From primeair at gmail.com Mon Apr 1 03:55:46 2013 From: primeair at gmail.com (Gotblade) Date: Sun, 31 Mar 2013 18:55:46 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg stalls when trying to transcode live content from HD PVR In-Reply-To: <1360941809099-4657324.post@n4.nabble.com> References: <1360941809099-4657324.post@n4.nabble.com> Message-ID: <1364781346214-4658189.post@n4.nabble.com> ffmpeg -i /dev/video0 -vcodec libx264 -preset ultrafast -qp 15 -acodec copy -y filename.mkv This is what worked for me. The -qp switch allows you to regulate the data stream so it's withing your hardware's capabilities. I started out with -qp 0 but it hung. Then I tried 1 & it went a lot further before hanging & I noticed a drop in the size in kb/s. 15 allowed me a complete capture at a high enough rate to have great quality video. Around 18000 kb/s. I notice I can scan these recordings much more cleanly as well. No after images as I jump forward or back! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-stalls-when-trying-to-transcode-live-content-from-HD-PVR-tp4657324p4658189.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From yugant at gmail.com Mon Apr 1 09:00:27 2013 From: yugant at gmail.com (Yugant Khanna) Date: Mon, 1 Apr 2013 12:30:27 +0530 Subject: [FFmpeg-user] Tee Muxer Message-ID: Hi , I am trying to capture the video in Segments and also want to stream the captured Video .So i tried using tee functionality for the same . I am getting an error "Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument" ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio= "Decklink Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec pcm_s24le -f segment -segment_time 10 -segment_list test.m3u8 -segment_format dvsd -f tee -map 0:0 -map 0:1 "D#%05d.mxf|[f=dvsd]rtp:// 127.0.0.1:1234/" ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 28 2013 21:52:57 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libg sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libo pencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-li bschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-lib twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enabl e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [dshow @ 0209c800] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, dshow, from 'video=Decklink Video Capture:audio=Decklink Audio Capture ': Duration: N/A, start: 10478.151000, bitrate: 1536 kb/s Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25 tbr, 10000k tbn, 25 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s [NULL @ 02e70020] Requested output format 'dvsd' is not a suitable output format Output #0, tee, to 'D#%05d.mxf|[f=dvsd]rtp://127.0.0.1:1234/': Metadata: encoder : Lavf55.0.100 Stream #0:0: Video: dvvideo, yuv411p, 720x576, q=2-31, 2000 kb/s, 90k tbn, 2 5 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> dvvideo) Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s24le) Could not write header for output file #0 (incorrect codec parameters ?): Invali d argument From nicolas.george at normalesup.org Mon Apr 1 10:36:06 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Mon, 1 Apr 2013 10:36:06 +0200 Subject: [FFmpeg-user] Tee Muxer In-Reply-To: References: Message-ID: <20130401083606.GA32217@phare.normalesup.org> Le duodi 12 germinal, an CCXXI, Yugant Khanna a ?crit?: > ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio= > "Decklink Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec > pcm_s24le -f segment -segment_time 10 -segment_list test.m3u8 > -segment_format dvsd -f tee -map 0:0 -map 0:1 "D#%05d.mxf|[f=dvsd]rtp:// > 127.0.0.1:1234/" You can not use the tee muxer like that: format options must go inside the segment pseudo-URL, just like you wrote "[f=dvsd]". That would mean something like that, with your options: -f tee '[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=dvsd]D#%05d.mxf|[f=dvsd]rtp://127.0.0.1:1234/' Also, I believe segment_format=dvsd does not exist. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From yugant at gmail.com Mon Apr 1 12:36:10 2013 From: yugant at gmail.com (Yugant Khanna) Date: Mon, 1 Apr 2013 16:06:10 +0530 Subject: [FFmpeg-user] Tee Muxer In-Reply-To: <20130401083606.GA32217@phare.normalesup.org> References: <20130401083606.GA32217@phare.normalesup.org> Message-ID: still i am missing some minor thingy ...I tried now i am getting Output file #0 does not contain any stream . ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio="Decklink Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec pcm_s24le -f tee "[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=dvsd] D#%05d.mxf | [f=mpegts] udp://127.0.0.1:12345/" ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 28 2013 21:52:57 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libg sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libo pencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-li bschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-lib twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enabl e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [dshow @ 022607e0] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, dshow, from 'video=Decklink Video Capture:audio=Decklink Audio Capture ': Duration: N/A, start: 23770.226000, bitrate: 1536 kb/s Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25 tbr, 10000k tbn, 25 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Output #0, tee, to '[f=segment:segment_time=10:segment_list=test.m3u8:segment_fo rmat=dvsd] D#%05d.mxf | [f=mpegts] udp://127.0.0.1:12345/': Output file #0 does not contain any stream On Mon, Apr 1, 2013 at 2:06 PM, Nicolas George < nicolas.george at normalesup.org> wrote: > Le duodi 12 germinal, an CCXXI, Yugant Khanna a ?crit : > > ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio= > > "Decklink Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec > > pcm_s24le -f segment -segment_time 10 -segment_list test.m3u8 > > -segment_format dvsd -f tee -map 0:0 -map 0:1 "D#%05d.mxf|[f=dvsd]rtp:// > > 127.0.0.1:1234/" > > You can not use the tee muxer like that: format options must go inside the > segment pseudo-URL, just like you wrote "[f=dvsd]". That would mean > something like that, with your options: > > -f tee > '[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=dvsd]D#%05d.mxf|[f=dvsd]rtp:// > 127.0.0.1:1234/' > > Also, I believe segment_format=dvsd does not exist. > > Regards, > > -- > Nicolas George > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.12 (GNU/Linux) > > iEYEARECAAYFAlFZRvYACgkQsGPZlzblTJOe7wCgvFBGlbBUI5yFlTyHq74ldLwi > +MUAn04ZQVhyWWxTcQwle5zQ7Txoi7jx > =ndxX > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From nicolas.george at normalesup.org Mon Apr 1 12:42:47 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Mon, 1 Apr 2013 12:42:47 +0200 Subject: [FFmpeg-user] Tee Muxer In-Reply-To: References: <20130401083606.GA32217@phare.normalesup.org> Message-ID: <20130401104247.GA32532@phare.normalesup.org> Le duodi 12 germinal, an CCXXI, Yugant Khanna a ?crit?: > still i am missing some minor thingy ...I tried now i am getting Output > file #0 does not contain any stream . > > ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio="Decklink > Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec pcm_s24le -f > tee "[f=segment:segment_time:segment_list=test.m3u8:segment_format=dvsd] > D#%05d.mxf | [f=mpegts] udp://127.0.0.1:12345/" You removed the -map options. Special muxers need the -map options because they can not guess what streams are wanted. > On Mon, Apr 1, 2013 at 2:06 PM, Nicolas George < > nicolas.george at normalesup.org> wrote: Please do not top-post on this mailing-list; if you do not know what it means, look it up. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From natgeo.operations at gmail.com Mon Apr 1 12:45:41 2013 From: natgeo.operations at gmail.com (Steve C) Date: Mon, 1 Apr 2013 06:45:41 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: has anyone had a chance to download and test this clip? https://dl.dropbox.com/u/1765157/NTSC_ProResHQ_DF_25sec.mov (~650 MB) On Sat, Mar 30, 2013 at 12:16 PM, Steve C wrote: > Carl, > > you can download a test file here: > https://dl.dropbox.com/u/1765157/NTSC_ProResHQ_DF_25sec.mov (~650 MB) > > Please let me know if you can re-produce the same error. > > thanks, > Steve > > > > On Fri, Mar 29, 2013 at 1:18 AM, Carl Eugen Hoyos wrote: > >> Steve C gmail.com> writes: >> >> > No, removing the -timecode option does not fix the issue, >> > i see get the same error. >> >> Please provide the input sample. >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From yugant at gmail.com Mon Apr 1 13:10:09 2013 From: yugant at gmail.com (Yugant Khanna) Date: Mon, 1 Apr 2013 16:40:09 +0530 Subject: [FFmpeg-user] Tee Muxer In-Reply-To: <20130401104247.GA32532@phare.normalesup.org> References: <20130401083606.GA32217@phare.normalesup.org> <20130401104247.GA32532@phare.normalesup.org> Message-ID: Added "-map 0:v -map 0:a" still that same error ffmpeg -f dshow -r 25 -i video="Decklink Video Capture":audio="Decklink Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M -acodec pcm_s24le-f tee -map 0:v -map 0:a "[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=dvsd] D#%05d.mxf | [f=mpegts] udp://127.0.0.1:12345/" ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 28 2013 21:52:57 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libg sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libo pencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-li bschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-lib twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enabl e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [dshow @ 02149880] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, dshow, from 'video=Decklink Video Capture:audio=Decklink Audio Capture ': Duration: N/A, start: 25972.844000, bitrate: 1536 kb/s Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25 tbr, 10000k tbn, 25 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s [segment @ 02eac040] Codec for stream 0 does not use global headers but containe r format requires global headers [segment @ 02eac040] Codec for stream 1 does not use global headers but containe r format requires global headers [tee @ 0214ec00] Slave '[f=mpegts] udp://127.0.0.1:12345/': error opening: Error number -22 occurred Output #0, tee, to '[f=segment:segment_time=10:segment_list=test.m3u8:segment_fo rmat=dvsd] D#%05d.mxf | [f=mpegts] udp://127.0.0.1:12345/': [dshow @ 02149880] real-time buffer 82% full! frame dropped! Metadata: encoder : Lavf55.0.100 Stream #0:0: Video: dvvideo, yuv411p, 720x576, q=2-31, 2000 kb/s, 90k tbn, 2 5 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> dvvideo) Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s24le) [dshow @ 02149880] real-time buffer 82% full! frame dropped! Could not write header for output file #0 (incorrect codec parameters ?): Invali d argument From nicolas.george at normalesup.org Mon Apr 1 13:50:58 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Mon, 1 Apr 2013 13:50:58 +0200 Subject: [FFmpeg-user] Tee Muxer In-Reply-To: References: <20130401083606.GA32217@phare.normalesup.org> <20130401104247.GA32532@phare.normalesup.org> Message-ID: <20130401115058.GA24171@phare.normalesup.org> Le duodi 12 germinal, an CCXXI, Yugant Khanna a ?crit?: > Added "-map 0:v -map 0:a" still that same error No, not the same error. Please try to read and understand the error messages. Your previous error message was: # Output file #0 does not contain any stream The error is now: > [segment @ 02eac040] Codec for stream 0 does not use global headers but > containe > r format requires global headers > [segment @ 02eac040] Codec for stream 1 does not use global headers but > containe > r format requires global headers This specific issue is addressed in the documentation: http://ffmpeg.org/ffmpeg-formats.html#tee Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From cehoyos at ag.or.at Mon Apr 1 16:58:06 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 1 Apr 2013 14:58:06 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg stalls when trying to transcode live content from HD PVR References: <1360941809099-4657324.post@n4.nabble.com> <1364781346214-4658189.post@n4.nabble.com> Message-ID: Gotblade gmail.com> writes: > ffmpeg -i /dev/video0 -vcodec libx264 -preset ultrafast > -qp 15 -acodec copy -y filename.mkv Complete, uncut console output missing. Is the problem only reproducible with an external library (libx264) or also if you are using "-vcodec mpeg4)? If it is only reproducible with -vcodec libx264, did you test the x264 executable if it also allows to reproduce the problem? Carl Eugen From cehoyos at ag.or.at Mon Apr 1 16:59:12 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 1 Apr 2013 14:59:12 +0000 (UTC) Subject: [FFmpeg-user] ffplay can not play a mp4 video References: <1364701556788-4658185.post@n4.nabble.com> <20130331084838.GE29565@leki> <1364743034314-4658188.post@n4.nabble.com> Message-ID: xxtina hotmail.com> writes: > How could I share the video that I can not play with you? Either upload to http://www.datafilehost.com/ or read https://ffmpeg.org/bugreports.html > Is this the appropriate way to contact for help? Yes, definitely. Carl Eugen From cehoyos at ag.or.at Mon Apr 1 17:13:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 1 Apr 2013 15:13:22 +0000 (UTC) Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 References: Message-ID: Steve C gmail.com> writes: > this error happens on quicktime 7, both mac and pc > platforms, if i try Quicktime X i can open and play > back the file. the file will also open and play back > with VLC. But most importantly, the file will not open > in Final Cut Pro. Do I understand correctly that in every case, the output file only fails to open with FCP? Since I don't have FCP, I can neither test if this is a regression nor how a possible work-around could look like. Carl Eugen From h.reindl at thelounge.net Mon Apr 1 17:21:55 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 01 Apr 2013 17:21:55 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: References: Message-ID: <5159A613.9010409@thelounge.net> am i really the only one receiving this randomly? i get a lot of mailing-lists and around 800 messages per day and this is the only list with such behavior surely the list-software is unable to distinct between a bounce and a reject at all, but as long this messages does not give any basic information WHICH message was "bounced" like date, time, subject. message-id it is impossible to do anything, even for me as admin of the mailserver and so i can only guess that some messages are rejected as spam from the MX -------- Original-Nachricht -------- Betreff: confirm a58e0237ee6751f7ce38287d432088783dcc782d Datum: Mon, 01 Apr 2013 17:13:49 +0200 Von: ffmpeg-user-request at ffmpeg.org An: h.reindl at thelounge.net Your membership in the mailing list ffmpeg-user has been disabled due to excessive bounces The last bounce received from you was dated 01-Apr-2013. You will not get any more messages from this list until you re-enable your membership. You will receive 3 more reminders like this before your membership in the list is deleted -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From andrey.aleksandrovich at googlemail.com Mon Apr 1 17:48:00 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Mon, 1 Apr 2013 18:48:00 +0300 Subject: [FFmpeg-user] Several audio tracks encoding issue Message-ID: Hello all, I have to make the film from 3 files: v.avi (huffyuv); a1.wav (pcm_s16le); a2.wav (pcm_s16le). I need that it will be mpeg4 and mp3 codecs in the final avi file. I have tried the command below, but have no luck :( localhost $ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 -map 1:0 -map 2:0 -c:v mpeg4 -qscale:v:0 3 -c:a:0 libmp3lame -qscale:a:0 4 final.avi -c:a:1 libmp3lame -qscale:a:1 4 -newaudio ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 12 2013 22:36:50 with gcc 4.5.4 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -march=native -pipe' --extra-cflags='-O2 -march=native -pipe' --extra-cxxflags='-O2 -march=native -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --enable-libmp3lame --enable-libvo-amrwbenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libaacplus --enable-nonfree --disable-indev=v4l --disable-indev=v4l2 --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-libfreetype --enable-libass --enable-pthreads --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-libopenjpeg --disable-altivec --disable-avx --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, avi, from 'v.avi': Duration: 01:27:36.54, start: 0.000000, bitrate: 49017 kb/s Stream #0:0: Video: huffyuv (HFYU / 0x55594648), yuv422p, 720x384, 23.98 tbr, 23.98 tbn, 23.98 tbc [wav @ 0x4847be48b0] max_analyze_duration 5000000 reached at 5013333 Input #1, wav, from 'a1.wav': Metadata: TCOD : 0 TCDO : 52565846458 Duration: 01:27:36.58, bitrate: 1536 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s [wav @ 0x4847cdd2d0] max_analyze_duration 5000000 reached at 5013333 Input #2, wav, from 'a2.wav': Duration: 01:27:36.51, bitrate: 1536 kb/s Stream #2:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s Missing argument for option 'newaudio' localhost $ What's wrong? ____ Gentoo Linux From hcoin at quietfountain.com Mon Apr 1 18:19:57 2013 From: hcoin at quietfountain.com (hcoin) Date: Mon, 01 Apr 2013 11:19:57 -0500 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? Message-ID: <5159B3AD.5050406@quietfountain.com> Hi all, thanks for ffmpeg! I'm struggling to drop two of three audio streams from an mpeg2 audio+video source, and can't puzzle out my mistake. I've given the following command on a mpeg2 file with one video and three ac3 audio streams. The source and all three streams play perfectly on vlc. I want the result to be exactly what was input, except with one audio stream: only the second of the three in the original. But, when I give the command the result is a file which when played in vlc shows around 56 or so audio streams, and otherwise produces no sound whatever, though the picture is correct. Am I making some newbie mistake? ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 -map 0.2 ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 -map 0.2 -ac 1 Both cases process normally, showing the entire and correct length, then exiting. Yet - no sound on playback. I tried it on freebsd 9.1 release, and I tried it on debian wheezy (0:0 and 0:2 instead of 0.0 and 0.2) -- same result. What have I overlooked? Thanks! Harry -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4261 bytes Desc: S/MIME Cryptographic Signature URL: From andrey.aleksandrovich at googlemail.com Mon Apr 1 18:45:06 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Mon, 1 Apr 2013 19:45:06 +0300 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? In-Reply-To: <5159B3AD.5050406@quietfountain.com> References: <5159B3AD.5050406@quietfountain.com> Message-ID: On 4/1/13, hcoin wrote: > Hi all, thanks for ffmpeg! > > I'm struggling to drop two of three audio streams from an mpeg2 > audio+video source, and can't puzzle out my mistake. > > I've given the following command on a mpeg2 file with one video and > three ac3 audio streams. The source and all three streams play > perfectly on vlc. I want the result to be exactly what was input, > except with one audio stream: only the second of the three in the > original. But, when I give the command the result is a file which when > played in vlc shows around 56 or so audio streams, and otherwise > produces no sound whatever, though the picture is correct. Am I making > some newbie mistake? > > ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 > -map 0.2 > > ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 > -map 0.2 -ac 1 > > Both cases process normally, showing the entire and correct length, then > exiting. Yet - no sound on playback. > > I tried it on freebsd 9.1 release, and I tried it on debian wheezy (0:0 > and 0:2 instead of 0.0 and 0.2) -- same result. What have I overlooked? > > Thanks! > > Harry Try this: ffmpeg -i INPUT.file -vcodec copy -acodec copy -map 0:0 -map 0:2 OUTPUT.file From hcoin at quietfountain.com Mon Apr 1 19:32:04 2013 From: hcoin at quietfountain.com (hcoin) Date: Mon, 01 Apr 2013 12:32:04 -0500 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? References: <5159B3AD.5050406@quietfountain.com> Message-ID: <5159C494.5080807@quietfountain.com> On 4/1/2013 11:45 AM, Andrey Aleksandrovich wrote: > On 4/1/13, hcoin wrote: >> Hi all, thanks for ffmpeg! >> >> I'm struggling to drop two of three audio streams from an mpeg2 >> audio+video source, and can't puzzle out my mistake. >> >> I've given the following command on a mpeg2 file with one video and >> three ac3 audio streams. The source and all three streams play >> perfectly on vlc. I want the result to be exactly what was input, >> except with one audio stream: only the second of the three in the >> original. But, when I give the command the result is a file which when >> played in vlc shows around 56 or so audio streams, and otherwise >> produces no sound whatever, though the picture is correct. Am I making >> some newbie mistake? >> >> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >> -map 0.2 >> >> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >> -map 0.2 -ac 1 >> >> Both cases process normally, showing the entire and correct length, then >> exiting. Yet - no sound on playback. >> >> I tried it on freebsd 9.1 release, and I tried it on debian wheezy (0:0 >> and 0:2 instead of 0.0 and 0.2) -- same result. What have I overlooked? >> >> Thanks! >> >> Harry > > Try this: > ffmpeg -i INPUT.file -vcodec copy -acodec copy -map 0:0 -map 0:2 OUTPUT.file > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Thanks for the quick response. The ffmpeg processing was normal. The video was normal. vlc reports 26 audio streams. No sound was produced. Same as before, basically, though the number of audio streams in the output file (which should be 1) is 26 instead of 50 something on my prior run. Actually I checked it again during the same vlc playback, somehow the number of audio streams has grown to 72 now. mplayer too produces no sound. Both vlc and mplayer work normally on the source mpg. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4261 bytes Desc: S/MIME Cryptographic Signature URL: From andrey.aleksandrovich at googlemail.com Mon Apr 1 19:42:02 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Mon, 1 Apr 2013 20:42:02 +0300 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? In-Reply-To: <5159C494.5080807@quietfountain.com> References: <5159B3AD.5050406@quietfountain.com> <5159C494.5080807@quietfountain.com> Message-ID: Meybe your source mpg is broken. Try to demux it first: $ projectx file.mpg -demux (projectx homepage - http://project-x.sourceforge.net/ ) and then to make your target file from 2 appropriate sources. ffmpeg -i video.file -i audio.file -vcodec copy -acodec copy output.mpg On 4/1/13, hcoin wrote: > On 4/1/2013 11:45 AM, Andrey Aleksandrovich wrote: >> On 4/1/13, hcoin wrote: >>> Hi all, thanks for ffmpeg! >>> >>> I'm struggling to drop two of three audio streams from an mpeg2 >>> audio+video source, and can't puzzle out my mistake. >>> >>> I've given the following command on a mpeg2 file with one video and >>> three ac3 audio streams. The source and all three streams play >>> perfectly on vlc. I want the result to be exactly what was input, >>> except with one audio stream: only the second of the three in the >>> original. But, when I give the command the result is a file which when >>> played in vlc shows around 56 or so audio streams, and otherwise >>> produces no sound whatever, though the picture is correct. Am I making >>> some newbie mistake? >>> >>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>> -map 0.2 >>> >>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>> -map 0.2 -ac 1 >>> >>> Both cases process normally, showing the entire and correct length, then >>> exiting. Yet - no sound on playback. >>> >>> I tried it on freebsd 9.1 release, and I tried it on debian wheezy (0:0 >>> and 0:2 instead of 0.0 and 0.2) -- same result. What have I >>> overlooked? >>> >>> Thanks! >>> >>> Harry >> >> Try this: >> ffmpeg -i INPUT.file -vcodec copy -acodec copy -map 0:0 -map 0:2 >> OUTPUT.file >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > Thanks for the quick response. The ffmpeg processing was normal. The > video was normal. vlc reports 26 audio streams. No sound was > produced. Same as before, basically, though the number of audio streams > in the output file (which should be 1) is 26 instead of 50 something on > my prior run. Actually I checked it again during the same vlc > playback, somehow the number of audio streams has grown to 72 now. > mplayer too produces no sound. Both vlc and mplayer work normally on > the source mpg. > > > > From natgeo.operations at gmail.com Mon Apr 1 20:00:10 2013 From: natgeo.operations at gmail.com (Steve C) Date: Mon, 1 Apr 2013 14:00:10 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: Carl, The error effects opening and playback with Quicktime 7, on both Mac and Windows platforms. Final Cut Pro works off of Quictime 7, so it makes sense that you cannot import the file into Final Cut. On Mon, Apr 1, 2013 at 11:13 AM, Carl Eugen Hoyos wrote: > Steve C gmail.com> writes: > > > this error happens on quicktime 7, both mac and pc > > platforms, if i try Quicktime X i can open and play > > back the file. the file will also open and play back > > with VLC. But most importantly, the file will not open > > in Final Cut Pro. > > Do I understand correctly that in every case, the output > file only fails to open with FCP? > Since I don't have FCP, I can neither test if this is a > regression nor how a possible work-around could look like. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From oliver at fromme.com Mon Apr 1 18:37:26 2013 From: oliver at fromme.com (Oliver Fromme) Date: Mon, 1 Apr 2013 18:37:26 +0200 (CEST) Subject: [FFmpeg-user] Correct fps value for h.264 stream Message-ID: <201304011637.r31GbQM3008151@haluter.fromme.com> Hello, This is probably a very special problem, but maybe someone has an answer ... I've got an .m4v file that contains a h.264 video stream. First thing I did is to convert it to .mkv because I can handle Matroska files much better. So far, so good. The container format (both .m4v and .mkv) reports a frame rate of 25 fps, but there is something wrong with the video stream: While mplayer plays the .mkv file fine, my media runs riot on it. It reports "-1 fps" (!), so I extracted the video stream from the container for examination: When I run "mplayer -identify" on it, it reports 180000 fps, and when I feed it into ffmpeg, it says 6203.12 fps. Something is definitely wrong with the stream. My first attempt was to re-encode the video stream with ffmpeg. This worked fine, and now the media player can play it, too. But this is not satisfactory because the re-encoding is not lossless. So I tried various options for ffmpeg with -vcodec copy, but it didn't work. No matter what I did, the h.264 stream didn't change (I checked with MD5) and the fps information was still wrong. I assume this is to be expected when using -vcodec copy, but, well, I had to try anyway. Just in case this information is useful, this is what I get without any options except "-vcodec copy": $ ffmpeg1 -i vstream-orig.h264 -vcodec copy vstream-new.h264 ffmpeg version 1.0.1 Copyright (c) 2000-2012 the FFmpeg developers built on Feb 16 2013 13:55:27 with gcc 4.2.1 (GCC) 20070831 patched [FreeBSD] configuration: --enable-x11grab --prefix=/usr/local --mandir=/usr/local/man --incdir=/usr/local/include/ffmpeg1 --libdir=/usr/local/lib/ffmpeg1 --shlibdir=/usr/local/lib/ffmpeg1 --datadir=/usr/local/share/ffmpeg1 --build-suffix=1 --progs-suffix=1 --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-pthreads --enable-memalign-hack --disable-libstagefright-h264 --disable-libutvideo --cc=cc --extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include' --extra-ldflags='-L/usr/local/lib ' --extra-libs=-pthread --disable-debug --disable-libaacplus --disable-indev=alsa --disable-outdev=alsa --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libass --disable-libcdio --disable-libcelt --enable-libfaac --disable-libfdk-aac --enable-fontconfig --enable-libfreetype --disable-frei0r --disable-gnutls --disable-libgsm --enable-libmp3lame --enable-libbluray --enable-libv4l2 --disable-libmodplug --disable-openal --enable-libopencv --enable-libo! penjpeg --disable-libpulse --disab libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [h264 @ 0x8094c1420] Stream #0: not enough frames to estimate rate; consider increasing probesize [h264 @ 0x8094c1420] Estimating duration from bitrate, this may be inaccurate Input #0, h264, from 'vstream-orig.h264': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (Main), yuv420p, 720x432 [SAR 64:45 DAR 64:27], 6203.12 fps, 90k tbr, 1200k tbn, 180k tbc Output #0, h264, to 'vstream-new.h264': Metadata: encoder : Lavf54.29.104 Stream #0:0: Video: h264, yuv420p, 720x432 [SAR 64:45 DAR 64:27], q=2-31, 6203.12 fps, 90k tbn, 90k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36023 < 36024 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36024 < 36025 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36025 < 36026 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36026 < 36027 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36027 < 36028 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36028 < 36029 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36029 < 36030 invalid, clipping [h264 @ 0x8094c1a20] st:0 PTS: -9223372036854775808 DTS: 36030 < 36031 invalid, clipping ... and so on ... Those last lines reapeat A LOT. I guess one line per frame. When I add "-r 25" for both input and output, the only thing that changes is this: Output #0, h264, to 'vstream-new.h264': Metadata: encoder : Lavf54.29.104 Stream #0:0: Video: h264, yuv420p, 720x432 [SAR 64:45 DAR 64:27], q=2-31, 6203.12 fps, 90k tbn, 25 tbc i.e. the tbc value changes from 90k to 25. I still get the "DTS invalid, clipping" lines, and it still says 6203.12 fps for both input and output. The next thing I tried was to add "-vsync drop". Now the "... clipping" lines were gone, but of course the resulting file still didn't change. I tried a lot of other options, but to no avail. So, now my question is: Is there a way to correct the wrong fps setting in the h.264 stream without re-encoding it, i.e. without loss? Please let me know if I have to provide more output or if you want me to try more commands. Unfortunately I cannot attach or upload the original file. Best regards Oliver -- From natgeo.operations at gmail.com Mon Apr 1 23:37:04 2013 From: natgeo.operations at gmail.com (Steve C) Date: Mon, 1 Apr 2013 17:37:04 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: I have been testing a workflow with FFMPEG that muxes one .MOV video file with 16 discreet mono WAV files. The resulting output file has one video stream and 16 audio streams. (please see full command line out put attached below) I have noticed the command statement that I'm use will correctly map all the streams, but it creates a warning : "Guessed Channel Layout for Input Stream" And this warning seems to cause FFMPEG to assign all 16 audio streams to the wrong channel position. "Channel positions : Front: C" - is used for all audio streams in the output file. See an example below. of the MediaInfo for for the ouput file. ID/String : 2 Format : PCM Format_Settings_Endianness : Little Format_Settings_Sign : Unsigned CodecID : in24 Duration/String : 1mn 10s BitRate_Mode/String : Constant BitRate/String : 1 152 Kbps Channel(s)/String : 1 channel ChannelPositions : Front: C SamplingRate/String : 48.0 KHz BitDepth/String : 24 bits StreamSize/String : 9.62 MiB (1%) Language/String : English I'm trying to figure out what how to correctly assign the streams with FFMPEG? I would like the channel assignment to match the 5.1+2 (7.1) arrangement. FL+FR+FC+LFE+BL+BR+SL+SR I've searched through the FFMPEG documentation, but I can't find out a way to do this. thanks, steve #################################################### ffmpeg started on 2013-04-01 at 14:55:34 Report written to "ffmpeg-20130401-145534.log" Command line: "C:\\ffmpeg.exe" -i "C:\\Test\\NTSC_ProResHQ_NA.mov" -i "C:\\Test\\01_5.1.FM.L.wav" -i "C:\\Test\\02_5.1.FM.R.wav" -i "C:\\Test\\03_5.1.FM.C.wav" -i "C:\\Test\\04_5.1.FM.LFE.wav" -i "C:\\Test\\05_5.1.FM.Ls.wav" -i "C:\\Test\\06_5.1.FM.Rs.wav" -i "C:\\Test\\07_Stereo.FM.L.wav" -i "C:\\Test\\08_Stereo.FM.R.wav" -i "C:\\Test\\09_5.1.MDE.L.wav" -i "C:\\Test\\10_5.1.MDE.R.wav" -i "C:\\Test\\11_5.1.MDE.C.wav" -i "C:\\Test\\12_5.1.MDE.LFE.wav" -i "C:\\Test\\13_5.1.MDE.Ls.wav" -i "C:\\Test\\14_5.1.MDE.Rs.wav" -i "C:\\Test\\15_Stereo.MDE.L.wav" -i "C:\\Test\\16_Stereo.MDE.R.wav" -c:v copy -c:a copy -timecode "09:59:50;00" -r 29.97 -report -map 0:0 -map 1:0 -map 2:0 -map 3:0 -map 4:0 -map 5:0 -map 6:0 -map 7:0 -map 8:0 -map 9:0 -map 10:0 -map 11:0 -map 12:0 -map 13:0 -map 14:0 -map 15:0 -map 16:0 -report "C:\\output-3.mov" ffmpeg version N-51338-g914f48d Copyright (c) 2000-2013 the FFmpeg developers built on Mar 27 2013 01:04:22 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 1.100 / 55. 1.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Splitting the commandline. Reading option '-i' ... matched as input file with argument 'C:\Test\NTSC_ProResHQ_NA.mov'. Reading option '-i' ... matched as input file with argument 'C:\Test\01_5.1.FM.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\02_5.1.FM.R.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\03_5.1.FM.C.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\04_5.1.FM.LFE.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\05_5.1.FM.Ls.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\06_5.1.FM.Rs.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\07_Stereo.FM.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\08_Stereo.FM.R.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\09_5.1.MDE.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\10_5.1.MDE.R.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\11_5.1.MDE.C.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\12_5.1.MDE.LFE.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\13_5.1.MDE.Ls.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\14_5.1.MDE.Rs.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\15_Stereo.MDE.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\16_Stereo.MDE.R.wav'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-timecode' ... matched as option 'timecode' (set initial TimeCode value.) with argument '09:59:50;00'. Reading option '-r' ... matched as option 'r' (set frame rate (Hz value, fraction or abbreviation)) with argument '29.97'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '1:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '2:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '3:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '4:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '5:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '6:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '7:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '8:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '9:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '10:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '11:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '12:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '13:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '14:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '15:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '16:0'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option 'C:\output-3.mov' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file C:\Test\NTSC_ProResHQ_NA.mov. Successfully parsed a group of options. Opening an input file: C:\Test\NTSC_ProResHQ_NA.mov. [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] ISO: File Type Major Brand: qt [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] File position before avformat_find_stream_info() is 19318 [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] Ignoring outlier non primary stream starttime -80.013344 [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] Ignoring outlier non primary stream starttime -80.013344 [mov,mp4,m4a,3gp,3g2,mj2 @ 03644860] File position after avformat_find_stream_info() is 889286 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Test\NTSC_ProResHQ_NA.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-03-20 16:19:14 Duration: 00:01:10.07, start: 0.000000, bitrate: 207771 kb/s Stream #0:0(eng), 1, 1/2997: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 207768 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Stream #0:1(eng), 1, 1/2997: Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Successfully opened the file. Parsing a group of options: input file C:\Test\01_5.1.FM.L.wav. Successfully parsed a group of options. Opening an input file: C:\Test\01_5.1.FM.L.wav. [wav @ 036ccfa0] Format wav probed with size=2048 and score=99 [wav @ 036ccfa0] File position before avformat_find_stream_info() is 1024 [wav @ 036ccfa0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036ccfa0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036ccfa0] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 036ccfa0] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'C:\Test\01_5.1.FM.L.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdI18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #1:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\02_5.1.FM.R.wav. Successfully parsed a group of options. Opening an input file: C:\Test\02_5.1.FM.R.wav. [wav @ 036bfb60] Format wav probed with size=2048 and score=99 [wav @ 036bfb60] File position before avformat_find_stream_info() is 1024 [wav @ 036bfb60] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036bfb60] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036bfb60] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 036bfb60] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #2.0 : mono Input #2, wav, from 'C:\Test\02_5.1.FM.R.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdP18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #2:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\03_5.1.FM.C.wav. Successfully parsed a group of options. Opening an input file: C:\Test\03_5.1.FM.C.wav. [wav @ 03eb2060] Format wav probed with size=2048 and score=99 [wav @ 03eb2060] File position before avformat_find_stream_info() is 1024 [wav @ 03eb2060] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03eb2060] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03eb2060] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 03eb2060] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #3.0 : mono Input #3, wav, from 'C:\Test\03_5.1.FM.C.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdM18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #3:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\04_5.1.FM.LFE.wav. Successfully parsed a group of options. Opening an input file: C:\Test\04_5.1.FM.LFE.wav. [wav @ 03f3f580] Format wav probed with size=2048 and score=99 [wav @ 03f3f580] File position before avformat_find_stream_info() is 1024 [wav @ 03f3f580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03f3f580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03f3f580] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 03f3f580] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #4.0 : mono Input #4, wav, from 'C:\Test\04_5.1.FM.LFE.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdW18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #4:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\05_5.1.FM.Ls.wav. Successfully parsed a group of options. Opening an input file: C:\Test\05_5.1.FM.Ls.wav. [wav @ 0409ba20] Format wav probed with size=2048 and score=99 [wav @ 0409ba20] File position before avformat_find_stream_info() is 1024 [wav @ 0409ba20] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0409ba20] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0409ba20] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 0409ba20] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #5.0 : mono Input #5, wav, from 'C:\Test\05_5.1.FM.Ls.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdS18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #5:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\06_5.1.FM.Rs.wav. Successfully parsed a group of options. Opening an input file: C:\Test\06_5.1.FM.Rs.wav. [wav @ 04016760] Format wav probed with size=2048 and score=99 [wav @ 04016760] File position before avformat_find_stream_info() is 1024 [wav @ 04016760] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04016760] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04016760] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 04016760] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #6.0 : mono Input #6, wav, from 'C:\Test\06_5.1.FM.Rs.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdT18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #6:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\07_Stereo.FM.L.wav. Successfully parsed a group of options. Opening an input file: C:\Test\07_Stereo.FM.L.wav. [wav @ 036b1da0] Format wav probed with size=2048 and score=99 [wav @ 036b1da0] File position before avformat_find_stream_info() is 1024 [wav @ 036b1da0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036b1da0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 036b1da0] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 036b1da0] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #7.0 : mono Input #7, wav, from 'C:\Test\07_Stereo.FM.L.wav': Metadata: encoded_by : Pro Tools originator_reference: bBAc908u07maaaGk date : 2012-12-10 creation_time : 20:59:48 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #7:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\08_Stereo.FM.R.wav. Successfully parsed a group of options. Opening an input file: C:\Test\08_Stereo.FM.R.wav. [wav @ 04170440] Format wav probed with size=2048 and score=99 [wav @ 04170440] File position before avformat_find_stream_info() is 1024 [wav @ 04170440] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04170440] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04170440] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 04170440] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #8.0 : mono Input #8, wav, from 'C:\Test\08_Stereo.FM.R.wav': Metadata: encoded_by : Pro Tools originator_reference: bBAcc18u07maaaGk date : 2012-12-10 creation_time : 20:59:48 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #8:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\09_5.1.MDE.L.wav. Successfully parsed a group of options. Opening an input file: C:\Test\09_5.1.MDE.L.wav. [wav @ 04099320] Format wav probed with size=2048 and score=99 [wav @ 04099320] File position before avformat_find_stream_info() is 1024 [wav @ 04099320] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04099320] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 04099320] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 04099320] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #9.0 : mono [TRUNCATED] Parsing a group of options: output file C:\output-3.mov. Applying option c:v (codec name) with argument copy. Applying option c:a (codec name) with argument copy. Applying option timecode (set initial TimeCode value.) with argument 09:59:50;00. Applying option r (set frame rate (Hz value, fraction or abbreviation)) with argument 29.97. Applying option map (set input stream mapping) with argument 0:0. Applying option map (set input stream mapping) with argument 1:0. Applying option map (set input stream mapping) with argument 2:0. Applying option map (set input stream mapping) with argument 3:0. Applying option map (set input stream mapping) with argument 4:0. Applying option map (set input stream mapping) with argument 5:0. Applying option map (set input stream mapping) with argument 6:0. Applying option map (set input stream mapping) with argument 7:0. Applying option map (set input stream mapping) with argument 8:0. Applying option map (set input stream mapping) with argument 9:0. Applying option map (set input stream mapping) with argument 10:0. Applying option map (set input stream mapping) with argument 11:0. Applying option map (set input stream mapping) with argument 12:0. Applying option map (set input stream mapping) with argument 13:0. Applying option map (set input stream mapping) with argument 14:0. Applying option map (set input stream mapping) with argument 15:0. Applying option map (set input stream mapping) with argument 16:0. Successfully parsed a group of options. Opening an output file: C:\output-3.mov. Codec AVOption gop_timecode (MPEG GOP Timecode in hh:mm:ss[:;.]ff format) specified for output file #0 (C:\output-3.mov) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream. Successfully opened the file. Output #0, mov, to 'C:\output-3.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt timecode : 09:59:50;00 encoder : Lavf55.0.100 Stream #0:0(eng), 0, 1/11988: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 207768 kb/s, 29.97 fps, 11988 tbn, 29.97 tbc Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Stream #0:1, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:2, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:3, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:4, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:5, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:6, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:7, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:8, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:9, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:10, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:11, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:12, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:13, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:14, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:15, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream #0:16, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, 1152 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #1:0 -> #0:1 (copy) Stream #2:0 -> #0:2 (copy) Stream #3:0 -> #0:3 (copy) Stream #4:0 -> #0:4 (copy) Stream #5:0 -> #0:5 (copy) Stream #6:0 -> #0:6 (copy) Stream #7:0 -> #0:7 (copy) Stream #8:0 -> #0:8 (copy) Stream #9:0 -> #0:9 (copy) Stream #10:0 -> #0:10 (copy) Stream #11:0 -> #0:11 (copy) Stream #12:0 -> #0:12 (copy) Stream #13:0 -> #0:13 (copy) Stream #14:0 -> #0:14 (copy) Stream #15:0 -> #0:15 (copy) Stream #16:0 -> #0:16 (copy) Press [q] to stop, [?] for help frame= 26 fps=0.0 q=-1.0 size= 24071kB time=00:00:05.23 bitrate=37684.9kbits/s frame= 29 fps= 29 q=-1.0 size= 26757kB time=00:00:09.78 bitrate=22406.9kbits/s frame= 38 fps= 25 q=-1.0 size= 34941kB time=00:00:13.42 bitrate=21325.0kbits/s frame= 43 fps= 21 q=-1.0 size= 39502kB time=00:00:14.33 bitrate=22577.9kbits/s [TRUNCATED] frame= 2092 fps= 15 q=-1.0 size= 1926358kB time=00:01:09.80 bitrate=226074.7kbits/s frame= 2099 fps= 15 q=-1.0 size= 1933695kB time=00:01:10.03 bitrate=226179.0kbits/s No more output streams to write to, finishing. frame= 2100 fps= 15 q=-1.0 Lsize= 1934985kB time=00:01:10.07 bitrate=226222.1kbits/s video:1777147kB audio:157658kB subtitle:0 global headers:0kB muxing overhead 0.009320% [AVIOContext @ 0246b9e0] Statistics: 1819896788 bytes read, 4 seeks [AVIOContext @ 03631800] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 036c0080] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 0371ff40] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 03eb2780] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 03639500] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 04016c80] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 036b4a00] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 036b24e0] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 036c0980] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 03f402c0] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 04099840] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 043c80a0] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 043c8160] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 0429db00] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 0429dc20] Statistics: 10158082 bytes read, 4 seeks [AVIOContext @ 04abf780] Statistics: 10158082 bytes read, 4 seeks From 550399108 at qq.com Tue Apr 2 04:15:31 2013 From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=) Date: Tue, 2 Apr 2013 10:15:31 +0800 Subject: [FFmpeg-user] How to control the bitrate between a min and max in VBR encoding Message-ID: Hello All, Now I use ffmpeg 2-pass encoding to transcoding my video to flv format. How can I make the bitrate under control? Which version of ffmpeg can be useful? From nlewis at mail.crawford.com Tue Apr 2 04:55:16 2013 From: nlewis at mail.crawford.com (Nathan Lewis) Date: Mon, 1 Apr 2013 22:55:16 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: On Mon, Apr 1, 2013 at 2:00 PM, Steve C wrote: > Carl, > > The error effects opening and playback with Quicktime 7, on both Mac and > Windows platforms. > Final Cut Pro works off of Quictime 7, so it makes sense that you cannot > import the file into Final Cut. > > Hi Steve, I just downloaded the file from the dropbox link provided and I am able to open and playback the file in both FCP and QT player on Mac OS 10.8.3 . Both report the file to be MOS (no audio tracks). Details: QT Player: version 7.6.6 (1709) QT Version 7.7.1 (2599.24) (Note: I'm not sure what to make of the discrepancy between version and QT version) Final Cut Pro: Version 7.0.3 I've lost some of the history of the thread. So, please let me know if I am missing something important or need to do any additional testing. Regards, Nathan From nlewis at mail.crawford.com Tue Apr 2 05:14:08 2013 From: nlewis at mail.crawford.com (Nathan Lewis) Date: Mon, 1 Apr 2013 23:14:08 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: On Mon, Apr 1, 2013 at 10:55 PM, Nathan Lewis wrote: > On Mon, Apr 1, 2013 at 2:00 PM, Steve C wrote: > >> Carl, >> >> The error effects opening and playback with Quicktime 7, on both Mac and >> Windows platforms. >> Final Cut Pro works off of Quictime 7, so it makes sense that you cannot >> import the file into Final Cut. >> >> > Hi Steve, > > I just downloaded the file from the dropbox link provided and I am able to > open and playback the file in both FCP and QT player on Mac OS 10.8.3 . > Both report the file to be MOS (no audio tracks). Details: > > QT Player: version 7.6.6 (1709) QT Version 7.7.1 (2599.24) (Note: I'm > not sure what to make of the discrepancy between version and QT version) > > Final Cut Pro: Version 7.0.3 > > I've lost some of the history of the thread. So, please let me know if I > am missing something important or need to do any additional testing. > > Regards, > > Nathan > > I need to do some research to figure out what I'm missing from the history of the thread, but trying to do a "-c:v copy" results in the same error and QT/FCP problems you've run into. Note that I'm not running the current git head version. $ ffmpeg -y -i NTSC_ProResHQ_DF_25sec.mov -c:v copy ~/Documents/natgeo_test.mov ffmpeg version N-49805-gfd6a021 Copyright (c) 2000-2013 the FFmpeg developers built on Feb 10 2013 22:16:33 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-nonfree --enable-gpl --enable-pthreads --enable-libx264 --enable-version3 libavutil 52. 17.101 / 52. 17.101 libavcodec 54. 91.102 / 54. 91.102 libavformat 54. 62.100 / 54. 62.100 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 37.101 / 3. 37.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'NTSC_ProResHQ_DF_25sec.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-03-30 15:58:59 Duration: 00:00:25.06, start: 0.000000, bitrate: 208519 kb/s Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 208485 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2013-03-30 15:58:59 handler_name : Apple Alias Data Handler timecode : 09:59:50;00 Stream #0:1(eng): Data: none (tmcd / 0x64636D74), 0 kb/s Metadata: creation_time : 2013-03-30 15:59:35 handler_name : Apple Alias Data Handler timecode : 09:59:50;00 Output #0, mov, to '/Users/nel/Documents/natgeo_test.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf54.62.100 Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 208485 kb/s, 29.97 fps, 11988 tbn, 2997 tbc Metadata: creation_time : 2013-03-30 15:58:59 handler_name : Apple Alias Data Handler timecode : 09:59:50;00 Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help fps 2997 is too large=-1.0 size= 635915kB time=00:00:24.95 bitrate=208722.1kbits/s frame= 751 fps= 50 q=-1.0 Lsize= 637740kB time=00:00:25.02 bitrate=208763.0kbits/s video:637733kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.001105% From hcoin at quietfountain.com Tue Apr 2 05:32:27 2013 From: hcoin at quietfountain.com (hcoin) Date: Mon, 01 Apr 2013 22:32:27 -0500 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? References: <5159B3AD.5050406@quietfountain.com> <5159C494.5080807@quietfountain.com> Message-ID: <515A514B.5070808@quietfountain.com> On 4/1/2013 12:42 PM, Andrey Aleksandrovich wrote: > Meybe your source mpg is broken. Try to demux it first: > $ projectx file.mpg -demux > (projectx homepage - http://project-x.sourceforge.net/ ) > and then to make your target file from 2 appropriate sources. > ffmpeg -i video.file -i audio.file -vcodec copy -acodec copy output.mpg > > On 4/1/13, hcoin wrote: >> On 4/1/2013 11:45 AM, Andrey Aleksandrovich wrote: >>> On 4/1/13, hcoin wrote: >>>> Hi all, thanks for ffmpeg! >>>> >>>> I'm struggling to drop two of three audio streams from an mpeg2 >>>> audio+video source, and can't puzzle out my mistake. >>>> >>>> I've given the following command on a mpeg2 file with one video and >>>> three ac3 audio streams. The source and all three streams play >>>> perfectly on vlc. I want the result to be exactly what was input, >>>> except with one audio stream: only the second of the three in the >>>> original. But, when I give the command the result is a file which when >>>> played in vlc shows around 56 or so audio streams, and otherwise >>>> produces no sound whatever, though the picture is correct. Am I making >>>> some newbie mistake? >>>> >>>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>>> -map 0.2 >>>> >>>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>>> -map 0.2 -ac 1 >>>> >>>> Both cases process normally, showing the entire and correct length, then >>>> exiting. Yet - no sound on playback. >>>> >>>> I tried it on freebsd 9.1 release, and I tried it on debian wheezy (0:0 >>>> and 0:2 instead of 0.0 and 0.2) -- same result. What have I >>>> overlooked? >>>> >>>> Thanks! >>>> >>>> Harry >>> Try this: >>> ffmpeg -i INPUT.file -vcodec copy -acodec copy -map 0:0 -map 0:2 >>> OUTPUT.file >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> Thanks for the quick response. The ffmpeg processing was normal. The >> video was normal. vlc reports 26 audio streams. No sound was >> produced. Same as before, basically, though the number of audio streams >> in the output file (which should be 1) is 26 instead of 50 something on >> my prior run. Actually I checked it again during the same vlc >> playback, somehow the number of audio streams has grown to 72 now. >> mplayer too produces no sound. Both vlc and mplayer work normally on >> the source mpg. >> >> >> >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Well, some more detail and an interesting development. As I wrote, both vlc and mplayer play the whole original thing perfectly allowing any audio stream to be chosen. In my world that means 'no file corruption'. Project-x completed and managed to produce audio and video files that left out a huge chunk of the middle of the file. But, the 'demux' idea worked with ffmpeg. It split the whole file into a video only mpg, and one ac3 stream. Both the broken out files played perfectly. Here's the interesting bit. When I gave this command: ffmpeg -i video.mpg -i audio.ac3 -acodec copy -vcodec copy whole_show.mpg The result was identical to my initial failed runs: No audio produced by the combined result, and vlc shows 56 audio streams. So, either I've missed something obvious, or ffmpeg can't handle muxing together an mpeg2 file and an ac3 audio stream. Or? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4261 bytes Desc: S/MIME Cryptographic Signature URL: From hcoin at quietfountain.com Tue Apr 2 06:51:57 2013 From: hcoin at quietfountain.com (hcoin) Date: Mon, 01 Apr 2013 23:51:57 -0500 Subject: [FFmpeg-user] Dropping audio streams confuses vlc? References: <5159B3AD.5050406@quietfountain.com> <5159C494.5080807@quietfountain.com> <515A514B.5070808@quietfountain.com> Message-ID: <515A63ED.2060800@quietfountain.com> On 4/1/2013 10:32 PM, hcoin wrote: > On 4/1/2013 12:42 PM, Andrey Aleksandrovich wrote: >> Meybe your source mpg is broken. Try to demux it first: >> $ projectx file.mpg -demux >> (projectx homepage - http://project-x.sourceforge.net/ ) >> and then to make your target file from 2 appropriate sources. >> ffmpeg -i video.file -i audio.file -vcodec copy -acodec copy output.mpg >> >> On 4/1/13, hcoin wrote: >>> On 4/1/2013 11:45 AM, Andrey Aleksandrovich wrote: >>>> On 4/1/13, hcoin wrote: >>>>> Hi all, thanks for ffmpeg! >>>>> >>>>> I'm struggling to drop two of three audio streams from an mpeg2 >>>>> audio+video source, and can't puzzle out my mistake. >>>>> >>>>> I've given the following command on a mpeg2 file with one video and >>>>> three ac3 audio streams. The source and all three streams play >>>>> perfectly on vlc. I want the result to be exactly what was input, >>>>> except with one audio stream: only the second of the three in the >>>>> original. But, when I give the command the result is a file >>>>> which when >>>>> played in vlc shows around 56 or so audio streams, and otherwise >>>>> produces no sound whatever, though the picture is correct. Am I >>>>> making >>>>> some newbie mistake? >>>>> >>>>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>>>> -map 0.2 >>>>> >>>>> ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy outvid.mpg -map 0.0 >>>>> -map 0.2 -ac 1 >>>>> >>>>> Both cases process normally, showing the entire and correct >>>>> length, then >>>>> exiting. Yet - no sound on playback. >>>>> >>>>> I tried it on freebsd 9.1 release, and I tried it on debian wheezy >>>>> (0:0 >>>>> and 0:2 instead of 0.0 and 0.2) -- same result. What have I >>>>> overlooked? >>>>> >>>>> Thanks! >>>>> >>>>> Harry >>>> Try this: >>>> ffmpeg -i INPUT.file -vcodec copy -acodec copy -map 0:0 -map 0:2 >>>> OUTPUT.file >>>> _______________________________________________ >>>> ffmpeg-user mailing list >>>> ffmpeg-user at ffmpeg.org >>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>> Thanks for the quick response. The ffmpeg processing was normal. The >>> video was normal. vlc reports 26 audio streams. No sound was >>> produced. Same as before, basically, though the number of audio >>> streams >>> in the output file (which should be 1) is 26 instead of 50 something on >>> my prior run. Actually I checked it again during the same vlc >>> playback, somehow the number of audio streams has grown to 72 now. >>> mplayer too produces no sound. Both vlc and mplayer work normally on >>> the source mpg. >>> >>> >>> >>> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > Well, some more detail and an interesting development. > > As I wrote, both vlc and mplayer play the whole original thing > perfectly allowing any audio stream to be chosen. In my world that > means 'no file corruption'. Project-x completed and managed to > produce audio and video files that left out a huge chunk of the middle > of the file. But, the 'demux' idea worked with ffmpeg. It split the > whole file into a video only mpg, and one ac3 stream. Both the broken > out files played perfectly. > > Here's the interesting bit. When I gave this command: > > ffmpeg -i video.mpg -i audio.ac3 -acodec copy -vcodec copy whole_show.mpg > > The result was identical to my initial failed runs: No audio produced > by the combined result, and vlc shows 56 audio streams. > > So, either I've missed something obvious, or ffmpeg can't handle > muxing together an mpeg2 file and an ac3 audio stream. > > Or? > > > > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Replying to my own post: While the command: ffmpeg -i video.mpg -i audio.ac3 -acodec copy -vcodec copy whole_show.mpg fails as noted above, ffmpeg -i video.mpg -i audio.ac3 -acodec copy -vcodec copy whole_show.mp4 works. Would those who know how alert whoever it is that fixes ffmpeg bugs? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4261 bytes Desc: S/MIME Cryptographic Signature URL: From dave at dericed.com Tue Apr 2 07:17:32 2013 From: dave at dericed.com (Dave Rice) Date: Tue, 2 Apr 2013 01:17:32 -0400 Subject: [FFmpeg-user] using ebur128 with multiple streams of audio Message-ID: <0112308C-FD1F-4E33-9277-76C04130F11C@dericed.com> Hi ffmpeg-user, I have many files with multiple audio streams and would like to analyze all audio streams with ebur128 in a single pass. I can accomplish this with -map, but the output of ebur128 doesn't provide the stream index in the default output. Here's how to reproduce. Make a file with multiple audio streams. Example: ffmpeg -f lavfi -i aevalsrc="sin(440*2*PI*t)::s=8000" -f lavfi -i aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" -f lavfi -i aevalsrc="-2+random(0)" -map 0 -map 1 -map 2 -c:a pcm_s16le -t 1 -y 3_tracks.mov Then analyze with ebur128 like this: ffmpeg -i 3_tracks.mov -map 0:a -af ebur128 -f null - And the output is: ffmpeg version 1.1.git Copyright (c) 2000-2013 the FFmpeg developers built on Mar 31 2013 19:26:29 with Apple LLVM version 4.2 (clang-425.0.27) (based on LLVM 3.2svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-ffplay --enable-libopenjpeg --extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.1/include/openjpeg-1.5 ' libavutil 52. 24.100 / 52. 24.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '3_tracks.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt encoder : Lavf55.1.100 Duration: 00:00:01.02, start: 0.000000, bitrate: 1552 kb/s Stream #0:0(eng): Audio: pcm_s16le (sowt / 0x74776F73), 8000 Hz, mono, s16, 128 kb/s Metadata: handler_name : DataHandler Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 44100 Hz, mono, s16, 705 kb/s Metadata: handler_name : DataHandler Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 44100 Hz, mono, s16, 705 kb/s Metadata: handler_name : DataHandler Output #0, null, to 'pipe:': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt encoder : Lavf55.1.100 Stream #0:0(eng): Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: handler_name : DataHandler Stream #0:1(eng): Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: handler_name : DataHandler Stream #0:2(eng): Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s Metadata: handler_name : DataHandler Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le) Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s16le) Stream #0:2 -> #0:2 (pcm_s16le -> pcm_s16le) Press [q] to stop, [?] for help [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.0999792 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.0999792 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.0999792 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.199979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.199979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.199979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.299979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.299979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.299979 M:-120.7 S:-120.7 I: -70.0 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.399979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.499979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.399979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.399979 M: -26.3 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.499979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.499979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.599979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.599979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.599979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.699979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.699979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.699979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.799979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.799979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.799979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.899979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.899979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.899979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.999979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c15640] t: 0.999979 M: -6.3 S:-120.7 I: -6.3 LUFS LRA: 0.0 LU [Parsed_ebur128_0 @ 0x7fc8b8c16500] t: 0.999979 M: -86.7 S:-120.7 I: -26.3 LUFS LRA: 0.0 LU size=N/A time=00:00:01.02 bitrate=N/A video:0kB audio:288kB subtitle:0 global headers:0kB muxing overhead -100.007471% Summary: Integrated loudness: I: -3.7 LUFS Threshold: -13.7 LUFS Loudness range: LRA: 0.0 LU Threshold: 0.0 LUFS LRA low: 0.0 LUFS LRA high: 0.0 LUFS [Parsed_ebur128_0 @ 0x7fc8b8c15640] Summary: Integrated loudness: I: -6.3 LUFS Threshold: -16.3 LUFS Loudness range: LRA: 0.0 LU Threshold: 0.0 LUFS LRA low: 0.0 LUFS LRA high: 0.0 LUFS [Parsed_ebur128_0 @ 0x7fc8b8c16500] Summary: Integrated loudness: I: -26.3 LUFS Threshold: -36.3 LUFS Loudness range: LRA: 0.0 LU Threshold: 0.0 LUFS LRA low: 0.0 LUFS LRA high: 0.0 LUFS The loudness info is presented for all tracks but none of the data is labeled with a stream index, so this is a parsing challenge. What I'm hoping to derive is something like: stream=0:0 I=-3.7 stream=0:1 I=-6.3 stream=0:2 I=-26.3 Is there a better way to get there besides counting the input audio streams and then count the occurrences of the integrated loudness summaries and matching them up. Similary I'd also like to plot the loudness data per stream but the frame reporting lines such as [Parsed_ebur128_0 @ 0x7fc8b8c11ac0] t: 0.699979 M: -3.7 S:-120.7 I: -3.7 LUFS LRA: 0.0 LU do not report the stream index either. Is there an existing way to add stream index data to the output of ebur128? Should I file an enhancement request? Dave Rice From cehoyos at ag.or.at Tue Apr 2 07:37:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 05:37:55 +0000 (UTC) Subject: [FFmpeg-user] How to control the bitrate between a min and max in VBR encoding References: Message-ID: ??? <550399108 qq.com> writes: > How can I make the bitrate under control? -vb allows to control the bitrate. > Which version of ffmpeg can be useful? On this mailing list, only current git head is supported. Carl Eugen From cehoyos at ag.or.at Tue Apr 2 07:40:27 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 05:40:27 +0000 (UTC) Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) References: Message-ID: Steve C gmail.com> writes: > I would like the channel assignment to match > the 5.1+2 (7.1) arrangement. How are 16 channels supposed to match any eight channel channel layout? > I've searched through the FFMPEG documentation, but > I can't find out a way to do this. ffmpeg -h full|grep channel_layout shows a few results here. Carl Eugen From cehoyos at ag.or.at Tue Apr 2 07:41:42 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 05:41:42 +0000 (UTC) Subject: [FFmpeg-user] Dropping audio streams confuses vlc? References: <5159B3AD.5050406@quietfountain.com> Message-ID: hcoin quietfountain.com> writes: > ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy > outvid.mpg -map 0.0 -map 0.2 > > ffmpeg -i sourcevid.mpg -vcodec copy -acodec copy > outvid.mpg -map 0.0 -map 0.2 -ac 1 > > Both cases process normally, showing the entire and correct > length, then exiting. Yet - no sound on playback. Please provide the complete, uncut console output, I suspect a sample will also be needed. Carl Eugen From 550399108 at qq.com Tue Apr 2 08:01:29 2013 From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=) Date: Tue, 2 Apr 2013 14:01:29 +0800 Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ICBIb3cgdG8gY29udHJvbCB0aGUg?= =?gb18030?q?bitrate_between_a_min_and_maxin_VBR_encoding?= Message-ID: Thank you for answer first. Now I use command below: ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 1 -vcodec libx264 -vpre fastfirstpass -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -an -f flv -map_chapters -1:-1 -g 50 /dev/null ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 2 -vcodec libx264 -vpre normal -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -acodec copy -f flv -map_chapters -1:-1 -g 50 output.flv The output.flv is a video with VBR,The part of maxbitrate is bigger than 1024k ,I want to control the bitrate between 1500k because of my poor boundwidth. I searchd more on google .There are some options can be used.Such as -minrate -maxrate -bufsize. I used them,but the quality of the output is not satisfactory. Someone said that this command only effective in CBR mode. Is there any other options can make me control the bitrate? ------------------ ???? ------------------ ???: "Carl Eugen Hoyos"; ????: 2013?4?2?(???) ??1:37 ???: "ffmpeg-user"; ??: Re: [FFmpeg-user] How to control the bitrate between a min and maxin VBR encoding ??? <550399108 qq.com> writes: > How can I make the bitrate under control? -vb allows to control the bitrate. > Which version of ffmpeg can be useful? On this mailing list, only current git head is supported. Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From michael.rampe at gmail.com Tue Apr 2 09:51:25 2013 From: michael.rampe at gmail.com (FFmichael) Date: Tue, 2 Apr 2013 00:51:25 -0700 (PDT) Subject: [FFmpeg-user] Determine q factor from encoded video [H.264] Message-ID: <1364889085665-4658219.post@n4.nabble.com> Is there a way (maybe with ffplay or ffprobe) to show the average q factor of an already encoded video (or individual video frames)? something like this that is shown after encoding: [libx264 @ 0x7fdfeb82ae00] frame I:29 Avg QP:18.52 size: 74048 [libx264 @ 0x7fdfeb82ae00] frame P:1261 Avg QP:21.62 size: 17495 [libx264 @ 0x7fdfeb82ae00] frame B:3557 Avg QP:26.92 size: 2912 ...or is this unknowable? Michael -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Determine-q-factor-from-encoded-video-H-264-tp4658219.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From andrew2012 at flight.us Tue Apr 2 11:32:44 2013 From: andrew2012 at flight.us (Andrew RT) Date: Tue, 2 Apr 2013 05:32:44 -0400 (EDT) Subject: [FFmpeg-user] adding an ac3 stream to an a/v stream In-Reply-To: References: <142598738.443731.1364683467582.open-xchange@email.1and1.com> Message-ID: <1516889934.548681.1364895164808.open-xchange@email.1and1.com> > On March 30, 2013 at 7:19 PM Carl Eugen Hoyos wrote: > > > Andrew RT flight.us> writes: > > > AFAICT (correct me if i'm wrong), the "-acodec copy" > > and similar options are obsolete > > Why? > > > as well as "-newaudio" and things like that. > > -new* does not exist anymore for two years. > > > So, one command I found did not throw any errors, but > > the result is a file with static/buzzing audio: > > > > ffmpeg -i myvideo.avi -i extra_audio.ac3 -map 0:0 > > -map 0:1 -map 1:0 -codec:v:0 copy -codec:a:0 copy > > -codec:a:1 copy muxed_output.avi > > Please provide the complete, uncut console output. > ffmpeg -i myvideo.avi -i extra_audio.ac3 -map 0:0 -map 0:1 -map 1:0 -codec:v:0 copy -codec:a:0 copy -codec:a:1 copy muxed_output.avi ffmpeg version N-51346-gace0cfe Copyright (c) 2000-2013 the FFmpeg developers built on Mar 27 2013 16:23:10 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1) configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab WARNING: library configuration mismatch avutil configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared avcodec configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared avformat configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared avdevice configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared avfilter configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared swscale configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared swresample configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared postproc configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, avi, from 'myvideo.avi': Metadata: encoder : Lavf55.0.100 Duration: 02:20:39.44, start: 0.000000, bitrate: 1486 kb/s Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 720x304 [SAR 1:1 DAR 45:19], 23.98 tbr, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 320 kb/s [ac3 @ 0x97d8940] max_analyze_duration 5000000 reached at 5024000 microseconds [ac3 @ 0x97d8940] Estimating duration from bitrate, this may be inaccurate Input #1, ac3, from 'extra_audio.ac3': Duration: 02:07:42.24, start: 0.000000, bitrate: 447 kb/s Stream #1:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s Output #0, avi, to 'muxed_output.avi': Metadata: ISFT : Lavf55.0.100 Stream #0:0: Video: mpeg4 (XVID / 0x44495658), yuv420p, 720x304 [SAR 1:1 DAR 45:19], q=2-31, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), 320 kb/s Stream #0:2: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), 448 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #1:0 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 1354 fps=0.0 q=-1.0 size= 8099kB time=00:00:56.47 bitrate=1174.9kbits/s frame= 3056 fps=3055 q=-1.0 size= 16887kB time=00:02:07.46 bitrate=1085.3kbits/s frame= 4481 fps=2983 q=-1.0 size= 30164kB time=00:03:06.91 bitrate=1322.1kbits/s frame= 5585 fps=2786 q=-1.0 size= 40466kB time=00:03:54.75 bitrate=1412.1kbits/s frame= 6629 fps=2647 q=-1.0 size= 50754kB time=00:04:36.51 bitrate=1503.7kbits/s frame= 7782 fps=2583 q=-1.0 size= 65733kB time=00:05:31.48 bitrate=1624.4kbits/s frame= 8047 fps=2289 q=-1.0 size= 71709kB time=00:05:46.49 bitrate=1695.4kbits/s frame= 8048 fps=1721 q=-1.0 size= 71736kB time=00:05:46.75 bitrate=1694.8kbits/s frame= 8075 fps=1560 q=-1.0 size= 72373kB time=00:05:50.01 bitrate=1693.9kbits/s frame= 9303 fps=1636 q=-1.0 size= 90285kB time=00:06:28.03 bitrate=1906.1kbits/s frame= 9939 fps=1607 q=-1.0 size= 104210kB time=00:06:57.08 bitrate=2046.8kbits/s frame=11002 fps=1645 q=-1.0 size= 113531kB time=00:07:38.88 bitrate=2026.8kbits/s frame=11680 fps=1625 q=-1.0 size= 120748kB time=00:08:07.52 bitrate=2029.0kbits/s frame=12558 fps=1634 q=-1.0 size= 128849kB time=00:08:43.77 bitrate=2015.2kbits/s frame=14294 fps=1746 q=-1.0 size= 133602kB time=00:09:56.17 bitrate=1835.8kbits/s frame=14606 fps=1679 q=-1.0 size= 146250kB time=00:10:19.42 bitrate=1934.2kbits/s frame=15446 fps=1679 q=-1.0 size= 152360kB time=00:10:44.25 bitrate=1937.3kbits/s frame=16184 fps=1667 q=-1.0 size= 158815kB time=00:11:15.00 bitrate=1927.4kbits/s frame=17344 fps=1699 q=-1.0 size= 166798kB time=00:12:03.39 bitrate=1888.9kbits/s frame=18006 fps=1682 q=-1.0 size= 173723kB time=00:12:34.33 bitrate=1886.6kbits/s frame=18970 fps=1693 q=-1.0 size= 180930kB time=00:13:14.14 bitrate=1866.4kbits/s frame=20477 fps=1749 q=-1.0 size= 190904kB time=00:14:14.06 bitrate=1831.1kbits/s frame=21586 fps=1767 q=-1.0 size= 200876kB time=00:15:00.32 bitrate=1827.8kbits/s frame=22393 fps=1761 q=-1.0 size= 209959kB time=00:15:33.98 bitrate=1841.6kbits/s frame=23387 fps=1770 q=-1.0 size= 218759kB time=00:16:15.45 bitrate=1837.2kbits/s frame=23615 fps=1755 q=-1.0 Lsize= 222070kB time=00:16:26.33 bitrate=1844.4kbits/s video:127607kB audio:92441kB subtitle:0 global headers:0kB muxing overhead 0.919301% Received signal 2: terminating. From francois.visagie at gmail.com Tue Apr 2 12:17:23 2013 From: francois.visagie at gmail.com (Francois Visagie) Date: Tue, 2 Apr 2013 12:17:23 +0200 Subject: [FFmpeg-user] =?gb2312?b?u9i4tKO6ICBIb3cgdG8gY29udHJvbCB0aGUgYml0?= =?gb2312?b?cmF0ZSBiZXR3ZWVuIGEgbWluIGFuZCBtYXhpbiBWQlIgZW5jb2Rp?= =?gb2312?b?bmc=?= In-Reply-To: References: Message-ID: <00c101ce2f8b$4bcb5720$e3620560$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of ??? > Sent: 02 April 2013 08:01 > To: FFmpeg user questions > Subject: [FFmpeg-user] ??? How to control the bitrate between a min > and maxin VBR encoding > > Thank you for answer first. > > > Now I use command below: > ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 1 -vcodec libx264 -vpre > fastfirstpass -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -an -f flv - > map_chapters -1:-1 -g 50 /dev/null > ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 2 -vcodec libx264 -vpre > normal -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -acodec copy -f flv - > map_chapters -1:-1 -g 50 output.flv > > > The output.flv is a video with VBR,The part of maxbitrate is bigger than 1024k > ,I want to control the bitrate between 1500k because of my poor > boundwidth. > > > I searchd more on google .There are some options can be used.Such as - > minrate -maxrate -bufsize. > I used them,but the quality of the output is not satisfactory. > > > Someone said that this command only effective in CBR mode. > > > Is there any other options can make me control the bitrate? If you haven't already, have a look at http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide http://mewiki.project357.com/wiki/X264_Settings#Ratecontrol http://git.videolan.org/?p=x264.git;a=blob_plain;f=doc/ratecontrol.txt;hb=HE AD http://www-ee.uta.edu/dip/courses/ee5351/h.264spic.pdf (although somewhat technical, explains some of the concepts) > > > > > > > > > ------------------ ???? ------------------ > ???: "Carl Eugen Hoyos"; > ????: 2013?4?2?(???) ??1:37 > ???: "ffmpeg-user"; > > ??: Re: [FFmpeg-user] How to control the bitrate between a min and > maxin VBR encoding > > > > ??? <550399108 qq.com> writes: > > > How can I make the bitrate under control? > > -vb allows to control the bitrate. > > > Which version of ffmpeg can be useful? > > On this mailing list, only current git head is supported. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From natgeo.operations at gmail.com Tue Apr 2 12:57:11 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 06:57:11 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > How are 16 channels supposed to match any eight > channel channel layout? the files have two different 8 channel mixes - Streams 1 thru 8 = 5.1+2 Full Mix Streams 9 thru 16 = 5.1+2 MDE Mix (Music, Dialogue, and Effects) The first mix is used for domestic versions, the second mix is used for international version where the show needs to be re-voiced with a local narrators voice. do you think there is some way to apply the -channel_layout twice, so to cover all 16 tracks? if no, is there some way to prevent FFMPEG from guessing the channel layout and making all channels "Channel positions : Front: C" In the event that I cannot correctly assign a layout to all 16, I would prefer that there was no layout information at all. thanks Steve On Tue, Apr 2, 2013 at 1:40 AM, Carl Eugen Hoyos wrote: > Steve C gmail.com> writes: > >> I would like the channel assignment to match >> the 5.1+2 (7.1) arrangement. > > How are 16 channels supposed to match any eight > channel channel layout? > >> I've searched through the FFMPEG documentation, but >> I can't find out a way to do this. > > ffmpeg -h full|grep channel_layout shows a few results here. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From natgeo.operations at gmail.com Tue Apr 2 13:03:35 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 07:03:35 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: Nathan, Yes, that file should play back fine, that is an example of the source file. The problem starts when you try to use -c:v copy FFMPEG shows an error: "fps 2997 is too large" play back the output file with quicktime it says: "Error -2002: a bad public movie atom was found in the movie" You can get -c:v copy to work, but you have to also use -timecode 00:00:00:00 -r 29.97 in the command statement. I don't understand why -timecode and -r are needed to make this work. I'm guessing it has something to do with the drop-frame timecode on a ProResHQ.mov, because when I take a ProResHQ.mov with non-drop-frame timecode (PAL, 25fps) i can get the -c:v copy command to work with no errors. hope that helps. thanks for testing this out, glad its not just me. steve On Mon, Apr 1, 2013 at 10:55 PM, Nathan Lewis wrote: > On Mon, Apr 1, 2013 at 2:00 PM, Steve C wrote: > >> Carl, >> >> The error effects opening and playback with Quicktime 7, on both Mac and >> Windows platforms. >> Final Cut Pro works off of Quictime 7, so it makes sense that you cannot >> import the file into Final Cut. >> >> > Hi Steve, > > I just downloaded the file from the dropbox link provided and I am able to > open and playback the file in both FCP and QT player on Mac OS 10.8.3 . > Both report the file to be MOS (no audio tracks). Details: > > QT Player: version 7.6.6 (1709) QT Version 7.7.1 (2599.24) (Note: I'm > not sure what to make of the discrepancy between version and QT version) > > Final Cut Pro: Version 7.0.3 > > I've lost some of the history of the thread. So, please let me know if I > am missing something important or need to do any additional testing. > > Regards, > > Nathan > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Tue Apr 2 13:26:13 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 11:26:13 +0000 (UTC) Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) References: Message-ID: Steve C gmail.com> writes: > do you think there is some way to apply the -channel_layout > twice, so to cover all 16 tracks? I don't understand this question;-( Did you already test the option? If not, that would explain your question and why I cannot answer it. (You have to specify the option 16 - or at least 14 - times.) > if no, is there some way to prevent FFMPEG from guessing > the channel layout and making all channels > "Channel positions : Front: C" The fact that FFmpeg tries hard to guess a channel_layout for every audio input is a regression; it couldn't be fixed until very recently because of some (very broken) changes to FFmpeg, I believe this has been fixed now (ie audio without channel layouts should be supported again). I am not convinced anybody is working on reverting "always guess a channel layout", probably because developers do not see this as a bug. Carl Eugen From cehoyos at ag.or.at Tue Apr 2 13:29:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 11:29:15 +0000 (UTC) Subject: [FFmpeg-user] Correct fps value for h.264 stream References: <201304011637.r31GbQM3008151@haluter.fromme.com> Message-ID: Oliver Fromme fromme.com> writes: > $ ffmpeg1 -i vstream-orig.h264 -vcodec copy vstream-new.h264 I probably don't understand your issue (mostly because you are not posting output for your original sample that you had problems with and because you are using an outdated version of FFmpeg), but the fact that libavcodec/FFmpeg is unable to extract timestamps from some h264 streams is a known limitation. Patch very welcome, but this apparently is not trivial;-( Carl Eugen From cehoyos at ag.or.at Tue Apr 2 13:32:25 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 11:32:25 +0000 (UTC) Subject: [FFmpeg-user] Several audio tracks encoding issue References: Message-ID: Andrey Aleksandrovich writes: > localhost $ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 > -map 1:0 -map 2:0 -c:v mpeg4 -qscale:v:0 3 -c:a:0 libmp3lame > -qscale:a:0 4 final.avi -c:a:1 libmp3lame -qscale:a:1 4 -newaudio The newaudio option was dropped years ago, map is sufficient now. Note that options after an output file name have no effect on the output file - in your case "-c:a:1 libmp3lame". (Does qscale really work for lame?) > ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers Please update to current git head, 0.10 is quite outdated. Carl Eugen From cehoyos at ag.or.at Tue Apr 2 13:40:42 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 11:40:42 +0000 (UTC) Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 References: Message-ID: Steve C gmail.com> writes: > The error effects opening and playback with Quicktime 7 Sorry, I had misread that part of your original mail. Could you test if 9362f31 and c071b00 also fail? Please stop top-posting, it is considered rude here. Carl Eugen From carlsson.marcus at gmail.com Tue Apr 2 15:11:49 2013 From: carlsson.marcus at gmail.com (Marcus Carlsson) Date: Tue, 2 Apr 2013 15:11:49 +0200 Subject: [FFmpeg-user] ffserver preset issue Message-ID: I'm trying to setup ffserver without any success. Configuration and ffmpeg output can be found here: http://pastie.org/private/6euxa2ilif8pzmuwikhwma If I run ffmpeg and output to a file (with the same arguments as specified in the ffserver configuration) it works just fine, it reads the x264 preset and fills in the "blank" settings but for some reason that doesn't work with the ffserver setup. Another thing to mention is that "Strict -2" is set in ffserver but ffmpeg will still complain about experimental "aac" if I don't explicitly add `-strict -2` to the command line. Any information on what might be the problem or if my setup is all wrong? Currently running ffmpeg 1.2 but will give git/master a go as well. /Marcus From natgeo.operations at gmail.com Tue Apr 2 15:56:51 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 09:56:51 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: > Could you test if 9362f31 and c071b00 also fail? Sorry, I'm not sure what you are referring to - are "9362f31" and "c071b00" different builds of FFMPEG? From paziu at yahoo.com Tue Apr 2 18:59:31 2013 From: paziu at yahoo.com (Mike F) Date: Tue, 2 Apr 2013 09:59:31 -0700 (PDT) Subject: [FFmpeg-user] =?utf-8?b?5Zue5aSN77yaICBIb3cgdG8gY29udHJvbCB0aGUg?= =?utf-8?q?bitrate_between_a_min_and_maxin_VBR_encoding?= In-Reply-To: References: Message-ID: <1364921971.8343.YahooMailNeo@web121702.mail.ne1.yahoo.com> my suggestion: specify bitrate with? ? -b:v??? & ? -b:a # ffmpeg -y -i 1080p-sample.mkv -analyzeduration 2G? -b:v 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -f flv -psnr -acodec libfaac -b:a 128k -g 50 1080p-sample.mkv.flv ----- Original Message ----- > From: ??? <550399108 at qq.com> > To: FFmpeg user questions > Cc: > Sent: Tuesday, April 2, 2013 2:01 AM > Subject: [FFmpeg-user] ??? How to control the bitrate between a min and maxin VBR encoding > >T hank you for answer first. > > > Now I use command below: > ? ? ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 1 -vcodec libx264 -vpre > fastfirstpass -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -an -f flv > -map_chapters -1:-1 -g 50 /dev/null > ? ? ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 2 -vcodec libx264 -vpre > normal -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -acodec copy -f flv > -map_chapters -1:-1 -g 50 output.flv > > > The output.flv is a video with VBR,The part of maxbitrate is bigger than 1024k > ,I want to control the bitrate between 1500k because of my poor boundwidth. > > > I searchd more on google .There are some options can be used.Such as -minrate > -maxrate -bufsize. > I used them,but the quality of the output is not satisfactory. > > > Someone said that this command only effective in CBR mode. > > > Is there any other options can make me control the bitrate? > > > > > > > > > ------------------ ???? ------------------ > ???: "Carl Eugen Hoyos"; > ????: 2013?4?2?(???) ??1:37 > ???: "ffmpeg-user"; > > ??: Re: [FFmpeg-user] How to control the bitrate between a min and maxin VBR > encoding > > > > ??? <550399108 qq.com> writes: > >> How can I make the bitrate under control? > > -vb allows to control the bitrate. > >> Which version of ffmpeg can be useful? > > On this mailing list, only current git head is supported. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From paziu at yahoo.com Tue Apr 2 20:16:22 2013 From: paziu at yahoo.com (Mike F) Date: Tue, 2 Apr 2013 11:16:22 -0700 (PDT) Subject: [FFmpeg-user] =?utf-8?b?5Zue5aSN77yaICBIb3cgdG8gY29udHJvbCB0aGUg?= =?utf-8?q?bitrate_between_a_min_and_maxin_VBR_encoding?= In-Reply-To: References: Message-ID: <1364926582.76912.YahooMailNeo@web121701.mail.ne1.yahoo.com> you could also try to play with refs, subq, qmin this is what I do for a single pass variable ffmpeg -y -i sample.mkv -vcodec libx264 -coder 1 -bf 3 -refs 16 -subq 10 -partitions +parti4x4+parti8x8+partp8x8 -g 200 -keyint_min 25 -qmin 20 -qmax 51 -trellis 2 -b_strategy 2 -sc_threshold 40 -acodec libaacplus -ab 64k -ar 44100 -ac 2 -f flv sample.flv the encoding is SLOW ( 45 fps on a quad e3-1245 xeon * 8 threads ), the bitrate for a 720p does not exceed 500-650kbit/s, the quality is very good for such low bitrate. play with qmin, subq, to get the desired the bitrate and quality ........and lower refs to speed up the encoding process ( default refs is 3 I think ) - 16 reference frames kills the cpu most. ----- Original Message ----- > From: ??? <550399108 at qq.com> > To: FFmpeg user questions > Cc: > Sent: Tuesday, April 2, 2013 2:01 AM > Subject: [FFmpeg-user] ??? How to control the bitrate between a min and maxin VBR encoding > >T hank you for answer first. > > > Now I use command below: > ? ? ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 1 -vcodec libx264 -vpre > fastfirstpass -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -an -f flv > -map_chapters -1:-1 -g 50 /dev/null > ? ? ffmpeg -y -i input.mp4 -analyzeduration 2G -pass 2 -vcodec libx264 -vpre > normal -b 1024k -s 1056x576 -aspect 1056:576 -r 25 -psnr -acodec copy -f flv > -map_chapters -1:-1 -g 50 output.flv > > > The output.flv is a video with VBR,The part of maxbitrate is bigger than 1024k > ,I want to control the bitrate between 1500k because of my poor boundwidth. > > > I searchd more on google .There are some options can be used.Such as -minrate > -maxrate -bufsize. > I used them,but the quality of the output is not satisfactory. > > > Someone said that this command only effective in CBR mode. > > > Is there any other options can make me control the bitrate? > > > > > > > > > ------------------ ???? ------------------ > ???: "Carl Eugen Hoyos"; > ????: 2013?4?2?(???) ??1:37 > ???: "ffmpeg-user"; > > ??: Re: [FFmpeg-user] How to control the bitrate between a min and maxin VBR > encoding > > > > ??? <550399108 qq.com> writes: > >> How can I make the bitrate under control? > > -vb allows to control the bitrate. > >> Which version of ffmpeg can be useful? > > On this mailing list, only current git head is supported. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From natgeo.operations at gmail.com Tue Apr 2 20:20:15 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 14:20:15 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > (You have to specify the option 16 - or at least > 14 - times.) sorry, i was not clear, i have tried to apply -channel_layout after each audio input, but that does not seem to work. FFMPEG just takes that last -channel_layout option and applies that to all audio inputs. I have also tried to apply -channel_layout as one statement for all inputs -channel_layout FL+FR+FC+LFE+BL+BR+SL+SR this option also does not seem to work, FFMPEG does not apply the layout to each input channel inscucession, but applies the full layout to each channel. so it creates an output file that has a 5.1 layout applied to each mono stream. What is the correct way to apply -channel_layout in my situation? I have 16 mono files, each represent one channel of the audio mix 5.1+2 (Full Mix) and 5.1+2 (MDE Mix) > The fact that FFmpeg tries hard to guess a channel_layout > for every audio input is a regression; it couldn't be > > I am not convinced anybody is working on reverting > "always guess a channel layout", probably because > developers do not see this as a bug. is there a way to bring up this issue with the developers? my preference would be to create a output file with no channel layout assigned. if that is not possible, i can try to make the -channel_layout work From andrey.aleksandrovich at googlemail.com Tue Apr 2 20:20:44 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Tue, 2 Apr 2013 21:20:44 +0300 Subject: [FFmpeg-user] Determine q factor from encoded video [H.264] In-Reply-To: <1364889085665-4658219.post@n4.nabble.com> References: <1364889085665-4658219.post@n4.nabble.com> Message-ID: MediaInfo? http://mediainfo.sourceforge.net On 4/2/13, FFmichael wrote: > Is there a way (maybe with ffplay or ffprobe) to show the average q factor > of > an already encoded video (or individual video frames)? > > something like this that is shown after encoding: > [libx264 @ 0x7fdfeb82ae00] frame I:29 Avg QP:18.52 size: 74048 > [libx264 @ 0x7fdfeb82ae00] frame P:1261 Avg QP:21.62 size: 17495 > [libx264 @ 0x7fdfeb82ae00] frame B:3557 Avg QP:26.92 size: 2912 > > ...or is this unknowable? > > Michael > > > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/Determine-q-factor-from-encoded-video-H-264-tp4658219.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From andrey.aleksandrovich at googlemail.com Tue Apr 2 20:30:09 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Tue, 2 Apr 2013 21:30:09 +0300 Subject: [FFmpeg-user] Correct fps value for h.264 stream In-Reply-To: <201304011637.r31GbQM3008151@haluter.fromme.com> References: <201304011637.r31GbQM3008151@haluter.fromme.com> Message-ID: Please, show the MediaInfo (http://mediainfo.sourceforge.net) output for your problem file. We will see. On 4/1/13, Oliver Fromme wrote: > Hello, > > This is probably a very special problem, but maybe someone > has an answer ... From andrey.aleksandrovich at googlemail.com Tue Apr 2 20:38:02 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Tue, 2 Apr 2013 21:38:02 +0300 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: You can also try sox (http://sox.sourceforge.net/). On 4/2/13, Steve C wrote: >> (You have to specify the option 16 - or at least >> 14 - times.) > > sorry, i was not clear, i have tried to apply -channel_layout after > each audio input, > but that does not seem to work. FFMPEG just takes that last > -channel_layout option > and applies that to all audio inputs. > > I have also tried to apply -channel_layout as one statement for all inputs > > -channel_layout FL+FR+FC+LFE+BL+BR+SL+SR > > this option also does not seem to work, FFMPEG does not apply the > layout to each input channel inscucession, > but applies the full layout to each channel. > so it creates an output file that has a 5.1 layout applied to each mono > stream. > > What is the correct way to apply -channel_layout in my situation? > I have 16 mono files, each represent one channel of the audio mix > 5.1+2 (Full Mix) and 5.1+2 (MDE Mix) > >> The fact that FFmpeg tries hard to guess a channel_layout >> for every audio input is a regression; it couldn't be >> >> I am not convinced anybody is working on reverting >> "always guess a channel layout", probably because >> developers do not see this as a bug. > > is there a way to bring up this issue with the developers? > > my preference would be to create a output file with no channel layout > assigned. > if that is not possible, i can try to make the -channel_layout work > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From andrey.aleksandrovich at googlemail.com Tue Apr 2 20:41:05 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Tue, 2 Apr 2013 21:41:05 +0300 Subject: [FFmpeg-user] Several audio tracks encoding issue In-Reply-To: References: Message-ID: Ok, will try. Thanks. On 4/2/13, Carl Eugen Hoyos wrote: > Andrey Aleksandrovich writes: > >> localhost $ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 >> -map 1:0 -map 2:0 -c:v mpeg4 -qscale:v:0 3 -c:a:0 libmp3lame >> -qscale:a:0 4 final.avi -c:a:1 libmp3lame -qscale:a:1 4 -newaudio > > The newaudio option was dropped years ago, map is sufficient now. > Note that options after an output file name have no effect on > the output file - in your case "-c:a:1 libmp3lame". > (Does qscale really work for lame?) > >> ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers > > Please update to current git head, 0.10 is > quite outdated. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From natgeo.operations at gmail.com Tue Apr 2 20:44:27 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 14:44:27 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > You can also try sox (http://sox.sourceforge.net/). thanks, but I'm having enough trouble just trying to use FFMPEG :) From natgeo.operations at gmail.com Tue Apr 2 20:48:31 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 14:48:31 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: >> Could you test if 9362f31 and c071b00 also fail? > > Sorry, I'm not sure what you are referring to - > are "9362f31" and "c071b00" different builds of FFMPEG? Sorry, I have not been able to find any references to "9362f31" and "c071b00" that you mention. I'm not that familiar with building software or using Git, so if anyone else already has these versions or knows how I can obtain them, I would appreciate the help. I have re-tested with the latest build for Windows 32- git-599866f (2013-04-01) And that build produced the same results - same error as before. From cehoyos at ag.or.at Tue Apr 2 21:28:51 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 19:28:51 +0000 (UTC) Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) References: Message-ID: Steve C gmail.com> writes: > > (You have to specify the option 16 - or at least > > 14 - times.) > > sorry, i was not clear, i have tried to apply > -channel_layout after each audio input, That cannot work;-) > but that does not seem to work. FFMPEG just takes > that last -channel_layout option > and applies that to all audio inputs. Since it works fine here, please provide the command line you tried together with complete, uncut console output. Carl Eugen From cehoyos at ag.or.at Tue Apr 2 21:31:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 19:31:50 +0000 (UTC) Subject: [FFmpeg-user] Correct fps value for h.264 stream References: <201304011637.r31GbQM3008151@haluter.fromme.com> Message-ID: Oliver Fromme fromme.com> writes: > $ ffmpeg1 -i vstream-orig.h264 -vcodec copy vstream-new.h264 A bug was fixed today so it should be possible now to set a framerate for raw (h264) input streams. Please test and report back, Carl Eugen PS: Please do not provide mediainfo output on this mailing list unless it is absolutely necessary, it typically only leads to confusion. From natgeo.operations at gmail.com Tue Apr 2 23:53:38 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 17:53:38 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > Since it works fine here, please provide the command line > you tried together with complete, uncut console output. Please see below for the full output. I must be inputting the -channel_layout option incorrectly, see below for the test i've run with 8 audio channels (5.1 +2 audio) "C:\\ffmpeg.exe" -i "C:\\Test\\NTSC_ProResHQ_NA.mov" -i "C:\\Test\\01_5.1.FM.L.wav" -i "C:\\Test\\02_5.1.FM.R.wav" -i "C:\\Test\\03_5.1.FM.C.wav" -i "C:\\Test\\04_5.1.FM.LFE.wav" -i "C:\\Test\\05_5.1.FM.Ls.wav" -i "C:\\Test\\06_5.1.FM.Rs.wav" -i "C:\\Test\\07_Stereo.FM.L.wav" -i "C:\\Test\\08_Stereo.FM.R.wav" -dn -c:v copy -c:a copy -timecode "00:00:00;00" -r 29.97 -channel_layout FL+FR+FC+LFE+BL+BR+SL+SR -report -map 0:0 -map 1:0 -map 2:0 -map 3:0 -map 4:0 -map 5:0 -map 6:0 -map 7:0 -map 8:0 -report "C:\\output-3.mov" This results in an output file with the 5.1+2 channel layout assigned to each audio stream Channel positions : Front: L C R, Side: L R, Rear: L R, LFE but for what i'm trying to do, each audio stream only represents one audio channel in the 5.1+2 mix. For the output to be correct it would have to look like this: Audio Stream 1 = FL Audio Stream 2= FR Audio Stream 3 = FC does that make sense? it doesn't seem like there is any way to do this through -channel_layout? ############################################# ffmpeg started on 2013-04-02 at 17:42:59 Report written to "ffmpeg-20130402-174259.log" Command line: "C:\\ffmpeg.exe" -i "C:\\Test\\NTSC_ProResHQ_NA.mov" -i "C:\\Test\\01_5.1.FM.L.wav" -i "C:\\Test\\02_5.1.FM.R.wav" -i "C:\\Test\\03_5.1.FM.C.wav" -i "C:\\Test\\04_5.1.FM.LFE.wav" -i "C:\\Test\\05_5.1.FM.Ls.wav" -i "C:\\Test\\06_5.1.FM.Rs.wav" -i "C:\\Test\\07_Stereo.FM.L.wav" -i "C:\\Test\\08_Stereo.FM.R.wav" -dn -c:v copy -c:a copy -timecode "00:00:00;00" -r 29.97 -channel_layout FL+FR+FC+LFE+BL+BR+SL+SR -report -map 0:0 -map 1:0 -map 2:0 -map 3:0 -map 4:0 -map 5:0 -map 6:0 -map 7:0 -map 8:0 -report "C:\\output-3.mov" ffmpeg version N-51511-g599866f Copyright (c) 2000-2013 the FFmpeg developers built on Apr 1 2013 12:44:46 with gcc 4.8.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 24.100 / 52. 24.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Splitting the commandline. Reading option '-i' ... matched as input file with argument 'C:\Test\NTSC_ProResHQ_NA.mov'. Reading option '-i' ... matched as input file with argument 'C:\Test\01_5.1.FM.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\02_5.1.FM.R.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\03_5.1.FM.C.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\04_5.1.FM.LFE.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\05_5.1.FM.Ls.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\06_5.1.FM.Rs.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\07_Stereo.FM.L.wav'. Reading option '-i' ... matched as input file with argument 'C:\Test\08_Stereo.FM.R.wav'. Reading option '-dn' ... matched as option 'dn' (disable data) with argument '1'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-timecode' ... matched as option 'timecode' (set initial TimeCode value.) with argument '00:00:00;00'. Reading option '-r' ... matched as option 'r' (set frame rate (Hz value, fraction or abbreviation)) with argument '29.97'. Reading option '-channel_layout' ... matched as option 'channel_layout' (set channel layout) with argument 'FL+FR+FC+LFE+BL+BR+SL+SR'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '1:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '2:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '3:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '4:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '5:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '6:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '7:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '8:0'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option 'C:\output-3.mov' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file C:\Test\NTSC_ProResHQ_NA.mov. Successfully parsed a group of options. Opening an input file: C:\Test\NTSC_ProResHQ_NA.mov. [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] ISO: File Type Major Brand: qt [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] File position before avformat_find_stream_info() is 19318 [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] Ignoring outlier non primary stream starttime -80.013344 [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] Ignoring outlier non primary stream starttime -80.013344 [mov,mp4,m4a,3gp,3g2,mj2 @ 03071800] File position after avformat_find_stream_info() is 889286 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Test\NTSC_ProResHQ_NA.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-03-20 16:19:14 Duration: 00:01:10.07, start: 0.000000, bitrate: 207771 kb/s Stream #0:0(eng), 1, 1/2997: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 207768 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Stream #0:1(eng), 1, 1/2997: Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Successfully opened the file. Parsing a group of options: input file C:\Test\01_5.1.FM.L.wav. Successfully parsed a group of options. Opening an input file: C:\Test\01_5.1.FM.L.wav. [wav @ 0212e580] Format wav probed with size=2048 and score=99 [wav @ 0212e580] File position before avformat_find_stream_info() is 1024 [wav @ 0212e580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0212e580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0212e580] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 0212e580] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'C:\Test\01_5.1.FM.L.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdI18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #1:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\02_5.1.FM.R.wav. Successfully parsed a group of options. Opening an input file: C:\Test\02_5.1.FM.R.wav. [wav @ 030d3ba0] Format wav probed with size=2048 and score=99 [wav @ 030d3ba0] File position before avformat_find_stream_info() is 1024 [wav @ 030d3ba0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 030d3ba0] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 030d3ba0] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 030d3ba0] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #2.0 : mono Input #2, wav, from 'C:\Test\02_5.1.FM.R.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdP18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #2:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\03_5.1.FM.C.wav. Successfully parsed a group of options. Opening an input file: C:\Test\03_5.1.FM.C.wav. [wav @ 0307a840] Format wav probed with size=2048 and score=99 [wav @ 0307a840] File position before avformat_find_stream_info() is 1024 [wav @ 0307a840] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0307a840] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0307a840] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 0307a840] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #3.0 : mono Input #3, wav, from 'C:\Test\03_5.1.FM.C.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdM18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #3:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\04_5.1.FM.LFE.wav. Successfully parsed a group of options. Opening an input file: C:\Test\04_5.1.FM.LFE.wav. [wav @ 03f24900] Format wav probed with size=2048 and score=99 [wav @ 03f24900] File position before avformat_find_stream_info() is 1024 [wav @ 03f24900] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03f24900] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03f24900] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 03f24900] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #4.0 : mono Input #4, wav, from 'C:\Test\04_5.1.FM.LFE.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdW18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #4:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\05_5.1.FM.Ls.wav. Successfully parsed a group of options. Opening an input file: C:\Test\05_5.1.FM.Ls.wav. [wav @ 03fafa40] Format wav probed with size=2048 and score=99 [wav @ 03fafa40] File position before avformat_find_stream_info() is 1024 [wav @ 03fafa40] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03fafa40] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03fafa40] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 03fafa40] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #5.0 : mono Input #5, wav, from 'C:\Test\05_5.1.FM.Ls.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdS18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #5:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\06_5.1.FM.Rs.wav. Successfully parsed a group of options. Opening an input file: C:\Test\06_5.1.FM.Rs.wav. [wav @ 041a3580] Format wav probed with size=2048 and score=99 [wav @ 041a3580] File position before avformat_find_stream_info() is 1024 [wav @ 041a3580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 041a3580] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 041a3580] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 041a3580] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #6.0 : mono Input #6, wav, from 'C:\Test\06_5.1.FM.Rs.wav': Metadata: encoded_by : Pro Tools originator_reference: !gKdT18u0RmaaaGk date : 2012-12-10 creation_time : 20:59:47 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #6:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\07_Stereo.FM.L.wav. Successfully parsed a group of options. Opening an input file: C:\Test\07_Stereo.FM.L.wav. [wav @ 030c5a80] Format wav probed with size=2048 and score=99 [wav @ 030c5a80] File position before avformat_find_stream_info() is 1024 [wav @ 030c5a80] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 030c5a80] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 030c5a80] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 030c5a80] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #7.0 : mono Input #7, wav, from 'C:\Test\07_Stereo.FM.L.wav': Metadata: encoded_by : Pro Tools originator_reference: bBAc908u07maaaGk date : 2012-12-10 creation_time : 20:59:48 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #7:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: input file C:\Test\08_Stereo.FM.R.wav. Successfully parsed a group of options. Opening an input file: C:\Test\08_Stereo.FM.R.wav. [wav @ 03ee9e40] Format wav probed with size=2048 and score=99 [wav @ 03ee9e40] File position before avformat_find_stream_info() is 1024 [wav @ 03ee9e40] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03ee9e40] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 03ee9e40] max_analyze_duration 5000000 reached at 5005000 microseconds [wav @ 03ee9e40] File position after avformat_find_stream_info() is 734029 Guessed Channel Layout for Input Stream #8.0 : mono Input #8, wav, from 'C:\Test\08_Stereo.FM.R.wav': Metadata: encoded_by : Pro Tools originator_reference: bBAcc18u07maaaGk date : 2012-12-10 creation_time : 20:59:48 time_reference : 1727517792 Duration: 00:01:10.07, bitrate: 1194 kb/s Stream #8:0, 178, 1/48000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32, 1152 kb/s Successfully opened the file. Parsing a group of options: output file C:\output-3.mov. Applying option dn (disable data) with argument 1. Applying option c:v (codec name) with argument copy. Applying option c:a (codec name) with argument copy. Applying option timecode (set initial TimeCode value.) with argument 00:00:00;00. Applying option r (set frame rate (Hz value, fraction or abbreviation)) with argument 29.97. Applying option channel_layout (set channel layout) with argument FL+FR+FC+LFE+BL+BR+SL+SR. Applying option map (set input stream mapping) with argument 0:0. Applying option map (set input stream mapping) with argument 1:0. Applying option map (set input stream mapping) with argument 2:0. Applying option map (set input stream mapping) with argument 3:0. Applying option map (set input stream mapping) with argument 4:0. Applying option map (set input stream mapping) with argument 5:0. Applying option map (set input stream mapping) with argument 6:0. Applying option map (set input stream mapping) with argument 7:0. Applying option map (set input stream mapping) with argument 8:0. Successfully parsed a group of options. Opening an output file: C:\output-3.mov. Codec AVOption gop_timecode (MPEG GOP Timecode in hh:mm:ss[:;.]ff format) specified for output file #0 (C:\output-3.mov) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream. Successfully opened the file. Output #0, mov, to 'C:\output-3.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt timecode : 00:00:00;00 encoder : Lavf55.1.100 Stream #0:0(eng), 0, 1/11988: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 207768 kb/s, 29.97 fps, 11988 tbn, 29.97 tbc Metadata: creation_time : 2013-03-20 16:19:14 handler_name : Apple Alias Data Handler timecode : 09:58:30;00 Stream #0:1, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:2, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:3, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:4, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:5, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:6, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:7, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream #0:8, 0, 1/48000: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL+FR+FC+LFE+BL+BR+SL+SR), 1152 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #1:0 -> #0:1 (copy) Stream #2:0 -> #0:2 (copy) Stream #3:0 -> #0:3 (copy) Stream #4:0 -> #0:4 (copy) Stream #5:0 -> #0:5 (copy) Stream #6:0 -> #0:6 (copy) Stream #7:0 -> #0:7 (copy) Stream #8:0 -> #0:8 (copy) Press [q] to stop, [?] for help frame= 13 fps=0.0 q=-1.0 size= 11546kB time=00:00:55.73 bitrate=1696.9kbits/s [TRUNCATED] frame= 2093 fps= 41 q=-1.0 size= 1848864kB time=00:01:09.83 bitrate=216876.4kbits/s No more output streams to write to, finishing. frame= 2100 fps= 41 q=-1.0 Lsize= 1856075kB time=00:01:10.07 bitrate=216996.6kbits/s video:1777147kB audio:78829kB subtitle:0 global headers:0kB muxing overhead 0.005335% From cehoyos at ag.or.at Wed Apr 3 00:03:13 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 2 Apr 2013 22:03:13 +0000 (UTC) Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) References: Message-ID: Steve C gmail.com> writes: > "C:\\ffmpeg.exe" -i "C:\\Test\\NTSC_ProResHQ_NA.mov" -i > "C:\\Test\\01_5.1.FM.L.wav" -i "C:\\Test\\02_5.1.FM.R.wav" -i > "C:\\Test\\03_5.1.FM.C.wav" -i "C:\\Test\\04_5.1.FM.LFE.wav" Please try: $ ffmpeg.exe -channel_layout FL -i 01_5.1.FM.L.wav -channel_layout FR -i 02_5.1.FM.R.wav -channel_layout FC -i 03_5.1.FM.C.wav -channel_layout LFE -i 04_5.1.FM.LFE.wav ... Carl Eugen From natgeo.operations at gmail.com Wed Apr 3 04:26:51 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 2 Apr 2013 22:26:51 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > Please try: > $ ffmpeg.exe -channel_layout FL -i 01_5.1.FM.L.wav > -channel_layout FR -i 02_5.1.FM.R.wav -channel_layout FC > -i 03_5.1.FM.C.wav -channel_layout LFE -i 04_5.1.FM.LFE.wav okay, that seems to work! thanks for the help!! I had tried something similar, but was putting the -channel_layout option after the input for each audio channel. putting the -channel_layout option before the input obviously makes all the difference. the output file that is created works with Quicktime 7 and can be imported into Final Cut Pro (although FCP gives a warning that the file is not optimized). the only issue i see with the output file is that Quicktime 7, by default activates only the first audio stream, all other audio streams are turned off, and have to be enabled manually through the Quicktime movie properties. not sure why this would happen - here is a screen shot of the Quicktime Movie properties to illustrate what I'm describing: http://www.datafilehost.com/download-4aa3f9cf.html I don't know if there is anything that can be done through FFMPEG to change this, seems like an issue with Quicktime? also, FYI here is the FFMPEG statement that worked for me: ffmpeg -i /Test-Files-02/035925_NTSC_ProResHQ_NA.mov -channel_layout FL -i /Test-Files-02/5.1.FM.L.wav -channel_layout FR -i /Test-Files-02/5.1.FM.R.wav -channel_layout FC -i /Test-Files-02/5.1.FM.C.wav -channel_layout LFE -i /Test-Files-02/5.1.FM.LFE.wav -channel_layout BL -i /Test-Files-02/5.1.FM.Ls.wav -channel_layout BR -i /Test-Files-02/5.1.FM.Rs.wav -channel_layout SL -i /Test-Files-02/Stereo.FM.L.wav -channel_layout SR -i /Test-Files-02/Stereo.FM.R.wav -c:v copy -c:a copy -r 29.97 -map 0:0 -map 1:0 -map 2:0 -map 3:0 -map 4:0 -map 5:0 -map 6:0 -map 7:0 -map 8:0 -report /Test-Files-02/output.mov From cehoyos at ag.or.at Wed Apr 3 09:38:18 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 3 Apr 2013 07:38:18 +0000 (UTC) Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) References: Message-ID: Steve C gmail.com> writes: > the only issue i see with the output file is that > Quicktime 7, by default activates only the first > audio stream This is probably a result of fixing ticket #468. Since your use case appear less common to me, this is unlikely to get fixed by default. (We could change it for the "mono" case, but that would make it even more confusing imo.) But I wonder how this should ever work in your specific case: FFmpeg would set all 16 channels as default, meaning that QuickTime would play both languages simultaneously... Carl Eugen From tevans.uk at googlemail.com Wed Apr 3 11:42:16 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 3 Apr 2013 10:42:16 +0100 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: On Tue, Apr 2, 2013 at 7:48 PM, Steve C wrote: >>> Could you test if 9362f31 and c071b00 also fail? >> >> Sorry, I'm not sure what you are referring to - >> are "9362f31" and "c071b00" different builds of FFMPEG? > > > Sorry, I have not been able to find any references to "9362f31" and > "c071b00" that you mention. > I'm not that familiar with building software or using Git, so if > anyone else already has these versions > or knows how I can obtain them, I would appreciate the help. > Presumably you have a git checkout of ffmpeg you are building from? >From the command line, simply go the ffmpeg folder, and type "git checkout 9362f31". This will update your version of ffmpeg to that revision. You can then rebuild ffmpeg (remember to clean first) and test to see if that version works. Repeat the process for the c071b00 revision. To go back to the tip, 'git checkout HEAD'. Cheers Tom From natgeo.operations at gmail.com Wed Apr 3 13:05:33 2013 From: natgeo.operations at gmail.com (Steve C) Date: Wed, 3 Apr 2013 07:05:33 -0400 Subject: [FFmpeg-user] Fwd: "Guessed Channel Layout for Input Stream" - how to assign channels? (FL+FR+FC+LFE+BL+BR) In-Reply-To: References: Message-ID: > This is probably a result of fixing ticket #468. > Since your use case appear less common to me, > this is unlikely to get fixed by default. > (We could change it for the "mono" case, but > that would make it even more confusing imo.) Yeah, it only causes an issue if someone opens the file with Quicktime, and doesnt see all 16 tracks present. They will need to know that they have to turned on the tracks in Movie Properties in order to hear them. When you drop the file into Final Cut Pro, all audio tracks are present, even if they are turned off in properties. > But I wonder how this should ever work in your > specific case: FFmpeg would set all 16 channels > as default, meaning that QuickTime would play > both languages simultaneously... We would not need the tracks active for playback simultaneously, but more for when the QC technician is checking the files - so they have a quick visual way to know all tracks are present. I also have not tested how this may affect transcoding down the workflow chain. If you take an MOV file that has its audio tracks turned off, will those tracks transcode through our workflow? I imagine it would be okay, but need to test. thanks again. From sorcchan at gmail.com Wed Apr 3 14:14:20 2013 From: sorcchan at gmail.com (=?UTF-8?B?0KPRiNCw0LrQvtCyINCQ0L3QtNGA0LXQuQ==?=) Date: Wed, 3 Apr 2013 16:14:20 +0400 Subject: [FFmpeg-user] Migration from avcodec_open to avcodec_open2 Message-ID: Hello all. I got a code: vcontext->width = 720; vcontext->height = 576; vcontext->time_base = (AVRational) {1, PAL_FPS}; vcontext->pix_fmt = PIX_FMT_YUV420P; where vcontex is AVCodecContext and I want to migrate to avcodec_open2 what requires options dictionary and my question is what equivalent options i should set into dictionary? E.g.: av_dict_set(&dict, "width", "720", 0); av_dict_set(&dict, "height", "576", 0); av_dict_set(&dict, "time_base", ????, 0); av_dict_set(&dict, "pix_fmt", "PIX_FMT_YUV420P", 0); Where I can see the list of codec context and codec specific options, it's description and text equivalent for dictionary? From heinrich.fink at hfink.eu Wed Apr 3 15:53:08 2013 From: heinrich.fink at hfink.eu (Heinrich Fink) Date: Wed, 3 Apr 2013 15:53:08 +0200 Subject: [FFmpeg-user] Sync troubles with input pipes from raw -> AAC/Scaling/H.264 -> live RTMP In-Reply-To: References: <6AA7CF67-52FD-4CF0-8B74-A2B3144924FA@hfink.eu> Message-ID: Hi, thanks for taking time to comment on this. Carl Eugen Hoyos wrote: > Heinrich Fink hfink.eu> writes: > >> I have troubles with A/V sync for a live RTMP output stream >> generated by ffmpeg. > > Is your problem only reproducible with rtmp output or also if > you output to a (flv) file? It is also reproducible with other output types / output format types (see below). > >> However, unfortunately A/V seems to drift after running >> for a couple of hours. > > (I wrote the above before reading this sentence.) > Could this be a problem with the framerate? > Ie, are you missing a ?000/1001" somewhere? My test case uses 25p only, so this shouldn?t be an issue. > >> Here?s the full command: > > It is unfortunately missing the complete, uncut console > output that is always needed if you want support on this > mailing list. sorry for that, my new test case (using flv1 as you suggested) includes now both the command the console output. > Generally, if your problem is reproducible without external > libraries (libx264), you should not use them as not using > them often makes reproducing the problem easier. > (Is it reproducible with -vcodec flv1 ?) yes, it is. I ran a test for about 3 hours, recording into a flv file, and I was able to observe a A/V shift of 4 frames. Here is the command: ./ffmpeg \ -nostdin \ -vsync cfr \ -c:v rawvideo\ -top:v 1 \ -video_size 1920x1080 \ -pixel_format yuv420p \ -f rawvideo \ -fflags nobuffer \ -i /tmp/pipevideo \ -c:a pcm_s16le \ -ac 2 \ -ar 48000 \ -f s16le \ -fflags nobuffer \ -analyzeduration 0 \ -i /tmp/pipeaudio \ -c:v flv1 \ -map 0:0,1:0 \ -map 1:0 \ -loglevel debug \ -vf "yadif=0:-1:0, scale=640:360" \ -af "aresample=async=1" \ -strict -2 \ -acodec aac \ -ab 192000 \ -ac 2 \ -ar 48000 \ -r 25 \ -fflags nobuffer \ -y \ -f flv capture.flv and here is the uncut console output of the full test session: ## CONSOLE OUTPUT BEGIN ## ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 15 2013 18:24:17 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --arch=x86_64 --enable-runtime-cpudetect libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Splitting the commandline. Reading option '-nostdin' ... matched as option 'stdin' (enable or disable interaction on standard input) with argument 0. Reading option '-vsync' ... matched as option 'vsync' (video sync method) with argument 'cfr'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'rawvideo'. Reading option '-top:v' ... matched as option 'top' (top=1/bottom=0/auto=-1 field first) with argument '1'. Reading option '-video_size' ... matched as AVOption 'video_size' with argument '1920x1080'. Reading option '-pixel_format' ... matched as AVOption 'pixel_format' with argument 'yuv420p'. Reading option '-f' ... matched as option 'f' (force format) with argument 'rawvideo'. Reading option '-fflags' ... matched as AVOption 'fflags' with argument 'nobuffer'. Reading option '-i' ... matched as input file with argument '/tmp/pipevideo'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_s16le'. Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '48000'. Reading option '-f' ... matched as option 'f' (force format) with argument 's16le'. Reading option '-fflags' ... matched as AVOption 'fflags' with argument 'nobuffer'. Reading option '-analyzeduration' ... matched as AVOption 'analyzeduration' with argument '0'. Reading option '-i' ... matched as input file with argument '/tmp/pipeaudio'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'flv1'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:0,1:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '1:0'. Reading option '-loglevel' ... matched as option 'loglevel' (set libav* logging level) with argument 'debug'. Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'yadif=0:-1:0, scale=640:360'. Reading option '-af' ... matched as option 'af' (set audio filters) with argument 'aresample=async=1'. Reading option '-strict' ... matched as AVOption 'strict' with argument '-2'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'. Reading option '-ab' ... matched as AVOption 'ab' with argument '192000'. Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '48000'. Reading option '-r' ... matched as option 'r' (set frame rate (Hz value, fraction or abbreviation)) with argument '25'. Reading option '-fflags' ... matched as AVOption 'fflags' with argument 'nobuffer'. Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'. Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'. Reading option 'capture.flv' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option nostdin (enable or disable interaction on standard input) with argument 0. Applying option vsync (video sync method) with argument cfr. Applying option loglevel (set libav* logging level) with argument debug. Applying option y (overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file /tmp/pipevideo. Applying option c:v (codec name) with argument rawvideo. Applying option top:v (top=1/bottom=0/auto=-1 field first) with argument 1. Applying option f (force format) with argument rawvideo. Successfully parsed a group of options. Opening an input file: /tmp/pipevideo. [rawvideo @ 0x7f9fea020200] File position before avformat_find_stream_info() is 0 [rawvideo @ 0x7f9fea020200] All info found [rawvideo @ 0x7f9fea020200] Estimating duration from bitrate, this may be inaccurate [rawvideo @ 0x7f9fea020200] File position after avformat_find_stream_info() is 3110400 Input #0, rawvideo, from '/tmp/pipevideo': Duration: N/A, start: 0.000000, bitrate: 622080 kb/s Stream #0:0, 1, 1/25: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080, 1/25, 622080 kb/s, 25 tbr, 25 tbn, 25 tbc Successfully opened the file. Parsing a group of options: input file /tmp/pipeaudio. Applying option c:a (codec name) with argument pcm_s16le. Applying option ac (set number of audio channels) with argument 2. Applying option ar (set audio sampling rate (in Hz)) with argument 48000. Applying option f (force format) with argument s16le. Successfully parsed a group of options. Opening an input file: /tmp/pipeaudio. [s16le @ 0x7f9fea029600] File position before avformat_find_stream_info() is 0 [s16le @ 0x7f9fea029600] max_analyze_duration 0 reached at 0 microseconds [s16le @ 0x7f9fea029600] Estimating duration from bitrate, this may be inaccurate [s16le @ 0x7f9fea029600] File position after avformat_find_stream_info() is 12288 Guessed Channel Layout for Input Stream #1.0 : stereo Input #1, s16le, from '/tmp/pipeaudio': Duration: N/A, bitrate: 1536 kb/s Stream #1:0, 2, 1/48000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Successfully opened the file. Parsing a group of options: output file capture.flv. Applying option c:v (codec name) with argument flv1. Applying option map (set input stream mapping) with argument 0:0,1:0. Applying option map (set input stream mapping) with argument 1:0. Applying option vf (set video filters) with argument yadif=0:-1:0, scale=640:360. Applying option af (set audio filters) with argument aresample=async=1. Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac. Applying option ac (set number of audio channels) with argument 2. Applying option ar (set audio sampling rate (in Hz)) with argument 48000. Applying option r (set frame rate (Hz value, fraction or abbreviation)) with argument 25. Applying option f (force format) with argument flv. Successfully parsed a group of options. Opening an output file: capture.flv. Matched encoder 'flv' for codec 'flv1'. Successfully opened the file. [yadif @ 0x7f9fe9c03880] Setting 'mode' to value '0' [yadif @ 0x7f9fe9c03880] Setting 'parity' to value '-1' [yadif @ 0x7f9fe9c03880] Setting 'deint' to value '0' [Parsed_yadif_0 @ 0x7f9fe9c04240] mode:0 parity:-1 deint:0 [scale @ 0x7f9fe9c03c40] Setting 'w' to value '640' [scale @ 0x7f9fe9c03c40] Setting 'h' to value '360' [scale @ 0x7f9fe9c03c40] Setting 'flags' to value '0x4' [Parsed_scale_1 @ 0x7f9fe9c03be0] w:640 h:360 flags:'0x4' interl:0 [buffer @ 0x7f9fe9c03f80] Setting entry with key 'video_size' to value '1920x1080' [buffer @ 0x7f9fe9c03f80] Setting entry with key 'pix_fmt' to value '0' [buffer @ 0x7f9fe9c03f80] Setting entry with key 'time_base' to value '1/25' [buffer @ 0x7f9fe9c03f80] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x7f9fe9c03f80] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x7f9fe9c03f80] Setting entry with key 'frame_rate' to value '25/1' [graph 0 input from stream 0:0 @ 0x7f9fe9c04060] w:1920 h:1080 pixfmt:yuv420p tb:1/25 fr:25/1 sar:0/1 sws_param:flags=2 [Parsed_scale_1 @ 0x7f9fe9c03be0] w:1920 h:1080 fmt:yuv420p sar:0/1 -> w:640 h:360 fmt:yuv420p sar:0/1 flags:0x4 [abuffer @ 0x7f9fe9c08220] Setting entry with key 'time_base' to value '1/48000' [abuffer @ 0x7f9fe9c08220] Setting entry with key 'sample_rate' to value '48000' [abuffer @ 0x7f9fe9c08220] Setting entry with key 'sample_fmt' to value 's16' [abuffer @ 0x7f9fe9c08220] Setting entry with key 'channel_layout' to value '0x3' [graph 1 input from stream 1:0 @ 0x7f9fe9c08160] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3 [aformat @ 0x7f9fe9c087a0] Setting entry with key 'sample_fmts' to value 'fltp' [aformat @ 0x7f9fe9c087a0] Setting entry with key 'sample_rates' to value '48000' [aformat @ 0x7f9fe9c087a0] Setting entry with key 'channel_layouts' to value '0x3' [Parsed_aresample_0 @ 0x7f9fe9c06f20] ch:2 chl:stereo fmt:s16 r:48000Hz -> ch:2 chl:stereo fmt:fltp r:48000Hz [flv @ 0x7f9fea03b400] intra_quant_bias = 0 inter_quant_bias = -64 Output #0, flv, to 'capture.flv': Metadata: encoder : Lavf54.63.104 Stream #0:0, 0, 1/1000: Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 640x360, 1/25, q=2-31, 200 kb/s, 1k tbn, 25 tbc Stream #0:1, 0, 1/1000: Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, stereo, fltp, 192 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> flv) Stream #1:0 -> #0:1 (pcm_s16le -> aac) *** 1 dup! EOF on sink link output stream 0:0:default. time=03:09:12.68 bitrate= 559.1kbits/s dup=1 drop=0 [output stream 0:1 @ 0x7f9fe9c084e0] EOF on sink link output stream 0:1:default. No more output streams to write to, finishing. [aac @ 0x7f9fea03ee00] Trying to remove 768 more samples than there are in the queue frame=283826 fps= 25 q=31.0 Lsize= 774856kB time=03:09:13.04 bitrate= 559.1kbits/s dup=1 drop=0 video:540535kB audio:221051kB subtitle:0 global headers:0kB muxing overhead 1.742400% Statistics: 882812390400 bytes read, 0 seeks [AVIOContext @ 0x7f9fe9c04c20] Statistics: 2179752960 bytes read, 0 seeks ## CONSOLE OUTPUT END ## I would appreciate any comments on this. many thanks, Heinrich From tag at gmx.com Wed Apr 3 16:35:33 2013 From: tag at gmx.com (Timm Ganske) Date: Wed, 3 Apr 2013 16:35:33 +0200 (CEST) Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? Message-ID: Using my smartphone I sometimes do videos. It is very convenient that they can be played directly in Firefox. Currently I am using ffmpeg2theora to create Theora videos, but I want to change to webm (VP8/vorbis) because it uses less space for same quality resp. better quality for same space. The videos are in different resolutions, and sometimes I crop them. I want to have the bitrate adopting to what is reasonable for the resolution. If I set a fixed bitrate let us say 800 kb/s, it is too much for low resolutions (some are below 320x240) and not enough for high resolutions (my S3 is capable of Full HD). But I want to convert them using a single command line for all of them. Is there a parameter for ffmpeg which lets me choose -video-quality (low, reasonable, good, excellent) which depends on the resolution? If not, is it possible to support such an option in the future? Or do I need some script which analyzes the video and sets the bitrate accordingly? From tevans.uk at googlemail.com Wed Apr 3 17:09:15 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Wed, 3 Apr 2013 16:09:15 +0100 Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? In-Reply-To: References: Message-ID: On Wed, Apr 3, 2013 at 3:35 PM, Timm Ganske wrote: > Using my smartphone I sometimes do videos. It is very convenient that they can be played directly in Firefox. Currently I am using ffmpeg2theora to create Theora videos, but I want to change to webm (VP8/vorbis) because it uses less space for same quality resp. better quality for same space. The videos are in different resolutions, and sometimes I crop them. I want to have the bitrate adopting to what is reasonable for the resolution. If I set a fixed bitrate let us say 800 kb/s, it is too much for low resolutions (some are below 320x240) and not enough for high resolutions (my S3 is capable of Full HD). But I want to convert them using a single command line for all of them. > > Is there a parameter for ffmpeg which lets me choose -video-quality (low, reasonable, good, excellent) which depends on the resolution? If not, is it possible to support such an option in the future? Or do I need some script which analyzes the video and sets the bitrate accordingly? Please, line breaks are free on the internet, use them every 80 characters or so! You can use the -crf argument, which sets a quality setting for the final video, not a bitrate. I believe this works with libvpx. Cheers Tom From classysnet at gmail.com Wed Apr 3 17:34:35 2013 From: classysnet at gmail.com (Rares Pop) Date: Wed, 3 Apr 2013 18:34:35 +0300 Subject: [FFmpeg-user] hls segmenter and error 'dimensions not set' Message-ID: Dear all, I have recently upgraded ffmpeg to its latest from git and the ffmpeg segmenter stopped working. Here is my log: [root at enc6 t]# /usr/local/bin/ffmpeg -i hi.ts -map 0 -c copy -f segment -segment_time 10 -segment_list hi.m3u8 hi_%05d.ts ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 29 2013 08:50:28 with gcc 4.5.3 (GCC) 20120403 (ALT Linux 4.5.3-alt1.M60P.1) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-gpl --disable-postproc --enable-pthreads --enable-shared --disable-static --disable-libvorbis --disable-libfaac --disable-libxvid --enable-libx264 --disable-libmp3lame --disable-libgsm --disable-libdc1394 --disable-zlib --disable-memalign-hack --disable-ffserver --disable-ffplay --disable-libschroedinger --disable-avisynth --disable-libnut --enable-libtheora --enable-version3 --enable-libopencore-amrwb --enable-libopencore-amrnb --disable-ffprobe --disable-libvpx --disable-libv4l2 --enable-nonfree --enable-librtmp --enable-libfdk-aac --enable-libvo-aacenc --disable-debug --extra-cflags='-pipe -Wall -g -O2 -frename-registers -fPIC -DPIC' libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [mpegts @ 0x20dd1c0] max_analyze_duration 5000000 reached at 5013333 microseconds [mpegts @ 0x20dd1c0] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [NULL @ 0x20e1440] start time is not set in estimate_timings_from_pts Input #0, mpegts, from 'hi.ts': Duration: 00:09:59.80, start: 1.400000, bitrate: 782 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 ([27][0][0][0] / 0x001B), 90k tbr, 90k tbn, 180k tbc Stream #0:1[0x101](rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 102 kb/s [segment @ 0x20e2780] dimensions not set Output #0, segment, to 'hi_%05d.ts': Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), q=2-31, 90k tbn, 90k tbc Stream #0:1(rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, 102 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument Is this a regression in ffmpeg or am I missing something. Previous ffmpeg (1.1 build in November) is working fine with the same data. Thanks for your help, Rares From tag at gmx.com Wed Apr 3 17:47:02 2013 From: tag at gmx.com (tag at gmx.com) Date: Wed, 3 Apr 2013 17:47:02 +0200 (CEST) Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? In-Reply-To: References: , Message-ID: Tom Evans wrote: > You can use the -crf argument, which sets a quality setting for the > final video, not a bitrate. I believe this works with libvpx. ffmpeg -i 12082012.mp4 -crf 10 12082012.mp4_crf10.webm ffmpeg -i 12082012.mp4 -crf 20 12082012.mp4_crf20.webm ffmpeg -i 12082012.mp4 -crf 40 12082012.mp4_crf40.webm -crf 10: Filesize 11716 KB -crf 20: Filesize 11716 KB -crf 40: Filesize 11716 KB Seems it does not work. Default 200 kb/s stays active all the time. Are there other options? From andrey.aleksandrovich at googlemail.com Wed Apr 3 18:19:28 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Wed, 3 Apr 2013 19:19:28 +0300 Subject: [FFmpeg-user] Remove 'anamorph' from video? Message-ID: Hi all, I have video: --- 720x390 [SAR 1:1 DAR 24:13], SAR 403:490 DAR 372:245 --- When it's playing the resolution scaled to 720x474. Is there way to remove 'aspect ratio' information from video? So I want square pixels in the video and it will be playing without scaling. From natgeo.operations at gmail.com Wed Apr 3 18:26:18 2013 From: natgeo.operations at gmail.com (Steve C) Date: Wed, 3 Apr 2013 12:26:18 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: > Presumably you have a git checkout of ffmpeg you are building from? Actually, I don't I've just been working off pre-built binaries. > From the command line, simply go the ffmpeg folder, and type "git > checkout 9362f31". This will update your version of ffmpeg to that > revision. You can then rebuild ffmpeg (remember to clean first) and > test to see if that version works. I will try to figure this out. From classysnet at gmail.com Wed Apr 3 19:03:47 2013 From: classysnet at gmail.com (Rares Pop) Date: Wed, 3 Apr 2013 20:03:47 +0300 Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter Message-ID: Dear all, I have recently upgraded ffmpeg to its latest from git and the ffmpeg segmenter stopped working. Here is my log: [root at enc6 t]# /usr/local/bin/ffmpeg -i hi.ts -map 0 -c copy -f segment -segment_time 10 -segment_list hi.m3u8 hi_%05d.ts ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 29 2013 08:50:28 with gcc 4.5.3 (GCC) 20120403 (ALT Linux 4.5.3-alt1.M60P.1) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-gpl --disable-postproc --enable-pthreads --enable-shared --disable-static --disable-libvorbis --disable-libfaac --disable-libxvid --enable-libx264 --disable-libmp3lame --disable-libgsm --disable-libdc1394 --disable-zlib --disable-memalign-hack --disable-ffserver --disable-ffplay --disable-libschroedinger --disable-avisynth --disable-libnut --enable-libtheora --enable-version3 --enable-libopencore-amrwb --enable-libopencore-amrnb --disable-ffprobe --disable-libvpx --disable-libv4l2 --enable-nonfree --enable-librtmp --enable-libfdk-aac --enable-libvo-aacenc --disable-debug --extra-cflags='-pipe -Wall -g -O2 -frename-registers -fPIC -DPIC' libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [mpegts @ 0x20dd1c0] max_analyze_duration 5000000 reached at 5013333 microseconds [mpegts @ 0x20dd1c0] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [NULL @ 0x20e1440] start time is not set in estimate_timings_from_pts Input #0, mpegts, from 'hi.ts': Duration: 00:09:59.80, start: 1.400000, bitrate: 782 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 ([27][0][0][0] / 0x001B), 90k tbr, 90k tbn, 180k tbc Stream #0:1[0x101](rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 102 kb/s [segment @ 0x20e2780] dimensions not set Output #0, segment, to 'hi_%05d.ts': Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), q=2-31, 90k tbn, 90k tbc Stream #0:1(rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, 102 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument Is this a regression in ffmpeg or am I missing something. Previous ffmpeg (1.1 build in November) is working fine with the same data. Thanks for your help, Rares From lou at lrcd.com Wed Apr 3 19:33:23 2013 From: lou at lrcd.com (Lou) Date: Wed, 3 Apr 2013 09:33:23 -0800 Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? In-Reply-To: References: Message-ID: <20130403093323.1e52e26b@lrcd.com> On Wed, 3 Apr 2013 17:47:02 +0200 (CEST) tag at gmx.com wrote: > Tom Evans wrote: > > You can use the -crf argument, which sets a quality setting for the > > final video, not a bitrate. I believe this works with libvpx. > > ffmpeg -i 12082012.mp4 -crf 10 12082012.mp4_crf10.webm > ffmpeg -i 12082012.mp4 -crf 20 12082012.mp4_crf20.webm > ffmpeg -i 12082012.mp4 -crf 40 12082012.mp4_crf40.webm > > -crf 10: Filesize 11716 KB > -crf 20: Filesize 11716 KB > -crf 40: Filesize 11716 KB > > Seems it does not work. Default 200 kb/s stays active all the time. > > Are there other options? See the vpx (WebM) Encoding Guide: https://ffmpeg.org/trac/ffmpeg/wiki/vpxEncodingGuide From mohancloudworld at gmail.com Wed Apr 3 20:24:44 2013 From: mohancloudworld at gmail.com (Mohan G) Date: Wed, 3 Apr 2013 13:24:44 -0500 Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? In-Reply-To: <20130403093323.1e52e26b@lrcd.com> References: <20130403093323.1e52e26b@lrcd.com> Message-ID: > Seems it does not work. Default 200 kb/s stays active all the time. > I use this, *ffmpeg -i inp.avi -codec:v libvpx -quality good -cpu-used 0 -qmin 10 -qmax 42 -threads 2 -codec:a libvorbis -q:a 10 -ac 2 out.webm * You can change the output quality with "-quality" option. -Mohan G From cehoyos at ag.or.at Wed Apr 3 20:26:03 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 3 Apr 2013 18:26:03 +0000 (UTC) Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter References: Message-ID: Rares Pop gmail.com> writes: > Is this a regression in ffmpeg or am I missing something. > Previous ffmpeg (1.1 build in November) is working fine > with the same data. Please provide the input sample hi.ts Thank you, Carl Eugen From cehoyos at ag.or.at Wed Apr 3 20:27:33 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 3 Apr 2013 18:27:33 +0000 (UTC) Subject: [FFmpeg-user] Remove 'anamorph' from video? References: Message-ID: Andrey Aleksandrovich googlemail.com> writes: > Is there way to remove 'aspect ratio' information from video? Did you already read the filter documentation? http://ffmpeg.org/ffmpeg-filters.html The setsar and setdar filters are probably what you are searching for. Carl Eugen From cehoyos at ag.or.at Wed Apr 3 20:31:04 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 3 Apr 2013 18:31:04 +0000 (UTC) Subject: [FFmpeg-user] adding an ac3 stream to an a/v stream References: <142598738.443731.1364683467582.open-xchange@email.1and1.com> <1516889934.548681.1364895164808.open-xchange@email.1and1.com> Message-ID: Andrew RT flight.us> writes: > WARNING: library configuration mismatch There is nothing obviously wrong in your command line and the only thing in the output are these warnings, consider recompiling without --enable-shared. What produces the static that your hear? ffplay? Carl Eugen From andrey.aleksandrovich at googlemail.com Wed Apr 3 20:53:59 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Wed, 3 Apr 2013 21:53:59 +0300 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: References: Message-ID: So maybe, but I don't understand why there is two values of SAR&DAR... What does mean the values in square brackets and outside them? On 4/3/13, Carl Eugen Hoyos wrote: > Andrey Aleksandrovich googlemail.com> writes: > >> Is there way to remove 'aspect ratio' information from video? > > Did you already read the filter documentation? > http://ffmpeg.org/ffmpeg-filters.html > The setsar and setdar filters are probably what > you are searching for. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From zenicanin at gmail.com Wed Apr 3 20:33:26 2013 From: zenicanin at gmail.com (zenicanin) Date: Wed, 3 Apr 2013 11:33:26 -0700 (PDT) Subject: [FFmpeg-user] Latest source doesn't support filter_complex? Message-ID: <1365014006480-4658260.post@n4.nabble.com> I just upgraded to the latest version of ffmpeg and keep getting Unrecognized option 'filter_complex'. Is this option really deprecated or is there some other problem? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Latest-source-doesn-t-support-filter-complex-tp4658260.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From francois.visagie at gmail.com Wed Apr 3 21:35:01 2013 From: francois.visagie at gmail.com (Francois Visagie) Date: Wed, 3 Apr 2013 21:35:01 +0200 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: References: Message-ID: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Andrey Aleksandrovich > Sent: 03 April 2013 20:54 > To: FFmpeg user questions > Subject: Re: [FFmpeg-user] Remove 'anamorph' from video? > > So maybe, but I don't understand why there is two values of SAR&DAR... > What does mean the values in square brackets and outside them? Here's a reference to get you started: http://en.wikipedia.org/wiki/Pixel_aspect_ratio Cheers, Francois > On 4/3/13, Carl Eugen Hoyos wrote: > > Andrey Aleksandrovich googlemail.com> > writes: > > > >> Is there way to remove 'aspect ratio' information from video? > > > > Did you already read the filter documentation? > > http://ffmpeg.org/ffmpeg-filters.html > > The setsar and setdar filters are probably what > > you are searching for. > > > > Carl Eugen > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From andrey.aleksandrovich at googlemail.com Wed Apr 3 21:58:23 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Wed, 3 Apr 2013 22:58:23 +0300 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> References: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> Message-ID: No. The question was: Why there's two values of PAR and DAR - one is inside of square brackets and another is outside? ---cut--- 720x390 [SAR 1:1 DAR 24:13], SAR 403:490 DAR 372:245 ---cut--- On 4/3/13, Francois Visagie wrote: >> -----Original Message----- >> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- >> bounces at ffmpeg.org] On Behalf Of Andrey Aleksandrovich >> Sent: 03 April 2013 20:54 >> To: FFmpeg user questions >> Subject: Re: [FFmpeg-user] Remove 'anamorph' from video? >> >> So maybe, but I don't understand why there is two values of SAR&DAR... >> What does mean the values in square brackets and outside them? > > Here's a reference to get you started: > http://en.wikipedia.org/wiki/Pixel_aspect_ratio > > Cheers, > Francois > >> On 4/3/13, Carl Eugen Hoyos wrote: >> > Andrey Aleksandrovich googlemail.com> >> writes: >> > >> >> Is there way to remove 'aspect ratio' information from video? >> > >> > Did you already read the filter documentation? >> > http://ffmpeg.org/ffmpeg-filters.html >> > The setsar and setdar filters are probably what >> > you are searching for. >> > >> > Carl Eugen >> > >> > _______________________________________________ >> > ffmpeg-user mailing list >> > ffmpeg-user at ffmpeg.org >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lou at lrcd.com Wed Apr 3 22:25:16 2013 From: lou at lrcd.com (Lou) Date: Wed, 3 Apr 2013 12:25:16 -0800 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <5159A613.9010409@thelounge.net> References: <5159A613.9010409@thelounge.net> Message-ID: <20130403122516.4cde565e@lrcd.com> On Mon, 01 Apr 2013 17:21:55 +0200 Reindl Harald wrote: > am i really the only one receiving this randomly? > > i get a lot of mailing-lists and around 800 messages > per day and this is the only list with such behavior > > surely the list-software is unable to distinct between > a bounce and a reject at all, but as long this messages > does not give any basic information WHICH message was > "bounced" like date, time, subject. message-id it is > impossible to do anything, even for me as admin of the > mailserver and so i can only guess that some messages > are rejected as spam from the MX > > -------- Original-Nachricht -------- > Betreff: confirm a58e0237ee6751f7ce38287d432088783dcc782d > Datum: Mon, 01 Apr 2013 17:13:49 +0200 > Von: ffmpeg-user-request at ffmpeg.org > An: h.reindl at thelounge.net > > Your membership in the mailing list ffmpeg-user has been disabled due > to excessive bounces The last bounce received from you was dated > 01-Apr-2013. You will not get any more messages from this list until > you re-enable your membership. You will receive 3 more reminders like > this before your membership in the list is deleted > Here is a log message (Mar 5 14:07:26, Mar 10 18:03:47, Apr 3 04:27:02): to=, relay=barracuda.thelounge.net[91.118.73.20]:25, delay=8.2, delays=0.04/0.23/6.8/1.2, dsn=5.0.0, status=bounced (host barracuda.thelounge.net[91.118.73.20] said: 554 rejected due to spam content (in reply to end of DATA command)) I'm guessing you have anti-spam measures of questionable effectiveness and are bouncing messages back (which is not a recommended behavior). From onemda at gmail.com Wed Apr 3 22:35:55 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 3 Apr 2013 20:35:55 +0000 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: References: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> Message-ID: On 4/3/13, Andrey Aleksandrovich wrote: > No. The question was: Why there's two values of PAR and DAR - one is > inside of square brackets and another is outside? PAR - pixel aspect ratio DAR - display aspect ratio From h.reindl at thelounge.net Thu Apr 4 00:07:17 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 00:07:17 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130403122516.4cde565e@lrcd.com> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> Message-ID: <515CA815.2030907@thelounge.net> Am 03.04.2013 22:25, schrieb Lou: > On Mon, 01 Apr 2013 17:21:55 +0200 > Reindl Harald wrote: > >> am i really the only one receiving this randomly? >> >> i get a lot of mailing-lists and around 800 messages >> per day and this is the only list with such behavior >> >> surely the list-software is unable to distinct between >> a bounce and a reject at all, but as long this messages >> does not give any basic information WHICH message was >> "bounced" like date, time, subject. message-id it is >> impossible to do anything, even for me as admin of the >> mailserver and so i can only guess that some messages >> are rejected as spam from the MX >> >> -------- Original-Nachricht -------- >> Betreff: confirm a58e0237ee6751f7ce38287d432088783dcc782d >> Datum: Mon, 01 Apr 2013 17:13:49 +0200 >> Von: ffmpeg-user-request at ffmpeg.org >> An: h.reindl at thelounge.net >> >> Your membership in the mailing list ffmpeg-user has been disabled due >> to excessive bounces The last bounce received from you was dated >> 01-Apr-2013. You will not get any more messages from this list until >> you re-enable your membership. You will receive 3 more reminders like >> this before your membership in the list is deleted >> > > Here is a log message (Mar 5 14:07:26, Mar 10 18:03:47, Apr 3 04:27:02): > > to=, > relay=barracuda.thelounge.net[91.118.73.20]:25, delay=8.2, > delays=0.04/0.23/6.8/1.2, dsn=5.0.0, status=bounced (host > barracuda.thelounge.net[91.118.73.20] said: 554 rejected due to spam > content (in reply to end of DATA command)) > > I'm guessing you have anti-spam measures of questionable effectiveness > and are bouncing messages back (which is not a recommended behavior) don't get me wrong but you should learn to read bounces/maillog relay=barracuda.thelounge.net[91.118.73.20]:25 said: 554 rejected due to spam IS A REJECT AND NOT A BOUNCE IF SOMETHING IN YOUR INFRASTRUCTURE BOUNCES AFTER YOU SMTP-CLIENT GET A REJECT THIS IS NOT A PROBLEM OF OUR MX AND YOUR CONFIG ERROR -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From andrew2012 at flight.us Thu Apr 4 00:35:20 2013 From: andrew2012 at flight.us (Andrew RT) Date: Wed, 3 Apr 2013 18:35:20 -0400 (EDT) Subject: [FFmpeg-user] adding an ac3 stream to an a/v stream In-Reply-To: References: <142598738.443731.1364683467582.open-xchange@email.1and1.com> <1516889934.548681.1364895164808.open-xchange@email.1and1.com> Message-ID: <648146021.636849.1365028520737.open-xchange@email.1and1.com> > On April 3, 2013 at 2:31 PM Carl Eugen Hoyos wrote: > > > Andrew RT flight.us> writes: > > > WARNING: library configuration mismatch > > There is nothing obviously wrong in your command line > and the only thing in the output are these warnings, > consider recompiling without --enable-shared. > > What produces the static that your hear? ffplay? > mplayer produces static. mplayer plays the solitary extra audio file just fine. It also plays the a/v avi file fine. I tried ffplay; mine is silent, for any file I try (a/v files, this one and others, and this external ac3 audio track. I don't know how to operate ffplay, as I haven't been using it.) I guess I'll try to recompile without the *.so's. see mplayer, vlc, ffplay output below. (mplayer often spits out errors even if it plays everything OK). This is a truncated muxed file, but I think it's same as a full one (unless ffmpeg does something special at end of encoding. I *HAVE* tried going all the way to the end, the result is what I described) ------------------------------ mplayer: ----------------------------- Playing muxed_output.avi. AVI file format detected. [aviheader] Video stream found, -vid 0 [aviheader] Audio stream found, -aid 1 [aviheader] Audio stream found, -aid 2 AVI Ogg : Initial audio header is too small !!!!! VIDEO: [XVID] 720x304 24bpp 23.976 fps 1061.3 kbps (129.6 kbyte/s) Clip info: Software: Lavf55.0.100 Failed to open VDPAU backend libvdpau_nvidia.so: cannot open shared object file: No such file or directory [vdpau] Error when calling vdp_device_create_x11: 1 ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 48000 Hz, 6 ch, s16le, 320.0 kbit/6.94% (ratio: 40000->576000) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== AO: [jack] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 2.37:1 - prescaling to correct movie aspect. VO: [xv] 720x304 => 720x304 Planar YV12 A: 21.5 V: 1.2 A-V: 20.327 ct: 0.000 29/ 29 9% 5% 12.0% 1 0 A: 45.7 V: 28.0 A-V: 17.712 ct: 0.100 672/672 4% 0% 9.0% 1 0 -------------------------- vlc: ------------------------------------- [0x94e7668] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface. [0x986cce8] main decoder error: no suitable decoder module for fourcc `undf'. VLC probably does not support this sound or video format. [0x9882ab0] main decoder error: no suitable decoder module for fourcc `undf'. VLC probably does not support this sound or video format. QPainter::begin: Paint device returned engine == 0, type: 1 QPainter::begin: Paint device returned engine == 0, type: 1 ------------------------- ffplay: ----------------------------------------- --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared postproc configuration: --enable-version3 --enable-nonfree --enable-libtheora --enable-gpl --enable-postproc --enable-libvorbis --enable-libgsm --enable-libmp3lame --enable-libfaac --enable-libxvid --enable-pthreads --enable-libx264 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-shared libavutil 52. 22.101 / 52. 22.101 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 0.100 / 55. 0.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, avi, from 'muxed_output.avi':vq= 0KB sq= 0B f=0/0 Metadata: encoder : Lavf55.0.100 Duration: 00:16:26.34, start: 0.000000, bitrate: 1844 kb/s Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 720x304 [SAR 1:1 DAR 45:19], 23.98 tbr, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 320 kb/s Stream #0:2: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/s > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lou at lrcd.com Thu Apr 4 00:36:12 2013 From: lou at lrcd.com (Lou) Date: Wed, 3 Apr 2013 14:36:12 -0800 Subject: [FFmpeg-user] Latest source doesn't support filter_complex? In-Reply-To: <1365014006480-4658260.post@n4.nabble.com> References: <1365014006480-4658260.post@n4.nabble.com> Message-ID: <20130403143612.68e4e471@lrcd.com> On Wed, 3 Apr 2013 11:33:26 -0700 (PDT) zenicanin wrote: > I just upgraded to the latest version of ffmpeg and keep getting Unrecognized > option 'filter_complex'. Is this option really deprecated or is there some > other problem? Please show your ffmpeg command and the complete console output. From cehoyos at ag.or.at Thu Apr 4 00:43:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 3 Apr 2013 22:43:22 +0000 (UTC) Subject: [FFmpeg-user] adding an ac3 stream to an a/v stream References: <142598738.443731.1364683467582.open-xchange@email.1and1.com> <1516889934.548681.1364895164808.open-xchange@email.1and1.com> Message-ID: Andrew RT flight.us> writes: > ffmpeg -i myvideo.avi -i extra_audio.ac3 Please provide the file extra_audio.ac3 - I currently don't understand what should be wrong... Carl Eugen From andrey.aleksandrovich at googlemail.com Thu Apr 4 07:09:03 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Thu, 4 Apr 2013 08:09:03 +0300 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: References: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> Message-ID: :facepalm: --- 720x390 [SAR 1:1 DAR 24:13], SAR 403:490 DAR 372:245 --- What does mean SAR 1:1 and SAR 403:490? Why there's two values? What is the real SAR? (Similarly about DAR). On 4/3/13, Paul B Mahol wrote: > On 4/3/13, Andrey Aleksandrovich > wrote: >> No. The question was: Why there's two values of PAR and DAR - one is >> inside of square brackets and another is outside? > > PAR - pixel aspect ratio > DAR - display aspect ratio > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From francois.visagie at gmail.com Thu Apr 4 07:55:14 2013 From: francois.visagie at gmail.com (Francois Visagie) Date: Thu, 4 Apr 2013 07:55:14 +0200 Subject: [FFmpeg-user] Remove 'anamorph' from video? In-Reply-To: References: <01c001ce30a2$5850ab40$08f201c0$@gmail.com> Message-ID: <001201ce30f8$fc9a8210$f5cf8630$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Andrey Aleksandrovich > Sent: 04 April 2013 07:09 > To: FFmpeg user questions > Subject: Re: [FFmpeg-user] Remove 'anamorph' from video? > > :facepalm: > --- 720x390 [SAR 1:1 DAR 24:13], SAR 403:490 DAR 372:245 --- What does > mean SAR 1:1 and SAR 403:490? Why there's two values? What is the real > SAR? > (Similarly about DAR). I understand now what you mean. You may be able to get close to the truth yourself by calculating from known values. The bracketed set of values seem to indicate the aspect ratio represented by resolution at square pixels, and the 2nd set after applying the encoded aspect ratio. Use something like MediaInfo to independently report resolution and encoded aspect ratio and see what you come up with. > > On 4/3/13, Paul B Mahol wrote: > > On 4/3/13, Andrey Aleksandrovich > > > > wrote: > >> No. The question was: Why there's two values of PAR and DAR - one is > >> inside of square brackets and another is outside? > > > > PAR - pixel aspect ratio > > DAR - display aspect ratio > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From classysnet at gmail.com Thu Apr 4 10:42:09 2013 From: classysnet at gmail.com (Rares Pop) Date: Thu, 4 Apr 2013 11:42:09 +0300 Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter In-Reply-To: References: Message-ID: Carl, thanks for looking at this. Here is the smallest I could find: https://dl.dropbox.com/u/12255037/hi.ts Thanks again, Rares On Wed, Apr 3, 2013 at 9:26 PM, Carl Eugen Hoyos wrote: > Rares Pop gmail.com> writes: > > > Is this a regression in ffmpeg or am I missing something. > > Previous ffmpeg (1.1 build in November) is working fine > > with the same data. > > Please provide the input sample hi.ts > > Thank you, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Thu Apr 4 11:39:35 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 4 Apr 2013 09:39:35 +0000 (UTC) Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter References: Message-ID: Rares Pop gmail.com> writes: > [mpegts 0x20dd1c0] max_analyze_duration 5000000 reached > at 5013333 microseconds > [mpegts 0x20dd1c0] Could not find codec parameters for > stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size > Consider increasing the value for the 'analyzeduration' and > 'probesize' options Works fine with -analyzeduration 6M > Is this a regression in ffmpeg or am I missing something. I wasn't able to find a version that sees the video without -analyzeduration. Do you know what command line produced the file? > Previous ffmpeg (1.1 build in November) is working fine > with the same data. Doesn't work here: $ ffmpeg -i hi.ts ffmpeg version 1.1 Copyright (c) 2000-2013 the FFmpeg developers built on Feb 10 2013 11:46:21 with gcc 4.7 (SUSE Linux) configuration: --enable-gpl libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 86.100 / 54. 86.100 libavformat 54. 59.106 / 54. 59.106 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mpegts @ 0x167f540] max_analyze_duration 5000000 reached at 5013333 [mpegts @ 0x167f540] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [NULL @ 0x1683740] start time is not set in estimate_timings_from_pts Input #0, mpegts, from 'hi.ts': Duration: 00:09:59.80, start: 1.400000, bitrate: 782 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 ([27][0][0][0] / 0x001B), 90k tbr, 90k tbn, 180k tbc Stream #0:1[0x101](rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 102 kb/s At least one output file must be specified Carl Eugen From cehoyos at ag.or.at Thu Apr 4 11:41:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 4 Apr 2013 09:41:15 +0000 (UTC) Subject: [FFmpeg-user] Remove 'anamorph' from video? References: Message-ID: Andrey Aleksandrovich googlemail.com> writes: > So maybe, but I don't understand why there is two > values of SAR&DAR... > What does mean the values in square brackets and > outside them? Video stream and container may specify different values for SAR and DAR. Please do not top-post here, it is considered rude. Carl Eugen From mike at redtux.org.uk Thu Apr 4 16:11:32 2013 From: mike at redtux.org.uk (Mike Martin) Date: Thu, 4 Apr 2013 15:11:32 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module Message-ID: Hi Has anyone had any joy with building this against proper ffmpeg and not the obnoxious fork which gstreamer seems to have gone with thanks From av at bsbc.nb.ca Thu Apr 4 16:37:15 2013 From: av at bsbc.nb.ca (Anthony Brown) Date: Thu, 04 Apr 2013 11:37:15 -0300 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130403122516.4cde565e@lrcd.com> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> Message-ID: <515D901B.8090104@bsbc.nb.ca> On 13-04-03 05:25 PM, Lou wrote: > On Mon, 01 Apr 2013 17:21:55 +0200 > Reindl Harald wrote: > >> am i really the only one receiving this randomly? I've had a couple of these in the past few months as well. I certainly don't run anything that would bounce spam back, but I suppose it's possible my ISP does.... Anthony -- Anthony Brown Audiovisual coordinator Brunswick Street Baptist Church Telephone: (506)-458-8348 (leave message) Email: av at bsbc.nb.ca -------------- next part -------------- A non-text attachment was scrubbed... Name: av.vcf Type: text/x-vcard Size: 163 bytes Desc: not available URL: From h.reindl at thelounge.net Thu Apr 4 16:42:22 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 16:42:22 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <515D901B.8090104@bsbc.nb.ca> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> Message-ID: <515D914E.6080508@thelounge.net> Am 04.04.2013 16:37, schrieb Anthony Brown: > On 13-04-03 05:25 PM, Lou wrote: >> On Mon, 01 Apr 2013 17:21:55 +0200 >> Reindl Harald wrote: >> >>> am i really the only one receiving this randomly? > > I've had a couple of these in the past few months as well. I certainly don't run anything that would bounce spam > back, but I suppose it's possible my ISP does.... looking at the reply yesterday our spam-firewall rejected some messages as spam - REJECTED, this is not the same as bounce and a REJHECT with 554 is NOT a reason for disable a account -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Thu Apr 4 16:48:51 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 4 Apr 2013 16:48:51 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <515D914E.6080508@thelounge.net> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> Message-ID: <20130404144851.GA1660@phare.normalesup.org> Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit?: > looking at the reply yesterday our spam-firewall rejected some > messages as spam - REJECTED, this is not the same as bounce > and a REJHECT with 554 is NOT a reason for disable a account If a MTA rejects a mail, the previous MTA is supposed to generate a bounce. So the distinction between a reject and a bounce is akin to the excuse "I did not punch him, I just held my fist at face level and he ran in it". The only time the distinction is relevant is when the return address may have been fabricated, but that is not the case here. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From andrey.krieger.utkin at gmail.com Thu Apr 4 16:54:02 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 4 Apr 2013 17:54:02 +0300 Subject: [FFmpeg-user] Migration from avcodec_open to avcodec_open2 In-Reply-To: References: Message-ID: 2013/4/3 ?????? ?????? : > Where I can see the list of codec context and codec specific options, it's > description and text equivalent for dictionary? libavcodec/options_table.h - common options. For codec-specific options, look for AVOption array in specific codec source code files. -- Andrey Utkin From andrey.krieger.utkin at gmail.com Thu Apr 4 16:55:28 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Thu, 4 Apr 2013 17:55:28 +0300 Subject: [FFmpeg-user] Migration from avcodec_open to avcodec_open2 In-Reply-To: References: Message-ID: Also `ffmpeg -h full` will print it for you. -- Andrey Utkin From h.reindl at thelounge.net Thu Apr 4 17:06:07 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 17:06:07 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130404144851.GA1660@phare.normalesup.org> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> Message-ID: <515D96DF.3070403@thelounge.net> Am 04.04.2013 16:48, schrieb Nicolas George: > Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit : >> looking at the reply yesterday our spam-firewall rejected some >> messages as spam - REJECTED, this is not the same as bounce >> and a REJHECT with 554 is NOT a reason for disable a account > > If a MTA rejects a mail, the previous MTA is supposed to generate a bounce. > So the distinction between a reject and a bounce is akin to the excuse "I > did not punch him, I just held my fist at face level and he ran in it". The > only time the distinction is relevant is when the return address may have > been fabricated, but that is not the case here as my job is mail-admin and software-developer including development of admin-backends for mailsystems and bounce-management systems i am able to read a maillog line 554 "in reply to end of DATA command" is NOT a valid reason to disable a account, this would only be valid "in reply to RCPT TO command" which indicated that the RCPT address does not accept email (host barracuda.thelounge.net[91.118.73.20] said: 554 rejected due to spam content (in reply to end of DATA command)) and as long as i have no information of the specific rejected messages in the suspend-notify it is impossible for me to observe the reason why a message was blocked and mark it as ham to train the filter http://www.faqs.org/rfcs/rfc2476.html Unless covered by a more precise response code, response code 554 is to be used to reject a MAIL FROM, RCPT TO, or DATA command that contains something improper. Enhanced status code 5.6.0 is to be used if no other code is more specific. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Thu Apr 4 17:13:50 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 4 Apr 2013 17:13:50 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <515D96DF.3070403@thelounge.net> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> Message-ID: <20130404151349.GA8686@phare.normalesup.org> Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit?: > 554 "in reply to end of DATA command" is NOT a valid reason to disable > a account, this would only be valid "in reply to RCPT TO command" which > indicated that the RCPT address does not accept email Says who? > http://www.faqs.org/rfcs/rfc2476.html Obsolete and irrelevant, since it is about message submission, and this is relaying. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From sorcchan at gmail.com Thu Apr 4 17:16:24 2013 From: sorcchan at gmail.com (=?UTF-8?B?0KPRiNCw0LrQvtCyINCQ0L3QtNGA0LXQuQ==?=) Date: Thu, 4 Apr 2013 19:16:24 +0400 Subject: [FFmpeg-user] Migration from avcodec_open to avcodec_open2 Message-ID: Thank you very much! Exactly what I looking for. 2013/4/4 Andrey Utkin > 2013/4/3 ?????? ?????? : > > Where I can see the list of codec context and codec specific options, > it's > > description and text equivalent for dictionary? > > libavcodec/options_table.h - common options. > For codec-specific options, look for AVOption array in specific codec > source code files. > > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From h.reindl at thelounge.net Thu Apr 4 17:21:33 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 17:21:33 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130404151349.GA8686@phare.normalesup.org> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> Message-ID: <515D9A7D.2010009@thelounge.net> Am 04.04.2013 17:13, schrieb Nicolas George: > Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit : >> 554 "in reply to end of DATA command" is NOT a valid reason to disable >> a account, this would only be valid "in reply to RCPT TO command" which >> indicated that the RCPT address does not accept email > > Says who? anybody who can read and understand that the message was rejected because it's specific content and not because the address does not exist, and yes this can be parsed easily from the logfiles, been there, done that and doing that daily to unsubscribe customer newsletters while "in reply to end of DATA command" is on the strict "do never unsubscribe in this case"-list -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Thu Apr 4 17:24:59 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 4 Apr 2013 17:24:59 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <515D9A7D.2010009@thelounge.net> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> <515D9A7D.2010009@thelounge.net> Message-ID: <20130404152459.GA12333@phare.normalesup.org> Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit?: > anybody who can read and understand that the message was > rejected because it's specific content and not because > the address does not exist I did not ask why the message was rejected, I asked what principle allows you to deduce it is not a valid reason to disable a recipient. For me, "this guy has a stupid spam filter that rejects too many of our mails, let him fix it" is a perfectly valid reason to disable a recipient on a mailing list. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From h.reindl at thelounge.net Thu Apr 4 17:37:14 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 17:37:14 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130404152459.GA12333@phare.normalesup.org> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> <515D9A7D.2010009@thelounge.net> <20130404152459.GA12333@phare.normalesup.org> Message-ID: <515D9E2A.9030503@thelounge.net> Am 04.04.2013 17:24, schrieb Nicolas George: > Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit : >> anybody who can read and understand that the message was >> rejected because it's specific content and not because >> the address does not exist > > I did not ask why the message was rejected, I asked what principle allows > you to deduce it is not a valid reason to disable a recipient. > > For me, "this guy has a stupid spam filter that rejects too many of our > mails, let him fix it" is a perfectly valid reason to disable a recipient on > a mailing list. hahaha - show me how to do with no hint about the specific message and since this is the only mailing-list of around 30 which acts this way this is all a bit strange the reason for detect messages is often that posters are not willing to clean up their messages to a readable form and include all sorts of repeatet footers, disclaimers and what else exists -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Thu Apr 4 17:41:29 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 4 Apr 2013 17:41:29 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <515D9E2A.9030503@thelounge.net> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> <515D9A7D.2010009@thelounge.net> <20130404152459.GA12333@phare.normalesup.org> <515D9E2A.9030503@thelounge.net> Message-ID: <20130404154129.GC16550@phare.normalesup.org> Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit?: > hahaha - show me how to do with no hint about the specific message > and since this is the only mailing-list of around 30 which acts > this way this is all a bit strange > > the reason for detect messages is often that posters are not willing > to clean up their messages to a readable form and include all sorts > of repeatet footers, disclaimers and what else exists The only thing I did manage to parse in this message is that you have decided it was not your fault and you will not change your mind. No point in discussing it further. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From milu71 at gmx.de Thu Apr 4 18:34:44 2013 From: milu71 at gmx.de (Michael Ludwig) Date: Thu, 4 Apr 2013 18:34:44 +0200 Subject: [FFmpeg-user] Docs, filtering introduction example Message-ID: <20130404163444.GD4840@Dago> http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction There's an example to show how filters work. The description is: "The result will be that in output the top half of the video is mirrored onto the bottom half." The command is: ffmpeg -i input -vf "[in] split [T1], [T2] overlay=0:H/2 [out]; [T1] crop=iw:ih/2:0:ih/2, vflip [T2]" output I'm getting an error running that with ffmpeg 1.1.3, verified for three different input sizes: 320x180: Overlay area (0,45)<->(320,225) not within the main area (0,0)<->(320,90) or zero-sized 480x270: Overlay area (0,67)<->(480,337) not within the main area (0,0)<->(480,134) or zero-sized 240x134: Overlay area (0,33)<->(240,167) not within the main area (0,0)<->(240,66) or zero-sized See complete output at the end of this mail. (Okay, no error with N-51511-g599866f from 2013-04-01, but not the desired result either.) The following filter works for me and produces a horizontally mirrored video, which I think is the objective of the example: -vf "[in] split [T1], crop=iw:ih/2:0:0, [T2] overlay [out]; [T1] vflip [T2]" This puts the vflipped video as the background and overlays the cropped top part of the video so it is visible twice, cropped in the top half, and vflipped in the bottom half. Hope I didn't misunderstand completely ? but here's my two cents: Two things made the original example a bit hard to understand for me: (1) One of the two outputs of the split filter is named, but the other one is not. (2) One of the inputs of the overlay filter is unnamed, and it's not clear whether that input comes first or second. Maybe this could be pointed out, given that it's the introduction, to make it a bit easier for newbies ? Okay, I just tested with a newer version N-51511-g599866f ? http://ffmpeg.zeranoe.com/builds/win64/shared/ffmpeg-20130401-git-599866f-win64-shared.7z ? which does not error out, but does not produce the mirrored video either: [Parsed_overlay_1 @ 0000000000313660] Overlay area with coordinates x1:0 y1:33 x2:240 y2:167 is not completely contained within the output with size 240x66 My modified example works correctly with both versions, by the way, so if you think it's correct and illustrative enough maybe substitute it to the original example. Sorry for this long post. Michael PS: Here's the complete uncut console output of running the command from the doc page: :: ffmpeg -i m2.mp4 -vf "[in] split [T1], [T2] overlay=0:H/2 [out]; [T1] crop=iw:ih/2:0:ih/2, vflip [T2]" out.mp4 -y > e.txt 2>&1 ==== For the older version: ffmpeg version 1.1.3 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 3 2013 02:04:56 with gcc 4.7.2 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 86.100 / 54. 86.100 libavformat 54. 59.106 / 54. 59.106 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'm2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.59.106 Duration: 00:00:49.13, start: 0.000000, bitrate: 125 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 122 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc Metadata: handler_name : VideoHandler [Parsed_overlay_1 @ 00000000005637c0] Overlay area (0,45)<->(320,225) not within the main area (0,0)<->(320,90) or zero-sized [auto-inserted scaler 0 @ 0000000000563460] Failed to configure input pad on Parsed_overlay_1 Error opening filters! ==== For the newer version: ffmpeg version N-51511-g599866f Copyright (c) 2000-2013 the FFmpeg developers built on Apr 1 2013 12:52:07 with gcc 4.8.0 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 24.100 / 52. 24.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 48.105 / 3. 48.105 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'm2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.59.106 Duration: 00:00:49.13, start: 0.000000, bitrate: 125 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 122 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc Metadata: handler_name : VideoHandler [Parsed_overlay_1 @ 00000000003136c0] Overlay area with coordinates x1:0 y1:45 x2:320 y2:225 is not completely contained within the output with size 320x90 [libx264 @ 00000000003dfb20] using SAR=1/1 [libx264 @ 00000000003dfb20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 [libx264 @ 00000000003dfb20] profile High, level 1.2 [libx264 @ 00000000003dfb20] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'out.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.1.100 Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 320x90 [SAR 1:1 DAR 32:9], q=-1--1, 15360 tbn, 30 tbc Metadata: handler_name : VideoHandler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Press [q] to stop, [?] for help frame= 340 fps=0.0 q=29.0 size= 46kB time=00:00:09.60 bitrate= 38.9kbits/s frame= 568 fps=552 q=29.0 size= 124kB time=00:00:17.20 bitrate= 58.9kbits/s frame= 845 fps=547 q=29.0 size= 179kB time=00:00:26.43 bitrate= 55.6kbits/s frame= 1072 fps=521 q=29.0 size= 258kB time=00:00:34.00 bitrate= 62.3kbits/s frame= 1335 fps=519 q=29.0 size= 324kB time=00:00:42.76 bitrate= 62.1kbits/s frame= 1474 fps=497 q=-1.0 Lsize= 403kB time=00:00:49.06 bitrate= 67.3kbits/s video:386kB audio:0kB subtitle:0 global headers:0kB muxing overhead 4.606866% [libx264 @ 00000000003dfb20] frame I:9 Avg QP:19.49 size: 2054 [libx264 @ 00000000003dfb20] frame P:503 Avg QP:22.40 size: 583 [libx264 @ 00000000003dfb20] frame B:962 Avg QP:26.56 size: 86 [libx264 @ 00000000003dfb20] consecutive B-frames: 5.4% 18.5% 12.6% 63.5% [libx264 @ 00000000003dfb20] mb I I16..4: 20.5% 51.7% 27.9% [libx264 @ 00000000003dfb20] mb P I16..4: 2.9% 2.1% 0.8% P16..4: 49.1% 16.7% 7.6% 0.0% 0.0% skip:20.8% [libx264 @ 00000000003dfb20] mb B I16..4: 0.1% 0.0% 0.0% B16..8: 34.6% 2.0% 0.2% direct: 0.8% skip:62.4% L0:41.1% L1:55.0% BI: 3.9% [libx264 @ 00000000003dfb20] 8x8 transform intra:39.7% inter:69.6% [libx264 @ 00000000003dfb20] coded y,uvDC,uvAC intra: 33.8% 78.1% 42.1% inter: 7.6% 18.8% 1.9% [libx264 @ 00000000003dfb20] i16 v,h,dc,p: 38% 43% 12% 7% [libx264 @ 00000000003dfb20] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 31% 35% 2% 2% 2% 2% 2% 4% [libx264 @ 00000000003dfb20] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 43% 17% 2% 4% 2% 5% 2% 5% [libx264 @ 00000000003dfb20] i8c dc,h,v,p: 46% 37% 14% 3% [libx264 @ 00000000003dfb20] Weighted P-Frames: Y:15.9% UV:10.5% [libx264 @ 00000000003dfb20] ref P L0: 62.1% 14.1% 16.6% 6.7% 0.5% [libx264 @ 00000000003dfb20] ref B L0: 86.2% 11.4% 2.5% [libx264 @ 00000000003dfb20] ref B L1: 92.9% 7.1% [libx264 @ 00000000003dfb20] kb/s:64.16 From h.reindl at thelounge.net Thu Apr 4 19:08:01 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 19:08:01 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130404154129.GC16550@phare.normalesup.org> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> <515D9A7D.2010009@thelounge.net> <20130404152459.GA12333@phare.normalesup.org> <515D9E2A.9030503@thelounge.net> <20130404154129.GC16550@phare.normalesup.org> Message-ID: <515DB371.3080909@thelounge.net> Am 04.04.2013 17:41, schrieb Nicolas George: > Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit : >> hahaha - show me how to do with no hint about the specific message >> and since this is the only mailing-list of around 30 which acts >> this way this is all a bit strange >> >> the reason for detect messages is often that posters are not willing >> to clean up their messages to a readable form and include all sorts >> of repeatet footers, disclaimers and what else exists > > The only thing I did manage to parse in this message is that you have > decided it was not your fault and you will not change your mind. No point in > discussing it further. what exactly did you not understand in the fact that it is impossible to mark specific messages as "not spam" as long you have ZEROP information about the rejected ones in this "membership has been disabled" message? "ffmpeg-user-bounces at ffmpeg.org" is white-listed onthe spam-firewall since my subscription as other lists to, but NOT for ZeroHour Intents if people send messages with http://www.datafilehost.com/download-4aa3f9cf.html which are blocked by "ZeroHour Intent (datafilehost.com)" because there are thousands of spam messages with links to the same domain it can happen that single messages are blocked why is NO REASON the suspend a account -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From h.reindl at thelounge.net Thu Apr 4 19:11:24 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 04 Apr 2013 19:11:24 +0200 Subject: [FFmpeg-user] Fwd: Your membership in the mailing list ffmpeg-user has been disabled due, to excessive bounces In-Reply-To: <20130404154129.GC16550@phare.normalesup.org> References: <5159A613.9010409@thelounge.net> <20130403122516.4cde565e@lrcd.com> <515D901B.8090104@bsbc.nb.ca> <515D914E.6080508@thelounge.net> <20130404144851.GA1660@phare.normalesup.org> <515D96DF.3070403@thelounge.net> <20130404151349.GA8686@phare.normalesup.org> <515D9A7D.2010009@thelounge.net> <20130404152459.GA12333@phare.normalesup.org> <515D9E2A.9030503@thelounge.net> <20130404154129.GC16550@phare.normalesup.org> Message-ID: <515DB43C.3060701@thelounge.net> Am 04.04.2013 17:41, schrieb Nicolas George: > Le quintidi 15 germinal, an CCXXI, Reindl Harald a ?crit : >> hahaha - show me how to do with no hint about the specific message >> and since this is the only mailing-list of around 30 which acts >> this way this is all a bit strange >> >> the reason for detect messages is often that posters are not willing >> to clean up their messages to a readable form and include all sorts >> of repeatet footers, disclaimers and what else exists > > The only thing I did manage to parse in this message is that you have > decided it was not your fault and you will not change your mind. No point in > discussing it further. and by the way "has been disabled due to excessive bounces" BULLSHIT! EXACTLY THREE MESSAGES since 2013-04-01 until now and ALL because "ZeroHour Intent (datafilehost.com)" well, english is not my native language but my definition of "excessive" is different -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From cehoyos at ag.or.at Thu Apr 4 19:48:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 4 Apr 2013 17:48:15 +0000 (UTC) Subject: [FFmpeg-user] Sync troubles with input pipes from raw -> AAC/Scaling/H.264 -> live RTMP References: <6AA7CF67-52FD-4CF0-8B74-A2B3144924FA@hfink.eu> Message-ID: Heinrich Fink hfink.eu> writes: > I ran a test for about 3 hours, recording into a flv file, > and I was able to observe a A/V shift of 4 frames. Is the shift not reproducible on the first frames of the output file, ie are you sure it is something that accumulates during the encode? Does it also happen with another audio encoder? (I believe it is very unlikely that the video pts are wrong, it at least would be easy to verify, there is -debug_ts.) Carl Eugen From cehoyos at ag.or.at Thu Apr 4 22:14:27 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 4 Apr 2013 20:14:27 +0000 (UTC) Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module References: Message-ID: Mike Martin redtux.org.uk> writes: > Has anyone had any joy with building this against > proper ffmpeg and not the obnoxious fork which > gstreamer seems to have gone with (Sorry if this is unrelated, I don't know much about GStreamer) A patch was posted that allows to build a GStreamer module with FFmpeg: https://bugzilla.gnome.org/show_bug.cgi?id=694937 Testing the patch and reporting on gnome bugzilla is probably very welcome! Carl Eugen From cehoyos at ag.or.at Thu Apr 4 22:30:01 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 4 Apr 2013 20:30:01 +0000 (UTC) Subject: [FFmpeg-user] Docs, filtering introduction example References: <20130404163444.GD4840@Dago> Message-ID: Michael Ludwig gmx.de> writes: > I'm getting an error running that with ffmpeg 1.1.3, > verified for three different input sizes: (I hope you understand that it does not make much sense to compare the current documentation with an old release.) [...] > (Okay, no error with N-51511-g599866f from 2013-04-01, > but not the desired result either.) This is ticket #2413, a patch is currently discussed on ffmpeg-devel. Further improvements are of course welcome! Carl Eugen From zenicanin at gmail.com Thu Apr 4 19:27:22 2013 From: zenicanin at gmail.com (zenicanin) Date: Thu, 4 Apr 2013 10:27:22 -0700 (PDT) Subject: [FFmpeg-user] Latest source doesn't support filter_complex? In-Reply-To: <20130403143612.68e4e471@lrcd.com> References: <1365014006480-4658260.post@n4.nabble.com> <20130403143612.68e4e471@lrcd.com> Message-ID: <1365096442991-4658292.post@n4.nabble.com> Sorry nevermind. It turns out I didn't have libmp3lame module in there. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Latest-source-doesn-t-support-filter-complex-tp4658260p4658292.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From milu71 at gmx.de Fri Apr 5 10:45:37 2013 From: milu71 at gmx.de (Michael Ludwig) Date: Fri, 5 Apr 2013 10:45:37 +0200 Subject: [FFmpeg-user] Docs, filtering introduction example In-Reply-To: References: <20130404163444.GD4840@Dago> Message-ID: <20130405084537.GF4840@Dago> Carl Eugen Hoyos schrieb am 04.04.2013 um 20:30 (+0000): > Michael Ludwig gmx.de> writes: > > (Okay, no error with N-51511-g599866f from 2013-04-01, > > but not the desired result either.) > > This is ticket #2413, a patch is currently discussed > on ffmpeg-devel. > Further improvements are of course welcome! Okay, I agree with the #2413 author. Will see if I can improve the docs. By the way, the reason the original example on the doc page is wrong is because (a) it crops the bottom half of the image which isn't supposed to appear in the output at all, and (b) it uses the cropped part as the main input for the overlay, so the resulting output ends up too small. The #2413 author posted a correct example, at the end of the ticket description, where he uses a fifo filter, which could be removed, I think (although I don't understand its purpose), and which has been removed recently. http://ffmpeg.org/trac/ffmpeg/ticket/2413 [FFmpeg-devel] [PATCH] doc: remove mention of fifo filter in the introduction. http://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/135982.html I think the following examples are clear, in decreasing order of explicitness: -vf "[in] split [S1][S2]; [S2] crop=iw:ih/2:0:0, vflip [Mir]; [S1][Mir] overlay=0:H/2 [out]" -vf "[in] split [Whole], crop=iw:ih/2:0:0, vflip [Mir]; [Whole][Mir] overlay=0:H/2 [out]" -vf "[in] split [Whole], crop=iw:ih/2:0:0, vflip, [Whole] overlay=0:H/2 [out]" Michael From dashing.meng at gmail.com Fri Apr 5 11:00:21 2013 From: dashing.meng at gmail.com (littlebat) Date: Fri, 5 Apr 2013 17:00:21 +0800 Subject: [FFmpeg-user] Docs, filtering introduction example In-Reply-To: <20130405084537.GF4840@Dago> References: <20130404163444.GD4840@Dago> <20130405084537.GF4840@Dago> Message-ID: <20130405170021.c3e62ab8.dashing.meng@gmail.com> On Fri, 5 Apr 2013 10:45:37 +0200 Michael Ludwig wrote: > Carl Eugen Hoyos schrieb am 04.04.2013 um 20:30 (+0000): > > Michael Ludwig gmx.de> writes: > > > > (Okay, no error with N-51511-g599866f from 2013-04-01, > > > but not the desired result either.) > > > > This is ticket #2413, a patch is currently discussed > > on ffmpeg-devel. > > Further improvements are of course welcome! > > Okay, I agree with the #2413 author. > Will see if I can improve the docs. > > By the way, the reason the original example on the doc page > is wrong is because (a) it crops the bottom half of the > image which isn't supposed to appear in the output at all, > and (b) it uses the cropped part as the main input for the > overlay, so the resulting output ends up too small. > > The #2413 author posted a correct example, at the end of > the ticket description, where he uses a fifo filter, which > could be removed, I think (although I don't understand its > purpose), and which has been removed recently. > > http://ffmpeg.org/trac/ffmpeg/ticket/2413 > > [FFmpeg-devel] [PATCH] doc: remove mention of fifo filter in the > introduction. > http://ffmpeg.org/pipermail/ffmpeg-devel/2012-December/135982.html > > I think the following examples are clear, in decreasing > order of explicitness: > > -vf "[in] split [S1][S2]; [S2] crop=iw:ih/2:0:0, vflip [Mir]; [S1] > [Mir] overlay=0:H/2 [out]" -vf "[in] split [Whole], crop=iw:ih/2:0:0, > vflip [Mir]; [Whole][Mir] overlay=0:H/2 [out]" -vf "[in] split > [Whole], crop=iw:ih/2:0:0, vflip, [Whole] overlay=0:H/2 [out]" You can join to discuss this question at FFmpeg-devel list(seems need subscribe the mail list at first), this is the lastest replay about this question so far: http://ffmpeg.org/pipermail/ffmpeg-devel/2013-April/141758.html From francois.visagie at gmail.com Fri Apr 5 11:17:38 2013 From: francois.visagie at gmail.com (Francois Visagie) Date: Fri, 5 Apr 2013 11:17:38 +0200 Subject: [FFmpeg-user] Muxer Catch-22 with '-target pal-dvd' Message-ID: <004a01ce31de$6dfb9cb0$49f2d610$@gmail.com> Hi All, Just a heads-up for the odd DVD being authored in SD resolution still, there's a Catch-22 with the '-target pal-dvd' option. When encoding without that option, the (grantedly intermediate) output file's audio cannot be played by some DirectShow applications like Windows Media Player (see ticket 1720 ). The work-around is to apply '-target pal-dvd'. However, when encoding 704-wide Rec.601 source like DV (perfectly legal resolution for DVD), this option causes distortion by forcing horizontal resolution to 720 pixels (see ticket 2434 ). Some background on Rec.601 sources and 704-pixel horizontal resolution at http://en.wikipedia.org/wiki/Pixel_aspect_ratio. The work-around in this case is _not_ to use '-target pal-dvd' but to specify the DVD-compliant encode using individual options, e.g. '-codec:a ac3 -b:a 128000 -codec:v mpeg2video -g 15 -bf 2 -bufsize 1835000 -b:v 6090000 -maxrate 9800000 -muxrate 10080000'. In this case the output file's audio won't be playable by Windows Media Player and similar and can't be demuxed by e.g. Avidemux for authoring. Work-arounds are remuxing the output in another application, and/or using another player like Media Player Classic, and/or demuxing for authoring with ffmpeg itself or another application. Cheers, Francois Francois Visagie T: +27 (83) 326-4358 F: +27 (86) 511-3490 E: francois.visagie at gmail.com From vkroutik at gmail.com Fri Apr 5 17:26:17 2013 From: vkroutik at gmail.com (Vlad Kroutik) Date: Fri, 5 Apr 2013 08:26:17 -0700 Subject: [FFmpeg-user] saving HLS stream onto S3 and CF challenge Message-ID: Hello, We are trying to use ffmpeg to save HLS stream into S3 bucket and then pick it up from CF. We have a few issues, maybe someone has a better solution: Here is a command line to capture a stream: C:\Dev\Tools\ffmpeg\ffmpeg.exe -re -i "http://X.X.X.X:1935/live/playlist.m3u8" -codec copy -y "v:\mylist\mylist.m3u8" It works, BUT chunks only 5 seconds long, and if origin starters, then chunks do it as well. we also would like to record that stream, I guess so user can rewind if needed, anyway to do it with the same command line without writing all the code? Thank you for recommendation on best practice here. From mike at redtux.org.uk Fri Apr 5 21:42:50 2013 From: mike at redtux.org.uk (Mike Martin) Date: Fri, 5 Apr 2013 20:42:50 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: Thats the patch I'm using (had to manually apply last two hunks, but thats probably because I'm patching against 1.0.6) Got it to compile but had to use a different ffmpeg version (0.10.6), will try against 0.11 as well The issue seems to be that URLProtocol became private at some stage in 2012 which causes make to bail TBH seeing how hostile the comments on that bug were bit wary of going on there On 4 April 2013 21:14, Carl Eugen Hoyos wrote: > Mike Martin redtux.org.uk> writes: > > > Has anyone had any joy with building this against > > proper ffmpeg and not the obnoxious fork which > > gstreamer seems to have gone with > > (Sorry if this is unrelated, I don't know much about > GStreamer) > A patch was posted that allows to build a GStreamer > module with FFmpeg: > https://bugzilla.gnome.org/show_bug.cgi?id=694937 > > Testing the patch and reporting on gnome bugzilla > is probably very welcome! > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From natgeo.operations at gmail.com Fri Apr 5 23:06:46 2013 From: natgeo.operations at gmail.com (Steve C) Date: Fri, 5 Apr 2013 17:06:46 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: okay, I figured out how to use git, at least enough to test with -checkout 9362f31. here are the results of testing with the 'git checkout 9362f3' of ffmeg used the same command statment on three or four different ProResHQ.mov source files. ffmpeg -i /Source-Files/035925_NTSC_ProResHQ_EM.mov -c:v copy -c:a copy -report /Users/ffmpeg/Desktop/output-4.mov with -checkout 9362f31 the results were the same each time. I dont see the same timecode error as before ("fps 2997 is too large"), but the timecode on the output file is still not correct. it seems the FFMPEG is still having trouble with the the timecode track. each time the log message shows: [mov @ 0x101cc1800] timecode: tbc=2997/1 invalid, fallback on 2997/100 Please see below for the full output from one of the test files. Anyone have any other suggestions as to what else I can try? ###################################### ffmpeg started on 2013-04-05 at 16:28:45 Report written to "ffmpeg-20130405-162845.log" Command line: ffmpeg -i /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov -c:v copy -c:a copy -report /Users/ffmpeg/Desktop/output-3.mov ffmpeg version N-49254-g9362f31 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 4 2013 17:28:58 with gcc 4.2.1 (GCC) (Apple Inc. build 5666) (dot 3) configuration: libavutil 52. 15.102 / 52. 15.102 libavcodec 54. 90.100 / 54. 90.100 libavformat 54. 61.104 / 54. 61.104 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 33.100 / 3. 33.100 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 Splitting the commandline. Reading option '-i' ... matched as input file with argument '/Source-Files/039486_NTSC_ProResHQ_NA_DF.mov'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '/Users/ffmpeg/Desktop/output-3.mov' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov. Successfully parsed a group of options. Opening an input file: /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] ISO: File Type Major Brand: qt [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] File position before avformat_find_stream_info() is 1820088290 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] File position after avformat_find_stream_info() is 870464 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Source-Files/039486_NTSC_ProResHQ_NA_DF.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-04-05 20:25:13 Duration: 00:01:10.07, start: 0.000000, bitrate: 207802 kb/s Stream #0:0(eng), 1, 1/2997: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 207768 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2013-04-05 20:25:13 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Stream #0:1(eng), 0, 1/2997: Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-04-05 20:27:42 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Successfully openened the file. Parsing a group of options: output file /Users/ffmpeg/Desktop/output-3.mov. Applying option c:v (codec name) with argument copy. Applying option c:a (codec name) with argument copy. Successfully parsed a group of options. Opening an output file: /Users/ffmpeg/Desktop/output-3.mov. Successfully openened the file. [mov @ 0x10187e000] timecode: tbc=2997/1 invalid, fallback on 2997/100 Output #0, mov, to '/Users/ffmpeg/Desktop/output-3.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf54.61.104 Stream #0:0(eng), 0, 1/11988: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 207768 kb/s, 29.97 fps, 11988 tbn, 2997 tbc Metadata: creation_time : 2013-04-05 20:25:13 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help frame= 21 fps=0.0 q=-1.0 size= 17898kB time=00:00:00.66 bitrate=219606.8kbits/s [TRUNCATED] frame= 2092 fps= 17 q=-1.0 size= 1769340kB time=00:01:09.77 bitrate=207745.7kbits/s frame= 2100 fps= 17 q=-1.0 size= 1777147kB time=-577014:-32:-22.-77 bitrate=N/A No more output streams to write to, finishing. frame= 2100 fps= 17 q=-1.0 Lsize= 1777164kB time=00:01:10.03 bitrate=207869.0kbits/s video:1777147kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.000983% Statistics: 1820149677 bytes read, 3 seeks From natgeo.operations at gmail.com Fri Apr 5 23:25:42 2013 From: natgeo.operations at gmail.com (Steve C) Date: Fri, 5 Apr 2013 17:25:42 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: I forgot to mention that not only was the timecode wrong on the output file, but it contained special characters example - starting timecode on input file = 09:59:50;00 - starting timecode on output file = 01:39:21;<9 (where is the < character come from?) Also, i just built the other version of FFMPEG that was suggested by Tom - "git checkout c071b00" This version produced the same results as the previous one. In the output report, I see this line: [mov @ 0x10187e000] timecode: tbc=2997/1 invalid, fallback on 2997/100 it also produces timecode in the output file which is wrong and has the "<" character in the place of the frames. Please see below for the full output report. What can I try now? Seems like this is bug in the way timecode is handled by FFMPEG for ProRes.mov? #################### ffmpeg started on 2013-04-05 at 17:18:02 Report written to "ffmpeg-20130405-171802.log" Command line: ffmpeg -i /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov -c:v copy -c:a copy -report /Users/ffmpeg/Desktop/output-5.mov ffmpeg version N-49251-gc071b00 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 5 2013 17:16:18 with gcc 4.2.1 (GCC) (Apple Inc. build 5666) (dot 3) configuration: libavutil 52. 15.102 / 52. 15.102 libavcodec 54. 90.100 / 54. 90.100 libavformat 54. 61.104 / 54. 61.104 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 33.100 / 3. 33.100 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 Splitting the commandline. Reading option '-i' ... matched as input file with argument '/Source-Files/039486_NTSC_ProResHQ_NA_DF.mov'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '/Users/ffmpeg/Desktop/output-5.mov' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov. Successfully parsed a group of options. Opening an input file: /Source-Files/039486_NTSC_ProResHQ_NA_DF.mov. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] ISO: File Type Major Brand: qt [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] File position before avformat_find_stream_info() is 1820088290 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x10180f600] File position after avformat_find_stream_info() is 870464 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Source-Files/039486_NTSC_ProResHQ_NA_DF.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-04-05 20:25:13 Duration: 00:01:10.07, start: 0.000000, bitrate: 207802 kb/s Stream #0:0(eng), 1, 1/2997: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080, 207768 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc Metadata: creation_time : 2013-04-05 20:25:13 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Stream #0:1(eng), 0, 1/2997: Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2013-04-05 20:27:42 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Successfully openened the file. Parsing a group of options: output file /Users/ffmpeg/Desktop/output-5.mov. Applying option c:v (codec name) with argument copy. Applying option c:a (codec name) with argument copy. Successfully parsed a group of options. Opening an output file: /Users/ffmpeg/Desktop/output-5.mov. Successfully openened the file. [mov @ 0x10187e000] timecode: tbc=2997/1 invalid, fallback on 2997/100 Output #0, mov, to '/Users/ffmpeg/Desktop/output-5.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf54.61.104 Stream #0:0(eng), 0, 1/11988: Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 207768 kb/s, 29.97 fps, 11988 tbn, 2997 tbc Metadata: creation_time : 2013-04-05 20:25:13 handler_name : Apple Alias Data Handler timecode : 09:59:59;00 Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help frame= 30 fps=0.0 q=-1.0 size= 25472kB time=00:00:00.96 bitrate=215569.5kbits/s [TRUNCATED] frame= 2094 fps= 23 q=-1.0 size= 1771305kB time=00:01:09.83 bitrate=207777.6kbits/s No more output streams to write to, finishing. frame= 2100 fps= 23 q=-1.0 Lsize= 1777164kB time=00:01:10.03 bitrate=207869.0kbits/s video:1777147kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.000983% Statistics: 1820149677 bytes read, 3 seeks From andrey.krieger.utkin at gmail.com Fri Apr 5 23:27:25 2013 From: andrey.krieger.utkin at gmail.com (Andrey Utkin) Date: Sat, 6 Apr 2013 00:27:25 +0300 Subject: [FFmpeg-user] saving HLS stream onto S3 and CF challenge In-Reply-To: References: Message-ID: 2013/4/5 Vlad Kroutik : > It works, BUT chunks only 5 seconds long, and if origin starters, then > chunks do it as well. we also would like to record that stream, I > guess so user can rewind if needed, anyway to do it with the same > command line without writing all the code? Personally i didn't understand this part at all. Please elaborate. -- Andrey Utkin From vkroutik at gmail.com Fri Apr 5 23:51:33 2013 From: vkroutik at gmail.com (Vlad Kroutik) Date: Fri, 5 Apr 2013 14:51:33 -0700 Subject: [FFmpeg-user] saving HLS stream onto S3 and CF challenge In-Reply-To: References: Message-ID: Sure, We would like to control how ffmpeg saves hls into the files, for some reason we so no place to control settings. Can someone suggest a best way to save hls stream into local files with some control, like DVR functionality? Thank yo. On Fri, Apr 5, 2013 at 2:27 PM, Andrey Utkin wrote: > 2013/4/5 Vlad Kroutik : >> It works, BUT chunks only 5 seconds long, and if origin starters, then >> chunks do it as well. we also would like to record that stream, I >> guess so user can rewind if needed, anyway to do it with the same >> command line without writing all the code? > > Personally i didn't understand this part at all. Please elaborate. > -- > Andrey Utkin > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From mike at redtux.org.uk Sat Apr 6 04:53:50 2013 From: mike at redtux.org.uk (Mike Martin) Date: Sat, 6 Apr 2013 03:53:50 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: FYI - the version of of ffmpeg which builds is between 0.10.6 and 0.11 I will upload source and rpm to www.redtux.org.uk shortly On 5 April 2013 20:42, Mike Martin wrote: > Thats the patch I'm using (had to manually apply last two hunks, but thats > probably because I'm patching against 1.0.6) > > Got it to compile but had to use a different ffmpeg version (0.10.6), will > try against 0.11 as well > > The issue seems to be that URLProtocol became private at some stage in > 2012 which causes make to bail > > TBH seeing how hostile the comments on that bug were bit wary of going on > there > > > On 4 April 2013 21:14, Carl Eugen Hoyos wrote: > >> Mike Martin redtux.org.uk> writes: >> >> > Has anyone had any joy with building this against >> > proper ffmpeg and not the obnoxious fork which >> > gstreamer seems to have gone with >> >> (Sorry if this is unrelated, I don't know much about >> GStreamer) >> A patch was posted that allows to build a GStreamer >> module with FFmpeg: >> https://bugzilla.gnome.org/show_bug.cgi?id=694937 >> >> Testing the patch and reporting on gnome bugzilla >> is probably very welcome! >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From mandy.wu at intel.com Sat Apr 6 12:13:54 2013 From: mandy.wu at intel.com (Wu, Mandy) Date: Sat, 6 Apr 2013 10:13:54 +0000 Subject: [FFmpeg-user] ffmpeg to get yuv file Message-ID: Hi, All, What is difference of decoding mp4 to yuv and decoding the demuxed H.264 file to yuv? I want to get yuv file of a source video (mp4). I tried below 2 commands: Decode to yuv file directly from mp4: ffmpeg -i souce.mp4 source1.yuv Demux the source file, and Decode the .H264 file to yuv: ffmpeg -i source.mp4 -vcodec copy -an source.h264 ffmpeg -i source.h264 source2.yuv I found size of the 2 yuv files is different. previous source1.yuv is bigger than the second one. Which one is correct? And which one should I use for psnr calculation? Thanks! Thanks, Mandy From andrey.aleksandrovich at googlemail.com Sat Apr 6 19:10:22 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Sat, 6 Apr 2013 20:10:22 +0300 Subject: [FFmpeg-user] Several audio tracks encoding issue In-Reply-To: References: Message-ID: I have still no result :( As it is mentioned there http://ffmpeg.org/trac/ffmpeg/wiki/Encoding VBR (Variable Bit Rate) mp3 audio the parameter -q:a:1 5 tells encoder (libmp3lame) to use "audio quality" at value 5, which means ~120-150Kbps VBR. But, the command: $ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 -map 1:0 -map 2:0 -q:v:0 3 -q:a:0 5 -q:a:1 5 final.avi generates bad second audio track (32Kbps, instead of specified quality setting). What's wrong? How to set this correctly? On 4/2/13, Andrey Aleksandrovich wrote: > Ok, will try. Thanks. > > On 4/2/13, Carl Eugen Hoyos wrote: >> Andrey Aleksandrovich writes: >> >>> localhost $ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 >>> -map 1:0 -map 2:0 -c:v mpeg4 -qscale:v:0 3 -c:a:0 libmp3lame >>> -qscale:a:0 4 final.avi -c:a:1 libmp3lame -qscale:a:1 4 -newaudio >> >> The newaudio option was dropped years ago, map is sufficient now. >> Note that options after an output file name have no effect on >> the output file - in your case "-c:a:1 libmp3lame". >> (Does qscale really work for lame?) >> >>> ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers >> >> Please update to current git head, 0.10 is >> quite outdated. >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > From 23liam at googlemail.com Sat Apr 6 19:34:10 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Sat, 6 Apr 2013 18:34:10 +0100 Subject: [FFmpeg-user] Several audio tracks encoding issue In-Reply-To: References: Message-ID: >But, the command: > >$ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 -map 1:0 -map 2:0 >-q:v:0 3 -q:a:0 5 -q:a:1 5 final.avi > >generates bad second audio track (32Kbps, instead of specified quality setting). >What's wrong? How to set this correctly? `-q` sets a specific quality, rather than a bit rate; the bit rate will vary depending on the nature of the audio track. Is the audio quality bad? From andrey.aleksandrovich at googlemail.com Sat Apr 6 20:08:46 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Sat, 6 Apr 2013 21:08:46 +0300 Subject: [FFmpeg-user] Several audio tracks encoding issue In-Reply-To: References: Message-ID: The first audio track is OK (~128KB VBR). But the second one is bad (32KB CBR). Why it is happened? On 4/6/13, Liam Condron-Farnos <23liam at googlemail.com> wrote: >>But, the command: >> >>$ ffmpeg -i v.avi -i a1.wav -i a2.wav -map 0:0 -map 1:0 -map 2:0 >>-q:v:0 3 -q:a:0 5 -q:a:1 5 final.avi >> >>generates bad second audio track (32Kbps, instead of specified quality > setting). >>What's wrong? How to set this correctly? > > `-q` sets a specific quality, rather than a bit rate; the bit rate will > vary depending on the nature of the audio track. Is the audio quality bad? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From hxxhancsj at hotmail.com Sun Apr 7 11:15:51 2013 From: hxxhancsj at hotmail.com (xxtina) Date: Sun, 7 Apr 2013 02:15:51 -0700 (PDT) Subject: [FFmpeg-user] ffplay can not play a mp4 video In-Reply-To: References: <1364701556788-4658185.post@n4.nabble.com> <20130331084838.GE29565@leki> <1364743034314-4658188.post@n4.nabble.com> Message-ID: <1365326151458-4658310.post@n4.nabble.com> Hello, I am so appreciated that I could turn to you for help. As your response, I have uploaded the small sample of the video that I can not play at the link http://www1.datafilehost.com/d/18f96aec (named 'mp4-can-not-play'). Could you give me some suggestions for decode videos like this one? Thank you so much! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffplay-can-not-play-a-mp4-video-tp4658185p4658310.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Sun Apr 7 11:57:46 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 7 Apr 2013 09:57:46 +0000 (UTC) Subject: [FFmpeg-user] ffplay can not play a mp4 video References: <1364701556788-4658185.post@n4.nabble.com> <20130331084838.GE29565@leki> <1364743034314-4658188.post@n4.nabble.com> <1365326151458-4658310.post@n4.nabble.com> Message-ID: xxtina hotmail.com> writes: > I have uploaded the small sample of the video that I can > not play at the link http://www1.datafilehost.com/d/18f96aec > (named 'mp4-can-not-play'). The file is not an mp4 file but a recording from a surveillance camera from a company called "Hikvision". To the best of my knowledge there is no way to decode these samples except with the Windows binary driver on WMP. Carl Eugen From cehoyos at ag.or.at Sun Apr 7 11:59:12 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 7 Apr 2013 09:59:12 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg to get yuv file References: Message-ID: Wu, Mandy intel.com> writes: > I want to get yuv file of a source video (mp4). > I tried below 2 commands: > Decode to yuv file directly from mp4: > ffmpeg -i souce.mp4 source1.yuv > Demux the source file, and Decode the .H264 file to yuv: > ffmpeg -i source.mp4 -vcodec copy -an source.h264 > ffmpeg -i source.h264 source2.yuv You probably want to use -vsync 0 in both cases. (If not, FFmpeg may duplicate / drop frames.) Carl Eugen From cehoyos at ag.or.at Sun Apr 7 12:00:56 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 7 Apr 2013 10:00:56 +0000 (UTC) Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module References: Message-ID: Mike Martin redtux.org.uk> writes: > Thats the patch I'm using (had to manually apply last > two hunks, but thats probably because I'm patching > against 1.0.6) > > Got it to compile but had to use a different ffmpeg > version (0.10.6), will try against 0.11 as well I was under the impression that the patch at least allows to compile with 1.2, is that wrong? There is no advantage in using older versions (but many disadvantages.) Carl Eugen From mike at redtux.org.uk Sun Apr 7 18:54:14 2013 From: mike at redtux.org.uk (Mike Martin) Date: Sun, 7 Apr 2013 17:54:14 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: My understanding is that the version of libav used in gst-libav is an old version Everything newer fails at the URLProtocols issue On 7 April 2013 11:00, Carl Eugen Hoyos wrote: > Mike Martin redtux.org.uk> writes: > > > Thats the patch I'm using (had to manually apply last > > two hunks, but thats probably because I'm patching > > against 1.0.6) > > > > Got it to compile but had to use a different ffmpeg > > version (0.10.6), will try against 0.11 as well > > I was under the impression that the patch at least allows > to compile with 1.2, is that wrong? > There is no advantage in using older versions (but many > disadvantages.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mike at redtux.org.uk Sun Apr 7 19:28:48 2013 From: mike at redtux.org.uk (Mike Martin) Date: Sun, 7 Apr 2013 18:28:48 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: On 7 April 2013 17:54, Mike Martin wrote: > My understanding is that the version of libav used in gst-libav is an old > version > > Everything newer fails at the URLProtocols issue > > > On 7 April 2013 11:00, Carl Eugen Hoyos wrote: > >> Mike Martin redtux.org.uk> writes: >> >> > Thats the patch I'm using (had to manually apply last >> > two hunks, but thats probably because I'm patching >> > against 1.0.6) >> > >> > Got it to compile but had to use a different ffmpeg >> > version (0.10.6), will try against 0.11 as well >> >> I was under the impression that the patch at least allows >> to compile with 1.2, is that wrong? >> There is no advantage in using older versions (but many >> disadvantages.) >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > Just checked and gst-libav includes libav-0.8.5 and current libav is 0.9.4 From andrey.aleksandrovich at googlemail.com Sun Apr 7 20:02:31 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Sun, 7 Apr 2013 21:02:31 +0300 Subject: [FFmpeg-user] Strange Wav (PCM) file (isn't playing correctly) Message-ID: I have old game (Anno 1602), and there's nice soundtrack playing during the game. The file is here http://rghost.net/45123707 But I can't understand, what codec this audio is compressed by. All known media players play it with distortion. In the Game this music is playing correctly. Here is that music recorded by 'what I hear' method (system sound grabber): http://rghost.net/45124011 If anyone know what is that codec, please help me to convert it. From timbot at oddlystudios.com Sun Apr 7 21:44:07 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Sun, 7 Apr 2013 15:44:07 -0400 Subject: [FFmpeg-user] alphamerge filter use problem Message-ID: Hi, this is my first question ever to a mailing list (or anywhere!). I have been using ffmpeg for about a year and a half and have figured out many things on my own but I'm stumped regarding the alphaextract and alphamerge filters. I am able to get the proper result every time with images, but have yet to succeed with videos. I have tried many formats and searched extensively but have not found a single documented case of someone successfully using this filter combination and sharing an example. The documentation on the ffmpeg-filters page is very sparse, and does not give any details of what formats/codecs are supported. I think this filter would be very useful if I can figure it out. Step 1 - alphaextract This seems to always work, and produces a grayscale video as expected. In my tests I have used rgba video, as well as quicktime with animation codec (bgra). Step 2 - applying the extracted alpha This works every time for images. However while the encode completes with video it never provides the masking expected by mapping the extracted alpha information to the alpha channel of the target video. In my tests with still images, I found that the output format must be a format that supports an alpha channel, and I have tried several with video (which I determined by looking at the source code of the filter). My command (taken from the docs) looks like this: ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov [alpha]; [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png composite.mov ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro 4.6.1-9ubuntu3) configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3 --enable-libfreetype --enable-filter=drawtext libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.100 / 3. 49.100 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 creation_time : 2013-04-07 18:35:55 Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc Metadata: creation_time : 2013-04-07 18:35:55 handler_name : ?Mainconcept Video Media Handler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s Metadata: creation_time : 2013-04-07 18:35:55 handler_name : #Mainconcept MP4 Sound Media Handler Output #0, mov, to 'composite.mov': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf55.1.100 Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc Metadata: creation_time : 2013-04-07 18:35:55 handler_name : ?Mainconcept Video Media Handler Stream mapping: Stream #0:0 -> #0:0 (h264 -> png) Press [q] to stop, [?] for help frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 bitrate=3177.5kbits/s dup=2 drop=0 video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.067323% After this, I see only my red background (timbotundelay.mp4) and no alpha applied. Thanks so much to anyone who can help me out here. I'm running out of things to try. I'd particularly like to see how people are using this filter successfully. From zbernie at icloud.com Mon Apr 8 03:30:43 2013 From: zbernie at icloud.com (Bernard Barton Jr.) Date: Sun, 07 Apr 2013 21:30:43 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? Message-ID: I'm using the following command to convert Apple iTunes .m4a files to .mp3: ffmpeg -i "$fname" -acodec libmp3lame -ab 256k -vn -y -f mp3 "$dir/$basefile.mp3" I was getting warnings about frame rates, even though these are audio files, so I added the -vn switch, which eliminated complaints about frame rate. But I'm still getting the "Missing png signature", and Video:png errors shown below. How can I eliminate these warnings about about the decoding for stream 1 failing? [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc393832600] stream 0, timescale not set [png @ 0x7fc393838800] Missing png signature [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc393832600] max_analyze_duration 5000000 reached at 5015510 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc393832600] decoding for stream 1 failed [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc393832600] Could not find codec parameters for stream 1 (Video: png): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options -Thanks From cehoyos at ag.or.at Sun Apr 7 22:13:39 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 7 Apr 2013 20:13:39 +0000 (UTC) Subject: [FFmpeg-user] Strange Wav (PCM) file (isn't playing correctly) References: Message-ID: Andrey Aleksandrovich writes: > I have old game (Anno 1602), and there's nice > soundtrack playing during the game. The file is here > http://rghost.net/45123707 $ ffmpeg -acodec adpcm_ima_apc -i 1st.wav works fine, I don't know if it is possible to autodetect the codec, there is a fact chunk and I am not sure what it contains. Carl Eugen From hxxhancsj at hotmail.com Mon Apr 8 08:55:29 2013 From: hxxhancsj at hotmail.com (xxtina) Date: Sun, 7 Apr 2013 23:55:29 -0700 (PDT) Subject: [FFmpeg-user] ffplay can not play a mp4 video In-Reply-To: References: <1364701556788-4658185.post@n4.nabble.com> <20130331084838.GE29565@leki> <1364743034314-4658188.post@n4.nabble.com> <1365326151458-4658310.post@n4.nabble.com> Message-ID: <1365404129422-4658319.post@n4.nabble.com> Thanks for the answers. But I am still a little confused. I am so sorry to bother you. Could you please tell me more about the Windows binary driver on WMP, such as where to dig for the introduction and usage? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffplay-can-not-play-a-mp4-video-tp4658185p4658319.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mike at redtux.org.uk Mon Apr 8 10:23:23 2013 From: mike at redtux.org.uk (Mike Martin) Date: Mon, 8 Apr 2013 09:23:23 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: On 7 April 2013 18:28, Mike Martin wrote: > > On 7 April 2013 17:54, Mike Martin wrote: > >> My understanding is that the version of libav used in gst-libav is an old >> version >> >> Everything newer fails at the URLProtocols issue >> >> >> On 7 April 2013 11:00, Carl Eugen Hoyos wrote: >> >>> Mike Martin redtux.org.uk> writes: >>> >>> > Thats the patch I'm using (had to manually apply last >>> > two hunks, but thats probably because I'm patching >>> > against 1.0.6) >>> > >>> > Got it to compile but had to use a different ffmpeg >>> > version (0.10.6), will try against 0.11 as well >>> >>> I was under the impression that the patch at least allows >>> to compile with 1.2, is that wrong? >>> There is no advantage in using older versions (but many >>> disadvantages.) >>> >>> Carl Eugen >>> >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>> >> >> > Just checked and gst-libav includes libav-0.8.5 and current libav is 0.9.4 > Also currently I am building against Fedora 18, whhich is gstreamer-1.0.6, so ffmpeg-0.10.6, released 26/10/2012, so this isn;t exactly ancient. gst-libav pulls libav-0.8.5 From ubitux at gmail.com Mon Apr 8 10:26:21 2013 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Mon, 8 Apr 2013 10:26:21 +0200 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: <20130408082621.GF25743@leki> On Mon, Apr 08, 2013 at 09:23:23AM +0100, Mike Martin wrote: [...] > Also currently I am building against Fedora 18, whhich is gstreamer-1.0.6, > so ffmpeg-0.10.6, released 26/10/2012, so this isn;t exactly ancient. > It's the date of the latest revision. 0.10 was released around January 2012. And so, it's actually pretty old. The difference between 0.10 and 0.10.6 is just little backported bugfixes. -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From tag at gmx.com Mon Apr 8 11:28:53 2013 From: tag at gmx.com (tag at gmx.com) Date: Mon, 8 Apr 2013 11:28:53 +0200 (CEST) Subject: [FFmpeg-user] Which parameter for WEBM VP8/Vorbis quality depending on resolution? In-Reply-To: <20130403093323.1e52e26b@lrcd.com> References: , <20130403093323.1e52e26b@lrcd.com> Message-ID: > On 03.04.2013 19:33 Lou wrote: >> Tom Evans wrote: >>> You can use the -crf argument, which sets a quality setting for the >>> final video, not a bitrate. I believe this works with libvpx. >> [...] >> Seems it does not work. Default 200 kb/s stays active all the time. >> Are there other options? > See the vpx (WebM) Encoding Guide: > https://ffmpeg.org/trac/ffmpeg/wiki/vpxEncodingGuide It is solved now. My problem was: -crf is trying to use some bitrate, but -b:v must be set much higher - the default is so ridiculous low that it is useless. It makes no sense for me to treat some value, that is being used as real bitrate for variable bitrate, as upper bound for constant bitrate which may not be achieved by far. Max bitrate is a complete different meaning than target bitrate and imho should be a different parameter. With a high default value. Maybe the documentation of -crf could state that more clearly. Mohan G: It seems the -quality setting is more a minor tweak than setting a real quality or bitrate. The hint did not help, but thank you for the try also ;). I will unsubscribe soon as my problem is solved. Thank you for the help! From mike at redtux.org.uk Mon Apr 8 11:38:11 2013 From: mike at redtux.org.uk (Mike Martin) Date: Mon, 8 Apr 2013 10:38:11 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: <20130408082621.GF25743@leki> References: <20130408082621.GF25743@leki> Message-ID: On 8 April 2013 09:26, Cl?ment B?sch wrote: > On Mon, Apr 08, 2013 at 09:23:23AM +0100, Mike Martin wrote: > [...] > > Also currently I am building against Fedora 18, whhich is > gstreamer-1.0.6, > > so ffmpeg-0.10.6, released 26/10/2012, so this isn;t exactly ancient. > > > > It's the date of the latest revision. 0.10 was released around January > 2012. And so, it's actually pretty old. The difference between 0.10 and > 0.10.6 is just little backported bugfixes. > > -- > Cl?ment B. > > Although as the latest released gstreamer is 1.0.6 and the plugin is based off libav-0.8.5, it is pretty much like for like Seeing if I can build gstreamer from head now From natgeo.operations at gmail.com Mon Apr 8 16:04:06 2013 From: natgeo.operations at gmail.com (Steve C) Date: Mon, 8 Apr 2013 10:04:06 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: never received a response on my last posts. Do the admins on this list agree that this is a bug in FFMPEG? Should I submit to the bug tracker? Or is there still something left to try? thanks again. From andrey.aleksandrovich at googlemail.com Mon Apr 8 16:27:27 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Mon, 8 Apr 2013 17:27:27 +0300 Subject: [FFmpeg-user] Strange Wav (PCM) file (isn't playing correctly) In-Reply-To: References: Message-ID: Oh, Thank you very much! This is it. On 4/7/13, Carl Eugen Hoyos wrote: > Andrey Aleksandrovich writes: > >> I have old game (Anno 1602), and there's nice >> soundtrack playing during the game. The file is here >> http://rghost.net/45123707 > > $ ffmpeg -acodec adpcm_ima_apc -i 1st.wav > works fine, I don't know if it is possible > to autodetect the codec, there is a fact > chunk and I am not sure what it contains. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From tevans.uk at googlemail.com Mon Apr 8 16:48:18 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Mon, 8 Apr 2013 15:48:18 +0100 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: On Mon, Apr 8, 2013 at 3:04 PM, Steve C wrote: > never received a response on my last posts. > > Do the admins on this list agree that this is a bug in FFMPEG? > Should I submit to the bug tracker? Or is there still something left to try? Carl is probably just waiting for a response to this: On Tue, Apr 2, 2013 at 12:40 PM, Carl Eugen Hoyos wrote: > Steve C gmail.com> writes: > >> The error effects opening and playback with Quicktime 7 > > Sorry, I had misread that part of your original mail. > Could you test if 9362f31 and c071b00 also fail? Cheers Tom From natgeo.operations at gmail.com Mon Apr 8 16:58:06 2013 From: natgeo.operations at gmail.com (Steve C) Date: Mon, 8 Apr 2013 10:58:06 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: > Carl is probably just waiting for a response to this: > > On Tue, Apr 2, 2013 at 12:40 PM, Carl Eugen Hoyos wrote: >> Steve C gmail.com> writes: >> >>> The error effects opening and playback with Quicktime 7 >> >> Sorry, I had misread that part of your original mail. >> Could you test if 9362f31 and c071b00 also fail? Tom, I did test these two and posted my results to the list on Friday of last week. While they didn't give the same error as the master version, neither of those two versions produced an output file with correct timecode. in some cases the timecode track did not even contain all number values (please see the previous posts from last week for the details). From cehoyos at ag.or.at Mon Apr 8 10:58:39 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 8 Apr 2013 08:58:39 +0000 (UTC) Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module References: Message-ID: Mike Martin redtux.org.uk> writes: > My understanding is that the version of libav used > in gst-libav is an old version > > Everything newer fails at the URLProtocols issue I just found the mail that Dirk (who wrote the FFmpeg patch for gst-libav) sent me and afaiu, it confirms that gst-libav master works fine with FFmpeg 1.2 If it does not work, please report here! Carl Eugen From cehoyos at ag.or.at Mon Apr 8 10:41:38 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 8 Apr 2013 08:41:38 +0000 (UTC) Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? References: Message-ID: Bernard Barton Jr. icloud.com> writes: > ffmpeg -i "$fname" -acodec libmp3lame -ab 256k -vn > -y -f mp3 "$dir/$basefile.mp3" Please provide your actual command line (one without variables but actual paths) together with the complete, uncut console output. Carl Eugen From lulebo at gmail.com Mon Apr 8 18:05:07 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Mon, 8 Apr 2013 18:05:07 +0200 Subject: [FFmpeg-user] Segmenting using pipe Message-ID: Hello I have been experimenting with encoding a single file and removing parts of the stream. I need this function when encoding material with breaks for commercials to remove bars and other funky stuff and sync with the subtitles. I have done some tests with piping (windows), and I can get this to work pretty well for two arbitrary segments as an example: (ffmpeg -i inputfile.mp4 -ss 00:10:00 -t 60 -bsf h264_mp4toannexb -vcodec copy -acodec copy -f mpegts - ffmpeg -i inputfile.mp4 -ss 00:20:00 -t 60 -bsf h264_mp4toannexb -vcodec copy -acodec copy -f mpegts - )| ffmpeg -y -f mpegts -i - -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 704k -minrate 704k -maxrate 704k -bufsize 1500k -profile:v baseline -aspect 16:9 -c:a libvo_aacenc -b:a 96k -ac 2 test.mp4 (this will create a two minute file from two different sections of the input file) The problem is the codec copying in the first parts. Since the timecodes might begin in the middle of a GOP, the final ffmpeg will not know what to do until the first I-frame shows up, so it just shows a freeze frame until that happens. There is a requirement of a fixed gop size of 3 secs for the final output file, so I need to do the encoding again for the final step, no cat can be used on segments. Is there a good intermediate codec to decode to in the first step? (Preferrably uncompressed to gain performance) I have found yuv4mpegvideo but that only does video. I need the audio as well. Anyone got any smart ideas? /Carl From shakiramer at gmail.com Mon Apr 8 20:58:22 2013 From: shakiramer at gmail.com (amer shakir) Date: Mon, 8 Apr 2013 11:58:22 -0700 Subject: [FFmpeg-user] conversion error Message-ID: when we convert .wav to mp3 it cannot be played by some players due to time issues. the players are not recognizing the length of the file. can you help? From andrey.aleksandrovich at googlemail.com Mon Apr 8 21:00:21 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Mon, 8 Apr 2013 22:00:21 +0300 Subject: [FFmpeg-user] Scanty information about video when using libxvid or 'vtag' option Message-ID: Hello, I observed, that using libxvid makes scanty info about video. For example, this command: ffmpeg -y -i video.avi -an -q:v 3 -t 20 mpeg4.avi makes video with characteristics (MediaInfo): http://bpaste.net/show/89938/ and the one below ffmpeg -y -i video.avi -an -c:v libxvid -qscale 3 -t 20 xvid.avi makes appropriate: http://bpaste.net/show/89939/ The same is when using '-vtag xvid' option with default mpeg4 codec. I need fourcc 'xvid' to play video on my standalone DVD-player. But I also want to understand why there's so little information in these cases. Thanks. From lou at lrcd.com Mon Apr 8 21:01:16 2013 From: lou at lrcd.com (Lou) Date: Mon, 8 Apr 2013 11:01:16 -0800 Subject: [FFmpeg-user] conversion error In-Reply-To: References: Message-ID: <20130408110116.3447ca4d@lrcd.com> On Mon, 8 Apr 2013 11:58:22 -0700 amer shakir wrote: > when we convert .wav to mp3 it cannot be played by some players due to time > issues. the players are not recognizing the length of the file. > > can you help? Please include your ffmpeg command and the complete console output. From primeair at gmail.com Mon Apr 8 18:32:20 2013 From: primeair at gmail.com (Gotblade) Date: Mon, 8 Apr 2013 09:32:20 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg stalls when trying to transcode live content from HD PVR In-Reply-To: References: <1360941809099-4657324.post@n4.nabble.com> <1364781346214-4658189.post@n4.nabble.com> Message-ID: <1365438740727-4658332.post@n4.nabble.com> The set of commands I used are not a problem they work without hanging so I didn't bother with console output. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-stalls-when-trying-to-transcode-live-content-from-HD-PVR-tp4657324p4658332.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From zbernie at icloud.com Tue Apr 9 01:07:19 2013 From: zbernie at icloud.com (Bernard Barton Jr.) Date: Mon, 08 Apr 2013 19:07:19 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? In-Reply-To: References: Message-ID: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> Here's the ffmpeg command without variables, and the complete output as requested: bernie at imac:/Volumes/Media Library/iTunes/iTunes Media/Music/Andy Williams/Merry Christmas> ffmpeg -i 02\ Have\ Yourself\ A\ Merry\ Little\ Christmas.m4a -acodec libmp3lame -ab 256k -vn -y -f mp3 /tmp/test.mp3 ffmpeg version 1.1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Feb 14 2013 13:28:28 with Apple clang version 4.0 (tags/Apple/clang-421.0.57) (based on LLVM 3.1svn) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-nonfree --enable-libfaac libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 86.100 / 54. 86.100 libavformat 54. 59.106 / 54. 59.106 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fd21a032600] stream 0, timescale not set [png @ 0x7fd21a038800] Missing png signature [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fd21a032600] max_analyze_duration 5000000 reached at 5015510 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fd21a032600] decoding for stream 1 failed [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fd21a032600] Could not find codec parameters for stream 1 (Video: png): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '02 Have Yourself A Merry Little Christmas.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2011-07-26 04:49:15 title : Have Yourself A Merry Little Christmas artist : Andy Williams album_artist : Andy Williams composer : R. Blane album : Merry Christmas genre : Holiday track : 2/12 disc : 1/1 date : 2004 media_type : 1 Duration: 00:03:23.43, start: 0.000000, bitrate: 304 kb/s Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 281 kb/s Metadata: creation_time : 2011-07-26 04:49:15 Stream #0:1: Video: png, 90k tbr, 90k tbn, 90k tbc Output #0, mp3, to '/tmp/test.mp3': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom media_type : 1 TIT2 : Have Yourself A Merry Little Christmas TPE1 : Andy Williams TPE2 : Andy Williams TCOM : R. Blane TALB : Merry Christmas TCON : Holiday TRCK : 2/12 TPOS : 1/1 TDRL : 2004 TSSE : Lavf54.59.106 Stream #0:0(eng): Audio: mp3, 44100 Hz, stereo, fltp, 256 kb/s Metadata: creation_time : 2011-07-26 04:49:15 Stream mapping: Stream #0:0 -> #0:0 (aac -> libmp3lame) Press [q] to stop, [?] for help size= 6358kB time=00:03:23.43 bitrate= 256.0kbits/s video:0kB audio:6357kB subtitle:0 global headers:0kB muxing overhead 0.018497% On Apr 8, 2013, at 4:41 AM, Carl Eugen Hoyos wrote: > Bernard Barton Jr. icloud.com> writes: > >> ffmpeg -i "$fname" -acodec libmp3lame -ab 256k -vn >> -y -f mp3 "$dir/$basefile.mp3" > > Please provide your actual command line (one without > variables but actual paths) together with the complete, > uncut console output. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Tue Apr 9 01:16:12 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 8 Apr 2013 23:16:12 +0000 (UTC) Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> Message-ID: Bernard Barton Jr. icloud.com> writes: > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '02 Have > Yourself A Merry Little Christmas.m4a': > Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), > 44100 Hz, stereo, fltp, 281 kb/s > Metadata: > creation_time : 2011-07-26 04:49:15 > Stream #0:1: Video: png, 90k tbr, 90k tbn, 90k tbc It appears that the mov file contains an image (thumbnail) that FFmpeg cannot read (or does ffmpeg -i 02.m4a out.jpg work?), consider providing the sample. (If you really only want to silence the warning, you probably have to disable the png decoder.) Carl Eugen From zbernie at icloud.com Tue Apr 9 02:54:38 2013 From: zbernie at icloud.com (Bernard Barton Jr.) Date: Mon, 08 Apr 2013 20:54:38 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? In-Reply-To: References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> Message-ID: Yes, the m4a file contains album artwork, which ffmpeg is detecting. I get an error when I try to output a jpg file. But your suggestion to disable the png decoder sounds accurate. However, I just read the man page, and do not see how to disable the png decoder. Can you please explain how this accomplished? On Apr 8, 2013, at 7:16 PM, Carl Eugen Hoyos wrote: > Bernard Barton Jr. icloud.com> writes: > >> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '02 Have >> Yourself A Merry Little Christmas.m4a': > >> Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), >> 44100 Hz, stereo, fltp, 281 kb/s >> Metadata: >> creation_time : 2011-07-26 04:49:15 >> Stream #0:1: Video: png, 90k tbr, 90k tbn, 90k tbc > > It appears that the mov file contains an image (thumbnail) > that FFmpeg cannot read (or does ffmpeg -i 02.m4a out.jpg > work?), consider providing the sample. > > (If you really only want to silence the warning, you > probably have to disable the png decoder.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Tue Apr 9 03:23:57 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 01:23:57 +0000 (UTC) Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> Message-ID: Bernard Barton Jr. icloud.com> writes: > Yes, the m4a file contains album artwork, which > ffmpeg is detecting. I get an error when I try > to output a jpg file. Please provide the sample. --disable-decoder=png disables the png decoder;-) Carl Eugen From zbernie at icloud.com Tue Apr 9 03:37:33 2013 From: zbernie at icloud.com (Bernard Barton Jr.) Date: Mon, 08 Apr 2013 21:37:33 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? In-Reply-To: References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> Message-ID: <7879988F-8935-46FA-A800-5CEA1AA9DC03@icloud.com> I tried attaching the 6MB m4a file, but received an email stating that the limit is 100KB. I then tried adding the --disable-decoder=png to my ffmpeg command in several places, but it always returns an "Unrecognized option" error: bernie at MacBook:~/Music> ffmpeg -i 04\ It\'s\ The\ Most\ Wonderful\ Time\ Of\ The\ Year.m4a --disable-decoder=png -acodec libmp3lame -ab 256k -vn -y -f mp3 /tmp/test.mp3 Unrecognized option '-disable-decoder=png' Failed to set value '-acodec' for option '-disable-decoder=png' On Apr 8, 2013, at 9:23 PM, Carl Eugen Hoyos wrote: > Bernard Barton Jr. icloud.com> writes: > >> Yes, the m4a file contains album artwork, which >> ffmpeg is detecting. I get an error when I try >> to output a jpg file. > > Please provide the sample. > > --disable-decoder=png disables the png decoder;-) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Tue Apr 9 03:55:51 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 01:55:51 +0000 (UTC) Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> <7879988F-8935-46FA-A800-5CEA1AA9DC03@icloud.com> Message-ID: Bernard Barton Jr. icloud.com> writes: > I tried attaching the 6MB m4a file, but received an > email stating that the limit is 100KB. Could you upload the sample to http://www1.datafilehost.com/ and post the link here? --disable-decoder is a configure option. Carl Eugen From zbernie at icloud.com Tue Apr 9 04:01:31 2013 From: zbernie at icloud.com (Bernard Barton Jr.) Date: Mon, 08 Apr 2013 22:01:31 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? In-Reply-To: References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> <7879988F-8935-46FA-A800-5CEA1AA9DC03@icloud.com> Message-ID: Here's the link: http://www1.datafilehost.com/d/5ec321bf On Apr 8, 2013, at 9:55 PM, Carl Eugen Hoyos wrote: > Bernard Barton Jr. icloud.com> writes: > >> I tried attaching the 6MB m4a file, but received an >> email stating that the limit is 100KB. > > Could you upload the sample to http://www1.datafilehost.com/ > and post the link here? > > --disable-decoder is a configure option. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From simondaniels23 at gmail.com Tue Apr 9 06:09:11 2013 From: simondaniels23 at gmail.com (Simon Daniels) Date: Mon, 8 Apr 2013 21:09:11 -0700 Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why Message-ID: Hey guys, All of the sudden many of my users are getting a "timebase not supported" error when using ffmpeg to trim a video. My app hasn't changed in months and it definitely seems like this is a more recent error. This is with ffmpeg 0.11.1. The weird thing is that I've had other videos from the same camera trim just fine with the similar ffmpeg arguments. Any ideas? running file://localhost/Applications/MyApp.app/Contents/Resources/ffmpeg with arguments ( "-ss", "52.402666", "-t", "15.000000", "-y", "-i", "/Users/user/Documents/GOPR9605.MP4", "-acodec", copy, "-copyts", "-copytb", 1, "-vf", "movie=watermark-1280x720.png [watermark]; [in][watermark]overlay=0:0 [out]", "-r", "59.94006065616729", "-b:v", 15215367, "/Users/user/Movies/GOPR9605 highlight.MP4" ) I 04/08/13 21:02:27 SaveWorker parseFfmpegOutput: | Ffmpeg results: ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers built on Aug 5 2012 09:38:45 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00) configuration: --enable-static --disable-shared --disable-outdev=sdl --enable-runtime-cpudetect --disable-bzlib --disable-libfreetype --disable-libopenjpeg --enable-zlib --arch=x86_64 --sysroot=/Developer/SDKs/MacOSX10.7.sdk --extra-cflags='-isysroot /Developer/SDKs/MacOSX10.7.sdk' libavutil 51. 54.100 / 51. 54.100 libavcodec 54. 23.100 / 54. 23.100 libavformat 54. 6.100 / 54. 6.100 libavdevice 54. 0.100 / 54. 0.100 libavfilter 2. 77.100 / 2. 77.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 15.100 / 0. 15.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/user/Documents/GOPR9605.MP4': Metadata: major_brand : avc1 minor_version : 0 compatible_brands: avc1isom creation_time : 2013-03-08 02:34:46 Duration: 00:07:45.01, start: 0.000000, bitrate: 15216 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 15073 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AVC Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AAC [buffer @ 0x10fb16e20] w:1280 h:720 pixfmt:yuv420p tb:1/90000 sar:1/1 sws_param:flags=2 [buffersink @ 0x10fb17100] No opaque field provided [movie @ 0x10fb17740] seek_point:0 format_name:(null) file_name:watermark-1280x720.png stream_index:0 [overlay @ 0x10fb17dc0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_movie_0' and the filter 'Parsed_overlay_1' [scale @ 0x10fb18560] w:1264 h:704 fmt:rgba sar:1/1 -> w:1264 h:704 fmt:yuva420p sar:1/1 flags:0x4 [overlay @ 0x10fb17dc0] main w:1280 h:720 fmt:yuv420p overlay x:0 y:0 w:1264 h:704 fmt:yuva420p [overlay @ 0x10fb17dc0] main_tb:1/90000 overlay_tb:1/25 -> tb:1/90000 exact:1 [mpeg4 @ 0x7fe2ab82fc00] timebase 15499/929011 not supported by MPEG 4 standard, the maximum admitted value for the timebase denominator is 65535 Output #0, mp4, to '/Users/user/Movies/GOPR9605 highlight.MP4': Metadata: major_brand : avc1 minor_version : 0 compatible_brands: avc1isom creation_time : 2013-03-08 02:34:46 Stream #0:0(eng): Video: mpeg4, yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 15215 kb/s, 90k tbn, 59.94 tbc Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AVC Stream #0:1(eng): Audio: aac, 48000 Hz, stereo, 128 kb/s Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AAC Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (copy) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height here is the MediaInfo output for this file: General Count : 280 Count of stream of this kind : 1 Kind of stream : General Kind of stream : General Stream identifier : 0 Count of video streams : 1 Count of audio streams : 1 Video_Format_List : AVC Video_Format_WithHint_List : AVC Codecs Video : AVC Video_Language_List : English Audio_Format_List : AAC Audio_Format_WithHint_List : AAC Audio codecs : AAC LC Audio_Language_List : English Complete name : /Users/user/Documents/ GOPR9605.MP4 Folder name : /Users/user/Documents/ File name : GOPR9605 File extension : mp4 Format : MPEG-4 Format : MPEG-4 Format/Extensions usually used : mp4 m4v m4a m4b m4p 3gpp 3gp 3gpp2 3g2 k3g jpm jpx mqv ismv isma f4v Commercial name : MPEG-4 Format profile : JVT Internet media type : video/mp4 Codec ID : avc1 Codec ID/Url : http://www.apple.com/quicktime/download/standalone.html Codec : MPEG-4 Codec : MPEG-4 Codec/Extensions usually used : mp4 m4v m4a m4b m4p 3gpp 3gp 3gpp2 3g2 k3g jpm jpx mqv ismv isma f4v File size : 884484368 File size : 844 MiB File size : 844 MiB File size : 844 MiB File size : 844 MiB File size : 843.5 MiB Duration : 465015 Duration : 7mn 45s Duration : 7mn 45s 15ms Duration : 7mn 45s Duration : 00:07:45.015 Overall bit rate mode : CBR Overall bit rate mode : Constant Overall bit rate : 15216445 Overall bit rate : 15.2 Mbps Stream size : 884740 Stream size : 864 KiB (0%) Stream size : 864 KiB Stream size : 864 KiB Stream size : 864 KiB Stream size : 864.0 KiB Stream size : 864 KiB (0%) Proportion of this stream : 0.00100 HeaderSize : 884732 DataSize : 883599636 FooterSize : 0 IsStreamable : Yes Encoded date : UTC 2013-03-08 02:34:46 Tagged date : UTC 2013-03-08 02:34:46 File last modification date : UTC 2013-03-08 09:42:30 File last modification date (local) : 2013-03-08 01:42:30 AMBA : Video Count : 248 Count of stream of this kind : 1 Kind of stream : Video Kind of stream : Video Stream identifier : 0 StreamOrder : 0 ID : 1 ID : 1 Format : AVC Format/Info : Advanced Video Codec Format/Url : http://developers.videolan.org/x264.html Commercial name : AVC Format profile From simondaniels23 at gmail.com Tue Apr 9 07:05:20 2013 From: simondaniels23 at gmail.com (Simon Daniels) Date: Mon, 8 Apr 2013 22:05:20 -0700 Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why In-Reply-To: References: Message-ID: I've updated to ffmpeg 1.2 and get different results. The video trims, but the entire thing is "frozen" and there's no audio. The file is also super small, probably because it's basically 15 seconds of a single frame. Users-MacBook-Pro:ffmpeg-1.2 user$ ./ffmpeg -ss 52 -y -i "/Users/user/Documents/GOPR9605.MP4" -t 15 -acodec copy -copyts -copytb 1 -vf "movie=watermark-1280x720.png [watermark]; [in] [watermark]overlay=0:0 [out]" -r 59.94006065616729 -b:v 15215367 "/Users/user/Movies/GOPR9605 highlight.MP4" ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 8 2013 21:40:26 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-static --disable-shared --enable-gpl --enable-nonfree --enable-libx264 --disable-outdev=sdl --enable-runtime-cpudetect --disable-bzlib --disable-libfreetype --disable-libopenjpeg --enable-zlib --arch=x86_64 --sysroot=/Developer/SDKs/MacOSX10.7.sdk --extra-cflags='-isysroot /Developer/SDKs/MacOSX10.7.sdk' libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/user/Documents/GOPR9605.MP4': Metadata: major_brand : avc1 minor_version : 0 compatible_brands: avc1isom creation_time : 2013-03-08 02:34:46 Duration: 00:07:45.01, start: 0.000000, bitrate: 15216 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 15073 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AVC Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AAC [libx264 @ 0x7fdcd9017600] using SAR=1/1 [libx264 @ 0x7fdcd9017600] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x7fdcd9017600] profile High, level 4.1 [libx264 @ 0x7fdcd9017600] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=15215 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to '/Users/user/Movies/GOPR9605 highlight.MP4': Metadata: major_brand : avc1 minor_version : 0 compatible_brands: avc1isom encoder : Lavf54.63.104 Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 15215 kb/s, 929011.00 tbn, 59.94 tbc Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AVC Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 128 kb/s Metadata: creation_time : 2013-03-08 02:34:46 handler_name : GoPro AAC Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 900 fps= 30 q=-2.0 Lsize= 2210kB time=00:00:14.98 bitrate=1208.2kbits/s dup=3060 drop=0 video:2198kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.515732% [libx264 @ 0x7fdcd9017600] frame I:4 Avg QP: 2.10 size:469148 [libx264 @ 0x7fdcd9017600] frame P:227 Avg QP: 0.33 size: 1467 [libx264 @ 0x7fdcd9017600] frame B:669 Avg QP: 1.17 size: 61 [libx264 @ 0x7fdcd9017600] consecutive B-frames: 0.9% 0.0% 0.0% 99.1% [libx264 @ 0x7fdcd9017600] mb I I16..4: 27.2% 16.1% 56.6% [libx264 @ 0x7fdcd9017600] mb P I16..4: 0.0% 0.0% 0.0% P16..4: 2.5% 0.0% 0.0% 0.0% 0.0% skip:97.5% [libx264 @ 0x7fdcd9017600] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.6% 0.0% 0.0% direct: 0.0% skip:99.4% L0:29.6% L1:70.3% BI: 0.0% [libx264 @ 0x7fdcd9017600] final ratefactor: -20.24 [libx264 @ 0x7fdcd9017600] 8x8 transform intra:16.1% inter:40.9% [libx264 @ 0x7fdcd9017600] coded y,uvDC,uvAC intra: 99.9% 87.5% 79.4% inter: 0.4% 0.2% 0.2% [libx264 @ 0x7fdcd9017600] i16 v,h,dc,p: 4% 7% 65% 24% [libx264 @ 0x7fdcd9017600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 8% 17% 29% 7% 8% 5% 9% 6% 12% [libx264 @ 0x7fdcd9017600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 11% 15% 22% 8% 9% 7% 10% 7% 11% [libx264 @ 0x7fdcd9017600] i8c dc,h,v,p: 53% 21% 22% 3% [libx264 @ 0x7fdcd9017600] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x7fdcd9017600] ref P L0: 99.2% 0.0% 0.7% 0.1% [libx264 @ 0x7fdcd9017600] ref B L0: 48.8% 51.1% 0.0% [libx264 @ 0x7fdcd9017600] ref B L1: 87.6% 12.4% [libx264 @ 0x7fdcd9017600] kb/s:1198.93 That help? On Mon, Apr 8, 2013 at 9:09 PM, Simon Daniels wrote: > Hey guys, > > All of the sudden many of my users are getting a "timebase not supported" > error when using ffmpeg to trim a video. My app hasn't changed in months > and it definitely seems like this is a more recent error. > > This is with ffmpeg 0.11.1. The weird thing is that I've had other videos > from the same camera trim just fine with the similar ffmpeg arguments. > > Any ideas? > > > running file://localhost/Applications/MyApp.app/Contents/Resources/ffmpeg > with arguments ( > "-ss", > "52.402666", > "-t", > "15.000000", > "-y", > "-i", > "/Users/user/Documents/GOPR9605.MP4", > "-acodec", > copy, > "-copyts", > "-copytb", > 1, > "-vf", > "movie=watermark-1280x720.png [watermark]; [in][watermark]overlay=0:0 > [out]", > "-r", > "59.94006065616729", > "-b:v", > 15215367, > "/Users/user/Movies/GOPR9605 highlight.MP4" > ) > I 04/08/13 21:02:27 SaveWorker parseFfmpegOutput: | Ffmpeg results: > ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers > built on Aug 5 2012 09:38:45 with llvm_gcc 4.2.1 (Based on Apple Inc. > build 5658) (LLVM build 2336.1.00) > configuration: --enable-static --disable-shared --disable-outdev=sdl > --enable-runtime-cpudetect --disable-bzlib --disable-libfreetype > --disable-libopenjpeg --enable-zlib --arch=x86_64 > --sysroot=/Developer/SDKs/MacOSX10.7.sdk --extra-cflags='-isysroot > /Developer/SDKs/MacOSX10.7.sdk' > libavutil 51. 54.100 / 51. 54.100 > libavcodec 54. 23.100 / 54. 23.100 > libavformat 54. 6.100 / 54. 6.100 > libavdevice 54. 0.100 / 54. 0.100 > libavfilter 2. 77.100 / 2. 77.100 > libswscale 2. 1.100 / 2. 1.100 > libswresample 0. 15.100 / 0. 15.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from > '/Users/user/Documents/GOPR9605.MP4': > Metadata: > major_brand : avc1 > minor_version : 0 > compatible_brands: avc1isom > creation_time : 2013-03-08 02:34:46 > Duration: 00:07:45.01, start: 0.000000, bitrate: 15216 kb/s > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > 1280x720 [SAR 1:1 DAR 16:9], 15073 kb/s, 59.94 fps, 59.94 tbr, 90k tbn, > 119.88 tbc > Metadata: > creation_time : 2013-03-08 02:34:46 > handler_name : GoPro AVC > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, > s16, 128 kb/s > Metadata: > creation_time : 2013-03-08 02:34:46 > handler_name : GoPro AAC > [buffer @ 0x10fb16e20] w:1280 h:720 pixfmt:yuv420p tb:1/90000 sar:1/1 > sws_param:flags=2 > [buffersink @ 0x10fb17100] No opaque field provided > [movie @ 0x10fb17740] seek_point:0 format_name:(null) > file_name:watermark-1280x720.png stream_index:0 > [overlay @ 0x10fb17dc0] auto-inserting filter 'auto-inserted scaler 0' > between the filter 'Parsed_movie_0' and the filter 'Parsed_overlay_1' > [scale @ 0x10fb18560] w:1264 h:704 fmt:rgba sar:1/1 -> w:1264 h:704 > fmt:yuva420p sar:1/1 flags:0x4 > [overlay @ 0x10fb17dc0] main w:1280 h:720 fmt:yuv420p overlay x:0 y:0 > w:1264 h:704 fmt:yuva420p > [overlay @ 0x10fb17dc0] main_tb:1/90000 overlay_tb:1/25 -> tb:1/90000 > exact:1 > [mpeg4 @ 0x7fe2ab82fc00] timebase 15499/929011 not supported by MPEG 4 > standard, the maximum admitted value for the timebase denominator is 65535 > Output #0, mp4, to '/Users/user/Movies/GOPR9605 highlight.MP4': > Metadata: > major_brand : avc1 > minor_version : 0 > compatible_brands: avc1isom > creation_time : 2013-03-08 02:34:46 > Stream #0:0(eng): Video: mpeg4, yuv420p, 1280x720 [SAR 1:1 DAR 16:9], > q=2-31, 15215 kb/s, 90k tbn, 59.94 tbc > Metadata: > creation_time : 2013-03-08 02:34:46 > handler_name : GoPro AVC > Stream #0:1(eng): Audio: aac, 48000 Hz, stereo, 128 kb/s > Metadata: > creation_time : 2013-03-08 02:34:46 > handler_name : GoPro AAC > Stream mapping: > Stream #0:0 -> #0:0 (h264 -> mpeg4) > Stream #0:1 -> #0:1 (copy) > Error while opening encoder for output stream #0:0 - maybe incorrect > parameters such as bit_rate, rate, width or height > > > > here is the MediaInfo output for this file: > General > Count : 280 > Count of stream of this kind : 1 > Kind of stream : General > Kind of stream : General > Stream identifier : 0 > Count of video streams : 1 > Count of audio streams : 1 > Video_Format_List : AVC > Video_Format_WithHint_List : AVC > Codecs Video : AVC > Video_Language_List : English > Audio_Format_List : AAC > Audio_Format_WithHint_List : AAC > Audio codecs : AAC LC > Audio_Language_List : English > Complete name : /Users/user/Documents/ > GOPR9605.MP4 > Folder name : /Users/user/Documents/ > File name : GOPR9605 > File extension : mp4 > Format : MPEG-4 > Format : MPEG-4 > Format/Extensions usually used : mp4 m4v m4a m4b m4p 3gpp 3gp > 3gpp2 3g2 k3g jpm jpx mqv ismv isma f4v > Commercial name : MPEG-4 > Format profile : JVT > Internet media type : video/mp4 > Codec ID : avc1 > Codec ID/Url : > http://www.apple.com/quicktime/download/standalone.html > Codec : MPEG-4 > Codec : MPEG-4 > Codec/Extensions usually used : mp4 m4v m4a m4b m4p 3gpp 3gp > 3gpp2 3g2 k3g jpm jpx mqv ismv isma f4v > File size : 884484368 > File size : 844 MiB > File size : 844 MiB > File size : 844 MiB > File size : 844 MiB > File size : 843.5 MiB > Duration : 465015 > Duration : 7mn 45s > Duration : 7mn 45s 15ms > Duration : 7mn 45s > Duration : 00:07:45.015 > Overall bit rate mode : CBR > Overall bit rate mode : Constant > Overall bit rate : 15216445 > Overall bit rate : 15.2 Mbps > Stream size : 884740 > Stream size : 864 KiB (0%) > Stream size : 864 KiB > Stream size : 864 KiB > Stream size : 864 KiB > Stream size : 864.0 KiB > Stream size : 864 KiB (0%) > Proportion of this stream : 0.00100 > HeaderSize : 884732 > DataSize : 883599636 > FooterSize : 0 > IsStreamable : Yes > Encoded date : UTC 2013-03-08 02:34:46 > Tagged date : UTC 2013-03-08 02:34:46 > File last modification date : UTC 2013-03-08 09:42:30 > File last modification date (local) : 2013-03-08 01:42:30 > AMBA : > > Video > Count : 248 > Count of stream of this kind : 1 > Kind of stream : Video > Kind of stream : Video > Stream identifier : 0 > StreamOrder : 0 > ID : 1 > ID : 1 > Format : AVC > Format/Info : Advanced Video Codec > Format/Url : > http://developers.videolan.org/x264.html > Commercial name : AVC > Format profile > > From cehoyos at ag.or.at Tue Apr 9 10:35:42 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 08:35:42 +0000 (UTC) Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> <7879988F-8935-46FA-A800-5CEA1AA9DC03@icloud.com> Message-ID: Bernard Barton Jr. icloud.com> writes: > http://www1.datafilehost.com/d/5ec321bf Does iTunes show the cover correctly? (A picture of Andy Williams on red background.) Workaround to decode the cover art of this specific sample is to specify the decoder: $ ffmpeg -vcodec mjpeg -i 04...m4a If you only want to silence the error message, configure FFmpeg with --disable-decoder=png. Carl Eugen From cehoyos at ag.or.at Tue Apr 9 10:36:57 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 08:36:57 +0000 (UTC) Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why References: Message-ID: Simon Daniels gmail.com> writes: > All of the sudden many of my users are getting a > "timebase not supported" error when using ffmpeg > to trim a video. Please provide the input sample (or find the commit introducing the problem). Carl Eugen From ticks.sunny84 at gmail.com Tue Apr 9 11:46:42 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Tue, 9 Apr 2013 15:16:42 +0530 Subject: [FFmpeg-user] video created using ffmpeg and H264 codec does not work on android devices Message-ID: Hi, I have created a video using ffmpeg and H264 codec ,but the video does not play on the android devices. The code i am using is as follows: JNIEXPORT void Java_com_canvasm_mediclinic_VideoGenerator_generate(JNIEnv *pEnv, jobject pObj,jobjectArray stringArray,int famerate,int width,int height,jstring videoFilename) { AVCodec *codec; AVCodecContext *c= NULL; //int framesnum=5; int i,looper, out_size, size, x, y,j; int ret,pts,got_pkt_ptr; int imagecount= (*pEnv)->GetArrayLength(pEnv, stringArray); int retval=-10; uint8_t endcode[]={0,0,1,0xb7}; AVPacket outpacket; FILE *f; AVFrame *picture,*encoded_avframe; jbyte *raw_record; char logdatadata[100]; int returnvalue = -1,numBytes =-1; const char *gVideoFileName = (char *)(*pEnv)->GetStringUTFChars(pEnv, videoFilename, NULL); /* find the mpeg1 video encoder */ codec = avcodec_find_encoder_by_name("libx264"); if (!codec) { __android_log_write(ANDROID_LOG_INFO, "record","codec not found"); exit(1); } c= avcodec_alloc_context(); c->bit_rate = 500000; c->width = width; c->height = height; c->time_base= (AVRational){1,famerate}; c->gop_size = 12; // emit one intra frame every ten frames c->max_b_frames=1; c->pix_fmt = PIX_FMT_YUV420P; c->codec_type = AVMEDIA_TYPE_VIDEO; c->codec_id = codec->id; c->max_b_frames = 0; c->me_range = 16; c->max_qdiff = 4; c->qmin = 10; c->qmax = 26; c->qcompress = 0.6; c->trellis=0; c->level = 30; c->refs = 5; c->coder_type = 0; c->scenechange_threshold = 0; //new c->flags|=CODEC_FLAG_LOOP_FILTER;//new c->scenechange_threshold = 40; //new c-> rc_buffer_size = 0; c->gop_size=250; //new c->max_b_frames=1;//new c->me_method=7; c->me_cmp|= 1; c->me_subpel_quality = 6; c->qmax=51; c->keyint_min=25; av_opt_set(c->priv_data,"subq","6",0); av_opt_set(c->priv_data,"crf","20.0",0); av_opt_set(c->priv_data,"weighted_p_pred","0",0); av_opt_set(c->priv_data,"profile","baseline",AV_OPT_SEARCH_CHILDREN); av_opt_set(c->priv_data,"preset","medium",0); av_opt_set(c->priv_data,"tune","zerolatency",0); av_opt_set(c->priv_data,"x264opts","rc-lookahead=0",0); /* open it */ retval = avcodec_open(c, codec); if ( retval < 0) { __android_log_write(ANDROID_LOG_INFO, "record","could not open codec"); exit(1); } f = fopen(gVideoFileName, "ab"); if (!f) { __android_log_write(ANDROID_LOG_INFO, "record","could not open video file"); exit(1); } pts = 0; for(i=0;i<=imagecount;i++) { jboolean isCp; int progress = 0; float temp; jstring string; if(i==imagecount) string = (jstring) (*pEnv)->GetObjectArrayElement(pEnv, stringArray, imagecount-1); else string = (jstring) (*pEnv)->GetObjectArrayElement(pEnv, stringArray, i); const char *rawString = (*pEnv)->GetStringUTFChars(pEnv, string, &isCp); picture = OpenImage(rawString,width,height,i); av_init_packet(&outpacket); fflush(stdout); { picture->pts=i ;//c->frame_number; do{ out_size = avcodec_encode_video2(c, &outpacket, picture,&got_pkt_ptr); }while(!got_pkt_ptr); } returnvalue = fwrite(outpacket.data, 1, outpacket.size, f); av_free_packet(&outpacket); } /* get the delayed frames */ for(got_pkt_ptr =1; got_pkt_ptr; i++) { fflush(stdout); avcodec_encode_video2(c, &outpacket, NULL,&got_pkt_ptr); if(got_pkt_ptr) { fwrite(outpacket.data, 1, outpacket.size, f); av_free_packet(&outpacket); } } fwrite(endcode,1,sizeof(endcode),f); fclose(f); avcodec_close(c); av_free(c); } AVFrame* OpenImage(const char* imageFileName,int w,int h,int index) { AVFrame *pFrame; AVCodec *pCodec ; AVFormatContext *pFormatCtx = NULL; AVCodecContext *pCodecCtx; uint8_t *buffer; int frameFinished,framesNumber = 0,retval = -1,numBytes=0; AVPacket packet; char logdatadata[100]; int result = -1; result=avformat_open_input(&pFormatCtx,imageFileName,NULL,NULL); if(result!=0) { __android_log_write(ANDROID_LOG_INFO, "record", "Can't open image file "); return NULL; } pCodecCtx = pFormatCtx->streams[0]->codec; pCodecCtx->width = w; pCodecCtx->height = h; pCodecCtx->pix_fmt = PIX_FMT_YUV420P; // Find the decoder for the video stream pCodec = avcodec_find_decoder(pCodecCtx->codec_id); if (!pCodec) { __android_log_write(ANDROID_LOG_INFO, "record", "Can't open image file "); return NULL; } pFrame = avcodec_alloc_frame(); numBytes = avpicture_get_size(PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height); buffer = (uint8_t *) av_malloc(numBytes * sizeof(uint8_t)); retval = avpicture_fill((AVPicture *) pFrame, buffer, PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height); // Open codec if(avcodec_open(pCodecCtx, pCodec)<0) { __android_log_write(ANDROID_LOG_INFO, "record","Could not open codec"); return NULL; } if (!pFrame) { __android_log_write(ANDROID_LOG_INFO, "record","Can't allocate memory for AVFrame\n"); return NULL; } int readval = -5; while (readval = av_read_frame(pFormatCtx, &packet) >= 0) { if(packet.stream_index != 0) continue; int ret = avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet); if (ret > 0) { __android_log_write(ANDROID_LOG_INFO, "record","Frame is decoded\n"); pFrame->quality = 4; av_free_packet(&packet); av_close_input_file(pFormatCtx); return pFrame; } else { __android_log_write(ANDROID_LOG_INFO, "record","error while decoding frame \n"); } } } Please let me know where i am going wrong? I have also tried the sampel example code that comes with ffmpeg doc/examples/decoding_encoding.c but the video created using that too is not playing on android devices. I am using ffmpeg version 1.2; any help will be greatly appriciated: regards, susheel tickoo From cehoyos at ag.or.at Tue Apr 9 11:46:59 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 09:46:59 +0000 (UTC) Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 References: Message-ID: Tom Evans googlemail.com> writes: > > Do the admins on this list agree that this is a bug in FFMPEG? > > Should I submit to the bug tracker? Or is there still > > something left to try? > > Carl is probably just waiting for a response to this: Not necessarily, I suspect this is not a regression, the only difference being that in older versions, an invalid timecode was written without any warning, now an error is shown. The actual problem is that I still wonder what a timecode is good for: http://thread.gmane.org/gmane.comp.video.ffmpeg.user/44794/focus=44796 Carl Eugen From cehoyos at ag.or.at Tue Apr 9 11:47:58 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 09:47:58 +0000 (UTC) Subject: [FFmpeg-user] video created using ffmpeg and H264 codec does not work on android devices References: Message-ID: Susheel Tickoo gmail.com> writes: > I have created a video using ffmpeg and H264 codec, > but the video does not play on the android devices. Does it play with any software? ffplay, vlc, MPlayer? Carl Eugen From ticks.sunny84 at gmail.com Tue Apr 9 11:52:07 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Tue, 9 Apr 2013 15:22:07 +0530 Subject: [FFmpeg-user] video created using ffmpeg and H264 codec does not work on android devices In-Reply-To: References: Message-ID: Hi Carl, It plays with Vplayer in the android market but the player skips some frames. On Tue, Apr 9, 2013 at 3:17 PM, Carl Eugen Hoyos wrote: > Susheel Tickoo gmail.com> writes: > > > I have created a video using ffmpeg and H264 codec, > > but the video does not play on the android devices. > > Does it play with any software? > ffplay, vlc, MPlayer? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From tps at netmaster.dk Tue Apr 9 12:26:36 2013 From: tps at netmaster.dk (Thomas Seilund) Date: Tue, 09 Apr 2013 12:26:36 +0200 Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio Message-ID: <5163ECDC.4020202@netmaster.dk> Hi All I have problems creating thumbnails that respect the aspect ratio of the video. I have two videos. One is 16:9 and another one is 4:3. I would expect the thumbnail filter to create an image that has the same aspect ratio as the input video! Below I run two commands for each video. 1. ffmpeg -i video -ss 0 -vf thumbnail -frames:v 1 thumbnail.jpg 2. identify thumbnail.jpg Command `identify` is from ImageMagick and describes the format and characteristics of an image file. In this case the thumbnail created by ffmpeg. Note that the output from `identify` is the same (thumbnail.jpg JPEG 720x576) for both videos. I would have expected pixel width and height to give 16:9 and 4:3 ratio. 16:9 video: ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail -frames:v 1 thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_6 4-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-c flags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postp roc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau - -disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-l ibfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --d isable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host - -enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpeg @ 0x7fb0d9a0f620] max_analyze_duration 5000000 reached at 5000000 Input #0, mpeg, from 'p20120122142736.mpg': Duration: 00:01:24.80, start: 1.000000, bitrate: 6577 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 9000 kb/s, 25 fps, 25 tbr, 90k tbn , 50 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 448 kb/s Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x7fb0d9a17a80] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:64/45 sws_param: [thumbnail @ 0x7fb0d9a23b90] batch size: 100 frames [buffersink @ 0x7fb0d9a17b00] auto-inserting filter 'auto-inserted scale 0' between the filter 'Parsed_thumbnail_0' and the filter 'out' [thumbnail @ 0x7fb0d9a23b90] auto-inserting filter 'auto-inserted scale 1' between the filter 'src' and the filter 'Parsed_ thumbnail_0' [scale @ 0x7fb0d9a243e0] w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:rgb24 flags:0x4 [scale @ 0x7fb0d9a23f70] w:720 h:576 fmt:rgb24 -> w:720 h:576 fmt:yuvj420p flags:0x4 Output #0, image2, to 'thumbnail.jpg': Metadata: encoder : Lavf53.32.100 Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> mjpeg) Press [q] to stop, [?] for help frame id #34 (pts_time=1.360000) selected time=00:00:00.00 bitrate= 0.0kbits/s frame= 1 fps= 1 q=5.9 Lsize= 0kB time=00:00:00.04 bitrate= 0.0kbits/s dup=34 drop=0 video:40kB audio:0kB global headers:0kB muxing overhead -100.000000% identify thumbnail.jpg thumbnail.jpg JPEG 720x576 720x576+0+0 8-bit DirectClass 41.4KB 0.000u 0:00.000 4:3 video: ffmpeg -i p20130120155621.ts -ss 0 -vf thumbnail -frames:v 1 thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-cflags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpegts @ 0x7fc790518620] max_analyze_duration 5000000 reached at 5000000 Input #0, mpegts, from 'p20130120155621.ts': Duration: 00:02:19.44, start: 1.376978, bitrate: 2479 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16, 128 kb/s Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x7fc79051e6f0] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: [thumbnail @ 0x7fc79051ebf0] batch size: 100 frames [buffersink @ 0x7fc79051e950] auto-inserting filter 'auto-inserted scale 0' between the filter 'Parsed_thumbnail_0' and the filter 'out' [thumbnail @ 0x7fc79051ebf0] auto-inserting filter 'auto-inserted scale 1' between the filter 'src' and the filter 'Parsed_thumbnail_0' [scale @ 0x7fc79051f470] w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:rgb24 flags:0x4 [scale @ 0x7fc79051efd0] w:720 h:576 fmt:rgb24 -> w:720 h:576 fmt:yuvj420p flags:0x4 Output #0, image2, to 'thumbnail.jpg': Metadata: encoder : Lavf53.32.100 Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (h264 -> mjpeg) Press [q] to stop, [?] for help [thumbnail @ 0x7fc79051ebf0] frame id #64 (pts_time=2.583022) selected frame= 1 fps= 0 q=6.1 Lsize= 0kB time=00:00:00.04 bitrate= 0.0kbits/s dup=65 drop=0 video:47kB audio:0kB global headers:0kB muxing overhead -100.000000% identify thumbnail.jpg thumbnail.jpg JPEG 720x576 720x576+0+0 8-bit DirectClass 48.1KB 0.000u 0:00.000 Thanks in advance Thomas S From cehoyos at ag.or.at Tue Apr 9 12:33:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 10:33:14 +0000 (UTC) Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio References: <5163ECDC.4020202@netmaster.dk> Message-ID: Thomas Seilund netmaster.dk> writes: > 16:9 video: > > ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail > -frames:v 1 thumbnail.jpg What does ffmpeg -i thumbnail.jpg report? I am also (often) using ImageMagick to test issues, but it is definitely not a bug-free project (see for example 61a9f09). Carl Eugen From tps at netmaster.dk Tue Apr 9 12:43:18 2013 From: tps at netmaster.dk (Thomas Seilund) Date: Tue, 09 Apr 2013 12:43:18 +0200 Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio In-Reply-To: References: <5163ECDC.4020202@netmaster.dk> Message-ID: <5163F0C6.4040007@netmaster.dk> On 09-04-2013 12:33, Carl Eugen Hoyos wrote: > Thomas Seilund netmaster.dk> writes: > >> 16:9 video: >> >> ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail >> -frames:v 1 thumbnail.jpg > What does ffmpeg -i thumbnail.jpg report? > I am also (often) using ImageMagick to test > issues, but it is definitely not a bug-free > project (see for example 61a9f09). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Hi Eugen Thanks! 16:9 ffmpeg -i thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-cflags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, image2, from 'thumbnail.jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 64:45 DAR 16:9], 25 tbr, 25 tbn, 25 tbc At least one output file must be specified 4:3: ffmpeg -i thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-cflags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, image2, from 'thumbnail.jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 16:15 DAR 4:3], 25 tbr, 25 tbn, 25 tbc At least one output file must be specified Thanks Thomas S From cehoyos at ag.or.at Tue Apr 9 12:45:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 10:45:29 +0000 (UTC) Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio References: <5163ECDC.4020202@netmaster.dk> Message-ID: Thomas Seilund netmaster.dk> writes: > 16:9 video: > > ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail > -frames:v 1 thumbnail.jpg I suspect you want to use -vf thumbnail,scale=iw*sar:ih to get an output file with sar=1:1 which will be displayed as expected by applications that ignore the JFIF aspect. Carl Eugen From cehoyos at ag.or.at Tue Apr 9 12:47:09 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 10:47:09 +0000 (UTC) Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio References: <5163ECDC.4020202@netmaster.dk> <5163F0C6.4040007@netmaster.dk> Message-ID: Thomas Seilund netmaster.dk> writes: > Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 64:45 DAR 16:9] > Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 16:15 DAR 4:3] This looks as if FFmpeg does exactly what you want, don't you agree? Carl Eugen From tps at netmaster.dk Tue Apr 9 12:58:10 2013 From: tps at netmaster.dk (Thomas Seilund) Date: Tue, 09 Apr 2013 12:58:10 +0200 Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio In-Reply-To: References: <5163ECDC.4020202@netmaster.dk> Message-ID: <5163F442.6090508@netmaster.dk> On 09-04-2013 12:45, Carl Eugen Hoyos wrote: > Thomas Seilund netmaster.dk> writes: > >> 16:9 video: >> >> ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail >> -frames:v 1 thumbnail.jpg > I suspect you want to use -vf thumbnail,scale=iw*sar:ih > to get an output file with sar=1:1 which will be displayed > as expected by applications that ignore the JFIF aspect. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Hi Carl Thanks a lot. That did the trick! Now I get this and this is what I wanted. 16:9 ffmpeg -i p20120122142736.mpg -ss 0 -vf thumbnail -frames:v 1 -vf scale=iw*sar:ih thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-cflags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpeg @ 0x7f2943d6b620] max_analyze_duration 5000000 reached at 5000000 Input #0, mpeg, from 'p20120122142736.mpg': Duration: 00:01:24.80, start: 1.000000, bitrate: 6577 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 9000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 448 kb/s Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x7f2943d737a0] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:64/45 sws_param: [scale @ 0x7f2943d7fb00] w:720 h:576 fmt:yuv420p -> w:1024 h:576 fmt:yuvj420p flags:0x4 Output #0, image2, to 'thumbnail.jpg': Metadata: encoder : Lavf53.32.100 Stream #0:0: Video: mjpeg, yuvj420p, 1024x576 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> mjpeg) Press [q] to stop, [?] for help frame= 1 fps= 0 q=6.4 Lsize= 0kB time=00:00:00.04 bitrate= 0.0kbits/s identify thumbnail.jpg thumbnail.jpg JPEG 1024x576 1024x576+0+0 8-bit DirectClass 50.2KB 0.000u 0:00.000 4:3 ffmpeg -i p20130120155621.ts -ss 0 -vf thumbnail -frames:v 1 -vf scale=iw*sar:ih thumbnail.jpg ffmpeg version 0.10.3 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 13 2013 20:25:12 with gcc 4.6.3 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-march=native -O2 -pipe' --extra-cflags='-march=native -O2 -pipe' --extra-cxxflags='-O2 -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --disable-ffplay --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=alsa --disable-indev=oss --disable-indev=jack --enable-libv4l2 --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-libfreetype --enable-libvpx --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=host --enable-hardcoded-tables libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpegts @ 0x7f8921a2e620] max_analyze_duration 5000000 reached at 5000000 Input #0, mpegts, from 'p20130120155621.ts': Duration: 00:02:19.44, start: 1.376978, bitrate: 2479 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16, 128 kb/s Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x7f8921a33250] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: [scale @ 0x7f8921a34bd0] w:720 h:576 fmt:yuv420p -> w:768 h:576 fmt:yuvj420p flags:0x4 Output #0, image2, to 'thumbnail.jpg': Metadata: encoder : Lavf53.32.100 Stream #0:0: Video: mjpeg, yuvj420p, 768x576 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (h264 -> mjpeg) Press [q] to stop, [?] for help frame= 1 fps= 0 q=6.2 Lsize= 0kB time=00:00:00.04 bitrate= 0.0kbits/s dup=4 drop=0 identify thumbnail.jpg thumbnail.jpg JPEG 768x576 768x576+0+0 8-bit DirectClass 48.9KB 0.000u 0:00.000 Thanks Thomas S From tps at netmaster.dk Tue Apr 9 13:11:13 2013 From: tps at netmaster.dk (Thomas Seilund) Date: Tue, 09 Apr 2013 13:11:13 +0200 Subject: [FFmpeg-user] filter thumbnail does not respect aspect ratio - SOLVED In-Reply-To: References: <5163ECDC.4020202@netmaster.dk> <5163F0C6.4040007@netmaster.dk> Message-ID: <5163F751.4060805@netmaster.dk> On 09-04-2013 12:47, Carl Eugen Hoyos wrote: > Thomas Seilund netmaster.dk> writes: > >> Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 64:45 DAR 16:9] >> Stream #0:0: Video: mjpeg, yuvj420p, 720x576 [SAR 16:15 DAR 4:3] > This looks as if FFmpeg does exactly what you want, > don't you agree? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Hi Carl I agree. My mistake that I did not apply the scale filter. I learn more about this tool each day! ffmpeg is great! Thanks Thomas S From natgeo.operations at gmail.com Tue Apr 9 13:54:40 2013 From: natgeo.operations at gmail.com (Steve C) Date: Tue, 9 Apr 2013 07:54:40 -0400 Subject: [FFmpeg-user] "fps 2997 is too large" - creates mov with bad public atom (error -2002 In-Reply-To: References: Message-ID: > The actual problem is that I still wonder what a > timecode is good for: > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/44794/focus=44796 Carl, the timecode track is still very important in a broadcast / professional environment. it is used by video editors and audio mixers as a reference tool for numerous functions. again, I have only noticed the problem with Drop Frame Timecode. For some reason, FFMPEG 1.2 and the two other builds tested cannot correctly copy the DF timecode track using "-c:v copy" From mike at redtux.org.uk Tue Apr 9 16:14:01 2013 From: mike at redtux.org.uk (Mike Martin) Date: Tue, 9 Apr 2013 15:14:01 +0100 Subject: [FFmpeg-user] Building Gstreamer v1 ffmpeg module In-Reply-To: References: Message-ID: Just finishing compiling. Yes gst-libav master works, but not gst-libav-1.0.6 (release) On 8 April 2013 09:58, Carl Eugen Hoyos wrote: > Mike Martin redtux.org.uk> writes: > > > My understanding is that the version of libav used > > in gst-libav is an old version > > > > Everything newer fails at the URLProtocols issue > > I just found the mail that Dirk (who wrote the FFmpeg > patch for gst-libav) sent me and afaiu, it confirms > that gst-libav master works fine with FFmpeg 1.2 > > If it does not work, please report here! > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Tue Apr 9 17:18:19 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 9 Apr 2013 15:18:19 +0000 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: Message-ID: On 4/7/13, Tim Willison wrote: > Hi, this is my first question ever to a mailing list (or anywhere!). > I have been using ffmpeg for about a year and a half and have figured out > many things on my own but I'm stumped regarding the alphaextract and > alphamerge filters. > I am able to get the proper result every time with images, but have yet to > succeed with videos. I have tried many formats and searched extensively but > have not found a single documented case of someone successfully using this > filter combination and sharing an example. The documentation on the > ffmpeg-filters page is very sparse, and does not give any details of what > formats/codecs are supported. > > I think this filter would be very useful if I can figure it out. > > Step 1 - alphaextract > This seems to always work, and produces a grayscale video as expected. In > my tests I have used rgba video, as well as quicktime with animation codec > (bgra). > > Step 2 - applying the extracted alpha > This works every time for images. However while the encode completes with > video it never provides the masking expected by mapping the extracted alpha > information to the alpha channel of the target video. In my tests with > still images, I found that the output format must be a format that supports > an alpha channel, and I have tried several with video (which I determined > by looking at the source code of the filter). > > My command (taken from the docs) looks like this: > > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov [alpha]; > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png > composite.mov > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > 4.6.1-9ubuntu3) > configuration: --enable-gpl --enable-libass --enable-libfaac > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-nonfree --enable-version3 --enable-libfreetype > --enable-filter=drawtext > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.100 / 3. 49.100 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42mp41 > creation_time : 2013-04-07 18:35:55 > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : ?Mainconcept Video Media Handler > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, > fltp, 189 kb/s > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : #Mainconcept MP4 Sound Media Handler > Output #0, mov, to 'composite.mov': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42mp41 > encoder : Lavf55.1.100 > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 [SAR > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : ?Mainconcept Video Media Handler > Stream mapping: > Stream #0:0 -> #0:0 (h264 -> png) > Press [q] to stop, [?] for help > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > bitrate=3177.5kbits/s dup=2 drop=0 > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead > 0.067323% > > After this, I see only my red background (timbotundelay.mp4) and no alpha > applied. > Thanks so much to anyone who can help me out here. I'm running out of > things to try. I'd particularly like to see how people are using this > filter successfully. If it does not work but should as it works with images than it is bug. And bugs should be reported so they get fixed. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jshupert at pps-inc.com Tue Apr 9 19:21:56 2013 From: jshupert at pps-inc.com (Jim Shupert) Date: Tue, 09 Apr 2013 13:21:56 -0400 Subject: [FFmpeg-user] dvd ntsc complaint mpg2 with lower than preset rate Message-ID: <51644E34.4050304@pps-inc.com> Friends, I wish to know the full parameters that i need to declare to make a dvd compliant mpg2 that I can author with dvdauthor I have success using the preset -target ntsc-dvd but I have some content that is 3 hours long. I wish to lower the data rate to ~ 4.5 Mbps I find I can use a -maxrate of 6mbps but if i go below 6 mbps it fails the below seems to fail ffmpeg -y -i /media/data/dvd-m/Hunt-003-b.mpg -vcodec mpeg2video -pix_fmt yuv420p -target ntsc-dvd -sample_fmt flt -maxrate 4500k -minrate 1200k -aspect 4:3 /media/data/dvd-m/Hunt-003.mpg ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg developers built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1) configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3 --enable-x11grab libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 58.100 / 54. 58.100 libavformat 54. 28.101 / 54. 28.101 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [mpeg @ 0x282d240] max_analyze_duration 5000000 reached at 5005000 Input #0, mpeg, from '/media/data/dvd-m/Hunt-003-b.mpg': Duration: 02:02:21.53, start: 0.994667, bitrate: 7582 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 9000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 448 kb/s [mpeg2video @ 0x282e4e0] Warning min_rate > 0 but min_rate != max_rate isn't recommended! [mpeg2video @ 0x282e4e0] bitrate above max bitrate Output #0, dvd, to '/media/data/dvd-m/Hunt-003.mpg': Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 8:9 DAR 4:3], q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc Stream #0:1: Audio: none, 48000 Hz, stereo, flt Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> mpeg2video) Stream #0:1 -> #0:1 (ac3 -> ac3) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height ----------------------------------- basically I wish to come from an avi to a dvd compliant ( dvdauthor compliant with nav packs ) mpg2 that has a definable datarate not using the datarate in the preset -target ntsc-dvd thanks From cehoyos at ag.or.at Tue Apr 9 20:07:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 18:07:29 +0000 (UTC) Subject: [FFmpeg-user] dvd ntsc complaint mpg2 with lower than preset rate References: <51644E34.4050304@pps-inc.com> Message-ID: Jim Shupert pps-inc.com> writes: > ffmpeg -y -i /media/data/dvd-m/Hunt-003-b.mpg > -vcodec mpeg2video -pix_fmt yuv420p -target ntsc-dvd > -sample_fmt flt -maxrate 4500k -minrate 1200k > -aspect 4:3 /media/data/dvd-m/Hunt-003.mpg (I suspect that -sample_fmt flt is unneeded and you don't have to specify the aspect at least for your input sample.) You have to specify the encoder bitrate if you specify a (muxing) maxrate. Carl Eugen From oliver at fromme.com Tue Apr 9 17:19:15 2013 From: oliver at fromme.com (Oliver Fromme) Date: Tue, 9 Apr 2013 17:19:15 +0200 (CEST) Subject: [FFmpeg-user] Correct fps value for h.264 stream In-Reply-To: Message-ID: <201304091519.r39FJFRS050342@haluter.fromme.com> Carl Eugen Hoyos wrote: > A bug was fixed today so it should be possible now > to set a framerate for raw (h264) input streams. > > Please test and report back, Carl Eugen When I download the "nightly snapshot", I should get that fix, right? So far I've been using the ffmpeg version of the FreeBSD ports collection, which is currently 1.0.5. I downloaded the snapshot and installed it (in parallel to the "standard" version 1.0.5 so I can use both). Unfortunately, it didn't change anything with regard to my problem. To be honest, I don't even know myself what exactly the problem is ... I just see the symptoms: My hardware player can't play the movie, it behaves like continuous fast-forward, and it displays "-1fps" when I press the info button on the remote. That's why i *think* that the fps value of the h.264 stream is wrong. Does a raw h.264 stream even contain an fps value? Does it contain time stamps? I don't know ... Interestingly, ffmpeg 1.0.5 and the new snapshot display very different fps values for the stream: This is ffprobe 1.0.5: $ ffprobe video.h264 ffprobe version 1.0.5 Copyright (c) 2007-2012 the FFmpeg developers [...] libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [h264 @ 0x80a4c1420] Stream #0: not enough frames to estimate rate; consider increasing probesize [h264 @ 0x80a4c1420] Estimating duration from bitrate, this may be inaccurate Input #0, h264, from '/tmp/foo': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (Main), yuv420p, 720x432 [SAR 64:45 DAR 64:27], 6203.12 fps, 90k tbr, 1200k tbn, 180k tbc And this is ffprobe from the snapshot of yesterday: $ ffprobe video.h264 ffprobe version 1.1.git Copyright (c) 2007-2013 the FFmpeg developers [...] libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.100 / 3. 49.100 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [h264 @ 0x80a0c1420] Stream #0: not enough frames to estimate rate; consider increasing probesize Input #0, h264, from '/tmp/foo': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (Main), yuv420p, 720x432 [SAR 64:45 DAR 64:27], 25 fps, 90k tbr, 1200k tbn, 180k tbc So, the former says 6203.12 fps, the latter says 25 fps which *should* be the correct value, but I'm not sure how this is determined by the ffprobe utility. If have (many) other h.264 files for which *both* versions of ffmpeg report the correct fps value. These files play fine. This is also the case when I re-encode the stream with ffmpeg: then both versions report the correct value of 25 fps for the new h.264 stream. So, the question remains: What is wrong with the original h.264 stream? Does it contain some corrupt header values (fps, time stamps, whatever)? And is there a way to correct it without loss, i.e. without re-encoding? Best regards Oliver -- From yanglm488 at hotmail.com Tue Apr 9 17:05:26 2013 From: yanglm488 at hotmail.com (jih488) Date: Tue, 9 Apr 2013 08:05:26 -0700 (PDT) Subject: [FFmpeg-user] The different about time_base in the AVStream and AVCodecContext Message-ID: <1365519926205-4658360.post@n4.nabble.com> Hello everyone! I find the AVRational time_base in the AVStream and AVCodecContext,but what are the differences about time_base in the AVStream and AVCodecContext? BR. Aaron -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/The-different-about-time-base-in-the-AVStream-and-AVCodecContext-tp4658360.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From zbernie at me.com Tue Apr 9 13:23:10 2013 From: zbernie at me.com (Bernard Barton Jr.) Date: Tue, 09 Apr 2013 07:23:10 -0400 Subject: [FFmpeg-user] How to mitigate stream 1 decoding error? In-Reply-To: References: <3BC83510-18CF-40B6-905E-AF0B8F3A83B1@icloud.com> <7879988F-8935-46FA-A800-5CEA1AA9DC03@icloud.com> Message-ID: Yes the cover art displays fine in iTunes. I don't think I want to decode the png, but it's interesting to know that you can decode artwork in an audio file. I don't think I want to maintain a custom version of ffmpeg either, so I'll just live with the error. Thanks for your help with this! On Apr 9, 2013, at 4:35 AM, Carl Eugen Hoyos wrote: > Bernard Barton Jr. icloud.com> writes: > >> http://www1.datafilehost.com/d/5ec321bf > > Does iTunes show the cover correctly? > (A picture of Andy Williams on red background.) > > Workaround to decode the cover art of this > specific sample is to specify the decoder: > $ ffmpeg -vcodec mjpeg -i 04...m4a > > If you only want to silence the error message, > configure FFmpeg with --disable-decoder=png. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Tue Apr 9 20:41:30 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 9 Apr 2013 18:41:30 +0000 (UTC) Subject: [FFmpeg-user] Correct fps value for h.264 stream References: <201304091519.r39FJFRS050342@haluter.fromme.com> Message-ID: Oliver Fromme fromme.com> writes: > So, the question remains: > What is wrong with the original h.264 stream? As I tried to explain there is not necessarily anything wrong with your stream, FFmpeg does not read all kinds of timestamps from H264 (this is apparently not a trivial task). I assumed (as said I am not sure I understand your original problem) that you want to manually fix the framerate when re-encoding your raw h264 stream with ffmpeg (this is at least what users have repeatedly tried to do, I did not even realize for quite some time that it is supposed to work): This was broken for some versions and is fixed in git head. Carl Eugen From fernandoiriase at gmail.com Wed Apr 10 01:09:10 2013 From: fernandoiriase at gmail.com (=?ISO-8859-1?Q?Fernando_Ir=EDas_Escher?=) Date: Tue, 9 Apr 2013 17:09:10 -0600 Subject: [FFmpeg-user] Add white padding with filter_complex Message-ID: I'm using this command to generate a white padding for a grid: * * *ffmpeg -i 1.mp4 -filter_complex "[0:0]pad=iw*2:ih*2::0:0:0xFFFFFF" -c:v libx264 -preset:v veryfast -threads 0 -s 640:360 -map 0:a -c:a libfaac -b:a 96k grid.mp4* Here is the output if it is somehow useful: http://dpaste.com/1052902/ The problem is that it is still showing the default black padding instead of a white one. Any hints what might be going on? ---- Fernando Irias Escher From onemda at gmail.com Wed Apr 10 01:25:44 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 9 Apr 2013 23:25:44 +0000 Subject: [FFmpeg-user] Add white padding with filter_complex In-Reply-To: References: Message-ID: On 4/9/13, Fernando Irias Escher wrote: > I'm using this command to generate a white padding for a grid: > * > * > > *ffmpeg -i 1.mp4 -filter_complex "[0:0]pad=iw*2:ih*2::0:0:0xFFFFFF" -c:v > libx264 -preset:v veryfast -threads 0 -s 640:360 -map 0:a -c:a > libfaac -b:a 96k grid.mp4* > > Here is the output if it is somehow useful: http://dpaste.com/1052902/ > > The problem is that it is still showing the default black padding instead > of a white one. Any hints what might be going on? Your example does not work here. This works fine: ffplay -f lavfi -i testsrc -vf "pad=iw*2:ih*2:iw/2:ih/2:0xFFFFFF" > > ---- > Fernando Irias Escher > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From fernandoiriase at gmail.com Wed Apr 10 01:35:51 2013 From: fernandoiriase at gmail.com (=?ISO-8859-1?Q?Fernando_Ir=EDas_Escher?=) Date: Tue, 9 Apr 2013 17:35:51 -0600 Subject: [FFmpeg-user] Add white padding with filter_complex In-Reply-To: References: Message-ID: You're right. Actually is a stupid bug. I'm adding two colons instead of just one in the padding. This: *pad=iw*2:ih*2::0:0:0xFFFFFF*, should be this: * pad=iw*2:ih*2:0:0:0xFFFFFF.* * * Thanks for making me look closer! On Tue, Apr 9, 2013 at 5:25 PM, Paul B Mahol wrote: > On 4/9/13, Fernando Irias Escher wrote: > > I'm using this command to generate a white padding for a grid: > > * > > * > > > > *ffmpeg -i 1.mp4 -filter_complex "[0:0]pad=iw*2:ih*2::0:0:0xFFFFFF" > -c:v > > libx264 -preset:v veryfast -threads 0 -s 640:360 -map 0:a -c:a > > libfaac -b:a 96k grid.mp4* > > > > Here is the output if it is somehow useful: http://dpaste.com/1052902/ > > > > The problem is that it is still showing the default black padding instead > > of a white one. Any hints what might be going on? > > Your example does not work here. > > This works fine: > > ffplay -f lavfi -i testsrc -vf "pad=iw*2:ih*2:iw/2:ih/2:0xFFFFFF" > > > > > ---- > > Fernando Irias Escher > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lou at lrcd.com Wed Apr 10 02:06:13 2013 From: lou at lrcd.com (Lou) Date: Tue, 9 Apr 2013 16:06:13 -0800 Subject: [FFmpeg-user] Add white padding with filter_complex In-Reply-To: References: Message-ID: <20130409160613.6178d143@lrcd.com> On Tue, 9 Apr 2013 17:09:10 -0600 Fernando Ir?as Escher wrote: > I'm using this command to generate a white padding for a grid: > * > * > > *ffmpeg -i 1.mp4 -filter_complex "[0:0]pad=iw*2:ih*2::0:0:0xFFFFFF" -c:v > libx264 -preset:v veryfast -threads 0 -s 640:360 -map 0:a -c:a > libfaac -b:a 96k grid.mp4* I don't quite understand what you're trying to do with -s 640:360. I think your pad will result in 1280x960. If you want that to be half that size it will be 640x480: the same size as your input. Forcing 640x360 will vertically squish your video. Also, you can avoid using -s outside of your filterchain so you have more control over when the scale is applied: -vf "pad=iw*2:ih*2:0:0:0xFFFFFF,scale=iw/2:-1" If you still want 640x360 you can crop instead of forcing the size with scale. This example will crop from the bottom (default will "center" the crop): -vf "pad=iw*2:ih*2:0:0:0xFFFFFF,scale=iw/2:-1,crop=iw:360:0:0" Since you only have one input and one output, resulting in a simple filtergraph [1], you can use -vf instead of -filter_complex. I'm not sure if using -filter_complex for a simple filtergraph will cause any issues (I don't think it would, but feel free to correct me). [1] http://ffmpeg.org/ffmpeg.html#Simple-filtergraphs From fernandoiriase at gmail.com Wed Apr 10 03:04:55 2013 From: fernandoiriase at gmail.com (=?ISO-8859-1?Q?Fernando_Ir=EDas_Escher?=) Date: Tue, 9 Apr 2013 19:04:55 -0600 Subject: [FFmpeg-user] Add white padding with filter_complex In-Reply-To: <20130409160613.6178d143@lrcd.com> References: <20130409160613.6178d143@lrcd.com> Message-ID: I'm actually going to receive four videos with size 320x180 after they were scaled in a previous step. The goal is to make a grid with four videos, and it will have a size of 640x360. The ffmpeg command I'm using is based on this one: *ffmpeg -f lavfi -i testsrc -f lavfi -i testsrc -f lavfi -i testsrc -f lavfi -i testsrc -filter_complex "[0:0]pad=iw*2:ih*2[a];[1:0]negate[b];[2:0]hflip[c];[3:0]edgedetect[d];[a][b]overlay=w[x];[x][c]overlay=0:h[y];[y][d]overlay=w:h" -y -c:v ffv1 -t 5 multiple_input_grid.avi* * * So the reason I'm using a complex filter is to allow filtering with multiple inputs and one output. The -s flag is kind of redundant once all the calculations were made from the size of the first input in the complex filter, but I have set it just to be explicit about it. Thanks for your insights! On Tue, Apr 9, 2013 at 6:06 PM, Lou wrote: > On Tue, 9 Apr 2013 17:09:10 -0600 > Fernando Ir?as Escher wrote: > > > I'm using this command to generate a white padding for a grid: > > * > > * > > > > *ffmpeg -i 1.mp4 -filter_complex "[0:0]pad=iw*2:ih*2::0:0:0xFFFFFF" > -c:v > > libx264 -preset:v veryfast -threads 0 -s 640:360 -map 0:a -c:a > > libfaac -b:a 96k grid.mp4* > > I don't quite understand what you're trying to do with -s 640:360. I > think your pad will result in 1280x960. If you want that to be half that > size it will be 640x480: the same size as your input. Forcing 640x360 > will vertically squish your video. Also, you can avoid using -s outside > of your filterchain so you have more control over when the scale is > applied: > > -vf "pad=iw*2:ih*2:0:0:0xFFFFFF,scale=iw/2:-1" > > If you still want 640x360 you can crop instead of forcing the size with > scale. This example will crop from the bottom (default will "center" > the crop): > > -vf "pad=iw*2:ih*2:0:0:0xFFFFFF,scale=iw/2:-1,crop=iw:360:0:0" > > Since you only have one input and one output, resulting in a simple > filtergraph [1], you can use -vf instead of -filter_complex. I'm not > sure if using -filter_complex for a simple filtergraph will cause any > issues (I don't think it would, but feel free to correct me). > > [1] http://ffmpeg.org/ffmpeg.html#Simple-filtergraphs > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From stredarts at gmail.com Wed Apr 10 03:58:22 2013 From: stredarts at gmail.com (Stuart Strader) Date: Tue, 9 Apr 2013 18:58:22 -0700 Subject: [FFmpeg-user] Flash Media Server .f4v aspect ratio weirdness. Message-ID: Hi, I'm trying to process webcam videos recorded by Flash Media Server 4.0. I'm having trouble with ffmpeg not preserving the aspect ratio. As you can see below, the info for stream #0.0 shows two aspect ratio entries. One associated with the actual resolution and another from who knows where. ... Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 211 kb/s, SAR 3:4 DAR 4:3, 12.65 fps, 29.97 tbr, 1k tbn, 59.94 tbc ... I can force the aspect ratio to 16:9, however this would be far less than ideal in my application, as users are able to free to choose the aspect ratio they would like to capture in. So, is there any way to sanitize the aspect ratio of these files? Here is the sample file: http://s3.amazonaws.com/ottw_dev/input_default_20130409_230929.f4v ``` ffmpeg -i input_default_20130409_230929.f4v -threads 0 -acodec libfaac -ab 96k -ar 44100 -b:v 700k output.mp4 ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 25 2013 13:34:30 with Apple clang version 4.1 (tags/Apple/clang-421.11.65) (based on LLVM 3.1svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libvpx --enable-librtmp libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f86d901ae00] max_analyze_duration 5000000 reached at 5025000 microseconds Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input_default_20130409_230929.f4v': Metadata: major_brand : f4v minor_version : 0 compatible_brands: isommp42m4v creation_time : 2018-07-29 17:46:23 Duration: 00:00:45.70, start: 0.000000, bitrate: 222 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 211 kb/s, SAR 3:4 DAR 4:3, 12.65 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 2018-07-29 17:46:23 handler_name : MainConcept Stream #0:1(eng): Audio: nellymoser (nmos / 0x736F6D6E), 8000 Hz, mono, flt, 16 kb/s Metadata: creation_time : 2018-07-29 17:46:23 handler_name : NellyMoser Handler Stream #0:2(eng): Data: none (amf0 / 0x30666D61) Metadata: creation_time : 2018-07-29 17:46:23 handler_name : Timed Metadata Handler File 'output.mp4' already exists. Overwrite ? [y/N] y using SAR=3/4 [libx264 @ 0x7f86d904ac00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x7f86d904ac00] profile High, level 3.0 [libx264 @ 0x7f86d904ac00] 264 - core 125 r15137+7 123778f - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=700 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'output.mp4': Metadata: major_brand : f4v minor_version : 0 compatible_brands: isommp42m4v encoder : Lavf54.63.104 Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 640x360 [SAR 3:4 DAR 4:3], q=-1--1, 700 kb/s, 16728413.00 tbn, 29.97 tbc Metadata: creation_time : 2018-07-29 17:46:23 handler_name : MainConcept Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono, s16, 96 kb/s Metadata: creation_time : 2018-07-29 17:46:23 handler_name : NellyMoser Handler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (nellymoser -> libfaac) Press [q] to stop, [?] for help frame= 1372 fps= 67 q=-1.0 Lsize= 4680kB time=00:00:45.74 bitrate= 838.0kbits/s dup=839 drop=0 video:4227kB audio:403kB subtitle:0 global headers:0kB muxing overhead 1.060890% [libx264 @ 0x7f86d904ac00] frame I:6 Avg QP: 7.94 size: 53563 [libx264 @ 0x7f86d904ac00] frame P:447 Avg QP: 9.51 size: 8077 [libx264 @ 0x7f86d904ac00] frame B:919 Avg QP:16.30 size: 431 [libx264 @ 0x7f86d904ac00] consecutive B-frames: 9.8% 1.5% 3.3% 85.4% [libx264 @ 0x7f86d904ac00] mb I I16..4: 19.4% 22.5% 58.1% [libx264 @ 0x7f86d904ac00] mb P I16..4: 0.9% 1.8% 2.9% P16..4: 24.7% 6.9% 4.5% 0.0% 0.0% skip:58.3% [libx264 @ 0x7f86d904ac00] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 16.4% 0.9% 0.2% direct: 0.3% skip:82.1% L0:49.0% L1:47.7% BI: 3.2% [libx264 @ 0x7f86d904ac00] final ratefactor: 11.82 [libx264 @ 0x7f86d904ac00] 8x8 transform intra:30.7% inter:28.3% [libx264 @ 0x7f86d904ac00] coded y,uvDC,uvAC intra: 84.6% 79.7% 72.2% inter: 7.6% 6.1% 3.7% [libx264 @ 0x7f86d904ac00] i16 v,h,dc,p: 34% 19% 17% 30% [libx264 @ 0x7f86d904ac00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 23% 13% 3% 4% 6% 3% 5% 4% [libx264 @ 0x7f86d904ac00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 22% 10% 4% 7% 7% 5% 5% 4% [libx264 @ 0x7f86d904ac00] i8c dc,h,v,p: 35% 23% 28% 15% [libx264 @ 0x7f86d904ac00] Weighted P-Frames: Y:0.7% UV:0.2% [libx264 @ 0x7f86d904ac00] ref P L0: 84.2% 5.6% 8.9% 1.3% 0.0% [libx264 @ 0x7f86d904ac00] ref B L0: 91.4% 8.1% 0.5% [libx264 @ 0x7f86d904ac00] ref B L1: 93.8% 6.2% [libx264 @ 0x7f86d904ac00] kb/s:756.31 ``` From simondaniels23 at gmail.com Wed Apr 10 07:40:59 2013 From: simondaniels23 at gmail.com (Simon Daniels) Date: Tue, 9 Apr 2013 22:40:59 -0700 Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why In-Reply-To: References: Message-ID: Here's the input file: https://dl.dropbox.com/u/28441949/GOPR9605.MP4. Unfortunately it's quite large. On Tue, Apr 9, 2013 at 1:36 AM, Carl Eugen Hoyos wrote: > Simon Daniels gmail.com> writes: > > > All of the sudden many of my users are getting a > > "timebase not supported" error when using ffmpeg > > to trim a video. > > Please provide the input sample (or find the > commit introducing the problem). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From bbutscheidt at yahoo.de Wed Apr 10 10:08:12 2013 From: bbutscheidt at yahoo.de (Bernd Butscheidt) Date: Wed, 10 Apr 2013 09:08:12 +0100 (BST) Subject: [FFmpeg-user] Correct fps value for h.264 stream References: <201304011637.r31GbQM3008151@haluter.fromme.com> Message-ID: <1365581292.76413.YahooMailNeo@web133203.mail.ir2.yahoo.com> >________________________________ > From: Carl Eugen Hoyos >To: ffmpeg-user at ffmpeg.org >Sent: Tuesday, April 2, 2013 9:31 PM >Subject: Re: [FFmpeg-user] Correct fps value for h.264 stream > > >PS: Please do not provide mediainfo output on this >mailing list unless it is absolutely necessary, it >typically only leads to confusion. > Is there an alternative to mediainfo when one wants to show the difference between -fps information given on the container level and -the stream level itselft? As far as I understood, ffprobe/ ffplay only shows (or correctly shows) the information given with the container (e.g. mkv)? From gabri.ns at gmail.com Wed Apr 10 10:22:56 2013 From: gabri.ns at gmail.com (Gabri Nurtinaz Shally) Date: Wed, 10 Apr 2013 15:22:56 +0700 Subject: [FFmpeg-user] Correct fps value for h.264 stream In-Reply-To: <201304011637.r31GbQM3008151@haluter.fromme.com> References: <201304011637.r31GbQM3008151@haluter.fromme.com> Message-ID: On Apr 2, 2013 1:15 AM, "Oliver Fromme" wrote: > > Hello, > > This is probably a very special problem, but maybe someone > has an answer ... > > I've got an .m4v file that contains a h.264 video stream. > First thing I did is to convert it to .mkv because I can > handle Matroska files much better. So far, so good. > since your purpose is canging container, have you try to use the m4v file directly?! ffmpeg -i input.m4v -c:a copy -c:v copy output.mkv or you can use mkvtoolnix to muxing stream into mkv container From cehoyos at ag.or.at Wed Apr 10 10:27:21 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 10 Apr 2013 08:27:21 +0000 (UTC) Subject: [FFmpeg-user] Flash Media Server .f4v aspect ratio weirdness. References: Message-ID: Stuart Strader gmail.com> writes: > I'm trying to process webcam videos recorded by > Flash Media Server 4.0. I'm having trouble with > ffmpeg not preserving the aspect ratio. > > As you can see below, the info for stream #0.0 shows > two aspect ratio entries. One associated with the > actual resolution and another from who knows where. The track header of your f4v file (see chapter 2.5.1 of the f4v specification) specifies a width of 320 and a height of 240 which corresponds to a display aspect ratio of 4:3. If you have any indication that the width and height of the tkhd atom should be generally ignored in f4v files please share it! I am curious: nellymoser is not specified for f4v (I have seen samples before), didn't you have to use some "trick" to save the stream? Could that be the reason that an incorrect track header is written? Carl Eugen From cehoyos at ag.or.at Wed Apr 10 10:37:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 10 Apr 2013 08:37:55 +0000 (UTC) Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why References: Message-ID: Simon Daniels gmail.com> writes: > "-r", > "59.94006065616729", Please use -r 60000/1001 to specify 59.9 fps Carl Eugen From lulebo at gmail.com Wed Apr 10 10:46:53 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Wed, 10 Apr 2013 10:46:53 +0200 Subject: [FFmpeg-user] Segmenting using pipe In-Reply-To: References: Message-ID: > I have been experimenting with encoding a single file and removing parts of the stream. I need this function when encoding material with breaks for commercials to remove bars and other funky stuff and sync with the subtitles. > > I have done some tests with piping (windows), and I can get this to work pretty well for two arbitrary segments as an example: > > (ffmpeg -i inputfile.mp4 -ss 00:10:00 -t 60 -bsf h264_mp4toannexb -vcodec copy -acodec copy -f mpegts - > ffmpeg -i inputfile.mp4 -ss 00:20:00 -t 60 -bsf h264_mp4toannexb -vcodec copy -acodec copy -f mpegts - )| ffmpeg -y -f mpegts -i - -c:v libx264 -preset veryfast -x264opts keyint=75:min-keyint=75:no-scenecut -b:v 704k -minrate 704k -maxrate 704k -bufsize 1500k -profile:v baseline -aspect 16:9 -c:a libvo_aacenc -b:a 96k -ac 2 test.mp4 > > (this will create a two minute file from two different sections of the input file) > > The problem is the codec copying in the first parts. Since the timecodes might begin in the middle of a GOP, the final ffmpeg will not know what to do until the first I-frame shows up, so it just shows a freeze frame until that happens. > > There is a requirement of a fixed gop size of 3 secs for the final output file, so I need to do the encoding again for the final step, no cat can be used on segments. > > Is there a good intermediate codec to decode to in the first step? (Preferrably uncompressed to gain performance) I have found yuv4mpegvideo but that only does video. I need the audio as well. Anyone got any smart ideas? Something that would solve this problem is if ffmpeg could fix(reencode) the first GOP in case it doesn't start on a keyframe while doing a codec copy on the remaining video. I have not found any info on this, so I guess it can't. If I would encode into a high bitrate i-frame-only in the first step, it would probably work. But that would kill performance. I think ffmpeg would benefit a lot from some form of edl and the possibility to skip over parts of the input. From cehoyos at ag.or.at Wed Apr 10 10:55:09 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 10 Apr 2013 08:55:09 +0000 (UTC) Subject: [FFmpeg-user] Correct fps value for h.264 stream References: <201304011637.r31GbQM3008151@haluter.fromme.com> <1365581292.76413.YahooMailNeo@web133203.mail.ir2.yahoo.com> Message-ID: Bernd Butscheidt yahoo.de> writes: > >PS: Please do not provide mediainfo output on this > >mailing list unless it is absolutely necessary, it > >typically only leads to confusion. > > Is there an alternative to mediainfo when one wants > to show the difference between > > -fps information given on the container level and > > -the stream level itselft? Usually what happens is the following: A user tries to play or reencode a media file and sees (possibly heavy) A/V desync. He tries to understand why this happens, makes several tests and finds out that mediainfo reports a different frame rate than ffmpeg. His conclusion is that the reason for the A/V desync is that ffmpeg reads an incorrect frame rate from the input file, so he opens a bug report (or writes to this mailing list) with the following information: * FFmpeg reports fps x * mediainfo reports fps y -> there is a bug in FFmpeg What actually happens is that FFmpeg reads incorrect timestamps from the input file and calculates an incorrect frame rate. Since the file in question typically does not specify a frame rate at all (it is often not a value written in a file but a value that gets calculated), above report makes no sense to the developers and does not get any attention. It is easier for all parties if the user reports his actual problem (A/V desync) and instead of reporting mediainfo output, reports that QT/Real/ WMP/vlc/MPlayer play the video with a different (correct) speed. (Note that this was completely different with MPlayer which iirc uses the frame rate to calculate video time stamps, so for MPlayer the assumption "wrong fps -> A/V desync" was often correct.) So there may be cases where such output is needed but I don't remember one atm (after having worked on many reports that did contain mediainfo output) and it is at least very unusual that it is useful. > As far as I understood, ffprobe/ ffplay only > shows (or correctly shows) the information > given with the container (e.g. mkv)? I may misunderstand but I believe it is exactly the opposite, ie it typically shows the actual calculated frame rate of the video, but a sample proving me wrong is of course welcome! Carl Eugen From sendtofaraway at qq.com Wed Apr 10 13:27:19 2013 From: sendtofaraway at qq.com (=?ISO-8859-1?B?YW5kcm8=?=) Date: Wed, 10 Apr 2013 19:27:19 +0800 Subject: [FFmpeg-user] how to mux MJPEG encoded data into AVI container Message-ID: I'm looking for a way to mux mjpeg (compressed) video data into a video container like mp4 or avi. (I'll also need to add audio in the future) . I do not want decode the MJpeg data first ,and then encode and mux it into AVI container . From timbot at oddlystudios.com Wed Apr 10 15:18:46 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Wed, 10 Apr 2013 09:18:46 -0400 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: Message-ID: Hi Paul, I'm reluctant to report it as a bug, since somebody clearly did a lot of work on this filter. It could be I'm using it wrong as I've found no full examples online despite a lot of searching. I am continuing to test combinations with yuv and rgb, but I thought if anyone has used this filter set perhaps they would be good enough to share (or some more experienced users might be prompted to try it and also share what they find). This person seems to have had a similar problem: http://ffmpeg.org/pipermail/ffmpeg-user/2012-December/012334.html If it turns out more people experience a problem I would certainly file a bug report with everything I've tried. I am now looking at the test units that were applied to the filters prior to them being committed to see if I'm doing something wrong. Clearly they passed so I may find something there. I also thought some folks might have another way of accomplishing the goal, which is essentially mapping the luma information of one video to the alpha channel of another. I've looked at the lut filter as well as the geq filter, and these can certainly write to the alpha channel but I can't see how to pass the luma information to those filters. I think this is why the alphamerge filter was created in the first place - to solve that problem. But if anyone has another approach I'd love to hear about it. Thanks for your reply! -Tim. On Tue, Apr 9, 2013 at 11:18 AM, Paul B Mahol wrote: > On 4/7/13, Tim Willison wrote: > > Hi, this is my first question ever to a mailing list (or anywhere!). > > I have been using ffmpeg for about a year and a half and have figured out > > many things on my own but I'm stumped regarding the alphaextract and > > alphamerge filters. > > I am able to get the proper result every time with images, but have yet > to > > succeed with videos. I have tried many formats and searched extensively > but > > have not found a single documented case of someone successfully using > this > > filter combination and sharing an example. The documentation on the > > ffmpeg-filters page is very sparse, and does not give any details of what > > formats/codecs are supported. > > > > I think this filter would be very useful if I can figure it out. > > > > Step 1 - alphaextract > > This seems to always work, and produces a grayscale video as expected. In > > my tests I have used rgba video, as well as quicktime with animation > codec > > (bgra). > > > > Step 2 - applying the extracted alpha > > This works every time for images. However while the encode completes with > > video it never provides the masking expected by mapping the extracted > alpha > > information to the alpha channel of the target video. In my tests with > > still images, I found that the output format must be a format that > supports > > an alpha channel, and I have tried several with video (which I determined > > by looking at the source code of the filter). > > > > My command (taken from the docs) looks like this: > > > > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov > [alpha]; > > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png > > composite.mov > > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg > > developers > > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > > 4.6.1-9ubuntu3) > > configuration: --enable-gpl --enable-libass --enable-libfaac > > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 > > --enable-nonfree --enable-version3 --enable-libfreetype > > --enable-filter=drawtext > > libavutil 52. 25.100 / 52. 25.100 > > libavcodec 55. 2.100 / 55. 2.100 > > libavformat 55. 1.100 / 55. 1.100 > > libavdevice 55. 0.100 / 55. 0.100 > > libavfilter 3. 49.100 / 3. 49.100 > > libswscale 2. 2.100 / 2. 2.100 > > libswresample 0. 17.102 / 0. 17.102 > > libpostproc 52. 2.100 / 52. 2.100 > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42mp41 > > creation_time : 2013-04-07 18:35:55 > > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : ?Mainconcept Video Media Handler > > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, > > fltp, 189 kb/s > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : #Mainconcept MP4 Sound Media Handler > > Output #0, mov, to 'composite.mov': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42mp41 > > encoder : Lavf55.1.100 > > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 [SAR > > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : ?Mainconcept Video Media Handler > > Stream mapping: > > Stream #0:0 -> #0:0 (h264 -> png) > > Press [q] to stop, [?] for help > > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > > bitrate=3177.5kbits/s dup=2 drop=0 > > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead > > 0.067323% > > > > After this, I see only my red background (timbotundelay.mp4) and no alpha > > applied. > > Thanks so much to anyone who can help me out here. I'm running out of > > things to try. I'd particularly like to see how people are using this > > filter successfully. > > > If it does not work but should as it works with images than it is bug. > And bugs should be reported so they get fixed. > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From ticks.sunny84 at gmail.com Wed Apr 10 15:23:19 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Wed, 10 Apr 2013 18:53:19 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg Message-ID: Hi All I have been trying to create a video using ffmpeg from images on android device. The settings of my H264 codec are as below: c->bit_rate = 500000; c->width = 480; c->height = 320; c->time_base= (AVRational){1,0}; c->gop_size = 12; // emit one intra frame every ten frames c->max_b_frames=1; c->pix_fmt = PIX_FMT_YUV420P; c->codec_type = AVMEDIA_TYPE_VIDEO; c->codec_id = codec->id; c->max_b_frames = 0; c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; c->me_range = 16; c->max_qdiff = 4; c->qmin = 10; c->qmax = 26; c->qcompress = 0.6; c->trellis=0; c->level = 13; c->refs = 5; c->coder_type = 0; c->scenechange_threshold = 0; //new c->flags|=CODEC_FLAG_LOOP_FILTER;//new //c->profile=FF_PROFILE_H264_BASELINE; //Baseline //new c->scenechange_threshold = 40; //new c-> rc_buffer_size = 0; c->gop_size=250; //new c->max_b_frames=1;//new c->me_method=7; c->me_cmp|= 1; c->me_subpel_quality = 6; c->qmax=51; c->keyint_min=25; av_opt_set(c->priv_data,"subq","6",0); av_opt_set(c->priv_data,"crf","20.0",0); av_opt_set(c->priv_data,"weighted_p_pred","0",0); av_opt_set(c->priv_data,"vprofile","baseline",0); av_opt_set(c->priv_data,"preset","medium",0); av_opt_set(c->priv_data,"tune","zerolatency",0); //av_opt_set(c->priv_data,"x264opts","rc-lookahead=0",0); I am not able to understand that if this is a bug in ffmpeg or my code that even after providing "*av_opt_set(c->priv_data,"vprofile","baseline",0);*" and "*av_opt_set(c->priv_data,"profile","baseline",0);*" as was told in the following ticket: http://ffmpeg.org/trac/ffmpeg/ticket/387 but nothing seems to activate the baseline profile. as when i do a ffmpef -i on my video it shows profile as hight. please help. regards, Susheel Tickoo From cehoyos at ag.or.at Wed Apr 10 15:43:16 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 10 Apr 2013 13:43:16 +0000 (UTC) Subject: [FFmpeg-user] how to mux MJPEG encoded data into AVI container References: Message-ID: andro qq.com> writes: > I'm looking for a way to mux mjpeg (compressed) video > data into a video container like mp4 or avi. $ ffmpeg -i input -vcodec mjpeg out.avi > (I'll also need to add audio in the future) . $ ffmpeg -i input -vcodec mjpeg -acodec ac3 out.avi > I do not want decode the MJpeg data first ,and then > encode and mux it into AVI container . $ ffmpeg -i mjpeg -vcodec copy out.avi Generally, it is a good idea to test something before posting on this mailing list and report here if it does not work (includig failing command line and complete, uncut console output). Carl Eugen From lulebo at gmail.com Wed Apr 10 16:04:40 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Wed, 10 Apr 2013 16:04:40 +0200 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: > c->max_b_frames=1; Then down a bit comes > c->max_b_frames = 0; Then comes > c->max_b_frames=1;//new This is just one example of things that look weird in those settings. Looks like you have copy/pasted a lot of different option around. I haven't gone through all the options you set, but for example, baseline can't have b-frames. /Carl From ticks.sunny84 at gmail.com Wed Apr 10 16:23:18 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Wed, 10 Apr 2013 19:53:18 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: Thanks Carl, I will suerly update my code to be more redable and presentable :). So do we need to remove the setting of max_b_frames at all or do we have to set it to c->max_b_frames = 0; ? On Wed, Apr 10, 2013 at 7:34 PM, Carl Lindqvist wrote: > > c->max_b_frames=1; > Then down a bit comes > > c->max_b_frames = 0; > Then comes > > c->max_b_frames=1;//new > > This is just one example of things that look weird in those settings. > Looks like you have copy/pasted a lot of different option around. > > I haven't gone through all the options you set, but for example, > baseline can't have b-frames. > > /Carl > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ticks.sunny84 at gmail.com Wed Apr 10 16:46:48 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Wed, 10 Apr 2013 20:16:48 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: Hi Carl, now i have removed the duplicate fields and here are my new settings: c->width = width; c->height = height; c->time_base= (AVRational){1,0}; c->gop_size = 12; // emit one intra frame every ten frames c->pix_fmt = PIX_FMT_YUV420P; c->codec_type = AVMEDIA_TYPE_VIDEO; c->codec_id = codec->id; c->max_b_frames = 0; c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; c->me_range = 16; c->max_qdiff = 4; c->qmin = 10; c->qcompress = 0.6; c->trellis=0; c->level = 13; c->refs = 5; c->coder_type = 0; c->flags|=CODEC_FLAG_LOOP_FILTER;//new c->scenechange_threshold = 40; //new c-> rc_buffer_size = 0; c->me_method=7; c->me_cmp|= 1; c->me_subpel_quality = 6; c->qmax=51; c->keyint_min=25; av_opt_set(c->priv_data,"subq","6",0); av_opt_set(c->priv_data,"crf","20.0",0); av_opt_set(c->priv_data,"weighted_p_pred","0",0); av_opt_set(c->priv_data,"vprofile","baseline",0); av_opt_set(c->priv_data,"preset","medium",0); av_opt_set(c->priv_data,"tune","zerolatency",0); I am setting max_b_frames=0 and i am also setting "* av_opt_set(c->priv_data,"vprofile","baseline",0);*" Do i need to set the *profile* or *vprofile* tag ? I am confused by the ticket http://ffmpeg.org/trac/ffmpeg/ticket/387 I have created the video using max_b_frames=0 and not setting it at all but in both cases the video is not baseline. Should i use only teh following line? c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; Please help me out. thanks & regards, Susheel Tickoo On Wed, Apr 10, 2013 at 7:53 PM, Susheel Tickoo wrote: > Thanks Carl, > > I will suerly update my code to be more redable and presentable :). So do > we need to remove the setting of max_b_frames at all or do we have to set > it to c->max_b_frames = 0; ? > > On Wed, Apr 10, 2013 at 7:34 PM, Carl Lindqvist wrote: > >> > c->max_b_frames=1; >> Then down a bit comes >> > c->max_b_frames = 0; >> Then comes >> > c->max_b_frames=1;//new >> >> This is just one example of things that look weird in those settings. >> Looks like you have copy/pasted a lot of different option around. >> >> I haven't gone through all the options you set, but for example, >> baseline can't have b-frames. >> >> /Carl >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From stefasab at gmail.com Wed Apr 10 17:51:14 2013 From: stefasab at gmail.com (Stefano Sabatini) Date: Wed, 10 Apr 2013 17:51:14 +0200 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: Message-ID: <20130410155114.GP7301@arborea> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: > Hi, this is my first question ever to a mailing list (or anywhere!). > I have been using ffmpeg for about a year and a half and have figured out > many things on my own but I'm stumped regarding the alphaextract and > alphamerge filters. > I am able to get the proper result every time with images, but have yet to > succeed with videos. I have tried many formats and searched extensively but > have not found a single documented case of someone successfully using this > filter combination and sharing an example. The documentation on the > ffmpeg-filters page is very sparse, and does not give any details of what > formats/codecs are supported. Yes that's because ffmpeg-filters is supposed to only document filters. > > I think this filter would be very useful if I can figure it out. > > Step 1 - alphaextract > This seems to always work, and produces a grayscale video as expected. In > my tests I have used rgba video, as well as quicktime with animation codec > (bgra). > > Step 2 - applying the extracted alpha > This works every time for images. However while the encode completes with > video it never provides the masking expected by mapping the extracted alpha > information to the alpha channel of the target video. In my tests with > still images, I found that the output format must be a format that supports > an alpha channel, and I have tried several with video (which I determined > by looking at the source code of the filter). The question is, are you sure your output codec supports alpha (most don't, indeed I couldn't even tell which ones support alpha). You can try with rawvideo and make sure it is not a codec bug. > My command (taken from the docs) looks like this: > > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov [alpha]; > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png composite.mov > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > 4.6.1-9ubuntu3) > configuration: --enable-gpl --enable-libass --enable-libfaac > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 > --enable-nonfree --enable-version3 --enable-libfreetype > --enable-filter=drawtext > libavutil 52. 25.100 / 52. 25.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 1.100 / 55. 1.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 49.100 / 3. 49.100 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42mp41 > creation_time : 2013-04-07 18:35:55 > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : ?Mainconcept Video Media Handler > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, > fltp, 189 kb/s > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : #Mainconcept MP4 Sound Media Handler > Output #0, mov, to 'composite.mov': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42mp41 > encoder : Lavf55.1.100 > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 [SAR > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > Metadata: > creation_time : 2013-04-07 18:35:55 > handler_name : ?Mainconcept Video Media Handler > Stream mapping: > Stream #0:0 -> #0:0 (h264 -> png) > Press [q] to stop, [?] for help > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > bitrate=3177.5kbits/s dup=2 drop=0 > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead > 0.067323% > > After this, I see only my red background (timbotundelay.mp4) and no alpha > applied. I tried a variant of your command and seems to work fine here. Also note that you need a background to show an image with overlay. Something like: ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" should do. > Thanks so much to anyone who can help me out here. I'm running out of > things to try. I'd particularly like to see how people are using this > filter successfully. From lukas.oesterreicher at inode.at Wed Apr 10 19:00:03 2013 From: lukas.oesterreicher at inode.at (=?iso-8859-15?Q?Lukas_=D6sterreicher?=) Date: Wed, 10 Apr 2013 19:00:03 +0200 Subject: [FFmpeg-user] Audio transcoding with metadata and album cover image Message-ID: <405017730.20130410190003@inode.at> Hi. I'm trying to re-encode from mp3 to he-aac and include metadata and an album cover image. All but the last works for me like this: ffmpeg -i 150.mp3 -c:a libaacplus -b:a 48k -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a However I can't get the last one to work. I tried something like this here: http://ffmpeg.org/ffmpeg-formats.html#mp3 which is in my case ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k -map 0 -map 1 -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a But that gives me this error: Encoder (codec none) not found for output stream #0:1 Any suggestions? Thanx, Lukas From lou at lrcd.com Wed Apr 10 21:17:00 2013 From: lou at lrcd.com (Lou) Date: Wed, 10 Apr 2013 11:17:00 -0800 Subject: [FFmpeg-user] Audio transcoding with metadata and album cover image In-Reply-To: <405017730.20130410190003@inode.at> References: <405017730.20130410190003@inode.at> Message-ID: <20130410111700.26092eae@lrcd.com> On Wed, 10 Apr 2013 19:00:03 +0200 Lukas ?sterreicher wrote: > Hi. > > I'm trying to re-encode from mp3 to he-aac and include metadata and an > album cover image. > > All but the last works for me like this: > ffmpeg -i 150.mp3 -c:a libaacplus -b:a 48k -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a > > However I can't get the last one to work. > I tried something like this here: http://ffmpeg.org/ffmpeg-formats.html#mp3 > which is in my case > ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k -map 0 -map 1 -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a > > But that gives me this error: > Encoder (codec none) not found for output stream #0:1 > > Any suggestions? Please show your complete ffmpeg console output. From adunaic at hotmail.co.uk Wed Apr 10 15:39:23 2013 From: adunaic at hotmail.co.uk (adunaic) Date: Wed, 10 Apr 2013 06:39:23 -0700 (PDT) Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) In-Reply-To: <4F51B7E9.5080005@gmail.com> References: <4F46A788.9070406@gmail.com> <4F51B7E9.5080005@gmail.com> Message-ID: <1365601163170-4658383.post@n4.nabble.com> I thought it best to re-open this than start a new. Do you know if this supported at all or will be? From Oggs pages they imply that dirac can be placed in an ogg container but I still get the error: Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted (ffmpeg version 1.0.5 ) Any thoughts? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Problem-transcoding-from-MP4-H264-AAC-to-OGG-DIRAC-FLAC-tp4416502p4658383.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ml at founder.com Wed Apr 10 11:03:41 2013 From: ml at founder.com (=?gb2312?B?bWFsaSjC7cD2KS5yZXBlYXQ=?=) Date: Wed, 10 Apr 2013 17:03:41 +0800 Subject: [FFmpeg-user] X264 chroma error Message-ID: <6AAE0F2CD95AC543A63780BBA5B1C9388F2D22@mail06.HOLD.FOUNDER.COM> Dears, Here is my string and dubug information: ---------------------------- >ffmpeg.exe -threads 16 -y -i test_lt_hh.ts -s 176x144 -r 12 -vcodec libx264 -b:v 112k -bt 112k -flags +loop -me_method umh -minrate 80k -maxrate 150k -flags +loop -cmp +chroma -qcomp 0.4 -b_strategy 1 -refs 4 -qblur 0.01 -qdiff 0.1 -me_range 16 -g 4 -keyint_min 1 -sc_threshold 40 -i_qfactor 0.71 -rc_eq "blurCplx^(1-qComp)" -pix_fmt yuv420p -qmin 2 -qmax 51 -coder 1 -bufsize 300k -level 40 -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -subq 5 -vprofile baseline -fmp4 -acodec libfaac -ab 16k -ar 22050 -ac 1 test.mp4 ------------------------------------ ffmpeg version 1.1.git Copyright (c) 2000-2013 the FFmpeg developers built on Apr 10 2013 15:06:10 with gcc 4.6.2 (GCC) configuration: --enable-shared --disable-static --enable-ffplay --enable-nonfree --enable-memalign -hack --enable-libmp3lame --enable-gpl --enable-libx264 --enable-libxvid --enable-libfaac --enable-l ibvo-aacenc --enable-version3 --extra-cflags='-If:/ffmpeg-BianYi/ffmpeg-20130126-git-c46943e/include ' --extra-ldflags='-Lf:/ffmpeg-BianYi/ffmpeg-20130126-git-c46943e/lib' libavutil 52. 17.100 / 52. 17.100 libavcodec 54. 91.100 / 54. 91.100 libavformat 54. 61.104 / 54. 61.104 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 34.101 / 3. 34.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mpegts @ 003ff160] max_analyze_duration 5000000 reached at 5000000 microseconds Input #0, mpegts, from 'test_lt_hh.ts': Duration: 00:02:13.04, start: 0.250000, bitrate: 1764 kb/s Program 1 Stream #0:0[0x3e9]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 12:11 DAR 15:11], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x3ea]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, mono, fltp, 63 kb/s [NULL @ 01bb9520] [Eval @ 0022b3b8] Undefined constant or missing '(' in 'chroma' [NULL @ 01bb9520] Unable to parse option value "+chroma" [NULL @ 01bb9520] Error setting option cmp to value +chroma. Output #0, mp4, to 'test.mp4': Stream #0:0: Video: none, yuv420p, 176x144 [SAR 135:121 DAR 15:11], q=-1--1, 90k tbn, 12 tbc Stream #0:1: Audio: none, 22050 Hz, mono, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> libfaac) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, ra te, width or height ------------------------------------------------------- Thanks for any help someone can provide. Best regards! Yours,mary From oliver at fromme.com Wed Apr 10 10:03:54 2013 From: oliver at fromme.com (Oliver Fromme) Date: Wed, 10 Apr 2013 10:03:54 +0200 (CEST) Subject: [FFmpeg-user] Correct fps value for h.264 stream In-Reply-To: Message-ID: <201304100803.r3A83srj055312@haluter.fromme.com> Carl Eugen Hoyos wrote: > Oliver Fromme fromme.com> writes: > > > So, the question remains: > > What is wrong with the original h.264 stream? > > As I tried to explain there is not necessarily > anything wrong with your stream, FFmpeg does not > read all kinds of timestamps from H264 (this is > apparently not a trivial task). I see. However, my hardware media player (black box) doesn't play that particular file correctly either, while it doesn't have any problems with other h.264 files. That's why I suspect there is something wrong with that particular stream. > I assumed (as said I am not sure I understand > your original problem) that you want to > manually fix the framerate when re-encoding > your raw h264 stream with ffmpeg (this is at > least what users have repeatedly tried to do, > I did not even realize for quite some time > that it is supposed to work): This was broken > for some versions and is fixed in git head. Ok. Indeed, when I re-encode the stream, the problem disappears, even when I use the "old" ffmpeg version from the FreeBSD ports collection (1.0.5). So ffmpeg works perfectly fine when re-encoding. My point is just that I would like to avoid re-encoding the stream, because this is not lossless. My question was whether it is possible to fix the stream without re-encoding. Unfortunately it seems that's not possible. (But that's not ffmpeg's fault, of course; it's probably rather a limitation of the x264 code, I guess.) Anyway -- Thank you very much for your efforts! Best regards Oliver -- From lou at lrcd.com Wed Apr 10 22:13:23 2013 From: lou at lrcd.com (Lou) Date: Wed, 10 Apr 2013 12:13:23 -0800 Subject: [FFmpeg-user] X264 chroma error In-Reply-To: <6AAE0F2CD95AC543A63780BBA5B1C9388F2D22@mail06.HOLD.FOUNDER.COM> References: <6AAE0F2CD95AC543A63780BBA5B1C9388F2D22@mail06.HOLD.FOUNDER.COM> Message-ID: <20130410121323.6dc26e08@lrcd.com> On Wed, 10 Apr 2013 17:03:41 +0800 mali(??).repeat wrote: > Dears, > > Here is my string and dubug information: > > ---------------------------- > > >ffmpeg.exe -threads 16 -y -i test_lt_hh.ts -s 176x144 -r 12 -vcodec libx264 -b:v 112k -bt 112k -flags +loop -me_method umh -minrate 80k -maxrate 150k -flags +loop > > -cmp +chroma -qcomp 0.4 -b_strategy 1 -refs 4 -qblur 0.01 -qdiff 0.1 -me_range 16 -g 4 -keyint_min 1 -sc_threshold 40 -i_qfactor 0.71 > > -rc_eq "blurCplx^(1-qComp)" -pix_fmt yuv420p -qmin 2 -qmax 51 -coder 1 -bufsize 300k -level 40 -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -subq 5 > > -vprofile baseline -fmp4 -acodec libfaac -ab 16k -ar 22050 -ac 1 test.mp4 > > ------------------------------------ It is not recommended to set each x264 option. Use the x264 encoding presets instead. They were created by the x264 developers so users will not have to declare a large number of confusing and outdated options. They make your commands much shorter, saner, and are kept up-to-date with any changes, additions, etc. Your command will then be shortened to: ffmpeg.exe -y -i test_lt_hh.ts -vcodec libx264 -preset medium \ -profile:v baseline -level 40 -b:v 112k -minrate 80k -maxrate 150k \ -acodec libfaac -b:a 16k -ar 22050 -ac 1 test.mp4 "-bt" is ignored by x264. You can see a list of current presets with "-preset help", and what settings they apply with "x264 --fullhelp". See: http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide From cehoyos at ag.or.at Thu Apr 11 00:03:32 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 10 Apr 2013 22:03:32 +0000 (UTC) Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) References: <4F46A788.9070406@gmail.com> <4F51B7E9.5080005@gmail.com> <1365601163170-4658383.post@n4.nabble.com> Message-ID: adunaic hotmail.co.uk> writes: > From Oggs pages they imply that dirac can be placed > in an ogg container but I still get the error: FFmpeg currently does not support muxing dirac in ogg, patch welcome! > Could not write header for output file #0 (incorrect codec parameters ?): > Operation not permitted > > (ffmpeg version 1.0.5 ) (This is outdated.) > Any thoughts? For future questions: Please always provide your failing command line together with complete, uncut console output. Carl Eugen From oliver at fromme.com Wed Apr 10 22:08:45 2013 From: oliver at fromme.com (Oliver Fromme) Date: Wed, 10 Apr 2013 22:08:45 +0200 (CEST) Subject: [FFmpeg-user] Correct fps value for h.264 stream Message-ID: <201304102008.r3AK8jPS059154@haluter.fromme.com> Gabri Nurtinaz Shally wrote: > On Apr 2, 2013 1:15 AM, "Oliver Fromme" wrote: > > I've got an .m4v file that contains a h.264 video stream. > > First thing I did is to convert it to .mkv because I can > > handle Matroska files much better. So far, so good. > > since your purpose is canging container, have you try to use the m4v file > directly?! > > ffmpeg -i input.m4v -c:a copy -c:v copy output.mkv > > or you can use mkvtoolnix to muxing stream into mkv container Yes, in fact I'm using my own script that's based on mkvmerge (from the mkvtoolnix package). However, it doesn't matter at all what tool I use to change the container, because the problem is not in the container. The problem seems to be caused by wrong or missing time stamps (these are called PTS, right?) in the h.264 stream. I tried to have a close look at the h.264 stream, using the report tools from the tstools package (tstools.berlios.de). The output is somewhat complicated, but it seems to indicate that the stream does not have any time stamps at all. This might explain why different tools display different fps values for the raw stream, and why mplayer plays it fine while my hardware media player doesn't. I'm not an h.264 expert, though, so I might be completely wrong. Maybe raw h.264 isn't supposed to contain time stamps at all. Maybe I have to put it into PES format (some googling seems to confirm that PES has time stamps), but I have no idea how to do that. Best regards Oliver From pj at illuminatedcomputing.com Thu Apr 11 00:54:39 2013 From: pj at illuminatedcomputing.com (Paul Jungwirth) Date: Wed, 10 Apr 2013 15:54:39 -0700 Subject: [FFmpeg-user] Concatenating mp4s Quickly Message-ID: Hello, I'm trying to use ffmpeg to concatenate about a half-dozen mp4 files in order to create a longer mp4. I don't need to do any other processing other than concating the files. Right now, my approach is to make mpgs out of each input file, literally `cat` them together, then re-encode back to mp4. But this takes over a minute per output video (for output about ~7 minutes long), and I'd like to get it down to just a few seconds if possible. Is there a better approach? I can pre-encode the input files to whatever format is appropriate; the only time that matters is the time to take all the inputs (in whatever format) and produce a final mp4. Any suggestions? Thanks, Paul -- _________________________________ Pulchritudo splendor veritatis. From 23liam at googlemail.com Thu Apr 11 01:04:28 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Thu, 11 Apr 2013 00:04:28 +0100 Subject: [FFmpeg-user] Concatenating mp4s Quickly In-Reply-To: References: Message-ID: >I'm trying to use ffmpeg to concatenate about a half-dozen mp4 files >in order to create a longer mp4. I don't need to do any other >processing other than concating the files. Look here: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join%2C%20merge%29%20media%20files#demuxer From andrew at sonar.sg Thu Apr 11 02:06:19 2013 From: andrew at sonar.sg (Andrew Crothers) Date: Thu, 11 Apr 2013 08:06:19 +0800 Subject: [FFmpeg-user] Unable to pipe fdk_aac output Message-ID: <4C7C18CF-7DDE-4B4E-B248-838436D9C3D0@sonar.sg> Hi all, I having major problems piping aac output. No problems writing directly to a file, but I just can't pipe! Any help out there? Input: ffmpeg -i source.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k pipe:1 Result: ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 11 2013 06:57:44 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-libaacplus --enable-libfdk_aac --enable-libopus --enable-libfaac --enable-nonfree libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 [wav @ 0x7fec2403d800] max_analyze_duration 5000000 reached at 5015510 microseconds Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'source.wav': Metadata: copyright : ? ueberschall 2003 Duration: 00:00:30.00, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s [NULL @ 0x7fec2405e200] Unable to find a suitable output format for 'pipe:1' pipe:1: Invalid argument Regards, Andrew From timothygu99 at gmail.com Thu Apr 11 05:34:00 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Thu, 11 Apr 2013 11:34:00 +0800 Subject: [FFmpeg-user] Unable to pipe fdk_aac output In-Reply-To: <4C7C18CF-7DDE-4B4E-B248-838436D9C3D0@sonar.sg> References: <4C7C18CF-7DDE-4B4E-B248-838436D9C3D0@sonar.sg> Message-ID: On Apr 10, 2013 5:07 PM, "Andrew Crothers" wrote: > > Hi all, > > I having major problems piping aac output. > > No problems writing directly to a file, but I just can't pipe! > > Any help out there? > > > Input: > > ffmpeg -i source.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k pipe:1 > > Result: > > ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers > built on Apr 11 2013 06:57:44 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) > configuration: --enable-libaacplus --enable-libfdk_aac --enable-libopus --enable-libfaac --enable-nonfree > libavutil 52. 18.100 / 52. 18.100 > libavcodec 54. 92.100 / 54. 92.100 > libavformat 54. 63.104 / 54. 63.104 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 42.103 / 3. 42.103 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > [wav @ 0x7fec2403d800] max_analyze_duration 5000000 reached at 5015510 microseconds > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, wav, from 'source.wav': > Metadata: > copyright : ? ueberschall 2003 > Duration: 00:00:30.00, bitrate: 1411 kb/s > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s > [NULL @ 0x7fec2405e200] Unable to find a suitable output format for 'pipe:1' > pipe:1: Invalid argument You did not specify the output format. Try this: ffmpeg -i source.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k -f adts pipe:1 Timothy Gu From ticks.sunny84 at gmail.com Thu Apr 11 07:00:43 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Thu, 11 Apr 2013 10:30:43 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: Hi Carl, Did you got any time to look into this? On Wed, Apr 10, 2013 at 8:16 PM, Susheel Tickoo wrote: > Hi Carl, > now i have removed the duplicate fields and here are my new settings: > > c->width = width; > c->height = height; > > c->time_base= (AVRational){1,0}; > c->gop_size = 12; // emit one intra frame every ten frames > c->pix_fmt = PIX_FMT_YUV420P; > c->codec_type = AVMEDIA_TYPE_VIDEO; > c->codec_id = codec->id; > c->max_b_frames = 0; > c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; > c->me_range = 16; > c->max_qdiff = 4; > c->qmin = 10; > c->qcompress = 0.6; > c->trellis=0; > c->level = 13; > c->refs = 5; > c->coder_type = 0; > c->flags|=CODEC_FLAG_LOOP_FILTER;//new > > c->scenechange_threshold = 40; //new > c-> rc_buffer_size = 0; > c->me_method=7; > c->me_cmp|= 1; > c->me_subpel_quality = 6; > c->qmax=51; > c->keyint_min=25; > av_opt_set(c->priv_data,"subq","6",0); > av_opt_set(c->priv_data,"crf","20.0",0); > av_opt_set(c->priv_data,"weighted_p_pred","0",0); > av_opt_set(c->priv_data,"vprofile","baseline",0); > av_opt_set(c->priv_data,"preset","medium",0); > av_opt_set(c->priv_data,"tune","zerolatency",0); > > I am setting max_b_frames=0 and i am also setting "* > av_opt_set(c->priv_data,"vprofile","baseline",0);*" > > Do i need to set the *profile* or *vprofile* tag ? > I am confused by the ticket http://ffmpeg.org/trac/ffmpeg/ticket/387 > I have created the video using max_b_frames=0 and not setting it at all > but in both cases the video is not baseline. > > Should i use only teh following line? > c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; > > Please help me out. > thanks & regards, > Susheel Tickoo > > > > On Wed, Apr 10, 2013 at 7:53 PM, Susheel Tickoo wrote: > >> Thanks Carl, >> >> I will suerly update my code to be more redable and presentable :). So do >> we need to remove the setting of max_b_frames at all or do we have to set >> it to c->max_b_frames = 0; ? >> >> On Wed, Apr 10, 2013 at 7:34 PM, Carl Lindqvist wrote: >> >>> > c->max_b_frames=1; >>> Then down a bit comes >>> > c->max_b_frames = 0; >>> Then comes >>> > c->max_b_frames=1;//new >>> >>> This is just one example of things that look weird in those settings. >>> Looks like you have copy/pasted a lot of different option around. >>> >>> I haven't gone through all the options you set, but for example, >>> baseline can't have b-frames. >>> >>> /Carl >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>> >> >> > From phpdev at ehrhardt.nl Thu Apr 11 07:10:48 2013 From: phpdev at ehrhardt.nl (Jan Ehrhardt) Date: Thu, 11 Apr 2013 07:10:48 +0200 Subject: [FFmpeg-user] Concatenating mp4s Quickly References: Message-ID: <90hcm8dt9il0hu7cb7e8i41bkeu9jn1t1i@4ax.com> Paul Jungwirth in gmane.comp.video.ffmpeg.user (Wed, 10 Apr 2013 15:54:39 -0700): >I'm trying to use ffmpeg to concatenate about a half-dozen mp4 files >in order to create a longer mp4. I don't need to do any other >processing other than concating the files. You are better off with mp4box. http://gpac.wp.mines-telecom.fr/mp4box/mp4box-documentation/ $ mp4box -align-cat -cat first.mp4 -cat second.mp4 -new concatenated.mp4 Jan From mimeini at gmail.com Thu Apr 11 07:38:15 2013 From: mimeini at gmail.com (mikkel meinike) Date: Thu, 11 Apr 2013 07:38:15 +0200 Subject: [FFmpeg-user] Concatenating mp4s Quickly In-Reply-To: <90hcm8dt9il0hu7cb7e8i41bkeu9jn1t1i@4ax.com> References: <90hcm8dt9il0hu7cb7e8i41bkeu9jn1t1i@4ax.com> Message-ID: Thank you Jan I think that mp4box could be usfull to me too. /mikkel Den 11/04/2013 07.11 skrev "Jan Ehrhardt" : > Paul Jungwirth in gmane.comp.video.ffmpeg.user (Wed, 10 Apr 2013 > 15:54:39 -0700): > >I'm trying to use ffmpeg to concatenate about a half-dozen mp4 files > >in order to create a longer mp4. I don't need to do any other > >processing other than concating the files. > > You are better off with mp4box. > http://gpac.wp.mines-telecom.fr/mp4box/mp4box-documentation/ > > $ mp4box -align-cat -cat first.mp4 -cat second.mp4 -new concatenated.mp4 > > Jan > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lukas.oesterreicher at inode.at Thu Apr 11 09:47:00 2013 From: lukas.oesterreicher at inode.at (=?utf-8?Q?Lukas_=C3=96sterreicher?=) Date: Thu, 11 Apr 2013 09:47:00 +0200 Subject: [FFmpeg-user] Audio transcoding with metadata and album cover image In-Reply-To: <20130410111700.26092eae@lrcd.com> References: <405017730.20130410190003@inode.at> <20130410111700.26092eae@lrcd.com> Message-ID: <941864805.20130411094700@inode.at> Hi. Here's my latest try: morkeleb at morkeleb-VirtualBox:~/Music$ ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k -map 0 -map 1 -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a ffmpeg version N-51694-g4e0130f Copyright (c) 2000-2013 the FFmpeg developers built on Apr 10 2013 16:12:06 with gcc 4.7 (Ubuntu/Linaro 4.7.2-2ubuntu1) configuration: --enable-gpl --enable-libaacplus --enable-libmp3lame --enable-version3 --enable-nonfree --enable-static libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.101 / 3. 49.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mp3 @ 0xa7d6240] max_analyze_duration 5000000 reached at 5015510 microseconds Input #0, mp3, from '150.mp3': Metadata: title : Jesus Gave Me Water artist : Sam Cooke & The Soul Stirrers album : Portrait Of A Legend Duration: 00:02:29.66, start: 0.000000, bitrate: 135 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 135 kb/s Input #1, image2, from '150.jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #1:0: Video: mjpeg, yuvj420p, 200x200 [SAR 1:1 DAR 1:1], 25 tbr, 25 tbn, 25 tbc File '150_encoded.m4a' already exists. Overwrite ? [y/N] Y Output #0, ipod, to '150_encoded.m4a': Metadata: title : Jesus Gave Me Water artist : Sam Cooke & The Soul Stirrers album : Portrait Of A Legend Stream #0:0: Audio: none, 44100 Hz, stereo, s16, 128 kb/s Stream #0:1: Video: none, q=2-31, 128 kb/s, 90k tbn Metadata: title : track title author : artist name Stream mapping: Stream #0:0 -> #0:0 (mp3 -> libaacplus) Stream #1:0 -> #0:1 (mjpeg -> ?) Encoder (codec none) not found for output stream #0:1 Hope this helps! Lukas > On Wed, 10 Apr 2013 19:00:03 +0200 > Lukas ?sterreicher wrote: >> Hi. >> >> I'm trying to re-encode from mp3 to he-aac and include metadata and an >> album cover image. >> >> All but the last works for me like this: >> ffmpeg -i 150.mp3 -c:a libaacplus -b:a 48k -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a >> >> However I can't get the last one to work. >> I tried something like this here: http://ffmpeg.org/ffmpeg-formats.html#mp3 >> which is in my case >> ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k -map 0 -map 1 -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a >> >> But that gives me this error: >> Encoder (codec none) not found for output stream #0:1 >> >> Any suggestions? > Please show your complete ffmpeg console output. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Mit freundlichen Gr??en Lukas ?sterreicher mailto:lukas.oesterreicher at inode.at From cehoyos at ag.or.at Thu Apr 11 09:58:08 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 11 Apr 2013 07:58:08 +0000 (UTC) Subject: [FFmpeg-user] Audio transcoding with metadata and album cover image References: <405017730.20130410190003@inode.at> <20130410111700.26092eae@lrcd.com> <941864805.20130411094700@inode.at> Message-ID: Hi Lukas! Lukas ?sterreicher inode.at> writes: > $ ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k > -map 0 -map 1 -metadata:s:v title="track title" > -metadata:s:v author="artist name" 150_encoded.m4a > Stream mapping: > Stream #0:0 -> #0:0 (mp3 -> libaacplus) > Stream #1:0 -> #0:1 (mjpeg -> ?) FFmpeg does not know what to do with your jpeg, specify either "-vcodec copy" (if jpeg is ok), or "-vcodec png" (if you prefer a png for some reason). Please do not top-post on this mailing list. Carl Eugen From lukas.oesterreicher at inode.at Thu Apr 11 10:45:26 2013 From: lukas.oesterreicher at inode.at (=?utf-8?Q?Lukas_=C3=96sterreicher?=) Date: Thu, 11 Apr 2013 10:45:26 +0200 Subject: [FFmpeg-user] Audio transcoding with metadata and album cover image In-Reply-To: References: <405017730.20130410190003@inode.at> <20130410111700.26092eae@lrcd.com> <941864805.20130411094700@inode.at> Message-ID: <1412792286.20130411104526@inode.at> > Hi Lukas! > Lukas ?sterreicher inode.at> writes: >> $ ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k >> -map 0 -map 1 -metadata:s:v title="track title" >> -metadata:s:v author="artist name" 150_encoded.m4a >> Stream mapping: >> Stream #0:0 -> #0:0 (mp3 -> libaacplus) >> Stream #1:0 -> #0:1 (mjpeg -> ?) > FFmpeg does not know what to do with your jpeg, > specify either "-vcodec copy" (if jpeg is ok), > or "-vcodec png" (if you prefer a png for some > reason). > Please do not top-post on this mailing list. > Carl Eugen Sorry, was just so used to put the most recent on top in other emails. Thanx for the help, however -vcodec does not help. I tried both with copy and with png, but it does not work: morkeleb at morkeleb-VirtualBox:~/Music$ ffmpeg -i 150.mp3 -i 150.jpg -c:a libaacplus -b:a 48k -map 0 -map 1 -vcodec png -metadata:s:v title="track title" -metadata:s:v author="artist name" 150_encoded.m4a ffmpeg version N-51694-g4e0130f Copyright (c) 2000-2013 the FFmpeg developers built on Apr 10 2013 16:12:06 with gcc 4.7 (Ubuntu/Linaro 4.7.2-2ubuntu1) configuration: --enable-gpl --enable-libaacplus --enable-libmp3lame --enable-version3 --enable-nonfree --enable-static libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.101 / 3. 49.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mp3 @ 0xa03d280] max_analyze_duration 5000000 reached at 5015510 microseconds Input #0, mp3, from '150.mp3': Metadata: title : Jesus Gave Me Water artist : Sam Cooke & The Soul Stirrers album : Portrait Of A Legend Duration: 00:02:29.66, start: 0.000000, bitrate: 135 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 135 kb/s Input #1, image2, from '150.jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #1:0: Video: mjpeg, yuvj420p, 200x200 [SAR 1:1 DAR 1:1], 25 tbr, 25 tbn, 25 tbc [ipod @ 0xa058b60] track 1: could not find tag, codec not currently supported in container Output #0, ipod, to '150_encoded.m4a': Metadata: title : Jesus Gave Me Water artist : Sam Cooke & The Soul Stirrers album : Portrait Of A Legend encoder : Lavf55.1.100 Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 48 kb/s Stream #0:1: Video: png, rgb24, 200x200 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k tbn, 25 tbc Metadata: title : track title author : artist name Stream mapping: Stream #0:0 -> #0:0 (mp3 -> libaacplus) Stream #1:0 -> #0:1 (mjpeg -> png) Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted I know cover images are generally supported in .mp4/mp4/3gp files - but does ffmpeg support them too? Or am I still doing something wrong? From cehoyos at ag.or.at Thu Apr 11 11:38:43 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 11 Apr 2013 09:38:43 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Audio_transcoding_with_metadata_and_album?= =?utf-8?q?_cover=09image?= References: <405017730.20130410190003@inode.at> <20130410111700.26092eae@lrcd.com> <941864805.20130411094700@inode.at> <1412792286.20130411104526@inode.at> Message-ID: Lukas ?sterreicher inode.at> writes: > I know cover images are generally supported in .mp4/mp4/3gp > files - but does ffmpeg support them too? Sorry for giving bad advice: cover images can be read from m4a files but not written. (mov supports writing mjpeg and png but I am not sure the resulting files are what you want, you have to test) Carl Eugen From classysnet at gmail.com Thu Apr 11 13:57:54 2013 From: classysnet at gmail.com (Rares Pop) Date: Thu, 11 Apr 2013 14:57:54 +0300 Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter In-Reply-To: References: Message-ID: Hi Carl, I have tried with analyzeduration 6M but same error ;dimensions not set.' This is my command: /usr/local/bin/ffmpeg -i hi.ts -loglevel debug -analyzeduration 6M -map 0 -c copy -f segment -segment_time 10 -segment_list hi.m3u8 hi_%05d.ts Perhaps there is a bug in the segmenter plugin.... Thanks again, Rares On Thu, Apr 4, 2013 at 12:39 PM, Carl Eugen Hoyos wrote: > Rares Pop gmail.com> writes: > > > [mpegts 0x20dd1c0] max_analyze_duration 5000000 reached > > at 5013333 microseconds > > [mpegts 0x20dd1c0] Could not find codec parameters for > > stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size > > Consider increasing the value for the 'analyzeduration' and > > 'probesize' options > > Works fine with -analyzeduration 6M > > > Is this a regression in ffmpeg or am I missing something. > > I wasn't able to find a version that sees the video > without -analyzeduration. > Do you know what command line produced the file? > > > Previous ffmpeg (1.1 build in November) is working fine > > with the same data. > > Doesn't work here: > $ ffmpeg -i hi.ts > ffmpeg version 1.1 Copyright (c) 2000-2013 the FFmpeg developers > built on Feb 10 2013 11:46:21 with gcc 4.7 (SUSE Linux) > configuration: --enable-gpl > libavutil 52. 13.100 / 52. 13.100 > libavcodec 54. 86.100 / 54. 86.100 > libavformat 54. 59.106 / 54. 59.106 > libavdevice 54. 3.102 / 54. 3.102 > libavfilter 3. 32.100 / 3. 32.100 > libswscale 2. 1.103 / 2. 1.103 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > [mpegts @ 0x167f540] max_analyze_duration 5000000 reached at 5013333 > [mpegts @ 0x167f540] Could not find codec parameters for stream 0 (Video: > h264 ([27][0][0][0] / 0x001B)): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > [NULL @ 0x1683740] start time is not set in estimate_timings_from_pts > Input #0, mpegts, from 'hi.ts': > Duration: 00:09:59.80, start: 1.400000, bitrate: 782 kb/s > Program 1 > Metadata: > service_name : Service01 > service_provider: FFmpeg > Stream #0:0[0x100]: Video: h264 ([27][0][0][0] / 0x001B), 90k tbr, 90k > tbn, 180k tbc > Stream #0:1[0x101](rus): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, > stereo, fltp, 102 kb/s > At least one output file must be specified > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From andrew at sonar.sg Thu Apr 11 14:11:14 2013 From: andrew at sonar.sg (andrew at sonar.sg) Date: Thu, 11 Apr 2013 05:11:14 -0700 Subject: [FFmpeg-user] Unable to pipe fdk_aac output In-Reply-To: References: <4C7C18CF-7DDE-4B4E-B248-838436D9C3D0@sonar.sg> Message-ID: <6b892f0daddc43f461062b0b78431e6c.squirrel@webmail.sonar.sg> > On Apr 10, 2013 5:07 PM, "Andrew Crothers" wrote: >> >> Hi all, >> >> I having major problems piping aac output. >> >> No problems writing directly to a file, but I just can't pipe! >> >> Any help out there? >> >> >> Input: >> >> ffmpeg -i source.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k >> pipe:1 >> >> Result: >> >> ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers >> built on Apr 11 2013 06:57:44 with llvm-gcc 4.2.1 (LLVM build > 2336.11.00) >> configuration: --enable-libaacplus --enable-libfdk_aac >> --enable-libopus > --enable-libfaac --enable-nonfree >> libavutil 52. 18.100 / 52. 18.100 >> libavcodec 54. 92.100 / 54. 92.100 >> libavformat 54. 63.104 / 54. 63.104 >> libavdevice 54. 3.103 / 54. 3.103 >> libavfilter 3. 42.103 / 3. 42.103 >> libswscale 2. 2.100 / 2. 2.100 >> libswresample 0. 17.102 / 0. 17.102 >> [wav @ 0x7fec2403d800] max_analyze_duration 5000000 reached at 5015510 > microseconds >> Guessed Channel Layout for Input Stream #0.0 : stereo >> Input #0, wav, from 'source.wav': >> Metadata: >> copyright : ? ueberschall 2003 >> Duration: 00:00:30.00, bitrate: 1411 kb/s >> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, > stereo, s16, 1411 kb/s >> [NULL @ 0x7fec2405e200] Unable to find a suitable output format for > 'pipe:1' >> pipe:1: Invalid argument > > You did not specify the output format. Try this: > > ffmpeg -i source.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k -f adts > pipe:1 > > Timothy Gu > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Perfect!! Many thanks Timothy From cehoyos at ag.or.at Thu Apr 11 14:17:16 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 11 Apr 2013 12:17:16 +0000 (UTC) Subject: [FFmpeg-user] error 'dimensions not set' and HLS segmenter References: Message-ID: Rares Pop gmail.com> writes: > ffmpeg -i hi.ts -loglevel debug -analyzeduration 6M -analyzeduration is an input option, do not specify it as an option for an output file. (Put it in front of the name of the input file that needs it.) Please do not top-post here, Carl Eugen From ticks.sunny84 at gmail.com Thu Apr 11 16:03:28 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Thu, 11 Apr 2013 19:33:28 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: Hi, I have been trying to create a baseline video using ffmpeg and h264 as below: But it seems that i can not map ffmpeg settigns properly to H264 settings. I am using ffmpeg ver 1.2 If anyone has done any work on this please share your valued openion as i am really stuck into this. On Thu, Apr 11, 2013 at 10:30 AM, Susheel Tickoo wrote: > Hi Carl, > > Did you got any time to look into this? > > On Wed, Apr 10, 2013 at 8:16 PM, Susheel Tickoo wrote: > >> Hi Carl, >> now i have removed the duplicate fields and here are my new settings: >> >> c->width = width; >> c->height = height; >> >> c->time_base= (AVRational){1,0}; >> c->gop_size = 12; // emit one intra frame every ten frames >> c->pix_fmt = PIX_FMT_YUV420P; >> c->codec_type = AVMEDIA_TYPE_VIDEO; >> c->codec_id = codec->id; >> c->max_b_frames = 0; >> c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; >> c->me_range = 16; >> c->max_qdiff = 4; >> c->qmin = 10; >> c->qcompress = 0.6; >> c->trellis=0; >> c->level = 13; >> c->refs = 5; >> c->coder_type = 0; >> c->flags|=CODEC_FLAG_LOOP_FILTER;//new >> >> c->scenechange_threshold = 40; //new >> c-> rc_buffer_size = 0; >> c->me_method=7; >> c->me_cmp|= 1; >> c->me_subpel_quality = 6; >> c->qmax=51; >> c->keyint_min=25; >> av_opt_set(c->priv_data,"subq","6",0); >> av_opt_set(c->priv_data,"crf","20.0",0); >> av_opt_set(c->priv_data,"weighted_p_pred","0",0); >> av_opt_set(c->priv_data,"vprofile","baseline",0); >> av_opt_set(c->priv_data,"preset","medium",0); >> av_opt_set(c->priv_data,"tune","zerolatency",0); >> >> I am setting max_b_frames=0 and i am also setting "* >> av_opt_set(c->priv_data,"vprofile","baseline",0);*" >> >> Do i need to set the *profile* or *vprofile* tag ? >> I am confused by the ticket http://ffmpeg.org/trac/ffmpeg/ticket/387 >> I have created the video using max_b_frames=0 and not setting it at all >> but in both cases the video is not baseline. >> >> Should i use only teh following line? >> c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; >> >> Please help me out. >> thanks & regards, >> Susheel Tickoo >> >> >> >> On Wed, Apr 10, 2013 at 7:53 PM, Susheel Tickoo wrote: >> >>> Thanks Carl, >>> >>> I will suerly update my code to be more redable and presentable :). So >>> do we need to remove the setting of max_b_frames at all or do we have to >>> set it to c->max_b_frames = 0; ? >>> >>> On Wed, Apr 10, 2013 at 7:34 PM, Carl Lindqvist wrote: >>> >>>> > c->max_b_frames=1; >>>> Then down a bit comes >>>> > c->max_b_frames = 0; >>>> Then comes >>>> > c->max_b_frames=1;//new >>>> >>>> This is just one example of things that look weird in those settings. >>>> Looks like you have copy/pasted a lot of different option around. >>>> >>>> I haven't gone through all the options you set, but for example, >>>> baseline can't have b-frames. >>>> >>>> /Carl >>>> _______________________________________________ >>>> ffmpeg-user mailing list >>>> ffmpeg-user at ffmpeg.org >>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>>> >>> >>> >> > From timbot at oddlystudios.com Thu Apr 11 17:36:38 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Thu, 11 Apr 2013 11:36:38 -0400 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: <20130410155114.GP7301@arborea> References: <20130410155114.GP7301@arborea> Message-ID: Hi Stefano! Thanks very much. Your comment was actually very helpful. I took a second look at how I was viewing the output of my alphamerge and sure enough - the alpha information was correct. I am now tackling my last problem. My video output from the alphamerge is excellent, but consistently shows a value of 16 in the alpha channel in areas that should be completely transparent. It has the effect of putting a very slight gray layer on everything. I am betting that this is some compression on my output (which is rgba, png codec into a mov). You mentioned using rawvideo. I haven't tried that before, but it sounds like what I might need. Thanks again for your help. -tim. On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini wrote: > On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: > > Hi, this is my first question ever to a mailing list (or anywhere!). > > I have been using ffmpeg for about a year and a half and have figured out > > many things on my own but I'm stumped regarding the alphaextract and > > alphamerge filters. > > I am able to get the proper result every time with images, but have yet > to > > succeed with videos. I have tried many formats and searched extensively > but > > have not found a single documented case of someone successfully using > this > > > filter combination and sharing an example. The documentation on the > > ffmpeg-filters page is very sparse, and does not give any details of what > > formats/codecs are supported. > > Yes that's because ffmpeg-filters is supposed to only document filters. > > > > > I think this filter would be very useful if I can figure it out. > > > > Step 1 - alphaextract > > This seems to always work, and produces a grayscale video as expected. In > > my tests I have used rgba video, as well as quicktime with animation > codec > > (bgra). > > > > Step 2 - applying the extracted alpha > > This works every time for images. However while the encode completes with > > video it never provides the masking expected by mapping the extracted > alpha > > information to the alpha channel of the target video. In my tests with > > still images, I found that the output format must be a format that > supports > > an alpha channel, and I have tried several with video (which I determined > > by looking at the source code of the filter). > > The question is, are you sure your output codec supports alpha (most > don't, indeed I couldn't even tell which ones support alpha). You can > try with rawvideo and make sure it is not a codec bug. > > > My command (taken from the docs) looks like this: > > > > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov > [alpha]; > > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png > composite.mov > > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg > > developers > > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > > 4.6.1-9ubuntu3) > > configuration: --enable-gpl --enable-libass --enable-libfaac > > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 > > --enable-nonfree --enable-version3 --enable-libfreetype > > --enable-filter=drawtext > > libavutil 52. 25.100 / 52. 25.100 > > libavcodec 55. 2.100 / 55. 2.100 > > libavformat 55. 1.100 / 55. 1.100 > > libavdevice 55. 0.100 / 55. 0.100 > > libavfilter 3. 49.100 / 3. 49.100 > > libswscale 2. 2.100 / 2. 2.100 > > libswresample 0. 17.102 / 0. 17.102 > > libpostproc 52. 2.100 / 52. 2.100 > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42mp41 > > creation_time : 2013-04-07 18:35:55 > > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, > > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : ?Mainconcept Video Media Handler > > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, > > fltp, 189 kb/s > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : #Mainconcept MP4 Sound Media Handler > > Output #0, mov, to 'composite.mov': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42mp41 > > encoder : Lavf55.1.100 > > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 [SAR > > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > > Metadata: > > creation_time : 2013-04-07 18:35:55 > > handler_name : ?Mainconcept Video Media Handler > > Stream mapping: > > Stream #0:0 -> #0:0 (h264 -> png) > > Press [q] to stop, [?] for help > > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > > bitrate=3177.5kbits/s dup=2 drop=0 > > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead > > 0.067323% > > > > After this, I see only my red background (timbotundelay.mp4) and no alpha > > applied. > > I tried a variant of your command and seems to work fine here. Also note > that you need a background to show an image with overlay. Something > like: > ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" > > should do. > > > Thanks so much to anyone who can help me out here. I'm running out of > > things to try. I'd particularly like to see how people are using this > > filter successfully. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From hariseldom at gmail.com Thu Apr 11 19:54:10 2013 From: hariseldom at gmail.com (Ricardo Martinez) Date: Thu, 11 Apr 2013 19:54:10 +0200 Subject: [FFmpeg-user] iPad live stream Message-ID: Hi List! i have read a lot of documentation, but i cannt see cleary the way. I don't know if this is possible: i want live stream to iPad using, ffmpeg and ffserver. By example ffmpeg -re -i file.flv http://localhost:8090/feed1.ffm and using ffserver make live stream for iPad devices. I did it ok for .flv files but don't know how can to do it for iPad devices. if possible, any website or ffserver.conf and ffmpeg example would be very helpful. thank you very much for reading -- Ricardo _______________________________________________ From seandarcy2 at gmail.com Thu Apr 11 23:46:38 2013 From: seandarcy2 at gmail.com (sean darcy) Date: Thu, 11 Apr 2013 17:46:38 -0400 Subject: [FFmpeg-user] -ab : syntax deprecated, use '|' to separate sample rate. ???? Message-ID: I'm transcoding from wav to m4a: ffmpeg -i $f -vn -acodec aac -ab 128000 -strict experimental $f1.m4a I get a deprecation warning on the sample rate that I can't understand: ffmpeg version N-51822-gf17651b Copyright (c) 2000-2013 the FFmpeg developers built on Apr 10 2013 20:37:57 with gcc 4.7.2 (GCC) 20120921 (Red Hat 4.7.2-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --shlibdir=/usr/lib64 --extra-cflags='-Ofast -march=native -mtune=native -fopenmp -fomit-frame-pointer -pipe' --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-gray --enable-vaapi --enable-hardcoded-tables --enable-libaacplus --enable-frei0r --enable-libfdk-aac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-pic --enable-libopencv --enable-openssl libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 50.103 / 3. 50.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [wav @ 0x1e899a0] max_analyze_duration 5000000 reached at 5015510 microseconds Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from '9.1_Politics.wav': Duration: 00:44:52.64, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s [audio format for output stream 0:0 @ 0x1e9d440] This syntax is deprecated, use '|' to separate sample rate. Output #0, ipod, to '09.1_Politics.m4a': Metadata: encoder : Lavf55.2.100 Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s From ffmpeg help: -ab E...A. set bitrate (in bits/s) (from 0 to INT_MAX) Thanks, sean From ubitux at gmail.com Thu Apr 11 23:49:38 2013 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 11 Apr 2013 23:49:38 +0200 Subject: [FFmpeg-user] -ab : syntax deprecated, use '|' to separate sample rate. ???? In-Reply-To: References: Message-ID: <20130411214938.GF25658@leki> On Thu, Apr 11, 2013 at 05:46:38PM -0400, sean darcy wrote: > I'm transcoding from wav to m4a: > > ffmpeg -i $f -vn -acodec aac -ab 128000 -strict experimental $f1.m4a > > I get a deprecation warning on the sample rate that I can't understand: > [...] > [audio format for output stream 0:0 @ 0x1e9d440] This syntax is > deprecated, use '|' to separate sample rate. This should not happen anymore, try to upgrade (it was introduced and fixed today). -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From seandarcy2 at gmail.com Fri Apr 12 00:46:50 2013 From: seandarcy2 at gmail.com (sean darcy) Date: Thu, 11 Apr 2013 18:46:50 -0400 Subject: [FFmpeg-user] -ab : syntax deprecated, use '|' to separate sample rate. ???? In-Reply-To: <20130411214938.GF25658@leki> References: <20130411214938.GF25658@leki> Message-ID: On 04/11/2013 05:49 PM, Cl?ment B?sch wrote: > On Thu, Apr 11, 2013 at 05:46:38PM -0400, sean darcy wrote: >> I'm transcoding from wav to m4a: >> >> ffmpeg -i $f -vn -acodec aac -ab 128000 -strict experimental $f1.m4a >> >> I get a deprecation warning on the sample rate that I can't understand: >> > [...] >> [audio format for output stream 0:0 @ 0x1e9d440] This syntax is >> deprecated, use '|' to separate sample rate. > > This should not happen anymore, try to upgrade (it was introduced and > fixed today). > Yup. Fixed. Thanks for the quick response. sean From jpolblanc at free.fr Thu Apr 11 23:24:28 2013 From: jpolblanc at free.fr (Jean-Paul BLANC) Date: Thu, 11 Apr 2013 23:24:28 +0200 Subject: [FFmpeg-user] ffmpeg 1.2 for Mac OS X 10.5.8 (Leopard) Message-ID: Hi, Where could I find binary for ffmpeg 1.2 for Mac OS X 10.5.8 (Leopard) ? Could I trust the one that I found on http://ffmpegmac.net/ ? JiPaul. From hancock at playerlync.com Thu Apr 11 18:33:19 2013 From: hancock at playerlync.com (scott80109) Date: Thu, 11 Apr 2013 09:33:19 -0700 (PDT) Subject: [FFmpeg-user] Help me speed up my transcoding command Message-ID: <1365697998860-4658412.post@n4.nabble.com> I'm new to ffmpeg and video transcoding. This command seems pretty slow. I'm using it to get rid of b-frames. If anyone has any tips on how to accomplish this faster, it would be much appreciated. ffmpeg videoToEncod.mp4 -y -preset faster -acodec libvo_aacenc -ac 2 -ab 128k -vcodec libx264 -b:v 2400k -r 29.97 -s 960x540 -g 15 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -me_method umh -subq 6 -trellis 1 -refs 1 -coder 0 -me_range 16 -sc_threshold 40 -i_qfactor 0.71 -bt 2400k -maxrate 10M -bufsize 10M -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -threads 0 theNewFile.mp4 2>&1'; -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Help-me-speed-up-my-transcoding-command-tp4658412.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From simondaniels23 at gmail.com Fri Apr 12 07:16:37 2013 From: simondaniels23 at gmail.com (Simon Daniels) Date: Thu, 11 Apr 2013 22:16:37 -0700 Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why In-Reply-To: References: Message-ID: That worked Carl! I also figured out that simply "59.94" works. Can I get away with that since the rest of my code already converts to a decimal? Or should I change over to fractional values? On Wed, Apr 10, 2013 at 1:37 AM, Carl Eugen Hoyos wrote: > Simon Daniels gmail.com> writes: > > > "-r", > > "59.94006065616729", > > Please use -r 60000/1001 to specify 59.9 fps > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From francois.visagie at gmail.com Fri Apr 12 08:00:56 2013 From: francois.visagie at gmail.com (Francois Visagie) Date: Fri, 12 Apr 2013 08:00:56 +0200 Subject: [FFmpeg-user] Help me speed up my transcoding command In-Reply-To: <1365697998860-4658412.post@n4.nabble.com> References: <1365697998860-4658412.post@n4.nabble.com> Message-ID: <002e01ce3743$1bdc1df0$539459d0$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of scott80109 > Sent: 11 April 2013 18:33 > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] Help me speed up my transcoding command > > I'm new to ffmpeg and video transcoding. This command seems pretty slow. > I'm using it to get rid of b-frames. If anyone has any tips on how to > accomplish this faster, it would be much appreciated. > > ffmpeg videoToEncod.mp4 -y -preset faster -acodec libvo_aacenc -ac 2 -ab > 128k -vcodec libx264 -b:v 2400k -r 29.97 -s 960x540 -g 15 -flags +loop -cmp > +chroma -partitions +parti4x4+partp8x8+partb8x8 -me_method umh -subq 6 > -trellis 1 -refs 1 -coder 0 -me_range 16 -sc_threshold 40 -i_qfactor 0.71 -bt > 2400k -maxrate 10M -bufsize 10M -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 - > level 30 -threads 0 theNewFile.mp4 2>&1'; I believe '-bf' specifies the number of B-frames to use, '0' in your case. Other than that and bitrate specifiers, I'd get rid of all command-line options and evaluate the various x264 presets for suitable speed vs. quality instead: http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide. Cheers, Francois > > > > -- > View this message in context: http://ffmpeg- > users.933282.n4.nabble.com/Help-me-speed-up-my-transcoding-command- > tp4658412.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bostjan.strojan at gmail.com Fri Apr 12 10:02:15 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Fri, 12 Apr 2013 10:02:15 +0200 Subject: [FFmpeg-user] Matrox MQSink Filter Message-ID: Hi, Having some trouble with converting certain avi format to anything else; ffprobe C0005.avi ffprobe version git-2013-04-12-9da3696 Copyright (c) 2007-2013 the FFmpeg developers built on Apr 12 2013 00:46:27 with gcc 4.7 (Debian 4.7.2-5) configuration: --enable-gpl --enable-libx264 --enable-libfdk-aac --enable-nonfree --enable-version3 --enable-x11grab libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 53.101 / 3. 53.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0x34e7ea0] non-interleaved AVI [mpeg2video @ 0x34e8920] Invalid frame dimensions 0x0. Input #0, avi, from 'C0005.avi': Metadata: encoder : Matrox MQSink Filter Duration: 00:00:00.40, start: 0.000000, bitrate: N/A Stream #0:0: Video: mpeg2video (4:2:2) (M701 / 0x3130374D), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc ------------- sample (md5 a845ed3747157797312ee236f4ebf1f2) will be here soon; https://dl.dropboxusercontent.com/u/79532365/matroxQsink/C0005.avi Thanks in advance, Bo?tjan From ahmedtariq.nu at gmail.com Fri Apr 12 13:15:52 2013 From: ahmedtariq.nu at gmail.com (Ahmad-D-Tariq) Date: Fri, 12 Apr 2013 16:15:52 +0500 Subject: [FFmpeg-user] need help with ffmpeg. Message-ID: Hi, i am working on an android app which modifies the audio of a video. i'm developing this app using Eclipse IDE and android SDK on ubuntu 10.04. i am stuck for one month and cannot figure out that how to do it. the documentation and tutorials on internet is not helping much.. please anybody help me that how can i perform this functionality using ffmpeg. a sample code would be much appreciated.. Best Regards, Ahmed From 23liam at googlemail.com Fri Apr 12 13:59:12 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Fri, 12 Apr 2013 12:59:12 +0100 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: >i am working on an android app which modifies the audio of a video. Modifies it how? What exactly do you want to do? Also, are you planning on bundling an ffmpeg executable with your app, oruse some of the libraries? From 23liam at googlemail.com Fri Apr 12 14:04:46 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Fri, 12 Apr 2013 13:04:46 +0100 Subject: [FFmpeg-user] Matrox MQSink Filter In-Reply-To: References: Message-ID: >Having some trouble with converting certain avi format to anything else; What have you tried? > Metadata: > encoder : Matrox MQSink Filter > Duration: 00:00:00.40, start: 0.000000, bitrate: N/A > Stream #0:0: Video: mpeg2video (4:2:2) (M701 / 0x3130374D), This is a matrox I-frame-only variation of MPEG2 (I know, because I've had to encode video to that format for use with Lightworks). It may not be possible to directly copy the bit stream (i.e. `-c:v copy` might not work). However, transcoding to another codec certainly should be possible. From bostjan.strojan at gmail.com Fri Apr 12 14:10:55 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Fri, 12 Apr 2013 14:10:55 +0200 Subject: [FFmpeg-user] Matrox MQSink Filter In-Reply-To: References: Message-ID: On Fri, Apr 12, 2013 at 2:04 PM, Liam Condron-Farnos <23liam at googlemail.com> wrote: >>Having some trouble with converting certain avi format to anything else; > > What have you tried? Nothing goes, example: ffmpeg -i C0005.avi test.mp4 ffmpeg version git-2013-04-12-9da3696 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 12 2013 00:46:27 with gcc 4.7 (Debian 4.7.2-5) configuration: --enable-gpl --enable-libx264 --enable-libfdk-aac --enable-nonfree --enable-version3 --enable-x11grab libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 53.101 / 3. 53.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0x236ef40] non-interleaved AVI [mpeg2video @ 0x236f940] Invalid frame dimensions 0x0. Input #0, avi, from 'C0005.avi': Metadata: encoder : Matrox MQSink Filter Duration: 00:00:00.40, start: 0.000000, bitrate: N/A Stream #0:0: Video: mpeg2video (4:2:2) (M701 / 0x3130374D), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc No pixel format specified, yuv422p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players. [libx264 @ 0x2371320] using SAR=1/1 [libx264 @ 0x2371320] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x2371320] profile High 4:2:2, level 4.0, 4:2:2 8-bit [libx264 @ 0x2371320] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf55.2.100 Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 12800 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> libx264) Press [q] to stop, [?] for help [mpeg2video @ 0x236f940] Invalid frame dimensions 0x0. [mpeg2video @ 0x236f940] ac-tex damaged at 119 67 [mpeg2video @ 0x236f940] Warning MVs not available [mpeg2video @ 0x236f940] concealing 120 DC, 120 AC, 120 MV errors in B frame [mpeg2video @ 0x236f940] ac-tex damaged at 119 67 [mpeg2video @ 0x236f940] concealing 50 DC, 50 AC, 50 MV errors in P frame [mpeg2video @ 0x236f940] ac-tex damaged at 119 67 [mpeg2video @ 0x236f940] Warning MVs not available [mpeg2video @ 0x236f940] concealing 120 DC, 120 AC, 120 MV errors in B frame frame= 11 fps=6.3 q=-1.0 Lsize= 775kB time=00:00:00.36 bitrate=17644.9kbits/s dup=1 drop=0 video:775kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.106912% [libx264 @ 0x2371320] frame I:2 Avg QP:23.02 size:102044 [libx264 @ 0x2371320] frame P:9 Avg QP:25.29 size: 65373 [libx264 @ 0x2371320] mb I I16..4: 7.9% 91.4% 0.6% [libx264 @ 0x2371320] mb P I16..4: 4.7% 44.5% 0.4% P16..4: 30.1% 4.8% 3.8% 0.0% 0.0% skip:11.7% [libx264 @ 0x2371320] 8x8 transform intra:90.2% inter:94.2% [libx264 @ 0x2371320] coded y,uvDC,uvAC intra: 71.3% 98.2% 50.8% inter: 24.5% 71.7% 3.6% [libx264 @ 0x2371320] i16 v,h,dc,p: 2% 3% 3% 92% [libx264 @ 0x2371320] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 9% 9% 55% 5% 6% 4% 4% 3% 4% [libx264 @ 0x2371320] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 12% 19% 8% 19% 12% 7% 4% 3% [libx264 @ 0x2371320] i8c dc,h,v,p: 49% 15% 26% 10% [libx264 @ 0x2371320] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x2371320] ref P L0: 50.2% 10.1% 24.7% 15.1% [libx264 @ 0x2371320] kb/s:14408.07 produces a 440 ms long avc/mp4 file. Bo?tjan From ahmedtariq.nu at gmail.com Fri Apr 12 14:11:46 2013 From: ahmedtariq.nu at gmail.com (Ahmad-D-Tariq) Date: Fri, 12 Apr 2013 17:11:46 +0500 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: i want to modify sound like in TomCat app. or like Helium Booth app(iphone). i want the functionality that when a person speaks then myapp records make video/audio and plays back it by changing its audio pitch, like tomcat app. and i builded the ffmpeg in ubuntu and included its libraries, libavformat and libavcodec. but i don't know now that what i am doing is right or wrong. On Fri, Apr 12, 2013 at 4:59 PM, Liam Condron-Farnos <23liam at googlemail.com>wrote: > >i am working on an android app which modifies the audio of a video. > > Modifies it how? What exactly do you want to do? Also, are you planning on > bundling an ffmpeg executable with your app, oruse some of the libraries? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ahmedtariq.nu at gmail.com Fri Apr 12 14:16:14 2013 From: ahmedtariq.nu at gmail.com (Ahmad-D-Tariq) Date: Fri, 12 Apr 2013 17:16:14 +0500 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: actually i followed this tutorial http://dmitrydzz-hobby.blogspot.com/2012/04/how-to-build-ffmpeg-and-use-it-in.html On Fri, Apr 12, 2013 at 5:11 PM, Ahmad-D-Tariq wrote: > i want to modify sound like in TomCat app. or like Helium Booth > app(iphone). i want the functionality that when a person speaks then myapp > records make video/audio and plays back it by changing its audio pitch, > like tomcat app. > and > i builded the ffmpeg in ubuntu and included its libraries, libavformat and > libavcodec. > > but i don't know now that what i am doing is right or wrong. > > > On Fri, Apr 12, 2013 at 4:59 PM, Liam Condron-Farnos < > 23liam at googlemail.com> wrote: > >> >i am working on an android app which modifies the audio of a video. >> >> Modifies it how? What exactly do you want to do? Also, are you planning on >> bundling an ffmpeg executable with your app, oruse some of the libraries? >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From 23liam at googlemail.com Fri Apr 12 14:38:59 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Fri, 12 Apr 2013 13:38:59 +0100 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: >i want the functionality that when a person speaks then myapp >records make video/audio and plays back it by changing its audio pitch, >like tomcat app. I'm not 100% certain, but I don't think ffmpeg can do this (that's only from a quick look over the documentation, so I may be mistaken). Maybe try another tool, like sox. From timbot at oddlystudios.com Fri Apr 12 14:43:25 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Fri, 12 Apr 2013 08:43:25 -0400 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: Hello folks, I thought I could share a tip I've used in my experiments with the alphaextract and alphamerge filters. I have found that the result of the alphamerge filter often leaves areas with some non-zero alpha. Comparing the mask file, areas that are pure black are not always reproduced as pure alpha after the merge. However, I found a way to deal with this by using the lut filter. It was as simple as adding this to my alphamerge filter chain: [text2][out]alphamerge,format=rgba,lutrgb=a=if(gte(val\,18)\,val) The lutrgb is using an expression that lets me add a threshold for the alpha channel. In this case, I'm saying that if the alpha drops below 18, it should likely be zero, otherwise keep it the same. This cleaned up any alpha artefacts very nicely. Hope that is helpful for somebody. -t. On Thu, Apr 11, 2013 at 11:36 AM, Tim Willison wrote: > Hi Stefano! > > Thanks very much. Your comment was actually very helpful. > I took a second look at how I was viewing the output of my alphamerge and > sure enough - the alpha information was correct. I am now tackling my last > problem. > > My video output from the alphamerge is excellent, but consistently shows a > value of 16 in the alpha channel in areas that should be completely > transparent. It has the effect of putting a very slight gray layer on > everything. > I am betting that this is some compression on my output (which is rgba, > png codec into a mov). > > You mentioned using rawvideo. I haven't tried that before, but it sounds > like what I might need. > > Thanks again for your help. > > -tim. > > > On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini wrote: > >> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: >> > Hi, this is my first question ever to a mailing list (or anywhere!). >> > I have been using ffmpeg for about a year and a half and have figured >> out >> > many things on my own but I'm stumped regarding the alphaextract and >> > alphamerge filters. >> > I am able to get the proper result every time with images, but have yet >> to >> > succeed with videos. I have tried many formats and searched extensively >> but >> > have not found a single documented case of someone successfully using >> this >> >> > filter combination and sharing an example. The documentation on the >> > ffmpeg-filters page is very sparse, and does not give any details of >> what >> > formats/codecs are supported. >> >> Yes that's because ffmpeg-filters is supposed to only document filters. >> >> > >> > I think this filter would be very useful if I can figure it out. >> > >> > Step 1 - alphaextract >> > This seems to always work, and produces a grayscale video as expected. >> In >> > my tests I have used rgba video, as well as quicktime with animation >> codec >> > (bgra). >> > >> > Step 2 - applying the extracted alpha >> > This works every time for images. However while the encode completes >> with >> > video it never provides the masking expected by mapping the extracted >> alpha >> > information to the alpha channel of the target video. In my tests with >> > still images, I found that the output format must be a format that >> supports >> > an alpha channel, and I have tried several with video (which I >> determined >> > by looking at the source code of the filter). >> >> The question is, are you sure your output codec supports alpha (most >> don't, indeed I couldn't even tell which ones support alpha). You can >> try with rawvideo and make sure it is not a codec bug. >> >> > My command (taken from the docs) looks like this: >> > >> > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov >> [alpha]; >> > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png >> composite.mov >> > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the FFmpeg >> > developers >> > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro >> > 4.6.1-9ubuntu3) >> > configuration: --enable-gpl --enable-libass --enable-libfaac >> > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb >> > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp >> > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 >> > --enable-nonfree --enable-version3 --enable-libfreetype >> > --enable-filter=drawtext >> > libavutil 52. 25.100 / 52. 25.100 >> > libavcodec 55. 2.100 / 55. 2.100 >> > libavformat 55. 1.100 / 55. 1.100 >> > libavdevice 55. 0.100 / 55. 0.100 >> > libavfilter 3. 49.100 / 3. 49.100 >> > libswscale 2. 2.100 / 2. 2.100 >> > libswresample 0. 17.102 / 0. 17.102 >> > libpostproc 52. 2.100 / 52. 2.100 >> > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': >> > Metadata: >> > major_brand : mp42 >> > minor_version : 0 >> > compatible_brands: mp42mp41 >> > creation_time : 2013-04-07 18:35:55 >> > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s >> > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, >> > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc >> > Metadata: >> > creation_time : 2013-04-07 18:35:55 >> > handler_name : ?Mainconcept Video Media Handler >> > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, >> > fltp, 189 kb/s >> > Metadata: >> > creation_time : 2013-04-07 18:35:55 >> > handler_name : #Mainconcept MP4 Sound Media Handler >> > Output #0, mov, to 'composite.mov': >> > Metadata: >> > major_brand : mp42 >> > minor_version : 0 >> > compatible_brands: mp42mp41 >> > encoder : Lavf55.1.100 >> > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 >> [SAR >> > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc >> > Metadata: >> > creation_time : 2013-04-07 18:35:55 >> > handler_name : ?Mainconcept Video Media Handler >> > Stream mapping: >> > Stream #0:0 -> #0:0 (h264 -> png) >> > Press [q] to stop, [?] for help >> > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 >> > bitrate=3177.5kbits/s dup=2 drop=0 >> > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead >> > 0.067323% >> > >> > After this, I see only my red background (timbotundelay.mp4) and no >> alpha >> > applied. >> >> I tried a variant of your command and seems to work fine here. Also note >> that you need a background to show an image with overlay. Something >> like: >> ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" >> >> should do. >> >> > Thanks so much to anyone who can help me out here. I'm running out of >> > things to try. I'd particularly like to see how people are using this >> > filter successfully. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > > > -- > Tim Willison > Lead Product Designer > www.oddlystudios.com > 495B Queen St. West > Toronto, ON. > 647 345 6275 > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From ahmedtariq.nu at gmail.com Fri Apr 12 14:49:27 2013 From: ahmedtariq.nu at gmail.com (Ahmad-D-Tariq) Date: Fri, 12 Apr 2013 17:49:27 +0500 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: Thanks very much. but i never used sox. actually never heard of it before. so by considering the functionality i want, could you plz give me some links for sox or like guidelines so that i may be able to achieve my desired functionality using sox. On Fri, Apr 12, 2013 at 5:38 PM, Liam Condron-Farnos <23liam at googlemail.com>wrote: > >i want the functionality that when a person speaks then myapp > >records make video/audio and plays back it by changing its audio pitch, > >like tomcat app. > > I'm not 100% certain, but I don't think ffmpeg can do this (that's only > from a quick look over the documentation, so I may be mistaken). Maybe try > another tool, like sox. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Fri Apr 12 15:12:21 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 12 Apr 2013 13:12:21 +0000 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: On 4/12/13, Tim Willison wrote: > Hello folks, > I thought I could share a tip I've used in my experiments with the > alphaextract and alphamerge filters. > > I have found that the result of the alphamerge filter often leaves areas > with some non-zero alpha. Comparing the mask file, areas that are pure > black are not always reproduced as pure alpha after the merge. Looks like you used YUV source for mask. And result is expected because YUV (limited range, not YUVJ) values can not be 0-16 or 235-255. > > However, I found a way to deal with this by using the lut filter. It was as > simple as adding this to my alphamerge filter chain: > > [text2][out]alphamerge,format=rgba,lutrgb=a=if(gte(val\,18)\,val) > > The lutrgb is using an expression that lets me add a threshold for the > alpha channel. In this case, I'm saying that if the alpha drops below 18, > it should likely be zero, otherwise keep it the same. > This cleaned up any alpha artefacts very nicely. > > Hope that is helpful for somebody. > > -t. > > > On Thu, Apr 11, 2013 at 11:36 AM, Tim Willison > wrote: > >> Hi Stefano! >> >> Thanks very much. Your comment was actually very helpful. >> I took a second look at how I was viewing the output of my alphamerge and >> sure enough - the alpha information was correct. I am now tackling my >> last >> problem. >> >> My video output from the alphamerge is excellent, but consistently shows >> a >> value of 16 in the alpha channel in areas that should be completely >> transparent. It has the effect of putting a very slight gray layer on >> everything. >> I am betting that this is some compression on my output (which is rgba, >> png codec into a mov). >> >> You mentioned using rawvideo. I haven't tried that before, but it sounds >> like what I might need. >> >> Thanks again for your help. >> >> -tim. >> >> >> On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini >> wrote: >> >>> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: >>> > Hi, this is my first question ever to a mailing list (or anywhere!). >>> > I have been using ffmpeg for about a year and a half and have figured >>> out >>> > many things on my own but I'm stumped regarding the alphaextract and >>> > alphamerge filters. >>> > I am able to get the proper result every time with images, but have >>> > yet >>> to >>> > succeed with videos. I have tried many formats and searched >>> > extensively >>> but >>> > have not found a single documented case of someone successfully using >>> this >>> >>> > filter combination and sharing an example. The documentation on the >>> > ffmpeg-filters page is very sparse, and does not give any details of >>> what >>> > formats/codecs are supported. >>> >>> Yes that's because ffmpeg-filters is supposed to only document filters. >>> >>> > >>> > I think this filter would be very useful if I can figure it out. >>> > >>> > Step 1 - alphaextract >>> > This seems to always work, and produces a grayscale video as expected. >>> In >>> > my tests I have used rgba video, as well as quicktime with animation >>> codec >>> > (bgra). >>> > >>> > Step 2 - applying the extracted alpha >>> > This works every time for images. However while the encode completes >>> with >>> > video it never provides the masking expected by mapping the extracted >>> alpha >>> > information to the alpha channel of the target video. In my tests with >>> > still images, I found that the output format must be a format that >>> supports >>> > an alpha channel, and I have tried several with video (which I >>> determined >>> > by looking at the source code of the filter). >>> >>> The question is, are you sure your output codec supports alpha (most >>> don't, indeed I couldn't even tell which ones support alpha). You can >>> try with rawvideo and make sure it is not a codec bug. >>> >>> > My command (taken from the docs) looks like this: >>> > >>> > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov >>> [alpha]; >>> > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png >>> composite.mov >>> > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the >>> > FFmpeg >>> > developers >>> > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro >>> > 4.6.1-9ubuntu3) >>> > configuration: --enable-gpl --enable-libass --enable-libfaac >>> > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb >>> > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp >>> > --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 >>> > --enable-nonfree --enable-version3 --enable-libfreetype >>> > --enable-filter=drawtext >>> > libavutil 52. 25.100 / 52. 25.100 >>> > libavcodec 55. 2.100 / 55. 2.100 >>> > libavformat 55. 1.100 / 55. 1.100 >>> > libavdevice 55. 0.100 / 55. 0.100 >>> > libavfilter 3. 49.100 / 3. 49.100 >>> > libswscale 2. 2.100 / 2. 2.100 >>> > libswresample 0. 17.102 / 0. 17.102 >>> > libpostproc 52. 2.100 / 52. 2.100 >>> > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': >>> > Metadata: >>> > major_brand : mp42 >>> > minor_version : 0 >>> > compatible_brands: mp42mp41 >>> > creation_time : 2013-04-07 18:35:55 >>> > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s >>> > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, >>> > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc >>> > Metadata: >>> > creation_time : 2013-04-07 18:35:55 >>> > handler_name : ?Mainconcept Video Media Handler >>> > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, >>> > stereo, >>> > fltp, 189 kb/s >>> > Metadata: >>> > creation_time : 2013-04-07 18:35:55 >>> > handler_name : #Mainconcept MP4 Sound Media Handler >>> > Output #0, mov, to 'composite.mov': >>> > Metadata: >>> > major_brand : mp42 >>> > minor_version : 0 >>> > compatible_brands: mp42mp41 >>> > encoder : Lavf55.1.100 >>> > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 >>> [SAR >>> > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc >>> > Metadata: >>> > creation_time : 2013-04-07 18:35:55 >>> > handler_name : ?Mainconcept Video Media Handler >>> > Stream mapping: >>> > Stream #0:0 -> #0:0 (h264 -> png) >>> > Press [q] to stop, [?] for help >>> > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 >>> > bitrate=3177.5kbits/s dup=2 drop=0 >>> > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead >>> > 0.067323% >>> > >>> > After this, I see only my red background (timbotundelay.mp4) and no >>> alpha >>> > applied. >>> >>> I tried a variant of your command and seems to work fine here. Also note >>> that you need a background to show an image with overlay. Something >>> like: >>> ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" >>> >>> should do. >>> >>> > Thanks so much to anyone who can help me out here. I'm running out of >>> > things to try. I'd particularly like to see how people are using this >>> > filter successfully. >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>> >> >> >> >> -- >> Tim Willison >> Lead Product Designer >> www.oddlystudios.com >> 495B Queen St. West >> Toronto, ON. >> 647 345 6275 >> > > > > -- > Tim Willison > Lead Product Designer > www.oddlystudios.com > 495B Queen St. West > Toronto, ON. > 647 345 6275 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Fri Apr 12 15:13:58 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 12 Apr 2013 13:13:58 +0000 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: On 4/12/13, Paul B Mahol wrote: > On 4/12/13, Tim Willison wrote: >> Hello folks, >> I thought I could share a tip I've used in my experiments with the >> alphaextract and alphamerge filters. >> >> I have found that the result of the alphamerge filter often leaves areas >> with some non-zero alpha. Comparing the mask file, areas that are pure >> black are not always reproduced as pure alpha after the merge. > > Looks like you used YUV source for mask. And result is expected because > YUV (limited range, not YUVJ) values can not be 0-16 or 235-255. Not including 16 and 235, thus allowed values are from 16 to 235. From cehoyos at ag.or.at Fri Apr 12 15:25:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 13:25:45 +0000 (UTC) Subject: [FFmpeg-user] need help with ffmpeg. References: Message-ID: Ahmad-D-Tariq gmail.com> writes: > actually i followed this tutorial > http://dmitrydzz-hobby.blogspot.com/2012/04/how-to-build-ffmpeg-and-use-it-in.html >From a (very) quick look, the build script on this site looks completely broken and should not be used. Please do not top-post here, it is considered rude. Carl Eugen From cehoyos at ag.or.at Fri Apr 12 15:26:35 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 13:26:35 +0000 (UTC) Subject: [FFmpeg-user] need help with ffmpeg. References: Message-ID: Liam Condron-Farnos <23liam googlemail.com> writes: > I'm not 100% certain, but I don't think ffmpeg can do > this (that's only from a quick look over > the documentation, so I may be mistaken). Maybe try > another tool, like sox. Given that FFmpeg can use sox, I am not sure above is 100% accurate. Carl Eugen From ozenozkaya at gmail.com Fri Apr 12 15:46:04 2013 From: ozenozkaya at gmail.com (=?ISO-8859-1?Q?=F6zen_=F6zkaya?=) Date: Fri, 12 Apr 2013 16:46:04 +0300 Subject: [FFmpeg-user] Record Webcam To a Different File Every Hour Message-ID: Hi, I want to record my webcam video to a different file every hour automatically. Does ffmpeg has this functionality? Here below you can see my current save command: ffmpeg -f video4linux2 -r 25 -s 640x480 -i /dev/video0 out.avi I want to get output files like out_1.avi to out_24.avi Does ffmpeg has such a property? Best Regards. -- ?zen ?zkaya From pkoshevoy at gmail.com Fri Apr 12 15:59:07 2013 From: pkoshevoy at gmail.com (Pavel Koshevoy) Date: Fri, 12 Apr 2013 07:59:07 -0600 Subject: [FFmpeg-user] need help with ffmpeg. In-Reply-To: References: Message-ID: On Apr 12, 2013 6:39 AM, "Liam Condron-Farnos" <23liam at googlemail.com> wrote: > > >i want the functionality that when a person speaks then myapp > >records make video/audio and plays back it by changing its audio pitch, > >like tomcat app. > > I'm not 100% certain, but I don't think ffmpeg can do this (that's only > from a quick look over the documentation, so I may be mistaken). Maybe try > another tool, like sox. Pitch shifting may be possible by chaining together audio tempo and audio resampling filters. The steps would be to double the audio tempo and resample the waveform at twice the sample rate (this may require to fix up the timestamps as well) -- the result should be audio of same duration at lower pitch. However, I haven't tried it myself. Pavel From timbot at oddlystudios.com Fri Apr 12 16:32:01 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Fri, 12 Apr 2013 10:32:01 -0400 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: That's interesting Paul, you might be on to something there. I found it very coincidental that the alpha from pure black areas was always 16. I'm actually doing two passes, and using rgba for everything. On my first pass, I have a moving gradient which is rgb24, and I apply a mask to it (with alphamerge) that was created using the alphaextract filter on an rgba png image (png color-type 6). I am using the output of that (which is now a nice animated gradient in the shape of text) as the luma source and doing another alphamerge against my original text image. Now my nice moving gradient reveals the original image. The only place I can think that YUV might be introduced would be in the alphaextract filter. But even that I encode into rgba png before I use it. I wonder if there is a better format to use than rgba png encoded. -t. On Fri, Apr 12, 2013 at 9:12 AM, Paul B Mahol wrote: > On 4/12/13, Tim Willison wrote: > > Hello folks, > > I thought I could share a tip I've used in my experiments with the > > alphaextract and alphamerge filters. > > > > I have found that the result of the alphamerge filter often leaves areas > > with some non-zero alpha. Comparing the mask file, areas that are pure > > black are not always reproduced as pure alpha after the merge. > > Looks like you used YUV source for mask. And result is expected because > YUV (limited range, not YUVJ) values can not be 0-16 or 235-255. > > > > > However, I found a way to deal with this by using the lut filter. It was > as > > simple as adding this to my alphamerge filter chain: > > > > [text2][out]alphamerge,format=rgba,lutrgb=a=if(gte(val\,18)\,val) > > > > The lutrgb is using an expression that lets me add a threshold for the > > alpha channel. In this case, I'm saying that if the alpha drops below 18, > > it should likely be zero, otherwise keep it the same. > > This cleaned up any alpha artefacts very nicely. > > > > Hope that is helpful for somebody. > > > > -t. > > > > > > On Thu, Apr 11, 2013 at 11:36 AM, Tim Willison > > wrote: > > > >> Hi Stefano! > >> > >> Thanks very much. Your comment was actually very helpful. > >> I took a second look at how I was viewing the output of my alphamerge > and > >> sure enough - the alpha information was correct. I am now tackling my > >> last > >> problem. > >> > >> My video output from the alphamerge is excellent, but consistently shows > >> a > >> value of 16 in the alpha channel in areas that should be completely > >> transparent. It has the effect of putting a very slight gray layer on > >> everything. > >> I am betting that this is some compression on my output (which is rgba, > >> png codec into a mov). > >> > >> You mentioned using rawvideo. I haven't tried that before, but it sounds > >> like what I might need. > >> > >> Thanks again for your help. > >> > >> -tim. > >> > >> > >> On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini > >> wrote: > >> > >>> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: > >>> > Hi, this is my first question ever to a mailing list (or anywhere!). > >>> > I have been using ffmpeg for about a year and a half and have figured > >>> out > >>> > many things on my own but I'm stumped regarding the alphaextract and > >>> > alphamerge filters. > >>> > I am able to get the proper result every time with images, but have > >>> > yet > >>> to > >>> > succeed with videos. I have tried many formats and searched > >>> > extensively > >>> but > >>> > have not found a single documented case of someone successfully using > >>> this > >>> > >>> > filter combination and sharing an example. The documentation on the > >>> > ffmpeg-filters page is very sparse, and does not give any details of > >>> what > >>> > formats/codecs are supported. > >>> > >>> Yes that's because ffmpeg-filters is supposed to only document filters. > >>> > >>> > > >>> > I think this filter would be very useful if I can figure it out. > >>> > > >>> > Step 1 - alphaextract > >>> > This seems to always work, and produces a grayscale video as > expected. > >>> In > >>> > my tests I have used rgba video, as well as quicktime with animation > >>> codec > >>> > (bgra). > >>> > > >>> > Step 2 - applying the extracted alpha > >>> > This works every time for images. However while the encode completes > >>> with > >>> > video it never provides the masking expected by mapping the extracted > >>> alpha > >>> > information to the alpha channel of the target video. In my tests > with > >>> > still images, I found that the output format must be a format that > >>> supports > >>> > an alpha channel, and I have tried several with video (which I > >>> determined > >>> > by looking at the source code of the filter). > >>> > >>> The question is, are you sure your output codec supports alpha (most > >>> don't, indeed I couldn't even tell which ones support alpha). You can > >>> try with rawvideo and make sure it is not a codec bug. > >>> > >>> > My command (taken from the docs) looks like this: > >>> > > >>> > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov > >>> [alpha]; > >>> > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png > >>> composite.mov > >>> > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the > >>> > FFmpeg > >>> > developers > >>> > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > >>> > 4.6.1-9ubuntu3) > >>> > configuration: --enable-gpl --enable-libass --enable-libfaac > >>> > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > >>> > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > >>> > --enable-libtheora --enable-libvorbis --enable-libvpx > --enable-libx264 > >>> > --enable-nonfree --enable-version3 --enable-libfreetype > >>> > --enable-filter=drawtext > >>> > libavutil 52. 25.100 / 52. 25.100 > >>> > libavcodec 55. 2.100 / 55. 2.100 > >>> > libavformat 55. 1.100 / 55. 1.100 > >>> > libavdevice 55. 0.100 / 55. 0.100 > >>> > libavfilter 3. 49.100 / 3. 49.100 > >>> > libswscale 2. 2.100 / 2. 2.100 > >>> > libswresample 0. 17.102 / 0. 17.102 > >>> > libpostproc 52. 2.100 / 52. 2.100 > >>> > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > >>> > Metadata: > >>> > major_brand : mp42 > >>> > minor_version : 0 > >>> > compatible_brands: mp42mp41 > >>> > creation_time : 2013-04-07 18:35:55 > >>> > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > >>> > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), > yuv420p, > >>> > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc > >>> > Metadata: > >>> > creation_time : 2013-04-07 18:35:55 > >>> > handler_name : ?Mainconcept Video Media Handler > >>> > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, > >>> > stereo, > >>> > fltp, 189 kb/s > >>> > Metadata: > >>> > creation_time : 2013-04-07 18:35:55 > >>> > handler_name : #Mainconcept MP4 Sound Media Handler > >>> > Output #0, mov, to 'composite.mov': > >>> > Metadata: > >>> > major_brand : mp42 > >>> > minor_version : 0 > >>> > compatible_brands: mp42mp41 > >>> > encoder : Lavf55.1.100 > >>> > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, 1280x720 > >>> [SAR > >>> > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > >>> > Metadata: > >>> > creation_time : 2013-04-07 18:35:55 > >>> > handler_name : ?Mainconcept Video Media Handler > >>> > Stream mapping: > >>> > Stream #0:0 -> #0:0 (h264 -> png) > >>> > Press [q] to stop, [?] for help > >>> > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > >>> > bitrate=3177.5kbits/s dup=2 drop=0 > >>> > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing overhead > >>> > 0.067323% > >>> > > >>> > After this, I see only my red background (timbotundelay.mp4) and no > >>> alpha > >>> > applied. > >>> > >>> I tried a variant of your command and seems to work fine here. Also > note > >>> that you need a background to show an image with overlay. Something > >>> like: > >>> ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" > >>> > >>> should do. > >>> > >>> > Thanks so much to anyone who can help me out here. I'm running out of > >>> > things to try. I'd particularly like to see how people are using this > >>> > filter successfully. > >>> _______________________________________________ > >>> ffmpeg-user mailing list > >>> ffmpeg-user at ffmpeg.org > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >>> > >> > >> > >> > >> -- > >> Tim Willison > >> Lead Product Designer > >> www.oddlystudios.com > >> 495B Queen St. West > >> Toronto, ON. > >> 647 345 6275 > >> > > > > > > > > -- > > Tim Willison > > Lead Product Designer > > www.oddlystudios.com > > 495B Queen St. West > > Toronto, ON. > > 647 345 6275 > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From onemda at gmail.com Fri Apr 12 16:38:48 2013 From: onemda at gmail.com (Paul B Mahol) Date: Fri, 12 Apr 2013 14:38:48 +0000 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: On 4/12/13, Tim Willison wrote: > That's interesting Paul, you might be on to something there. I found it > very coincidental that the alpha from pure black areas was always 16. > > I'm actually doing two passes, and using rgba for everything. > On my first pass, I have a moving gradient which is rgb24, and I apply a > mask to it (with alphamerge) that was created using the alphaextract filter > on an rgba png image (png color-type 6). > > I am using the output of that (which is now a nice animated gradient in the > shape of text) as the luma source and doing another alphamerge against my > original text image. Now my nice moving gradient reveals the original > image. > > The only place I can think that YUV might be introduced would be in the > alphaextract filter. But even that I encode into rgba png before I use it. > I wonder if there is a better format to use than rgba png encoded. Does your rgb source have pure black 0 pixels? (you can see it with histogram filter) When converting it to gray8 with alpha extract, is it preserved? 0 values should remain 0. > > -t. > > On Fri, Apr 12, 2013 at 9:12 AM, Paul B Mahol wrote: > >> On 4/12/13, Tim Willison wrote: >> > Hello folks, >> > I thought I could share a tip I've used in my experiments with the >> > alphaextract and alphamerge filters. >> > >> > I have found that the result of the alphamerge filter often leaves >> > areas >> > with some non-zero alpha. Comparing the mask file, areas that are pure >> > black are not always reproduced as pure alpha after the merge. >> >> Looks like you used YUV source for mask. And result is expected because >> YUV (limited range, not YUVJ) values can not be 0-16 or 235-255. >> >> > >> > However, I found a way to deal with this by using the lut filter. It >> > was >> as >> > simple as adding this to my alphamerge filter chain: >> > >> > [text2][out]alphamerge,format=rgba,lutrgb=a=if(gte(val\,18)\,val) >> > >> > The lutrgb is using an expression that lets me add a threshold for the >> > alpha channel. In this case, I'm saying that if the alpha drops below >> > 18, >> > it should likely be zero, otherwise keep it the same. >> > This cleaned up any alpha artefacts very nicely. >> > >> > Hope that is helpful for somebody. >> > >> > -t. >> > >> > >> > On Thu, Apr 11, 2013 at 11:36 AM, Tim Willison >> > wrote: >> > >> >> Hi Stefano! >> >> >> >> Thanks very much. Your comment was actually very helpful. >> >> I took a second look at how I was viewing the output of my alphamerge >> and >> >> sure enough - the alpha information was correct. I am now tackling my >> >> last >> >> problem. >> >> >> >> My video output from the alphamerge is excellent, but consistently >> >> shows >> >> a >> >> value of 16 in the alpha channel in areas that should be completely >> >> transparent. It has the effect of putting a very slight gray layer on >> >> everything. >> >> I am betting that this is some compression on my output (which is >> >> rgba, >> >> png codec into a mov). >> >> >> >> You mentioned using rawvideo. I haven't tried that before, but it >> >> sounds >> >> like what I might need. >> >> >> >> Thanks again for your help. >> >> >> >> -tim. >> >> >> >> >> >> On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini >> >> wrote: >> >> >> >>> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: >> >>> > Hi, this is my first question ever to a mailing list (or >> >>> > anywhere!). >> >>> > I have been using ffmpeg for about a year and a half and have >> >>> > figured >> >>> out >> >>> > many things on my own but I'm stumped regarding the alphaextract >> >>> > and >> >>> > alphamerge filters. >> >>> > I am able to get the proper result every time with images, but have >> >>> > yet >> >>> to >> >>> > succeed with videos. I have tried many formats and searched >> >>> > extensively >> >>> but >> >>> > have not found a single documented case of someone successfully >> >>> > using >> >>> this >> >>> >> >>> > filter combination and sharing an example. The documentation on the >> >>> > ffmpeg-filters page is very sparse, and does not give any details >> >>> > of >> >>> what >> >>> > formats/codecs are supported. >> >>> >> >>> Yes that's because ffmpeg-filters is supposed to only document >> >>> filters. >> >>> >> >>> > >> >>> > I think this filter would be very useful if I can figure it out. >> >>> > >> >>> > Step 1 - alphaextract >> >>> > This seems to always work, and produces a grayscale video as >> expected. >> >>> In >> >>> > my tests I have used rgba video, as well as quicktime with >> >>> > animation >> >>> codec >> >>> > (bgra). >> >>> > >> >>> > Step 2 - applying the extracted alpha >> >>> > This works every time for images. However while the encode >> >>> > completes >> >>> with >> >>> > video it never provides the masking expected by mapping the >> >>> > extracted >> >>> alpha >> >>> > information to the alpha channel of the target video. In my tests >> with >> >>> > still images, I found that the output format must be a format that >> >>> supports >> >>> > an alpha channel, and I have tried several with video (which I >> >>> determined >> >>> > by looking at the source code of the filter). >> >>> >> >>> The question is, are you sure your output codec supports alpha (most >> >>> don't, indeed I couldn't even tell which ones support alpha). You can >> >>> try with rawvideo and make sure it is not a codec bug. >> >>> >> >>> > My command (taken from the docs) looks like this: >> >>> > >> >>> > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov >> >>> [alpha]; >> >>> > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png >> >>> composite.mov >> >>> > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the >> >>> > FFmpeg >> >>> > developers >> >>> > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro >> >>> > 4.6.1-9ubuntu3) >> >>> > configuration: --enable-gpl --enable-libass --enable-libfaac >> >>> > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb >> >>> > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp >> >>> > --enable-libtheora --enable-libvorbis --enable-libvpx >> --enable-libx264 >> >>> > --enable-nonfree --enable-version3 --enable-libfreetype >> >>> > --enable-filter=drawtext >> >>> > libavutil 52. 25.100 / 52. 25.100 >> >>> > libavcodec 55. 2.100 / 55. 2.100 >> >>> > libavformat 55. 1.100 / 55. 1.100 >> >>> > libavdevice 55. 0.100 / 55. 0.100 >> >>> > libavfilter 3. 49.100 / 3. 49.100 >> >>> > libswscale 2. 2.100 / 2. 2.100 >> >>> > libswresample 0. 17.102 / 0. 17.102 >> >>> > libpostproc 52. 2.100 / 52. 2.100 >> >>> > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': >> >>> > Metadata: >> >>> > major_brand : mp42 >> >>> > minor_version : 0 >> >>> > compatible_brands: mp42mp41 >> >>> > creation_time : 2013-04-07 18:35:55 >> >>> > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s >> >>> > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), >> yuv420p, >> >>> > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 >> >>> > tbc >> >>> > Metadata: >> >>> > creation_time : 2013-04-07 18:35:55 >> >>> > handler_name : ?Mainconcept Video Media Handler >> >>> > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, >> >>> > stereo, >> >>> > fltp, 189 kb/s >> >>> > Metadata: >> >>> > creation_time : 2013-04-07 18:35:55 >> >>> > handler_name : #Mainconcept MP4 Sound Media Handler >> >>> > Output #0, mov, to 'composite.mov': >> >>> > Metadata: >> >>> > major_brand : mp42 >> >>> > minor_version : 0 >> >>> > compatible_brands: mp42mp41 >> >>> > encoder : Lavf55.1.100 >> >>> > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, >> >>> > 1280x720 >> >>> [SAR >> >>> > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc >> >>> > Metadata: >> >>> > creation_time : 2013-04-07 18:35:55 >> >>> > handler_name : ?Mainconcept Video Media Handler >> >>> > Stream mapping: >> >>> > Stream #0:0 -> #0:0 (h264 -> png) >> >>> > Press [q] to stop, [?] for help >> >>> > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 >> >>> > bitrate=3177.5kbits/s dup=2 drop=0 >> >>> > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing >> >>> > overhead >> >>> > 0.067323% >> >>> > >> >>> > After this, I see only my red background (timbotundelay.mp4) and no >> >>> alpha >> >>> > applied. >> >>> >> >>> I tried a variant of your command and seems to work fine here. Also >> note >> >>> that you need a background to show an image with overlay. Something >> >>> like: >> >>> ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" >> >>> >> >>> should do. >> >>> >> >>> > Thanks so much to anyone who can help me out here. I'm running out >> >>> > of >> >>> > things to try. I'd particularly like to see how people are using >> >>> > this >> >>> > filter successfully. >> >>> _______________________________________________ >> >>> ffmpeg-user mailing list >> >>> ffmpeg-user at ffmpeg.org >> >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >>> >> >> >> >> >> >> >> >> -- >> >> Tim Willison >> >> Lead Product Designer >> >> www.oddlystudios.com >> >> 495B Queen St. West >> >> Toronto, ON. >> >> 647 345 6275 >> >> >> > >> > >> > >> > -- >> > Tim Willison >> > Lead Product Designer >> > www.oddlystudios.com >> > 495B Queen St. West >> > Toronto, ON. >> > 647 345 6275 >> > _______________________________________________ >> > ffmpeg-user mailing list >> > ffmpeg-user at ffmpeg.org >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > > > -- > Tim Willison > Lead Product Designer > www.oddlystudios.com > 495B Queen St. West > Toronto, ON. > 647 345 6275 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From timbot at oddlystudios.com Fri Apr 12 16:57:13 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Fri, 12 Apr 2013 10:57:13 -0400 Subject: [FFmpeg-user] alphamerge filter use problem In-Reply-To: References: <20130410155114.GP7301@arborea> Message-ID: According to AfterEffects, both my moving gradient and the output from the alphaextract filter have pure zero black pixels. I also tested by having ffmpeg itself generate the inputs to be sure. Here's a test I tried: ------generate a fully alpha image (pure alpha) ffmpeg -f lavfi -i color=c=0x00000000:s=1280x166:r=30:sar=1 -y -pix_fmt rgba -vcodec png -s 1280x166 -r 30 -vframes 120 alphasquare.mov ------alpha extract the square (pure white - confirmed) ffmpeg -i alphasquare.mov -vf 'alphaextract' -y -pix_fmt rgb -vcodec png -s 1280x166 -r alphaextracted.mov ------apply white mask to pure white box ffmpeg -f lavfi -i color=c=0xFFFFFFFF:s=1280x166:r=30:sar=1 -i alphaextracted.mov -filter_complex '[0:0][1:0]alphamerge' -y -pix_fmt rgba -vcodec png -s 1280x166 -r 30 -vframes 120 alphacomposite.mov The resulting video should have pure white pixels, but it doesn't. They come out more like 235 than 255. If I skip the in-between step and do it all in one filter chain, however, I DO get a pure white: ------------------same thing in one line ffmpeg -f lavfi -i color=c=0x00000000:s=1280x166:r=30:sar=1 -f lavfi -i color=c=0xFFFFFF:s=1280x166:r=30:sar=1 -filter_complex '[0:0] alphaextract [alphaextracted];[1:0][alphaextracted] alphamerge' -y -pix_fmt rgba -vcodec png -s 1280x166 -r 30 -vframes 120 alphacomposite.mov So....after many combinations like this my best solution was to clean up the alpha channel rather than try to get the proper combination of encoding. And it worked pretty well. It seems reliable across various options and I've tried it even when I'm not using the alphaextract filter, but rather generating my own luma masks (with format=gray filter). -t. On Fri, Apr 12, 2013 at 10:38 AM, Paul B Mahol wrote: > On 4/12/13, Tim Willison wrote: > > That's interesting Paul, you might be on to something there. I found it > > very coincidental that the alpha from pure black areas was always 16. > > > > I'm actually doing two passes, and using rgba for everything. > > On my first pass, I have a moving gradient which is rgb24, and I apply a > > mask to it (with alphamerge) that was created using the alphaextract > filter > > on an rgba png image (png color-type 6). > > > > I am using the output of that (which is now a nice animated gradient in > the > > shape of text) as the luma source and doing another alphamerge against my > > original text image. Now my nice moving gradient reveals the original > > image. > > > > The only place I can think that YUV might be introduced would be in the > > alphaextract filter. But even that I encode into rgba png before I use > it. > > I wonder if there is a better format to use than rgba png encoded. > > Does your rgb source have pure black 0 pixels? > (you can see it with histogram filter) > > When converting it to gray8 with alpha extract, is it preserved? > > 0 values should remain 0. > > > > > -t. > > > > On Fri, Apr 12, 2013 at 9:12 AM, Paul B Mahol wrote: > > > >> On 4/12/13, Tim Willison wrote: > >> > Hello folks, > >> > I thought I could share a tip I've used in my experiments with the > >> > alphaextract and alphamerge filters. > >> > > >> > I have found that the result of the alphamerge filter often leaves > >> > areas > >> > with some non-zero alpha. Comparing the mask file, areas that are pure > >> > black are not always reproduced as pure alpha after the merge. > >> > >> Looks like you used YUV source for mask. And result is expected because > >> YUV (limited range, not YUVJ) values can not be 0-16 or 235-255. > >> > >> > > >> > However, I found a way to deal with this by using the lut filter. It > >> > was > >> as > >> > simple as adding this to my alphamerge filter chain: > >> > > >> > [text2][out]alphamerge,format=rgba,lutrgb=a=if(gte(val\,18)\,val) > >> > > >> > The lutrgb is using an expression that lets me add a threshold for the > >> > alpha channel. In this case, I'm saying that if the alpha drops below > >> > 18, > >> > it should likely be zero, otherwise keep it the same. > >> > This cleaned up any alpha artefacts very nicely. > >> > > >> > Hope that is helpful for somebody. > >> > > >> > -t. > >> > > >> > > >> > On Thu, Apr 11, 2013 at 11:36 AM, Tim Willison > >> > wrote: > >> > > >> >> Hi Stefano! > >> >> > >> >> Thanks very much. Your comment was actually very helpful. > >> >> I took a second look at how I was viewing the output of my alphamerge > >> and > >> >> sure enough - the alpha information was correct. I am now tackling my > >> >> last > >> >> problem. > >> >> > >> >> My video output from the alphamerge is excellent, but consistently > >> >> shows > >> >> a > >> >> value of 16 in the alpha channel in areas that should be completely > >> >> transparent. It has the effect of putting a very slight gray layer on > >> >> everything. > >> >> I am betting that this is some compression on my output (which is > >> >> rgba, > >> >> png codec into a mov). > >> >> > >> >> You mentioned using rawvideo. I haven't tried that before, but it > >> >> sounds > >> >> like what I might need. > >> >> > >> >> Thanks again for your help. > >> >> > >> >> -tim. > >> >> > >> >> > >> >> On Wed, Apr 10, 2013 at 11:51 AM, Stefano Sabatini > >> >> wrote: > >> >> > >> >>> On date Sunday 2013-04-07 15:44:07 -0400, Tim Willison wrote: > >> >>> > Hi, this is my first question ever to a mailing list (or > >> >>> > anywhere!). > >> >>> > I have been using ffmpeg for about a year and a half and have > >> >>> > figured > >> >>> out > >> >>> > many things on my own but I'm stumped regarding the alphaextract > >> >>> > and > >> >>> > alphamerge filters. > >> >>> > I am able to get the proper result every time with images, but > have > >> >>> > yet > >> >>> to > >> >>> > succeed with videos. I have tried many formats and searched > >> >>> > extensively > >> >>> but > >> >>> > have not found a single documented case of someone successfully > >> >>> > using > >> >>> this > >> >>> > >> >>> > filter combination and sharing an example. The documentation on > the > >> >>> > ffmpeg-filters page is very sparse, and does not give any details > >> >>> > of > >> >>> what > >> >>> > formats/codecs are supported. > >> >>> > >> >>> Yes that's because ffmpeg-filters is supposed to only document > >> >>> filters. > >> >>> > >> >>> > > >> >>> > I think this filter would be very useful if I can figure it out. > >> >>> > > >> >>> > Step 1 - alphaextract > >> >>> > This seems to always work, and produces a grayscale video as > >> expected. > >> >>> In > >> >>> > my tests I have used rgba video, as well as quicktime with > >> >>> > animation > >> >>> codec > >> >>> > (bgra). > >> >>> > > >> >>> > Step 2 - applying the extracted alpha > >> >>> > This works every time for images. However while the encode > >> >>> > completes > >> >>> with > >> >>> > video it never provides the masking expected by mapping the > >> >>> > extracted > >> >>> alpha > >> >>> > information to the alpha channel of the target video. In my tests > >> with > >> >>> > still images, I found that the output format must be a format that > >> >>> supports > >> >>> > an alpha channel, and I have tried several with video (which I > >> >>> determined > >> >>> > by looking at the source code of the filter). > >> >>> > >> >>> The question is, are you sure your output codec supports alpha (most > >> >>> don't, indeed I couldn't even tell which ones support alpha). You > can > >> >>> try with rawvideo and make sure it is not a codec bug. > >> >>> > >> >>> > My command (taken from the docs) looks like this: > >> >>> > > >> >>> > ffmpeg -i timbotundelay.mp4 -vf "movie=timbotoverlayextracted.mov > >> >>> [alpha]; > >> >>> > [in][alpha] alphamerge [out]" -an -y -pix_fmt rgba -vcodec png > >> >>> composite.mov > >> >>> > ffmpeg version git-2013-04-06-1177416 Copyright (c) 2000-2013 the > >> >>> > FFmpeg > >> >>> > developers > >> >>> > built on Apr 6 2013 17:31:47 with gcc 4.6.1 (Ubuntu/Linaro > >> >>> > 4.6.1-9ubuntu3) > >> >>> > configuration: --enable-gpl --enable-libass --enable-libfaac > >> >>> > --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb > >> >>> > --enable-libopencore-amrwb --enable-libspeex --enable-librtmp > >> >>> > --enable-libtheora --enable-libvorbis --enable-libvpx > >> --enable-libx264 > >> >>> > --enable-nonfree --enable-version3 --enable-libfreetype > >> >>> > --enable-filter=drawtext > >> >>> > libavutil 52. 25.100 / 52. 25.100 > >> >>> > libavcodec 55. 2.100 / 55. 2.100 > >> >>> > libavformat 55. 1.100 / 55. 1.100 > >> >>> > libavdevice 55. 0.100 / 55. 0.100 > >> >>> > libavfilter 3. 49.100 / 3. 49.100 > >> >>> > libswscale 2. 2.100 / 2. 2.100 > >> >>> > libswresample 0. 17.102 / 0. 17.102 > >> >>> > libpostproc 52. 2.100 / 52. 2.100 > >> >>> > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'timbotundelay.mp4': > >> >>> > Metadata: > >> >>> > major_brand : mp42 > >> >>> > minor_version : 0 > >> >>> > compatible_brands: mp42mp41 > >> >>> > creation_time : 2013-04-07 18:35:55 > >> >>> > Duration: 00:00:04.97, start: 0.000000, bitrate: 241 kb/s > >> >>> > Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), > >> yuv420p, > >> >>> > 1280x720 [SAR 1:1 DAR 16:9], 35 kb/s, 30 fps, 30 tbr, 30k tbn, 60 > >> >>> > tbc > >> >>> > Metadata: > >> >>> > creation_time : 2013-04-07 18:35:55 > >> >>> > handler_name : ?Mainconcept Video Media Handler > >> >>> > Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, > >> >>> > stereo, > >> >>> > fltp, 189 kb/s > >> >>> > Metadata: > >> >>> > creation_time : 2013-04-07 18:35:55 > >> >>> > handler_name : #Mainconcept MP4 Sound Media Handler > >> >>> > Output #0, mov, to 'composite.mov': > >> >>> > Metadata: > >> >>> > major_brand : mp42 > >> >>> > minor_version : 0 > >> >>> > compatible_brands: mp42mp41 > >> >>> > encoder : Lavf55.1.100 > >> >>> > Stream #0:0(eng): Video: png (png / 0x20676E70), rgba, > >> >>> > 1280x720 > >> >>> [SAR > >> >>> > 1:1 DAR 16:9], q=2-31, 200 kb/s, 15360 tbn, 30 tbc > >> >>> > Metadata: > >> >>> > creation_time : 2013-04-07 18:35:55 > >> >>> > handler_name : ?Mainconcept Video Media Handler > >> >>> > Stream mapping: > >> >>> > Stream #0:0 -> #0:0 (h264 -> png) > >> >>> > Press [q] to stop, [?] for help > >> >>> > frame= 151 fps= 64 q=0.0 Lsize= 1952kB time=00:00:05.03 > >> >>> > bitrate=3177.5kbits/s dup=2 drop=0 > >> >>> > video:1951kB audio:0kB subtitle:0 global headers:0kB muxing > >> >>> > overhead > >> >>> > 0.067323% > >> >>> > > >> >>> > After this, I see only my red background (timbotundelay.mp4) and > no > >> >>> alpha > >> >>> > applied. > >> >>> > >> >>> I tried a variant of your command and seems to work fine here. Also > >> note > >> >>> that you need a background to show an image with overlay. Something > >> >>> like: > >> >>> ffplay -f lavfi color -vf "movie=composite.mov, [in] overlay" > >> >>> > >> >>> should do. > >> >>> > >> >>> > Thanks so much to anyone who can help me out here. I'm running out > >> >>> > of > >> >>> > things to try. I'd particularly like to see how people are using > >> >>> > this > >> >>> > filter successfully. > >> >>> _______________________________________________ > >> >>> ffmpeg-user mailing list > >> >>> ffmpeg-user at ffmpeg.org > >> >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> >>> > >> >> > >> >> > >> >> > >> >> -- > >> >> Tim Willison > >> >> Lead Product Designer > >> >> www.oddlystudios.com > >> >> 495B Queen St. West > >> >> Toronto, ON. > >> >> 647 345 6275 > >> >> > >> > > >> > > >> > > >> > -- > >> > Tim Willison > >> > Lead Product Designer > >> > www.oddlystudios.com > >> > 495B Queen St. West > >> > Toronto, ON. > >> > 647 345 6275 > >> > _______________________________________________ > >> > ffmpeg-user mailing list > >> > ffmpeg-user at ffmpeg.org > >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > > > > > > > -- > > Tim Willison > > Lead Product Designer > > www.oddlystudios.com > > 495B Queen St. West > > Toronto, ON. > > 647 345 6275 > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From cehoyos at ag.or.at Fri Apr 12 17:04:24 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 15:04:24 +0000 (UTC) Subject: [FFmpeg-user] Matrox MQSink Filter References: Message-ID: Bo?tjan Strojan gmail.com> writes: > https://dl.dropboxusercontent.com/u/79532365/matroxQsink/C0005.avi I opened ticket #2464, thank you for the sample! (How long is the sample approximately supposed to play and which application plays it?) Carl Eugen From Guillaume.Chanel at unige.ch Fri Apr 12 17:09:03 2013 From: Guillaume.Chanel at unige.ch (Guillaume Chanel) Date: Fri, 12 Apr 2013 15:09:03 +0000 Subject: [FFmpeg-user] Problem of PTS for video segmentation Message-ID: Dear FFMPEG users, I am struggling with an issue using ffmpeg timestamps, especially -vsynch and -copyts. My goal is to cut a video according to a given time and number of frames (i.e. from time T up to F frames after) but keeping the same times stamps or at least the same non constant frame rate. This means that I would like to: - cut my video; - change the codec (since Camtasia cannot be encoded and codec copy does not work); - keep the same time between the PTS of two given frames. So far I am using the following FFMPEG command: ffmpeg -y -i inputVideo.asf -ss 353.035 -vcodec mpeg4 -vframes 100 -vsync 0 -copyts -vf showinfo outputVideob.avi 2> videoinfo.txt I tried without the -vsync and and -copyts which gave the same results. I also added the showinfo filter to try to understand the problem. Here is a reduced output of the command: ------------------------- ffmpeg version 0.10.6-6:0.10.6-0ubuntu0jon1~oneiric1 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 12 2012 13:11:34 with gcc 4.6.1 configuration: --arch=amd64 --enable-pthreads --enable-runtime-cpudetect --extra-version='6:0.10.6-0ubuntu0jon1~oneiric1' --libdir=/usr/lib/x86_64-linux-gnu --disable-stripping --prefix=/usr --enable-bzlib --enable-libdc1394 --enable-libfreetype --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-librtmp --enable-libopenjpeg --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-vaapi --enable-vdpau --enable-libvorbis --enable-libvpx --enable-zlib --enable-gpl --enable-postproc --enable-libcdio --enable-x11grab --enable-libx264 --shlibdir=/usr/lib/x86_64-linux-gnu --enable-shared --disable-static libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100 / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, asf, from '/user/mmi/emotion/data/EATMINT/ClientP02/TobiiExport/D02P02/ScreenRecordings_EATMI_Expe_D02P02_D02P02_2012-06-19_15-13-31.3831205_5725692445_8077573997.asf': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 Duration: 00:39:11.98, start: 0.000000, bitrate: 1581 kb/s Stream #0:0(fre): Video: camtasia (tscc / 0x63637374), bgra, 1280x1024, 1000000 kb/s, 59.67 tbr, 1k tbn, 1k tbc Incompatible pixel format 'bgra' for codec 'mpeg4', auto-selecting format 'yuv420p' [buffer @ 0xc9da80] w:1280 h:1024 pixfmt:bgra tb:1/1000000 sar:0/1 sws_param: [showinfo @ 0xc9f3c0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'Parsed_showinfo_0' [scale @ 0xca1580] w:1280 h:1024 fmt:bgra -> w:1280 h:1024 fmt:yuv420p flags:0x4 Output #0, avi, to '/home/chanel/D02P02_Screen_Collab.avi': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 ISFT : Lavf53.32.100 Stream #0:0(fre): Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1280x1024, q=2-31, 200 kb/s, 59.67 tbn, 59.67 tbc Stream mapping: Stream #0:0 -> #0:0 (camtasia -> mpeg4) Press [q] to stop, [?] for help n:0 pts:353035000 pts_time:353.035000 pos:156948836 fmt:yuv420p sar:0/1 s:1280x1024 i:P iskey:1 type:? checksum:13764275 plane_checksum:[F47E3FFC 6811D07D 1EC431ED 00000000] [showinfo @ 0xc9f3c0] n:1 pts:353161000 pts_time:353.161000 pos:156948836 fmt:yuv420p sar:0/1 s:1280x1024 i:P iskey:1 type:? checksum:0F8DFF59 plane_checksum:[484FE07F E077B316 DA056BB5 00000000] [showinfo @ 0xc9f3c0] n:2 pts:353236000 pts_time:353.236000 pos:157052836 fmt:yuv420p sar:0/1 s:1280x1024 i:P iskey:1 type:? checksum:E83B38CB plane_checksum:[23802B1D AAA6C229 F0374B76 00000000] [showinfo @ 0xc9f3c0] n:3 pts:353317000 pts_time:353.317000 pos:157068836 fmt:yuv420p sar:0/1 s:1280x1024 i:P iskey:1 type:? checksum:2C92157F plane_checksum:[5D2C0CB0 5CB2C00B BC8C48B5 00000000] . . . frame= 100 fps= 26 q=31.0 Lsize= 575kB time=00:00:09.88 bitrate= 476.2kbits/s video:555kB audio:0kB global headers:0kB muxing overhead 3.497475% ------------------------- As you can be seen from the output the video is cutted at the correct PTS: 353.035 and records 100 frames as requested. So far so good. Now I would like to see the output. For this I employ the ffprobe command: ffprobe -show_frames outputVideob.avi | more I will only list the values of the field pkt_pts_time for frame 0 to 3: Frame 0 : 0.000000 Frame 1 : 0.134078 Frame 2 : 0.201117 Frame 3 : 0.284916 As you can see the PTS time difference between frames has changed. This could be seen by comparing the showinfo filter output from the ffprobe output: Frame1 - Frame0 for the showinfo filter: 353.161000 ? 353.035000 = 0.126 Frame1 - Frame0 for the ffprobe output: 0.134078 ? 0.000000 = 0.134078 Is there a way to keep the same time difference as in the input ? Notice that the frame rate of my files are not constant I would not be satisfied with a constant frame rate. Also the absolute value are not the same despite of the -copyts. But this I do not care that much, I only need to keep the same non constant rate between frames from the input file to the output file. What am I doing wrong ? Thanks in advance for your help, Guillaume Chanel From cehoyos at ag.or.at Fri Apr 12 17:34:35 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 15:34:35 +0000 (UTC) Subject: [FFmpeg-user] Problem of PTS for video segmentation References: Message-ID: Guillaume Chanel unige.ch> writes: > ffmpeg version 0.10.6-6:0.10.6-0ubuntu0jon1~oneiric1 This is old, please test current git head. (It is possible that you are missing -copytb) Carl Eugen From cehoyos at ag.or.at Fri Apr 12 17:40:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 15:40:50 +0000 (UTC) Subject: [FFmpeg-user] Error: timebase not supported by MPEG 4 standard. Not sure why References: Message-ID: Simon Daniels gmail.com> writes: > > Please use -r 60000/1001 to specify 59.9 fps > > I also figured out that simply "59.94" works. Can > I get away with that since the rest of my code > already converts to a decimal? Or should I change > over to fractional values? To the best of my knowledge, the only way to specify a framerate of 60000/1001 is to pass "60000/1001" to one of the appropriate FFmpeg options. (Or do you want a framerate of 59.94 ?) Please do not top-post here. Carl Eugen From me at renecalles.de Fri Apr 12 19:04:43 2013 From: me at renecalles.de (me at renecalles.de) Date: Fri, 12 Apr 2013 19:04:43 +0200 Subject: [FFmpeg-user] ffmpeg 1.2 for Mac OS X 10.5.8 (Leopard) In-Reply-To: References: Message-ID: Hi, would compiling it also be an option for you? I wrote an tutorial on my blog: http://www.renevolution.com/how-to-install-ffmpeg-on-mac-os-x/ @Moderators: I hope providing the URL is not considered as rude, if it is please let me know and i'll never include such kind of link. Best, Ren? Am 2013-04-11 23:24, schrieb Jean-Paul BLANC: > Hi, > > Where could I find binary for ffmpeg 1.2 for Mac OS X 10.5.8 (Leopard) > ? > Could I trust the one that I found on http://ffmpegmac.net/ ? > > JiPaul. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From me at renecalles.de Fri Apr 12 19:08:51 2013 From: me at renecalles.de (me at renecalles.de) Date: Fri, 12 Apr 2013 19:08:51 +0200 Subject: [FFmpeg-user] Record Webcam To a Different File Every Hour In-Reply-To: References: Message-ID: <92e5e15de5f2620d3b755766e80f27c9@renecalles.de> Hi ?zen, two options are coming in my mind for this topic: 1) If you're on linux use a cronjob every hour and your command could look like ffmpeg -f video4linux2 -r 25 -s 640x480 -i /dev/video0 -t 3599 out.avi 2) Make use of the segmenter included in FFmpeg. But i'm not shure such long segments are supported and you might need to choose antoher format such as MP4/MPEG-TS as output. Both options are not tested but this is where i would start testing. Best Regards, Ren? Am 2013-04-12 15:46, schrieb ?zen ?zkaya: > Hi, > > I want to record my webcam video to a different file every hour > automatically. Does ffmpeg has this functionality? > > Here below you can see my current save command: > ffmpeg -f video4linux2 -r 25 -s 640x480 -i /dev/video0 out.avi > > I want to get output files like out_1.avi to out_24.avi > > Does ffmpeg has such a property? > > Best Regards. From nfrancia87 at gmail.com Fri Apr 12 19:12:41 2013 From: nfrancia87 at gmail.com (Nicolas Francia) Date: Fri, 12 Apr 2013 14:12:41 -0300 Subject: [FFmpeg-user] Statistical Multiplexing Message-ID: Dear users, I am writing to know if there is a way to achieve statistical multiplexing using ffmpeg, afert i?ve received a TS packet, demux it, decoded it and re encoded it. So i can manage better the band with available to stream. Thanks a lot! From seandarcy2 at gmail.com Fri Apr 12 19:50:08 2013 From: seandarcy2 at gmail.com (sean darcy) Date: Fri, 12 Apr 2013 13:50:08 -0400 Subject: [FFmpeg-user] AAC Encoding Guide: codecs for lectures ?? Message-ID: In the ffmpeg wiki AAC Encoding Guide: > whalesong, or a recorded lecture - though there are special speech codecs that you should consider for the latter Are we're talking about ilbc or amr? I'm encoding my lectures to play android and iphones. any suggestions? sean From cehoyos at ag.or.at Fri Apr 12 20:07:48 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 12 Apr 2013 18:07:48 +0000 (UTC) Subject: [FFmpeg-user] Record Webcam To a Different File Every Hour References: Message-ID: ?zen ?zkaya gmail.com> writes: > I want to record my webcam video to a different file every hour > automatically. Does ffmpeg has this functionality? Please try the segmenter, it supports all (?) fileformats and should be tested extensively: http://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment Carl Eugen From adunaic at hotmail.co.uk Fri Apr 12 11:17:42 2013 From: adunaic at hotmail.co.uk (adunaic) Date: Fri, 12 Apr 2013 02:17:42 -0700 (PDT) Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) In-Reply-To: References: <4F46A788.9070406@gmail.com> <4F51B7E9.5080005@gmail.com> <1365601163170-4658383.post@n4.nabble.com> Message-ID: <1365758262944-4658421.post@n4.nabble.com> > > From Oggs pages they imply that dirac can be placed > > in an ogg container but I still get the error: >FFmpeg currently does not support muxing dirac in ogg, >patch welcome! Thank you for clarifying this. I am just a casual user with not enough knowledge to add this, sorry. > > Could not write header for output file #0 (incorrect codec parameters > ?): > > Operation not permitted > > > > (ffmpeg version 1.0.5 ) > (This is outdated.) It is all that is offered in Fedora 18, I struggle to compile from source. > > Any thoughts? > For future questions: Please always provide > your failing command line together with complete, > uncut console output. > Carl Eugen Will do in the future, thanks again. Robin -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Problem-transcoding-from-MP4-H264-AAC-to-OGG-DIRAC-FLAC-tp4416502p4658421.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From bappa.mondal at senabi.com Fri Apr 12 18:09:54 2013 From: bappa.mondal at senabi.com (bappa.mondal at senabi.com) Date: Fri, 12 Apr 2013 21:39:54 +0530 Subject: [FFmpeg-user] FFmpeg issue Message-ID: Dear Team, I am using a bit simple command and that works fine through terminal. but when I use it from web it gives me a blank mp4 file. I have a centos 6 ser version and ffmpeg config is - ffmpeg version 1.2 built on Mar 25 2013 20:01:59 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3) configuration: --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libvo-aacenc --enable-libxvid --disable-ffplay --enable-shared --enable-gpl --enable-postproc --enable-nonfree --enable-avfilter --enable-pthreads --extra-cflags=-fPIC now I am running command like, "ffmpeg -i /path-to-videos/fail_1.flv -vcodec libx264 /path-to-videos/failed.mp4" Note: path-to-videos = actual path in the server. Can you please suggest on that. Waiting for your valuable response. 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I am using a bit simple command and that works fine through terminal. but when I use it from web it gives me a blank mp4 file. I have a centos 6 ser version and ffmpeg config is - I am running command like "ffmpeg -i /path-to-videos/fail_1.flv -vcodec libx264 /path-to-videos/failed.mp4" *path-to-videos = actual path in the server. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/AVI-WMV-FLV-MOV-and-MP4-to-libx264-MP4-conversion-bugs-tp4656171p4658442.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From nfrancia87 at gmail.com Fri Apr 12 16:57:16 2013 From: nfrancia87 at gmail.com (Nicolas Francia) Date: Fri, 12 Apr 2013 11:57:16 -0300 Subject: [FFmpeg-user] Statistical multiplexing Message-ID: Dear users, I am writing to know if there is a way to achieve statistical multiplexing using ffmpeg, afert i?ve received a TS packet, demux it, decoded it and re enconded. So i can manage better the band with available to stream. Thanks a lot! From 23liam at googlemail.com Fri Apr 12 22:07:19 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Fri, 12 Apr 2013 21:07:19 +0100 Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC) In-Reply-To: <1365758262944-4658421.post@n4.nabble.com> References: <4F46A788.9070406@gmail.com> <4F51B7E9.5080005@gmail.com> <1365601163170-4658383.post@n4.nabble.com> <1365758262944-4658421.post@n4.nabble.com> Message-ID: >> > (ffmpeg version 1.0.5 ) >> >> (This is outdated.) > >It is all that is offered in Fedora 18, I struggle to compile from source. You can get an up-to-date static build from the downloads page: https://ffmpeg.org/download.html From timothygu99 at gmail.com Sat Apr 13 02:04:59 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Fri, 12 Apr 2013 17:04:59 -0700 Subject: [FFmpeg-user] AAC Encoding Guide: codecs for lectures ?? In-Reply-To: References: Message-ID: Hi, On Apr 12, 2013 10:51 AM, "sean darcy" wrote: > > In the ffmpeg wiki AAC Encoding Guide: > > > whalesong, or a recorded lecture - though there are special speech codecs that you should consider for the latter > > Are we're talking about ilbc or amr? > > I'm encoding my lectures to play android and iphones. any suggestions? Opus is probably better. But without additional apps, iOS and Android will play neither of iLBC and opus, and iOS cannot play amr. Timothy From asorti6 at gmail.com Sat Apr 13 02:24:32 2013 From: asorti6 at gmail.com (Owen Jones) Date: Sat, 13 Apr 2013 01:24:32 +0100 Subject: [FFmpeg-user] channelsplit seems to mute one channel Message-ID: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> I am trying to use the channelsplit filter to split a single stereo channel into two separate channels. However when I look at the resulting file in Quicktime (sorry don't know a better way to check this with ffprobe) the second (right) channel appears as disabled. Am I doing something wrong, or is there a way to force a channel to be enabled? ffmpeg -i VTS_01_1.VOB -codec:v copy -codec:a pcm_s16le -map 0 -filter_complex 'channelsplit' test.mov ffmpeg version 1.1.git-0d194ee Copyright (c) 2000-2013 the FFmpeg developers built on Feb 8 2013 09:49:39 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn) configuration: --enable-nonfree --enable-gpl --enable-version3 --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libfaac --enable-libxvid --enable-libx264 --enable-libvpx --enable-hardcoded-tables --enable-shared --enable-pthreads --disable-indevs --cc=clang libavutil 52. 17.101 / 52. 17.101 libavcodec 54. 91.102 / 54. 91.102 libavformat 54. 61.104 / 54. 61.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 35.101 / 3. 35.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [mpeg @ 0x7fcc88806600] max_analyze_duration 5000000 reached at 5000000 microseconds Input #0, mpeg, from 'VTS_01_1.VOB': Duration: 00:00:10.60, start: 0.235211, bitrate: 803800 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 256 kb/s Output #0, mov, to 'test.mov': Metadata: encoder : Lavf54.61.104 Stream #0:0: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 1 channels (FL), s16, 768 kb/s Stream #0:1: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 1 channels (FR), s16, 768 kb/s Stream #0:2: Video: mpeg2video (m2v1 / 0x3176326D), yuv420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 90k tbn, 90k tbc Stream mapping: Stream #0:1 (ac3) -> channelsplit channelsplit:FL -> Stream #0:0 (pcm_s16le) channelsplit:FR -> Stream #0:1 (pcm_s16le) Stream #0:0 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 9061 fps=1606 q=-1.0 Lsize= 258633kB time=00:06:02.36 bitrate=5847.0kbits/s video:190544kB audio:67890kB subtitle:0 global headers:0kB muxing overhead 0.076662% From andrey.aleksandrovich at googlemail.com Sat Apr 13 08:46:31 2013 From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich) Date: Sat, 13 Apr 2013 09:46:31 +0300 Subject: [FFmpeg-user] channelsplit seems to mute one channel In-Reply-To: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> Message-ID: I think you have miss -map_channel option (see man ffmpeg at appropriate section). On 4/13/13, Owen Jones wrote: > I am trying to use the channelsplit filter to split a single stereo channel > into two separate channels. However when I look at the resulting file in > Quicktime (sorry don't know a better way to check this with ffprobe) the > second (right) channel appears as disabled. Am I doing something wrong, or > is there a way to force a channel to be enabled? > > ffmpeg -i VTS_01_1.VOB -codec:v copy -codec:a pcm_s16le -map 0 > -filter_complex 'channelsplit' test.mov > ffmpeg version 1.1.git-0d194ee Copyright (c) 2000-2013 the FFmpeg > developers > built on Feb 8 2013 09:49:39 with Apple clang version 4.1 > (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn) > configuration: --enable-nonfree --enable-gpl --enable-version3 > --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame > --enable-libvorbis --enable-libtheora --enable-libfaac --enable-libxvid > --enable-libx264 --enable-libvpx --enable-hardcoded-tables --enable-shared > --enable-pthreads --disable-indevs --cc=clang > libavutil 52. 17.101 / 52. 17.101 > libavcodec 54. 91.102 / 54. 91.102 > libavformat 54. 61.104 / 54. 61.104 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 35.101 / 3. 35.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > [mpeg @ 0x7fcc88806600] max_analyze_duration 5000000 reached at 5000000 > microseconds > Input #0, mpeg, from 'VTS_01_1.VOB': > Duration: 00:00:10.60, start: 0.235211, bitrate: 803800 kb/s > Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR > 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc > Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 256 kb/s > Output #0, mov, to 'test.mov': > Metadata: > encoder : Lavf54.61.104 > Stream #0:0: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 1 channels > (FL), s16, 768 kb/s > Stream #0:1: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 1 channels > (FR), s16, 768 kb/s > Stream #0:2: Video: mpeg2video (m2v1 / 0x3176326D), yuv420p, 720x576 > [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 90k tbn, 90k tbc > Stream mapping: > Stream #0:1 (ac3) -> channelsplit > channelsplit:FL -> Stream #0:0 (pcm_s16le) > channelsplit:FR -> Stream #0:1 (pcm_s16le) > Stream #0:0 -> #0:2 (copy) > Press [q] to stop, [?] for help > frame= 9061 fps=1606 q=-1.0 Lsize= 258633kB time=00:06:02.36 > bitrate=5847.0kbits/s > video:190544kB audio:67890kB subtitle:0 global headers:0kB muxing overhead > 0.076662% > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Sat Apr 13 09:16:32 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 13 Apr 2013 07:16:32 +0000 (UTC) Subject: [FFmpeg-user] channelsplit seems to mute one channel References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> Message-ID: Owen Jones gmail.com> writes: > I am trying to use the channelsplit filter to split > a single stereo channel into two separate channels. > However when I look at the resulting file in > Quicktime (sorry don't know a better way to check > this with ffprobe) The best way to test is: ffmpeg -i test.mov -map 0:0 out1.wav ffmpeg -i test.mov -map 0:1 out2.wav and test the resulting wav files. > the second (right) channel appears as disabled. The reason is that FFmpeg only marks the first audio track as default. The reason is that users typically don't like if all languages are played simultaneously with QT;-) (This was reported very often and only fixed recently.) It is not impossible to fix, but could you explain your use-case? Carl Eugen From ryan.lindsay at gmail.com Sat Apr 13 12:07:53 2013 From: ryan.lindsay at gmail.com (ryan lindsay) Date: Sat, 13 Apr 2013 20:07:53 +1000 Subject: [FFmpeg-user] Re encoding rtsp? Message-ID: Hi, just wondering if someone can help me. I have an ip camera that spits out an h.264 mpeg stream via rtsp. And I have it connected to a beagle board xm. What I would like to do is suck in the rtsp into the beagle board then I want to take an audio stream in from the beagle board, and I won't to munge them together and spit out the rtsp stream of the beagle board with video and audio. Is this possible to do with ffmpeg? And if so can someone give me an idea of how. Thanks in advance From wiebe at halfgaar.net Sat Apr 13 12:35:10 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sat, 13 Apr 2013 12:35:10 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> Message-ID: <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> Hi, Until now, I always had the LFE channel mixed in with the fronts 50%/50%. The mixing is done in my home-made multi-channel preamp/crossover. However, since I calibrated my subwoofer to be flat to 20 Hz with the rest of the audio bandwidth (it was too loud before), bass from films seems weak. That's when I learned the true method of reproducing the LFE channel. The channel is meant to be played back 10 dB louder than the other channels (see [2] and [3], so that one channel can have all the bass from the other speakers, and to be able to have more 'bang'. For example, [1] says: ==== This 10dB boost is achieved by calibration in the monitoring amplifiers at the studio; a full-scale LFE signal on the tape is set up to play 10dB louder than a full-scale signal on any other channel. (...snip...) One important point is that the 10dB boost to the LFE is performed at the final amplification stage. The potential extra volume of the LFE channel could not pass through interconnects, analogue recordings or digital recordings without going over the specified limits by a factor of 3. This, indeed, is the whole raison d'?tre of the LFE channel. It's a channel where by convention everything is recorded 10dB low to make room for louder bangs. ==== So, I was wondering if ffmpeg (using Mplayer, XMBC, etc) really doesn't apply the 10 dB boost and lets the amplifier do it. [1] has some detailed explanations why it really shouldn't be done in the digital domain. And it makes sense of course, because boosting LFE can't be done; it would mean reducing the rest by 10 dB to avoid clipping. This gives rise to a few questions: 1) As mentioned, does ffmpeg adhere to this and allow connect 5.1 speakers to apply the boost? And do all those Logitech/etc 5.1/7.1 PC speaker sets do so? [1] mentions that even home cinema receivers don't always get it right. 2) Are there downmixing routines in ffmpeg? The old liba52 did have them, and I saw in the source code that there actually was normalization applied so that mixing channels wouldn't clip them. However, I couldn't find anything about the LFE 10 dB boost, and how the channels were normalized and mixed in with left and right when not having .1 output on the sound card. I also know that when downmixing to 2.0, the AC-3 standard ([2]) requires dynamic compression to accommodate the extra signals, which makes sense, but I don't know of such a standard for downmixing to 4.0. Regards, Wiebe [1] http://www.avsforum.com/t/748147/lfe-subwoofers-and-interconnects-explained [2] http://www.atsc.org/cms/index.php/standards/standards/48-atsc-a52-standard [3] http://www.dolby.com/uploadedFiles/Assets/US/Doc/Professional/38_LFE.pdf From asorti6 at gmail.com Sat Apr 13 13:13:21 2013 From: asorti6 at gmail.com (Owen Jones) Date: Sat, 13 Apr 2013 12:13:21 +0100 Subject: [FFmpeg-user] channelsplit seems to mute one channel In-Reply-To: References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> Message-ID: <4461BDB2-F86E-4505-83D0-E813A34B095E@gmail.com> On 13 Apr 2013, at 08:16, Carl Eugen Hoyos wrote: > The best way to test is: > ffmpeg -i test.mov -map 0:0 out1.wav > ffmpeg -i test.mov -map 0:1 out2.wav > and test the resulting wav files. I tried this, there is a sound track in both channels > >> the second (right) channel appears as disabled. > > The reason is that FFmpeg only marks the first > audio track as default. The reason is that > users typically don't like if all languages > are played simultaneously with QT;-) > (This was reported very often and only fixed > recently.) > It is not impossible to fix, but could you > explain your use-case? Ah, thanks, can see that it makes sense to generally mute all except the first track. I am trying to transform a DVD video with joint stereo (which appears as a single track in Quicktime) into the format that iMovie uses which is two channel stereo (which appears as two tracks). Although iMovie will make its own transformation if fed joint stereo I don't really trust this and want to leave the video as untouched as possible following the ffmpeg transformation (basically using iMovie for its cataloguing only). I've tried using -map_channel and pan to try and re-enable the second channel but haven't managed to figure out a way of doing that. From cehoyos at ag.or.at Sat Apr 13 13:46:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 13 Apr 2013 11:46:22 +0000 (UTC) Subject: [FFmpeg-user] LFE channel + 10dB References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> Message-ID: Wiebe Cazemier halfgaar.net> writes: > 1) As mentioned, does ffmpeg adhere to this and > allow connect 5.1 speakers to apply the boost? Please test and tell us! Carl Eugen From cehoyos at ag.or.at Sat Apr 13 13:48:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 13 Apr 2013 11:48:29 +0000 (UTC) Subject: [FFmpeg-user] Re encoding rtsp? References: Message-ID: ryan lindsay gmail.com> writes: > What I would like to do is suck in the rtsp into the > beagle board then I want to take an audio stream in > from the beagle board, and I won't to munge them > together and spit out the rtsp stream of the beagle > board with video and audio. > > Is this possible to do with ffmpeg? Did you already try if ffmpeg allows you to receive the rtsp stream? If yes, I see no particular problem (except A/V sync, but that is something you have to test), if not, please provide the failing ffmpeg console output. Carl Eugen From ryan.lindsay at gmail.com Sat Apr 13 13:53:56 2013 From: ryan.lindsay at gmail.com (ryan lindsay) Date: Sat, 13 Apr 2013 21:53:56 +1000 Subject: [FFmpeg-user] Re encoding rtsp? In-Reply-To: References: Message-ID: So I can receive the rtsp stream, the thing I don't know how to do is encode the audio from the beagles device What would te options be for sync ing the audio to the video ? On Saturday, 13 April 2013, Carl Eugen Hoyos wrote: > ryan lindsay gmail.com> writes: > > > What I would like to do is suck in the rtsp into the > > beagle board then I want to take an audio stream in > > from the beagle board, and I won't to munge them > > together and spit out the rtsp stream of the beagle > > board with video and audio. > > > > Is this possible to do with ffmpeg? > > Did you already try if ffmpeg allows you to receive > the rtsp stream? > If yes, I see no particular problem (except A/V sync, > but that is something you have to test), if not, > please provide the failing ffmpeg console output. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From kqt4at5v at gmail.com Sat Apr 13 15:08:55 2013 From: kqt4at5v at gmail.com (kqt4at5v at gmail.com) Date: Sat, 13 Apr 2013 08:08:55 -0500 (CDT) Subject: [FFmpeg-user] maktroska and -ss Message-ID: I did a little recording with ffmpeg -y -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30000/1001 -s 714x404 -i :0.0+435,462 -acodec ac3 -ab 128k -vcodec libx264 -preset ultrafast -t 600 "file.mkv" When I try to edit with "ffplay -ss 13 -i file.mkv" it does not jump to the right time I do not seem to be able to jump to a specific second I am using packages from Fedora 18 What have I screwed up ffmpeg version 1.0.5 Copyright (c) 2000-2012 the FFmpeg developers built on Mar 10 2013 12:20:02 with gcc 4.7.2 (GCC) 20121109 (Red Hat 4.7.2-8) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib --mandir=/usr/share/man --arch=i686 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i686 -mtune=atom -fasynchronous-unwind-tables' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib --cpu=i686 --enable-runtime-cpudetect libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 Richard From wiebe at halfgaar.net Sat Apr 13 15:30:54 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sat, 13 Apr 2013 15:30:54 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> Message-ID: <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> ----- Original Message ----- > From: "Carl Eugen Hoyos" > To: ffmpeg-user at ffmpeg.org > Sent: Saturday, 13 April, 2013 1:46:22 PM > Subject: Re: [FFmpeg-user] LFE channel + 10dB > > Wiebe Cazemier halfgaar.net> writes: > > > 1) As mentioned, does ffmpeg adhere to this and > > allow connect 5.1 speakers to apply the boost? > > Please test and tell us! This is harder than it sounds, because you don't know if the test file already compensates for it. The reference I mentioned says: === Calibration tests generally hide this 10dB difference; a basic pink noise test will contain a signal that should play at 75dB SPL for each channel. For the normal channels, that means a signal 30dB below full scale (105dB-30dB = 75dB), but for the LFE channel it is a signal 40dB below full scale (115dB-40dB = 75dB): 1/3 of the signal amplitude. If those test signals are used to calibrate the amplifier so they all output at 75dB SPL, then everything is set up okay. The LFE channel now has 10dB of extra headroom above the other channels and can produce the full 115dB SPL when required. === But, I will do my best to find out, of course. I can tell you that the ALSA driver doesn't apply this boost to its .1 output (For my Audigy 2 ZS), because testfile [1] plays significantly louder when addressing the subwoofer. [1] http://www.halfgaar.net/media/chan-id.zip From cehoyos at ag.or.at Sat Apr 13 17:09:26 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 13 Apr 2013 15:09:26 +0000 (UTC) Subject: [FFmpeg-user] maktroska and -ss References: Message-ID: gmail.com> writes: > When I try to edit with "ffplay -ss 13 -i file.mkv" > it does not jump to the right time Please test a more recent version and / or provide the input sample. Carl Eugen From timbot at oddlystudios.com Sat Apr 13 17:44:29 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Sat, 13 Apr 2013 11:44:29 -0400 Subject: [FFmpeg-user] itsoffset breaking overlay filter Message-ID: Hi guys, I have actually filed a bug for this one (#2467) but I'm wondering if anyone else is experiencing a problem with using -itsoffset on the input pad for the overlay filter. I've used this technique many times to delay the start of an overlay, but the latest ffmpeg seems to have a problem. Every time I apply -itsoffset to an overlay, I get "Buffer queue overflow, dropping" repeated. This happens even when I'm just generating the video right inside ffmpeg: ffmpeg -f lavfi -i color=c=0xFFFFFF:s=1280x720:r=30:sar=1 -itsoffset -2 -f lavfi -i color=c=0x000000:s=1280x720:r=30:sar=1 -filter_complex 'overlay' -vframes 240 -y mytest.mp4 I have tried encoding to a wide variety of formats and the output is always the same. [Parsed_overlay_0 @ 0x34788c0] Buffer queue overflow, dropping. Is anybody else experiencing this? Unfortunately I can't think of many ways to offset an overlay without the -itsoffset flag, other than resorting to chopping the video up, overlaying, then joining it again. (Overlays that have no -itsoffset come out just fine). -Timbot From wiebe at halfgaar.net Sun Apr 14 10:56:13 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sun, 14 Apr 2013 10:56:13 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> Message-ID: <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> ----- Original Message ----- > I can tell you that the ALSA driver doesn't apply this boost to its > .1 output (For my Audigy 2 ZS), because testfile [1] plays > significantly louder when addressing the subwoofer. > > [1] http://www.halfgaar.net/media/chan-id.zip Sorry, I was typing this when I was talking; that never works. I forgot to add that currently, my ALSA mixer is set so that LFE is 10 dB louder, and then that test-file plays its LFE track louder (play with 'aplay -D surround51). If I set all channels equal in the mixer, all is equally loud. So, the driver leaves things alone, as it should. Now to find a proper test file, because this one is not played correctly by Mplayer (channel mapping is weird), and it's not AC-3, DTS, AAC, which is what I wanted to test. From cehoyos at ag.or.at Sun Apr 14 11:56:24 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 14 Apr 2013 09:56:24 +0000 (UTC) Subject: [FFmpeg-user] LFE channel + 10dB References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> Message-ID: Wiebe Cazemier halfgaar.net> writes: > this one is not played correctly by Mplayer > (channel mapping is weird), Did you report this? It does not sound like a known problem to me. Carl Eugen From wiebe at halfgaar.net Sun Apr 14 12:04:00 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sun, 14 Apr 2013 12:04:00 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> Message-ID: <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> ----- Original Message ----- > From: "Carl Eugen Hoyos" > To: ffmpeg-user at ffmpeg.org > Sent: Sunday, 14 April, 2013 11:56:24 AM > Subject: Re: [FFmpeg-user] LFE channel + 10dB > > Wiebe Cazemier halfgaar.net> writes: > > > this one is not played correctly by Mplayer > > (channel mapping is weird), > > Did you report this? > It does not sound like a known problem to me. I have yet to find out where this comes from, exactly. This is the result when played with Mplayer: * front left is played front left * front right is played front right * center is played rear left * lfe is played rear right * rear left is played over center out * rear right is played over LFE out From cehoyos at ag.or.at Sun Apr 14 12:19:06 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 14 Apr 2013 10:19:06 +0000 (UTC) Subject: [FFmpeg-user] LFE channel + 10dB References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> Message-ID: Wiebe Cazemier halfgaar.net> writes: > > > this one is not played correctly by Mplayer > > > (channel mapping is weird), > > > > Did you report this? > > It does not sound like a known problem to me. > > I have yet to find out where this comes from, exactly. > > This is the result when played with Mplayer: Sorry for being unclear: I do know how 5.1 is played incorrectly, no need to describe exactly. Afair, this works fine here, what it is needed is either a sample or a patch or (at least) complete, uncut console output. Carl Eugen From wiebe at halfgaar.net Sun Apr 14 13:48:40 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sun, 14 Apr 2013 13:48:40 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> Message-ID: <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> ----- Original Message ----- > From: "Carl Eugen Hoyos" > To: ffmpeg-user at ffmpeg.org > Sent: Sunday, 14 April, 2013 12:19:06 PM > Subject: Re: [FFmpeg-user] LFE channel + 10dB > > Sorry for being unclear: > I do know how 5.1 is played incorrectly, no > need to describe exactly. > Afair, this works fine here, what it is needed > is either a sample or a patch or (at least) > complete, uncut console output. > > Carl Eugen I did give the sample. It's a 6 channel wave file. Will that do? http://www.halfgaar.net/media/chan-id.zip Only this sample goes wrong. Films play normally. Xine plays the sample correctly. This is console output: MPlayer SVN-r35421-4.7.2 (C) 2000-2012 MPlayer Team 199 audio & 415 video codecs Playing chan-id.wav. Cache fill: 82.73% (3469916 bytes) libavformat version 54.36.100 (internal) Audio only file format detected. Load subtitles in ./ ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 44100 Hz, 6 ch, s16le, 4233.6 kbit/100.00% (ratio: 529200->529200) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory AO: [alsa] 48000Hz 6ch s16le (2 bytes per sample) Video: no video Starting playback... A: 2.5 (02.4) of 6.0 (06.0) 2.0% 44% From cehoyos at ag.or.at Sun Apr 14 14:14:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 14 Apr 2013 12:14:05 +0000 (UTC) Subject: [FFmpeg-user] LFE channel + 10dB References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> Message-ID: Wiebe Cazemier halfgaar.net> writes: > I did give the sample. It's a 6 channel wave file. Will that do? > > http://www.halfgaar.net/media/chan-id.zip > > Only this sample goes wrong. Films play normally. The sample plays correctly with MPlayer... > Xine plays the sample correctly. ... but incorrectly with xine. (Correct means: "Center" from left surround.) Carl Eugen From nicolas.george at normalesup.org Sun Apr 14 15:01:43 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 14 Apr 2013 15:01:43 +0200 Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> Message-ID: <20130414130143.GA11360@phare.normalesup.org> Le quintidi 25 germinal, an CCXXI, Wiebe Cazemier a ?crit?: > I did give the sample. It's a 6 channel wave file. Will that do? > > http://www.halfgaar.net/media/chan-id.zip > > Only this sample goes wrong. Films play normally. > > Xine plays the sample correctly. This sample is wrong, in the sense that the name of the channels that are spoken do not match the channel they are stored in. You should not use it as a base for your tests. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From wiebe at halfgaar.net Sun Apr 14 15:48:10 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Sun, 14 Apr 2013 15:48:10 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <20130414130143.GA11360@phare.normalesup.org> References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> <20130414130143.GA11360@phare.normalesup.org> Message-ID: <1173645536.20053.1365947289991.JavaMail.root@halfgaar.net> ----- Original Message ----- > From: "Nicolas George" > To: "FFmpeg user questions" > Sent: Sunday, 14 April, 2013 3:01:43 PM > Subject: Re: [FFmpeg-user] LFE channel + 10dB > > This sample is wrong, in the sense that the name of the channels that > are > spoken do not match the channel they are stored in. > > You should not use it as a base for your tests. > > Regards, Wrong? Xine and aplay play it correctly. The only confusing thing is that LFE plays all the words together as a mix, instead of saying LFE. From nicolas.george at normalesup.org Sun Apr 14 16:06:27 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 14 Apr 2013 16:06:27 +0200 Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <1173645536.20053.1365947289991.JavaMail.root@halfgaar.net> References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> <1935339432.19620.1365929773309.JavaMail.root@halfgaar.net> <673378601.19852.1365933840431.JavaMail.root@halfgaar.net> <1218228674.19901.1365940120226.JavaMail.root@halfgaar.net> <20130414130143.GA11360@phare.normalesup.org> <1173645536.20053.1365947289991.JavaMail.root@halfgaar.net> Message-ID: <20130414140627.GA26806@phare.normalesup.org> Le quintidi 25 germinal, an CCXXI, Wiebe Cazemier a ?crit?: > Wrong? Xine and aplay play it correctly. Yes, and that is a bug. For aplay, that is obvious: it does not have the code to convert the "FL FR FC LF RL RR" order in WAVE into the "FL FR RL RR FC LF" order used by the ALSA drivers, that is obvious from the source. I suspect the same applies to Xine. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From bostjan.strojan at gmail.com Sun Apr 14 16:44:01 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Sun, 14 Apr 2013 16:44:01 +0200 Subject: [FFmpeg-user] Matrox MQSink Filter In-Reply-To: References: Message-ID: On Fri, Apr 12, 2013 at 5:04 PM, Carl Eugen Hoyos wrote: > I opened ticket #2464, thank you for the sample! > (How long is the sample approximately supposed to play > and which application plays it?) As far as i know the only apps that are playing it are machines with dedicated matrox cards (Not sure about details either and i don't have such a machine). Not sure about the length either, but its longer than 440ms. (I have also tested windows xp with virtualdub and some soft matrox codecs, and that was a failure as well, directshow players did not work either) Thanks, Bo?tjan From cehoyos at ag.or.at Sun Apr 14 18:39:28 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 14 Apr 2013 16:39:28 +0000 (UTC) Subject: [FFmpeg-user] Matrox MQSink Filter References: Message-ID: Bo?tjan Strojan gmail.com> writes: > > I opened ticket #2464, thank you for the sample! > > (How long is the sample approximately supposed to play > > and which application plays it?) > > As far as i know the only apps that are playing it are machines with > dedicated matrox cards (Not sure about details either and i don't have > such a machine). So you actually don't know if the file is broken? =-( Carl Eugen From bostjan.strojan at gmail.com Sun Apr 14 20:40:33 2013 From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=) Date: Sun, 14 Apr 2013 20:40:33 +0200 Subject: [FFmpeg-user] Matrox MQSink Filter In-Reply-To: References: Message-ID: The file is not broken. b. On Sun, Apr 14, 2013 at 6:39 PM, Carl Eugen Hoyos wrote: > Bo?tjan Strojan gmail.com> writes: > >> > I opened ticket #2464, thank you for the sample! >> > (How long is the sample approximately supposed to play >> > and which application plays it?) >> >> As far as i know the only apps that are playing it are machines with >> dedicated matrox cards (Not sure about details either and i don't have >> such a machine). > > So you actually don't know if the file is broken? > =-( > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From subscriptions at znet.ca Sun Apr 14 22:27:00 2013 From: subscriptions at znet.ca (Subscriptions) Date: Sun, 14 Apr 2013 16:27:00 -0400 Subject: [FFmpeg-user] =?utf-8?q?ATSC_TS_live_transcode_on_DL360_G4=3F?= In-Reply-To: <4d4f41dac925a684ced5ba347d9614dd@znet.ca> References: <4d4f41dac925a684ced5ba347d9614dd@znet.ca> Message-ID: <979712125c1dd0a1e95f895cdbc99eb9@znet.ca> Hello again, This is to conclude this topic and share information which may be useful for others: Live transcode of one mpeg2 TS coming at ~17Mb/s to h264 at ~5Mb/s is now working on DL360 G4 server with 2 double core Intel Xeon @ 3.4 GHz (4 cores in total). Needs **LATEST** lib264 (x264-snapshot-20130331-2245). It looks there were performance enhancements to this library recently. A DL360 G5 with 2 quad core Intel(R) Xeon(R) CPU E5440 @ 2.83GHz (8 cores in total) is able to transcode 4 transponders coming at this speed with the same x264 library. Channels for transcode are captured via Hdhomerun box over-the-air. Thanks, and hope this is helpful - George. Le 2013-02-01 09:20, Subscriptions a ?crit?: > Hello again, > > It simply looks that 2 double-core Xeon at 3.4Ghz (HP DL 360) can not > transcode mpeg2 ts live stream coming at ~20 Mb/s to h264. Even with > 'preset ultrafast' the load goes above 4 and the buffer overruns. The > minimal processor that is able to do this up to now is a single > Phenom > 2 X6 Black at 3.4 Ghz, and it sometimes gets with load ~5 on it's six > cores. > > Thanks, George > > > Hello, > > I want to reduce bit rate of ~18Mb/s ATSC live stream which I capture > over the air with HDHomerun box, and it looks that an HP DL360 G4 > server (two 64 bit double core Xeon processors) can not do the job. > Is > there some way to make this functional? My idea is to reduce the > stream from 18Mb/s to about 4-5Mb/s at hd720. The problem is that > processor load goes over 4 and starts to overrun the buffer. Is there > a way to make the transcoding somehow work with x264 on that > computer? > > Thanks, George. > > nice -n -20 ffmpeg -i > "udp://239.255.254.1:1234?fifo_size=6000000&overrun_nonfatal=1&" > -acodec copy -c:v libx264 -crf 22 -preset fast -f mpegts > udp://239.255.254.240:1234 > > ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers > built on Oct 16 2012 09:46:44 with gcc 4.7 (Debian 4.7.2-4) > > Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), > yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 15000 kb/s, 63.81 fps, 59.94 > tbr, 90k tbn, 119.88 tbc > > using cpu capabilities: MMX2 SSE2 SSE3 Cache64 > Output #0, mpegts, to 'udp://239.255.254.240:1234': > Metadata: > encoder : Lavf54.29.104 > Stream #0:0: Video: h264, yuv420p, 1280x720 [SAR 1:1 DAR 16:9], > q=-1--1, 90k tbn, 59.94 tbc > Stream #0:1(eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, > 5.1(side), 384 kb/s > > > Tasks: 82 total, 2 running, 80 sleeping, 0 stopped, 0 zombie > %Cpu0 : 96.3 us, 1.0 sy, 0.0 ni, 0.3 id, 0.0 wa, 0.0 hi, 2.3 > si, 0.0 st > %Cpu1 : 95.7 us, 1.0 sy, 0.0 ni, 1.0 id, 0.0 wa, 0.0 hi, 2.3 > si, 0.0 st > %Cpu2 : 97.7 us, 1.3 sy, 0.0 ni, 0.3 id, 0.0 wa, 0.0 hi, 0.7 > si, 0.0 st > %Cpu3 : 96.0 us, 0.7 sy, 0.0 ni, 2.0 id, 0.0 wa, 0.0 hi, 1.3 > si, 0.0 st > KiB Mem: 3995312 total, 2685548 used, 1309764 free, 74516 > buffers > > processor : 3 > vendor_id : GenuineIntel > cpu family : 15 > model : 4 > model name : Intel(R) Xeon(TM) CPU 3.40GHz > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lsoe at alumni.dtu.dk Sun Apr 14 17:04:45 2013 From: lsoe at alumni.dtu.dk (=?ISO-8859-1?Q?Lars_S=F8rensen?=) Date: Sun, 14 Apr 2013 17:04:45 +0200 Subject: [FFmpeg-user] Installing FFmpeg on a Zentyal 3 server Message-ID: Hi I followed the excellent guide on theWiki UbuntuCompilationGuide but had a few problems: X11, Xlib, Xext, XShm and Xfixes were not found - I checked config.log This can be solved by installing libx11-dev, libxv-dev and libfixes-dev (sudo apt-get install libxv-dev etc.) Riteman From webb.jordan.zh at 163.com Mon Apr 15 04:20:44 2013 From: webb.jordan.zh at 163.com (webb.jordan.zh) Date: Mon, 15 Apr 2013 10:20:44 +0800 (CST) Subject: [FFmpeg-user] how to define the GOP frame types and GOD display order Message-ID: <15a53b84.fb.13e0b7e9069.Coremail.webb.jordan.zh@163.com> Hello I want to convert dv-avi format video to mp4 format video with h264 encoding and the the GOP display order and GOP frame types must be: GOP Display Order: 3 2 0 1 7 6 4 5 11 10 8 9 15 14 12 13 19 18 16 17 23 22 20 21 27 26 24 25 GOP Frame Types: IBBBPBBBPBBBPBBBPBBBPBBBPBBB I using command line for this: "..\ffmpeg.exe" -i "D:\Video\03000.avi" -vcodec libx264 -pix_fmt yuv420p -r 59.94 -aspect 16:9 -s 1280x720 -b:v 8000k -level 4 -refs 4 -preset faster -x264opts keyint=28:bframes=3:b_pyramid=none:open_gop=1 -mbd 2 -an -y "D:\xxTest\DV-MP4HD.mp4" But the GOP Frame Types of the converted video is : IBPBPBPBPBPBPBPBPBPBPBPBPBPB What wrong of the command line? I need any helps from anyones, Thank you very much! From Guillaume.Chanel at unige.ch Mon Apr 15 10:17:50 2013 From: Guillaume.Chanel at unige.ch (Guillaume Chanel) Date: Mon, 15 Apr 2013 08:17:50 +0000 Subject: [FFmpeg-user] Problem of PTS for video segmentation In-Reply-To: References: Message-ID: Thanks Carl for getting interest in my question, I am not very sure I can compile a version of ffmpeg for Ubuntu easily (as I understood several conflicts can occurs with libav ?). This is why I use the the recommended PPA here: https://launchpad.net/~jon-severinsson/+archive/ffmpeg If you have any advice on this point (like don't be worried and just go for the compilation) I will take it with pleasure. Anyway, I moved to windows and installed the pre-compiled version N-52045-g694fa00 which is from 12 April 2013. The result is almost exactly the same. The only difference it in the showinfo filter which now displays all the frames information, starting from the beginning of the file. Before it was starting at the time indicated by the -ss option. But the segmentation remains the same and the time stamps obtained by ffprobe also. Using the "-copytb 1" option did not change anything either. While waiting for your answer I will try the segmentation filter proposed as an answer to another post this past week-end. Keep me aware, Guillaume From jacobhameiri at gmail.com Mon Apr 15 08:49:22 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Sun, 14 Apr 2013 23:49:22 -0700 (PDT) Subject: [FFmpeg-user] bmp images to stream Message-ID: <1366008562471-4658482.post@n4.nabble.com> Hi, I have a process that generates bmp images and I need to create a video stream from these images, this is the command I am trying but it seems that ffmpeg is just stuck and his memory consumption is growing fast, what is the correct command I should use ? ffmpeg started on 2013-04-15 at 09:42:52 Report written to "ffmpeg-20130415-094252.log" Command line: /bin/ffmpeg -an -f image2pipe -vcodec bmp -s 1280x800 -pix_fmt argb -r 8 -i - -vcodec libx264 -preset ultrafast -tune zerolatency -r 8 -bsf:v h264_mp4toannexb -b 600000 -f mpegts udp://224.1.1.1:1234 -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [image2pipe @ 0x2bca620] File position before avformat_find_stream_info() is 0 -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bmp-images-to-stream-tp4658482.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jacobhameiri at gmail.com Mon Apr 15 09:35:32 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Mon, 15 Apr 2013 00:35:32 -0700 (PDT) Subject: [FFmpeg-user] bmp images to stream In-Reply-To: <1366008562471-4658482.post@n4.nabble.com> References: <1366008562471-4658482.post@n4.nabble.com> Message-ID: <1366011332588-4658483.post@n4.nabble.com> I tried rawvideo codec and this is the output: ffmpeg started on 2013-04-15 at 10:23:24 Report written to "ffmpeg-20130415-102324.log" Command line: /bin/ffmpeg -rtbufsize 4096000 -an -f image2pipe -vcodec rawvideo -bufsize 14336k -s 1280x800 -pix_fmt rgb32 -r 8 -i - -vcodec libx264 /sdcard/viddd44.avi -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [image2pipe @ 0x171b6d0] File position before avformat_find_stream_info() is 0 [image2pipe @ 0x171b6d0] parser not found for codec rawvideo, packets or times may be invalid. [image2pipe @ 0x171b6d0] parser not found for codec rawvideo, packets or times may be invalid. [image2pipe @ 0x171b6d0] max_analyze_duration 5000000 reached at 5000000 [image2pipe @ 0x171b6d0] Estimating duration from bitrate, this may be inaccurate [image2pipe @ 0x171b6d0] File position after avformat_find_stream_info() is 176128 Input #0, image2pipe, from 'pipe:': Duration: N/A, bitrate: N/A Stream #0:0, 42, 1/8: Video: rawvideo (BGRA / 0x41524742), bgra, 1280x800, 8 fps, 8 tbr, 8 tbn, 8 tbc [buffer @ 0x17159d0] Setting entry with key 'video_size' to value '1280x800' [buffer @ 0x17159d0] Setting entry with key 'pix_fmt' to value '30' [buffer @ 0x17159d0] Setting entry with key 'time_base' to value '1/8' [buffer @ 0x17159d0] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x17159d0] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x17159d0] Setting entry with key 'frame_rate' to value '8/1' [graph 0 input from stream 0:0 @ 0x1715990] w:1280 h:800 pixfmt:bgra tb:1/8 fr:8/1 sar:0/1 sws_param:flags=2 [format @ 0x17160a0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format' [auto-inserted scaler 0 @ 0x17186c0] picking yuv420p out of 4 ref:bgra alpha:1 [graph 0 input from stream 0:0 @ 0x1715990] TB:0.125000 FRAME_RATE:8.000000 SAMPLE_RATE:NaN [auto-inserted scaler 0 @ 0x17186c0] w:1280 h:800 fmt:bgra sar:0/1 -> w:1280 h:800 fmt:yuv420p sar:0/1 flags:0x4 [libx264 @ 0x1714e60] using cpu capabilities: ARMv6 NEON [libx264 @ 0x1714e60] profile High, level 3.2 Output #0, avi, to '/sdcard/viddd44.avi': Metadata: ISFT : Lavf54.29.104 Stream #0:0, 0, 1/8: Video: h264 (H264 / 0x34363248), yuv420p, 1280x800, q=-1--1, 8 tbn, 8 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument [rawvideo @ 0x172a050] Invalid buffer size, packet size 4096 < expected length 4096000 Error while decoding stream #0:0: Invalid argument -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bmp-images-to-stream-tp4658482p4658483.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jpolblanc at free.fr Mon Apr 15 09:40:00 2013 From: jpolblanc at free.fr (Jean-Paul BLANC) Date: Mon, 15 Apr 2013 09:40:00 +0200 Subject: [FFmpeg-user] ffmpeg 1.2 for Mac OS X 10.5.8 (Leopard) Message-ID: @me at renecalles Thanks for your answer. OK I'll try to compile but I'm afraid my computer is too old and my disk too filled to do that. JiPaul. From onemda at gmail.com Mon Apr 15 10:38:49 2013 From: onemda at gmail.com (Paul B Mahol) Date: Mon, 15 Apr 2013 08:38:49 +0000 Subject: [FFmpeg-user] bmp images to stream In-Reply-To: <1366008562471-4658482.post@n4.nabble.com> References: <1366008562471-4658482.post@n4.nabble.com> Message-ID: On 4/15/13, Jacobs wrote: > Hi, I have a process that generates bmp images and I need to create a video > stream from these images, this is the command I am trying but it seems that > ffmpeg is just stuck and his memory consumption is growing fast, what is > the > correct command I should use ? What creates such bmp images? Could you upload sample of one of them? > > ffmpeg started on 2013-04-15 at 09:42:52 > Report written to "ffmpeg-20130415-094252.log" > Command line: > /bin/ffmpeg -an -f image2pipe -vcodec bmp -s 1280x800 -pix_fmt argb -r 8 -i > - -vcodec libx264 -preset ultrafast -tune zerolatency -r 8 -bsf:v > h264_mp4toannexb -b 600000 -f mpegts udp://224.1.1.1:1234 -report > ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers > built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 > (prerelease) > configuration: --arch=arm --cpu=cortex-a8 --target-os=linux > --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static > --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- > --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm > --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' > --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc > --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver > --enable-pthreads --enable-libx264 --disable-fast-unaligned > libavutil 51. 73.101 / 51. 73.101 > libavcodec 54. 59.100 / 54. 59.100 > libavformat 54. 29.104 / 54. 29.104 > libavdevice 54. 2.101 / 54. 2.101 > libavfilter 3. 17.100 / 3. 17.100 > libswscale 2. 1.101 / 2. 1.101 > libswresample 0. 15.100 / 0. 15.100 > libpostproc 52. 0.100 / 52. 0.100 > [image2pipe @ 0x2bca620] File position before avformat_find_stream_info() > is > 0 > > > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/bmp-images-to-stream-tp4658482.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ticks.sunny84 at gmail.com Mon Apr 15 13:57:07 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Mon, 15 Apr 2013 17:27:07 +0530 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: Hi all, Is there any one woh is able to do this in c code. If yes please share the settings that can activate baseline video creation in ffmpeg. As right now i am not able to activate the baseline profile through c code. I am getting close enough to conclude that this can be a bug in ffmeg code. On Wed, Apr 10, 2013 at 6:53 PM, Susheel Tickoo wrote: > Hi All > I have been trying to create a video using ffmpeg from images on android > device. > > The settings of my H264 codec are as below: > c->bit_rate = 500000; > c->width = 480; > c->height = 320; > c->time_base= (AVRational){1,0}; > c->gop_size = 12; // emit one intra frame every ten frames > c->max_b_frames=1; > c->pix_fmt = PIX_FMT_YUV420P; > c->codec_type = AVMEDIA_TYPE_VIDEO; > c->codec_id = codec->id; > c->max_b_frames = 0; > c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; > c->me_range = 16; > c->max_qdiff = 4; > c->qmin = 10; > c->qmax = 26; > c->qcompress = 0.6; > c->trellis=0; > c->level = 13; > c->refs = 5; > c->coder_type = 0; > c->scenechange_threshold = 0; > //new > c->flags|=CODEC_FLAG_LOOP_FILTER;//new > //c->profile=FF_PROFILE_H264_BASELINE; //Baseline //new > c->scenechange_threshold = 40; //new > c-> rc_buffer_size = 0; > c->gop_size=250; //new > c->max_b_frames=1;//new > c->me_method=7; > c->me_cmp|= 1; > c->me_subpel_quality = 6; > c->qmax=51; > c->keyint_min=25; > av_opt_set(c->priv_data,"subq","6",0); > av_opt_set(c->priv_data,"crf","20.0",0); > av_opt_set(c->priv_data,"weighted_p_pred","0",0); > av_opt_set(c->priv_data,"vprofile","baseline",0); > av_opt_set(c->priv_data,"preset","medium",0); > av_opt_set(c->priv_data,"tune","zerolatency",0); > //av_opt_set(c->priv_data,"x264opts","rc-lookahead=0",0); > > > I am not able to understand that if this is a bug in ffmpeg or my code > that even after providing "* > av_opt_set(c->priv_data,"vprofile","baseline",0);*" and "* > av_opt_set(c->priv_data,"profile","baseline",0);*" > as was told in the following ticket: > http://ffmpeg.org/trac/ffmpeg/ticket/387 > > but nothing seems to activate the baseline profile. as when i do a ffmpef > -i on my video it shows profile as hight. > > please help. > regards, > Susheel Tickoo > > > > From lulebo at gmail.com Mon Apr 15 16:04:37 2013 From: lulebo at gmail.com (Carl Lindqvist) Date: Mon, 15 Apr 2013 16:04:37 +0200 Subject: [FFmpeg-user] baseline profile not getting activated in ffmpeg In-Reply-To: References: Message-ID: I have not been able to check how I did it, and I don't know it from the top of my head. But I think you will get more help in the libavusers list which is more oriented towards using the libs in code. /Carl 2013/4/15 Susheel Tickoo : > Hi all, > > Is there any one woh is able to do this in c code. If yes please share the > settings that can activate baseline video creation in ffmpeg. > As right now i am not able to activate the baseline profile through c code. > I am getting close enough to conclude that this can be a bug in ffmeg code. > On Wed, Apr 10, 2013 at 6:53 PM, Susheel Tickoo wrote: > >> Hi All >> I have been trying to create a video using ffmpeg from images on android >> device. >> >> The settings of my H264 codec are as below: >> c->bit_rate = 500000; >> c->width = 480; >> c->height = 320; >> c->time_base= (AVRational){1,0}; >> c->gop_size = 12; // emit one intra frame every ten frames >> c->max_b_frames=1; >> c->pix_fmt = PIX_FMT_YUV420P; >> c->codec_type = AVMEDIA_TYPE_VIDEO; >> c->codec_id = codec->id; >> c->max_b_frames = 0; >> c->profile = FF_PROFILE_H264_CONSTRAINED_BASELINE; >> c->me_range = 16; >> c->max_qdiff = 4; >> c->qmin = 10; >> c->qmax = 26; >> c->qcompress = 0.6; >> c->trellis=0; >> c->level = 13; >> c->refs = 5; >> c->coder_type = 0; >> c->scenechange_threshold = 0; >> //new >> c->flags|=CODEC_FLAG_LOOP_FILTER;//new >> //c->profile=FF_PROFILE_H264_BASELINE; //Baseline //new >> c->scenechange_threshold = 40; //new >> c-> rc_buffer_size = 0; >> c->gop_size=250; //new >> c->max_b_frames=1;//new >> c->me_method=7; >> c->me_cmp|= 1; >> c->me_subpel_quality = 6; >> c->qmax=51; >> c->keyint_min=25; >> av_opt_set(c->priv_data,"subq","6",0); >> av_opt_set(c->priv_data,"crf","20.0",0); >> av_opt_set(c->priv_data,"weighted_p_pred","0",0); >> av_opt_set(c->priv_data,"vprofile","baseline",0); >> av_opt_set(c->priv_data,"preset","medium",0); >> av_opt_set(c->priv_data,"tune","zerolatency",0); >> //av_opt_set(c->priv_data,"x264opts","rc-lookahead=0",0); >> >> >> I am not able to understand that if this is a bug in ffmpeg or my code >> that even after providing "* >> av_opt_set(c->priv_data,"vprofile","baseline",0);*" and "* >> av_opt_set(c->priv_data,"profile","baseline",0);*" >> as was told in the following ticket: >> http://ffmpeg.org/trac/ffmpeg/ticket/387 >> >> but nothing seems to activate the baseline profile. as when i do a ffmpef >> -i on my video it shows profile as hight. >> >> please help. >> regards, >> Susheel Tickoo >> >> >> >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jacobhameiri at gmail.com Mon Apr 15 10:40:48 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Mon, 15 Apr 2013 01:40:48 -0700 (PDT) Subject: [FFmpeg-user] bmp images to stream In-Reply-To: References: <1366008562471-4658482.post@n4.nabble.com> Message-ID: <1366015248255-4658487.post@n4.nabble.com> thanks for you response, I changed to -f rawvideo instead of image2pipe and it works now -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bmp-images-to-stream-tp4658482p4658487.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jacobhameiri at gmail.com Mon Apr 15 11:43:11 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Mon, 15 Apr 2013 02:43:11 -0700 (PDT) Subject: [FFmpeg-user] reduce laltency when streaming udp multicast Message-ID: <1366018991831-4658488.post@n4.nabble.com> I am streaming a raw rgb live video via multicast using mpegts, currently I have a 1-2 seconds delay what can I do to reduce this delay ? (I am willing to lower the quality to gain less delay time) this is my command and output: ffmpeg started on 2013-04-15 at 12:37:59 Report written to "ffmpeg-20130415-123759.log" Command line: /bin/ffmpeg -an -f rawvideo -vcodec rawvideo -s 800x1280 -pix_fmt rgba -r 9 -i - -vcodec mpeg4 -preset ultrafast -tune zerolatency -r 9 -filter:v "scale=400:-1" -sws_flags fast_bilinear -b 200000 -f mpegts udp://224.1.1.1:1234 -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [rawvideo @ 0x2ac0620] File position before avformat_find_stream_info() is 0 [rawvideo @ 0x2ac0620] All info found [rawvideo @ 0x2ac0620] Estimating duration from bitrate, this may be inaccurate [rawvideo @ 0x2ac0620] File position after avformat_find_stream_info() is 4096000 Input #0, rawvideo, from 'pipe:': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 1, 1/9: Video: rawvideo (RGBA / 0x41424752), rgba, 800x1280, 9 tbr, 9 tbn, 9 tbc Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x2acd6c0] Setting entry with key 'video_size' to value '800x1280' [buffer @ 0x2acd6c0] Setting entry with key 'pix_fmt' to value '28' [buffer @ 0x2acd6c0] Setting entry with key 'time_base' to value '1/9' [buffer @ 0x2acd6c0] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x2acd6c0] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x2acd6c0] Setting entry with key 'frame_rate' to value '9/1' [graph 0 input from stream 0:0 @ 0x2acd670] w:800 h:1280 pixfmt:rgba tb:1/9 fr:9/1 sar:0/1 sws_param:flags=2 [graph 0 input from stream 0:0 @ 0x2acd670] TB:0.111111 FRAME_RATE:9.000000 SAMPLE_RATE:NaN [Parsed_scale_0 @ 0x2ac05e0] w:800 h:1280 fmt:rgba sar:0/1 -> w:400 h:640 fmt:yuv420p sar:0/1 flags:0x1 [mpeg4 @ 0x2ac9970] detected 4 logical cores [mpeg4 @ 0x2ac9970] intra_quant_bias = 0 inter_quant_bias = -64 [mpegts @ 0x2ac91f0] muxrate VBR, pcr every 1 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://224.1.1.1:1234': Metadata: encoder : Lavf54.29.104 Stream #0:0, 0, 1/90000: Video: mpeg4, yuv420p, 400x640, q=2-31, 200 kb/s, 90k tbn, 9 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mpeg4) frame= 6 fps=0.0 q=2.0 size= 52kB time=00:00:00.66 bitrate= 638.4kbits/s frame= 12 fps= 11 q=2.0 size= 54kB time=00:00:01.33 bitrate= 332.8kbits/s frame= 17 fps= 10 q=2.0 size= 112kB time=00:00:01.88 bitrate= 484.9kbits/s frame= 23 fps= 10 q=2.0 size= 114kB time=00:00:02.55 bitrate= 365.5kbits/s frame= 28 fps= 10 q=2.0 size= 172kB time=00:00:03.11 bitrate= 452.5kbits/s frame= 34 fps= 10 q=2.0 size= 174kB time=00:00:03.77 bitrate= 377.4kbits/s frame= 39 fps= 10 q=2.0 size= 232kB time=00:00:04.33 bitrate= 438.0kbits/s frame= 45 fps= 10 q=2.0 size= 234kB time=00:00:05.00 bitrate= 383.5kbits/s frame= 50 fps=9.9 q=2.0 size= 294kB time=00:00:05.55 bitrate= 434.2kbits/s frame= 55 fps=9.7 q=2.0 size= 304kB time=00:00:06.11 bitrate= 408.0kbits/s frame= 59 fps=9.3 q=2.0 size= 307kB time=00:00:06.55 bitrate= 383.1kbits/s frame= 63 fps=9.1 q=2.0 size= 352kB time=00:00:07.00 bitrate= 412.5kbits/s frame= 67 fps=8.9 q=2.0 size= 360kB time=00:00:07.44 bitrate= 396.6kbits/s frame= 71 fps=8.8 q=2.0 size= 366kB time=00:00:07.88 bitrate= 380.2kbits/s frame= 76 fps=8.8 q=2.0 size= 411kB time=00:00:08.44 bitrate= 398.2kbits/s frame= 81 fps=8.8 q=2.0 size= 417kB time=00:00:09.00 bitrate= 379.3kbits/s frame= 86 fps=8.8 q=2.0 size= 477kB time=00:00:09.55 bitrate= 408.6kbits/s frame= 91 fps=8.8 q=2.0 size= 488kB time=00:00:10.11 bitrate= 395.1kbits/s frame= 96 fps=8.8 q=2.0 size= 500kB time=00:00:10.66 bitrate= 383.8kbits/s frame= 101 fps=8.8 q=2.1 size= 554kB time=00:00:11.22 bitrate= 404.2kbits/s frame= 106 fps=8.8 q=2.0 size= 563kB time=00:00:11.77 bitrate= 391.5kbits/s frame= 110 fps=8.7 q=2.3 size= 615kB time=00:00:12.22 bitrate= 412.0kbits/s frame= 115 fps=8.7 q=6.6 size= 646kB time=00:00:12.77 bitrate= 414.3kbits/s frame= 120 fps=8.7 q=5.2 size= 661kB time=00:00:13.33 bitrate= 406.2kbits/s frame= 125 fps=8.7 q=5.3 size= 704kB time=00:00:13.88 bitrate= 415.2kbits/s frame= 130 fps=8.7 q=6.0 size= 714kB time=00:00:14.44 bitrate= 405.0kbits/s frame= 135 fps=8.7 q=9.0 size= 754kB time=00:00:15.00 bitrate= 411.9kbits/s frame= 140 fps=8.7 q=11.1 size= 769kB time=00:00:15.55 bitrate= 404.8kbits/s frame= 145 fps=8.7 q=8.3 size= 801kB time=00:00:16.11 bitrate= 407.3kbits/s frame= 150 fps=8.7 q=11.5 size= 811kB time=00:00:16.66 bitrate= 398.7kbits/s frame= 155 fps=8.7 q=14.1 size= 823kB time=00:00:17.22 bitrate= 391.7kbits/s frame= 160 fps=8.7 q=16.9 size= 853kB time=00:00:17.77 bitrate= 393.1kbits/s frame= 164 fps=8.7 q=10.6 size= 857kB time=00:00:18.22 bitrate= 385.2kbits/s frame= 169 fps=8.7 q=7.6 size= 885kB time=00:00:18.77 bitrate= 385.9kbits/s frame= 174 fps=8.7 q=12.6 size= 892kB time=00:00:19.33 bitrate= 377.9kbits/s frame= 179 fps=8.7 q=12.9 size= 901kB time=00:00:19.88 bitrate= 371.3kbits/s frame= 184 fps=8.7 q=15.6 size= 931kB time=00:00:20.44 bitrate= 373.0kbits/s frame= 189 fps=8.7 q=16.9 size= 939kB time=00:00:21.00 bitrate= 366.5kbits/s frame= 194 fps=8.6 q=14.8 size= 963kB time=00:00:21.55 bitrate= 366.0kbits/s frame= 198 fps=8.6 q=18.5 size= 970kB time=00:00:22.00 bitrate= 361.3kbits/s frame= 203 fps=8.6 q=15.8 size= 978kB time=00:00:22.55 bitrate= 355.3kbits/s frame= 208 fps=8.6 q=18.0 size= 1004kB time=00:00:23.11 bitrate= 355.7kbits/s frame= 213 fps=8.6 q=19.3 size= 1013kB time=00:00:23.66 bitrate= 350.7kbits/s -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/reduce-laltency-when-streaming-udp-multicast-tp4658488.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jacobhameiri at gmail.com Mon Apr 15 12:09:58 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Mon, 15 Apr 2013 03:09:58 -0700 (PDT) Subject: [FFmpeg-user] reduce laltency when streaming udp multicast In-Reply-To: <1366018991831-4658488.post@n4.nabble.com> References: <1366018991831-4658488.post@n4.nabble.com> Message-ID: <1366020598074-4658489.post@n4.nabble.com> this is the ffprobe report: ffprobe started on 2013-04-15 at 13:08:15 Report written to "ffprobe-20130415-130815.log" Command line: "d:\\ffmpeg\\ffprobe.exe" udp://224.1.1.1:1234 -report ffprobe version N-46469-gc995644 Copyright (c) 2007-2012 the FFmpeg developers built on Nov 5 2012 17:58:15 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [mpegts @ 00000000020df460] Format mpegts probed with size=2048 and score=100 [mpegts @ 00000000020df460] Unable to seek back to the start [mpegts @ 00000000020df460] stream=0 stream_type=10 pid=100 prog_reg_desc= [mpegts @ 00000000020df460] File position before avformat_find_stream_info() is 9400 [mpeg4 @ 00000000020f0620] hmm, seems the headers are not complete, trying to guess time_increment_bits [mpeg4 @ 00000000020f0620] my guess is 4 bits ;) [mpeg4 @ 00000000020f0620] Error, header damaged or not MPEG4 header (qscale=0) [mpegts @ 00000000020df460] invalid dts/pts combination 35476000 [mpeg4 @ 00000000020f0620] looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35486000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35496000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35506000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35516000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35526000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35536000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35546000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35556000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35566000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] invalid dts/pts combination 35576000 [mpeg4 @ 00000000020f0620] [IMGUTILS @ 000000000022f420] Picture size 0x0 is invalid [mpeg4 @ 00000000020f0620] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f0620] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020df460] max_analyze_duration 5000000 reached at 5000000 [mpegts @ 00000000020df460] Estimating duration from bitrate, this may be inaccurate [mpegts @ 00000000020df460] File position after avformat_find_stream_info() is 208680 Input #0, mpegts, from 'udp://224.1.1.1:1234': Duration: N/A, start: 394.177778, bitrate: N/A Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100], 56, 1/90000: Video: mpeg4 (Simple Profile) ([16][0][0][0] / 0x0010), yuv420p, 400x640 [SAR 1:1 DAR 5:8], 9 fps, 9 tbr, 90k tbn, 9 tbc [mpeg4 @ 00000000020f0620] detected 8 logical cores [AVIOContext @ 00000000020f02a0] Statistics: 208680 bytes read, 0 seeks -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/reduce-laltency-when-streaming-udp-multicast-tp4658488p4658489.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ory at drilon.com Mon Apr 15 10:42:22 2013 From: ory at drilon.com (Ory Drilon) Date: Mon, 15 Apr 2013 16:42:22 +0800 Subject: [FFmpeg-user] Behavior of igndts Message-ID: Could I get a more detailed explanation of the igndts flag? The most detail I can dig up is the header comment that says it "Ignore[s] DTS on frames that contain both DTS & PTS" during decoding. What does ffmpeg do with these DTSes if the flag is not specified? I'm wondering if I can blindly enable this flag for every file I convert, since a few of my video input files result in output that's twice the duration of input! I'm guessing that erroneous DTSes from input files cause ffmpeg to double the amount of frames or the duration of each frame in the output, since enabling igndts fixes these bad conversions. From ddhung at vasc.com.vn Tue Apr 16 04:36:36 2013 From: ddhung at vasc.com.vn (=?utf-8?B?xJDhurduZyDEkOG7qWMgSMawbmc=?=) Date: Tue, 16 Apr 2013 09:36:36 +0700 Subject: [FFmpeg-user] Transcode multicast IP video over ffmpeg Message-ID: <000d01ce3a4b$3a8e5e10$afab1a30$@vasc.com.vn> Hi all. I?m using your ffmpeg to transcode Multicast multicast UDP IP video input with 3.5 Mbps and codec is x264 to output with the the same input codec but low bit rate about 1.5Mbps. When I have done I found the quality of image is very low, the image is often blured. I have read in the codec x264 there is a encoding mode which is pass 2 can improve the quality of the output image. Could you support for me the step by step to configure this pass 2 method. Here is the command line I am using: C:/ffmpeg_32/bin/ffmpeg.exe -i udp://233.19.204.1:5501 -filter:v yadif=0:-1:1,fps=25 -vcodec libx264 -g 16 -vprofile main -level 30 -b:v 1050k -minrate 1000k -maxrate 1100k -bufsize 100k -s: 720x576 -r 25 -ac 2 -ar 48000 -b:a 128k -filter:a volume=2 -f mpegts -muxrate 1300k udp://233.19.209.1:5501?pkt_size=1316 Thank you so much! Best Regards ??ng ??c H?ng | Broadcast Technical Department | IPTV Center | VASC Software and Media Company | | 33 Thai Ha Street | ??ng ?a District | Hanoi | Vietnam| | Tel (+84) 37722728 ext 681/684 | Mobile (+84) 915 070 921| | ddhung at vasc.com.vn | www.mytv.com.vn | From danghuy83 at gmail.com Tue Apr 16 07:22:20 2013 From: danghuy83 at gmail.com (viet ha) Date: Tue, 16 Apr 2013 12:22:20 +0700 Subject: [FFmpeg-user] Encode/decode h263-1996 in java Message-ID: <516CE00C.1010202@gmail.com> Hi all! I'm trying to use ffmpeg to encode/decode h263-1996 in java. But I don't how to do ( I have known sip_communicator using H264 but it didn't use h263-1996). Can anyone help me? Thank you so much! From palani at i-waves.com Tue Apr 16 07:52:34 2013 From: palani at i-waves.com (Palaniappan N) Date: Tue, 16 Apr 2013 11:22:34 +0530 Subject: [FFmpeg-user] FFMPEG | zlib support is required Message-ID: Hi, We are executing the following FFMPEG command to convert swf to mp4 and we get an i/o error ffmpeg -i /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf -vcodec mpeg4 -s 1024x768 -b 512k -acodec libfaac -strict experimental -ac 2 -ab 256k -ar 44100 /var/www/html/WorkFiles/output/swf.mp4 ffmpeg version N-49823-g7ff1a4b Copyright (c) 2000-2013 the FFmpeg developers built on Feb 12 2013 02:34:44 with gcc 4.4.6 (GCC) 20110731 (Red Hat 4.4.6-3) configuration: --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 libavutil 52. 17.101 / 52. 17.101 libavcodec 54. 91.102 / 54. 91.102 libavformat 54. 62.100 / 54. 62.100 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 37.101 / 3. 37.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [swf @ 0x2ae62c0] SWF compressed file detected [swf @ 0x2ae62c0] zlib support is required to read SWF compressed files /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf: Input/output error Can anyone suggest us what we have to do further? Thanks! Regards, Palani From asorti6 at gmail.com Tue Apr 16 08:20:53 2013 From: asorti6 at gmail.com (Owen Jones) Date: Tue, 16 Apr 2013 07:20:53 +0100 Subject: [FFmpeg-user] channelsplit seems to mute one channel In-Reply-To: References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> Message-ID: <9AE0B411-5B92-4E1C-A0C5-82B3529FCE45@gmail.com> On 13 Apr 2013, at 08:16, Carl Eugen Hoyos wrote: > The reason is that FFmpeg only marks the first > audio track as default. The reason is that > users typically don't like if all languages > are played simultaneously with QT;-) > (This was reported very often and only fixed > recently.) > It is not impossible to fix, but could you > explain your use-case? Is there any possibility of a fix or a way around this? I've also tried looking into the metadata to see if I can re-enable the second channel but can't see how to do that either. The only fix I can see right now is to post-process every output file with quicktime but this will be rather painful. If ffmpeg is making a decision to disable a channel without an override option that feels rather like like a bug or at least a missing feature? I guess I could have a go at trying to write the code for a new filter/option myself although I think it might take me a while to get up to speed on how to do this... Regards Owen From onemda at gmail.com Tue Apr 16 10:50:04 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 16 Apr 2013 08:50:04 +0000 Subject: [FFmpeg-user] FFMPEG | zlib support is required In-Reply-To: References: Message-ID: On 4/16/13, Palaniappan N wrote: > Hi, > > > We are executing the following FFMPEG command to convert swf to mp4 and we > get an i/o error > > ffmpeg -i /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf > -vcodec mpeg4 -s 1024x768 -b 512k -acodec libfaac -strict experimental -ac > 2 -ab 256k -ar 44100 /var/www/html/WorkFiles/output/swf.mp4 > ffmpeg version N-49823-g7ff1a4b Copyright (c) 2000-2013 the FFmpeg > developers > built on Feb 12 2013 02:34:44 with gcc 4.4.6 (GCC) 20110731 (Red Hat > 4.4.6-3) > configuration: --enable-gpl --enable-libmp3lame --enable-libvorbis > --enable-libvpx --enable-libx264 > libavutil 52. 17.101 / 52. 17.101 > libavcodec 54. 91.102 / 54. 91.102 > libavformat 54. 62.100 / 54. 62.100 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 37.101 / 3. 37.101 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > > [swf @ 0x2ae62c0] SWF compressed file detected > [swf @ 0x2ae62c0] zlib support is required to read SWF compressed files > /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf: Input/output > error > > Can anyone suggest us what we have to do further? Recompile with --enable-zlib enabled. > > > Thanks! > > Regards, > Palani > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Tue Apr 16 10:53:00 2013 From: onemda at gmail.com (Paul B Mahol) Date: Tue, 16 Apr 2013 08:53:00 +0000 Subject: [FFmpeg-user] channelsplit seems to mute one channel In-Reply-To: <9AE0B411-5B92-4E1C-A0C5-82B3529FCE45@gmail.com> References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> <9AE0B411-5B92-4E1C-A0C5-82B3529FCE45@gmail.com> Message-ID: On 4/16/13, Owen Jones wrote: > On 13 Apr 2013, at 08:16, Carl Eugen Hoyos wrote: > >> The reason is that FFmpeg only marks the first >> audio track as default. The reason is that >> users typically don't like if all languages >> are played simultaneously with QT;-) >> (This was reported very often and only fixed >> recently.) >> It is not impossible to fix, but could you >> explain your use-case? > > Is there any possibility of a fix or a way around this? I've also tried > looking into the metadata to see if I can re-enable the second channel but > can't see how to do that either. The only fix I can see right now is to > post-process every output file with quicktime but this will be rather > painful. > > If ffmpeg is making a decision to disable a channel without an override > option that feels rather like like a bug or at least a missing feature? I > guess I could have a go at trying to write the code for a new filter/option > myself although I think it might take me a while to get up to speed on how > to do this... I think you are confused a little. You have file where each single channel is in its own separate stream. There is no point in writing filter for this. Just read and follow documentation how to pick annother audio stream and not default one. > > Regards > Owen > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Tue Apr 16 11:07:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 09:07:29 +0000 (UTC) Subject: [FFmpeg-user] channelsplit seems to mute one channel References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> <9AE0B411-5B92-4E1C-A0C5-82B3529FCE45@gmail.com> Message-ID: Owen Jones gmail.com> writes: > On 13 Apr 2013, at 08:16, Carl Eugen Hoyos wrote: > > > The reason is that FFmpeg only marks the first > > audio track as default. > Is there any possibility of a fix or a way around this? You can either revert 981cf0c or search for the line "avio_wb24(pb, track->secondary ? 0x2 : 0xf);" in libavformat/movenc.c and make is always write "0xF" or - ideally - as a temporary workaround, you add a new flag to movenc.c to force all tracks as primary: See the "AVOption"s on top of the file and MOVMuxContext in libavformat/movenc.h (and don't forget to send the result to ffmpeg-devel). Carl Eugen From palani at i-waves.com Tue Apr 16 11:07:49 2013 From: palani at i-waves.com (Palaniappan N) Date: Tue, 16 Apr 2013 14:37:49 +0530 Subject: [FFmpeg-user] FFMPEG | zlib support is required In-Reply-To: References: Message-ID: On Tue, Apr 16, 2013 at 2:20 PM, Paul B Mahol wrote: > On 4/16/13, Palaniappan N wrote: > > Hi, > > > > > > We are executing the following FFMPEG command to convert swf to mp4 and > we > > get an i/o error > > > > ffmpeg -i /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf > > -vcodec mpeg4 -s 1024x768 -b 512k -acodec libfaac -strict experimental > -ac > > 2 -ab 256k -ar 44100 /var/www/html/WorkFiles/output/swf.mp4 > > ffmpeg version N-49823-g7ff1a4b Copyright (c) 2000-2013 the FFmpeg > > developers > > built on Feb 12 2013 02:34:44 with gcc 4.4.6 (GCC) 20110731 (Red Hat > > 4.4.6-3) > > configuration: --enable-gpl --enable-libmp3lame --enable-libvorbis > > --enable-libvpx --enable-libx264 > > libavutil 52. 17.101 / 52. 17.101 > > libavcodec 54. 91.102 / 54. 91.102 > > libavformat 54. 62.100 / 54. 62.100 > > libavdevice 54. 3.103 / 54. 3.103 > > libavfilter 3. 37.101 / 3. 37.101 > > libswscale 2. 2.100 / 2. 2.100 > > libswresample 0. 17.102 / 0. 17.102 > > libpostproc 52. 2.100 / 52. 2.100 > > > > [swf @ 0x2ae62c0] SWF compressed file detected > > [swf @ 0x2ae62c0] zlib support is required to read SWF compressed files > > /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf: > Input/output > > error > > > > Can anyone suggest us what we have to do further? > > Recompile with --enable-zlib enabled. > > > > > > Shouldn't I Install "zlib", before recompiling it along with FFMPEG? -Palani From cehoyos at ag.or.at Tue Apr 16 11:08:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 09:08:55 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?FFMPEG_=7C_zlib_support_is_required?= References: Message-ID: Palaniappan N i-waves.com> writes: > [swf 0x2ae62c0] SWF compressed file detected > [swf 0x2ae62c0] zlib support is required to read SWF compressed files Install zlib and recompile (it gets auto-detected). Note that only some swf files contain audio and video and can be decoded at all with FFmpeg, other contain "sprites" and "shapes", FFmpeg does not support them. Carl Eugen From cehoyos at ag.or.at Tue Apr 16 11:12:15 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 09:12:15 +0000 (UTC) Subject: [FFmpeg-user] Behavior of igndts References: Message-ID: Ory Drilon drilon.com> writes: > a few of my video input files result in output > that's twice the duration of input! Please provide such sample(s)! Carl Eugen From cehoyos at ag.or.at Tue Apr 16 11:10:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 09:10:55 +0000 (UTC) Subject: [FFmpeg-user] Transcode multicast IP video over ffmpeg References: <000d01ce3a4b$3a8e5e10$afab1a30$@vasc.com.vn> Message-ID: ??ng ??c H?ng vasc.com.vn> writes: > I have read in the codec x264 there is a encoding mode > which is pass 2 can improve the quality of the output > image. Could you support for me the step by step to > configure this pass 2 method. > > Here is the command line I am using: > > C:/ffmpeg_32/bin/ffmpeg.exe -i udp://233.19.204.1:5501 I may misunderstand but I don't think 2-pass methods can be used for udp input. Try different x264 speed presets to find the one that still is real-time but has best quality (and choose high profile if you interested in quality). Carl Eugen From cehoyos at ag.or.at Tue Apr 16 11:26:30 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 09:26:30 +0000 (UTC) Subject: [FFmpeg-user] reduce laltency when streaming udp multicast References: <1366018991831-4658488.post@n4.nabble.com> Message-ID: Jacobs gmail.com> writes: > I am streaming a raw rgb live video via multicast > using mpegts, currently I have a 1-2 seconds delay > what can I do to reduce this delay ? > /bin/ffmpeg -an -f rawvideo -vcodec rawvideo > -s 800x1280 -pix_fmt rgba -r 9 -i - It is possible that -probesize and -analyzeduration help for your problem (I don't know). > -vcodec mpeg4 -preset ultrafast -tune zerolatency ultrafast and zerolatency have no effect on mpeg4 encoding. Unrelated: Some of your configure flags are default flags and have no effect: --disable-shared --enable-static --enable-hwaccels --enable-ffserver --enable-pthreads That makes me wonder why you are using these options: --enable-version3 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16 Carl Eugen From ddhung at vasc.com.vn Tue Apr 16 11:58:55 2013 From: ddhung at vasc.com.vn (=?utf-8?B?xJDhurduZyDEkOG7qWMgSMawbmc=?=) Date: Tue, 16 Apr 2013 16:58:55 +0700 Subject: [FFmpeg-user] Transcode multicast IP video over ffmpeg In-Reply-To: References: <000d01ce3a4b$3a8e5e10$afab1a30$@vasc.com.vn> Message-ID: <002801ce3a89$05ea1300$11be3900$@vasc.com.vn> I am trying to use the presets for codec x264 but I still understand the method to chose the suitable video bit rate, profile, level for each preset. Could you give me some useful advices for that. Thank you so much Best Regards ??ng ??c H?ng | Broadcast Technical Department | IPTV Center | VASC Software and Media Company | | 33 Thai Ha Street | ??ng ?a District | Hanoi | Vietnam| | Tel (+84) 37722728 ext 681/684 | Mobile (+84) 915 070 921| | ddhung at vasc.com.vn | www.mytv.com.vn | -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: Tuesday, April 16, 2013 4:11 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Transcode multicast IP video over ffmpeg ??ng ??c H?ng vasc.com.vn> writes: > I have read in the codec x264 there is a encoding mode which is pass 2 > can improve the quality of the output image. Could you support for me > the step by step to configure this pass 2 method. > > Here is the command line I am using: > > C:/ffmpeg_32/bin/ffmpeg.exe -i udp://233.19.204.1:5501 I may misunderstand but I don't think 2-pass methods can be used for udp input. Try different x264 speed presets to find the one that still is real-time but has best quality (and choose high profile if you interested in quality). Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ory at drilon.com Tue Apr 16 12:57:25 2013 From: ory at drilon.com (Ory Drilon) Date: Tue, 16 Apr 2013 18:57:25 +0800 Subject: [FFmpeg-user] Behavior of igndts Message-ID: Uploaded a chopped down file to http://www1.datafilehost.com/d/01827038since I'm behind a proxy that won't let me go on ftp. The following ffmpeg call produces a 46 second output. The original file actually runs for 27 seconds. ffmpeg -y -loglevel warning -i doubling_dts -map 0:0 -map 0:1 -vf scale=292:240,fps=fps\=25 -ac 2 -vcodec libx264 -acodec libvo_aacenc -scodec mov_text -b:v 150k -b:a 16k -f mp4 without_igndts.mp4 Adding +igndts like below produces a video of the correct length and size. Note that the filesize of the erroneous output above is larger. ffmpeg -fflags +igndts -y -loglevel warning -i doubling_dts -map 0:0 -map 0:1 -vf scale=292:240,fps=fps\=25 -ac 2 -vcodec libx264 -acodec libvo_aacenc -scodec mov_text -b:v 150k -b:a 16k -f mp4 with_igndts.mp4 In the meantime, are there any consequences to simply adding +igndts for the rest of files I'm converting even if most of them don't have this issue? From cehoyos at ag.or.at Tue Apr 16 13:27:57 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 11:27:57 +0000 (UTC) Subject: [FFmpeg-user] Behavior of igndts References: Message-ID: Ory Drilon drilon.com> writes: > Uploaded a chopped down file to > http://www1.datafilehost.com/d/01827038 Thank you! Michael seems to assume that these kind of files are broken (I cannot say): http://thread.gmane.org/gmane.comp.video.ffmpeg.issues/4952 Without doing too much investigation, I would guess that without -flags +igndts, the dts is used to determine the dts of the output stream. Carl Eugen From ory at drilon.com Tue Apr 16 14:11:26 2013 From: ory at drilon.com (Ory Drilon) Date: Tue, 16 Apr 2013 20:11:26 +0800 Subject: [FFmpeg-user] Behavior of igndts In-Reply-To: References: Message-ID: >Michael seems to assume that these kind of files are broken (I cannot say) Yeah, the DTSes in these files are probably broken which is why igndts fixes it. What I'm wondering is if I can just enable igndts even on files that aren't broken, just to be safe. Actually, I find it weird that ffmpeg would use the input stream's DTSes as a basis for the output DTSes. Wouldn't the output DTSes be vastly different if you were transcoding from one codec to another? My best guess is that the input DTSes are used for some sort of performance optimization that's safe for most modern streams. It's just occasionally failing in my case because I'm working with some really old files. From 23liam at googlemail.com Tue Apr 16 16:20:29 2013 From: 23liam at googlemail.com (Liam Condron-Farnos) Date: Tue, 16 Apr 2013 15:20:29 +0100 Subject: [FFmpeg-user] Transcode multicast IP video over ffmpeg In-Reply-To: <002801ce3a89$05ea1300$11be3900$@vasc.com.vn> References: <000d01ce3a4b$3a8e5e10$afab1a30$@vasc.com.vn> <002801ce3a89$05ea1300$11be3900$@vasc.com.vn> Message-ID: >I am trying to use the presets for codec x264 but I still understand the method to chose the suitable video bit rate, profile, level for each preset. Could >you give me some useful advices for that. Read this: http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide From hyperhacker at gmail.com Wed Apr 17 00:18:45 2013 From: hyperhacker at gmail.com (Rena) Date: Tue, 16 Apr 2013 18:18:45 -0400 Subject: [FFmpeg-user] FLV streaming, severe audio sync issues Message-ID: I'm trying to capture from my screen to stream to twitch.tv RTMP server, but the audio is always going way out of sync. After one minute of recording the audio is 4 seconds behind. I've tried everything I can find and nothing is helping. Updating to a recent ffmpeg build only seems to have made it worse. For testing I have it recording to a local file instead of to RTMP server: $ ffmpeg -f pulse -sample_rate 22050 -i alsa_output.pci-0000_00_14.2.analog-stereo.monitor -f x11grab -show_region 1 -s 850x480 -r 12 -i :0.0+3160,200 -vcodec libx264 -s 850x480 -b:v 384k -b:a 64k -threads 0 -g 3 -f flv out.flv ffmpeg version N-51651-g1fabd95 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 6 2013 21:44:28 with gcc 4.7 (Ubuntu/Linaro 4.7.2-2ubuntu1) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-avresample --enable-avisynth --enable-libass --enable-libfreetype --enable-libmp3lame --enable-libopencv --enable-libpulse --enable-librtmp --enable-libtheora --enable-libtwolame --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-openssl --enable-x11grab libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.100 / 3. 49.100 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, pulse, from 'alsa_output.pci-0000_00_14.2.analog-stereo.monitor': Duration: N/A, start: 0.000000, bitrate: 705 kb/s Stream #0:0: Audio: pcm_s16le, 22050 Hz, stereo, s16, 705 kb/s [x11grab @ 0x25b84a0] device: :0.0+3160,200 -> display: :0.0 x: 3160 y: 200 width: 850 height: 480 [x11grab @ 0x25b84a0] shared memory extension found Input #1, x11grab, from ':0.0+3160,200': Duration: N/A, start: 1366150061.077213, bitrate: 156672 kb/s Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 850x480, 156672 kb/s, 12 tbr, 1000k tbn, 12 tbc No pixel format specified, yuv444p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players. [libx264 @ 0x25d0180] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX XOP FMA4 SSEMisalign LZCNT [libx264 @ 0x25d0180] profile High 4:4:4 Predictive, level 2.2, 4:4:4 8-bit [libx264 @ 0x25d0180] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=2 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=3 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=3 rc=abr mbtree=1 bitrate=384 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'out.flv': Metadata: encoder : Lavf55.1.100 Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv444p, 850x480, q=-1--1, 384 kb/s, 1k tbn, 12 tbc Stream #0:1: Audio: mp3 ([2][0][0][0] / 0x0002), 22050 Hz, stereo, s16p, 64 kb/s Stream mapping: Stream #1:0 -> #0:0 (rawvideo -> libx264) Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help [swscaler @ 0x2592320] Warning: data is not aligned! This can lead to a speedloss frame= 1269 fps= 12 q=32766.0 Lsize= 6223kB time=00:01:50.54 bitrate= 461.2kbits/s video:5267kB audio:864kB subtitle:0 global headers:0kB muxing overhead 1.488966% [libx264 @ 0x25d0180] frame I:424 Avg QP:38.55 size: 11698 [libx264 @ 0x25d0180] frame P:558 Avg QP:46.10 size: 622 [libx264 @ 0x25d0180] frame B:287 Avg QP:47.69 size: 300 [libx264 @ 0x25d0180] consecutive B-frames: 54.8% 45.2% 0.0% [libx264 @ 0x25d0180] mb I I16..4: 34.9% 47.8% 17.3% [libx264 @ 0x25d0180] mb P I16..4: 2.0% 2.7% 0.2% P16..4: 5.2% 0.7% 0.2% 0.0% 0.0% skip:88.9% [libx264 @ 0x25d0180] mb B I16..4: 0.1% 0.4% 0.0% B16..8: 6.4% 0.3% 0.0% direct: 0.2% skip:92.5% L0:39.6% L1:59.5% BI: 0.9% [libx264 @ 0x25d0180] final ratefactor: 39.24 [libx264 @ 0x25d0180] 8x8 transform intra:48.4% inter:80.1% [libx264 @ 0x25d0180] coded y,u,v intra: 28.2% 18.4% 18.5% inter: 0.4% 0.5% 0.5% [libx264 @ 0x25d0180] i16 v,h,dc,p: 46% 48% 4% 2% [libx264 @ 0x25d0180] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 26% 29% 4% 3% 3% 4% 2% 6% [libx264 @ 0x25d0180] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 31% 13% 4% 6% 4% 4% 3% 3% [libx264 @ 0x25d0180] Weighted P-Frames: Y:0.9% UV:0.9% [libx264 @ 0x25d0180] ref P L0: 94.7% 1.6% 3.7% [libx264 @ 0x25d0180] kb/s:407.99 The system has no trouble handling the load, but the resulting video always has severe audio lag. -- Sent from my Game Boy. From cehoyos at ag.or.at Wed Apr 17 00:44:29 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 16 Apr 2013 22:44:29 +0000 (UTC) Subject: [FFmpeg-user] FLV streaming, severe audio sync issues References: Message-ID: Rena gmail.com> writes: > Duration: N/A, start: 0.000000, bitrate: 705 kb/s > Duration: N/A, start: 1366150061.077213, bitrate: 156672 kb/s This unfortunately cannot work;-( You can try if the alsa input provides proper timestamps. Carl Eugen From hyperhacker at gmail.com Wed Apr 17 01:39:30 2013 From: hyperhacker at gmail.com (Rena) Date: Tue, 16 Apr 2013 19:39:30 -0400 Subject: [FFmpeg-user] FLV streaming, severe audio sync issues In-Reply-To: References: Message-ID: On Tue, Apr 16, 2013 at 6:44 PM, Carl Eugen Hoyos wrote: > Rena gmail.com> writes: > > > Duration: N/A, start: 0.000000, bitrate: 705 kb/s > > > Duration: N/A, start: 1366150061.077213, bitrate: 156672 kb/s > > This unfortunately cannot work;-( > You can try if the alsa input provides proper timestamps. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > It's a Pulse issue? Can it be fixed somehow, or do I need to report a bug against PulseAudio? I can try alsa, but that's somewhat limiting as I use Pulse streams to mix different sources during the broadcast. -- Sent from my Game Boy. From cehoyos at ag.or.at Wed Apr 17 09:28:36 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 07:28:36 +0000 (UTC) Subject: [FFmpeg-user] FLV streaming, severe audio sync issues References: Message-ID: Rena gmail.com> writes: > > > Duration: N/A, start: 0.000000, bitrate: 705 kb/s > > > > > Duration: N/A, start: 1366150061.077213, bitrate: 156672 kb/s > > > > This unfortunately cannot work;-( > > You can try if the alsa input provides proper timestamps. > > It's a Pulse issue? I would say it is an issue with FFmpeg's pulse device that does not allow to use absolute timestamps. (See v4l2 options for an example.) Carl Eugen From ticks.sunny84 at gmail.com Wed Apr 17 14:25:13 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Wed, 17 Apr 2013 17:55:13 +0530 Subject: [FFmpeg-user] how to compile ffmpeg with H264 as an executable for android armv6 devices Message-ID: Hi All, I have been trying to compile ffmpeg with H264 enabled for ARMV6 devices but i am not able to be successful. I have followed the following link https://github.com/guardianproject/android-ffmpeg I was able to create an ffmpeg executable for ARMV7 but that does not seem to work on ARMV6 devices. Any help will be greatly appriciated. thanks & regards, Susheel Tickoo From sullafelix at ymail.com Wed Apr 17 14:50:04 2013 From: sullafelix at ymail.com (Mark Haney) Date: Wed, 17 Apr 2013 05:50:04 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg 1.0.5 and buffer underflow Message-ID: <1366203004.83081.YahooMailNeo@web122303.mail.ne1.yahoo.com> I've been using ffmpeg for several years to encode video into DVD format.? Mostly old home movies to hand out to the rest of the family.? Lately, I've been getting hammered with buffer underflow errors, even with video I've encoded successfully.? I'm using Fedora 18 and ffmpeg 1.0.5.? I've googled this problem, but found only posts that are ~1 year old or more and have tried a few things to no avail.? Anyone got ideas on how to fix this? From cehoyos at ag.or.at Wed Apr 17 14:53:34 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 12:53:34 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg 1.0.5 and buffer underflow References: <1366203004.83081.YahooMailNeo@web122303.mail.ne1.yahoo.com> Message-ID: Mark Haney ymail.com> writes: > I'm using Fedora 18 and ffmpeg 1.0.5. This is over five months old, please try current git head (or 1.2) Don't forget to post your command line together with the complete, uncut console output if it still fails. Carl Eugen From timbot at oddlystudios.com Wed Apr 17 14:54:57 2013 From: timbot at oddlystudios.com (Tim Willison) Date: Wed, 17 Apr 2013 08:54:57 -0400 Subject: [FFmpeg-user] ffmpeg 1.0.5 and buffer underflow In-Reply-To: <1366203004.83081.YahooMailNeo@web122303.mail.ne1.yahoo.com> References: <1366203004.83081.YahooMailNeo@web122303.mail.ne1.yahoo.com> Message-ID: I have likewise been having buffer overflow problems with the latest git HEAD and have submitted a bug report that is currently open. (#2467) Keeping an eye on that in the bug tracker might give some clues. On Wed, Apr 17, 2013 at 8:50 AM, Mark Haney wrote: > I've been using ffmpeg for several years to encode video into DVD format. > Mostly old home movies to hand out to the rest of the family. Lately, I've > been getting hammered with buffer underflow errors, even with video I've > encoded successfully. I'm using Fedora 18 and ffmpeg 1.0.5. I've googled > this problem, but found only posts that are ~1 year old or more and have > tried a few things to no avail. Anyone got ideas on how to fix this? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Tim Willison Lead Product Designer www.oddlystudios.com 495B Queen St. West Toronto, ON. 647 345 6275 From cehoyos at ag.or.at Wed Apr 17 14:56:40 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 12:56:40 +0000 (UTC) Subject: [FFmpeg-user] how to compile ffmpeg with H264 as an executable for android armv6 devices References: Message-ID: Susheel Tickoo gmail.com> writes: > I have followed the following link [...] Please allow me to repeat that external build scripts are *not* supported on this mailing list! (I have not yet seen one that is not at least broken and does not offer any advantages.) If you believe that our configure script has deficiencies, please report them (here)! See http://fate.ffmpeg.org/ for examples of several successful arm compiles. (I suspect all you have to use is --cpu=armv6) Carl Eugen From Guillaume.Chanel at unige.ch Wed Apr 17 15:20:29 2013 From: Guillaume.Chanel at unige.ch (Guillaume Chanel) Date: Wed, 17 Apr 2013 13:20:29 +0000 Subject: [FFmpeg-user] RE : Problem of PTS for video segmentation In-Reply-To: References: , Message-ID: Dear Carl, I still have the same problem with a git head version. Do you have any idea why the time stamps are modified despite of using -copyts and -copytb ? I also have another question: is it possible to use the -vcodec copy option with the -force-key-frame option ? I tried but it seems it does not work. The idea is to be able to cut a video file without re-encoding the whole video (which implies loss) but just the first frame and the following up to the next keyframe. Guillaume From cehoyos at ag.or.at Wed Apr 17 15:22:10 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 13:22:10 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg 1.0.5 and buffer underflow References: <1366203004.83081.YahooMailNeo@web122303.mail.ne1.yahoo.com> Message-ID: Tim Willison oddlystudios.com> writes: > I have likewise been having buffer overflow problems > with the latest git HEAD and have submitted a bug > report that is currently open. (#2467) Your problem is strongly libavfilter-related (and not reproducible with 1.0 as you know). Please do not top-post here, it is considered rude. Carl Eugen From ticks.sunny84 at gmail.com Wed Apr 17 15:35:11 2013 From: ticks.sunny84 at gmail.com (Susheel Tickoo) Date: Wed, 17 Apr 2013 19:05:11 +0530 Subject: [FFmpeg-user] how to compile ffmpeg with H264 as an executable for android armv6 devices In-Reply-To: References: Message-ID: Thanks Carl, I am new to ffmpeg compilation and didn't new about the scripts. I am not able to find a script for ARMV6 for android on the below link http://fate.ffmpeg.org/. On Wed, Apr 17, 2013 at 6:26 PM, Carl Eugen Hoyos wrote: > Susheel Tickoo gmail.com> writes: > > > I have followed the following link > > [...] > > Please allow me to repeat that external build scripts > are *not* supported on this mailing list! > (I have not yet seen one that is not at least > broken and does not offer any advantages.) > If you believe that our configure script has > deficiencies, please report them (here)! > > See http://fate.ffmpeg.org/ for examples of several > successful arm compiles. > (I suspect all you have to use is --cpu=armv6) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Apr 17 15:58:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 13:58:22 +0000 (UTC) Subject: [FFmpeg-user] how to compile ffmpeg with H264 as an executable for android armv6 devices References: Message-ID: Susheel Tickoo gmail.com> writes: > I am not able to find a script for ARMV6 for > android on the below link http://fate.ffmpeg.org/ As said, on this mailing list only compilation *without* a script is supported, just use configure as several of the test machines on above links. (Click on the numbers on the right to see the configure options.) How does compilation fail for you currently / which configure options did you test but did not produce the wanted binary? Please do not top-post here, it is considered rude. Carl Eugen From jacobhameiri at gmail.com Wed Apr 17 17:04:33 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Wed, 17 Apr 2013 08:04:33 -0700 (PDT) Subject: [FFmpeg-user] multicast streaming real time raw rgb images Message-ID: <1366211073584-4658525.post@n4.nabble.com> Hi, I am trying to stream real time raw data via multicast but I am having 2 issues. 1. I cant get rid of a 1-2 seconds lag I have. 2. my played video stream has many glitches and corrupted frames. this is the ffmpeg command and output: and ffplay command and output: ffmpeg: ffmpeg started on 2013-04-17 at 11:38:57 Report written to "ffmpeg-20130417-113857.log" Command line: /bin/ffmpeg -an -f rawvideo -vcodec rawvideo -s 800x1280 -pix_fmt rgba -r 9 -i - -vcodec mpeg4 -r 7 -filter:v "scale=400:-1" -sws_flags fast_bilinear -b 300000 -f mpegts udp://224.1.1.1:1234 -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [rawvideo @ 0x291d620] File position before avformat_find_stream_info() is 0 [rawvideo @ 0x291d620] All info found [rawvideo @ 0x291d620] Estimating duration from bitrate, this may be inaccurate [rawvideo @ 0x291d620] File position after avformat_find_stream_info() is 4096000 Input #0, rawvideo, from 'pipe:': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 1, 1/9: Video: rawvideo (RGBA / 0x41424752), rgba, 800x1280, 9 tbr, 9 tbn, 9 tbc Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x292a6d0] Setting entry with key 'video_size' to value '800x1280' [buffer @ 0x292a6d0] Setting entry with key 'pix_fmt' to value '28' [buffer @ 0x292a6d0] Setting entry with key 'time_base' to value '1/9' [buffer @ 0x292a6d0] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x292a6d0] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x292a6d0] Setting entry with key 'frame_rate' to value '9/1' [graph 0 input from stream 0:0 @ 0x292a680] w:800 h:1280 pixfmt:rgba tb:1/9 fr:9/1 sar:0/1 sws_param:flags=2 [graph 0 input from stream 0:0 @ 0x292a680] TB:0.111111 FRAME_RATE:9.000000 SAMPLE_RATE:NaN [Parsed_scale_0 @ 0x291d5e0] w:800 h:1280 fmt:rgba sar:0/1 -> w:400 h:640 fmt:yuv420p sar:0/1 flags:0x1 [mpeg4 @ 0x2926970] detected 4 logical cores [mpeg4 @ 0x2926970] intra_quant_bias = 0 inter_quant_bias = -64 [mpegts @ 0x29261f0] muxrate VBR, pcr every 1 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://224.1.1.1:1234': Metadata: encoder : Lavf54.29.104 Stream #0:0, 0, 1/90000: Video: mpeg4, yuv420p, 400x640, q=2-31, 300 kb/s, 90k tbn, 7 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mpeg4) frame= 6 fps=0.0 q=2.0 size= 69kB time=00:00:00.85 bitrate= 656.2kbits/s frame= 11 fps= 10 q=2.0 size= 70kB time=00:00:01.57 bitrate= 363.7kbits/s *** drop! frame= 15 fps=9.4 q=2.0 size= 146kB time=00:00:02.14 bitrate= 559.4kbits/s dup=0 drop=1 *** drop! frame= 19 fps=9.1 q=2.0 size= 147kB time=00:00:02.71 bitrate= 443.8kbits/s dup=0 drop=2 *** drop! *** drop! frame= 22 fps=8.4 q=2.0 size= 149kB time=00:00:03.14 bitrate= 388.6kbits/s dup=0 drop=4 *** drop! frame= 26 fps=8.3 q=2.0 size= 225kB time=00:00:03.71 bitrate= 496.4kbits/s dup=0 drop=5 *** drop! frame= 30 fps=8.2 q=2.0 size= 226kB time=00:00:04.28 bitrate= 432.0kbits/s dup=0 drop=6 *** drop! frame= 34 fps=8.1 q=2.0 size= 227kB time=00:00:04.85 bitrate= 382.4kbits/s dup=0 drop=7 *** drop! frame= 38 fps=8.0 q=2.0 size= 303kB time=00:00:05.42 bitrate= 457.4kbits/s dup=0 drop=8 *** drop! frame= 42 fps=8.0 q=2.0 size= 304kB time=00:00:06.00 bitrate= 415.1kbits/s dup=0 drop=9 *** drop! frame= 46 fps=7.9 q=2.0 size= 305kB time=00:00:06.57 bitrate= 379.9kbits/s dup=0 drop=10 *** drop! frame= 50 fps=7.9 q=2.0 size= 381kB time=00:00:07.14 bitrate= 437.1kbits/s dup=0 drop=11 *** drop! frame= 54 fps=7.9 q=2.0 size= 382kB time=00:00:07.71 bitrate= 405.7kbits/s dup=0 drop=12 *** drop! *** drop! frame= 57 fps=7.7 q=2.0 size= 383kB time=00:00:08.14 bitrate= 384.9kbits/s dup=0 drop=14 *** drop! frame= 61 fps=7.7 q=1.6 size= 459kB time=00:00:08.71 bitrate= 431.3kbits/s dup=0 drop=15 *** drop! frame= 64 fps=7.5 q=2.0 size= 501kB time=00:00:09.14 bitrate= 449.2kbits/s dup=0 drop=16 frame= 68 fps=7.5 q=4.7 size= 656kB time=00:00:09.71 bitrate= 553.5kbits/s dup=0 drop=16 *** drop! frame= 71 fps=7.3 q=2.9 size= 672kB time=00:00:10.14 bitrate= 542.9kbits/s dup=0 drop=17 *** drop! frame= 73 fps=7.1 q=2.2 size= 713kB time=00:00:10.42 bitrate= 559.9kbits/s dup=0 drop=18 *** drop! frame= 76 fps=7.0 q=2.9 size= 735kB time=00:00:10.85 bitrate= 554.8kbits/s dup=0 drop=19 *** drop! frame= 80 fps=7.1 q=2.0 size= 746kB time=00:00:11.42 bitrate= 534.8kbits/s dup=0 drop=20 *** drop! frame= 82 fps=6.9 q=2.9 size= 766kB time=00:00:11.71 bitrate= 535.8kbits/s dup=0 drop=21 *** drop! frame= 85 fps=6.9 q=1.6 size= 782kB time=00:00:12.14 bitrate= 527.8kbits/s dup=0 drop=22 *** drop! frame= 89 fps=6.9 q=4.5 size= 850kB time=00:00:12.71 bitrate= 547.7kbits/s dup=0 drop=23 *** drop! frame= 93 fps=6.9 q=2.9 size= 857kB time=00:00:13.28 bitrate= 528.6kbits/s dup=0 drop=24 *** drop! frame= 97 fps=6.9 q=1.9 size= 910kB time=00:00:13.85 bitrate= 537.9kbits/s dup=0 drop=25 *** drop! frame= 101 fps=6.9 q=2.0 size= 912kB time=00:00:14.42 bitrate= 518.0kbits/s dup=0 drop=26 *** drop! frame= 105 fps=6.9 q=2.0 size= 915kB time=00:00:15.00 bitrate= 499.6kbits/s dup=0 drop=27 *** drop! frame= 109 fps=6.9 q=1.6 size= 959kB time=00:00:15.57 bitrate= 504.4kbits/s dup=0 drop=28 *** drop! frame= 112 fps=6.8 q=2.0 size= 964kB time=00:00:16.00 bitrate= 493.4kbits/s dup=0 drop=29 *** drop! frame= 116 fps=6.8 q=2.0 size= 968kB time=00:00:16.57 bitrate= 478.3kbits/s dup=0 drop=30 *** drop! frame= 120 fps=6.8 q=2.0 size= 970kB time=00:00:17.14 bitrate= 463.3kbits/s dup=0 drop=31 *** drop! frame= 124 fps=6.8 q=2.0 size= 1011kB time=00:00:17.71 bitrate= 467.6kbits/s dup=0 drop=32 *** drop! *** drop! frame= 127 fps=6.7 q=2.0 size= 1013kB time=00:00:18.14 bitrate= 457.3kbits/s dup=0 drop=34 frame= 131 fps=6.7 q=2.0 size= 1015kB time=00:00:18.71 bitrate= 444.4kbits/s dup=0 drop=34 *** drop! frame= 134 fps=6.7 q=2.0 size= 1056kB time=00:00:19.14 bitrate= 452.0kbits/s dup=0 drop=35 *** drop! frame= 138 fps=6.7 q=2.0 size= 1058kB time=00:00:19.71 bitrate= 439.8kbits/s dup=0 drop=36 *** drop! *** drop! frame= 141 fps=6.7 q=2.0 size= 1060kB time=00:00:20.14 bitrate= 431.1kbits/s dup=0 drop=38 *** drop! frame= 145 fps=6.6 q=1.6 size= 1102kB time=00:00:20.71 bitrate= 435.6kbits/s dup=0 drop=39 *** drop! frame= 149 fps=6.6 q=2.0 size= 1104kB time=00:00:21.28 bitrate= 424.8kbits/s dup=0 drop=40 *** drop! frame= 152 fps=6.6 q=2.0 size= 1105kB time=00:00:21.71 bitrate= 417.0kbits/s dup=0 drop=41 *** drop! frame= 155 fps=6.6 q=2.0 size= 1107kB time=00:00:22.14 bitrate= 409.6kbits/s dup=0 drop=42 *** drop! frame= 159 fps=6.6 q=2.0 size= 1148kB time=00:00:22.71 bitrate= 413.9kbits/s dup=0 drop=43 *** drop! frame= 163 fps=6.6 q=2.0 size= 1149kB time=00:00:23.28 bitrate= 404.3kbits/s dup=0 drop=44 *** drop! frame= 166 fps=6.6 q=2.0 size= 1176kB time=00:00:23.71 bitrate= 406.1kbits/s dup=0 drop=45 *** drop! frame= 169 fps=6.6 q=1.6 size= 1237kB time=00:00:24.14 bitrate= 419.9kbits/s dup=0 drop=46 frame= 173 fps=6.6 q=2.0 size= 1240kB time=00:00:24.71 bitrate= 411.1kbits/s dup=0 drop=46 *** drop! frame= 176 fps=6.6 q=2.0 size= 1242kB time=00:00:25.14 bitrate= 404.6kbits/s dup=0 drop=47 *** drop! frame= 179 fps=6.6 q=2.0 size= 1244kB time=00:00:25.57 bitrate= 398.4kbits/s dup=0 drop=48 *** drop! frame= 182 fps=6.5 q=2.0 size= 1327kB time=00:00:26.00 bitrate= 418.2kbits/s dup=0 drop=49 *** drop! frame= 186 fps=6.6 q=5.0 size= 1374kB time=00:00:26.57 bitrate= 423.7kbits/s dup=0 drop=50 *** drop! frame= 190 fps=6.6 q=6.6 size= 1406kB time=00:00:27.14 bitrate= 424.3kbits/s dup=0 drop=51 *** drop! frame= 194 fps=6.6 q=7.2 size= 1463kB time=00:00:27.71 bitrate= 432.4kbits/s dup=0 drop=52 *** drop! *** drop! frame= 197 fps=6.6 q=8.6 size= 1485kB time=00:00:28.14 bitrate= 432.3kbits/s dup=0 drop=54 *** drop! frame= 201 fps=6.6 q=9.1 size= 1510kB time=00:00:28.71 bitrate= 430.7kbits/s dup=0 drop=55 *** drop! frame= 205 fps=6.6 q=8.4 size= 1550kB time=00:00:29.28 bitrate= 433.5kbits/s dup=0 drop=56 *** drop! frame= 209 fps=6.6 q=8.5 size= 1566kB time=00:00:29.85 bitrate= 429.6kbits/s dup=0 drop=57 *** drop! frame= 213 fps=6.6 q=8.8 size= 1583kB time=00:00:30.42 bitrate= 426.3kbits/s dup=0 drop=58 *** drop! frame= 217 fps=6.6 q=7.5 size= 1619kB time=00:00:31.00 bitrate= 427.8kbits/s dup=0 drop=59 *** drop! frame= 221 fps=6.6 q=9.9 size= 1636kB time=00:00:31.57 bitrate= 424.4kbits/s dup=0 drop=60 *** drop! frame= 225 fps=6.6 q=10.6 size= 1655kB time=00:00:32.14 bitrate= 421.9kbits/s dup=0 drop=61 *** drop! frame= 229 fps=6.6 q=9.1 size= 1693kB time=00:00:32.71 bitrate= 424.0kbits/s dup=0 drop=62 *** drop! *** drop! frame= 232 fps=6.6 q=14.4 size= 1709kB time=00:00:33.14 bitrate= 422.4kbits/s dup=0 drop=64 *** drop! frame= 236 fps=6.6 q=13.8 size= 1724kB time=00:00:33.71 bitrate= 419.0kbits/s dup=0 drop=65 *** drop! frame= 239 fps=6.6 q=8.5 size= 1731kB time=00:00:34.14 bitrate= 415.3kbits/s dup=0 drop=66 frame= 243 fps=6.6 q=5.6 size= 1769kB time=00:00:34.71 bitrate= 417.4kbits/s dup=0 drop=66 *** drop! frame= 246 fps=6.6 q=3.4 size= 1778kB time=00:00:35.14 bitrate= 414.6kbits/s dup=0 drop=67 *** drop! frame= 249 fps=6.6 q=3.6 size= 1780kB time=00:00:35.57 bitrate= 409.8kbits/s dup=0 drop=68 *** drop! frame= 252 fps=6.6 q=2.2 size= 1795kB time=00:00:36.00 bitrate= 408.4kbits/s dup=0 drop=69 *** drop! frame= 255 fps=6.6 q=4.1 size= 1856kB time=00:00:36.42 bitrate= 417.5kbits/s dup=0 drop=70 *** drop! frame= 259 fps=6.6 q=2.1 size= 1858kB time=00:00:37.00 bitrate= 411.4kbits/s dup=0 drop=71 *** drop! frame= 263 fps=6.6 q=7.5 size= 1893kB time=00:00:37.57 bitrate= 412.8kbits/s dup=0 drop=72 *** drop! frame= 267 fps=6.6 q=14.2 size= 1926kB time=00:00:38.14 bitrate= 413.7kbits/s dup=0 drop=73 *** drop! frame= 271 fps=6.6 q=15.0 size= 1941kB time=00:00:38.71 bitrate= 410.8kbits/s dup=0 drop=74 *** drop! *** drop! frame= 274 fps=6.6 q=14.1 size= 1952kB time=00:00:39.14 bitrate= 408.6kbits/s dup=0 drop=76 *** drop! frame= 278 fps=6.6 q=14.1 size= 1983kB time=00:00:39.71 bitrate= 409.0kbits/s dup=0 drop=77 *** drop! frame= 282 fps=6.6 q=14.0 size= 1997kB time=00:00:40.28 bitrate= 406.1kbits/s dup=0 drop=78 *** drop! frame= 286 fps=6.6 q=13.7 size= 2010kB time=00:00:40.85 bitrate= 403.0kbits/s dup=0 drop=79 *** drop! frame= 290 fps=6.6 q=11.9 size= 2041kB time=00:00:41.42 bitrate= 403.7kbits/s dup=0 drop=80 *** drop! frame= 294 fps=6.6 q=13.5 size= 2056kB time=00:00:42.00 bitrate= 401.0kbits/s dup=0 drop=81 *** drop! frame= 298 fps=6.6 q=11.5 size= 2069kB time=00:00:42.57 bitrate= 398.1kbits/s dup=0 drop=82 *** drop! frame= 302 fps=6.6 q=11.6 size= 2101kB time=00:00:43.14 bitrate= 399.0kbits/s dup=0 drop=83 *** drop! frame= 306 fps=6.6 q=12.4 size= 2115kB time=00:00:43.71 bitrate= 396.4kbits/s dup=0 drop=84 *** drop! *** drop! frame= 309 fps=6.6 q=11.8 size= 2125kB time=00:00:44.14 bitrate= 394.4kbits/s dup=0 drop=86 *** drop! frame= 313 fps=6.6 q=4.5 size= 2159kB time=00:00:44.71 bitrate= 395.5kbits/s dup=0 drop=87 *** drop! frame= 317 fps=6.6 q=5.7 size= 2165kB time=00:00:45.28 bitrate= 391.7kbits/s dup=0 drop=88 *** drop! frame= 321 fps=6.6 q=6.7 size= 2182kB time=00:00:45.85 bitrate= 389.8kbits/s dup=0 drop=89 *** drop! frame= 325 fps=6.6 q=5.8 size= 2220kB time=00:00:46.42 bitrate= 391.7kbits/s dup=0 drop=90 *** drop! frame= 329 fps=6.6 q=5.8 size= 2227kB time=00:00:47.00 bitrate= 388.1kbits/s dup=0 drop=91 *** drop! frame= 333 fps=6.6 q=5.0 size= 2234kB time=00:00:47.57 bitrate= 384.8kbits/s dup=0 drop=92 *** drop! frame= 337 fps=6.6 q=3.2 size= 2294kB time=00:00:48.14 bitrate= 390.3kbits/s dup=0 drop=93 *** drop! --------------------------------------------------------------------------------------------------------------------------- ffplay: ffplay started on 2013-04-17 at 11:38:14 Report written to "ffplay-20130417-113814.log" Command line: "d:\\ffmpeg\\ffplay.exe" udp://224.1.1.1:1234 -probesize 50000 -analyzeduration 50 -report ffplay version N-46469-gc995644 Copyright (c) 2003-2012 the FFmpeg developers built on Nov 5 2012 17:58:15 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [mpegts @ 00000000020c22c0] Format mpegts probed with size=2048 and score=100 [mpegts @ 00000000020c22c0] Unable to seek back to the start [mpegts @ 00000000020c22c0] stream=0 stream_type=10 pid=100 prog_reg_desc= [mpegts @ 00000000020c22c0] File position before avformat_find_stream_info() is 16356 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 4 got 12 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 00000000020f1a60] hmm, seems the headers are not complete, trying to guess time_increment_bits [mpeg4 @ 00000000020f1a60] my guess is 3 bits ;) [mpegts @ 00000000020c22c0] invalid dts/pts combination 138857 [mpeg4 @ 00000000020f1a60] looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 151714 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 164571 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 177429 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 190286 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 203143 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 216000 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 228857 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 241714 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 254571 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 15 [mpegts @ 00000000020c22c0] invalid dts/pts combination 267429 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] Probe buffer size limit 50000 reached [mpegts @ 00000000020c22c0] Estimating duration from bitrate, this may be inaccurate [mpegts @ 00000000020c22c0] File position after avformat_find_stream_info() is 140628 Input #0, mpegts, from 'udp://224.1.1.1:1234': Duration: N/A, start: 1.542856, bitrate: N/A Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100], 12, 1/90000: Video: mpeg4 (Simple Profile) ([16][0][0][0] / 0x0010), yuv420p, 400x640 [SAR 1:1 DAR 5:8], 7 fps, 7 tbr, 90k tbn, 7 tbc [mpeg4 @ 00000000020f1a60] detected 8 logical cores [mpeg4 @ 0000000005f98f60] warning: first frame is no keyframe [mpeg4 @ 0000000005f98f60] ac-tex damaged at 12 16 [mpeg4 @ 0000000005f98f60] Error at MB: 428 [mpeg4 @ 0000000005f98f60] Error at MB: 685 [mpeg4 @ 0000000005f98f60] concealing 395 DC, 395 AC, 395 MV errors in P frame Frame changed from size:0x0 to size:400x640 [buffer @ 00000000021343e0] Setting entry with key 'video_size' to value '400x640' [buffer @ 00000000021343e0] Setting entry with key 'pix_fmt' to value '0' [buffer @ 00000000021343e0] Setting entry with key 'time_base' to value '1/90000' [buffer @ 00000000021343e0] Setting entry with key 'pixel_aspect' to value '1/1' [ffplay_buffer @ 00000000020fffe0] w:400 h:640 pixfmt:yuv420p tb:1/90000 fr:0/1 sar:1/1 sws_param: [ffplay_crop @ 0000000002100080] w:400 h:640 sar:1/1 -> w:400 h:640 sar:1/1 [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 19 20 [mpeg4 @ 0000000005f9a0a0] Error at MB: 539 [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 18 26 [mpeg4 @ 0000000005f9a0a0] Error at MB: 694 [mpeg4 @ 0000000005f9a0a0] concealing 311 DC, 311 AC, 311 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 4 got 5 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 12 got 2 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 9 got 10 [mpeg4 @ 00000000020efbc0] ac-tex damaged at 23 14 [mpeg4 @ 00000000020efbc0] Error at MB: 387 [mpeg4 @ 00000000020efbc0] concealing 76 DC, 76 AC, 76 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 14 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 3 got 9 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 1 got 9 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 7 got 8 [mpeg4 @ 00000000020f1140] ac-tex damaged at 10 21 [mpeg4 @ 00000000020f1140] Error at MB: 556 [mpeg4 @ 00000000020f1140] ac-tex damaged at 4 26 [mpeg4 @ 00000000020f1140] Error at MB: 680 [mpeg4 @ 00000000020f1140] concealing 309 DC, 309 AC, 309 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 17 expected 7 got 8 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 6 got 14 [mpeg4 @ 0000000005f996a0] ac-tex damaged at 24 11 [mpeg4 @ 0000000005f996a0] Error at MB: 310 [mpeg4 @ 0000000005f996a0] concealing 150 DC, 150 AC, 150 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 13 [mpeg4 @ 00000000020efbc0] 1. marker bit missing in 3. esc [mpeg4 @ 00000000020efbc0] Error at MB: 169 [mpeg4 @ 00000000020efbc0] concealing 86 DC, 86 AC, 86 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 13 got 3 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 10 got 11 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 4 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 11 got 2 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 15 got 0 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 0 got 8 [mpeg4 @ 000000000210d500] ac-tex damaged at 15 22 [mpeg4 @ 000000000210d500] Error at MB: 587 [mpeg4 @ 000000000210d500] concealing 284 DC, 284 AC, 284 MV errors in P frame [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f98f60] ac-tex damaged at 23 15 [mpeg4 @ 0000000005f98f60] Error at MB: 413 [mpeg4 @ 0000000005f98f60] concealing 259 DC, 259 AC, 259 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 15 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 24 4 [mpeg4 @ 0000000005f996a0] Error at MB: 128 [mpeg4 @ 0000000005f996a0] concealing 127 DC, 127 AC, 127 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 9 got 10 [mpeg4 @ 0000000005f9a0a0] concealing 25 DC, 25 AC, 25 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 1 got 10 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 00000000020f1140] I cbpy damaged at 1 14 [mpeg4 @ 00000000020f1140] Error at MB: 365 [mpeg4 @ 00000000020f1140] concealing 98 DC, 98 AC, 98 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 2 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 18 33 [mpeg4 @ 0000000005f9a0a0] Error at MB: 876 [mpeg4 @ 0000000005f9a0a0] concealing 200 DC, 200 AC, 200 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 13 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 2 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 3 2 [mpeg4 @ 0000000005f996a0] Error at MB: 55 [mpeg4 @ 0000000005f996a0] 2. marker bit missing in 3. esc [mpeg4 @ 0000000005f996a0] Error at MB: 456 [mpeg4 @ 0000000005f996a0] concealing 396 DC, 396 AC, 396 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 8 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 2 got 10 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f98f60] I cbpy damaged at 16 18 [mpeg4 @ 0000000005f98f60] Error at MB: 484 [mpeg4 @ 0000000005f98f60] concealing 183 DC, 183 AC, 183 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 8 got 15 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 1 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 11 got 3 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 000000000210d500] Error at MB: 198 [mpeg4 @ 000000000210d500] concealing 268 DC, 268 AC, 268 MV errors in P frame [mpeg4 @ 0000000005f98f60] illegal dc vlc [mpeg4 @ 0000000005f98f60] Error at MB: 83 [mpeg4 @ 0000000005f98f60] I cbpy damaged at 15 18 [mpeg4 @ 0000000005f98f60] Error at MB: 483 [mpeg4 @ 0000000005f98f60] concealing 353 DC, 353 AC, 353 MV errors in I frame [mpegts @ 00000000020c22c0] first_dts 138857 not matching first dts 2903143 in que [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 10 [mpeg4 @ 00000000020f1140] ac-tex damaged at 22 17 [mpeg4 @ 00000000020f1140] Error at MB: 464 [mpeg4 @ 00000000020f1140] concealing 600 DC, 600 AC, 600 MV errors in P frame [mpeg4 @ 000000000210d500] ac-tex damaged at 8 3 [mpeg4 @ 000000000210d500] Error at MB: 86 [mpeg4 @ 000000000210d500] concealing 1000 DC, 1000 AC, 1000 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 0 got 6 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f9a7e0] concealing 296 DC, 296 AC, 296 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 13 got 14 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 0 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 9 9 [mpeg4 @ 0000000005f996a0] Error at MB: 243 [mpeg4 @ 0000000005f996a0] concealing 400 DC, 400 AC, 400 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 11 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 11 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 6 got 7 -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/multicast-streaming-real-time-raw-rgb-images-tp4658525.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From jacobhameiri at gmail.com Wed Apr 17 16:51:39 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Wed, 17 Apr 2013 07:51:39 -0700 (PDT) Subject: [FFmpeg-user] multicast streaming real time raw rgb images Message-ID: <1366210298888-4658524.post@n4.nabble.com> Hi, I am trying to stream real time raw data via multicast but I am having 2 issues. 1. I cant get rid of a 1-2 seconds lag I have. 2. my played video stream has many glitches and corrupted frames. this is the ffmpeg command and output: and ffplay command and output: ffmpeg: ffmpeg started on 2013-04-17 at 11:38:57 Report written to "ffmpeg-20130417-113857.log" Command line: /bin/ffmpeg -an -f rawvideo -vcodec rawvideo -s 800x1280 -pix_fmt rgba -r 9 -i - -vcodec mpeg4 -r 7 -filter:v "scale=400:-1" -sws_flags fast_bilinear -b 300000 -f mpegts udp://224.1.1.1:1234 -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [rawvideo @ 0x291d620] File position before avformat_find_stream_info() is 0 [rawvideo @ 0x291d620] All info found [rawvideo @ 0x291d620] Estimating duration from bitrate, this may be inaccurate [rawvideo @ 0x291d620] File position after avformat_find_stream_info() is 4096000 Input #0, rawvideo, from 'pipe:': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 1, 1/9: Video: rawvideo (RGBA / 0x41424752), rgba, 800x1280, 9 tbr, 9 tbn, 9 tbc Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x292a6d0] Setting entry with key 'video_size' to value '800x1280' [buffer @ 0x292a6d0] Setting entry with key 'pix_fmt' to value '28' [buffer @ 0x292a6d0] Setting entry with key 'time_base' to value '1/9' [buffer @ 0x292a6d0] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x292a6d0] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x292a6d0] Setting entry with key 'frame_rate' to value '9/1' [graph 0 input from stream 0:0 @ 0x292a680] w:800 h:1280 pixfmt:rgba tb:1/9 fr:9/1 sar:0/1 sws_param:flags=2 [graph 0 input from stream 0:0 @ 0x292a680] TB:0.111111 FRAME_RATE:9.000000 SAMPLE_RATE:NaN [Parsed_scale_0 @ 0x291d5e0] w:800 h:1280 fmt:rgba sar:0/1 -> w:400 h:640 fmt:yuv420p sar:0/1 flags:0x1 [mpeg4 @ 0x2926970] detected 4 logical cores [mpeg4 @ 0x2926970] intra_quant_bias = 0 inter_quant_bias = -64 [mpegts @ 0x29261f0] muxrate VBR, pcr every 1 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://224.1.1.1:1234': Metadata: encoder : Lavf54.29.104 Stream #0:0, 0, 1/90000: Video: mpeg4, yuv420p, 400x640, q=2-31, 300 kb/s, 90k tbn, 7 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mpeg4) frame= 6 fps=0.0 q=2.0 size= 69kB time=00:00:00.85 bitrate= 656.2kbits/s frame= 11 fps= 10 q=2.0 size= 70kB time=00:00:01.57 bitrate= 363.7kbits/s *** drop! frame= 15 fps=9.4 q=2.0 size= 146kB time=00:00:02.14 bitrate= 559.4kbits/s dup=0 drop=1 *** drop! frame= 19 fps=9.1 q=2.0 size= 147kB time=00:00:02.71 bitrate= 443.8kbits/s dup=0 drop=2 *** drop! *** drop! frame= 22 fps=8.4 q=2.0 size= 149kB time=00:00:03.14 bitrate= 388.6kbits/s dup=0 drop=4 *** drop! frame= 26 fps=8.3 q=2.0 size= 225kB time=00:00:03.71 bitrate= 496.4kbits/s dup=0 drop=5 *** drop! frame= 30 fps=8.2 q=2.0 size= 226kB time=00:00:04.28 bitrate= 432.0kbits/s dup=0 drop=6 *** drop! frame= 34 fps=8.1 q=2.0 size= 227kB time=00:00:04.85 bitrate= 382.4kbits/s dup=0 drop=7 *** drop! frame= 38 fps=8.0 q=2.0 size= 303kB time=00:00:05.42 bitrate= 457.4kbits/s dup=0 drop=8 *** drop! frame= 42 fps=8.0 q=2.0 size= 304kB time=00:00:06.00 bitrate= 415.1kbits/s dup=0 drop=9 *** drop! frame= 46 fps=7.9 q=2.0 size= 305kB time=00:00:06.57 bitrate= 379.9kbits/s dup=0 drop=10 *** drop! frame= 50 fps=7.9 q=2.0 size= 381kB time=00:00:07.14 bitrate= 437.1kbits/s dup=0 drop=11 *** drop! frame= 54 fps=7.9 q=2.0 size= 382kB time=00:00:07.71 bitrate= 405.7kbits/s dup=0 drop=12 *** drop! *** drop! frame= 57 fps=7.7 q=2.0 size= 383kB time=00:00:08.14 bitrate= 384.9kbits/s dup=0 drop=14 *** drop! frame= 61 fps=7.7 q=1.6 size= 459kB time=00:00:08.71 bitrate= 431.3kbits/s dup=0 drop=15 *** drop! frame= 64 fps=7.5 q=2.0 size= 501kB time=00:00:09.14 bitrate= 449.2kbits/s dup=0 drop=16 frame= 68 fps=7.5 q=4.7 size= 656kB time=00:00:09.71 bitrate= 553.5kbits/s dup=0 drop=16 *** drop! frame= 71 fps=7.3 q=2.9 size= 672kB time=00:00:10.14 bitrate= 542.9kbits/s dup=0 drop=17 *** drop! frame= 73 fps=7.1 q=2.2 size= 713kB time=00:00:10.42 bitrate= 559.9kbits/s dup=0 drop=18 *** drop! frame= 76 fps=7.0 q=2.9 size= 735kB time=00:00:10.85 bitrate= 554.8kbits/s dup=0 drop=19 *** drop! frame= 80 fps=7.1 q=2.0 size= 746kB time=00:00:11.42 bitrate= 534.8kbits/s dup=0 drop=20 *** drop! frame= 82 fps=6.9 q=2.9 size= 766kB time=00:00:11.71 bitrate= 535.8kbits/s dup=0 drop=21 *** drop! frame= 85 fps=6.9 q=1.6 size= 782kB time=00:00:12.14 bitrate= 527.8kbits/s dup=0 drop=22 *** drop! frame= 89 fps=6.9 q=4.5 size= 850kB time=00:00:12.71 bitrate= 547.7kbits/s dup=0 drop=23 *** drop! frame= 93 fps=6.9 q=2.9 size= 857kB time=00:00:13.28 bitrate= 528.6kbits/s dup=0 drop=24 *** drop! frame= 97 fps=6.9 q=1.9 size= 910kB time=00:00:13.85 bitrate= 537.9kbits/s dup=0 drop=25 *** drop! frame= 101 fps=6.9 q=2.0 size= 912kB time=00:00:14.42 bitrate= 518.0kbits/s dup=0 drop=26 *** drop! frame= 105 fps=6.9 q=2.0 size= 915kB time=00:00:15.00 bitrate= 499.6kbits/s dup=0 drop=27 *** drop! frame= 109 fps=6.9 q=1.6 size= 959kB time=00:00:15.57 bitrate= 504.4kbits/s dup=0 drop=28 *** drop! frame= 112 fps=6.8 q=2.0 size= 964kB time=00:00:16.00 bitrate= 493.4kbits/s dup=0 drop=29 *** drop! frame= 116 fps=6.8 q=2.0 size= 968kB time=00:00:16.57 bitrate= 478.3kbits/s dup=0 drop=30 *** drop! frame= 120 fps=6.8 q=2.0 size= 970kB time=00:00:17.14 bitrate= 463.3kbits/s dup=0 drop=31 *** drop! frame= 124 fps=6.8 q=2.0 size= 1011kB time=00:00:17.71 bitrate= 467.6kbits/s dup=0 drop=32 *** drop! *** drop! frame= 127 fps=6.7 q=2.0 size= 1013kB time=00:00:18.14 bitrate= 457.3kbits/s dup=0 drop=34 frame= 131 fps=6.7 q=2.0 size= 1015kB time=00:00:18.71 bitrate= 444.4kbits/s dup=0 drop=34 *** drop! frame= 134 fps=6.7 q=2.0 size= 1056kB time=00:00:19.14 bitrate= 452.0kbits/s dup=0 drop=35 *** drop! frame= 138 fps=6.7 q=2.0 size= 1058kB time=00:00:19.71 bitrate= 439.8kbits/s dup=0 drop=36 *** drop! *** drop! frame= 141 fps=6.7 q=2.0 size= 1060kB time=00:00:20.14 bitrate= 431.1kbits/s dup=0 drop=38 *** drop! frame= 145 fps=6.6 q=1.6 size= 1102kB time=00:00:20.71 bitrate= 435.6kbits/s dup=0 drop=39 *** drop! frame= 149 fps=6.6 q=2.0 size= 1104kB time=00:00:21.28 bitrate= 424.8kbits/s dup=0 drop=40 *** drop! frame= 152 fps=6.6 q=2.0 size= 1105kB time=00:00:21.71 bitrate= 417.0kbits/s dup=0 drop=41 *** drop! frame= 155 fps=6.6 q=2.0 size= 1107kB time=00:00:22.14 bitrate= 409.6kbits/s dup=0 drop=42 *** drop! frame= 159 fps=6.6 q=2.0 size= 1148kB time=00:00:22.71 bitrate= 413.9kbits/s dup=0 drop=43 *** drop! frame= 163 fps=6.6 q=2.0 size= 1149kB time=00:00:23.28 bitrate= 404.3kbits/s dup=0 drop=44 *** drop! frame= 166 fps=6.6 q=2.0 size= 1176kB time=00:00:23.71 bitrate= 406.1kbits/s dup=0 drop=45 *** drop! frame= 169 fps=6.6 q=1.6 size= 1237kB time=00:00:24.14 bitrate= 419.9kbits/s dup=0 drop=46 frame= 173 fps=6.6 q=2.0 size= 1240kB time=00:00:24.71 bitrate= 411.1kbits/s dup=0 drop=46 *** drop! frame= 176 fps=6.6 q=2.0 size= 1242kB time=00:00:25.14 bitrate= 404.6kbits/s dup=0 drop=47 *** drop! frame= 179 fps=6.6 q=2.0 size= 1244kB time=00:00:25.57 bitrate= 398.4kbits/s dup=0 drop=48 *** drop! frame= 182 fps=6.5 q=2.0 size= 1327kB time=00:00:26.00 bitrate= 418.2kbits/s dup=0 drop=49 *** drop! frame= 186 fps=6.6 q=5.0 size= 1374kB time=00:00:26.57 bitrate= 423.7kbits/s dup=0 drop=50 *** drop! frame= 190 fps=6.6 q=6.6 size= 1406kB time=00:00:27.14 bitrate= 424.3kbits/s dup=0 drop=51 *** drop! frame= 194 fps=6.6 q=7.2 size= 1463kB time=00:00:27.71 bitrate= 432.4kbits/s dup=0 drop=52 *** drop! *** drop! frame= 197 fps=6.6 q=8.6 size= 1485kB time=00:00:28.14 bitrate= 432.3kbits/s dup=0 drop=54 *** drop! frame= 201 fps=6.6 q=9.1 size= 1510kB time=00:00:28.71 bitrate= 430.7kbits/s dup=0 drop=55 *** drop! frame= 205 fps=6.6 q=8.4 size= 1550kB time=00:00:29.28 bitrate= 433.5kbits/s dup=0 drop=56 *** drop! frame= 209 fps=6.6 q=8.5 size= 1566kB time=00:00:29.85 bitrate= 429.6kbits/s dup=0 drop=57 *** drop! frame= 213 fps=6.6 q=8.8 size= 1583kB time=00:00:30.42 bitrate= 426.3kbits/s dup=0 drop=58 *** drop! frame= 217 fps=6.6 q=7.5 size= 1619kB time=00:00:31.00 bitrate= 427.8kbits/s dup=0 drop=59 *** drop! frame= 221 fps=6.6 q=9.9 size= 1636kB time=00:00:31.57 bitrate= 424.4kbits/s dup=0 drop=60 *** drop! frame= 225 fps=6.6 q=10.6 size= 1655kB time=00:00:32.14 bitrate= 421.9kbits/s dup=0 drop=61 *** drop! frame= 229 fps=6.6 q=9.1 size= 1693kB time=00:00:32.71 bitrate= 424.0kbits/s dup=0 drop=62 *** drop! *** drop! frame= 232 fps=6.6 q=14.4 size= 1709kB time=00:00:33.14 bitrate= 422.4kbits/s dup=0 drop=64 *** drop! frame= 236 fps=6.6 q=13.8 size= 1724kB time=00:00:33.71 bitrate= 419.0kbits/s dup=0 drop=65 *** drop! frame= 239 fps=6.6 q=8.5 size= 1731kB time=00:00:34.14 bitrate= 415.3kbits/s dup=0 drop=66 frame= 243 fps=6.6 q=5.6 size= 1769kB time=00:00:34.71 bitrate= 417.4kbits/s dup=0 drop=66 *** drop! frame= 246 fps=6.6 q=3.4 size= 1778kB time=00:00:35.14 bitrate= 414.6kbits/s dup=0 drop=67 *** drop! frame= 249 fps=6.6 q=3.6 size= 1780kB time=00:00:35.57 bitrate= 409.8kbits/s dup=0 drop=68 *** drop! frame= 252 fps=6.6 q=2.2 size= 1795kB time=00:00:36.00 bitrate= 408.4kbits/s dup=0 drop=69 *** drop! frame= 255 fps=6.6 q=4.1 size= 1856kB time=00:00:36.42 bitrate= 417.5kbits/s dup=0 drop=70 *** drop! frame= 259 fps=6.6 q=2.1 size= 1858kB time=00:00:37.00 bitrate= 411.4kbits/s dup=0 drop=71 *** drop! frame= 263 fps=6.6 q=7.5 size= 1893kB time=00:00:37.57 bitrate= 412.8kbits/s dup=0 drop=72 *** drop! frame= 267 fps=6.6 q=14.2 size= 1926kB time=00:00:38.14 bitrate= 413.7kbits/s dup=0 drop=73 *** drop! frame= 271 fps=6.6 q=15.0 size= 1941kB time=00:00:38.71 bitrate= 410.8kbits/s dup=0 drop=74 *** drop! *** drop! frame= 274 fps=6.6 q=14.1 size= 1952kB time=00:00:39.14 bitrate= 408.6kbits/s dup=0 drop=76 *** drop! frame= 278 fps=6.6 q=14.1 size= 1983kB time=00:00:39.71 bitrate= 409.0kbits/s dup=0 drop=77 *** drop! frame= 282 fps=6.6 q=14.0 size= 1997kB time=00:00:40.28 bitrate= 406.1kbits/s dup=0 drop=78 *** drop! frame= 286 fps=6.6 q=13.7 size= 2010kB time=00:00:40.85 bitrate= 403.0kbits/s dup=0 drop=79 *** drop! frame= 290 fps=6.6 q=11.9 size= 2041kB time=00:00:41.42 bitrate= 403.7kbits/s dup=0 drop=80 *** drop! frame= 294 fps=6.6 q=13.5 size= 2056kB time=00:00:42.00 bitrate= 401.0kbits/s dup=0 drop=81 *** drop! frame= 298 fps=6.6 q=11.5 size= 2069kB time=00:00:42.57 bitrate= 398.1kbits/s dup=0 drop=82 *** drop! frame= 302 fps=6.6 q=11.6 size= 2101kB time=00:00:43.14 bitrate= 399.0kbits/s dup=0 drop=83 *** drop! frame= 306 fps=6.6 q=12.4 size= 2115kB time=00:00:43.71 bitrate= 396.4kbits/s dup=0 drop=84 *** drop! *** drop! frame= 309 fps=6.6 q=11.8 size= 2125kB time=00:00:44.14 bitrate= 394.4kbits/s dup=0 drop=86 *** drop! frame= 313 fps=6.6 q=4.5 size= 2159kB time=00:00:44.71 bitrate= 395.5kbits/s dup=0 drop=87 *** drop! frame= 317 fps=6.6 q=5.7 size= 2165kB time=00:00:45.28 bitrate= 391.7kbits/s dup=0 drop=88 *** drop! frame= 321 fps=6.6 q=6.7 size= 2182kB time=00:00:45.85 bitrate= 389.8kbits/s dup=0 drop=89 *** drop! frame= 325 fps=6.6 q=5.8 size= 2220kB time=00:00:46.42 bitrate= 391.7kbits/s dup=0 drop=90 *** drop! frame= 329 fps=6.6 q=5.8 size= 2227kB time=00:00:47.00 bitrate= 388.1kbits/s dup=0 drop=91 *** drop! frame= 333 fps=6.6 q=5.0 size= 2234kB time=00:00:47.57 bitrate= 384.8kbits/s dup=0 drop=92 *** drop! frame= 337 fps=6.6 q=3.2 size= 2294kB time=00:00:48.14 bitrate= 390.3kbits/s dup=0 drop=93 *** drop! --------------------------------------------------------------------------------------------------------------------------- ffplay: ffplay started on 2013-04-17 at 11:38:14 Report written to "ffplay-20130417-113814.log" Command line: "d:\\ffmpeg\\ffplay.exe" udp://224.1.1.1:1234 -probesize 50000 -analyzeduration 50 -report ffplay version N-46469-gc995644 Copyright (c) 2003-2012 the FFmpeg developers built on Nov 5 2012 17:58:15 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [mpegts @ 00000000020c22c0] Format mpegts probed with size=2048 and score=100 [mpegts @ 00000000020c22c0] Unable to seek back to the start [mpegts @ 00000000020c22c0] stream=0 stream_type=10 pid=100 prog_reg_desc= [mpegts @ 00000000020c22c0] File position before avformat_find_stream_info() is 16356 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 4 got 12 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 00000000020f1a60] hmm, seems the headers are not complete, trying to guess time_increment_bits [mpeg4 @ 00000000020f1a60] my guess is 3 bits ;) [mpegts @ 00000000020c22c0] invalid dts/pts combination 138857 [mpeg4 @ 00000000020f1a60] looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 151714 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 164571 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 177429 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 190286 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 203143 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 216000 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 228857 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] invalid dts/pts combination 241714 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] invalid dts/pts combination 254571 [mpeg4 @ 00000000020f1a60] header damaged [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 15 [mpegts @ 00000000020c22c0] invalid dts/pts combination 267429 [mpeg4 @ 00000000020f1a60] [IMGUTILS @ 0000000005d4f400] Picture size 0x0 is invalid [mpeg4 @ 00000000020f1a60] video_get_buffer: image parameters invalid [mpeg4 @ 00000000020f1a60] get_buffer() failed (-1 0 0000000000000000) [mpegts @ 00000000020c22c0] Probe buffer size limit 50000 reached [mpegts @ 00000000020c22c0] Estimating duration from bitrate, this may be inaccurate [mpegts @ 00000000020c22c0] File position after avformat_find_stream_info() is 140628 Input #0, mpegts, from 'udp://224.1.1.1:1234': Duration: N/A, start: 1.542856, bitrate: N/A Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100], 12, 1/90000: Video: mpeg4 (Simple Profile) ([16][0][0][0] / 0x0010), yuv420p, 400x640 [SAR 1:1 DAR 5:8], 7 fps, 7 tbr, 90k tbn, 7 tbc [mpeg4 @ 00000000020f1a60] detected 8 logical cores [mpeg4 @ 0000000005f98f60] warning: first frame is no keyframe [mpeg4 @ 0000000005f98f60] ac-tex damaged at 12 16 [mpeg4 @ 0000000005f98f60] Error at MB: 428 [mpeg4 @ 0000000005f98f60] Error at MB: 685 [mpeg4 @ 0000000005f98f60] concealing 395 DC, 395 AC, 395 MV errors in P frame Frame changed from size:0x0 to size:400x640 [buffer @ 00000000021343e0] Setting entry with key 'video_size' to value '400x640' [buffer @ 00000000021343e0] Setting entry with key 'pix_fmt' to value '0' [buffer @ 00000000021343e0] Setting entry with key 'time_base' to value '1/90000' [buffer @ 00000000021343e0] Setting entry with key 'pixel_aspect' to value '1/1' [ffplay_buffer @ 00000000020fffe0] w:400 h:640 pixfmt:yuv420p tb:1/90000 fr:0/1 sar:1/1 sws_param: [ffplay_crop @ 0000000002100080] w:400 h:640 sar:1/1 -> w:400 h:640 sar:1/1 [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 19 20 [mpeg4 @ 0000000005f9a0a0] Error at MB: 539 [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 18 26 [mpeg4 @ 0000000005f9a0a0] Error at MB: 694 [mpeg4 @ 0000000005f9a0a0] concealing 311 DC, 311 AC, 311 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 4 got 5 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 12 got 2 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 9 got 10 [mpeg4 @ 00000000020efbc0] ac-tex damaged at 23 14 [mpeg4 @ 00000000020efbc0] Error at MB: 387 [mpeg4 @ 00000000020efbc0] concealing 76 DC, 76 AC, 76 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 7 got 14 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 3 got 9 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 1 got 9 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 7 got 8 [mpeg4 @ 00000000020f1140] ac-tex damaged at 10 21 [mpeg4 @ 00000000020f1140] Error at MB: 556 [mpeg4 @ 00000000020f1140] ac-tex damaged at 4 26 [mpeg4 @ 00000000020f1140] Error at MB: 680 [mpeg4 @ 00000000020f1140] concealing 309 DC, 309 AC, 309 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 17 expected 7 got 8 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 6 got 14 [mpeg4 @ 0000000005f996a0] ac-tex damaged at 24 11 [mpeg4 @ 0000000005f996a0] Error at MB: 310 [mpeg4 @ 0000000005f996a0] concealing 150 DC, 150 AC, 150 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 13 [mpeg4 @ 00000000020efbc0] 1. marker bit missing in 3. esc [mpeg4 @ 00000000020efbc0] Error at MB: 169 [mpeg4 @ 00000000020efbc0] concealing 86 DC, 86 AC, 86 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 13 got 3 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 10 got 11 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 4 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 11 got 2 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 15 got 0 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 0 got 8 [mpeg4 @ 000000000210d500] ac-tex damaged at 15 22 [mpeg4 @ 000000000210d500] Error at MB: 587 [mpeg4 @ 000000000210d500] concealing 284 DC, 284 AC, 284 MV errors in P frame [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f98f60] ac-tex damaged at 23 15 [mpeg4 @ 0000000005f98f60] Error at MB: 413 [mpeg4 @ 0000000005f98f60] concealing 259 DC, 259 AC, 259 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 15 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 24 4 [mpeg4 @ 0000000005f996a0] Error at MB: 128 [mpeg4 @ 0000000005f996a0] concealing 127 DC, 127 AC, 127 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 9 got 10 [mpeg4 @ 0000000005f9a0a0] concealing 25 DC, 25 AC, 25 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 1 got 10 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 00000000020f1140] I cbpy damaged at 1 14 [mpeg4 @ 00000000020f1140] Error at MB: 365 [mpeg4 @ 00000000020f1140] concealing 98 DC, 98 AC, 98 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 2 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f9a0a0] ac-tex damaged at 18 33 [mpeg4 @ 0000000005f9a0a0] Error at MB: 876 [mpeg4 @ 0000000005f9a0a0] concealing 200 DC, 200 AC, 200 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 13 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 2 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 3 2 [mpeg4 @ 0000000005f996a0] Error at MB: 55 [mpeg4 @ 0000000005f996a0] 2. marker bit missing in 3. esc [mpeg4 @ 0000000005f996a0] Error at MB: 456 [mpeg4 @ 0000000005f996a0] concealing 396 DC, 396 AC, 396 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 8 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 2 got 10 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f98f60] I cbpy damaged at 16 18 [mpeg4 @ 0000000005f98f60] Error at MB: 484 [mpeg4 @ 0000000005f98f60] concealing 183 DC, 183 AC, 183 MV errors in I frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 8 got 15 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 1 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 11 got 3 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 000000000210d500] Error at MB: 198 [mpeg4 @ 000000000210d500] concealing 268 DC, 268 AC, 268 MV errors in P frame [mpeg4 @ 0000000005f98f60] illegal dc vlc [mpeg4 @ 0000000005f98f60] Error at MB: 83 [mpeg4 @ 0000000005f98f60] I cbpy damaged at 15 18 [mpeg4 @ 0000000005f98f60] Error at MB: 483 [mpeg4 @ 0000000005f98f60] concealing 353 DC, 353 AC, 353 MV errors in I frame [mpegts @ 00000000020c22c0] first_dts 138857 not matching first dts 2903143 in que [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 9 got 10 [mpeg4 @ 00000000020f1140] ac-tex damaged at 22 17 [mpeg4 @ 00000000020f1140] Error at MB: 464 [mpeg4 @ 00000000020f1140] concealing 600 DC, 600 AC, 600 MV errors in P frame [mpeg4 @ 000000000210d500] ac-tex damaged at 8 3 [mpeg4 @ 000000000210d500] Error at MB: 86 [mpeg4 @ 000000000210d500] concealing 1000 DC, 1000 AC, 1000 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 0 got 6 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f9a7e0] concealing 296 DC, 296 AC, 296 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 13 got 14 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 10 got 0 [mpegts @ 00000000020c22c0] PES packet size mismatch [mpeg4 @ 0000000005f996a0] ac-tex damaged at 9 9 [mpeg4 @ 0000000005f996a0] Error at MB: 243 [mpeg4 @ 0000000005f996a0] concealing 400 DC, 400 AC, 400 MV errors in P frame [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 11 got 12 [mpegts @ 00000000020c22c0] Continuity check failed for pid 256 expected 5 got 11 [mpegts @ 00000000020c22c0] Continuity check failed for pid 0 expected 6 got 7 -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/multicast-streaming-real-time-raw-rgb-images-tp4658524.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Wed Apr 17 18:40:50 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 16:40:50 +0000 (UTC) Subject: [FFmpeg-user] multicast streaming real time raw rgb images References: <1366210298888-4658524.post@n4.nabble.com> Message-ID: Jacobs gmail.com> writes: > 1. I cant get rid of a 1-2 seconds lag I have. Did you already test -probesize and -analyzeduration ? Your configure line still looks slightly broken.. Carl Eugen From jacobhameiri at gmail.com Wed Apr 17 18:56:39 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Wed, 17 Apr 2013 09:56:39 -0700 (PDT) Subject: [FFmpeg-user] multicast streaming real time raw rgb images In-Reply-To: References: <1366210298888-4658524.post@n4.nabble.com> Message-ID: <1366217799156-4658528.post@n4.nabble.com> I added -probesize and -analyzeduration to ffplay. is it also a good idea to add these params in the ffmpeg command ? in what part of my command ? what values ? I am also trying libx264 as I understand it should get me better streaming results but with no luck, I get a lagged result and many corrupted frames. Also, is there a good way to find out what part of the process is causing the delay ? and are there any buffers I can ( and should ) disable ? Again thanks for your help, I am on this for days now with no real progress. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/multicast-streaming-real-time-raw-rgb-images-tp4658526p4658528.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From llee040 at sbcglobal.net Wed Apr 17 19:48:55 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Wed, 17 Apr 2013 12:48:55 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? Message-ID: I've developed a highly refined method for applying detelecine using MEncoder's pullup filter methods. I get excellent results using that method for the examples I'm including in this message. However, for reasons that are probably obvious, I'm now trying to get comparable results using ffmpeg. How can I use the fieldmatch filter to detelecine these excerpts to get better results using ffmpeg? This source (for both source 1 and source 2) is 720P. I don't see any difference in my results whether or not I use decimate in the filtergraph as shown in the 8.22.2 Examples (http://ffmpeg.org/ffmpeg-filters.html#toc-Examples-3). I've also included a slightly longer mpg-muxed version (source 2) which concludes with the source 1 excerpt. I get different results when encoding the longer mpg-muxed excerpt (detelecine seems correct, however some areas are pixilated, but only when encoding the longer excerpt) Although the included filtergraph for my source 2 command uses the shorter of the two examples provided in the 8.22.2 section of the documentation, I've tried both and can't observe any difference in the output. Here's my source 1: https://dl.dropboxusercontent.com/u/42718751/Blockyclip.zip Here's my output 1: https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output.zip Here's my source 2 (mpg): https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip Here's my output 2: https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output_mpg.zip Here's the command used on source 1 and results (please find the command used on source 2 below). --------- ffmpeg -i 'blockyclip.mp4' -vf "fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced" -pix_fmt yuv420p -c:v libx264 -preset veryslow -tune film -qp 22 -r 23.976 -s 1280x720 -aspect 16:9 'Blockyclip_output.mp4' ffmpeg version N-37959-g20e99a9 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 16 2013 13:28:23 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-filter=fieldmatch --enable-libmp3lame --enable-gpl --enable-libfaac --enable-pthreads --enable-ffplay --enable-nonfree --enable-libx264 libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 56.101 / 3. 56.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9701819400] stream 0, timescale not set [h264 @ 0x7f9701830c00] mmco: unref short failure Last message repeated 1 times [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9701819400] Stream #2: not enough frames to estimate rate; consider increasing probesize Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'blockyclip.mp4': Metadata: major_brand : M4VH minor_version : 1 compatible_brands: M4VHmp42avc1isom creation_time : 2013-04-17 15:28:37 title : blockyclip artist : blockyclip media_type : 10 show : blockyclip network : genre : TV Shows date : 2013-04-13 Duration: 00:00:15.46, start: 0.000000, bitrate: 7475 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 7051 kb/s, 59.75 fps, 59.94 tbr, 59940 tbn, 119.88 tbc Metadata: creation_time : 2013-04-17 15:28:37 handler_name : ?Apple Video Media Handler Stream #0:1(eng): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1(side), fltp, 384 kb/s Metadata: creation_time : 2013-04-17 15:28:37 handler_name : ?Apple Sound Media Handler Stream #0:2: Video: mjpeg, yuvj420p, 512x512 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc [libx264 @ 0x7f9701831800] using SAR=1/1 [libx264 @ 0x7f9701831800] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7f9701831800] profile High, level 5.0 [libx264 @ 0x7f9701831800] 264 - core 130 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=22 ip_ratio=1.40 pb_ratio=1.30 aq=0 Output #0, mp4, to 'Blockyclip_output.mp4': Metadata: major_brand : M4VH minor_version : 1 compatible_brands: M4VHmp42avc1isom date : 2013-04-13 title : blockyclip artist : blockyclip media_type : 10 show : blockyclip network : genre : TV Shows encoder : Lavf55.2.100 Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 23.98 tbc Metadata: creation_time : 2013-04-17 15:28:37 handler_name : ?Apple Video Media Handler Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, 5.1, s16, 128 kb/s Metadata: creation_time : 2013-04-17 15:28:37 handler_name : ?Apple Sound Media Handler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (ac3 -> libfaac) Press [q] to stop, [?] for help [h264 @ 0x7f9702631400] mmco: unref short failure [h264 @ 0x7f9702631a00] mmco: unref short failure frame CRC mismatch1 q=24.0 size= 995kB time=00:00:14.46 bitrate= 563.7kbits/s dup=2 drop=527 frame= 372 fps= 20 q=-1.0 Lsize= 1219kB time=00:00:15.46 bitrate= 645.4kbits/s dup=2 drop=550 video:891kB audio:316kB subtitle:0 global headers:0kB muxing overhead 0.974321% [libx264 @ 0x7f9701831800] frame I:2 Avg QP:19.00 size: 8307 [libx264 @ 0x7f9701831800] frame P:97 Avg QP:22.00 size: 6083 [libx264 @ 0x7f9701831800] frame B:273 Avg QP:23.69 size: 1117 [libx264 @ 0x7f9701831800] consecutive B-frames: 2.2% 3.2% 10.5% 59.1% 17.5% 3.2% 0.0% 4.3% 0.0% [libx264 @ 0x7f9701831800] mb I I16..4: 92.3% 4.8% 2.9% [libx264 @ 0x7f9701831800] mb P I16..4: 8.0% 2.5% 1.0% P16..4: 9.4% 2.6% 1.4% 0.1% 0.0% skip:75.0% [libx264 @ 0x7f9701831800] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 7.7% 0.6% 0.1% direct: 0.9% skip:90.5% L0:53.0% L1:39.3% BI: 7.8% [libx264 @ 0x7f9701831800] 8x8 transform intra:19.8% inter:59.9% [libx264 @ 0x7f9701831800] direct mvs spatial:97.4% temporal:2.6% [libx264 @ 0x7f9701831800] coded y,uvDC,uvAC intra: 18.8% 28.3% 9.9% inter: 2.2% 3.6% 0.2% [libx264 @ 0x7f9701831800] i16 v,h,dc,p: 76% 16% 2% 6% [libx264 @ 0x7f9701831800] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 13% 26% 5% 9% 12% 9% 6% 8% [libx264 @ 0x7f9701831800] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 42% 8% 3% 5% 6% 6% 4% 7% [libx264 @ 0x7f9701831800] i8c dc,h,v,p: 64% 20% 11% 5% [libx264 @ 0x7f9701831800] Weighted P-Frames: Y:5.2% UV:3.1% [libx264 @ 0x7f9701831800] ref P L0: 53.5% 8.6% 20.4% 5.3% 3.4% 2.5% 1.7% 1.0% 0.7% 0.6% 0.5% 0.5% 0.4% 0.4% 0.4% 0.3% [libx264 @ 0x7f9701831800] ref B L0: 84.2% 9.4% 3.1% 1.2% 0.7% 0.4% 0.3% 0.1% 0.1% 0.1% 0.2% 0.1% 0.1% 0.0% 0.0% [libx264 @ 0x7f9701831800] ref B L1: 97.0% 3.0% [libx264 @ 0x7f9701831800] kb/s:470.08 --------- Here's the command used on source 2: --------- ffmpeg -i 'Blockyclip_mpg.mpg' -vf "fieldmatch=order=tff:combmatch=none" -pix_fmt yuv420p -c:v libx264 -preset veryslow -tune film -qp 22 -r 23.976 -s 1280x720 -aspect 16:9 'Blockyclip_output_mpg.mp4' ffmpeg version N-37959-g20e99a9 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 16 2013 13:28:23 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-filter=fieldmatch --enable-libmp3lame --enable-gpl --enable-libfaac --enable-pthreads --enable-ffplay --enable-nonfree --enable-libx264 libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 56.101 / 3. 56.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 0x7fc6f4021a00] mmco: unref short failure Last message repeated 1 times Input #0, mpegts, from 'Blockyclip_mpg.mpg': Duration: 02:44:21.62, start: 1.404767, bitrate: 12513 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (Main) (HDMV / 0x564D4448), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s [libx264 @ 0x7fc6f401a000] using SAR=1/1 [libx264 @ 0x7fc6f401a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7fc6f401a000] profile High, level 5.0 [libx264 @ 0x7fc6f401a000] 264 - core 130 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=22 ip_ratio=1.40 pb_ratio=1.30 aq=0 Output #0, mp4, to '/Blockyclip_output_mpg.mp4': Metadata: encoder : Lavf55.2.100 Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 23.98 tbc Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, 5.1, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (ac3 -> libfaac) Press [q] to stop, [?] for help [h264 @ 0x7fc6f5112000] mmco: unref short failure [h264 @ 0x7fc6f5113800] mmco: unref short failure frame= 6144 fps=9.9 q=-1.0 Lsize= 53333kB time=00:04:16.32 bitrate=1704.5kbits/s dup=0 drop=9209 video:47140kB audio:6027kB subtitle:0 global headers:0kB muxing overhead 0.312380% [libx264 @ 0x7fc6f401a000] frame I:58 Avg QP:19.00 size: 32860 [libx264 @ 0x7fc6f401a000] frame P:1451 Avg QP:22.00 size: 15946 [libx264 @ 0x7fc6f401a000] frame B:4635 Avg QP:23.73 size: 5011 [libx264 @ 0x7fc6f401a000] consecutive B-frames: 2.5% 3.9% 12.8% 21.9% 24.4% 25.6% 5.0% 3.1% 0.7% [libx264 @ 0x7fc6f401a000] mb I I16..4: 43.3% 47.9% 8.8% [libx264 @ 0x7fc6f401a000] mb P I16..4: 7.7% 10.2% 1.5% P16..4: 30.2% 7.1% 5.0% 0.3% 0.1% skip:37.9% [libx264 @ 0x7fc6f401a000] mb B I16..4: 0.4% 0.8% 0.2% B16..8: 28.3% 3.5% 0.7% direct: 2.8% skip:63.3% L0:42.9% L1:48.8% BI: 8.3% [libx264 @ 0x7fc6f401a000] 8x8 transform intra:52.5% inter:73.8% [libx264 @ 0x7fc6f401a000] direct mvs spatial:99.8% temporal:0.2% [libx264 @ 0x7fc6f401a000] coded y,uvDC,uvAC intra: 45.8% 56.1% 19.9% inter: 9.0% 15.6% 0.4% [libx264 @ 0x7fc6f401a000] i16 v,h,dc,p: 49% 25% 8% 18% [libx264 @ 0x7fc6f401a000] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 14% 12% 10% 7% 12% 12% 12% 10% 11% [libx264 @ 0x7fc6f401a000] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 22% 8% 5% 10% 10% 10% 7% 11% [libx264 @ 0x7fc6f401a000] i8c dc,h,v,p: 48% 26% 18% 7% [libx264 @ 0x7fc6f401a000] Weighted P-Frames: Y:5.0% UV:2.7% [libx264 @ 0x7fc6f401a000] ref P L0: 51.4% 16.3% 12.7% 4.5% 3.1% 2.6% 1.9% 1.4% 1.0% 1.0% 0.8% 0.9% 0.6% 0.7% 0.6% 0.5% [libx264 @ 0x7fc6f401a000] ref B L0: 84.2% 7.1% 2.9% 1.3% 1.0% 0.8% 0.7% 0.4% 0.3% 0.3% 0.3% 0.2% 0.2% 0.2% 0.1% [libx264 @ 0x7fc6f401a000] ref B L1: 95.1% 4.9% [libx264 @ 0x7fc6f401a000] kb/s:1506.95 --------- Thanks. Laine Lee From sullafelix at ymail.com Wed Apr 17 20:47:06 2013 From: sullafelix at ymail.com (Mark Haney) Date: Wed, 17 Apr 2013 11:47:06 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg 1.0.5 and buffer underflow Message-ID: <1366224426.47288.androidMobile@web122303.mail.ne1.yahoo.com> I'll do that. Thanks. Sent from Yahoo! Mail on Android From onemda at gmail.com Wed Apr 17 22:01:17 2013 From: onemda at gmail.com (Paul B Mahol) Date: Wed, 17 Apr 2013 20:01:17 +0000 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: References: Message-ID: On 4/17/13, L. Lee wrote: > I've developed a highly refined method for applying detelecine using > MEncoder's pullup filter methods. I get excellent results using that > method for the examples I'm including in this message. However, for > reasons that are probably obvious, I'm now trying to get comparable > results using ffmpeg. FYI, ffmpeg have mp=pullup filter too From andyqos at ukfsn.org Wed Apr 17 22:56:59 2013 From: andyqos at ukfsn.org (Andy Furniss) Date: Wed, 17 Apr 2013 21:56:59 +0100 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: References: Message-ID: <516F0C9B.6000001@ukfsn.org> L. Lee wrote: > I've developed a highly refined method for applying detelecine using > MEncoder's pullup filter methods. I get excellent results using that > method for the examples I'm including in this message. However, for > reasons that are probably obvious, I'm now trying to get comparable > results using ffmpeg. > > How can I use the fieldmatch filter to detelecine these excerpts to get > better results using ffmpeg? This source (for both source 1 and source 2) > is 720P. I don't know the answer and have no experience of this, but if the input is progressive then I guess fieldmatch isn't going to find any fields. From andyqos at ukfsn.org Wed Apr 17 23:57:18 2013 From: andyqos at ukfsn.org (Andy Furniss) Date: Wed, 17 Apr 2013 22:57:18 +0100 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: References: Message-ID: <516F1ABE.80407@ukfsn.org> L. Lee wrote: > I've developed a highly refined method for applying detelecine using > MEncoder's pullup filter methods. I get excellent results using that > method for the examples I'm including in this message. However, for > reasons that are probably obvious, I'm now trying to get comparable > results using ffmpeg. > > How can I use the fieldmatch filter to detelecine these excerpts to get > better results using ffmpeg? This source (for both source 1 and source 2) > is 720P. I don't see any difference in my results whether or not I use > decimate in the filtergraph as shown in the 8.22.2 Examples > (http://ffmpeg.org/ffmpeg-filters.html#toc-Examples-3). I've also included > a slightly longer mpg-muxed version (source 2) which concludes with the > source 1 excerpt. I get different results when encoding the longer > mpg-muxed excerpt (detelecine seems correct, however some areas are > pixilated, but only when encoding the longer excerpt) Although the > included filtergraph for my source 2 command uses the shorter of the two > examples provided in the 8.22.2 section of the documentation, I've tried > both and can't observe any difference in the output. > > Here's my source 1: > > https://dl.dropboxusercontent.com/u/42718751/Blockyclip.zip > > Here's my output 1: > > https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output.zip This looks wrong - 12fps with frames duped. From cehoyos at ag.or.at Thu Apr 18 00:36:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 17 Apr 2013 22:36:14 +0000 (UTC) Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? References: Message-ID: L. Lee sbcglobal.net> writes: > https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output.zip > https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip Both samples look progressive to me, I don't think the fieldmatch filter can have any effect on these files. What do you expect the filter to do? Please note that the pullup filter in FFmpeg currently does not work! Carl Eugen From timothygu99 at gmail.com Thu Apr 18 01:43:02 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Wed, 17 Apr 2013 16:43:02 -0700 Subject: [FFmpeg-user] "Libav" comments in ARM asm In-Reply-To: References: Message-ID: Hi, I found that here are many asm files in libavcodec/arm contains "This file is part of Libav." instead of other things for obvious reasons. For example, libavcodec/arm/neon.S. Please fix it. Regards, Timothy G. From Jonathan.Haws at sdl.usu.edu Thu Apr 18 01:44:21 2013 From: Jonathan.Haws at sdl.usu.edu (Jonathan Haws) Date: Wed, 17 Apr 2013 23:44:21 +0000 Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel Message-ID: Question: I am trying to benchmark the difference between VA-API accelerated decoding and non-accelerated. I have an H.264 encoded video that I am using. Here is what I have tried: Build latest from GIT with --enable-vaapi: ./configure --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-indev=jack --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect --enable-vaapi --enable-gpl make &&make install Then I run: ffmpeg -benchmark -i myfile.mp4 -f null - With htop I see that the ffmpeg threads are consuming no more than 1.3% of the CPU (totalling around 10% overall). Then I rebuild ffmpeg: ./configure --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-indev=jack --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect --enable-gpl make &&make install And rerun the same test. Then I see nearly the exact same output - CPU usage at about 10%. I am running this on an Intel i7 (Ivy Bridge) with the following installed: libva.x86_64 (1.1.0-5) libva-devel.x86_64 (1.1.0-5) libva-utils.x86_64 (1.1.0-5) libva-intel-driver.x86_64 (1.0.20-1) xorg-x11-drv-intel.x86_64 (2.21.5-1) xorg-x11-proto-devel.noarch (7.6-24) xorg-x11-server-Xorg.x86_64 (1.13.3-2) xorg-x11-server-common.x86_64 (1.13.3-2) xorg-x11-server-devel.x86_64 (1.13.3-2) xorg-x11-util-macros.noarch (1.17-2) xorg-x11-xkb-utils.x86_64 (7.7-5) mesa-dri-drivers.x86_64 (9.1-3) mesa-dri-filesystem.x86_64 (9.1-3) mesa-libEGL.x86_64 (9.1-3) mesa-libGL.x86_64 (9.1-3) mesa-libGL-devel.x86_64 (9.1-3) mesa-libGLU.x86_64 (9.0.0-1) mesa-libgbm.x86_64 (9.1-3) mesa-libglapi.x86_64 (9.1-3) Why can't I see a difference between the two? Shouldn't the CPU usage be lower for the accelerated build? What am I missing? Thanks! Jon From edwardsmarkf at gmail.com Thu Apr 18 02:29:19 2013 From: edwardsmarkf at gmail.com (mark edwards) Date: Wed, 17 Apr 2013 17:29:19 -0700 Subject: [FFmpeg-user] determining average audio level Message-ID: hello all - i am sorry to ask such a basic question but my brain is shot today. i need to determine if i have a totally blank audio file or not. apparently just looking at the file size is not an answer! if i wanted to use ffmpeg (or ffprobe?) to determine if there is any audio content there, what is the best way to do it? average audio level or something equally easy? thank you in advance for tolerating such a basic question. From cehoyos at ag.or.at Thu Apr 18 02:29:12 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 00:29:12 +0000 (UTC) Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel References: Message-ID: Jonathan Haws sdl.usu.edu> writes: > I am trying to benchmark the difference between > VA-API accelerated decoding and non-accelerated. I don't think you can do that with ffmpeg (the application). Carl Eugen From Jonathan.Haws at sdl.usu.edu Thu Apr 18 02:31:38 2013 From: Jonathan.Haws at sdl.usu.edu (Jonathan Haws) Date: Thu, 18 Apr 2013 00:31:38 +0000 Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel In-Reply-To: References: , Message-ID: <6dgxfdp4sowodym0dt0fe8pu.1366245098572@email.android.com> But doesn't ffmpeg use vaapi? What other options are there to do this? Sent from my smartphone - sorry for any typos! -------- Original message -------- From: Carl Eugen Hoyos Date: 04/17/2013 6:29 PM (GMT-07:00) To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] No visible difference between hwaccel and no hwaccel Jonathan Haws sdl.usu.edu> writes: > I am trying to benchmark the difference between > VA-API accelerated decoding and non-accelerated. I don't think you can do that with ffmpeg (the application). Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ubitux at gmail.com Thu Apr 18 02:34:43 2013 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 18 Apr 2013 02:34:43 +0200 Subject: [FFmpeg-user] determining average audio level In-Reply-To: References: Message-ID: <20130418003443.GG5653@leki> On Wed, Apr 17, 2013 at 05:29:19PM -0700, mark edwards wrote: > hello all - > > i am sorry to ask such a basic question but my brain is shot today. > > i need to determine if i have a totally blank audio file or not. > apparently just looking at the file size is not an answer! > > if i wanted to use ffmpeg (or ffprobe?) to determine if there is any audio > content there, what is the best way to do it? average audio level or > something equally easy? > Potential solutions here: https://www.ffmpeg.org/ffmpeg-filters.html#volumedetect https://www.ffmpeg.org/ffmpeg-filters.html#silencedetect https://www.ffmpeg.org/ffmpeg-filters.html#ebur128 [...] -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From cehoyos at ag.or.at Thu Apr 18 02:41:03 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 00:41:03 +0000 (UTC) Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel References: , <6dgxfdp4sowodym0dt0fe8pu.1366245098572@email.android.com> Message-ID: Jonathan Haws sdl.usu.edu> writes: > But doesn't ffmpeg use vaapi? ffmpeg (the application) does not use vaapi. > What other options are there to do this? I believe vlc supports VA-API using libavcodec. Please do not top-post here, it is considered rude. Carl Eugen From Jonathan.Haws at sdl.usu.edu Thu Apr 18 02:54:58 2013 From: Jonathan.Haws at sdl.usu.edu (Jonathan Haws) Date: Thu, 18 Apr 2013 00:54:58 +0000 Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel Message-ID: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> -------- Original message -------- From: Carl Eugen Hoyos Date: 04/17/2013 6:41 PM (GMT-07:00) To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] No visible difference between hwaccel and no hwaccel >ffmpeg (the application) does not use vaapi. Then what does --enable-vaapi do if not enable VAAPI support in ffmpeg? >I believe vlc supports VA-API using libavcodec. I am running on a headless system so I want to pipe the output to null (basically benchmark just the decode path). Unless vlc allows for that then I don't think it will work for what I am trying to do. >Please do not top-post here, it is considered rude. Sorry, didn't think about that when replying from my phone. From cehoyos at ag.or.at Thu Apr 18 03:04:23 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 01:04:23 +0000 (UTC) Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel References: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> Message-ID: Jonathan Haws sdl.usu.edu> writes: > >ffmpeg (the application) does not use vaapi. > > Then what does --enable-vaapi do if not enable > VAAPI support in ffmpeg? (It does nothing because it is auto-detected.) It enables VA-API support in libavcodec (to allow applications like vlc and xbmc to use it). Carl Eugen From edwardsmarkf at gmail.com Thu Apr 18 03:06:55 2013 From: edwardsmarkf at gmail.com (mark edwards) Date: Wed, 17 Apr 2013 18:06:55 -0700 Subject: [FFmpeg-user] determining average audio level In-Reply-To: <20130418003443.GG5653@leki> References: <20130418003443.GG5653@leki> Message-ID: awesome! thank you clement! /usr/local/bin/ffmpeg -i /blah/blah/clement.flv -af volumedetect -f null - ; ### volume From pulche at gmail.com Thu Apr 18 07:13:19 2013 From: pulche at gmail.com (Jeeyoung Byun) Date: Thu, 18 Apr 2013 14:13:19 +0900 Subject: [FFmpeg-user] Questions to developers.. Message-ID: Usually developers write the codes by themselves directly referencing the technical specification? Or if some existing open sources or non-official codes are available, then they sometimes refer the codes? Actually, I have some legal issues using Dolby AC-3 codec from FFmpeg. Does anyone know that how the "FFmpeg AC-3" was developed in a sense of the references? Referencing the specification directly? Or referencing the existed unofficial but available source codes from Dolby? Oh, absolutely no offense to developers.. :) Thanks, Jenny From ryan.lindsay at gmail.com Thu Apr 18 09:04:23 2013 From: ryan.lindsay at gmail.com (ryan lindsay) Date: Thu, 18 Apr 2013 17:04:23 +1000 Subject: [FFmpeg-user] re-streaming rtsp on arm Message-ID: Hi I am trying to take an rtsp stream and send it to ffserver, and have that transcode it to webm I have got the version 1.2 from the website and compiled from scratch on my beagleboard and it's running ubuntu 12 When I run ffmpeg I get root at arm:~/ffmpeg-1.2# ffmpeg -i rtsp:// 192.168.1.45:8557/PSIA/Streaming/channels/2?videoCodecType=H.264 -f alsa -ac 2 -i hw:0 http://localhost:8090/feed1.ffmffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1, Copyright (c) 2000-2013 the Libav developers built on Apr 2 2013 19:45:31 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] non-existing PPS referenced [h264 @ 0x1d62980] non-existing PPS 0 referenced [h264 @ 0x1d62980] decode_slice_header error [h264 @ 0x1d62980] no frame! [h264 @ 0x1d62980] left block unavailable for requested intra mode at 0 21 [h264 @ 0x1d62980] error while decoding MB 0 21, bytestream (13442) [h264 @ 0x1d62980] concealing 2929 DC, 2929 AC, 2929 MV errors [rtsp @ 0x1d5eae0] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 60.00 (60/1) -> 1000.00 (1000/1) Input #0, rtsp, from 'rtsp:// 192.168.1.45:8557/PSIA/Streaming/channels/2?videoCodecType=H.264': Metadata: title : RTSP/RTP stream from IPNC comment : PSIA/Streaming/channels/2?videoCodecType=H.264 Duration: N/A, start: 0.064000, bitrate: N/A Stream #0.0: Video: h264 (High), yuvj420p, 1920x720, 1k tbr, 90k tbn, 60 tbc [alsa @ 0x1d61280] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'hw:0': Duration: N/A, start: 612993.999981, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible sample format '(null)' for codec 'libvorbis', auto-selecting format 's16' [buffer @ 0x1d721c0] w:1920 h:720 pixfmt:yuvj420p [avsink @ 0x1d5b6a0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x1d5a920] w:1920 h:720 fmt:yuvj420p -> w:1920 h:720 fmt:yuv420p flags:0x4 Segmentation fault I'm a little confused why it segfaults, and also wondering what the Seems stream 0 codec frame rate differs from container frame rate: 60.00 (60/1) -> 1000.00 (1000/1) refers to I have set the frame rate to 30 in the ffserver con-fig, because I know the frame rate coming from the camera is 30fps Is the segfault related?? From asorti6 at gmail.com Thu Apr 18 09:49:41 2013 From: asorti6 at gmail.com (Owen Jones) Date: Thu, 18 Apr 2013 08:49:41 +0100 Subject: [FFmpeg-user] channelsplit seems to mute one channel In-Reply-To: References: <7FEC3D2C-458C-4A5A-AE32-2D9478DC8078@gmail.com> <9AE0B411-5B92-4E1C-A0C5-82B3529FCE45@gmail.com> Message-ID: <4C70C419-054D-4EB9-8879-340EF1E2BA29@gmail.com> On 16 Apr 2013, at 10:07, Carl Eugen Hoyos wrote: > You can either revert 981cf0c or search for the > line "avio_wb24(pb, track->secondary ? 0x2 : 0xf);" > in libavformat/movenc.c and make is always write > "0xF" or - ideally - as a temporary workaround, > you add a new flag to movenc.c to force all > tracks as primary: See the "AVOption"s on top of > the file and MOVMuxContext in libavformat/movenc.h > (and don't forget to send the result to > ffmpeg-devel). Thanks very much. I will do a quick fix for my local version and then have a go at creating a patch. Quite a few years since I've done much coding so may take a bit to get up to speed. Owen From besucher80 at gmx.de Thu Apr 18 12:00:40 2013 From: besucher80 at gmx.de (Stefan Kruse) Date: Thu, 18 Apr 2013 12:00:40 +0200 Subject: [FFmpeg-user] Problem with ffmpeg frames Message-ID: <000c01ce3c1b$95f380b0$c1da8210$@de> Hi, i try to use ffmpeg to make thumbnails from images. I try it on server with: shell_exec( 'ffmpeg -i '.$inputVideoPath.' -ss 00:00:00.00 -f image2 -vframes 1 '.$outputVideoImage ); But then comes the error: Ff_frame_encoder_init failed What does this mean. Thanks for helping. Stefan From cehoyos at ag.or.at Thu Apr 18 12:02:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 10:02:05 +0000 (UTC) Subject: [FFmpeg-user] Problem with ffmpeg frames References: <000c01ce3c1b$95f380b0$c1da8210$@de> Message-ID: Stefan Kruse gmx.de> writes: > shell_exec( 'ffmpeg -i '.$inputVideoPath.' -ss 00:00:00.00 > -f image2 -vframes 1 '.$outputVideoImage ); Instead of using variables, please post an actual (working) command line that does not contain variables and please post the complete, uncut console output (not just one line). Carl Eugen From cehoyos at ag.or.at Thu Apr 18 12:10:33 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 10:10:33 +0000 (UTC) Subject: [FFmpeg-user] re-streaming rtsp on arm References: Message-ID: ryan lindsay gmail.com> writes: > ffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1 This is not FFmpeg but an intentionally broken fork of it. It contains several hundred (known, user reported) bugs that are not reproducible with FFmpeg, some of them security relevant. Please understand that we cannot support this broken version. Please see http://ffmpeg.org/download.html for supported versions, if you are not a distributor, current git head is always recommended. Carl Eugen From cehoyos at ag.or.at Thu Apr 18 12:12:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 10:12:55 +0000 (UTC) Subject: [FFmpeg-user] Questions to developers.. References: Message-ID: Jeeyoung Byun gmail.com> writes: > Actually, I have some legal issues using > Dolby AC-3 codec from FFmpeg. Since no FFmpeg developer is a lawyer (afaik), this is not the right place to ask. But if you have any indication that something about the AC-3 codec source code is different from the rest of FFmpeg, please feel free to share your information. Carl Eugen From besucher80 at gmx.de Thu Apr 18 12:19:21 2013 From: besucher80 at gmx.de (Stefan Kruse) Date: Thu, 18 Apr 2013 12:19:21 +0200 Subject: [FFmpeg-user] Problem with ffmpeg frames In-Reply-To: References: <000c01ce3c1b$95f380b0$c1da8210$@de> Message-ID: <001101ce3c1e$31fa8470$95ef8d50$@de> Okay, Here is what I entered on shell: ./ffmpeg -i smk-ebay.avi -ss 00:00:00 -f image2 -vframes 1 test.jpg And the console output is the follow: ffmpeg version 1.1.git Copyright (c) 2000-2013 the FFmpeg developers built on Apr 18 2013 10:53:59 with gcc 4.4.5 (Gentoo 4.4.5 p1.3, pie-0.4.5) configuration: --prefix=XYZ --disable-yasm libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 56.102 / 3. 56.102 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 Input #0, avi, from 'smk-ebay.avi': Duration: 00:03:57.88, start: 0.000000, bitrate: 196 kb/s Stream #0:0: Video: h264 (High) (H264 / 0x34363248), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: aac ([255][0][0][0] / 0x00FF), 44100 Hz, mono, fltp, 64 kb/s [mjpeg @ 0x8fecca0] ff_frame_thread_encoder_init failed Output #0, image2, to 'test.jpg': Stream #0:0: Video: mjpeg, yuvj420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (h264 -> mjpeg) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height Hope this helps. Because an working example I cant give. But maybe this helps. Thanks Stefan -----Urspr?ngliche Nachricht----- Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Carl Eugen Hoyos Gesendet: Donnerstag, 18. April 2013 12:02 An: ffmpeg-user at ffmpeg.org Betreff: Re: [FFmpeg-user] Problem with ffmpeg frames Stefan Kruse gmx.de> writes: > shell_exec( 'ffmpeg -i '.$inputVideoPath.' -ss 00:00:00.00 -f image2 > -vframes 1 '.$outputVideoImage ); Instead of using variables, please post an actual (working) command line that does not contain variables and please post the complete, uncut console output (not just one line). Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Thu Apr 18 12:28:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 10:28:22 +0000 (UTC) Subject: [FFmpeg-user] Problem with ffmpeg frames References: <000c01ce3c1b$95f380b0$c1da8210$@de> <001101ce3c1e$31fa8470$95ef8d50$@de> Message-ID: Stefan Kruse gmx.de> writes: > ./ffmpeg -i smk-ebay.avi -ss 00:00:00 > -f image2 -vframes 1 test.jpg (It will probably work with -threads 1, but the interesting question is why it fails.) > ffmpeg version 1.1.git Copyright (c) 2000-2013 the FFmpeg developers > built on Apr 18 2013 10:53:59 with gcc 4.4.5 (Gentoo 4.4.5 p1.3, > pie-0.4.5) > configuration: --prefix=XYZ --disable-yasm > libavutil 52. 26.100 / 52. 26.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 2.100 / 55. 2.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 56.102 / 3. 56.102 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > Input #0, avi, from 'smk-ebay.avi': > Duration: 00:03:57.88, start: 0.000000, bitrate: 196 kb/s > Stream #0:0: Video: h264 (High) (H264 / 0x34363248), yuv420p, 640x480 > [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 50 tbc > [mjpeg 0x8fecca0] ff_frame_thread_encoder_init failed (I have not seen something similar yet.) Does this only happen for smk-ebay.avi or for every input file? Is something special about your system? No memory? No pthreads? Hundreds of CPUs (although that shouldn't be a problem)? Carl Eugen From besucher80 at gmx.de Thu Apr 18 12:49:53 2013 From: besucher80 at gmx.de (Stefan Kruse) Date: Thu, 18 Apr 2013 12:49:53 +0200 Subject: [FFmpeg-user] Problem with ffmpeg frames In-Reply-To: References: <000c01ce3c1b$95f380b0$c1da8210$@de> <001101ce3c1e$31fa8470$95ef8d50$@de> Message-ID: <001201ce3c22$7613b240$623b16c0$@de> Hi Carl, theres no special to my system. And yes, this happens to all input files. Maybe a missing plugin? Regards Stefan -----Urspr?ngliche Nachricht----- Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Carl Eugen Hoyos Gesendet: Donnerstag, 18. April 2013 12:28 An: ffmpeg-user at ffmpeg.org Betreff: Re: [FFmpeg-user] Problem with ffmpeg frames Stefan Kruse gmx.de> writes: > ./ffmpeg -i smk-ebay.avi -ss 00:00:00 -f image2 -vframes 1 test.jpg (It will probably work with -threads 1, but the interesting question is why it fails.) > ffmpeg version 1.1.git Copyright (c) 2000-2013 the FFmpeg developers > built on Apr 18 2013 10:53:59 with gcc 4.4.5 (Gentoo 4.4.5 p1.3, > pie-0.4.5) > configuration: --prefix=XYZ --disable-yasm > libavutil 52. 26.100 / 52. 26.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 2.100 / 55. 2.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 56.102 / 3. 56.102 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > Input #0, avi, from 'smk-ebay.avi': > Duration: 00:03:57.88, start: 0.000000, bitrate: 196 kb/s > Stream #0:0: Video: h264 (High) (H264 / 0x34363248), yuv420p, > 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 50 tbc > [mjpeg 0x8fecca0] ff_frame_thread_encoder_init failed (I have not seen something similar yet.) Does this only happen for smk-ebay.avi or for every input file? Is something special about your system? No memory? No pthreads? Hundreds of CPUs (although that shouldn't be a problem)? Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Thu Apr 18 12:57:07 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 10:57:07 +0000 (UTC) Subject: [FFmpeg-user] Problem with ffmpeg frames References: <000c01ce3c1b$95f380b0$c1da8210$@de> <001101ce3c1e$31fa8470$95ef8d50$@de> <001201ce3c22$7613b240$623b16c0$@de> Message-ID: Stefan Kruse gmx.de> writes: > theres no special to my system. And yes, this > happens to all input files. > > Maybe a missing plugin? No. You can try to recompile after "make distclean", you can try with a different (newer) compiler or you could add a few printfs in ff_frame_thread_encoder_init() in libavcodec/frame_thread_encoder.c to find out what operation fails (av_fifo_alloc, avcodec_open2 avcodec_alloc_context3 or pthread_create). Does -loglevel 99 show any additional information? Please avoid top-posting here, it is considered rude. Carl Eugen From guillaume at apiculture-warre.fr Thu Apr 18 14:45:04 2013 From: guillaume at apiculture-warre.fr (Guillaume Fontaine) Date: Thu, 18 Apr 2013 14:45:04 +0200 Subject: [FFmpeg-user] .MTS (sony) file with LPCM song Message-ID: <516FEAD0.2040907@apiculture-warre.fr> Hello, I have a HD sony camera (so it write the video into a MTS file). when I choose the song format LPCM (Linear PCM -> it's a professional camera) ffmpeg seems to not see the song... (but with the dolby digital, it see the song). I want to use my file in kdenlive (on ubuntu 12.10 and in few days 13.04) who use ffmpeg... My others softwares don't see to my song (winFF, totem, vlc, openshot video) except xine. Totem tell me : "Impossible to find the plugin Python (v2.7) need to install the plugin to read multimedia file of type : audio/x-private-ts-lpcm" Do you have any ideas to read my video with LPCM song ? I forgot to tell you my version of ffmpeg : "ffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1, Copyright (c) 2000-2013 the Libav developers built on Apr 2 2013 17:07:34 with gcc 4.7.2" best regards Guillaume From cehoyos at ag.or.at Thu Apr 18 14:54:56 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 12:54:56 +0000 (UTC) Subject: [FFmpeg-user] .MTS (sony) file with LPCM song References: <516FEAD0.2040907@apiculture-warre.fr> Message-ID: Guillaume Fontaine writes: > I have a HD sony camera (so it write the video into > a MTS file). when I choose the song format LPCM > (Linear PCM -> it's a professional camera) ffmpeg > seems to not see the song. Consider providing a sample. > "ffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1 This is not FFmpeg but an intentionally broken version that contains several hundred user reported bugs that are not reproducible with FFmpeg, some of them security relevant, please understand that we cannot support this fork. Please see http://ffmpeg.org/download.html for supported versions, current git head is recommended if you are not a distributor yourself. Carl Eugen From cehoyos at ag.or.at Thu Apr 18 14:59:35 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 12:59:35 +0000 (UTC) Subject: [FFmpeg-user] (no subject) References: Message-ID: Timothy Gu gmail.com> writes: > I found that here are many asm files in > libavcodec/arm contains > > "This file is part of Libav." The files were written for Libav, some people believe that changing this line would violate copyrights. IANAL, Carl Eugen From cehoyos at ag.or.at Thu Apr 18 15:02:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 13:02:05 +0000 (UTC) Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel References: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> Message-ID: Carl Eugen Hoyos ag.or.at> writes: > > Then what does --enable-vaapi do if not enable > > VAAPI support in ffmpeg? > (It does nothing because it is auto-detected.) To clarify: It used to be auto-detected for several years, I consider it a regression (bug) that it is not auto-detecting currently. > It enables VA-API support in libavcodec (to allow > applications like vlc and xbmc to use it). That should still be true. Carl Eugen From h.reindl at thelounge.net Thu Apr 18 15:12:15 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 18 Apr 2013 15:12:15 +0200 Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel In-Reply-To: References: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> Message-ID: <516FF12F.60509@thelounge.net> Am 18.04.2013 15:02, schrieb Carl Eugen Hoyos: > Carl Eugen Hoyos ag.or.at> writes: > >>> Then what does --enable-vaapi do if not enable >>> VAAPI support in ffmpeg? > >> (It does nothing because it is auto-detected.) > > To clarify: It used to be auto-detected for > several years, I consider it a regression (bug) > that it is not auto-detecting currently. > >> It enables VA-API support in libavcodec (to allow >> applications like vlc and xbmc to use it). > > That should still be true however, you need to install "libva-devel" or however the devel-packages are named on the used distribution -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From llee040 at sbcglobal.net Wed Apr 17 23:57:13 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Wed, 17 Apr 2013 16:57:13 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: Message-ID: On 4/17/13 3:01 PM, "Paul B Mahol" wrote: >On 4/17/13, L. Lee wrote: >> I've developed a highly refined method for applying detelecine using >> MEncoder's pullup filter methods. I get excellent results using that >> method for the examples I'm including in this message. However, for >> reasons that are probably obvious, I'm now trying to get comparable >> results using ffmpeg. > >FYI, ffmpeg have mp=pullup filter too Thanks, Paul, but there are additional mplayer features needed to fine tune the pullup process in most situations to make it work with any degree of reliability, and not all of those are available in ffmpeg, I think. Laine Lee From cehoyos at ag.or.at Thu Apr 18 15:30:49 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 13:30:49 +0000 (UTC) Subject: [FFmpeg-user] multicast streaming real time raw rgb images References: <1366210298888-4658524.post@n4.nabble.com> <1366217799156-4658528.post@n4.nabble.com> Message-ID: Jacobs gmail.com> writes: > Also, is there a good way to find out what part of the > process is causing the delay ? Yes, by testing -analyzeduration and -probesize, you are eliminating possible reasons for the delay. > and are there any buffers I can ( and should ) disable ? Yes, -analyzeduration and -probesize both have an impact on the used buffers. Carl Eugen From cehoyos at ag.or.at Thu Apr 18 15:33:47 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 13:33:47 +0000 (UTC) Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? References: Message-ID: L. Lee sbcglobal.net> writes: > there are additional mplayer features needed to fine > tune the pullup process in most situations to make it > work with any degree of reliability, and not all of > those are available in ffmpeg, I think. You can use the exact same options (I suspect that is what you mean with "features") with pullup in FFmpeg as in MPlayer but I would expect A/V-desync if you use mp=pullup. Carl Eugen From Jonathan.Haws at sdl.usu.edu Thu Apr 18 15:38:18 2013 From: Jonathan.Haws at sdl.usu.edu (Jonathan Haws) Date: Thu, 18 Apr 2013 13:38:18 +0000 Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel In-Reply-To: <516FF12F.60509@thelounge.net> References: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> <516FF12F.60509@thelounge.net> Message-ID: >>>> Then what does --enable-vaapi do if not enable VAAPI support in >>>> ffmpeg? >> >>> (It does nothing because it is auto-detected.) >> >> To clarify: It used to be auto-detected for several years, I consider >> it a regression (bug) that it is not auto-detecting currently. >> >>> It enables VA-API support in libavcodec (to allow applications like >>> vlc and xbmc to use it). > >> That should still be true > >however, you need to install "libva-devel" or however the devel-packages are named on the used distribution Okay, so what I am gathering is that if I want to do this benchmarking, I will need to write my own app using libavcodec that will do the decoding. I can then compile libavcodec with --enable-vaapi, build my program, benchmark; then repeay without --enable-vaapi. Is that correct? Any leads on where to start? Are there some stripped down examples of libavcodec? Thanks! Jon From guillaume at apiculture-warre.fr Thu Apr 18 15:48:15 2013 From: guillaume at apiculture-warre.fr (Guillaume Fontaine) Date: Thu, 18 Apr 2013 15:48:15 +0200 Subject: [FFmpeg-user] .MTS (sony) file with LPCM song In-Reply-To: References: <516FEAD0.2040907@apiculture-warre.fr> Message-ID: <516FF99F.9080102@apiculture-warre.fr> Hey ! It's a problem of this fork... With the version of Jon Severinsson's FFmpeg PPA (link from http://ffmpeg.org/download.html) my song appear with kdenlive ! Thanks Guillaume Le 18/04/2013 14:54, Carl Eugen Hoyos a ?crit : > Guillaume Fontaine writes: > >> I have a HD sony camera (so it write the video into >> a MTS file). when I choose the song format LPCM >> (Linear PCM -> it's a professional camera) ffmpeg >> seems to not see the song. > Consider providing a sample. > >> "ffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1 > This is not FFmpeg but an intentionally broken version > that contains several hundred user reported bugs that > are not reproducible with FFmpeg, some of them security > relevant, please understand that we cannot support this > fork. > Please see http://ffmpeg.org/download.html for supported > versions, current git head is recommended if you are not > a distributor yourself. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Thu Apr 18 15:57:44 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 13:57:44 +0000 (UTC) Subject: [FFmpeg-user] No visible difference between hwaccel and no hwaccel References: <6oxkhqn37uy9ds8dei6hw631.1366246216238@email.android.com> <516FF12F.60509@thelounge.net> Message-ID: Jonathan Haws sdl.usu.edu> writes: > Okay, so what I am gathering is that if I want to do > this benchmarking, I will need to write my own app > using libavcodec that will do the decoding. It may be simpler to test with one of the existing applications. Carl Eugen From guillaume at apiculture-warre.fr Thu Apr 18 16:23:19 2013 From: guillaume at apiculture-warre.fr (Guillaume Fontaine) Date: Thu, 18 Apr 2013 16:23:19 +0200 Subject: [FFmpeg-user] .MTS (sony) file with LPCM song In-Reply-To: <516FF99F.9080102@apiculture-warre.fr> References: <516FEAD0.2040907@apiculture-warre.fr> <516FF99F.9080102@apiculture-warre.fr> Message-ID: <517001D7.1030501@apiculture-warre.fr> So now, I have some softwares who can read my file (kdenlive, winFF, xine, ffmpeg) and some others not (totem, vlc). I imagine that vlc and totem use libraries of libav and not of ffmpeg. Do you know how to correct it and have totem and vlc who can read my file ? A sample of MTS file with LPCM song is on : http://dl.free.fr/uPa683LrZ Guillaume Fontaine Le 18/04/2013 15:48, Guillaume Fontaine a ?crit : > Hey ! > > It's a problem of this fork... > With the version of Jon Severinsson's FFmpeg PPA (link from > http://ffmpeg.org/download.html) my song appear with kdenlive ! > > Thanks > > Guillaume > > Le 18/04/2013 14:54, Carl Eugen Hoyos a ?crit : >> Guillaume Fontaine writes: >> >>> I have a HD sony camera (so it write the video into >>> a MTS file). when I choose the song format LPCM >>> (Linear PCM -> it's a professional camera) ffmpeg >>> seems to not see the song. >> Consider providing a sample. >> >>> "ffmpeg version 0.8.6-6:0.8.6-0ubuntu0.12.10.1 >> This is not FFmpeg but an intentionally broken version >> that contains several hundred user reported bugs that >> are not reproducible with FFmpeg, some of them security >> relevant, please understand that we cannot support this >> fork. >> Please see http://ffmpeg.org/download.html for supported >> versions, current git head is recommended if you are not >> a distributor yourself. >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From timothygu99 at gmail.com Thu Apr 18 16:31:25 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Thu, 18 Apr 2013 07:31:25 -0700 Subject: [FFmpeg-user] (no subject) In-Reply-To: References: Message-ID: On Apr 18, 2013 6:00 AM, "Carl Eugen Hoyos" wrote: > > Timothy Gu gmail.com> writes: > > > I found that here are many asm files in > > libavcodec/arm contains > > > > "This file is part of Libav." > > The files were written for Libav, some people > believe that changing this line would violate > copyrights. OK. Timothy G. From jacobhameiri at gmail.com Thu Apr 18 16:16:33 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Thu, 18 Apr 2013 07:16:33 -0700 (PDT) Subject: [FFmpeg-user] multicast streaming real time raw rgb images In-Reply-To: References: <1366210298888-4658524.post@n4.nabble.com> <1366217799156-4658528.post@n4.nabble.com> Message-ID: <1366294593955-4658567.post@n4.nabble.com> setting -analyzeduration -probesize to low values had some effect on the delay ( but still more delay than required ). vlc was able to play the stream but with many errors ( see output below ) ffmpeg wasn't able to play the stream ( see output below ) Also, I had moved to libx264 ( hoping for better network streaming results ) but I have many frame errors. maybe you can understand from the output what my problems are. Can you please examine my command and output and suggest proper modifications ? As mentioned before I have 2 issues, a. delay in played stream b. errors in played stream ( corrupted images ) full commands and debug output: >>>>>>>>>>>>>> ffmpeg: ffmpeg started on 2013-04-18 at 16:55:24 Report written to "ffmpeg-20130418-165524.log" Command line: /bin/ffmpeg -loglevel debug -probesize 32 -analyzeduration 32 -threads 8 -an -f rawvideo -vcodec rawvideo -s 800x1280 -pix_fmt rgba -r 9 -i - -vcodec libx264 -skip_loop_filter all -threads 8 -probesize 32 -analyzeduration 32 -preset ultrafast -tune zerolatency,fastdecode -r 9 -filter:v "scale=400:-1" -sws_flags fast_bilinear -bsf:v h264_mp4toannexb -b 200000 -f mpegts udp://224.1.1.1:1234 -report ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 18 2012 21:53:10 with gcc 4.6.x-google (GCC) 20120106 (prerelease) configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/ --disable-shared --enable-static --cross-prefix=/home/oz/android-ndk-r8b//toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/oz/android-ndk-r8b//platforms/android-8/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon -mfpu=vfpv3-d16' --extra-ldflags='-L../x264 ' --enable-version3 --enable-gpl --disable-doc --enable-libfreetype --enable-hwaccels --disable-ffprobe --enable-ffserver --enable-pthreads --enable-libx264 --disable-fast-unaligned libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 59.100 / 54. 59.100 libavformat 54. 29.104 / 54. 29.104 libavdevice 54. 2.101 / 54. 2.101 libavfilter 3. 17.100 / 3. 17.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 [rawvideo @ 0x24ad7b0] File position before avformat_find_stream_info() is 0 [rawvideo @ 0x24ad7b0] All info found [rawvideo @ 0x24ad7b0] Estimating duration from bitrate, this may be inaccurate [rawvideo @ 0x24ad7b0] File position after avformat_find_stream_info() is 4096000 Input #0, rawvideo, from 'pipe:': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 1, 1/9: Video: rawvideo (RGBA / 0x41424752), rgba, 800x1280, 1/9, 9 tbr, 9 tbn, 9 tbc Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x24a8a80] Setting entry with key 'video_size' to value '800x1280' [buffer @ 0x24a8a80] Setting entry with key 'pix_fmt' to value '28' [buffer @ 0x24a8a80] Setting entry with key 'time_base' to value '1/9' [buffer @ 0x24a8a80] Setting entry with key 'pixel_aspect' to value '0/1' [buffer @ 0x24a8a80] Setting entry with key 'sws_param' to value 'flags=2' [buffer @ 0x24a8a80] Setting entry with key 'frame_rate' to value '9/1' [graph 0 input from stream 0:0 @ 0x24a8a40] w:800 h:1280 pixfmt:rgba tb:1/9 fr:9/1 sar:0/1 sws_param:flags=2 [Parsed_scale_0 @ 0x24a8610] picking yuv420p out of 4 ref:rgba alpha:1 [graph 0 input from stream 0:0 @ 0x24a8a40] TB:0.111111 FRAME_RATE:9.000000 SAMPLE_RATE:NaN [Parsed_scale_0 @ 0x24a8610] w:800 h:1280 fmt:rgba sar:0/1 -> w:400 h:640 fmt:yuv420p sar:0/1 flags:0x1 [libx264 @ 0x24a70d0] not compiled with thread support! [libx264 @ 0x24a70d0] using cpu capabilities: ARMv6 NEON [libx264 @ 0x24a70d0] profile Constrained Baseline, level 2.2 [mpegts @ 0x24a6970] muxrate VBR, pcr every 1 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://224.1.1.1:1234': Metadata: encoder : Lavf54.29.104 Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 400x640, 1/9, q=-1--1, 200 kb/s, 90k tbn, 9 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) [libx264 @ 0x24a70d0] frame= 0 QP=20.00 NAL=3 Slice:I Poc:0 I:1000 P:0 SKIP:0 size=8302 bytes [libx264 @ 0x24a70d0] frame= 1 QP=24.00 NAL=2 Slice:P Poc:2 I:4 P:10 SKIP:986 size=329 bytes [libx264 @ 0x24a70d0] frame= 2 QP=20.00 NAL=2 Slice:P Poc:4 I:12 P:26 SKIP:962 size=631 bytes [libx264 @ 0x24a70d0] frame= 3 QP=16.00 NAL=2 Slice:P Poc:6 I:97 P:260 SKIP:643 size=3237 bytes [libx264 @ 0x24a70d0] frame= 4 QP=15.00 NAL=2 Slice:P Poc:8 I:13 P:77 SKIP:910 size=929 bytes [libx264 @ 0x24a70d0] frame= 5 QP=14.00 NAL=2 Slice:P Poc:10 I:270 P:155 SKIP:575 size=2241 bytes frame= 6 fps=0.0 q=14.0 size= 18kB time=00:00:00.66 bitrate= 216.6kbits/s frame= 6 QP=13.00 NAL=2 Slice:P Poc:12 I:27 P:160 SKIP:813 size=2749 bytes [libx264 @ 0x24a70d0] frame= 7 QP=14.00 NAL=2 Slice:P Poc:14 I:5 P:14 SKIP:981 size=962 bytes [libx264 @ 0x24a70d0] frame= 8 QP=12.00 NAL=2 Slice:P Poc:16 I:202 P:176 SKIP:622 size=2551 bytes [libx264 @ 0x24a70d0] frame= 9 QP=12.00 NAL=2 Slice:P Poc:18 I:112 P:55 SKIP:833 size=1264 bytes frame= 10 fps=8.7 q=12.0 size= 26kB time=00:00:01.11 bitrate= 193.6kbits/s frame= 10 QP=11.00 NAL=2 Slice:P Poc:20 I:162 P:158 SKIP:680 size=2033 bytes [libx264 @ 0x24a70d0] frame= 11 QP=12.00 NAL=2 Slice:P Poc:22 I:108 P:82 SKIP:810 size=2372 bytes [libx264 @ 0x24a70d0] frame= 12 QP=11.00 NAL=2 Slice:P Poc:24 I:110 P:81 SKIP:809 size=1373 bytes [libx264 @ 0x24a70d0] frame= 13 QP=11.00 NAL=2 Slice:P Poc:26 I:105 P:70 SKIP:825 size=1379 bytes frame= 14 fps=8.2 q=11.0 size= 34kB time=00:00:01.55 bitrate= 180.8kbits/s frame= 14 QP=10.00 NAL=2 Slice:P Poc:28 I:112 P:156 SKIP:732 size=2564 bytes [libx264 @ 0x24a70d0] frame= 15 QP=10.00 NAL=2 Slice:P Poc:30 I:5 P:17 SKIP:978 size=1192 bytes [libx264 @ 0x24a70d0] frame= 16 QP=9.00 NAL=2 Slice:P Poc:32 I:9 P:106 SKIP:885 size=1822 bytes [libx264 @ 0x24a70d0] frame= 17 QP=9.00 NAL=2 Slice:P Poc:34 I:6 P:15 SKIP:979 size=1144 bytes frame= 18 fps=8.1 q=9.0 size= 42kB time=00:00:02.00 bitrate= 173.0kbits/s frame= 18 QP=9.00 NAL=2 Slice:P Poc:36 I:9 P:26 SKIP:965 size=2243 bytes [libx264 @ 0x24a70d0] frame= 19 QP=8.00 NAL=2 Slice:P Poc:38 I:89 P:191 SKIP:720 size=3340 bytes [libx264 @ 0x24a70d0] frame= 20 QP=8.00 NAL=2 Slice:P Poc:40 I:18 P:59 SKIP:923 size=1319 bytes [libx264 @ 0x24a70d0] frame= 21 QP=8.00 NAL=2 Slice:P Poc:42 I:4 P:16 SKIP:980 size=1444 bytes frame= 22 fps=8.0 q=8.0 size= 51kB time=00:00:02.44 bitrate= 172.3kbits/s frame= 22 QP=7.00 NAL=2 Slice:P Poc:44 I:8 P:132 SKIP:860 size=2115 bytes [libx264 @ 0x24a70d0] frame= 23 QP=7.00 NAL=2 Slice:P Poc:46 I:3 P:29 SKIP:968 size=1245 bytes [libx264 @ 0x24a70d0] frame= 24 QP=7.00 NAL=2 Slice:P Poc:48 I:13 P:14 SKIP:973 size=1265 bytes [libx264 @ 0x24a70d0] frame= 25 QP=7.00 NAL=2 Slice:P Poc:50 I:5 P:19 SKIP:976 size=1231 bytes frame= 26 fps=7.9 q=7.0 size= 58kB time=00:00:02.88 bitrate= 164.5kbits/s frame= 26 QP=7.00 NAL=2 Slice:P Poc:52 I:8 P:34 SKIP:958 size=2490 bytes [libx264 @ 0x24a70d0] frame= 27 QP=6.00 NAL=2 Slice:P Poc:54 I:6 P:45 SKIP:949 size=1067 bytes [libx264 @ 0x24a70d0] frame= 28 QP=6.00 NAL=2 Slice:P Poc:56 I:3 P:20 SKIP:977 size=1517 bytes [libx264 @ 0x24a70d0] frame= 29 QP=5.00 NAL=2 Slice:P Poc:58 I:332 P:261 SKIP:407 size=4384 bytes frame= 30 fps=7.8 q=5.0 size= 68kB time=00:00:03.33 bitrate= 167.4kbits/s frame= 30 QP=6.00 NAL=2 Slice:P Poc:60 I:4 P:19 SKIP:977 size=1591 bytes [libx264 @ 0x24a70d0] frame= 31 QP=5.00 NAL=2 Slice:P Poc:62 I:104 P:80 SKIP:816 size=1634 bytes [libx264 @ 0x24a70d0] frame= 32 QP=5.00 NAL=2 Slice:P Poc:64 I:102 P:75 SKIP:823 size=1575 bytes [libx264 @ 0x24a70d0] frame= 33 QP=5.00 NAL=2 Slice:P Poc:66 I:102 P:73 SKIP:825 size=1684 bytes frame= 34 fps=7.8 q=5.0 size= 75kB time=00:00:03.77 bitrate= 163.6kbits/s frame= 34 QP=5.00 NAL=2 Slice:P Poc:68 I:115 P:80 SKIP:805 size=3326 bytes [libx264 @ 0x24a70d0] frame= 35 QP=5.00 NAL=2 Slice:P Poc:70 I:108 P:73 SKIP:819 size=2064 bytes [libx264 @ 0x24a70d0] frame= 36 QP=5.00 NAL=2 Slice:P Poc:72 I:102 P:75 SKIP:823 size=1960 bytes [libx264 @ 0x24a70d0] frame= 37 QP=4.00 NAL=2 Slice:P Poc:74 I:107 P:218 SKIP:675 size=4105 bytes frame= 38 fps=7.8 q=4.0 size= 88kB time=00:00:04.22 bitrate= 171.3kbits/s frame= 38 QP=5.00 NAL=2 Slice:P Poc:76 I:11 P:27 SKIP:962 size=1420 bytes [libx264 @ 0x24a70d0] frame= 39 QP=5.00 NAL=2 Slice:P Poc:78 I:2 P:24 SKIP:974 size=1782 bytes [libx264 @ 0x24a70d0] frame= 40 QP=5.00 NAL=2 Slice:P Poc:80 I:6 P:20 SKIP:974 size=1655 bytes [libx264 @ 0x24a70d0] frame= 41 QP=5.00 NAL=2 Slice:P Poc:82 I:7 P:33 SKIP:960 size=2445 bytes frame= 42 fps=7.7 q=5.0 size= 96kB time=00:00:04.66 bitrate= 169.2kbits/s frame= 42 QP=4.00 NAL=2 Slice:P Poc:84 I:7 P:80 SKIP:913 size=2183 bytes [libx264 @ 0x24a70d0] frame= 43 QP=4.00 NAL=2 Slice:P Poc:86 I:5 P:55 SKIP:940 size=1311 bytes [libx264 @ 0x24a70d0] frame= 44 QP=4.00 NAL=2 Slice:P Poc:88 I:8 P:50 SKIP:942 size=1875 bytes [libx264 @ 0x24a70d0] frame= 45 QP=4.00 NAL=2 Slice:P Poc:90 I:5 P:52 SKIP:943 size=2009 bytes frame= 46 fps=7.7 q=4.0 size= 105kB time=00:00:05.11 bitrate= 168.0kbits/s frame= 46 QP=3.00 NAL=2 Slice:P Poc:92 I:9 P:143 SKIP:848 size=2747 bytes [libx264 @ 0x24a70d0] frame= 47 QP=4.00 NAL=2 Slice:P Poc:94 I:3 P:32 SKIP:965 size=2084 bytes [libx264 @ 0x24a70d0] frame= 48 QP=3.00 NAL=2 Slice:P Poc:96 I:3 P:31 SKIP:966 size=1468 bytes [libx264 @ 0x24a70d0] frame= 49 QP=3.00 NAL=2 Slice:P Poc:98 I:13 P:32 SKIP:955 size=1954 bytes frame= 50 fps=7.7 q=3.0 size= 114kB time=00:00:05.55 bitrate= 168.1kbits/s frame= 50 QP=3.00 NAL=2 Slice:P Poc:100 I:4 P:34 SKIP:962 size=1644 bytes [libx264 @ 0x24a70d0] frame= 51 QP=3.00 NAL=2 Slice:P Poc:102 I:7 P:26 SKIP:967 size=1504 bytes [libx264 @ 0x24a70d0] frame= 52 QP=3.00 NAL=2 Slice:P Poc:104 I:8 P:23 SKIP:969 size=1821 bytes [libx264 @ 0x24a70d0] frame= 53 QP=2.00 NAL=2 Slice:P Poc:106 I:3 P:159 SKIP:838 size=2845 bytes frame= 54 fps=7.7 q=2.0 size= 123kB time=00:00:06.00 bitrate= 167.9kbits/s frame= 54 QP=2.00 NAL=2 Slice:P Poc:108 I:10 P:34 SKIP:956 size=1507 bytes [libx264 @ 0x24a70d0] frame= 55 QP=2.00 NAL=2 Slice:P Poc:110 I:0 P:20 SKIP:980 size=786 bytes [libx264 @ 0x24a70d0] frame= 56 QP=3.00 NAL=2 Slice:P Poc:112 I:15 P:21 SKIP:964 size=3536 bytes [libx264 @ 0x24a70d0] frame= 57 QP=2.00 NAL=2 Slice:P Poc:114 I:7 P:34 SKIP:959 size=1475 bytes frame= 58 fps=7.7 q=2.0 size= 131kB time=00:00:06.44 bitrate= 166.9kbits/s frame= 58 QP=2.00 NAL=2 Slice:P Poc:116 I:2 P:23 SKIP:975 size=1303 bytes [libx264 @ 0x24a70d0] frame= 59 QP=1.00 NAL=2 Slice:P Poc:118 I:5 P:25 SKIP:970 size=1767 bytes [libx264 @ 0x24a70d0] frame= 60 QP=1.00 NAL=2 Slice:P Poc:120 I:0 P:20 SKIP:980 size=727 bytes [libx264 @ 0x24a70d0] frame= 61 QP=1.00 NAL=2 Slice:P Poc:122 I:2 P:24 SKIP:974 size=2010 bytes frame= 62 fps=7.7 q=1.0 size= 138kB time=00:00:06.88 bitrate= 164.2kbits/s frame= 62 QP=1.00 NAL=2 Slice:P Poc:124 I:10 P:15 SKIP:975 size=1017 bytes [libx264 @ 0x24a70d0] frame= 63 QP=1.00 NAL=2 Slice:P Poc:126 I:3 P:23 SKIP:974 size=1903 bytes [libx264 @ 0x24a70d0] frame= 64 QP=1.00 NAL=2 Slice:P Poc:128 I:9 P:32 SKIP:959 size=3250 bytes [libx264 @ 0x24a70d0] frame= 65 QP=1.00 NAL=2 Slice:P Poc:130 I:9 P:29 SKIP:962 size=1872 bytes frame= 66 fps=7.7 q=1.0 size= 147kB time=00:00:07.33 bitrate= 164.1kbits/s frame= 66 QP=1.00 NAL=2 Slice:P Poc:132 I:4 P:26 SKIP:970 size=1818 bytes [libx264 @ 0x24a70d0] frame= 67 QP=0.00 NAL=2 Slice:P Poc:134 I:9 P:84 SKIP:907 size=2241 bytes [libx264 @ 0x24a70d0] frame= 68 QP=0.00 NAL=2 Slice:P Poc:136 I:4 P:64 SKIP:932 size=1974 bytes [libx264 @ 0x24a70d0] frame= 69 QP=0.00 NAL=2 Slice:P Poc:138 I:9 P:56 SKIP:935 size=2081 bytes frame= 70 fps=7.7 q=0.0 size= 156kB time=00:00:07.77 bitrate= 164.8kbits/s frame= 70 QP=0.00 NAL=2 Slice:P Poc:140 I:3 P:60 SKIP:937 size=1877 bytes [libx264 @ 0x24a70d0] frame= 71 QP=0.00 NAL=2 Slice:P Poc:142 I:4 P:60 SKIP:936 size=2441 bytes [libx264 @ 0x24a70d0] frame= 72 QP=1.00 NAL=2 Slice:P Poc:144 I:22 P:37 SKIP:941 size=4017 bytes [libx264 @ 0x24a70d0] frame= 73 QP=1.00 NAL=2 Slice:P Poc:146 I:5 P:41 SKIP:954 size=2168 bytes frame= 74 fps=7.7 q=1.0 size= 168kB time=00:00:08.22 bitrate= 167.0kbits/s frame= 74 QP=0.00 NAL=2 Slice:P Poc:148 I:2 P:67 SKIP:931 size=2189 bytes [libx264 @ 0x24a70d0] frame= 75 QP=0.00 NAL=2 Slice:P Poc:150 I:7 P:59 SKIP:934 size=1934 bytes [libx264 @ 0x24a70d0] frame= 76 QP=0.00 NAL=2 Slice:P Poc:152 I:4 P:60 SKIP:936 size=2027 bytes [libx264 @ 0x24a70d0] frame= 77 QP=0.00 NAL=2 Slice:P Poc:154 I:9 P:55 SKIP:936 size=1958 bytes frame= 78 fps=7.7 q=0.0 size= 177kB time=00:00:08.66 bitrate= 166.9kbits/s frame= 78 QP=0.00 NAL=2 Slice:P Poc:156 I:2 P:63 SKIP:935 size=2272 bytes [libx264 @ 0x24a70d0] frame= 79 QP=0.00 NAL=2 Slice:P Poc:158 I:4 P:57 SKIP:939 size=1775 bytes [libx264 @ 0x24a70d0] frame= 80 QP=0.00 NAL=2 Slice:P Poc:160 I:23 P:53 SKIP:924 size=2222 bytes [libx264 @ 0x24a70d0] frame= 81 QP=0.00 NAL=2 Slice:P Poc:162 I:4 P:55 SKIP:941 size=2289 bytes frame= 82 fps=7.7 q=0.0 size= 186kB time=00:00:09.11 bitrate= 167.2kbits/s frame= 82 QP=0.00 NAL=2 Slice:P Poc:164 I:7 P:52 SKIP:941 size=2001 bytes [libx264 @ 0x24a70d0] frame= 83 QP=0.00 NAL=2 Slice:P Poc:166 I:2 P:56 SKIP:942 size=1894 bytes [libx264 @ 0x24a70d0] frame= 84 QP=0.00 NAL=2 Slice:P Poc:168 I:8 P:48 SKIP:944 size=1595 bytes [libx264 @ 0x24a70d0] frame= 85 QP=0.00 NAL=2 Slice:P Poc:170 I:7 P:49 SKIP:944 size=1455 bytes frame= 86 fps=7.7 q=0.0 size= 194kB time=00:00:09.55 bitrate= 166.7kbits/s frame= 86 QP=0.00 NAL=2 Slice:P Poc:172 I:3 P:55 SKIP:942 size=1815 bytes [libx264 @ 0x24a70d0] frame= 87 QP=0.00 NAL=2 Slice:P Poc:174 I:13 P:63 SKIP:924 size=3783 bytes [libx264 @ 0x24a70d0] frame= 88 QP=0.00 NAL=2 Slice:P Poc:176 I:6 P:59 SKIP:935 size=1401 bytes [libx264 @ 0x24a70d0] frame= 89 QP=0.00 NAL=2 Slice:P Poc:178 I:5 P:59 SKIP:936 size=2293 bytes frame= 90 fps=7.7 q=0.0 size= 205kB time=00:00:10.00 bitrate= 167.5kbits/s frame= 90 QP=0.00 NAL=2 Slice:P Poc:180 I:10 P:51 SKIP:939 size=2057 bytes [libx264 @ 0x24a70d0] frame= 91 QP=0.00 NAL=2 Slice:P Poc:182 I:4 P:57 SKIP:939 size=2433 bytes [libx264 @ 0x24a70d0] frame= 92 QP=0.00 NAL=2 Slice:P Poc:184 I:5 P:56 SKIP:939 size=1980 bytes [libx264 @ 0x24a70d0] frame= 93 QP=0.00 NAL=2 Slice:P Poc:186 I:6 P:55 SKIP:939 size=2394 bytes frame= 94 fps=7.7 q=0.0 size= 214kB time=00:00:10.44 bitrate= 168.0kbits/s frame= 94 QP=0.00 NAL=2 Slice:P Poc:188 I:9 P:64 SKIP:927 size=3500 bytes [libx264 @ 0x24a70d0] frame= 95 QP=0.00 NAL=2 Slice:P Poc:190 I:9 P:63 SKIP:928 size=2089 bytes >>>>>>>>>>>>>> ffplay: ffplay started on 2013-04-18 at 16:54:44 Report written to "ffplay-20130418-165444.log" Command line: ffplay.exe -loglevel debug -probesize 3000 -analyzeduration 32 udp://224.1.1.1:1234 -report ffplay version N-46469-gc995644 Copyright (c) 2003-2012 the FFmpeg developers built on Nov 5 2012 17:58:15 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 5.100 / 52. 5.100 libavcodec 54. 71.100 / 54. 71.100 libavformat 54. 36.100 / 54. 36.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 21.106 / 3. 21.106 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 [mpegts @ 0000000001f12ac0] Format mpegts probed with size=2048 and score=100 [mpegts @ 0000000001f12ac0] Unable to seek back to the start [mpegts @ 0000000001f12ac0] File position before avformat_find_stream_info() is 10904 [mpegts @ 0000000001f12ac0] stream=0 stream_type=0 pid=100 prog_reg_desc= [mpegts @ 0000000001f12ac0] probing stream 0 pp:2500 [mpegts @ 0000000001f12ac0] Probe with size=935, packets=1 detected mp3 with score=1 [mpegts @ 0000000001f12ac0] stream=0 stream_type=1b pid=100 prog_reg_desc= [NULL @ 0000000001f26b40] non-existing PPS referenced [h264 @ 0000000001f26b40] non-existing PPS 0 referenced [h264 @ 0000000001f26b40] decode_slice_header error [h264 @ 0000000001f26b40] no frame! [h264 @ 0000000001f26b40] non-existing PPS referenced [h264 @ 0000000001f26b40] non-existing PPS 0 referenced [h264 @ 0000000001f26b40] decode_slice_header error [h264 @ 0000000001f26b40] no frame! [mpegts @ 0000000001f12ac0] Probe buffer size limit 3000 reached [mpegts @ 0000000001f12ac0] Stream #0: not enough frames to estimate rate; consider increasing probesize [mpegts @ 0000000001f12ac0] decoding for stream 0 failed [mpegts @ 0000000001f12ac0] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options rfps: 1.000000 0.000000 rfps: 1.000000 0.000000 rfps: 1.083333 0.000000 rfps: 1.083333 0.000000 rfps: 1.166667 0.000000 rfps: 1.166667 0.000000 rfps: 1.250000 0.000000 rfps: 1.250000 0.000000 rfps: 1.333333 0.000000 rfps: 1.333333 0.000000 rfps: 1.416667 0.000000 rfps: 1.416667 0.000000 rfps: 1.500000 0.000000 rfps: 1.500000 0.000000 rfps: 1.583333 0.000000 rfps: 1.583333 0.000000 rfps: 1.666667 0.000000 rfps: 1.666667 0.000000 rfps: 1.750000 0.000000 rfps: 1.750000 0.000000 rfps: 1.833333 0.000000 rfps: 1.833333 0.000000 rfps: 1.916667 0.000000 rfps: 1.916667 0.000000 rfps: 2.000000 0.000000 rfps: 2.000000 0.000000 rfps: 2.083333 0.000000 rfps: 2.083333 0.000000 rfps: 2.166667 0.000000 rfps: 2.166667 0.000000 rfps: 2.250000 0.000000 rfps: 2.250000 0.000000 rfps: 2.333333 0.000000 rfps: 2.333333 0.000000 rfps: 2.416667 0.000000 rfps: 2.416667 0.000000 rfps: 2.500000 0.000000 rfps: 2.500000 0.000000 rfps: 2.583333 0.000000 rfps: 2.583333 0.000000 rfps: 2.666667 0.000000 rfps: 2.666667 0.000000 rfps: 2.750000 0.000000 rfps: 2.750000 0.000000 rfps: 2.833333 0.000000 rfps: 2.833333 0.000000 rfps: 2.916667 0.000000 rfps: 2.916667 0.000000 rfps: 3.000000 0.000000 rfps: 3.000000 0.000000 rfps: 3.083333 0.000000 rfps: 3.083333 0.000000 rfps: 3.166667 0.000000 rfps: 3.166667 0.000000 rfps: 3.250000 0.000000 rfps: 3.250000 0.000000 rfps: 3.333333 0.000000 rfps: 3.333333 0.000000 rfps: 3.416667 0.000000 rfps: 3.416667 0.000000 rfps: 3.500000 0.000000 rfps: 3.500000 0.000000 rfps: 3.583333 0.000000 rfps: 3.583333 0.000000 rfps: 3.666667 0.000000 rfps: 3.666667 0.000000 rfps: 3.750000 0.000000 rfps: 3.750000 0.000000 rfps: 3.833333 0.000000 rfps: 3.833333 0.000000 rfps: 3.916667 0.000000 rfps: 3.916667 0.000000 rfps: 4.000000 0.000000 rfps: 4.000000 0.000000 rfps: 4.083333 0.000000 rfps: 4.083333 0.000000 rfps: 4.166667 0.000000 rfps: 4.166667 0.000000 rfps: 4.250000 0.000000 rfps: 4.250000 0.000000 rfps: 4.333333 0.000000 rfps: 4.333333 0.000000 rfps: 4.416667 0.000000 rfps: 4.416667 0.000000 rfps: 4.500000 0.000000 rfps: 4.500000 0.000000 rfps: 4.583333 0.000000 rfps: 4.583333 0.000000 rfps: 4.666667 0.000000 rfps: 4.666667 0.000000 rfps: 4.750000 0.000000 rfps: 4.750000 0.000000 rfps: 4.833333 0.000000 rfps: 4.833333 0.000000 rfps: 4.916667 0.000000 rfps: 4.916667 0.000000 rfps: 5.000000 0.000000 rfps: 5.000000 0.000000 rfps: 5.083333 0.000000 rfps: 5.083333 0.000000 rfps: 5.166667 0.000000 rfps: 5.166667 0.000000 rfps: 5.250000 0.000000 rfps: 5.250000 0.000000 rfps: 5.333333 0.000000 rfps: 5.333333 0.000000 rfps: 5.416667 0.000000 rfps: 5.416667 0.000000 rfps: 5.500000 0.000000 rfps: 5.500000 0.000000 rfps: 5.583333 0.000000 rfps: 5.583333 0.000000 rfps: 5.666667 0.000000 rfps: 5.666667 0.000000 rfps: 5.750000 0.000000 rfps: 5.750000 0.000000 rfps: 5.833333 0.000000 rfps: 5.833333 0.000000 rfps: 5.916667 0.000000 rfps: 5.916667 0.000000 rfps: 6.000000 0.000000 rfps: 6.000000 0.000000 rfps: 6.083333 0.000000 rfps: 6.083333 0.000000 rfps: 6.166667 0.000000 rfps: 6.166667 0.000000 rfps: 6.250000 0.000000 rfps: 6.250000 0.000000 rfps: 6.333333 0.000000 rfps: 6.333333 0.000000 rfps: 6.416667 0.000000 rfps: 6.416667 0.000000 rfps: 6.500000 0.000000 rfps: 6.500000 0.000000 rfps: 6.583333 0.000000 rfps: 6.583333 0.000000 rfps: 6.666667 0.000000 rfps: 6.666667 0.000000 rfps: 6.750000 0.000000 rfps: 6.750000 0.000000 rfps: 6.833333 0.000000 rfps: 6.833333 0.000000 rfps: 6.916667 0.000000 rfps: 6.916667 0.000000 rfps: 7.000000 0.000000 rfps: 7.000000 0.000000 rfps: 7.083333 0.000000 rfps: 7.083333 0.000000 rfps: 7.166667 0.000000 rfps: 7.166667 0.000000 rfps: 7.250000 0.000000 rfps: 7.250000 0.000000 rfps: 7.333333 0.000000 rfps: 7.333333 0.000000 rfps: 7.416667 0.000000 rfps: 7.416667 0.000000 rfps: 7.500000 0.000000 rfps: 7.500000 0.000000 rfps: 7.583333 0.000000 rfps: 7.583333 0.000000 rfps: 7.666667 0.000000 rfps: 7.666667 0.000000 rfps: 7.750000 0.000000 rfps: 7.750000 0.000000 rfps: 7.833333 0.000000 rfps: 7.833333 0.000000 rfps: 7.916667 0.000000 rfps: 7.916667 0.000000 rfps: 8.000000 0.000000 rfps: 8.000000 0.000000 rfps: 8.083333 0.000000 rfps: 8.083333 0.000000 rfps: 8.166667 0.000000 rfps: 8.166667 0.000000 rfps: 8.250000 0.000000 rfps: 8.250000 0.000000 rfps: 8.333333 0.000000 rfps: 8.333333 0.000000 rfps: 8.416667 0.000000 rfps: 8.416667 0.000000 rfps: 8.500000 0.000000 rfps: 8.500000 0.000000 rfps: 8.583333 0.000000 rfps: 8.583333 0.000000 rfps: 8.666667 0.000000 rfps: 8.666667 0.000000 rfps: 8.750000 0.000000 rfps: 8.750000 0.000000 rfps: 8.833333 0.000000 rfps: 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10.583333 0.000000 rfps: 10.583333 0.000000 rfps: 10.666667 0.000000 rfps: 10.666667 0.000000 ... rfps: 59.833333 0.000000 rfps: 59.916667 0.000000 rfps: 59.916667 0.000000 rfps: 60.000000 0.000000 rfps: 60.000000 0.000000 rfps: 23.976024 0.000000 rfps: 23.976024 0.000000 rfps: 29.970030 0.000000 rfps: 29.970030 0.000000 rfps: 59.940060 0.000000 rfps: 59.940060 0.000000 rfps: 11.988012 0.000000 rfps: 11.988012 0.000000 rfps: 14.985015 0.000000 rfps: 14.985015 0.000000 rfps: 47.952048 0.000000 rfps: 47.952048 0.000000 [mpegts @ 0000000001f12ac0] Estimating duration from bitrate, this may be inaccurate [mpegts @ 0000000001f12ac0] File position after avformat_find_stream_info() is 21056 [AVIOContext @ 000000000203fc00] Statistics: 21056 bytes read, 0 seeks >>>>>>>>>>>>>> vlc: main debug: processing request item: udp://224.1.1.1:1234, node: Playlist, skip: 0 main debug: resyncing on udp://224.1.1.1:1234 main debug: udp://224.1.1.1:1234 is at 5 main debug: starting playback of the new playlist item main debug: resyncing on udp://224.1.1.1:1234 main debug: udp://224.1.1.1:1234 is at 5 main debug: creating new input thread main debug: Creating an input for 'udp://224.1.1.1:1234' main debug: using timeshift granularity of 50 MiB, in path 'C:\Users\admin\AppData\Local\Temp' main debug: `udp://@224.1.1.1:1234' gives access `udp' demux `' path `@224.1.1.1:1234' main debug: creating demux: access='udp' demux='' location='@224.1.1.1:1234' file='\\@224.1.1.1:1234' main debug: looking for access_demux module: 0 candidates main debug: no access_demux module matched "udp" main debug: TIMER module_need() : 0.158 ms - Total 0.158 ms / 1 intvls (Avg 0.158 ms) main debug: creating access 'udp' location='@224.1.1.1:1234', path='\\@224.1.1.1:1234' main debug: looking for access module: 1 candidate access_udp debug: opening server=:0 local=224.1.1.1:1234 main debug: net: opening 224.1.1.1 datagram port 1234 main debug: using access module "access_udp" main debug: TIMER module_need() : 0.791 ms - Total 0.791 ms / 1 intvls (Avg 0.791 ms) main debug: Using block method for AStream* main debug: starting pre-buffering qt4 debug: IM: Setting an input main debug: received first data after 3528 ms main debug: prebuffering done 1472 bytes in 3s - 0 KiB/s main debug: looking for stream_filter module: 4 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.162 ms - Total 0.162 ms / 1 intvls (Avg 0.162 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.113 ms - Total 0.113 ms / 1 intvls (Avg 0.113 ms) main debug: creating demux: access='udp' demux='' location='@224.1.1.1:1234' file='\\@224.1.1.1:1234' main debug: looking for demux module: 55 candidates ts debug: PATCallBack called ts debug: new PAT ts_id=1 version=0 current_next=1 ts debug: * number=1 pid=4096 ts debug: PMTCallBack called ts debug: new PMT program number=1 version=0 pid_pcr=256 ts debug: * es pid=256 type=27 fcc=h264 main debug: selecting program id=1 access_udp warning: unimplemented query in control main debug: using demux module "ts" main debug: TIMER module_need() : 63.822 ms - Total 63.822 ms / 1 intvls (Avg 63.822 ms) main debug: looking for decoder module: 32 candidates avcodec debug: libavcodec already initialized avcodec debug: trying to use direct rendering avcodec debug: allowing 4 thread(s) for decoding avcodec debug: ffmpeg codec (H264 - MPEG-4 AVC (part 10)) started avcodec debug: using frame thread mode with 4 threads main debug: using decoder module "avcodec" main debug: TIMER module_need() : 3.296 ms - Total 3.296 ms / 1 intvls (Avg 3.296 ms) main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_h264" main debug: TIMER module_need() : 0.192 ms - Total 0.192 ms / 1 intvls (Avg 0.192 ms) ts debug: DEMUX_SET_GROUP 0 00000000 access_udp warning: unimplemented query in control main debug: looking for meta reader module: 2 candidates lua debug: Trying Lua scripts in C:\Users\admin\AppData\Roaming\vlc\lua\meta\reader lua debug: Trying Lua scripts in D:\Program Files (x86)\VideoLAN\VLC\lua\meta\reader lua debug: Trying Lua playlist script D:\Program Files (x86)\VideoLAN\VLC\lua\meta\reader\filename.luac main debug: no meta reader module matching "any" could be loaded main debug: TIMER module_need() : 1.455 ms - Total 1.455 ms / 1 intvls (Avg 1.455 ms) main debug: `udp://@224.1.1.1:1234' successfully opened main debug: Buffering 0% packetizer_h264 debug: found NAL_SPS (sps_id=0) packetizer_h264 debug: found NAL_PPS (pps_id=0 sps_id=0) main debug: Stream buffering done (111 ms in 227 ms) main debug: Decoder buffering done in 0 ms main debug: looking for text renderer module: 2 candidates freetype debug: Using Arial as font from file C:\Windows\Fonts\arial.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 0.977 ms - Total 0.977 ms / 1 intvls (Avg 0.977 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 0.664 ms - Total 0.664 ms / 1 intvls (Avg 0.664 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.170 ms - Total 0.170 ms / 1 intvls (Avg 0.170 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates direct3d debug: creating Vout EventThread direct3d debug: DirectXCreateWindow main debug: looking for vout window hwnd module: 2 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window hwnd module "qt4" main debug: TIMER module_need() : 14.223 ms - Total 14.223 ms / 1 intvls (Avg 14.223 ms) direct3d debug: created video sub-window direct3d debug: Vout EventThread running direct3d debug: disabling screen saver direct3d debug: Direct3d Device: NVIDIA GeForce GT 330M 4318 2601 162 main debug: VoutDisplayEvent 'resize' 416x678 window direct3d warning: trying surface pixel format: YV12 direct3d debug: selected surface pixel format is YV12 direct3d debug: Using D3DTEXF_LINEAR for minification direct3d debug: Using D3DTEXF_LINEAR for magnification direct3d debug: Direct3D scene created successfully direct3d debug: Direct3D device adapter successfully initialized main debug: using vout display module "direct3d" main debug: TIMER module_need() : 190.594 ms - Total 190.594 ms / 1 intvls (Avg 190.594 ms) main debug: original format sz 400x640, of (0,0), vsz 400x640, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates main debug: VoutDisplayEvent 'resize' 422x640 window freetype debug: Using Arial as font from file C:\Windows\Fonts\arial.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 1.806 ms - Total 1.806 ms / 1 intvls (Avg 1.806 ms) avcodec debug: using direct rendering main debug: End of video preroll direct3d debug: Created 364x43 texture for OSD main debug: Post-processing available main debug: auto hiding mouse cursor ts debug: PSINewTableCallBack: table 0x42(66) ext=0x1(1) ts debug: SDTCallBack called ts debug: new SDT ts_id=1 version=0 current_next=1 network_id=1 ts debug: * service id=1 eit schedule=0 present=0 running=4 free_ca=0 ts debug: - type=1 provider=FFmpeg name=Service01 main debug: EsOutProgramMeta: number=1 main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 100 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Stream buffering done (111 ms in 106 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 116 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 95% main debug: Stream buffering done (222 ms in 330 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 121 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 91% main debug: Stream buffering done (222 ms in 204 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 143 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 77% main debug: Stream buffering done (222 ms in 306 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 144 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 76% main debug: Stream buffering done (222 ms in 183 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main warning: picture is too late to be displayed (missing 129 ms) main debug: picture might be displayed late (missing 18 ms) main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 191 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 58% main debug: Stream buffering done (222 ms in 203 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main warning: picture is too late to be displayed (missing 73 ms) main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 200 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 55% main debug: Stream buffering done (222 ms in 310 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 234 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 47% main debug: Buffering 94% main debug: Stream buffering done (333 ms in 415 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 263 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 42% main debug: Buffering 84% main debug: Stream buffering done (333 ms in 479 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 297 ms) main error: ES_OUT_RESET_PCR called main debug: Buffering 0% main debug: Buffering 37% main debug: Buffering 74% main debug: Stream buffering done (333 ms in 148 ms) main debug: End of video preroll main debug: Received first picture main debug: Decoder buffering done in 0 ms main debug: picture might be displayed late (missing 1 ms) main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 312 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 35% main debug: Buffering 71% main debug: Stream buffering done (333 ms in 409 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 314 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 35% main debug: Buffering 70% main debug: Stream buffering done (333 ms in 408 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 325 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 34% main debug: Buffering 68% main debug: Stream buffering done (333 ms in 309 ms) main debug: Decoder buffering done in 0 ms main debug: End of video preroll main debug: auto hiding mouse cursor main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 337 ms) main error: ES_OUT_RESET_PCR called main warning: early picture skipped main debug: Buffering 0% main debug: Buffering 32% main debug: Buffering 65% main debug: Buffering 98% main debug: Stream buffering done (444 ms in 512 ms) main debug: End of video preroll main debug: Received first picture main debug: Decoder buffering done in 0 ms main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 337 ms) main error: ES_OUT_RESET_PCR called main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 339 ms) main error: ES_OUT_RESET_PCR called main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 343 ms) main error: ES_OUT_RESET_PCR called main error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 351 ms) main error: ES_OUT_RESET_PCR called -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/multicast-streaming-real-time-raw-rgb-images-tp4658526p4658567.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From paulj at fastmail.es Thu Apr 18 16:52:55 2013 From: paulj at fastmail.es (paulj at fastmail.es) Date: Thu, 18 Apr 2013 16:52:55 +0200 Subject: [FFmpeg-user] Replacement for Flash Media Server? Message-ID: <517008C7.17424.16505B1@paulj.fastmail.es> Hi all: I need to stream video from Flash Media Live Encoder, not directly to the Web, but to an intermediate server, which will in turn re-transmit it to the Web (through justin.tv). Flash Media Live Encoder seems designed to work with Flash Media Server, but can I use ffmpeg instead? (i.e.: install ffmpeg in a server, point Flash Media Live Encoder to its URL and have ffmpeg receive correctly its stream). Thanks in advance, Paulo. ------------------------------------------- "Oh, don't worry kids. Most of you will never fall in love. You'll just marry for fear of dying alone." The Simpsons. ------------------------------------------- From cehoyos at ag.or.at Thu Apr 18 16:56:20 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 14:56:20 +0000 (UTC) Subject: [FFmpeg-user] .MTS (sony) file with LPCM song References: <516FEAD0.2040907@apiculture-warre.fr> <516FF99F.9080102@apiculture-warre.fr> <517001D7.1030501@apiculture-warre.fr> Message-ID: Guillaume Fontaine apiculture-warre.fr> writes: > So now, I have some softwares who can read my file > (kdenlive, winFF, xine, ffmpeg) and some others not > (totem, vlc). I don't think this has anything to do with FFmpeg vs fork, vlc and totem just don't support the file. Carl Eugen From wiebe at halfgaar.net Thu Apr 18 16:57:27 2013 From: wiebe at halfgaar.net (Wiebe Cazemier) Date: Thu, 18 Apr 2013 16:57:27 +0200 (CEST) Subject: [FFmpeg-user] LFE channel + 10dB In-Reply-To: <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> References: <2110300068.19174.1365848200960.JavaMail.root@halfgaar.net> <1805285104.19208.1365849310718.JavaMail.root@halfgaar.net> <984976981.19358.1365859854366.JavaMail.root@halfgaar.net> Message-ID: <783534887.24289.1366297047946.JavaMail.root@halfgaar.net> ----- Original Message ----- > From: "Wiebe Cazemier" > To: "FFmpeg user questions" > Sent: Saturday, 13 April, 2013 3:30:54 PM > Subject: Re: [FFmpeg-user] LFE channel + 10dB > > > Wiebe Cazemier halfgaar.net> writes: > > > > > 1) As mentioned, does ffmpeg adhere to this and > > > allow connect 5.1 speakers to apply the boost? > > > > Please test and tell us! > > This is harder than it sounds, because you don't know if the test > file already compensates for it. The reference I mentioned says: > > === > Calibration tests generally hide this 10dB difference; a basic pink > noise test will contain a signal that should play at 75dB SPL for > each channel. For the normal channels, that means a signal 30dB > below full scale (105dB-30dB = 75dB), but for the LFE channel it is > a signal 40dB below full scale (115dB-40dB = 75dB): 1/3 of the > signal amplitude. If those test signals are used to calibrate the > amplifier so they all output at 75dB SPL, then everything is set up > okay. The LFE channel now has 10dB of extra headroom above the other > channels and can produce the full 115dB SPL when required. > === > > But, I will do my best to find out, of course. > > I can tell you that the ALSA driver doesn't apply this boost to its > .1 output (For my Audigy 2 ZS), because testfile [1] plays > significantly louder when addressing the subwoofer. > > [1] http://www.halfgaar.net/media/chan-id.zip To get back to this; finding a calibration file seems impossible. This mailing list discussion, archived at different sites, is basically the only thing I find when I Google for it. I did test with the DVD 'Pulse' from Pink Floyd, which has a calibration routine on it. However, it says that the sub should play 4 dB (or was it 6?) louder than the rest. I don't understand why or what this is. It might be just a curious thing in their tests, because no standard specifies 4 or 6. I also listened to a lot of movies. By the sound of it, the 10 dB boost is not applied and with it applied externally, it seems to give a pretty balanced result. Bass that is obviously meant to shake the room only does so with the boost, yet normal sound doesn't seem too rich in bass. So, it would seem ffmpeg does it right. From cehoyos at ag.or.at Thu Apr 18 16:59:22 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 18 Apr 2013 14:59:22 +0000 (UTC) Subject: [FFmpeg-user] multicast streaming real time raw rgb images References: <1366210298888-4658524.post@n4.nabble.com> <1366217799156-4658528.post@n4.nabble.com> <1366294593955-4658567.post@n4.nabble.com> Message-ID: Jacobs gmail.com> writes: > Also, I had moved to libx264 I don't know much about libx264 encoding, but I don't see any options that force low delay in your encoding command. [...] > As mentioned before I have 2 issues, > a. delay in played stream > b. errors in played stream ( corrupted images ) Are they only reproducible if you stream or also if you output to a file? [...] > ffplay.exe -loglevel debug -probesize 3000 -analyzeduration 32 I don't think this can work on the reception site, you need larger values there. Carl Eugen From greg.p.wunder.ctr at mail.mil Thu Apr 18 17:02:31 2013 From: greg.p.wunder.ctr at mail.mil (Wunder, Greg P CTR (US)) Date: Thu, 18 Apr 2013 15:02:31 +0000 Subject: [FFmpeg-user] ffmpeg script to deal with RTSP (UNCLASSIFIED) Message-ID: <75E856F5C9F7FB46BCA8B05A9BB0C8B830D691A1@ucolhp9j.easf.csd.disa.mil> Classification: UNCLASSIFIED Caveats: NONE My infrastructure is setup such that I am running on a CentOS 6.x 64bit image with FFMPEG 0.10.2. I am attempting to deal with 2 problems: 1) Time-out and restart of RTSP video source 2) Errors in RTSP video source My script is as follows: ffmpeg -i "rtsp://192.168.12.3/myFeed" -f flv -an "rtmp://192.168.12.33/live/myFeed" For #1, every thirty minutes the feed get reset as the video is written off to disk which I have no control over. How can I have the script wait during this period and then pick-up the feed again rather than manually restart the script? For #2 I get the following stdout/stderr at runtime causing the feed to have issues (accessing the original RTSP feed with VLC it looks good, but RTMP from the FFMPEG script it does not). RTMP is hosted via Adobe Media Server 5.0.1. [h264 @ 0x1f63ba0] concealing 799 DC, 799 AC, 799 MV errors [h264 @ 0x1fbfe40] illegal short term buffer state detected [h264 @ 0x1fbfe40] concealing 750 DC, 750 AC, 750 MV errors [h264 @ 0x201c0c0] left block unavailable for requested intra4x4 mode -1 at 23 7 [h264 @ 0x201c0c0] error while decoding MB 23 7 [h264 @ 0x201c0c0] P sub_mb_type 4 out of range at 15 22 [h264 @ 0x201c0c0] error while decoding MB 15 22 [h264 @ 0x201c0c0] concealing 726 DC, 726 AC, 726 MV errors [h264 @ 0x2078340] illegal short term buffer state detected [h264 @ 0x2078340] left block unavailable for requested intra mode at 30 18 [h264 @ 0x2078340] error while decoding MB 30 18 [h264 @ 0x2078340] concealing 379 DC, 379 AC, 379 MV errors frame= 5413 fps= 15 q=31.0 size= 11191kB time=00:06:13.17 bitrate= 245.7kbits/concealing 750 DC, 750 AC, 750 MV errors [h264 @ 0x1c848e0] top block unavailable for requested intra mode at 0 15 [h264 @ 0x1c848e0] error while decoding MB 0 15 [h264 @ 0x1c848e0] P sub_mb_type 6 out of range at 7 22 [h264 @ 0x1c848e0] error while decoding MB 7 22 [h264 @ 0x1c848e0] concealing 727 DC, 727 AC, 727 MV errors [h264 @ 0x1c84dc0] illegal short term buffer state detected [h264 @ 0x1c84dc0] mb_type 791 in P slice too large at 30 18 [h264 @ 0x1c84dc0] error while decoding MB 30 18 [h264 @ 0x1c84dc0] top block unavailable for requested intra mode at 23 22 [h264 @ 0x1c84dc0] error while decoding MB 23 22 [h264 @ 0x1c84dc0] concealing 509 DC, 509 AC, 509 MV errors [h264 @ 0x1eab960] mb_type 31 in P slice too large at 30 18 [h264 @ 0x1eab960] error while decoding MB 30 18 [h264 @ 0x1eab960] top block unavailable for requested intra mode at 34 21 [h264 @ 0x1eab960] error while decoding MB 34 21 [h264 @ 0x1f07a80] concealing 750 DC, 750 AC, 750 MV errors [h264 @ 0x1eab960] top block unavailable for requested intra4x4 mode -1 at 21 27 [h264 @ 0x1eab960] error while decoding MB 21 27 [h264 @ 0x1eab960] concealing 483 DC, 483 AC, 483 MV errors [h264 @ 0x1f63ba0] mb_type 346 in P slice too large at 10 11 [h264 @ 0x1f63ba0] error while decoding MB 10 11 [h264 @ 0x1f63ba0] top block unavailable for requested intra mode at 26 24 [h264 @ 0x1f63ba0] error while decoding MB 26 24 [h264 @ 0x1f63ba0] concealing 811 DC, 811 AC, 811 MV errors [h264 @ 0x1fbfe40] illegal short term buffer state detected [h264 @ 0x1fbfe40] top block unavailable for requested intra mode at 30 18 [h264 @ 0x1fbfe40] error while decoding MB 30 18 [h264 @ 0x1fbfe40] mb_type 33 in P slice too large at 28 24 [h264 @ 0x1fbfe40] error while decoding MB 28 24 [h264 @ 0x1fbfe40] concealing 467 DC, 467 AC, 467 MV errors [h264 @ 0x201c0c0] concealing 750 DC, 750 AC, 750 MV errors [h264 @ 0x2078340] mb_type 121 in P slice too large at 10 11 [h264 @ 0x2078340] error while decoding MB 10 11 [h264 @ 0x2078340] top block unavailable for requested intra mode at 0 25 [h264 @ 0x2078340] error while decoding MB 0 25 [h264 @ 0x2078340] concealing 802 DC, 802 AC, 802 MV errors [h264 @ 0x1c83a40] illegal short term buffer state detected [h264 @ 0x1c83a40] top block unavailable for requested intra mode at 30 18 [h264 @ 0x1c83a40] error while decoding MB 30 18 [h264 @ 0x1c83a40] negative number of zero coeffs at 19 21 [h264 @ 0x1c83a40] error while decoding MB 19 21 [h264 @ 0x1c83a40] P sub_mb_type 6 out of range at 12 27 [h264 @ 0x1c83a40] error while decoding MB 12 27 frame= 5420 fps= 15 q=31.0 size= 11208kB time=00:06:13.64 bitrate= 245.7kbits/concealing 486 DC, 486 AC, 486 MV errors Thanks for any help you can be. Greg Classification: UNCLASSIFIED Caveats: NONE From me at renecalles.de Thu Apr 18 18:51:51 2013 From: me at renecalles.de (=?iso-8859-1?Q?Ren=E9_Calles?=) Date: Thu, 18 Apr 2013 18:51:51 +0200 Subject: [FFmpeg-user] Generation of Smooth Streaming Manifest files Message-ID: <51F01222-AF3C-4DC6-880F-EFD59F35BB63@renecalles.de> Hi there, i wonder if there is an option to create Smooth Streaming .ism- and .ismc-manifests with the segmented option. I haven't found anything in the documentation except for the external tool ismindex but this is actually not working for live streaming purposes as far is i could test it. If not possible via FFmpeg, does anyone know a script or something i could use in separate process? Thanks and best regards, Ren? From seandarcy2 at gmail.com Thu Apr 18 20:57:15 2013 From: seandarcy2 at gmail.com (sean darcy) Date: Thu, 18 Apr 2013 14:57:15 -0400 Subject: [FFmpeg-user] vbr with libfdk_aac ? Message-ID: ffmpeg --help full | grep -A 10 libfdk ffmpeg version N-52058-g423089e Copyright (c) 2000-2013 the FFmpeg developers ............ libfdk_aac AVOptions: -afterburner E...A. Afterburner (improved quality) (from 0 to 1) -eld_sbr E...A. Enable SBR for ELD (for SBR in other configurations, use the -profile parameter) (from 0 to 1) -signaling E...A. SBR/PS signaling style (from -1 to 2) default E...A. Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled) implicit E...A. Implicit backwards compatible signaling explicit_sbr E...A. Explicit SBR, implicit PS signaling explicit_hierarchical E...A. Explicit hierarchical signaling -latm E...A. Output LATM/LOAS encapsulated data (from 0 to 1) -header_period E...A. StreamMuxConfig and PCE repetition period (in frames) (from 0 to 65535) -vbr E...A. VBR mode (1-5) (from 0 to 5) But look at this. copying the wiki: ffmpeg -i 17.wav -vn -c:a libfdk_aac -vbr 2 -afterburner 1 17.m4a ffmpeg version N-52058-g423089e Copyright (c) 2000-2013 the FFmpeg developers built on Apr 13 2013 19:19:37 with gcc 4.7.2 (GCC) 20120921 (Red Hat 4.7.2-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --cpu=amdfam10 --optflags='-Ofast -march=native -mtune=native -fopenmp -fomit-frame-pointer -pipe -fPIC' --enable-avresample --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-frei0r --enable-libfdk-aac --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --enable-nonfree --enable-zlib libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 53.101 / 3. 53.101 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [wav @ 0xe1ba00] max_analyze_duration 5000000 reached at 5015510 microseconds Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from '17.wav': Duration: 00:30:31.07, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s [libfdk_aac @ 0xe43280] Note, the VBR setting is unsupported and only works with some parameter combinations .......... BTW, also ran it with -afterburner. sean From timothygu99 at gmail.com Fri Apr 19 01:36:26 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Thu, 18 Apr 2013 16:36:26 -0700 Subject: [FFmpeg-user] re-streaming rtsp on arm In-Reply-To: References: Message-ID: Hi, On Apr 18, 2013 12:05 AM, "ryan lindsay" wrote: > I have got the version 1.2 from the website and compiled from scratch on my > beagleboard > and it's running ubuntu 12 > When I run ffmpeg I get > (snip) > ffmpeg version > 0.8.6-6:0.8.6-0ubuntu0.12.10.1, Copyright (c) 2000-2013 the Libav developers You told us that you compiled ffmpeg 1.2 yourself, but the command line output does not match up. Did you forget to make install ? Regards, Timothy G. From llee040 at sbcglobal.net Fri Apr 19 02:44:22 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Thu, 18 Apr 2013 19:44:22 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: Message-ID: On 4/17/13 5:36 PM, "Carl Eugen Hoyos" wrote: >L. Lee sbcglobal.net> writes: > >> https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output.zip >> https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip > >Both samples look progressive to me, I don't think >the fieldmatch filter can have any effect on >these files. >What do you expect the filter to do? I don't know what gives me the results demonstrated by the conversion of https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip to https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output_mpg.zip if the fieldmatch filter isn't doing it, but that conversion is satisfactory, with the exception of two significant deficiencies. Some areas are pixillated (as you can observe) and converting an excerpt of longer duration results in a/v sync issues (audio is delayed, however, I haven't yet done enough testing to determine whether the sync problem increases cumulatively or whether there's just an offset that occurs initially. I also tried the same filtergraph with 1080i sources, but the results were so similar to the ones achieved for 720p that I didn't consider providing samples. Here's are the two methods that I've developed to apply pullup for the two denominations of source video using MEncoder in case the correlation will help clarify what is needed. It's been quite some time since I've needed to alter these options which result in very high quality conversions for the video samples I provided in my O.P. I'll be glad to provide corresponding MEncoder excerpts for the encoding results already submitted for comparison to those achieved through ffmpeg, if that would help you help me. 720p: -demuxer lavf -vf filmdint=dint_thres=256,fspp,softskip,scale=1280:720,framestep=2 -mc 0 -noskip -aspect 16/9 -of lavf -lavfopts format=mp4 -alang en -oac faac -faacopts br=192:mpeg=4:object=2:raw -channels 6 -fps 30000/1001 -ofps 24000/1001 -ovc x264 -x264encopts preset=slower:tune=film:crf=28:bitrate=1800:bframes=0:global_header:threads =auto:frameref=3 1080p: -demuxer lavf -vf filmdint=fast=0/sad_thres=512,fspp,softskip,crop=1920:1064:0:16,scale=1280: 720 -mc 0 -noskip -of lavf -lavfopts format=mp4 -aspect 16/9 -oac faac -faacopts br=192:mpeg=4:object=2:raw -channels 6 -fps 30000/1001 -ofps 24000/1001 -ovc x264 -x264encopts preset=veryslow:tune=film:crf=28:bitrate=1800:bframes=0:global_header:threa ds=auto:frameref=3 Thanks. Laine Lee > >Please note that the pullup filter in FFmpeg >currently does not work! > >Carl Eugen From timothygu99 at gmail.com Fri Apr 19 04:45:49 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Thu, 18 Apr 2013 19:45:49 -0700 Subject: [FFmpeg-user] vbr with libfdk_aac ? In-Reply-To: References: Message-ID: On Thu, Apr 18, 2013 at 11:57 AM, sean darcy wrote: > ffmpeg -i 17.wav -vn -c:a libfdk_aac -vbr 2 -afterburner 1 17.m4a > ffmpeg version N-52058-g423089e Copyright (c) 2000-2013 the FFmpeg > developers > built on Apr 13 2013 19:19:37 with gcc 4.7.2 (GCC) 20120921 (Red Hat > 4.7.2-2) > configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg > --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man > --arch=x86_64 --cpu=amdfam10 --optflags='-Ofast -march=native -mtune=native > -fopenmp -fomit-frame-pointer -pipe -fPIC' --enable-avresample > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd > --enable-frei0r --enable-libfdk-aac --enable-gnutls --enable-libass > --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack > --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal > --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse > --enable-librtmp --enable-libschroedinger --enable-libspeex > --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx > --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter > --enable-postproc --enable-pthreads --disable-static --enable-shared > --enable-gpl --enable-nonfree --enable-zlib libavutil 52. 26.100 / > 52. 26.100 > libavcodec 55. 2.100 / 55. 2.100 > libavformat 55. 2.100 / 55. 2.100 > libavdevice 55. 0.100 / 55. 0.100 > libavfilter 3. 53.101 / 3. 53.101 > libavresample 1. 1. 0 / 1. 1. 0 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 3.100 / 52. 3.100 > [wav @ 0xe1ba00] max_analyze_duration 5000000 reached at 5015510 > microseconds > Guessed Channel Layout for Input Stream #0.0 : stereo > Input #0, wav, from '17.wav': > Duration: 00:30:31.07, bitrate: 1411 kb/s > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, > s16, 1411 kb/s > [libfdk_aac @ 0xe43280] Note, the VBR setting is unsupported and only works > with some parameter combinations Question for Sean: Is the output OK? If it is, then it shouldn't be a problem. (Also see question a) below) I also have a couple of questions for developers for the library. Please correct me (and please don't consider me rude) if I am wrong. a) According to the official libfdk-aac available in mstorsjo's repo, it does not say anything about VBR, or values between 1-5 for param AACENC_BITRATEMODE. But in line 185 of libavcodec/libfdk-aacenc.c, ffmpeg is setting it for VBR. In my opinion, not say it can do VBR stuff means it can't. b) Shouldn't -vbr be -q:a? And if it is for abbreviating "-flags +qscale -global_quality [1-5]", shouldn't it also map to other codecs requiring this option? Timothy G. From volodomanov at gmail.com Fri Apr 19 05:24:59 2013 From: volodomanov at gmail.com (Dennis Volodomanov) Date: Fri, 19 Apr 2013 13:24:59 +1000 Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th Message-ID: <5170B90B.7060300@gmail.com> Hello all, The build on April 6th was decoding an MP3 stream just fine, but some time after that it broke unfortunately. Please check the following output: > ffmpeg_old -loglevel 99 -i http://radio.retro-retro.ru ffmpeg version N-51639-g7775992 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 6 2013 17:29:40 with gcc 4.7.2 (GCC) configuration: --disable-shared --enable-static --arch=x86_64 --target-os=mingw32 --cross-prefix=/home/dennis/cc/mingw-w64-x86_64/bin/x86_64-w64-mingw32- --pkg-config=pkg-config --enable-gpl --enable-libx264 --enable-avisynth --enable-libxvid --enable-libmp3lame --enable-version3 --enable-zlib --enable-libopenjpeg --enable-libfreetype --enable-fontconfig --enable-libass --disable-w32threads --enable-bzlib --enable-libbluray --enable-dxva2 --enable-memalign-hack --enable-avresample --enable-runtime-cpudetect --extra-cflags=-DPTW32_STATIC_LIB libavutil 52. 25.100 / 52. 25.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 1.100 / 55. 1.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 49.100 / 3. 49.100 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'. Reading option '-i' ... matched as input file with argument 'http://radio.retro-retro.ru'. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument 99. Successfully parsed a group of options. Parsing a group of options: input file http://radio.retro-retro.ru. Successfully parsed a group of options. Opening an input file: http://radio.retro-retro.ru. [mp3 @ 0000000001aa7460] Format mp3 probed with size=65536 and score=50 [mp3 @ 0000000001aa7460] File position before avformat_find_stream_info() is 0 [mp3 @ 0000000001ad2300] Header missing [mp3 @ 0000000001aa7460] max_analyze_duration 5000000 reached at 5015510 microseconds [mp3 @ 0000000001aa7460] File position after avformat_find_stream_info() is 61440 Input #0, mp3, from 'http://radio.retro-retro.ru': Duration: N/A, start: 0.000000, bitrate: 96 kb/s Stream #0:0, 194, 1/14112000: Audio: mp3, 44100 Hz, stereo, s16p, 96 kb/s Successfully opened the file. At least one output file must be specified [AVIOContext @ 0000000001a4fe00] Statistics: 65536 bytes read, 0 seeks ------------------------------------------------------- > ffmpeg -loglevel 99 -i http://radio.retro-retro.ru ffmpeg version N-52262-gb63f7b3 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 19 2013 12:03:13 with gcc 4.7.2 (GCC) configuration: --disable-shared --enable-static --arch=x86_64 --target-os=mingw32 --cross-prefix=/home/dennis/cc/mingw-w64-x86_64/bin/x86_64-w64-mingw32- --pkg-config=pkg-config --enable-gpl --enable-libx264 --enable-avisynth --enable-libxvid --enable-libmp3lame --enable-version3 --enable-zlib --enable-libopenjpeg --enable-libfreetype --enable-fontconfig --enable-libass --disable-w32threads --enable-bzlib --enable-libbluray --enable-dxva2 --enable-memalign-hack --enable-avresample --enable-runtime-cpudetect --extra-cflags=-DPTW32_STATIC_LIB libavutil 52. 27.100 / 52. 27.100 libavcodec 55. 3.100 / 55. 3.100 libavformat 55. 3.100 / 55. 3.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 58.100 / 3. 58.100 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'. Reading option '-i' ... matched as input file with argument 'http://radio.retro-retro.ru'. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument 99. Successfully parsed a group of options. Parsing a group of options: input file http://radio.retro-retro.ru. Successfully parsed a group of options. Opening an input file: http://radio.retro-retro.ru. [AVIOContext @ 0000000001a94f60] Statistics: 4924 bytes read, 0 seeks http://radio.retro-retro.ru: Invalid data found when processing input It fails even if I force "-f mp3". Thanks in advance for your help! Dennis From werner.robitza at gmail.com Fri Apr 19 08:58:00 2013 From: werner.robitza at gmail.com (Werner Robitza) Date: Fri, 19 Apr 2013 08:58:00 +0200 Subject: [FFmpeg-user] What is the (technical) difference between -r and the fps filter? Message-ID: One can extract thumbnails with ffmpeg -i input -vf fps="fps=1/60" out%03d.png or ffmpeg -i input -r 1/60 out%03d.png It looks like they'd do the same, but when trying it on a video I got different frame counts, the latter being incorrect (see output later). So, what is the difference between those two? I remember reading that the -r option is applied after the video filters, but what'd cause it to create a different number of output frames? --------- % ffmpeg -y -i input.mp4 -vf fps=fps=1/60 img%03d.png ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 2 2013 20:38:39 with Apple LLVM version 4.2 (clang-425.0.24) (based on LLVM 3.2svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libvo-aacenc --enable-libass --enable-libfdk-aac --enable-libopus --enable-frei0r --enable-libopenjpeg --extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.1/include/openjpeg-1.5 ' libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2012-01-08 11:16:19 encoder : Lavf53.24.0 Duration: 00:03:31.58, start: 0.000000, bitrate: 2359 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 2229 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: creation_time : 2012-01-08 11:16:19 handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s Metadata: creation_time : 2012-01-08 11:16:19 handler_name : SoundHandler Output #0, image2, to 'img%03d.png': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.63.104 Stream #0:0(eng): Video: png, rgb24, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 0.02 tbc Metadata: creation_time : 2012-01-08 11:16:19 handler_name : VideoHandler Stream mapping: Stream #0:0 -> #0:0 (h264 -> png) Press [q] to stop, [?] for help frame= 5 fps=0.8 q=0.0 Lsize=N/A time=00:05:00.00 bitrate=N/A video:2310kB audio:0kB subtitle:0 global headers:0kB muxing overhead -100.000930% % ffmpeg -y -i input.mp4 -r 1/60 rimg%03d.png ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 2 2013 20:38:39 with Apple LLVM version 4.2 (clang-425.0.24) (based on LLVM 3.2svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libvo-aacenc --enable-libass --enable-libfdk-aac --enable-libopus --enable-frei0r --enable-libopenjpeg --extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.1/include/openjpeg-1.5 ' libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2012-01-08 11:16:19 encoder : Lavf53.24.0 Duration: 00:03:31.58, start: 0.000000, bitrate: 2359 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 2229 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: creation_time : 2012-01-08 11:16:19 handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s Metadata: creation_time : 2012-01-08 11:16:19 handler_name : SoundHandler Output #0, image2, to 'rimg%03d.png': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.63.104 Stream #0:0(eng): Video: png, rgb24, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 0.02 tbc Metadata: creation_time : 2012-01-08 11:16:19 handler_name : VideoHandler Stream mapping: Stream #0:0 -> #0:0 (h264 -> png) Press [q] to stop, [?] for help frame= 7 fps=0.8 q=0.0 Lsize=N/A time=00:07:00.00 bitrate=N/A dup=0 drop=5280 video:2398kB audio:0kB subtitle:0 global headers:0kB muxing overhead -100.000896% From yogesh.bit2006 at gmail.com Fri Apr 19 09:05:23 2013 From: yogesh.bit2006 at gmail.com (Yogesh Tyagi) Date: Fri, 19 Apr 2013 12:35:23 +0530 Subject: [FFmpeg-user] Not able to generate interlaced video Message-ID: Hi, I was trying to generate h.264 interlaced video.I used following command: ./ffmpeg -i simpson.h264 -vcodec libx264 -x264opts tff=1 temp.h264 ffmpeg version N-31424-g1473833, Copyright (c) 2000-2011 the FFmpeg developers built on Feb 19 2013 14:39:47 with gcc 4.4.5 configuration: --enable-libx264 --enable-gpl --enable-static --disable-shared libavutil 51. 11. 0 / 51. 11. 0 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 24. 4 / 2. 24. 4 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [h264 @ 0x8fde380] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 47.95 (48000/1001) -> 23.98 (48000/2002) Input #0, h264, from 'simpson.h264': Duration: N/A, bitrate: N/A Stream #0.0: Video: h264 (High), yuv420p, 1920x800, 23.98 fps, 23.98 tbr, 1200k tbn, 47.95 tbc File 'temp.h264' already exists. Overwrite ? [y/N] y [buffer @ 0x8fe6780] w:1920 h:800 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 0x8fd9be0] Default settings detected, using medium profile [libx264 @ 0x8fd9be0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 0x8fd9be0] profile High, level 4.0 Output #0, h264, to 'temp.h264': Metadata: encoder : Lavf53.5.0 Stream #0.0: Video: libx264, yuv420p, 1920x800, q=2-31, 200 kb/s, 90k tbn, 23.98 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 10 fps= 0 q=28.0 Lsize= 30kB time=00:00:00.33 bitrate= 742.3kbits/s dup=0 drop=1 video:30kB audio:0kB global headers:0kB muxing overhead 0.000000% frame I:1 Avg QP:14.35 size: 30227 [libx264 @ 0x8fd9be0] frame P:3 Avg QP:24.38 size: 101 [libx264 @ 0x8fd9be0] frame B:6 Avg QP:27.82 size: 72 [libx264 @ 0x8fd9be0] consecutive B-frames: 20.0% 0.0% 0.0% 80.0% [libx264 @ 0x8fd9be0] mb I I16..4: 88.4% 0.4% 11.2% [libx264 @ 0x8fd9be0] mb P I16..4: 0.1% 0.0% 0.0% P16..4: 0.2% 0.0% 0.0% 0.0% 0.0% skip:99.7% [libx264 @ 0x8fd9be0] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.2% 0.0% 0.0% direct: 0.0% skip:99.8% L0:35.1% L1:64.9% BI: 0.0% [libx264 @ 0x8fd9be0] 8x8 transform intra:0.4% inter:25.0% [libx264 @ 0x8fd9be0] coded y,uvDC,uvAC intra: 8.4% 11.9% 11.5% inter: 0.0% 0.0% 0.0% [libx264 @ 0x8fd9be0] i16 v,h,dc,p: 95% 2% 2% 0% [libx264 @ 0x8fd9be0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 51% 14% 27% 0% 0% 0% 1% 0% 7% [libx264 @ 0x8fd9be0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 35% 26% 18% 2% 2% 5% 4% 5% 3% [libx264 @ 0x8fd9be0] i8c dc,h,v,p: 88% 6% 4% 1% [libx264 @ 0x8fd9be0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x8fd9be0] ref P L0: 96.7% 0.0% 3.3% [libx264 @ 0x8fd9be0] ref B L0: 81.5% 14.8% 3.7% [libx264 @ 0x8fd9be0] ref B L1: 98.0% 2.0% [libx264 @ 0x8fd9be0] kb/s:593.82 I inspected the video usinf mediainfo and it shows that the file generated is progressive. # mediainfo-gui temp.h264 General Complete name : temp.h264 Format : AVC Format/Info : Advanced Video Codec File size : 30.2 KiB Duration : 417ms Overall bit rate : 594 Kbps Video Format : AVC Format/Info : Advanced Video Codec Format profile : High at L4.0 Format settings, CABAC : Yes Format settings, ReFrames : 4 frames Duration : 417ms Width : 1 920 pixels Height : 800 pixels Display aspect ratio : 2.40:1 Frame rate mode : Variable Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Writing library : x264 core 120 Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=2 / keyint=250 / keyint_min=23 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=crf / mbtree=1 / crf=23.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00 Am I missing some options to be passed to ffmpeg to generate interlaced content ? Thanks, Yogesh From cehoyos at ag.or.at Fri Apr 19 09:15:08 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 07:15:08 +0000 (UTC) Subject: [FFmpeg-user] Not able to generate interlaced video References: Message-ID: Yogesh Tyagi gmail.com> writes: > I was trying to generate h.264 interlaced video. > I used following command: > > ./ffmpeg -i simpson.h264 -vcodec libx264 -x264opts tff=1 temp.h264 Please try -flags ildct Carl Eugen From cehoyos at ag.or.at Fri Apr 19 09:16:45 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 07:16:45 +0000 (UTC) Subject: [FFmpeg-user] What is the (technical) difference between -r and the fps filter? References: Message-ID: Werner Robitza gmail.com> writes: > ffmpeg -i input -vf fps="fps=1/60" out%03d.png > > or > > ffmpeg -i input -r 1/60 out%03d.png > > It looks like they'd do the same, but when trying it on a video I got > different frame counts, the latter being incorrect That is correct, please use -vf fps See also ticket #1578 Carl Eugen From cehoyos at ag.or.at Fri Apr 19 09:37:05 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 07:37:05 +0000 (UTC) Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th References: <5170B90B.7060300@gmail.com> Message-ID: Dennis Volodomanov gmail.com> writes: > ffmpeg -loglevel 99 -i http://radio.retro-retro.ru In the past, ffmpeg used a ffmpeg-specific user agent when opening http streams, it now uses a common string to allow access to http servers that expect a "normal" browser. Workaround is to use: $ ffmpeg -user-agent something_random -i http://radio.retro-retro.ru See ticket #2265 for a similar (Google-) page that also behaves strangely, unfortunately in the opposite way. Carl Eugen From cehoyos at ag.or.at Fri Apr 19 09:47:06 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 07:47:06 +0000 (UTC) Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? References: Message-ID: L. Lee sbcglobal.net> writes: > converting an excerpt of longer duration results > in a/v sync issues Please provide a sample that allows to reproduce this problem. Carl Eugen From volodomanov at gmail.com Fri Apr 19 09:55:22 2013 From: volodomanov at gmail.com (Dennis Volodomanov) Date: Fri, 19 Apr 2013 17:55:22 +1000 Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th In-Reply-To: References: <5170B90B.7060300@gmail.com> Message-ID: <5170F86A.7030103@gmail.com> Carl Eugen Hoyos wrote: > > In the past, ffmpeg used a ffmpeg-specific user agent > when opening http streams, it now uses a common string > to allow access to http servers that expect a "normal" > browser. > > Workaround is to use: > $ ffmpeg -user-agent something_random -i http://radio.retro-retro.ru > > See ticket #2265 for a similar (Google-) page that > also behaves strangely, unfortunately in the > opposite way. Thank you - that works. I saw that change, but would've never guessed that the server would fail to reply without a user-agent. Dennis From ubitux at gmail.com Fri Apr 19 10:05:57 2013 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Fri, 19 Apr 2013 10:05:57 +0200 Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th In-Reply-To: <5170F86A.7030103@gmail.com> References: <5170B90B.7060300@gmail.com> <5170F86A.7030103@gmail.com> Message-ID: <20130419080557.GT10843@leki> On Fri, Apr 19, 2013 at 05:55:22PM +1000, Dennis Volodomanov wrote: > Carl Eugen Hoyos wrote: > > > >In the past, ffmpeg used a ffmpeg-specific user agent > >when opening http streams, it now uses a common string > >to allow access to http servers that expect a "normal" > >browser. > > > >Workaround is to use: > >$ ffmpeg -user-agent something_random -i http://radio.retro-retro.ru > > > >See ticket #2265 for a similar (Google-) page that > >also behaves strangely, unfortunately in the > >opposite way. > > Thank you - that works. I saw that change, but would've never > guessed that the server would fail to reply without a user-agent. > I think the problem is that the website has two mode for its front page. If the user-agent has "Mozilla" in it, it is in browser mode and display some HTML for the stream, and otherwise it is in streaming mode... -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From volodomanov at gmail.com Fri Apr 19 09:53:29 2013 From: volodomanov at gmail.com (dvlpr) Date: Fri, 19 Apr 2013 00:53:29 -0700 (PDT) Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th In-Reply-To: References: <5170B90B.7060300@gmail.com> Message-ID: <1366358009086-4658588.post@n4.nabble.com> Carl Eugen Hoyos wrote > In the past, ffmpeg used a ffmpeg-specific user agent > when opening http streams, it now uses a common string > to allow access to http servers that expect a "normal" > browser. > > Workaround is to use: > $ ffmpeg -user-agent something_random -i http://radio.retro-retro.ru > > See ticket #2265 for a similar (Google-) page that > also behaves strangely, unfortunately in the > opposite way. Thank you - that works. I saw that change, but would've never guessed that the server would fail to reply without a user-agent. Dennis -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Problem-with-decoding-an-MP3-stream-introduced-after-April-6th-tp4658580p4658588.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From volodomanov at gmail.com Fri Apr 19 10:18:14 2013 From: volodomanov at gmail.com (Dennis Volodomanov) Date: Fri, 19 Apr 2013 18:18:14 +1000 Subject: [FFmpeg-user] Problem with decoding an MP3 stream introduced after April 6th In-Reply-To: <20130419080557.GT10843@leki> References: <5170B90B.7060300@gmail.com> <5170F86A.7030103@gmail.com> <20130419080557.GT10843@leki> Message-ID: <5170FDC6.9040703@gmail.com> Cl?ment B?sch wrote: > I think the problem is that the website has two mode for its front page. > If the user-agent has "Mozilla" in it, it is in browser mode and display > some HTML for the stream, and otherwise it is in streaming mode... Yes, that's possible. It was similar for some other streaming sites as well, but they could all be using a similar concept. Dennis From cehoyos at ag.or.at Fri Apr 19 10:19:32 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 08:19:32 +0000 (UTC) Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? References: Message-ID: L. Lee sbcglobal.net> writes: > https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip Could you test if the mpdecimate filter works better for this sample? $ ffmpeg -i sample -vf mpdecimate -r 24000/1001 -qscale 2 out.avi The fieldmatch filter does not help because your sample does not contain any interlaced frames, so there are no fields to match... In the future, the decimate filter will hopefully support dropping three of five frames. As an alternative, you could try to interlace your video to get the telecine pattern back (it appears your video was de-interlaced after telecine was used) and feed the result to fieldmatch. I am not saying that is the best option, but you could test: -vf interlace,fieldmatch... Carl Eugen From paulj at fastmail.es Fri Apr 19 12:52:56 2013 From: paulj at fastmail.es (paulj at fastmail.es) Date: Fri, 19 Apr 2013 12:52:56 +0200 Subject: [FFmpeg-user] Syntax for ffmpeg to listen on a port for a RTMP stream Message-ID: <51712208.19345.804377@paulj.fastmail.es> Hello: I want ffmpeg to listen on a given TCP port for an incoming video stream, but I can't get the right syntax. I have tried: ffmpeg -listen -i rtmp://localhost:6060 -vcodec copy -acodec copy bla.mp4 ffmpeg -i rtmp://localhost:6060 -listen -vcodec copy -acodec copy bla.mp4 ffmpeg -i rtmp://localhost:6060 listen -vcodec copy -acodec copy bla.mp4 And in all cases I get "TCP connection to localhost:6060 failed. Connection refused", as if ffmpeg was trying to open a connection, instead of listening. With RTSP, there's a "-rtsp_flags" option where you can add parameters, but the documentation doesn't say anything about RTMP. I'm using the binary build of 1.2.7 for OS X (the one linked to in the webpage). The build DOES support RTMP, as shown with the "-protocols" option. Paulo. ------------------------------------------- "Oh, don't worry kids. Most of you will never fall in love. You'll just marry for fear of dying alone." The Simpsons. ------------------------------------------- From h.reindl at thelounge.net Fri Apr 19 13:43:27 2013 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 19 Apr 2013 13:43:27 +0200 Subject: [FFmpeg-user] Syntax for ffmpeg to listen on a port for a RTMP stream In-Reply-To: <51712208.19345.804377@paulj.fastmail.es> References: <51712208.19345.804377@paulj.fastmail.es> Message-ID: <51712DDF.9030301@thelounge.net> Am 19.04.2013 12:52, schrieb paulj at fastmail.es: > I want ffmpeg to listen on a given TCP port for an incoming video stream, but I can't > get the right syntax. I have tried: > > ffmpeg -listen -i rtmp://localhost:6060 -vcodec copy -acodec copy bla.mp4 > > ffmpeg -i rtmp://localhost:6060 -listen -vcodec copy -acodec copy bla.mp4 > > ffmpeg -i rtmp://localhost:6060 listen -vcodec copy -acodec copy bla.mp4 > > And in all cases I get "TCP connection to localhost:6060 failed. Connection refused", > as if ffmpeg was trying to open a connection, instead of listening why should ffmpeg LISTEN on a port? what you search is ffserver -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature URL: From paulj at fastmail.es Fri Apr 19 13:58:06 2013 From: paulj at fastmail.es (paulj at fastmail.es) Date: Fri, 19 Apr 2013 13:58:06 +0200 Subject: [FFmpeg-user] Syntax for ffmpeg to listen on a port for a RTMP stream In-Reply-To: <51712DDF.9030301@thelounge.net> References: <51712208.19345.804377@paulj.fastmail.es>, <51712DDF.9030301@thelounge.net> Message-ID: <5171314E.20514.BBEB0D@paulj.fastmail.es> > Am 19.04.2013 12:52, schrieb paulj at fastmail.es: > > I want ffmpeg to listen on a given TCP port for an incoming video > stream, but I can't > > get the right syntax. I have tried: > > > > ffmpeg -listen -i rtmp://localhost:6060 -vcodec copy -acodec copy > bla.mp4 > > > > ffmpeg -i rtmp://localhost:6060 -listen -vcodec copy -acodec copy > bla.mp4 > > > > ffmpeg -i rtmp://localhost:6060 listen -vcodec copy -acodec copy > bla.mp4 > > > > And in all cases I get "TCP connection to localhost:6060 failed. > Connection refused", > > as if ffmpeg was trying to open a connection, instead of listening > > why should ffmpeg LISTEN on a port? > what you search is ffserver > > Nope. ffserver listens on a port and SERVES streams to clients connecting to it. I need ffmpeg to listen on a port and RECEIVE a stream sent by a client. With RTSP you can do it, i.e.: ????????ffmpeg -rtsp_flags listen -f rtsp -i rtsp://localhost:5050/ -vcodec copy -acodec copy video.mp4 What I'm asking is the equivalent syntax in RTMP. Paulo. ------------------------------------------- "Oh, don't worry kids. Most of you will never fall in love. You'll just marry for fear of dying alone." The Simpsons. ------------------------------------------- From tevans.uk at googlemail.com Fri Apr 19 15:19:34 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 19 Apr 2013 14:19:34 +0100 Subject: [FFmpeg-user] Syntax for ffmpeg to listen on a port for a RTMP stream In-Reply-To: <5171314E.20514.BBEB0D@paulj.fastmail.es> References: <51712208.19345.804377@paulj.fastmail.es> <51712DDF.9030301@thelounge.net> <5171314E.20514.BBEB0D@paulj.fastmail.es> Message-ID: On Fri, Apr 19, 2013 at 12:58 PM, wrote: > > Nope. ffserver listens on a port and SERVES streams to clients connecting to it. I > need ffmpeg to listen on a port and RECEIVE a stream sent by a client. > With RTSP you can do it, i.e.: > > ffmpeg -rtsp_flags listen -f rtsp -i rtsp://localhost:5050/ -vcodec copy -acodec copy > video.mp4 > > What I'm asking is the equivalent syntax in RTMP. > > > Paulo. Can you use netcat to do this, pushing the data to ffmpeg over stdin using pipe protocol: nc -l localhost 5050 | ffmpeg -f rtsp -i pipe:0 ... Cheers Tom From swapnil.singh1520 at gmail.com Fri Apr 19 15:23:42 2013 From: swapnil.singh1520 at gmail.com (Swapnil Singh) Date: Fri, 19 Apr 2013 18:53:42 +0530 Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc Message-ID: *Hi everyone, *I was trying to transcode a mp4 file to mpg container using libx264 avc. After some trails i was able to transcode it successfully.But was only able to play the the stream using ffplay but not with mplayer or Vlc.i am including the logs for the playback and transcode below.Please guide. trancode coomandline ffmpeg -i 20130301_oblivion_trailer2i_4000.mp4 -vcodec libx264 -bufsize 500000000 -profile:v baseline -level:v 3 -s 720x576 -b:v 3.3M -r 25 temp.mpg *Using ffplay* #ffplay temp.mpg FFplay version 0.6.3-rpmfusion, Copyright (c) 2003-2010 the FFmpeg developers built on May 5 2011 19:19:44 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib --mandir=/usr/share/man --arch=i686 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i686 -mtune=atom -fasynchronous-unwind-tables' --extra-version=rpmfusion --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib --cpu=i686 --enable-runtime-cpudetect libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0xb23017e0]max_analyze_duration reached Input #0, mpeg, from 'temp.mpg': Duration: 00:02:30.64, start: 0.500000, bitrate: 3889 kb/s Stream #0.0[0x1e0]: Video: h264, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x1c0]: Audio: mp2, 44100 Hz, 2 channels, s16, 128 kb/s 28.79 A-V: -0.000 s:0.0 aq= 319KB vq= 6662KB sq= 0B f=0/709 0/0 *Using mplayer * mplayer temp.mpg MPlayer SVN-r31628-4.4.4 (C) 2000-2010 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing temp.mpg. MPEG-PS file format detected. MPEG: FATAL: EOF while searching for sequence header. Video: Cannot read properties. ========================================================================== Opening audio decoder: [mp3lib] MPEG layer-2, layer-3 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400) Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3) ========================================================================== AO: [pulse] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 4.3 (04.3) of 148.8 (02:28.8) 0.2% Exiting... (End of file) *Using Vlc* #vlc temp.mpg VLC media player 1.1.12 The Luggage (revision exported) Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS") Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE") [0x8b2b8fc] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface. Blocked: call to setlocale(6, "") From tevans.uk at googlemail.com Fri Apr 19 15:41:26 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 19 Apr 2013 14:41:26 +0100 Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc In-Reply-To: References: Message-ID: On Fri, Apr 19, 2013 at 2:23 PM, Swapnil Singh wrote: > *Hi everyone, > *I was trying to transcode a mp4 file to mpg container using libx264 avc. > After some trails i was able to transcode it successfully.But was only able > to play the the stream using ffplay but not with mplayer or Vlc.i am > including the logs for the playback and transcode below.Please guide. > > trancode coomandline > ffmpeg -i 20130301_oblivion_trailer2i_4000.mp4 -vcodec libx264 -bufsize > 500000000 -profile:v baseline -level:v 3 -s 720x576 -b:v 3.3M -r 25 > temp.mpg MPEG PS is only supposed to have either MPEG 2 Part 1 or Mpeg 2 Part 2 video. ffmpeg can understand, so ffplay can also. mplayer would probably also understand if you used demuxer lavf. However, if you want a file that will play in most players, you will need to choose a format that can contain the codec you require. MPEG TS or MP4 or mkv would be my choices. Cheers Tom From cehoyos at ag.or.at Fri Apr 19 16:19:21 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 14:19:21 +0000 (UTC) Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc References: Message-ID: Swapnil Singh gmail.com> writes: > trancode coomandline > ffmpeg -i 20130301_oblivion_trailer2i_4000.mp4 -vcodec libx264 -bufsize > 500000000 -profile:v baseline -level:v 3 -s 720x576 -b:v 3.3M -r 25 > temp.mpg > > *Using ffplay* > #ffplay temp.mpg > FFplay version 0.6.3-rpmfusion This is ancient, please update to current git head. > MPlayer SVN-r31628-4.4.4 (C) 2000-2010 MPlayer Team This is also old, but it is possible that you need -demuxer lavf to play back h264 in mpeg. Carl Eugen From cehoyos at ag.or.at Fri Apr 19 16:20:39 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 19 Apr 2013 14:20:39 +0000 (UTC) Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc References: Message-ID: Tom Evans googlemail.com> writes: > MPEG PS is only supposed to have either MPEG 2 Part 1 > or Mpeg 2 Part 2 video. Program streams may also contain h264 (and several other video codecs iirc). Carl Eugen From werner.robitza at gmail.com Fri Apr 19 17:06:38 2013 From: werner.robitza at gmail.com (Werner Robitza) Date: Fri, 19 Apr 2013 17:06:38 +0200 Subject: [FFmpeg-user] What is the (technical) difference between -r and the fps filter? In-Reply-To: References: Message-ID: > > That is correct, please use -vf fps > See also ticket #1578 > Thanks Carl Eugen. However the question still remains: what's the actual difference between the filter and -r? Is it generally not recommended to use the latter at all? Seems all over the place in many examples? From ted at ukegoldberg.com Fri Apr 19 17:11:35 2013 From: ted at ukegoldberg.com (ted at ukegoldberg.com) Date: Fri, 19 Apr 2013 11:11:35 -0400 Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) Message-ID: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> I am trying out FFMPEG on many recommendations. I am working with recovered deleted videos that are now Protected MPEG-4 Movie. I don't think they were protected before they were deleated. They are from my point and shot camera. The are now jerky and blurry. I converted them to AVI with FFMPEG. They are still jerky and blurry. Any suggestions? Thanks, Ted From seandarcy2 at gmail.com Fri Apr 19 17:13:45 2013 From: seandarcy2 at gmail.com (sean darcy) Date: Fri, 19 Apr 2013 11:13:45 -0400 Subject: [FFmpeg-user] vbr with libfdk_aac ? In-Reply-To: References: Message-ID: On 04/18/2013 10:45 PM, Timothy Gu wrote: > On Thu, Apr 18, 2013 at 11:57 AM, sean darcy wrote: >> ffmpeg -i 17.wav -vn -c:a libfdk_aac -vbr 2 -afterburner 1 17.m4a >> ffmpeg version N-52058-g423089e Copyright (c) 2000-2013 the FFmpeg >> developers >> built on Apr 13 2013 19:19:37 with gcc 4.7.2 (GCC) 20120921 (Red Hat >> 4.7.2-2) >> configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg >> --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man >> --arch=x86_64 --cpu=amdfam10 --optflags='-Ofast -march=native -mtune=native >> -fopenmp -fomit-frame-pointer -pipe -fPIC' --enable-avresample >> --enable-libopencore-amrnb --enable-libopencore-amrwb >> --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd >> --enable-frei0r --enable-libfdk-aac --enable-gnutls --enable-libass >> --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack >> --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal >> --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse >> --enable-librtmp --enable-libschroedinger --enable-libspeex >> --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx >> --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter >> --enable-postproc --enable-pthreads --disable-static --enable-shared >> --enable-gpl --enable-nonfree --enable-zlib libavutil 52. 26.100 / >> 52. 26.100 >> libavcodec 55. 2.100 / 55. 2.100 >> libavformat 55. 2.100 / 55. 2.100 >> libavdevice 55. 0.100 / 55. 0.100 >> libavfilter 3. 53.101 / 3. 53.101 >> libavresample 1. 1. 0 / 1. 1. 0 >> libswscale 2. 2.100 / 2. 2.100 >> libswresample 0. 17.102 / 0. 17.102 >> libpostproc 52. 3.100 / 52. 3.100 >> [wav @ 0xe1ba00] max_analyze_duration 5000000 reached at 5015510 >> microseconds >> Guessed Channel Layout for Input Stream #0.0 : stereo >> Input #0, wav, from '17.wav': >> Duration: 00:30:31.07, bitrate: 1411 kb/s >> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, >> s16, 1411 kb/s >> [libfdk_aac @ 0xe43280] Note, the VBR setting is unsupported and only works >> with some parameter combinations > > Question for Sean: > > Is the output OK? If it is, then it shouldn't be a problem. (Also see > question a) below) > > I also have a couple of questions for developers for the library. > Please correct me (and please don't consider me rude) if I am wrong. > > a) According to the official libfdk-aac available in mstorsjo's repo, > it does not say anything about VBR, or values between 1-5 for param > AACENC_BITRATEMODE. But in line 185 of libavcodec/libfdk-aacenc.c, > ffmpeg is setting it for VBR. In my opinion, not say it can do VBR > stuff means it can't. > > b) Shouldn't -vbr be -q:a? And if it is for abbreviating "-flags > +qscale -global_quality [1-5]", shouldn't it also map to other codecs > requiring this option? > > Timothy G. > Well the output is ok, but I'm expecting a better result than 90kb/s for just speech. Wasted some time checking out results of different vbr settings. ffprobe for ffmpeg -i 17.wav -vn -c:a libfdk_aac -vbr 2 -afterburner 1 17-ffmpeg.m4a Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '17-ffmpeg.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.2.100 Duration: 00:30:31.11, start: 0.046440, bitrate: 91 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 90 kb/s Same but -vbr 5 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '17-ffmpeg.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.2.100 Duration: 00:30:31.11, start: 0.046440, bitrate: 187 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 185 kb/s So the vbr spec is doing something. But is it just changing a CBR or is it actually allocating bits to different passages? I also used fdkaac - a direct CLI for libfdk_aac. ffrobe on fdkaac -p 2 -m 2 17.wav Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '17-fdkaac.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2013-04-19 14:42:11 encoder : fdkaac 0.2.0, libfdk-aac 3.3.3, VBR mode 2 Duration: 00:30:31.13, start: 0.000000, bitrate: 91 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 90 kb/s And VBR 5: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '17-fdkaac.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2013-04-19 14:52:02 encoder : fdkaac 0.2.0, libfdk-aac 3.3.3, VBR mode 5 Duration: 00:30:31.13, start: 0.000000, bitrate: 187 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 185 kb/s So ffmpeg is generating identical bit rates for vbr settings as fdkaac. But again, hard to know if this is just a CBR setting or really VBR. sean From tevans.uk at googlemail.com Fri Apr 19 17:38:35 2013 From: tevans.uk at googlemail.com (Tom Evans) Date: Fri, 19 Apr 2013 16:38:35 +0100 Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc In-Reply-To: References: Message-ID: On Fri, Apr 19, 2013 at 3:20 PM, Carl Eugen Hoyos wrote: > Tom Evans googlemail.com> writes: > >> MPEG PS is only supposed to have either MPEG 2 Part 1 >> or Mpeg 2 Part 2 video. > > Program streams may also contain h264 (and several > other video codecs iirc). Hmm, if I try to put h264 into a PS, I get a file that mplayer quickly loses sync on: > $ ffmpeg -y -i Oblivion_Trailer_720p.mp4 -c:v libx264 -bufsize 500000000 -profile:v baseline -level:v 3 -s 720x576 -b:v 3.3M temp.mpg ffmpeg version N-52268-g42bcc40 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 19 2013 15:23:31 with gcc 4.6.4 (FreeBSD Ports Collection) 20121102 (prerelease) configuration: --cc=/usr/local/bin/gcc46 --extra-libs=-L/usr/local/lib --extra-cflags=-I/usr/local/include --prefix=/usr/local --mandir=/usr/local/man --enable-libx264 --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-nonfree --enable-gpl --enable-version3 --enable-libtheora --enable-libvorbis --enable-ffmpeg --enable-ffplay --enable-ffprobe libavutil 52. 27.100 / 52. 27.100 libavcodec 55. 4.100 / 55. 4.100 libavformat 55. 3.100 / 55. 3.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 58.100 / 3. 58.100 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Oblivion_Trailer_720p.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42mp41 creation_time : 2012-12-09 02:51:05 copyright : copyright-eng : Duration: 00:02:36.03, start: 0.000000, bitrate: 5641 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x544, 5506 kb/s, 23.98 fps, 23.98 tbr, 2997 tbn, 5994 tbc Metadata: creation_time : 2012-12-09 02:51:05 handler_name : Apple Video Media Handler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 129 kb/s Metadata: creation_time : 2012-12-09 02:51:05 handler_name : Apple Sound Media Handler [libx264 @ 0x805020400] VBV maxrate unspecified, assuming CBR [libx264 @ 0x805020400] VBV buffer (500000) > level limit (10000) [libx264 @ 0x805020400] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle Cache64 [libx264 @ 0x805020400] profile Constrained Baseline, level 3.0 Output #0, mpeg, to 'temp.mpg': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42mp41 copyright-eng : copyright : encoder : Lavf55.3.100 Stream #0:0(eng): Video: h264, yuv420p, 720x576, q=-1--1, 3300 kb/s, 90k tbn, 23.98 tbc Metadata: creation_time : 2012-12-09 02:51:05 handler_name : Apple Video Media Handler Stream #0:1(eng): Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2012-12-09 02:51:05 handler_name : Apple Sound Media Handler Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> mp2) Press [q] to stop, [?] for help frame= 3741 fps= 27 q=-2.0 Lsize= 74358kB time=00:02:36.03 bitrate=3903.8kbits/s video:71598kB audio:2438kB subtitle:0 global headers:0kB muxing overhead 0.433945% [libx264 @ 0x805020400] frame I:262 Avg QP:13.22 size: 36029 [libx264 @ 0x805020400] frame P:3479 Avg QP:15.24 size: 18361 [libx264 @ 0x805020400] mb I I16..4: 38.1% 0.0% 61.9% [libx264 @ 0x805020400] mb P I16..4: 6.9% 0.0% 12.6% P16..4: 33.8% 19.6% 7.7% 0.0% 0.0% skip:19.4% [libx264 @ 0x805020400] final ratefactor: 15.59 [libx264 @ 0x805020400] coded y,uvDC,uvAC intra: 63.0% 57.2% 38.6% inter: 34.3% 35.2% 8.4% [libx264 @ 0x805020400] i16 v,h,dc,p: 54% 13% 9% 24% [libx264 @ 0x805020400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 35% 15% 15% 5% 7% 8% 4% 7% 4% [libx264 @ 0x805020400] i8c dc,h,v,p: 62% 12% 21% 5% [libx264 @ 0x805020400] ref P L0: 78.3% 15.2% 6.5% [libx264 @ 0x805020400] kb/s:3759.09 > $ mplayer temp.mpg MPlayer UNKNOWN-4.6.4 (C) 2000-2013 MPlayer Team Playing temp.mpg. Cache fill: 0.00% (0 bytes) libavformat version 55.3.100 (internal) libavformat file format detected. [mpeg @ 0xded460]max_analyze_duration 5000000 reached at 5004000 microseconds [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (mp2), -aid 0 VIDEO: [H264] 720x576 0bpp 23.976 fps 0.0 kbps ( 0.0 kbyte/s) Could not find a UTF-8 locale, some keys will not be handled. ========================================================================== Forced video codec: ffh264vdpau Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.4.100 (internal) Selected video codec: [ffh264vdpau] vfm: ffmpeg (FFmpeg H.264 (VDPAU)) ========================================================================== ========================================================================== Trying to force audio codec driver family hwac3... Requested audio codec family [mpg123] (afm=mpg123) not available. Enable it at compilation. Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 2 ch, floatle, 128.0 kbit/4.17% (ratio: 16000->384000) Selected audio codec: [ffmp2float] afm: ffmpeg (FFmpeg MPEG layer-1 and layer-2 audio) ========================================================================== AO: [oss] 48000Hz 2ch s16le (2 bytes per sample) Starting playback... [VD_FFMPEG] Trying pixfmt=0. Movie-Aspect is undefined - no prescaling applied. VO: [vdpau] 720x576 => 720x576 H.264 VDPAU acceleration [fs] [VD_FFMPEG] XVMC-accelerated MPEG-2. [VD_FFMPEG] Trying pixfmt=0. [VD_FFMPEG] XVMC-accelerated MPEG-2. A: 0.9 V: 0.6 A-V: 0.311 ct: 0.003 0/ 0 ??% ??% ??,?% 1 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.0 V: 0.8 A-V: 0.147 ct: 0.032 0/ 0 ??% ??% ??,?% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.0 V: 0.9 A-V: 0.106 ct: 0.041 0/ 0 36% 89% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.0 V: 0.9 A-V: 0.117 ct: 0.045 0/ 0 34% 86% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.1 V: 0.9 A-V: 0.160 ct: 0.049 0/ 0 31% 84% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.1 V: 0.9 A-V: 0.192 ct: 0.053 0/ 0 30% 79% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.2 V: 0.9 A-V: 0.234 ct: 0.057 0/ 0 28% 74% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.2 V: 0.9 A-V: 0.266 ct: 0.062 0/ 0 26% 70% 0.4% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.5 V: 1.2 A-V: 0.305 ct: 0.070 0/ 0 18% 48% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.6 V: 1.2 A-V: 0.337 ct: 0.074 0/ 0 17% 47% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.6 V: 1.2 A-V: 0.380 ct: 0.078 0/ 0 17% 45% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.6 V: 1.2 A-V: 0.412 ct: 0.082 0/ 0 16% 43% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.7 V: 1.2 A-V: 0.454 ct: 0.087 0/ 0 16% 42% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 1.7 V: 1.2 A-V: 0.486 ct: 0.091 0/ 0 15% 41% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 2.0 V: 1.5 A-V: 0.514 ct: 0.099 0/ 0 12% 32% 0.3% 6 0 99% No pts value from demuxer to use for frame! pts after filters MISSING A: 2.1 V: 1.5 A-V: 0.557 ct: 0.103 0/ 0 12% 31% 0.3% 6 0 99% A-V keeps increasing after that. I don't know why mplayer says UNKNOWN, I built it about 10 minutes ago from f6ad441. Cheers Tom From dalezak at gmail.com Tue Apr 16 19:32:11 2013 From: dalezak at gmail.com (Dale Zak) Date: Tue, 16 Apr 2013 11:32:11 -0600 Subject: [FFmpeg-user] FFMPEG Fade In Watermark Only Message-ID: <5D47262B5E7D4C2D92F1BA14FE4A7857@gmail.com> Dear FFMPEG users, I'm trying to watermark a video using FFMPEG and would only like the watermark to fade in. The following code adds the watermark however the fade in effect is applied to the entire video and not just the watermark. ffmpeg -loop 1 -t 3 -i background.png -i silence.mp3 -shortest -s 640x360 -vf "movie=watermark.png [watermark]; [in] [watermark] overlay=main_w/2-overlay_w/2:main_h/2-overlay_h/2,fade=in:0:30 [out]" -y output.mpg The problem is my video clips have a white background so would like to maintain the white across all clips. I've searched online but have been unable to find a good example on how to apply the fade in only to the overlay. I posted the question on SuperUser http://superuser.com/questions/567059/ffmpeg-fade-in-watermark-only but unfortunately have been unable to find a solution. Any help regarding this issue would be greatly appreciated. Cheers, Dale From dennis.koch at outlook.com Fri Apr 19 12:32:27 2013 From: dennis.koch at outlook.com (Dennis Koch) Date: Fri, 19 Apr 2013 12:32:27 +0200 Subject: [FFmpeg-user] ffmpeg decoder color space / ventuz Message-ID: hello there, what colorspace is the ffmpeg decoder using? all our videos in 709 and rgb are darker than the original in the ventuz output monitor. and afaik ventuz uses ffmpeg as decoder. From doug at doughandy.com Wed Apr 17 19:37:56 2013 From: doug at doughandy.com (Doug Handy) Date: Wed, 17 Apr 2013 13:37:56 -0400 Subject: [FFmpeg-user] Intent of -copyinkf and non-key frames in video stream copy Message-ID: I am trying to clarify if I am misunderstanding the intent of the -copyinkf flag, or if what I want to do is not possible without transcoding. I would like to extract frame accurate subsets from an input movie while still getting the speed advantage of a video stream copy. Normally a video stream copy must start on key frame boundaries, and i understand why. But the documentation includes this subset: ?-copyinkf[:stream_specifier] (output,per-stream)? When doing stream copy, copy also non-key frames found at the beginning. I am not finding much more information about this option at all. Perhaps it is wishful thinking on my part and reading it to mean what I want it to mean, but I was hoping this would mean that when specified and your start location was not on a key frame, that it would generate a proper keyframe from the prior keyframe and intermediate frames up to the desired starting point. I did find a cvslog message from Jan 2012 which showed a bug with this option: This option only applies to streamcopy, but is currently processed only when encoding. The change log is at https://lists.ffmpeg.org/pipermail/ffmpeg-cvslog/2012-January/045759.html However, I am not getting it to produce the desired effect. It always subsets on a keyframe boundary. So prior to filing a bug report, I am trying to find out if I just misunderstand the intent of -copyinkf. So the question is: Should -copyinkf allow me to start a video stream copy on a non-key frame position? If it helps, I can typically copy the video stream only and discard the audio stream. Thanks From jacobhameiri at gmail.com Tue Apr 16 11:52:03 2013 From: jacobhameiri at gmail.com (Jacobs) Date: Tue, 16 Apr 2013 02:52:03 -0700 (PDT) Subject: [FFmpeg-user] reduce laltency when streaming udp multicast In-Reply-To: References: <1366018991831-4658488.post@n4.nabble.com> Message-ID: <1366105923271-4658505.post@n4.nabble.com> Thanks for your help. > It is possible that -probesize and -analyzeduration > help for your problem (I don't know). what values of -probesize and -analyzeduration would you recommend ? and in what part of the commandline they should be added ? > ultrafast and zerolatency have no effect on mpeg4 > encoding. -probesize and -analyzeduration are there because I am testing libx264 ang mpeg4. ( btw for less latency would you recommend mpeg4 or h264 ?) > Some of your configure flags are default flags and > have no effect: are the unnecessary configure flags affecting the performance ? also, is there a good way to find out what part of the process is causing the delay ? and are there any buffers I can ( and should ) disable ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/reduce-laltency-when-streaming-udp-multicast-tp4658488p4658505.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mingai426 at gmail.com Thu Apr 18 16:04:45 2013 From: mingai426 at gmail.com (JinMeng) Date: Thu, 18 Apr 2013 22:04:45 +0800 Subject: [FFmpeg-user] User-Question Message-ID: Hi, Thanks for ffmpeg. Now I want to use ffmpeg in iOS development. But ffmpeg doesn't support mp3 encoding. So I download libmp3lame kit and compiled it as iOS version. Now I used "--enable-libmp3lame" option, but configuration is failed. "ERROR: libmp3lame >= 3.98.3 not found" Please help me. How can I compile ffmpeg with libmp3lame as iOS version? Hope your kind reply. Thanks. From tovivekvijayan at gmail.com Fri Apr 19 09:24:47 2013 From: tovivekvijayan at gmail.com (vivek) Date: Fri, 19 Apr 2013 00:24:47 -0700 (PDT) Subject: [FFmpeg-user] RTMP Live Video Stream Mixing Message-ID: <1366356287938-4658584.post@n4.nabble.com> Hi, I have around four live rtmp streams coming to my server. Is there any way through ffmpeg i can mix these video streams in different ways? I would like to see the four streams as one stream in a matrix display. Please advice. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/RTMP-Live-Video-Stream-Mixing-tp4658584.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From blacktrash at gmx.net Sat Apr 20 06:01:12 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Sat, 20 Apr 2013 05:01:12 +0100 Subject: [FFmpeg-user] What is the (technical) difference between -r and the fps filter? In-Reply-To: References: Message-ID: <20130420040112.GB31816@krille.blacktrash.org> * Werner Robitza on Friday, April 19, 2013 at 17:06:38 +0200 >> That is correct, please use -vf fps >> See also ticket #1578 > > Thanks Carl Eugen. > > However the question still remains: what's the actual difference between > the filter and -r? > Is it generally not recommended to use the latter at all? Seems all over > the place in many examples? Or is the message that -r or -r:v is deprecated? Or in the process of being deprecated? -- Was hei?t hier Dogma, ich bin Underdogma! [ What the hell do you mean dogma, I am underdogma. ] free movies --->>> http://www.blacktrash.org/underdogma http://itunes.apple.com/podcast/underdogma-movies/id363423596 From llee040 at sbcglobal.net Sat Apr 20 07:43:38 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Sat, 20 Apr 2013 00:43:38 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: Message-ID: On 4/19/13 2:47 AM, "Carl Eugen Hoyos" wrote: > L. Lee sbcglobal.net> writes: > >> converting an excerpt of longer duration results >> in a/v sync issues > > Please provide a sample that allows to reproduce > this problem. > > Carl Eugen Thanks. Here's the two-minute source: https://dl.dropboxusercontent.com/u/42718751/audio_offset_and_pixelation.zip Here's the ffmpeg output with pixelation and a/v sync issues: https://dl.dropboxusercontent.com/u/42718751/audio_offset_and_pixelation_out put.zip Here's the practically flawless Mencoder output: https://dl.dropboxusercontent.com/u/42718751/audio_offset_and_pixelatio_menc oder.zip Here's the ffmpeg command and feedback: ffmpeg -i audio_offset_and_pixelation.mpg -vf "fieldmatch=order=tff:combmatch=none" -pix_fmt yuv420p -c:v libx264 -preset veryslow -tune film -qp 22 -r 23.976 -s 1280x720 -aspect 16:9 audio_offset_and_pixelation.mp4 ffmpeg version N-37959-g20e99a9 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 16 2013 13:28:23 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-filter=fieldmatch --enable-libmp3lame --enable-gpl --enable-libfaac --enable-pthreads --enable-ffplay --enable-nonfree --enable-libx264 libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 56.101 / 3. 56.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 0x7feff3821a00] mmco: unref short failure Last message repeated 1 times Input #0, mpegts, from '/audio_offset_and_pixelation.mpg': Duration: 00:02:00.69, start: 1.404767, bitrate: 11454 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (Main) (HDMV / 0x564D4448), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s [libx264 @ 0x7feff381a000] using SAR=1/1 [libx264 @ 0x7feff381a000] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7feff381a000] profile High, level 5.0 [libx264 @ 0x7feff381a000] 264 - core 130 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=22 ip_ratio=1.40 pb_ratio=1.30 aq=0 Output #0, mp4, to '/audio_offset_and_pixelation.mp4': Metadata: encoder : Lavf55.2.100 Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 23.98 tbc Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, 5.1, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (ac3 -> libfaac) Press [q] to stop, [?] for help [h264 @ 0x7feff505d400] mmco: unref short failure [h264 @ 0x7feff505ec00] mmco: unref short failure frame= 2896 fps=9.6 q=32750.0 Lsize= 26623kB time=00:02:00.70 bitrate=1806.9kbits/s dup=0 drop=4336 video:23709kB audio:2835kB subtitle:0 global headers:0kB muxing overhead 0.297858% [libx264 @ 0x7feff381a000] frame I:21 Avg QP:19.00 size: 34628 [libx264 @ 0x7feff381a000] frame P:725 Avg QP:22.00 size: 16724 [libx264 @ 0x7feff381a000] frame B:2150 Avg QP:23.73 size: 5314 [libx264 @ 0x7feff381a000] consecutive B-frames: 3.5% 4.6% 12.7% 23.5% 25.7% 22.4% 3.6% 3.0% 0.9% [libx264 @ 0x7feff381a000] mb I I16..4: 52.2% 39.7% 8.1% [libx264 @ 0x7feff381a000] mb P I16..4: 7.5% 8.9% 1.9% P16..4: 25.8% 6.8% 4.8% 0.3% 0.1% skip:44.0% [libx264 @ 0x7feff381a000] mb B I16..4: 0.4% 1.0% 0.2% B16..8: 23.1% 3.4% 0.8% direct: 3.2% skip:67.8% L0:42.9% L1:46.9% BI:10.2% [libx264 @ 0x7feff381a000] 8x8 transform intra:50.0% inter:71.2% [libx264 @ 0x7feff381a000] direct mvs spatial:99.6% temporal:0.4% [libx264 @ 0x7feff381a000] coded y,uvDC,uvAC intra: 47.9% 57.2% 21.3% inter: 9.1% 15.0% 0.6% [libx264 @ 0x7feff381a000] i16 v,h,dc,p: 57% 24% 7% 12% [libx264 @ 0x7feff381a000] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 11% 9% 8% 12% 13% 12% 10% 12% [libx264 @ 0x7feff381a000] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 21% 8% 6% 9% 9% 9% 8% 12% [libx264 @ 0x7feff381a000] i8c dc,h,v,p: 49% 26% 18% 7% [libx264 @ 0x7feff381a000] Weighted P-Frames: Y:8.1% UV:5.0% [libx264 @ 0x7feff381a000] ref P L0: 50.4% 15.1% 14.3% 4.6% 3.3% 2.7% 2.0% 1.3% 1.0% 1.1% 0.8% 0.9% 0.7% 0.8% 0.6% 0.6% [libx264 @ 0x7feff381a000] ref B L0: 80.3% 8.4% 3.5% 1.6% 1.3% 1.1% 0.9% 0.5% 0.5% 0.4% 0.4% 0.4% 0.4% 0.2% 0.1% [libx264 @ 0x7feff381a000] ref B L1: 94.2% 5.8% [libx264 @ 0x7feff381a000] kb/s:1607.94 Laine Lee From llee040 at sbcglobal.net Sat Apr 20 09:17:13 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Sat, 20 Apr 2013 02:17:13 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: Message-ID: On 4/19/13 3:19 AM, "Carl Eugen Hoyos" wrote: >L. Lee sbcglobal.net> writes: > >> https://dl.dropboxusercontent.com/u/42718751/Blockyclip_mpg.zip > >Could you test if the mpdecimate filter works better >for this sample? >$ ffmpeg -i sample -vf mpdecimate -r 24000/1001 -qscale 2 out.avi >The fieldmatch filter does not help because your sample >does not contain any interlaced frames, so there are no >fields to match... >In the future, the decimate filter will hopefully >support dropping three of five frames. Thanks. Well, that's interesting! I get a crash. Seems to be mpdecimate: ffmpeg -i /sample.mpg -vf mpdecimate -r 24000/1001 -qscale 2 /sample.avi ffmpeg version N-37959-g20e99a9 Copyright (c) 2000-2013 the FFmpeg developers built on Apr 16 2013 13:28:23 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --enable-filter=fieldmatch --enable-libmp3lame --enable-gpl --enable-libfaac --enable-pthreads --enable-ffplay --enable-nonfree --enable-libx264 libavutil 52. 26.100 / 52. 26.100 libavcodec 55. 2.100 / 55. 2.100 libavformat 55. 2.100 / 55. 2.100 libavdevice 55. 0.100 / 55. 0.100 libavfilter 3. 56.101 / 3. 56.101 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 0x7fcd92021a00] mmco: unref short failure Last message repeated 1 times Input #0, mpegts, from 'sample.mpg': Duration: 00:02:00.69, start: 1.404767, bitrate: 11454 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (Main) (HDMV / 0x564D4448), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 384 kb/s Please use -q:a or -q:v, -qscale is ambiguous Output #0, avi, to 'sample.avi': Metadata: ISFT : Lavf55.2.100 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 23.98 tbn, 23.98 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, fltp Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> libmp3lame) Press [q] to stop, [?] for help [h264 @ 0x7fcd928d2000] mmco: unref short failure [h264 @ 0x7fcd928d3800] mmco: unref short failure Segmentation fault: 11 > >As an alternative, you could try to interlace your video >to get the telecine pattern back (it appears your video >was de-interlaced after telecine was used) and feed the >result to fieldmatch. I am not saying that is the best >option, but you could test: >-vf interlace,fieldmatch... This strategy gave me good results, and it didn't seem to take significantly longer. Thanks! Laine Lee From palani at i-waves.com Sat Apr 20 10:31:43 2013 From: palani at i-waves.com (Palaniappan N) Date: Sat, 20 Apr 2013 14:01:43 +0530 Subject: [FFmpeg-user] FFMPEG | zlib support is required In-Reply-To: References: Message-ID: Hi, Initially we received the following error swf @ 0x2ae62c0] SWF compressed file detected [swf @ 0x2ae62c0] zlib support is required to read SWF compressed files /var/www/html/WorkFiles/uploads/Sponsor_tutorial_15Apr13.swf: Input/output error Then I did Installed "zlib-1.2.3-27.el6.x86_64" and recompiled FFMPEG using the following Command and the output is as follows ffmpeg]# ./configure --enable-gpl --enable-libfdk_aac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-zlib --enable-nonfree ERROR: libfdk_aac not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. How to proceed further? Please suggest me. Thanks, Palani From cehoyos at ag.or.at Sat Apr 20 11:26:25 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 09:26:25 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?FFMPEG_=7C_zlib_support_is_required?= References: Message-ID: Palaniappan N i-waves.com> writes: > ERROR: libfdk_aac not found > How to proceed further? Please suggest me. Either remove --enable-libfdk_aac from your configure line or install the libfdk_aac library together with its development part. Carl Eugen From cehoyos at ag.or.at Sat Apr 20 11:28:53 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 09:28:53 +0000 (UTC) Subject: [FFmpeg-user] Intent of -copyinkf and non-key frames in video stream copy References: Message-ID: Doug Handy doughandy.com> writes: > ?-copyinkf[:stream_specifier] (output,per-stream)? > When doing stream copy, copy also non-key frames > found at the beginning. > > I am not finding much more information about this > option at all. Perhaps it is wishful thinking on > my part and reading it to mean what I want it to > mean, but I was hoping this would mean that when > specified and your start location was not on a key > frame, that it would generate a proper keyframe No. The option just allows you not to skip the initial non-keyframes when doing stream-copy. Carl Eugen From cehoyos at ag.or.at Sat Apr 20 11:32:59 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 09:32:59 +0000 (UTC) Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) References: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> Message-ID: ukegoldberg.com> writes: > I am trying out FFMPEG on many recommendations. I > am working with recovered deleted videos that are > now Protected MPEG-4 Movie. I don't think they were > protected before they were deleated. They are from my > point and shot camera. The are now jerky and blurry. I am not a native speaker (and several other people on this list aren't either) ;-( I don't understand what you are asking (what is a "protected MPEG-4 movie"? What are recovered deleted videos?), the best thing you can do is provide the command line you are using together with the complete, uncut console output and explain (in simple) words what is wrong with the output file. > I converted them to AVI with FFMPEG. They are > still jerky and blurry. Any suggestions? FFmpeg has several postprocessing options, please see the fine documentation. Carl Eugen From cehoyos at ag.or.at Sat Apr 20 11:35:37 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 09:35:37 +0000 (UTC) Subject: [FFmpeg-user] unable to play mpeg transcoded stream on mplayer or Vlc References: Message-ID: Tom Evans googlemail.com> writes: > On Fri, Apr 19, 2013 at 3:20 PM, Carl Eugen Hoyos wrote: > > Tom Evans googlemail.com> writes: > > > >> MPEG PS is only supposed to have either MPEG 2 Part 1 > >> or Mpeg 2 Part 2 video. > > > > Program streams may also contain h264 (and several > > other video codecs iirc). (including vc1.) > Hmm, if I try to put h264 into a PS, I get a file > that mplayer quickly loses sync on: I just wanted to explain that - contrary to what you wrote - program streams may contain H264 streams. If you just wanted to say that ffmpeg does not mux h264 correctly into program streams, then this was just a misunderstanding, sorry. (I didn't test muxing.) Carl Eugen From cehoyos at ag.or.at Sat Apr 20 12:51:01 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 10:51:01 +0000 (UTC) Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? References: Message-ID: L. Lee sbcglobal.net> writes: > I get a crash. Seems to be mpdecimate: > > ffmpeg -i /sample.mpg -vf mpdecimate -r 24000/1001 > -qscale 2 /sample.avi > Segmentation fault: 11 I don't get this crash with Blockyclip_mpg.mpg. Is another sample needed? Carl Eugen From onemda at gmail.com Sat Apr 20 13:28:18 2013 From: onemda at gmail.com (Paul B Mahol) Date: Sat, 20 Apr 2013 11:28:18 +0000 Subject: [FFmpeg-user] ffmpeg decoder color space / ventuz In-Reply-To: References: Message-ID: On 4/19/13, Dennis Koch wrote: > hello there, > > what colorspace is the ffmpeg decoder using? all our videos in 709 and rgb > are darker than the original in the ventuz output monitor. > and afaik ventuz uses ffmpeg as decoder. More info is needed. Guessing: one of inputs or outputs are yuvj while inputs are not, and in some casese colorspace conversion is not done as it should, see colormatrix filter. > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From onemda at gmail.com Sat Apr 20 13:40:50 2013 From: onemda at gmail.com (Paul B Mahol) Date: Sat, 20 Apr 2013 11:40:50 +0000 Subject: [FFmpeg-user] FFMPEG Fade In Watermark Only In-Reply-To: <5D47262B5E7D4C2D92F1BA14FE4A7857@gmail.com> References: <5D47262B5E7D4C2D92F1BA14FE4A7857@gmail.com> Message-ID: On 4/16/13, Dale Zak wrote: > Dear FFMPEG users, > > I'm trying to watermark a video using FFMPEG and would only like the > watermark to fade in. > > The following code adds the watermark however the fade in effect is applied > to the entire video and not just the watermark. > > ffmpeg -loop 1 -t 3 -i background.png -i silence.mp3 -shortest -s 640x360 > -vf "movie=watermark.png [watermark]; [in] [watermark] > overlay=main_w/2-overlay_w/2:main_h/2-overlay_h/2,fade=in:0:30 [out]" -y > output.mpg I could do it, by applying fade prior to overlay. > > The problem is my video clips have a white background so would like to > maintain the white across all clips. > > I've searched online but have been unable to find a good example on how to > apply the fade in only to the overlay. > > I posted the question on SuperUser > http://superuser.com/questions/567059/ffmpeg-fade-in-watermark-only but > unfortunately have been unable to find a solution. > > Any help regarding this issue would be greatly appreciated. > > Cheers, > Dale > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ted at ukegoldberg.com Sat Apr 20 14:35:06 2013 From: ted at ukegoldberg.com (ted at ukegoldberg.com) Date: Sat, 20 Apr 2013 08:35:06 -0400 Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) In-Reply-To: References: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> Message-ID: <7A406BDC-FD91-4328-B306-D6BE0DD73FA4@ukegoldberg.com> Thanks for responding. My question was about an issue I am having that a number of people on mac forms said FFMPEG would be able to solve. I now think it probably can't. I am now receiving emails on a very technical nature that have nothing to do with my question. I would like to unsubscribe but don't see how. Would someone be so kind as to tell me how to unsubscribe? Thanks, Ted On Apr 20, 2013, at 5:32 AM, Carl Eugen Hoyos wrote: > ukegoldberg.com> writes: > >> I am trying out FFMPEG on many recommendations. I >> am working with recovered deleted videos that are >> now Protected MPEG-4 Movie. I don't think they were >> protected before they were deleated. They are from my >> point and shot camera. The are now jerky and blurry. > > I am not a native speaker (and several other people > on this list aren't either) ;-( > I don't understand what you are asking (what is a > "protected MPEG-4 movie"? What are recovered > deleted videos?), the best thing you can do is > provide the command line you are using together > with the complete, uncut console output and > explain (in simple) words what is wrong with the > output file. > >> I converted them to AVI with FFMPEG. They are >> still jerky and blurry. Any suggestions? > > FFmpeg has several postprocessing options, please > see the fine documentation. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From doug at doughandy.com Sat Apr 20 14:45:43 2013 From: doug at doughandy.com (Doug Handy) Date: Sat, 20 Apr 2013 08:45:43 -0400 Subject: [FFmpeg-user] Intent of -copyinkf and non-key frames in video stream copy In-Reply-To: References: Message-ID: <2AA1F30A-C534-41B0-AB6B-D6E70DC69786@doughandy.com> Carl, >> ?-copyinkf[:stream_specifier] (output,per-stream)? >> When doing stream copy, copy also non-key frames >> found at the beginning. > No. > The option just allows you not to skip > the initial non-keyframes when doing > stream-copy. > Not what I wanted to hear, but a definitive no is still better than continuing to try to chase a dream. So what does that suggest about the playability of the output video stream on those beginning frames? If there is no initial key frame generated, what is the value of using this option to include non-keyframes? I'm trying now to understand the potential uses for the -copyinkf option. Thanks, Doug From lou at lrcd.com Sat Apr 20 15:54:31 2013 From: lou at lrcd.com (Lou) Date: Sat, 20 Apr 2013 05:54:31 -0800 Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) In-Reply-To: <7A406BDC-FD91-4328-B306-D6BE0DD73FA4@ukegoldberg.com> References: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> <7A406BDC-FD91-4328-B306-D6BE0DD73FA4@ukegoldberg.com> Message-ID: <1366466071.1795.140661220389806.695C713A@webmail.messagingengine.com> On Sat, Apr 20, 2013, at 04:35 AM, ted at ukegoldberg.com wrote: > > I am now receiving emails on a very technical nature that have nothing to > do with my question. I would like to unsubscribe but don't see how. Would > someone be so kind as to tell me how to unsubscribe? You can unsubscribe on the same page that you subscribed (scroll to bottom of page): http://ffmpeg.org/mailman/listinfo/ffmpeg-user Or via email: From timothygu99 at gmail.com Sat Apr 20 19:17:02 2013 From: timothygu99 at gmail.com (Timothy Gu) Date: Sat, 20 Apr 2013 10:17:02 -0700 Subject: [FFmpeg-user] User-Question In-Reply-To: References: Message-ID: On Apr 19, 2013 8:28 PM, "JinMeng" wrote: > Now I used "--enable-libmp3lame" option, but configuration is failed. > > "ERROR: libmp3lame >= 3.98.3 not found" > Please help me. > How can I compile ffmpeg with libmp3lame as iOS version? You said you have already compiled the library. The problem should be that configure cannot find libmp3lame. Please post your config.log. Regards, Timothy G. From cehoyos at ag.or.at Sat Apr 20 21:26:55 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 19:26:55 +0000 (UTC) Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) References: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> <7A406BDC-FD91-4328-B306-D6BE0DD73FA4@ukegoldberg.com> Message-ID: ukegoldberg.com> writes: > My question was about an issue I am having that a > number of people on mac forms said FFMPEG would be > able to solve. As said, if you can explain the "issue" in a very simple way, there is some possibility that you get help on this mailing list. FFmpeg certainly allows to reduce jerkiness and blurry on videos, but this needs some trial-and-error. Carl Eugen From cehoyos at ag.or.at Sat Apr 20 21:30:14 2013 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 20 Apr 2013 19:30:14 +0000 (UTC) Subject: [FFmpeg-user] Intent of -copyinkf and non-key frames in video stream copy References: <2AA1F30A-C534-41B0-AB6B-D6E70DC69786@doughandy.com> Message-ID: Doug Handy doughandy.com> writes: > If there is no initial key frame generated, what is > the value of using this option to include non-keyframes? > I'm trying now to understand the potential uses for the > -copyinkf option. If your input video does not start with a keyframe, the option allows you not to start stream copying with the first keyframe (this is the defaul behaviour), but with the first frame of the input stream (no matter if it is a keyframe or not). If you really want to know "why", please be assured that for every option (of the huge number FFmpeg provides) a user has once requested it. (In the specific case of -copyinkf, it is also useful if you want to copy subtitles that are for some reason not marked as keyframes, but this is more a work-around than a feature.) Carl Eugen From ricardo at americasnet.com Sun Apr 21 02:35:13 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sat, 20 Apr 2013 17:35:13 -0700 Subject: [FFmpeg-user] HTML5 live streaming Message-ID: Hi, I've been searching around for solutions to use ffmpeg to do live streaming to html5. I understand that for iOS I need a segmenter? I've installed the latest ffmpeg and also m3u8-segmenter from git, but I'm not really sure how to use it. The m3u8-segmenter site has an example like this: ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 -u http://inodes.org/bigbuck/ But in terms of an HTML5 page source, how would I access the stream that is being generated by the segmenter? Is there a simpler way to use ffmpeg for live streaming? thanks Ricardo From onephatcat at earthlink.net Sun Apr 21 05:45:43 2013 From: onephatcat at earthlink.net (JoelB) Date: Sat, 20 Apr 2013 20:45:43 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: References: Message-ID: <3B09B9AD-D034-4246-9EB9-959BD0D425E2@earthlink.net> Usually a segmented stream URL ends with .m3u8 so for ios, you just give the user a link to that m3u8 playlist URL and if you have a valid stream, the phone will just play it. http://inodes.org/bigbuck/big_buck.m3u8 Would probably be the res Joel Sent from my iPhone On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann wrote: > Hi, > > I've been searching around for solutions to use ffmpeg to do live streaming > to html5. > > I understand that for iOS I need a segmenter? > > I've installed the latest ffmpeg and also m3u8-segmenter from git, but I'm > not really sure how to use it. The m3u8-segmenter site has an example like > this: > > ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 > -u http://inodes.org/bigbuck/ > > But in terms of an HTML5 page source, how would I access the stream that is > being generated by the segmenter? > > Is there a simpler way to use ffmpeg for live streaming? > > thanks > Ricardo > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ricardo at americasnet.com Sun Apr 21 05:56:42 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sat, 20 Apr 2013 20:56:42 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <6253590.7404.1366516016782.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> Message-ID: Hi Joel, Thanks, I'll play around with that. Another question... is the segmenter only for iOS? Would other devices also work with this? Ricardo On Sat, Apr 20, 2013 at 8:45 PM, JoelB wrote: > Usually a segmented stream URL ends with .m3u8 so for ios, you just give > the user a link to that m3u8 playlist URL and if you have a valid stream, > the phone will just play it. > > http://inodes.org/bigbuck/big_buck.m3u8 > > Would probably be the res > > > Joel > Sent from my iPhone > > On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann > wrote: > > > Hi, > > > > I've been searching around for solutions to use ffmpeg to do live > streaming > > to html5. > > > > I understand that for iOS I need a segmenter? > > > > I've installed the latest ffmpeg and also m3u8-segmenter from git, but > I'm > > not really sure how to use it. The m3u8-segmenter site has an example > like > > this: > > > > ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > > m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 > > -u http://inodes.org/bigbuck/ > > > > But in terms of an HTML5 page source, how would I access the stream that > is > > being generated by the segmenter? > > > > Is there a simpler way to use ffmpeg for live streaming? > > > > thanks > > Ricardo > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------------------------------ > Powered by Xeams. Visit xeams.com for more information > ------------------------------------------------------ > From doug at doughandy.com Sun Apr 21 05:57:48 2013 From: doug at doughandy.com (Doug Handy) Date: Sat, 20 Apr 2013 23:57:48 -0400 Subject: [FFmpeg-user] Intent of -copyinkf and non-key frames in video stream copy In-Reply-To: References: <2AA1F30A-C534-41B0-AB6B-D6E70DC69786@doughandy.com> Message-ID: Carl, > > If your input video does not start with a keyframe, > the option allows you not to start stream copying > with the first keyframe (this is the defaul > behaviour), but with the first frame of the > input stream (no matter if it is a keyframe or > not). > In my use case, it isn't the video stream per se that does not start with a keyframe. Rather, it is a trim point that I'd like to be frame accurate even while doing a stream copy instead of transcode. So that even when the point where I want to extract a subset does not start on a key frame, for performance reasons I was hoping -copyinkf would let me start anywhere I think I just into the argument description what I wanted it to mean. :) But that is why I was asking for clarification here first instead of submitting a bug report. Sounds like it could well be working as designed; just not the feature I was hoping it would be. Doug From onephatcat at earthlink.net Sun Apr 21 07:21:13 2013 From: onephatcat at earthlink.net (JoelB) Date: Sat, 20 Apr 2013 22:21:13 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: References: <6253590.7404.1366516016782.JavaMail.root@xeams> Message-ID: <50FBC452-CA6F-41D7-9A3F-62CB665F3487@earthlink.net> Should work for Roku and anything that supports HLS, I think android does, various flash players, QuickTime 7 for Mac, safari(?) supposedly supports it as well. Generally, you want to encode h.264 video with AAC audio. Joel Sent from my iPhone On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann wrote: > Hi Joel, > > Thanks, I'll play around with that. > > Another question... is the segmenter only for iOS? Would other devices also > work with this? > > Ricardo > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB wrote: > >> Usually a segmented stream URL ends with .m3u8 so for ios, you just give >> the user a link to that m3u8 playlist URL and if you have a valid stream, >> the phone will just play it. >> >> http://inodes.org/bigbuck/big_buck.m3u8 >> >> Would probably be the res >> >> >> Joel >> Sent from my iPhone >> >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann >> wrote: >> >>> Hi, >>> >>> I've been searching around for solutions to use ffmpeg to do live >> streaming >>> to html5. >>> >>> I understand that for iOS I need a segmenter? >>> >>> I've installed the latest ffmpeg and also m3u8-segmenter from git, but >> I'm >>> not really sure how to use it. The m3u8-segmenter site has an example >> like >>> this: >>> >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 >>> -u http://inodes.org/bigbuck/ >>> >>> But in terms of an HTML5 page source, how would I access the stream that >> is >>> being generated by the segmenter? >>> >>> Is there a simpler way to use ffmpeg for live streaming? >>> >>> thanks >>> Ricardo >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> >> ------------------------------------------------------ >> Powered by Xeams. Visit xeams.com for more information >> ------------------------------------------------------ > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From renaux.jacky at orange.fr Sun Apr 21 09:56:18 2013 From: renaux.jacky at orange.fr (jacky) Date: Sun, 21 Apr 2013 09:56:18 +0200 Subject: [FFmpeg-user] audiofilter Message-ID: <51739BA2.80009@orange.fr> Hi I have some trouble undestanding volumedetect audiofilter depending on mp3 or aac files I do not get the same values (including n_samples) I have a flv file AVC/AAC if I run ffmpeg -f lavfi -i amovie=toto.flv,volumedetect -f null - 2> toto.txt I am getting if I demux the audio ffmpeg -i %f_in% -vn -acodec libmp3lame -ar 44100 -ab 64k -y toto.mp3 and run again volumedetect ffmpeg -f lavfi -i amovie=toto.mp3,volumedetect -f null - 2> toto.txt n_samples: 194644271 mean_volume: -27.2 dB max_volume: 0.0 dB histogram_0db: 67 histogram_1db: 857 histogram_2db: 967 histogram_3db: 1669 histogram_4db: 9089 histogram_5db: 7714 the aac is 22050hz and 56k are these values responsable for these changes ? details from ffmpeg console input ffmpeg version N-48731-g7e5c5fa Copyright (c) 2000-2013 the FFmpeg developers built on Jan 10 2013 19:47:49 with gcc 4.7.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib --enable-filter=frei0r libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 86.100 / 54. 86.100 libavformat 54. 59.107 / 54. 59.107 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, flv, from 'chandigarh_architecture_qn4_new.flv': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf54.37.100 Duration: 01:13:33.68, start: 0.007000, bitrate: 446 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 720x576 [SAR 128:117 DAR 160:117], 380 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0:1: Audio: aac, 22050 Hz, mono, fltp, 56 kb/s Output #0, mp3, to 'toto.mp3': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt TSSE : Lavf54.59.107 Stream #0:0: Audio: mp3, 44100 Hz, mono, fltp, 64 kb/s Stream mapping: Stream #0:1 -> #0:0 (aac -> libmp3lame) what is wrong thanks for your help jacky From nicolas.george at normalesup.org Sun Apr 21 10:03:44 2013 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 21 Apr 2013 10:03:44 +0200 Subject: [FFmpeg-user] audiofilter In-Reply-To: <51739BA2.80009@orange.fr> References: <51739BA2.80009@orange.fr> Message-ID: <20130421080344.GA1328@phare.normalesup.org> Le duodi 2 flor?al, an CCXXI, jacky a ?crit?: > Hi > I have some trouble undestanding volumedetect audiofilter > depending on mp3 or aac files > I do not get the same values (including n_samples) > I have a flv file AVC/AAC if I run > ffmpeg -f lavfi -i amovie=toto.flv,volumedetect -f null - 2> toto.txt > I am getting You forgot to paste the output. > if I demux the audio > ffmpeg -i %f_in% -vn -acodec libmp3lame -ar 44100 -ab 64k -y toto.mp3 You are not just demuxing, you are transcoding. > and run again volumedetect > > ffmpeg -f lavfi -i amovie=toto.mp3,volumedetect -f null - 2> toto.txt > > n_samples: 194644271 > mean_volume: -27.2 dB > max_volume: 0.0 dB > histogram_0db: 67 > histogram_1db: 857 > histogram_2db: 967 > histogram_3db: 1669 > histogram_4db: 9089 > histogram_5db: 7714 > > the aac is 22050hz and 56k are these values responsable for these changes ? I can not answer that, not being able to know how large are the changes. > details from ffmpeg console input > ffmpeg version N-48731-g7e5c5fa Copyright (c) 2000-2013 the FFmpeg > developers > built on Jan 10 2013 19:47:49 with gcc 4.7.2 (GCC) This is a bit old, there was a lot of changes in filters since then. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From gabipetrovay at gmail.com Sun Apr 21 15:38:57 2013 From: gabipetrovay at gmail.com (Gabriel Petrovay) Date: Sun, 21 Apr 2013 15:38:57 +0200 Subject: [FFmpeg-user] How to make an mp4 file streamable / Why does ffmpeg "breaks streamability" of mp4 files? Message-ID: Hi, I am using the "copy" coder and the -f mp4 format to try and add "streamabilty" to an audio file. With streamability I mean that an HTML-5 enabled browser can start playing the file before it is completely downloaded. I asked this question on StackOverflow where you can see more details with what mp4 format arguments I tried and how ffmped deletes the "streamability" from a file that is already streamable: http://stackoverflow.com/questions/16131616/how-to-create-streamable-audio-file-with-ffmpeg Thanks! -- MSc Gabriel Petrovay Mobile: +41(0)787978034 From gabipetrovay at gmail.com Sun Apr 21 15:53:51 2013 From: gabipetrovay at gmail.com (Gabriel Petrovay) Date: Sun, 21 Apr 2013 15:53:51 +0200 Subject: [FFmpeg-user] How to make an mp4 file streamable / Why does ffmpeg "breaks streamability" of mp4 files? In-Reply-To: References: Message-ID: I found the solution: use ONLY the -movflags faststart format argument for the mp4 format. On Sun, Apr 21, 2013 at 3:38 PM, Gabriel Petrovay wrote: > Hi, > > I am using the "copy" coder and the -f mp4 format to try and add > "streamabilty" to an audio file. With streamability I mean that an HTML-5 > enabled browser can start playing the file before it is completely > downloaded. > > I asked this question on StackOverflow where you can see more details with > what mp4 format arguments I tried and how ffmped deletes the > "streamability" from a file that is already streamable: > > http://stackoverflow.com/questions/16131616/how-to-create-streamable-audio-file-with-ffmpeg > > Thanks! > > -- > MSc Gabriel Petrovay > Mobile: +41(0)787978034 > -- MSc Gabriel Petrovay Mobile: +41(0)787978034 From ricardo at americasnet.com Sun Apr 21 23:53:36 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sun, 21 Apr 2013 14:53:36 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <30369007.7466.1366521739275.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> Message-ID: On Sat, Apr 20, 2013 at 10:21 PM, JoelB wrote: > Should work for Roku and anything that supports HLS, I think android does, > various flash players, QuickTime 7 for Mac, safari(?) supposedly supports > it as well. Generally, you want to encode h.264 video with AAC audio. > > Joel > > Sent from my iPhone > So that's what I'm doing... I'm encoding to 264 and AAC, and piping to the segmenter. I see the m3u8 file being written as well as all the other ts files. But if I try to access from an iPad it gives me different errors, sometimes it says "cannot decode", sometimes it says cannot find the file. I'm assuming that the m3u8 file is supposed to be served up by a standard web server (apache). Do I need any special configuration in apache for this? Here's my ffmpeg command: ffmpeg -threads 4 -i "rtmp://server1/live/livestream1" -f mpegts -i_qfactor 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 -s 480x270 -b:v 512k -b:a 56k -ar 22050 - | m3u8-segmenter -i - -d 10 -p cidadeviva/live -m /shared/media/videos/cidadeviva/live.m3u8 -u http://web1.americasnet.com/videos/ It succeeds in reading the rtpm stream and does pipe the output to the segmenter since I see all of the m3u8 and ts files being written. I'm able to access the m3u8 file via http, although I see apache often complains that it can't find the file. Could that be because the file is constantly being written? Ricardo > On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann > wrote: > > > Hi Joel, > > > > Thanks, I'll play around with that. > > > > Another question... is the segmenter only for iOS? Would other devices > also > > work with this? > > > > Ricardo > > > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB wrote: > > > >> Usually a segmented stream URL ends with .m3u8 so for ios, you just give > >> the user a link to that m3u8 playlist URL and if you have a valid > stream, > >> the phone will just play it. > >> > >> http://inodes.org/bigbuck/big_buck.m3u8 > >> > >> Would probably be the res > >> > >> > >> Joel > >> Sent from my iPhone > >> > >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann > >> wrote: > >> > >>> Hi, > >>> > >>> I've been searching around for solutions to use ffmpeg to do live > >> streaming > >>> to html5. > >>> > >>> I understand that for iOS I need a segmenter? > >>> > >>> I've installed the latest ffmpeg and also m3u8-segmenter from git, but > >> I'm > >>> not really sure how to use it. The m3u8-segmenter site has an example > >> like > >>> this: > >>> > >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 > >>> -u http://inodes.org/bigbuck/ > >>> > >>> But in terms of an HTML5 page source, how would I access the stream > that > >> is > >>> being generated by the segmenter? > >>> > >>> Is there a simpler way to use ffmpeg for live streaming? > >>> > >>> thanks > >>> Ricardo > >>> _______________________________________________ > >>> ffmpeg-user mailing list > >>> ffmpeg-user at ffmpeg.org > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > >> > >> ------------------------------------------------------ > >> Powered by Xeams. Visit xeams.com for more information > >> ------------------------------------------------------ > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------------------------------ > Powered by Xeams. Visit xeams.com for more information > ------------------------------------------------------ > From ted at ukegoldberg.com Sat Apr 20 15:58:00 2013 From: ted at ukegoldberg.com (ted at ukegoldberg.com) Date: Sat, 20 Apr 2013 09:58:00 -0400 Subject: [FFmpeg-user] MPEG TO AVI (reposted after joining mailing list) In-Reply-To: <1366466071.1795.140661220389806.695C713A@webmail.messagingengine.com> References: <3E0B5B9B-4073-4931-9BF9-F55F2D82A0A3@ukegoldberg.com> <7A406BDC-FD91-4328-B306-D6BE0DD73FA4@ukegoldberg.com> <1366466071.1795.140661220389806.695C713A@webmail.messagingengine.com> Message-ID: Thank you. On Apr 20, 2013, at 9:54 AM, Lou wrote: > On Sat, Apr 20, 2013, at 04:35 AM, ted at ukegoldberg.com wrote: >> >> I am now receiving emails on a very technical nature that have nothing to >> do with my question. I would like to unsubscribe but don't see how. Would >> someone be so kind as to tell me how to unsubscribe? > > You can unsubscribe on the same page that you subscribed (scroll to > bottom of page): > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > Or via email: > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From armencho at gmail.com Sun Apr 21 17:17:22 2013 From: armencho at gmail.com (Armen Michaeli) Date: Sun, 21 Apr 2013 17:17:22 +0200 Subject: [FFmpeg-user] Encoding an XSUB (DivX subtitles) stream (from an SRT stream) Message-ID: <67BD5367-C004-4023-9799-6F5C756A4B94@gmail.com> Hi all, I am trying to include an XSUB stream in to an AVI container. The source of this stream comes from an SRT file, so I am using the following command line: ffmpeg -i movie.avi -i movie.srt -scodec xsub output.avi The subtitle encoding fails with "[xsub @ 0x7fb3f883ea00] No subtitle bitmap available.", as part of the following output: ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers built on Mar 15 2013 18:24:17 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --arch=x86_64 --enable-runtime-cpudetect libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [srt @ 0x7fb3f8818e00] Estimating duration from bitrate, this may be inaccurate Input #0, srt, from 'movie.srt': Duration: N/A, bitrate: N/A Stream #0:0: Subtitle: subrip Output #0, avi, to 'output.avi': Metadata: ISFT : Lavf54.63.104 Stream #0:0: Subtitle: xsub (DXSB / 0x42535844) Stream mapping: Stream #0:0 -> #0:0 (subrip -> xsub) Press [q] to stop, [?] for help [xsub @ 0x7fb3f883ea00] No subtitle bitmap available. Subtitle encoding failed Am I doing something wrong, or is ffmpeg incapabale of doing what I am trying to do? I am on a Macbook Air, Mac OS X 10.8.3, binary downloaded from http://ffmpegmac.net. I don't have much choice of binaries anyway, these are far and scarce for the platform. From llee040 at sbcglobal.net Mon Apr 22 01:15:09 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Sun, 21 Apr 2013 18:15:09 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: Message-ID: On 4/20/13 5:51 AM, "Carl Eugen Hoyos" wrote: >L. Lee sbcglobal.net> writes: > >> I get a crash. Seems to be mpdecimate: >> >> ffmpeg -i /sample.mpg -vf mpdecimate -r 24000/1001 >> -qscale 2 /sample.avi > >> Segmentation fault: 11 Thanks, Carl Eugen. I tried it on a couple of excerpts. This brings to mind the message you left on the MEncoder list about not needing to specifically enable the fieldmatch filter in build configuration, which apparently was necessary for me. My ffmpeg is the ancillary configuration provided by the MPlayer build process. Should I be building my ffmpeg a better way? Also, when I applied the "interlace,fieldmatch?" filtergraph to the conversion of a longer video 720p passage, sync and frames were fine, but I still got intermittent blocks (pixelization), so I'm still looking for a solution to that. Thanks again. Laine Lee From ubitux at gmail.com Mon Apr 22 01:19:13 2013 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Mon, 22 Apr 2013 01:19:13 +0200 Subject: [FFmpeg-user] Encoding an XSUB (DivX subtitles) stream (from an SRT stream) In-Reply-To: <67BD5367-C004-4023-9799-6F5C756A4B94@gmail.com> References: <67BD5367-C004-4023-9799-6F5C756A4B94@gmail.com> Message-ID: <20130421231913.GB10923@leki> On Sun, Apr 21, 2013 at 05:17:22PM +0200, Armen Michaeli wrote: > Hi all, > > I am trying to include an XSUB stream in to an AVI container. The source of this stream comes from an SRT file, so I am using the following command line: > > ffmpeg -i movie.avi -i movie.srt -scodec xsub output.avi > > The subtitle encoding fails with "[xsub @ 0x7fb3f883ea00] No subtitle bitmap available.", as part of the following output: > > ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers > built on Mar 15 2013 18:24:17 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) > configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --arch=x86_64 --enable-runtime-cpudetect > libavutil 52. 18.100 / 52. 18.100 > libavcodec 54. 92.100 / 54. 92.100 > libavformat 54. 63.104 / 54. 63.104 > libavdevice 54. 3.103 / 54. 3.103 > libavfilter 3. 42.103 / 3. 42.103 > libswscale 2. 2.100 / 2. 2.100 > libswresample 0. 17.102 / 0. 17.102 > libpostproc 52. 2.100 / 52. 2.100 > [srt @ 0x7fb3f8818e00] Estimating duration from bitrate, this may be inaccurate > Input #0, srt, from 'movie.srt': > Duration: N/A, bitrate: N/A > Stream #0:0: Subtitle: subrip > Output #0, avi, to 'output.avi': > Metadata: > ISFT : Lavf54.63.104 > Stream #0:0: Subtitle: xsub (DXSB / 0x42535844) > Stream mapping: > Stream #0:0 -> #0:0 (subrip -> xsub) > Press [q] to stop, [?] for help > [xsub @ 0x7fb3f883ea00] No subtitle bitmap available. > Subtitle encoding failed > > Am I doing something wrong, or is ffmpeg incapabale of doing what I am trying to do? > srt/subrip subtitles are text based. xsub are bitmap based. FFmpeg doesn't do rasterization internally. You have the possibility to use the subtitles filter to hardburn your subtitles. See https://ffmpeg.org/ffmpeg-filters.html#subtitles -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 490 bytes Desc: not available URL: From llee040 at sbcglobal.net Mon Apr 22 01:31:39 2013 From: llee040 at sbcglobal.net (L. Lee) Date: Sun, 21 Apr 2013 18:31:39 -0500 Subject: [FFmpeg-user] How can I use the fieldmatch filter to detelecine? In-Reply-To: <516F1ABE.80407@ukfsn.org> Message-ID: On 4/17/13 4:57 PM, "Andy Furniss" wrote: >L. Lee wrote: >> I've developed a highly refined method for applying detelecine using >> MEncoder's pullup filter methods. I get excellent results using that >> method for the examples I'm including in this message. However, for >> reasons that are probably obvious, I'm now trying to get comparable >> results using ffmpeg. >> >> How can I use the fieldmatch filter to detelecine these excerpts to get >> better results using ffmpeg? This source (for both source 1 and source >>2) >> is 720P. I don't see any difference in my results whether or not I use >> decimate in the filtergraph as shown in the 8.22.2 Examples >> (http://ffmpeg.org/ffmpeg-filters.html#toc-Examples-3). I've also >>included >> a slightly longer mpg-muxed version (source 2) which concludes with the >> source 1 excerpt. I get different results when encoding the longer >> mpg-muxed excerpt (detelecine seems correct, however some areas are >> pixilated, but only when encoding the longer excerpt) Although the >> included filtergraph for my source 2 command uses the shorter of the two >> examples provided in the 8.22.2 section of the documentation, I've tried >> both and can't observe any difference in the output. >> >> Here's my source 1: >> >> https://dl.dropboxusercontent.com/u/42718751/Blockyclip.zip >> >> Here's my output 1: >> >> https://dl.dropboxusercontent.com/u/42718751/Blockyclip_output.zip > >This looks wrong - 12fps with frames duped. > >_______________________________________________ >ffmpeg-user mailing list >ffmpeg-user at ffmpeg.org >http://ffmpeg.org/mailman/listinfo/ffmpeg-user Thanks, also, for your reply, Andy. Yes, I agree, it was wrong. I'm making progress now. The 1080i conversion is working OK with regard to fps and correct detelecine, and by using "interlace,fieldmatch" in the filtergraph, that's also true for the 720p. But I'm still having trouble with blockiness (pixelization) as observable in the sample provided, but now I'm trying to determine whether my method of building ffmpeg needs correcting (see my last post). Thanks again. Laine Lee From vkroutik at gmail.com Mon Apr 22 02:00:19 2013 From: vkroutik at gmail.com (Vlad Kroutik) Date: Sun, 21 Apr 2013 17:00:19 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> Message-ID: You do not need m3u8-segmenter as ffmpeg support segmented output and its built it. Look at the ffmpeg docs or example: /usr/local/bin/ffmpeg -i hi.ts -loglevel debug -analyzeduration 6M -map 0 -c copy -f segment -segment_time 10 -segment_list hi.m3u8 hi_%05d.ts On Sun, Apr 21, 2013 at 2:53 PM, Ricardo Kleemann wrote: > On Sat, Apr 20, 2013 at 10:21 PM, JoelB wrote: > > > Should work for Roku and anything that supports HLS, I think android > does, > > various flash players, QuickTime 7 for Mac, safari(?) supposedly supports > > it as well. Generally, you want to encode h.264 video with AAC audio. > > > > Joel > > > > Sent from my iPhone > > > > So that's what I'm doing... I'm encoding to 264 and AAC, and piping to the > segmenter. > > I see the m3u8 file being written as well as all the other ts files. > > But if I try to access from an iPad it gives me different errors, sometimes > it says "cannot decode", sometimes it says cannot find the file. > > I'm assuming that the m3u8 file is supposed to be served up by a standard > web server (apache). Do I need any special configuration in apache for > this? > > Here's my ffmpeg command: > > ffmpeg -threads 4 -i "rtmp://server1/live/livestream1" -f mpegts -i_qfactor > 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 -s > 480x270 -b:v 512k -b:a 56k -ar 22050 - | m3u8-segmenter -i - -d 10 -p > cidadeviva/live -m /shared/media/videos/cidadeviva/live.m3u8 -u > http://web1.americasnet.com/videos/ > > It succeeds in reading the rtpm stream and does pipe the output to the > segmenter since I see all of the m3u8 and ts files being written. > > I'm able to access the m3u8 file via http, although I see apache often > complains that it can't find the file. Could that be because the file is > constantly being written? > > Ricardo > > > > On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann > > wrote: > > > > > Hi Joel, > > > > > > Thanks, I'll play around with that. > > > > > > Another question... is the segmenter only for iOS? Would other devices > > also > > > work with this? > > > > > > Ricardo > > > > > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB > wrote: > > > > > >> Usually a segmented stream URL ends with .m3u8 so for ios, you just > give > > >> the user a link to that m3u8 playlist URL and if you have a valid > > stream, > > >> the phone will just play it. > > >> > > >> http://inodes.org/bigbuck/big_buck.m3u8 > > >> > > >> Would probably be the res > > >> > > >> > > >> Joel > > >> Sent from my iPhone > > >> > > >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann < > ricardo at americasnet.com> > > >> wrote: > > >> > > >>> Hi, > > >>> > > >>> I've been searching around for solutions to use ffmpeg to do live > > >> streaming > > >>> to html5. > > >>> > > >>> I understand that for iOS I need a segmenter? > > >>> > > >>> I've installed the latest ffmpeg and also m3u8-segmenter from git, > but > > >> I'm > > >>> not really sure how to use it. The m3u8-segmenter site has an example > > >> like > > >>> this: > > >>> > > >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > > >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 > > >>> -u http://inodes.org/bigbuck/ > > >>> > > >>> But in terms of an HTML5 page source, how would I access the stream > > that > > >> is > > >>> being generated by the segmenter? > > >>> > > >>> Is there a simpler way to use ffmpeg for live streaming? > > >>> > > >>> thanks > > >>> Ricardo > > >>> _______________________________________________ > > >>> ffmpeg-user mailing list > > >>> ffmpeg-user at ffmpeg.org > > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > >> _______________________________________________ > > >> ffmpeg-user mailing list > > >> ffmpeg-user at ffmpeg.org > > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > >> > > >> > > >> ------------------------------------------------------ > > >> Powered by Xeams. Visit xeams.com for more information > > >> ------------------------------------------------------ > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > ------------------------------------------------------ > > Powered by Xeams. Visit xeams.com for more information > > ------------------------------------------------------ > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From blacktrash at gmx.net Mon Apr 22 02:04:28 2013 From: blacktrash at gmx.net (Christian Ebert) Date: Mon, 22 Apr 2013 01:04:28 +0100 Subject: [FFmpeg-user] What is the (technical) difference between -r and the fps filter? In-Reply-To: <20130420040112.GB31816@krille.blacktrash.org> References: <20130420040112.GB31816@krille.blacktrash.org> Message-ID: <20130422000428.GB751@krille.blacktrash.org> * Christian Ebert on Saturday, April 20, 2013 at 05:01:12 +0100 > * Werner Robitza on Friday, April 19, 2013 at 17:06:38 +0200 >>> That is correct, please use -vf fps >>> See also ticket #1578 >> >> Thanks Carl Eugen. >> >> However the question still remains: what's the actual difference between >> the filter and -r? >> Is it generally not recommended to use the latter at all? Seems all over >> the place in many examples? > > Or is the message that -r or -r:v is deprecated? Or in the > process of being deprecated? ping? What about -r:v -force_fps ? Deprecated? Not the same? Not the same in what way? -- \black\trash movie _SAME TIME SAME PLACE_ --->> http://www.blacktrash.org/underdogma/stsp.php \black\trash audio _ANOTHER TIME ANOTHER PLACE_ --->> http://www.blacktrash.org/underdogma/atap.html From ricardo at americasnet.com Mon Apr 22 02:22:45 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sun, 21 Apr 2013 17:22:45 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <21636383.8570.1366581243220.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> <21636383.8570.1366581243220.JavaMail.root@xeams> Message-ID: Hi Guys, Some additional info if someone can help... I'm actually able to play my m3u8 stream directly using VLC... but in HTML5 video tag it is not working. The iPad says that it cannot decode. I must have something in the ts files that doesn't work for iPad, while it still works for VLC Any ideas on how to troubleshoot? thanks Ricardo On Sun, Apr 21, 2013 at 2:53 PM, Ricardo Kleemann wrote: > On Sat, Apr 20, 2013 at 10:21 PM, JoelB wrote: > > > Should work for Roku and anything that supports HLS, I think android > does, > > various flash players, QuickTime 7 for Mac, safari(?) supposedly supports > > it as well. Generally, you want to encode h.264 video with AAC audio. > > > > Joel > > > > Sent from my iPhone > > > > So that's what I'm doing... I'm encoding to 264 and AAC, and piping to the > segmenter. > > I see the m3u8 file being written as well as all the other ts files. > > But if I try to access from an iPad it gives me different errors, sometimes > it says "cannot decode", sometimes it says cannot find the file. > > I'm assuming that the m3u8 file is supposed to be served up by a standard > web server (apache). Do I need any special configuration in apache for > this? > > Here's my ffmpeg command: > > ffmpeg -threads 4 -i "rtmp://server1/live/livestream1" -f mpegts -i_qfactor > 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 -s > 480x270 -b:v 512k -b:a 56k -ar 22050 - | m3u8-segmenter -i - -d 10 -p > cidadeviva/live -m /shared/media/videos/cidadeviva/live.m3u8 -u > http://web1.americasnet.com/videos/ > > It succeeds in reading the rtpm stream and does pipe the output to the > segmenter since I see all of the m3u8 and ts files being written. > > I'm able to access the m3u8 file via http, although I see apache often > complains that it can't find the file. Could that be because the file is > constantly being written? > > Ricardo > > > > On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann > > wrote: > > > > > Hi Joel, > > > > > > Thanks, I'll play around with that. > > > > > > Another question... is the segmenter only for iOS? Would other devices > > also > > > work with this? > > > > > > Ricardo > > > > > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB > wrote: > > > > > >> Usually a segmented stream URL ends with .m3u8 so for ios, you just > give > > >> the user a link to that m3u8 playlist URL and if you have a valid > > stream, > > >> the phone will just play it. > > >> > > >> http://inodes.org/bigbuck/big_buck.m3u8 > > >> > > >> Would probably be the res > > >> > > >> > > >> Joel > > >> Sent from my iPhone > > >> > > >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann < > ricardo at americasnet.com> > > >> wrote: > > >> > > >>> Hi, > > >>> > > >>> I've been searching around for solutions to use ffmpeg to do live > > >> streaming > > >>> to html5. > > >>> > > >>> I understand that for iOS I need a segmenter? > > >>> > > >>> I've installed the latest ffmpeg and also m3u8-segmenter from git, > but > > >> I'm > > >>> not really sure how to use it. The m3u8-segmenter site has an example > > >> like > > >>> this: > > >>> > > >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > > >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m tmp/big_buck.m3u8 > > >>> -u http://inodes.org/bigbuck/ > > >>> > > >>> But in terms of an HTML5 page source, how would I access the stream > > that > > >> is > > >>> being generated by the segmenter? > > >>> > > >>> Is there a simpler way to use ffmpeg for live streaming? > > >>> > > >>> thanks > > >>> Ricardo > > >>> _______________________________________________ > > >>> ffmpeg-user mailing list > > >>> ffmpeg-user at ffmpeg.org > > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > >> _______________________________________________ > > >> ffmpeg-user mailing list > > >> ffmpeg-user at ffmpeg.org > > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > >> > > >> > > >> ------------------------------------------------------ > > >> Powered by Xeams. Visit xeams.com for more information > > >> ------------------------------------------------------ > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > ------------------------------------------------------ > > Powered by Xeams. Visit xeams.com for more information > > ------------------------------------------------------ > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------------------------------ > Powered by Xeams. Visit xeams.com for more information > ------------------------------------------------------ > From ricardo at americasnet.com Mon Apr 22 02:43:02 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sun, 21 Apr 2013 17:43:02 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <5518136.8718.1366588839795.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> <5518136.8718.1366588839795.JavaMail.root@xeams> Message-ID: That's great, I didn't realize it. I'm trying to convert the input to h.264 + AAC and output to the segmenter but it's not working. Can you help me find the error in the command? ffmpeg -loglevel debug -threads 4 -i "rtmp://server1/live/livestream1" -i_qfactor 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 -s 480x270 -b:v 512k -b:a 56k -ar 22050 -map 0 -f segment -segment_time 10 -segment_list live.m3u8 -segment_list_flags +live live_%05d.ts The debug output is too large, but here is the end section: Successfully opened the file. Parsing a group of options: output file live_%05d.ts. Applying option vcodec (force video codec ('copy' to copy stream)) with argument libx264. Applying option s (set frame size (WxH or abbreviation)) with argument 480x270. Applying option b:v (video bitrate (please use -b:v)) with argument 512k. Applying option b:a (video bitrate (please use -b:v)) with argument 56k. Applying option ar (set audio sampling rate (in Hz)) with argument 22050. Applying option map (set input stream mapping) with argument 0. Applying option f (force format) with argument segment. Successfully parsed a group of options. Opening an output file: live_%05d.ts. Successfully opened the file. [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'video_size' to value '490x270' [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'pix_fmt' to value '0' [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'time_base' to value '1/1000' [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'pixel_aspect' to value '1/1' [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'sws_param' to value 'flags=2' [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'frame_rate' to value '30000/1001' [graph 0 input from stream 0:0 @ 0x153eb20] w:490 h:270 pixfmt:yuv420p tb:1/1000 fr:30000/1001 sar:1/1 sws_param:flags=2 [scaler for output stream 0:0 @ 0x153fc40] Setting 'w' to value '480' [scaler for output stream 0:0 @ 0x153fc40] Setting 'h' to value '270' [scaler for output stream 0:0 @ 0x153fc40] Setting 'flags' to value '0x4' [scaler for output stream 0:0 @ 0x153fc40] w:480 h:270 flags:'0x4' interl:0 [format @ 0x160a2a0] compat: called with args=[yuv420p|yuvj420p|yuv422p|yuv444p] [format @ 0x160a2a0] Setting 'pix_fmts' to value 'yuv420p|yuvj420p|yuv422p|yuv444p' [AVFilterGraph @ 0x1535020] query_formats: 5 queried, 4 merged, 0 already done, 0 delayed [scaler for output stream 0:0 @ 0x153fc40] w:490 h:270 fmt:yuv420p sar:1/1 -> w:480 h:270 fmt:yuv420p sar:49/48 flags:0x4 [graph 1 input from stream 0:1 @ 0x1533300] Setting 'time_base' to value '1/44100' [graph 1 input from stream 0:1 @ 0x1533300] Setting 'sample_rate' to value '44100' [graph 1 input from stream 0:1 @ 0x1533300] Setting 'sample_fmt' to value 's16p' [graph 1 input from stream 0:1 @ 0x1533300] Setting 'channel_layout' to value '0x4' [graph 1 input from stream 0:1 @ 0x1533300] tb:1/44100 samplefmt:s16p samplerate:44100 chlayout:0x4 [audio format for output stream 0:1 @ 0x1536b40] Setting 'sample_fmts' to value 's16' [audio format for output stream 0:1 @ 0x1536b40] Setting 'sample_rates' to value '22050' [audio format for output stream 0:1 @ 0x1536b40] Setting 'channel_layouts' to value '0x4|0x3' [audio format for output stream 0:1 @ 0x1536b40] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1' [AVFilterGraph @ 0x153d200] query_formats: 4 queried, 9 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x153b540] ch:1 chl:mono fmt:s16p r:44100Hz -> ch:1 chl:mono fmt:s16 r:22050Hz [libx264 @ 0x1635460] using mv_range_thread = 24 [libx264 @ 0x1635460] using SAR=49/48 [libx264 @ 0x1635460] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.1 Cache64 [libx264 @ 0x1635460] profile High, level 2.1 [libx264 @ 0x1635460] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=512 ratetol=1.0 qcomp=0.60 qpmin=10 qpmax=63 qpstep=4 ip_ratio=1.41 aq=1:1.00 [segment @ 0x15d4080] Selected stream id:0 type:video [mpegts @ 0x23f2cc0] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, segment, to 'live_%05d.ts': Metadata: author : copyright : description : keywords : rating : title : presetname : Custom creationdate : Sun Apr 21 16:31:28 2013 : videodevice : AV/C Tape Recorder/Player avclevel : 30 avcprofile : 66 videokeyframe_frequency: 3 audiodevice : AV/C Tape Recorder/Player audiochannels : 1 audioinputvolume: 10 encoder : Lavf55.3.100 Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 480x270 [SAR 49:48 DAR 49:27], 1001/30000, q=10-63, 512 kb/s, 90k tbn, 29.97 tbc Stream #0:1, 0, 1/90000: Audio: mp2, 22050 Hz, mono, s16, 56 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (mp3 -> mp2) Press [q] to stop, [?] for help *** 1 dup! *** drop! Last message repeated 106 times [libx264 @ 0x1635460] frame= 0 QP=28.46 NAL=3 Slice:I Poc:0 I:510 P:0 SKIP:0 size=4440 bytes [segment @ 0x15d4080] segment:'live_00000.ts' starts with packet stream:0 pts:0 pts_time:0 frame:0 [mpegts @ 0x23f2cc0] H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) [AVIOContext @ 0x23ea6a0] Statistics: 0 seeks, 0 writeouts av_interleaved_write_frame(): Invalid argument Statistics: 287291 bytes read, 0 seeks [rtmp @ 0x153d800] Deleting stream... On Sun, Apr 21, 2013 at 5:00 PM, Vlad Kroutik wrote: > You do not need m3u8-segmenter as ffmpeg support segmented output and its > built it. Look at the ffmpeg docs or example: > /usr/local/bin/ffmpeg -i hi.ts -loglevel debug > -analyzeduration 6M -map 0 -c copy -f segment -segment_time 10 > -segment_list hi.m3u8 hi_%05d.ts > > > > On Sun, Apr 21, 2013 at 2:53 PM, Ricardo Kleemann > wrote: > > > On Sat, Apr 20, 2013 at 10:21 PM, JoelB > wrote: > > > > > Should work for Roku and anything that supports HLS, I think android > > does, > > > various flash players, QuickTime 7 for Mac, safari(?) supposedly > supports > > > it as well. Generally, you want to encode h.264 video with AAC audio. > > > > > > Joel > > > > > > Sent from my iPhone > > > > > > > So that's what I'm doing... I'm encoding to 264 and AAC, and piping to > the > > segmenter. > > > > I see the m3u8 file being written as well as all the other ts files. > > > > But if I try to access from an iPad it gives me different errors, > sometimes > > it says "cannot decode", sometimes it says cannot find the file. > > > > I'm assuming that the m3u8 file is supposed to be served up by a standard > > web server (apache). Do I need any special configuration in apache for > > this? > > > > Here's my ffmpeg command: > > > > ffmpeg -threads 4 -i "rtmp://server1/live/livestream1" -f mpegts > -i_qfactor > > 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 -s > > 480x270 -b:v 512k -b:a 56k -ar 22050 - | m3u8-segmenter -i - -d 10 -p > > cidadeviva/live -m /shared/media/videos/cidadeviva/live.m3u8 -u > > http://web1.americasnet.com/videos/ > > > > It succeeds in reading the rtpm stream and does pipe the output to the > > segmenter since I see all of the m3u8 and ts files being written. > > > > I'm able to access the m3u8 file via http, although I see apache often > > complains that it can't find the file. Could that be because the file is > > constantly being written? > > > > Ricardo > > > > > > > On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann > > > > wrote: > > > > > > > Hi Joel, > > > > > > > > Thanks, I'll play around with that. > > > > > > > > Another question... is the segmenter only for iOS? Would other > devices > > > also > > > > work with this? > > > > > > > > Ricardo > > > > > > > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB > > wrote: > > > > > > > >> Usually a segmented stream URL ends with .m3u8 so for ios, you just > > give > > > >> the user a link to that m3u8 playlist URL and if you have a valid > > > stream, > > > >> the phone will just play it. > > > >> > > > >> http://inodes.org/bigbuck/big_buck.m3u8 > > > >> > > > >> Would probably be the res > > > >> > > > >> > > > >> Joel > > > >> Sent from my iPhone > > > >> > > > >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann < > > ricardo at americasnet.com> > > > >> wrote: > > > >> > > > >>> Hi, > > > >>> > > > >>> I've been searching around for solutions to use ffmpeg to do live > > > >> streaming > > > >>> to html5. > > > >>> > > > >>> I understand that for iOS I need a segmenter? > > > >>> > > > >>> I've installed the latest ffmpeg and also m3u8-segmenter from git, > > but > > > >> I'm > > > >>> not really sure how to use it. The m3u8-segmenter site has an > example > > > >> like > > > >>> this: > > > >>> > > > >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > > > >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m > tmp/big_buck.m3u8 > > > >>> -u http://inodes.org/bigbuck/ > > > >>> > > > >>> But in terms of an HTML5 page source, how would I access the stream > > > that > > > >> is > > > >>> being generated by the segmenter? > > > >>> > > > >>> Is there a simpler way to use ffmpeg for live streaming? > > > >>> > > > >>> thanks > > > >>> Ricardo > > > >>> _______________________________________________ > > > >>> ffmpeg-user mailing list > > > >>> ffmpeg-user at ffmpeg.org > > > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > >> _______________________________________________ > > > >> ffmpeg-user mailing list > > > >> ffmpeg-user at ffmpeg.org > > > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > >> > > > >> > > > >> ------------------------------------------------------ > > > >> Powered by Xeams. Visit xeams.com for more information > > > >> ------------------------------------------------------ > > > > _______________________________________________ > > > > ffmpeg-user mailing list > > > > ffmpeg-user at ffmpeg.org > > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > > > > ------------------------------------------------------ > > > Powered by Xeams. Visit xeams.com for more information > > > ------------------------------------------------------ > > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------------------------------ > Powered by Xeams. Visit xeams.com for more information > ------------------------------------------------------ > From pulche at gmail.com Mon Apr 22 02:48:29 2013 From: pulche at gmail.com (Jeeyoung Byun) Date: Mon, 22 Apr 2013 09:48:29 +0900 Subject: [FFmpeg-user] Questions to developers.. In-Reply-To: References: Message-ID: Thank you for your response. I understand that this is not the right place to ask about licensing issues. But, other than the legal issues, *do you know whether any source codes from Dolby AC-3 is incorporated in the ffmpeg AC-3? *If it's not the right place even for only this question, then could you please let me know where I can ask this quesiton? Thank you so much for your consideration in advance. Jeeyoung On Thu, Apr 18, 2013 at 7:12 PM, Carl Eugen Hoyos wrote: > Jeeyoung Byun gmail.com> writes: > > > Actually, I have some legal issues using > > Dolby AC-3 codec from FFmpeg. > > Since no FFmpeg developer is a lawyer (afaik), > this is not the right place to ask. > But if you have any indication that something > about the AC-3 codec source code is different > from the rest of FFmpeg, please feel free to > share your information. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ddhung at vasc.com.vn Mon Apr 22 03:32:24 2013 From: ddhung at vasc.com.vn (=?utf-8?B?xJDhurduZyDEkOG7qWMgSMawbmc=?=) Date: Mon, 22 Apr 2013 08:32:24 +0700 Subject: [FFmpeg-user] Improvement the quality of H264 video signal with low bit rate Message-ID: <001601ce3ef9$414b0f20$c3e12d60$@vasc.com.vn> Dear All. I transcoded the IP input video which is UDP and multicast stream to other output with low bit rate. I found that the quality of video output is very bad. The image is foggy and blured when active scene. The below is the code I am using. Can you give me some advice to improve video ouput. Thank you so much. C:/ffmpeg_32/bin/ffmpeg -i udp://236.19.204.27:6000 -filter:v yadif=0:-1:1,fps=25 -vcodec libx264 -g 16 -vprofile main -level 30 -b:v 1000k -minrate 950k -maxrate 1050k -bufsize 100k -s: 720x576 -r 25 -ac 2 -ar 48000 -b:a 128k -filter:a volume=2 -f mpegts -muxrate 1200k udp://237.19.200.7:6000?pkt_size=1316 Best Regards ??ng ??c H?ng | Broadcast Technical Department | IPTV Center | VASC Software and Media Company | | 33 Thai Ha Street | ??ng ?a District | Hanoi | Vietnam| | Tel (+84) 37722728 ext 681/684 | Mobile (+84) 915 070 921| | ddhung at vasc.com.vn | www.mytv.com.vn | From ricardo at americasnet.com Mon Apr 22 05:05:58 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sun, 21 Apr 2013 20:05:58 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <23993583.8768.1366591402289.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> <5518136.8718.1366588839795.JavaMail.root@xeams> <23993583.8768.1366591402289.JavaMail.root@xeams> Message-ID: I did get the segmenter to work with h.264, I needed to add this to the command prior to -f segment: -vbsf h264_mp4toannexb -flags -global_header On Sun, Apr 21, 2013 at 5:43 PM, Ricardo Kleemann wrote: > That's great, I didn't realize it. > > I'm trying to convert the input to h.264 + AAC and output to the segmenter > but it's not working. Can you help me find the error in the command? > > ffmpeg -loglevel debug -threads 4 -i "rtmp://server1/live/livestream1" > -i_qfactor 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec > libx264 -s 480x270 -b:v 512k -b:a 56k -ar 22050 -map 0 -f segment > -segment_time 10 -segment_list live.m3u8 -segment_list_flags +live > live_%05d.ts > > The debug output is too large, but here is the end section: > > Successfully opened the file. > Parsing a group of options: output file live_%05d.ts. > Applying option vcodec (force video codec ('copy' to copy stream)) with > argument libx264. > Applying option s (set frame size (WxH or abbreviation)) with argument > 480x270. > Applying option b:v (video bitrate (please use -b:v)) with argument 512k. > Applying option b:a (video bitrate (please use -b:v)) with argument 56k. > Applying option ar (set audio sampling rate (in Hz)) with argument 22050. > Applying option map (set input stream mapping) with argument 0. > Applying option f (force format) with argument segment. > Successfully parsed a group of options. > Opening an output file: live_%05d.ts. > Successfully opened the file. > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'video_size' to value > '490x270' > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'pix_fmt' to value '0' > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'time_base' to value > '1/1000' > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'pixel_aspect' to value > '1/1' > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'sws_param' to value > 'flags=2' > [graph 0 input from stream 0:0 @ 0x153eb20] Setting 'frame_rate' to value > '30000/1001' > [graph 0 input from stream 0:0 @ 0x153eb20] w:490 h:270 pixfmt:yuv420p > tb:1/1000 fr:30000/1001 sar:1/1 sws_param:flags=2 > [scaler for output stream 0:0 @ 0x153fc40] Setting 'w' to value '480' > [scaler for output stream 0:0 @ 0x153fc40] Setting 'h' to value '270' > [scaler for output stream 0:0 @ 0x153fc40] Setting 'flags' to value '0x4' > [scaler for output stream 0:0 @ 0x153fc40] w:480 h:270 flags:'0x4' interl:0 > [format @ 0x160a2a0] compat: called with > args=[yuv420p|yuvj420p|yuv422p|yuv444p] > [format @ 0x160a2a0] Setting 'pix_fmts' to value > 'yuv420p|yuvj420p|yuv422p|yuv444p' > [AVFilterGraph @ 0x1535020] query_formats: 5 queried, 4 merged, 0 already > done, 0 delayed > [scaler for output stream 0:0 @ 0x153fc40] w:490 h:270 fmt:yuv420p sar:1/1 > -> w:480 h:270 fmt:yuv420p sar:49/48 flags:0x4 > [graph 1 input from stream 0:1 @ 0x1533300] Setting 'time_base' to value > '1/44100' > [graph 1 input from stream 0:1 @ 0x1533300] Setting 'sample_rate' to value > '44100' > [graph 1 input from stream 0:1 @ 0x1533300] Setting 'sample_fmt' to value > 's16p' > [graph 1 input from stream 0:1 @ 0x1533300] Setting 'channel_layout' to > value '0x4' > [graph 1 input from stream 0:1 @ 0x1533300] tb:1/44100 samplefmt:s16p > samplerate:44100 chlayout:0x4 > [audio format for output stream 0:1 @ 0x1536b40] Setting 'sample_fmts' to > value 's16' > [audio format for output stream 0:1 @ 0x1536b40] Setting 'sample_rates' to > value '22050' > [audio format for output stream 0:1 @ 0x1536b40] Setting 'channel_layouts' > to value '0x4|0x3' > [audio format for output stream 0:1 @ 0x1536b40] auto-inserting filter > 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the > filter 'audio format for output stream 0:1' > [AVFilterGraph @ 0x153d200] query_formats: 4 queried, 9 merged, 3 already > done, 0 delayed > [auto-inserted resampler 0 @ 0x153b540] ch:1 chl:mono fmt:s16p r:44100Hz -> > ch:1 chl:mono fmt:s16 r:22050Hz > [libx264 @ 0x1635460] using mv_range_thread = 24 > [libx264 @ 0x1635460] using SAR=49/48 > [libx264 @ 0x1635460] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.1 > Cache64 > [libx264 @ 0x1635460] profile High, level 2.1 > [libx264 @ 0x1635460] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec > - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: > cabac=1 > ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 > mixed_ref=1 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 > fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 > sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 > constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 > weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 > intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=512 ratetol=1.0 > qcomp=0.60 qpmin=10 qpmax=63 qpstep=4 ip_ratio=1.41 aq=1:1.00 > [segment @ 0x15d4080] Selected stream id:0 type:video > [mpegts @ 0x23f2cc0] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt > every 40 pkts > Output #0, segment, to 'live_%05d.ts': > Metadata: > author : > copyright : > description : > keywords : > rating : > title : > presetname : Custom > creationdate : Sun Apr 21 16:31:28 2013 > : > videodevice : AV/C Tape Recorder/Player > avclevel : 30 > avcprofile : 66 > videokeyframe_frequency: 3 > audiodevice : AV/C Tape Recorder/Player > audiochannels : 1 > audioinputvolume: 10 > encoder : Lavf55.3.100 > Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 480x270 [SAR 49:48 DAR > 49:27], 1001/30000, q=10-63, 512 kb/s, 90k tbn, 29.97 tbc > Stream #0:1, 0, 1/90000: Audio: mp2, 22050 Hz, mono, s16, 56 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (h264 -> libx264) > Stream #0:1 -> #0:1 (mp3 -> mp2) > Press [q] to stop, [?] for help > *** 1 dup! > *** drop! > Last message repeated 106 times > [libx264 @ 0x1635460] frame= 0 QP=28.46 NAL=3 Slice:I Poc:0 I:510 P:0 > SKIP:0 size=4440 bytes > [segment @ 0x15d4080] segment:'live_00000.ts' starts with packet stream:0 > pts:0 pts_time:0 frame:0 > [mpegts @ 0x23f2cc0] H.264 bitstream malformed, no startcode found, use the > h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) > [AVIOContext @ 0x23ea6a0] Statistics: 0 seeks, 0 writeouts > av_interleaved_write_frame(): Invalid argument > Statistics: 287291 bytes read, 0 seeks > [rtmp @ 0x153d800] Deleting stream... > > > > > On Sun, Apr 21, 2013 at 5:00 PM, Vlad Kroutik wrote: > > > You do not need m3u8-segmenter as ffmpeg support segmented output and its > > built it. Look at the ffmpeg docs or example: > > /usr/local/bin/ffmpeg -i hi.ts -loglevel debug > > -analyzeduration 6M -map 0 -c copy -f segment -segment_time 10 > > -segment_list hi.m3u8 hi_%05d.ts > > > > > > > > On Sun, Apr 21, 2013 at 2:53 PM, Ricardo Kleemann > > wrote: > > > > > On Sat, Apr 20, 2013 at 10:21 PM, JoelB > > wrote: > > > > > > > Should work for Roku and anything that supports HLS, I think android > > > does, > > > > various flash players, QuickTime 7 for Mac, safari(?) supposedly > > supports > > > > it as well. Generally, you want to encode h.264 video with AAC audio. > > > > > > > > Joel > > > > > > > > Sent from my iPhone > > > > > > > > > > So that's what I'm doing... I'm encoding to 264 and AAC, and piping to > > the > > > segmenter. > > > > > > I see the m3u8 file being written as well as all the other ts files. > > > > > > But if I try to access from an iPad it gives me different errors, > > sometimes > > > it says "cannot decode", sometimes it says cannot find the file. > > > > > > I'm assuming that the m3u8 file is supposed to be served up by a > standard > > > web server (apache). Do I need any special configuration in apache for > > > this? > > > > > > Here's my ffmpeg command: > > > > > > ffmpeg -threads 4 -i "rtmp://server1/live/livestream1" -f mpegts > > -i_qfactor > > > 0.71 -qcomp 0.6 -qmin 10 -qmax 63 -qdiff 4 -trellis 0 -vcodec libx264 > -s > > > 480x270 -b:v 512k -b:a 56k -ar 22050 - | m3u8-segmenter -i - -d 10 -p > > > cidadeviva/live -m /shared/media/videos/cidadeviva/live.m3u8 -u > > > http://web1.americasnet.com/videos/ > > > > > > It succeeds in reading the rtpm stream and does pipe the output to the > > > segmenter since I see all of the m3u8 and ts files being written. > > > > > > I'm able to access the m3u8 file via http, although I see apache often > > > complains that it can't find the file. Could that be because the file > is > > > constantly being written? > > > > > > Ricardo > > > > > > > > > > On Apr 20, 2013, at 8:56 PM, Ricardo Kleemann < > ricardo at americasnet.com > > > > > > > wrote: > > > > > > > > > Hi Joel, > > > > > > > > > > Thanks, I'll play around with that. > > > > > > > > > > Another question... is the segmenter only for iOS? Would other > > devices > > > > also > > > > > work with this? > > > > > > > > > > Ricardo > > > > > > > > > > On Sat, Apr 20, 2013 at 8:45 PM, JoelB > > > wrote: > > > > > > > > > >> Usually a segmented stream URL ends with .m3u8 so for ios, you > just > > > give > > > > >> the user a link to that m3u8 playlist URL and if you have a valid > > > > stream, > > > > >> the phone will just play it. > > > > >> > > > > >> http://inodes.org/bigbuck/big_buck.m3u8 > > > > >> > > > > >> Would probably be the res > > > > >> > > > > >> > > > > >> Joel > > > > >> Sent from my iPhone > > > > >> > > > > >> On Apr 20, 2013, at 5:35 PM, Ricardo Kleemann < > > > ricardo at americasnet.com> > > > > >> wrote: > > > > >> > > > > >>> Hi, > > > > >>> > > > > >>> I've been searching around for solutions to use ffmpeg to do live > > > > >> streaming > > > > >>> to html5. > > > > >>> > > > > >>> I understand that for iOS I need a segmenter? > > > > >>> > > > > >>> I've installed the latest ffmpeg and also m3u8-segmenter from > git, > > > but > > > > >> I'm > > > > >>> not really sure how to use it. The m3u8-segmenter site has an > > example > > > > >> like > > > > >>> this: > > > > >>> > > > > >>> ffmpeg -loglevel quiet -i big_buck_bunny.ogv -f mpegts - | \ > > > > >>> m3u8-segmenter -i - -d 10 -p tmp/big_buck_bunny -m > > tmp/big_buck.m3u8 > > > > >>> -u http://inodes.org/bigbuck/ > > > > >>> > > > > >>> But in terms of an HTML5 page source, how would I access the > stream > > > > that > > > > >> is > > > > >>> being generated by the segmenter? > > > > >>> > > > > >>> Is there a simpler way to use ffmpeg for live streaming? > > > > >>> > > > > >>> thanks > > > > >>> Ricardo > > > > >>> _______________________________________________ > > > > >>> ffmpeg-user mailing list > > > > >>> ffmpeg-user at ffmpeg.org > > > > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > >> _______________________________________________ > > > > >> ffmpeg-user mailing list > > > > >> ffmpeg-user at ffmpeg.org > > > > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > >> > > > > >> > > > > >> ------------------------------------------------------ > > > > >> Powered by Xeams. Visit xeams.com for more information > > > > >> ------------------------------------------------------ > > > > > _______________________________________________ > > > > > ffmpeg-user mailing list > > > > > ffmpeg-user at ffmpeg.org > > > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > _______________________________________________ > > > > ffmpeg-user mailing list > > > > ffmpeg-user at ffmpeg.org > > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > > > > > > > ------------------------------------------------------ > > > > Powered by Xeams. Visit xeams.com for more information > > > > ------------------------------------------------------ > > > > > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > ------------------------------------------------------ > > Powered by Xeams. Visit xeams.com for more information > > ------------------------------------------------------ > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------------------------------ > Powered by Xeams. Visit xeams.com for more information > ------------------------------------------------------ > From ricardo at americasnet.com Mon Apr 22 05:11:08 2013 From: ricardo at americasnet.com (Ricardo Kleemann) Date: Sun, 21 Apr 2013 20:11:08 -0700 Subject: [FFmpeg-user] HTML5 live streaming In-Reply-To: <12048931.8867.1366599988498.JavaMail.root@xeams> References: <6253590.7404.1366516016782.JavaMail.root@xeams> <30369007.7466.1366521739275.JavaMail.root@xeams> <5518136.8718.1366588839795.JavaMail.root@xeams> <23993583.8768.1366591402289.JavaMail.root@xeams> <12048931.8867.1366599988498.JavaMail.root@xeams> Message-ID: Alright... but back to my original question. I got the ffmpeg segmenter to work. I can playback via VLC. But it still does not work for HTML5