From phil_rhodes at rocketmail.com Tue Nov 1 00:22:46 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 31 Oct 2011 23:22:46 -0000 Subject: [FFmpeg-user] Encoding h264 with 4:4:4 chroma In-Reply-To: <4EAE57A1.50902@bbc.co.uk> References: <4EAE57A1.50902@bbc.co.uk> Message-ID: >> That is why a snapshot is offered on http://ffmpeg.org/download.html (it >> contains current git head). What on earth am I supposed to do with that? I am not a software engineer. If it was a Visual Studio project, I might just about be able to load it into the IDE and hit F6. But it isn't. So I can't. >> PS: Compiling natively on mingw for Windows if far easier imo than >> cross-compiling. > > Interesting that you should say that as I have heard others say quite > the reverse... Me too. > the inconsistent build requirements and need to 'tweak' Makefile's and > the like to accommodate complier version issues etc. (Not all libraries > have as refined a configure system as ffmpeg's) It's my direct experience that this is the case with almost all opensource software. Much as the configure/make/install line is given out quite a lot, from what I've seen it works rather rarely. Works, I mean, implying that all of the advertised features are available, it's properly integrated with the desktop environment, properly configured with sensible defaults, etc. P From dev at rarevision.com Tue Nov 1 00:32:59 2011 From: dev at rarevision.com (Thomas Worth) Date: Mon, 31 Oct 2011 16:32:59 -0700 Subject: [FFmpeg-user] Encoding h264 with 4:4:4 chroma In-Reply-To: References: <4EAE57A1.50902@bbc.co.uk> Message-ID: On Mon, Oct 31, 2011 at 4:22 PM, Phil Rhodes wrote: > >>> That is why a snapshot is offered on http://ffmpeg.org/download.html (it >>> contains current git head). > > What on earth am I supposed to do with that? I am not a software engineer. > If it was a Visual Studio project, I might just about be able to load it > into the IDE and hit F6. But it isn't. So I can't. > >>> PS: Compiling natively on mingw for Windows if far easier imo than >>> cross-compiling. >> >> Interesting that you should say that as I have heard others say quite the >> reverse... > > Me too. > >> the inconsistent build requirements and need to 'tweak' Makefile's and the >> like to accommodate complier version issues etc. (Not all libraries have as >> refined a configure system as ffmpeg's) > > It's my direct experience that this is the case with almost all opensource > software. Much as the configure/make/install line is given out quite a lot, > from what I've seen it works rather rarely. Works, I mean, implying that all > of the advertised features are available, it's properly integrated with the > desktop environment, properly configured with sensible defaults, etc. I sympathize. Compiling FFmpeg on Windows is much harder than on a Unix-ish sytem. However, it can be done. You'll want to install MinGW and MSYS, which give you the GNU utilities you need and a shell. You'll need to run configure from the MSYS shell. It won't work using the Windows command prompt, although you can run stuff you compile through the command prompt. Once you get MSYS working, it acts pretty much like a Unix system. I haven't tried cross-compiling. My own software uses Windows libraries, so I need to develop from the Windows machine anyway. From james.darnley at gmail.com Tue Nov 1 00:36:51 2011 From: james.darnley at gmail.com (James Darnley) Date: Tue, 01 Nov 2011 00:36:51 +0100 Subject: [FFmpeg-user] FFMpeg symbols are not defined In-Reply-To: References: Message-ID: <4EAF3113.4020801@gmail.com> On 2011-10-30 13:14, ??????? ????????? wrote: > After compilation the head revision of FFMPEG, my linker tells me that > some symbols are undefined. Those look like zlib functions to me. Check that the linker is trying to link to it and that you have the library installed. Configure should have disabled the features that require zlib if it isn't available. From dimka87 at gmail.com Tue Nov 1 04:20:50 2011 From: dimka87 at gmail.com (=?KOI8-R?B?5M3J1NLJyiDnz8zP18nOz9c=?=) Date: Tue, 1 Nov 2011 06:20:50 +0300 Subject: [FFmpeg-user] FFMpeg symbols are not defined In-Reply-To: <4EAF3113.4020801@gmail.com> References: <4EAF3113.4020801@gmail.com> Message-ID: Thank you. -lz saved my life! 2011/11/1 James Darnley > On 2011-10-30 13:14, ??????? ????????? wrote: > >> After compilation the head revision of FFMPEG, my linker tells me that >> some symbols are undefined. >> > > Those look like zlib functions to me. Check that the linker is trying to > link to it and that you have the library installed. Configure should have > disabled the features that require zlib if it isn't available. > > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > -- ? ?????????, ??????? ??????? ???? ?? ??? ??? "???" ????????? ??????? From cessen at cessen.com Tue Nov 1 09:24:36 2011 From: cessen at cessen.com (Nathan Vegdahl) Date: Tue, 1 Nov 2011 01:24:36 -0700 Subject: [FFmpeg-user] Encoding h264 with 4:4:4 chroma In-Reply-To: References: Message-ID: Compiled from git as per these instructions: http://ubuntuforums.org/showthread.php?t=786095 4:4:4 chroma with libx264 appears to work fine now. For example, I am able to do fully lossless h264 encoding (including full 4:4:4 chroma) via this command: ffmpeg -i input_file -vcodec libx264 -pix_fmt yuv444p -crf 0 output_file I've been testing with already compressed 4:2:0 input, however, since I am not at my workstation right now where my input files reside. But the format information (e.g. it claims to be 4:4:4) of the output is correct, at least. I will test more thoroughly when I have access to my workstation again. Thanks much, Carl! --Nathan On Mon, Oct 31, 2011 at 3:23 PM, Carl Eugen Hoyos wrote: > Nathan Vegdahl cessen.com> writes: > >> I will gladly test with git head next, but can I first get explicit >> confirmation that this is even supposed to work yet? > > It is supposed to work (and I suspect no -pix_fmt will be needed if your input > material is 4:4:4). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From thomas at pixelpartner.de Tue Nov 1 10:45:57 2011 From: thomas at pixelpartner.de (PixelPartner) Date: Tue, 1 Nov 2011 10:45:57 +0100 (CET) Subject: [FFmpeg-user] recent change to libavfilter hinders the movie filter to work on Windows Message-ID: <934154547.70593.1320140757704.JavaMail.open-xchange@oxgw01.schlund.de> Working with the following version under Windows 7 64 ? ffmpeg N-34031-ge403a97 libavutil??? 51. 22. 0 / 51. 22. 0 libavcodec?? 53. 23. 0 / 53. 23. 0 libavformat? 53. 17. 0 / 53. 17. 0 libavdevice? 53.? 4. 0 / 53.? 4. 0 libavfilter?? 2. 45. 0 /? 2. 45. 0 libswscale??? 2.? 1. 0 /? 2.? 1. 0 libpostproc? 51.? 2. 0 / 51.? 2. 0 ? There was recently a change that cripples the libavfilter movie= to accept Windows-Style file names. All of the following causes failure: movie=C:\folder\file.ext movie=C:\\folder\\file.ext movie='C:\folder\file.ext' movie=C:/folder/file.ext ? If I remove C: it works, but I assume it works only with files on the "current" drive. ? It seems to me that someone added paramter parsing \\ -> \ and : for values also to the movie filter option list. Does this make any sense ? Thomas Kumlehn PIXEL PARTNER? +49 177 6 990 990 Member of the networksFacebook,Twitter,LinkedIn,XINGandcrew-united. sent from myiPad 3-D From cehoyos at ag.or.at Tue Nov 1 13:00:07 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 12:00:07 +0000 (UTC) Subject: [FFmpeg-user] recent change to libavfilter hinders the movie filter to work on Windows References: <934154547.70593.1320140757704.JavaMail.open-xchange@oxgw01.schlund.de> Message-ID: PixelPartner pixelpartner.de> writes: > There was recently a change that cripples the libavfilter movie= to accept > Windows-Style file names. Please find the version introducing the problem (with git bisect) Carl Eugen From mcaramb at rocketmail.com Tue Nov 1 15:36:02 2011 From: mcaramb at rocketmail.com (Mike Carambat) Date: Tue, 1 Nov 2011 07:36:02 -0700 (PDT) Subject: [FFmpeg-user] Blue Cherry PV-143 (BT878 Chipset) not working! Message-ID: <1320158162.42207.YahooMailNeo@web113314.mail.gq1.yahoo.com> Anyone have a working command line for the Blue Cherry PV-143 (BT878 Chipset) capture card?? It's a rather standard chipset for linux v4l based systems, and should work well, but I keep getting ioctl errors when issuing the following simplified command: ffmpeg -an -f video4linux2 -i /dev/video0 test.avi I don't believe this card has any type of hardware compressor, and simply outputs raw YUV video. I can query the card with v4l2-ctl and I've got all the specifics on size, pixel-type, fps, etc. I tried verbatim specifying these details before in the command line, but it didn't help. I also tried using -vcodec rawvideo. No dice. Also, when doing: ffplay -f video4linux2 /dev/video0 I get the same ioctl error, but I can see in the error message that it correctly detected all the aforementioned specifics and applied them automatically. So the card is being seen, read and configured, but the data stream is coming across as unusable. I apologize for not having the exact error in front of me, but won't be back at that computer until later in the day. I'm posting this from memory, so I might have some ammunition to return with this evening. I'm running Ubuntu Server 11.10 with the latest standard apt installation package for ffmpeg (0.7.2-4:0.7.2-1ubuntu1) Thanks for any advice Mike From vlad.ion at gmail.com Tue Nov 1 15:36:30 2011 From: vlad.ion at gmail.com (oblivion) Date: Tue, 1 Nov 2011 07:36:30 -0700 (PDT) Subject: [FFmpeg-user] Muxing raw H264 to container format (MP4, MOV) In-Reply-To: <1319703002850-3943546.post@n4.nabble.com> References: <1319124086069-3922255.post@n4.nabble.com> <1319125913860-3922333.post@n4.nabble.com> <1319623314753-3940025.post@n4.nabble.com> <4EA7E4B2.6070304@mdsh.com> <1319703002850-3943546.post@n4.nabble.com> Message-ID: <1320158190449-3963745.post@n4.nabble.com> Considering nobody knows why my h264 file is skipped maybe it's a bug and I should submit a bug report? I find the fact that there is no error very strange. If I give an empty (or invalid) file as input to ffmpeg then I get an error so it means it does read my file but decides to ignore it without any message. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Muxing-raw-H264-to-container-format-MP4-MOV-tp3922255p3963745.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From diesql at googlemail.com Tue Nov 1 15:44:25 2011 From: diesql at googlemail.com (Jamie Tufnell) Date: Tue, 1 Nov 2011 10:44:25 -0400 Subject: [FFmpeg-user] VC-1 encoding on Linux with IPP Message-ID: Hi I'm looking for suggestions on how to encode to WMV3 (Windows Media Video 9) in a Linux remote server environment. As far as I know FFmpeg won't do this out of the box (or at all?) So here are the options as I see them so far: 1) Set up a Windows server/VM and use some Windows software. It needs to be automated and reliable. I have no idea if this is feasible. 2) Drop BIG$$$ on something from MainConcept or whoever. 3) Try the Intel Performance Primitives libraries. They have a code sample that encodes VC-1, which I think is basically WMV3? 4) Hope that FFmpeg/libavcodec can link against the IPP libraries and enable VC-1 as an output format? 5) Find evidence that H264 has just as good penetration/quality/filesize as WMV3 on all Windows systems and devices, and ditch WMV3. Any comments greatly appreciated. Thanks J From dev at rarevision.com Tue Nov 1 16:53:40 2011 From: dev at rarevision.com (Thomas Worth) Date: Tue, 1 Nov 2011 08:53:40 -0700 Subject: [FFmpeg-user] AC-3 encoding questions Message-ID: I'm trying to encode some 6 channel, 24-bit WAV files, but am getting some warnings from FFmpeg: [ac3 @ 0x101920200] channel_layout not specified Ok, now I've used -channel_layout XXX, but I don't know where these layouts are documented. I've tried all the way up to 120, and get stuff like this: Stream #0:0: Audio: ac3, 48000 Hz, 6 channels (LFE|BL|BR|FLC), flt, 320 kb/s I assume the "LFE|BL|BR|FLC" refers to how each channel will be mapped, but I've not been able to find exactly what I want, which is this: (FL|FR|FC|LFE|BL|BR) Assuming "BL" and "BR" mean "Back Left" and "Back Right." If so, this isn't really the correct terminology. It should be "Right Surround" and "Left Surround." The standard 5.1 channel layout is this: 1. Left 2. Right 3. Center 4. LFE 5. Left Surround 6. Right Surround That's how the channels are specified in the WAV file. I hope that if FFmpeg can't determine the channel layout, it just encodes each channel in the same order as they are in the WAV. Also, the duration of the AC-3 file is slightly different from the original WAV. With long projects, this can introduce audio drift. Is there anything I can do to avoid this? From cehoyos at ag.or.at Tue Nov 1 17:02:02 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 16:02:02 +0000 (UTC) Subject: [FFmpeg-user] Blue Cherry PV-143 (BT878 Chipset) not working! References: <1320158162.42207.YahooMailNeo@web113314.mail.gq1.yahoo.com> Message-ID: Mike Carambat rocketmail.com> writes: > It's a rather standard chipset for linux v4l based systems, and should work > well, but I keep getting ioctl errors when issuing the following simplified > command: > > ffmpeg -an -f video4linux2 -i /dev/video0 test.avi (Complete, uncut output missing) If your card really does not use compression, I assume this is a regression. If you want to help, please find the version introducing the bug with git bisect. Carl Eugen From cehoyos at ag.or.at Tue Nov 1 17:04:41 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 16:04:41 +0000 (UTC) Subject: [FFmpeg-user] AC-3 encoding questions References: Message-ID: Thomas Worth rarevision.com> writes: > I'm trying to encode some 6 channel, 24-bit WAV files, but am getting > some warnings from FFmpeg: > > [ac3 @ 0x101920200] channel_layout not specified (Complete, uncut output missing.) If the wav file really has no channel layout specified, the ac3 encoder will guess the layout (and says so if I am not wrong) as 5.1. Only specify a layout if you don't like what the encoder guesses. Carl Eugen From dev at rarevision.com Tue Nov 1 17:24:14 2011 From: dev at rarevision.com (Thomas Worth) Date: Tue, 1 Nov 2011 09:24:14 -0700 Subject: [FFmpeg-user] AC-3 encoding questions In-Reply-To: References: Message-ID: On Tue, Nov 1, 2011 at 9:04 AM, Carl Eugen Hoyos wrote: > Thomas Worth rarevision.com> writes: > >> I'm trying to encode some 6 channel, 24-bit WAV files, but am getting >> some warnings from FFmpeg: >> >> [ac3 @ 0x101920200] channel_layout not specified > > (Complete, uncut output missing.) > > If the wav file really has no channel layout specified, the ac3 encoder will > guess the layout (and says so if I am not wrong) as 5.1. > Only specify a layout if you don't like what the encoder guesses. Hi Carl. Sorry, here's full output. Obviously the channel layout is wrong, but you get the idea: macpro:AUDIO user$ ffmpeg -i Reel_02_0999.wav -vn -acodec ac3 -ab 320k -ac 6 -channel_layout 120 -y Reel_02_TEST.ac3 ffmpeg version 0.8.5.git, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 20 2011 02:47:48 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --disable-doc --enable-static --disable-shared --enable-gpl --enable-nonfree --enable-version3 --enable-libfaac --enable-libx264 --enable-libvpx --enable-libmp3lame --cc=gcc-4.2 --enable-pthreads --arch=x86_64 --target-os=darwin --extra-cflags='-I/sources/faac-1.28/include -I/sources/libvpx -I/sources/lame-3.98.4 -I/sources/x264 -isysroot /Developer/SDKs/MacOSX10.6.sdk -mmacosx-version-min=10.6' --extra-ldflags='-L/sources/faac-1.28/libfaac/.libs -L/sources/libvpx -L/sources/lame-3.98.4/libmp3lame/.libs -L/sources/x264 -isysroot /Developer/SDKs/MacOSX10.6.sdk -mmacosx-version-min=10.6' libavutil 51. 22. 0 / 51. 22. 0 libavcodec 53. 22. 0 / 53. 22. 0 libavformat 53. 17. 0 / 53. 17. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 44. 1 / 2. 44. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [wav @ 0x101805800] parser not found for codec pcm_s24le, packets or times may be invalid. [wav @ 0x101805800] max_analyze_duration 5000000 reached at 5003458 Input #0, wav, from 'Reel_02_0999.wav': Duration: 00:16:53.22, bitrate: 6912 kb/s Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 6 channels, s32, 6912 kb/s Incompatible sample format 's32' for codec 'ac3', auto-selecting format 'flt' [ac3 @ 0x101920200] Specified channel_layout is not supported Output #0, ac3, to 'Reel_02_TEST.ac3': Stream #0:0: Audio: ac3, 48000 Hz, 6 channels (LFE|BL|BR|FLC), flt, 320 kb/s Stream mapping: Stream #0.0 -> #0.0 (pcm_s24le -> ac3) Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height From cehoyos at ag.or.at Tue Nov 1 17:42:54 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 16:42:54 +0000 (UTC) Subject: [FFmpeg-user] AC-3 encoding questions References: Message-ID: Thomas Worth rarevision.com> writes: > >> [ac3 @ 0x101920200] channel_layout not specified > > > > (Complete, uncut output missing.) > > > > If the wav file really has no channel layout specified, the ac3 encoder will > > guess the layout (and says so if I am not wrong) as 5.1. > > Only specify a layout if you don't like what the encoder guesses. > > Hi Carl. Sorry, here's full output. Obviously the channel layout is > wrong, but you get the idea: No, because I suspect it is working fine if you do not specify -channel_layout (that should come before the input file imo), or which layout would you like for six cahnnel audio? Carl Eugen From dev at rarevision.com Tue Nov 1 17:57:31 2011 From: dev at rarevision.com (Thomas Worth) Date: Tue, 1 Nov 2011 09:57:31 -0700 Subject: [FFmpeg-user] AC-3 encoding questions In-Reply-To: References: Message-ID: On Tue, Nov 1, 2011 at 9:42 AM, Carl Eugen Hoyos wrote: > Thomas Worth rarevision.com> writes: > >> >> [ac3 @ 0x101920200] channel_layout not specified >> > >> > (Complete, uncut output missing.) >> > >> > If the wav file really has no channel layout specified, the ac3 encoder will >> > guess the layout (and says so if I am not wrong) as 5.1. >> > Only specify a layout if you don't like what the encoder guesses. >> >> Hi Carl. Sorry, here's full output. Obviously the channel layout is >> wrong, but you get the idea: > > No, because I suspect it is working fine if you do not specify -channel_layout > (that should come before the input file imo), or which layout would you like for > six cahnnel audio? Here is the layout I would like: FL|FR|FC|LFE|BL|BR Assuming that "BL" and "BR" mean "back left" and "back right," which would be surround channels. That is how the channels are arranged in the WAV file. Do you know which channel_layout number I need to use to achieve this? I'll try putting it before -i as you suggested. From asai at globalchangemusic.org Tue Nov 1 17:59:12 2011 From: asai at globalchangemusic.org (Asai) Date: Tue, 01 Nov 2011 09:59:12 -0700 Subject: [FFmpeg-user] Quickest Route From Final Cut Pro Message-ID: <4EB02560.4050801@globalchangemusic.org> Greetings, I've been working with a studio which uses Final Cut exclusively, but unfortunately it seems that the quickest export from Final Cut (that we know of) is straight to Apple Intermediate Codec. If they export to ProRes it takes 2 - 3 times the length of the video before we can get it into FFMPEG. Is there any way to force FFMPEG to accept icod by ignoring the container format? If not, is there a faster way to get from Final Cut to FFMPEG without having to go through an intermediate step? -- --asai From cehoyos at ag.or.at Tue Nov 1 18:15:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 17:15:36 +0000 (UTC) Subject: [FFmpeg-user] AC-3 encoding questions References: Message-ID: Thomas Worth rarevision.com> writes: > > I suspect it is working fine if you do not specify -channel_layout > > (that should come before the input file imo), or which layout would you like > > for six cahnnel audio? > > Here is the layout I would like: Please try: ffmpeg -i Reel_02_0999.wav -ab 320k -y Reel_02_TEST.ac3 That should work as expected. If not, please provide complete, uncut output and explain what went wrong. Carl Eugen From stefasab at gmail.com Tue Nov 1 19:00:04 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Tue, 1 Nov 2011 19:00:04 +0100 Subject: [FFmpeg-user] recent change to libavfilter hinders the movie filter to work on Windows In-Reply-To: <934154547.70593.1320140757704.JavaMail.open-xchange@oxgw01.schlund.de> References: <934154547.70593.1320140757704.JavaMail.open-xchange@oxgw01.schlund.de> Message-ID: <20111101180004.GA20302@arborea> On date Tuesday 2011-11-01 10:45:57 +0100, PixelPartner encoded: > Working with the following version under Windows 7 64 > ? > ffmpeg N-34031-ge403a97 > libavutil??? 51. 22. 0 / 51. 22. 0 > libavcodec?? 53. 23. 0 / 53. 23. 0 > libavformat? 53. 17. 0 / 53. 17. 0 > libavdevice? 53.? 4. 0 / 53.? 4. 0 > libavfilter?? 2. 45. 0 /? 2. 45. 0 > libswscale??? 2.? 1. 0 /? 2.? 1. 0 > libpostproc? 51.? 2. 0 / 51.? 2. 0 > ? > There was recently a change that cripples the libavfilter movie= to accept Windows-Style file names. > All of the following causes failure: AFAIK there is no regression, there was no change in the way the movie command is parsed. > movie=C:\folder\file.ext > movie=C:\\folder\\file.ext > movie='C:\folder\file.ext' > movie=C:/folder/file.ext > ? > If I remove C: it works, but I assume it works only with files on the "current" drive. > ? > It seems to me that someone added paramter parsing \\ -> \ and : for values also to the movie filter option list. > Does this make any sense ? A filename of this kind: movie=C:\folder\file.ext is de-escaped three times before to be used as a filename. So you need to perform three level of escaping when providing such a filename to libavfilter via commandline. 1. first escaping, de-escaping is performed by the movie filter itself, will convert \X -> X, and will preserve strings between single quotes. movie=C:\folder\file.ext -> movie='C:\folder\file.ext' 2. second escaping, de-escaping is performed when reading the movie file description: movie='C:\folder\file.ext' -> movie=\'C:\\folder\\file.ext\' 3. third escaping, de-escaping is performed by your shell, in the case of a bash shell: movie=\'C:\\folder\\file.ext\' -> "movie=\'C:\\\\folder\\\\file.ext\'" I understand that all this is confusing and should be documented better, but I see no other way to deal with strings which contain characters which are special to the parsing function used (libavutil/parseutils.h:av_get_token()). Please tell if the attached patch is helpful to you. -- ffmpeg-user random tip #6 Please follow netiquette rules while posting to ffmpeg-user: http://linux.sgms-centre.com/misc/netiquette.php -------------- next part -------------- A non-text attachment was scrubbed... Name: 0002-lavfi-movie-extend-documentation-regarding-filename-.patch Type: text/x-diff Size: 1718 bytes Desc: not available URL: From nicolas.george at normalesup.org Tue Nov 1 20:17:43 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Tue, 1 Nov 2011 20:17:43 +0100 Subject: [FFmpeg-user] how to make frame repeat? In-Reply-To: <1320088055.11760.YahooMailNeo@web46203.mail.sp1.yahoo.com> References: <1320031260.60090.YahooMailNeo@web46206.mail.sp1.yahoo.com> <1320087901.90174.YahooMailNeo@web46209.mail.sp1.yahoo.com> <1320088055.11760.YahooMailNeo@web46203.mail.sp1.yahoo.com> Message-ID: <20111101191743.GB19681@phare.normalesup.org> Le decadi 10 brumaire, an CCXX, Hardik Sharma a ?crit?: > Also I want exact number of frames to complete my experiment where I am > comparing some values frame by frame from different codecs. So if there is > less number of frame, it will go out of synch and gonna fail my > experiment. Your mails are barely readable, let alone understandable. I do not know if the culprit is yahoo mail or something else, but you need to change something, otherwise I do not know how long people will still make an effort to read you. As to what I understand on the heart of your problem, I believe the short answer is this: You need to use the timestamps of the frames and not just count them. Once you do that, you do not need to worry if a frame gets dropped in the first place. 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 dev at rarevision.com Tue Nov 1 20:33:16 2011 From: dev at rarevision.com (Thomas Worth) Date: Tue, 1 Nov 2011 12:33:16 -0700 Subject: [FFmpeg-user] AC-3 encoding questions In-Reply-To: References: Message-ID: On Tue, Nov 1, 2011 at 10:15 AM, Carl Eugen Hoyos wrote: > Thomas Worth rarevision.com> writes: > >> > I suspect it is working fine if you do not specify -channel_layout >> > (that should come before the input file imo), or which layout would you like >> > for six cahnnel audio? >> >> Here is the layout I would like: > > Please try: > ffmpeg -i Reel_02_0999.wav -ab 320k -y Reel_02_TEST.ac3 > > That should work as expected. If not, please provide complete, uncut output and > explain what went wrong. The channel mapping seems to work. The only issue I have now is that both Apple DVD Studio Pro and Adobe Encore are reporting that AC-3 files encoded with FFmpeg are slightly longer (about 1 frame at 23.976 fps) than files encoded with those programs. Any idea why this would be? This causes sync problems with long audio files. From cehoyos at ag.or.at Tue Nov 1 21:24:02 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 1 Nov 2011 20:24:02 +0000 (UTC) Subject: [FFmpeg-user] AC-3 encoding questions References: Message-ID: Thomas Worth rarevision.com> writes: > > Please try: > > ffmpeg -i Reel_02_0999.wav -ab 320k -y Reel_02_TEST.ac3 > > > > That should work as expected. If not, please provide complete, uncut output > > and explain what went wrong. > > The channel mapping seems to work. Thank you. Now back to your original question (how does -channel_layout work?) - I just could not believe it is needed to convert a 5.1 sample to ac3: If you look into libavutil/audioconvert.h, you will find defines for speaker positions (in hex). If you add the values for the speakers that are in your stream, you get the value for -channel_layout: Stereo: 3 LRC: 7 Quad: 51 etc. Note that channel order is always assumed to be wav order, you cannot specify a different channel order (or in other words: FFmpeg internally only supports wav channel order)! Ticket #98 contains a sample (and an unfinished patch) that requires channel reordering. Regarding your length problem: Please consider opening a ticket on trac, don't forget to provide a command line including complete, uncut output that allows to reproduce the problem. Carl Eugen From rogerdpack2 at gmail.com Tue Nov 1 22:32:38 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 1 Nov 2011 15:32:38 -0600 Subject: [FFmpeg-user] watchable video tester program In-Reply-To: <1319890847.17196.YahooMailNeo@web36406.mail.mud.yahoo.com> References: <1319890847.17196.YahooMailNeo@web36406.mail.mud.yahoo.com> Message-ID: mplayer? mencoder? On Sat, Oct 29, 2011 at 6:20 AM, Tom Sparks wrote: > I am looking for a program that can scan though a videoand check to see if it is watchable? > Are there macroblock errors? > Is there dead/missing sound? > Are there Frame errors? > > it needs to be able to work on non-original (re-compressed) video (mpg2/H.264 to Xvid) > > > > > -- > tom_a_sparks "It's a nerdy thing I like to do" > Please use ISO approved file formats excluding Office Open XML - http://www.gnu.org/philosophy/no-word-attachments.html > Ubuntu wiki page https://wiki.ubuntu.com/tomsparks > 3 x (x)Ubuntu 10.04, Amiga A1200 WB 3.1, UAE AF 2006 Premium Edition, AF 2012 Plus Edition, Sam440 AOS 4.1.2, Roland DXY-1300 pen plotter, Cutok DC330 cutter/pen plotter > Wanted: RiscOS system, GEOS system (C64/C128), Atari ST, Apple Macintosh (6502/68k/PPC only) > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From amit.dor.shifer at gmail.com Wed Nov 2 00:33:55 2011 From: amit.dor.shifer at gmail.com (Amit Dor-Shifer) Date: Wed, 2 Nov 2011 10:33:55 +1100 Subject: [FFmpeg-user] decoding raw RTP? Message-ID: Attached is a sample of mulaw-encoded audio, packeted as RTP. Is there a way for ffmpeg to decode this? $ ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample /tmp/out.wav ffmpeg version N-34304-gc0dbab9, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 31 2011 17:54:58 with gcc 4.4.5 configuration: --disable-vaapi --enable-libtheora --enable-libvorbis --enable-nonfree --enable-postproc --enable-version3 --enable-libspeex libavutil 51. 22. 0 / 51. 22. 0 libavcodec 53. 26. 0 / 53. 26. 0 libavformat 53. 18. 0 / 53. 18. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 45. 2 / 2. 45. 2 libswscale 2. 1. 0 / 2. 1. 0 [rtp @ 0x1a2c8c0] Guessing on RTP content - if not received properly you need an SDP file describing it [rtp @ 0x1a2c8c0] Estimating duration from bitrate, this may be inaccurate Input #0, rtp, from 'testing-123.mulaw.rtp.sample': Duration: N/A, bitrate: N/A Stream #0:0: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s Output #0, wav, to '/tmp/out.wav': Metadata: encoder : Lavf53.18.0 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1 channels, s16, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 (pcm_mulaw -> pcm_s16le) Press [q] to stop, [?] for help size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s video:0kB audio:0kB global headers:0kB muxing overhead inf% ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample -------------- next part -------------- A non-text attachment was scrubbed... Name: testing-123.mulaw.rtp.sample Type: application/octet-stream Size: 20468 bytes Desc: not available URL: From lamia at jeack.com.au Wed Nov 2 01:45:09 2011 From: lamia at jeack.com.au (Bazza) Date: Wed, 02 Nov 2011 11:45:09 +1100 Subject: [FFmpeg-user] Muxing raw H264 to container format (MP4, MOV) References: <1319124086069-3922255.post@n4.nabble.com> <1319125913860-3922333.post@n4.nabble.com> <1319623314753-3940025.post@n4.nabble.com> <4EA7E4B2.6070304@mdsh.com> <1319703002850-3943546.post@n4.nabble.com> <1320158190449-3963745.post@n4.nabble.com> Message-ID: On Tue, 1 Nov 2011 07:36:30 -0700 (PDT), oblivion wrote: >Considering nobody knows why my h264 file is skipped maybe it's a bug and I >should submit a bug report? > >I find the fact that there is no error very strange. If I give an empty (or >invalid) file as input to ffmpeg then I get an error so it means it does >read my file but decides to ignore it without any message. Well I'm most certainly no expert but I tried your little example. The line that worked for me ... ffmpeg -threads 3 -i in.h264 -vcodec mpeg4 -r 25 -sameq outfile.mp4 I didn't need to " -f h264 " but I did need to force the codec which, running the way you expressed it, complained about the frame rates being incompatible (and quite rightfully ?? so because MP4 is only 25 fps ???) ---------------------------------------------- As reported C:\Users\Administrator>ffmpeg -threads 3 -i in.h264 -vcodec mpeg4 -r 25 -sameq .mp4 ffmpeg version git-N-29638-g95f163b, Copyright (c) 2000-2011 the FFmpeg developers built on May 6 2011 12:50:01 with gcc 4.5.3 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable nable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-l able-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-lib le-zlib --pkg-config=pkg-config libavutil 51. 2. 1 / 51. 2. 1 libavcodec 53. 3. 0 / 53. 3. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 0. 14. 0 / 0. 14. 0 [h264 @ 018AAE60] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (50/2) Input #0, h264, from 'e:\molly\in.h264': Duration: N/A, bitrate: N/A Stream #0.0: Video: h264 (High), yuv420p, 1280x960, 25 fps, 25 tbr, 1200k tbn, 50 tbc [buffer @ 03E6FD20] w:1280 h:960 pixfmt:yuv420p Output #0, mp4, to 'e:\molly\outfile.mp4': Metadata: encoder : Lavf53.0.3 Stream #0.0: Video: mpeg4, yuv420p, 1280x960, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 85 fps= 15 q=0.0 Lsize= 8933kB time=3.40 bitrate=21523.8kbits/s dup=0 drop=172 video:8932kB audio:0kB global headers:0kB muxing overhead 0.016171% ---------------------------------------------- From lamia at jeack.com.au Wed Nov 2 02:14:01 2011 From: lamia at jeack.com.au (Bazza) Date: Wed, 02 Nov 2011 12:14:01 +1100 Subject: [FFmpeg-user] Muxing raw H264 to container format (MP4, MOV) References: <1319124086069-3922255.post@n4.nabble.com> <1319125913860-3922333.post@n4.nabble.com> <1319623314753-3940025.post@n4.nabble.com> <4EA7E4B2.6070304@mdsh.com> <1319703002850-3943546.post@n4.nabble.com> <1320158190449-3963745.post@n4.nabble.com> Message-ID: On Tue, 1 Nov 2011 07:36:30 -0700 (PDT), oblivion wrote: >Considering nobody knows why my h264 file is skipped maybe it's a bug and I >should submit a bug report? > >I find the fact that there is no error very strange. If I give an empty (or >invalid) file as input to ffmpeg then I get an error so it means it does >read my file but decides to ignore it without any message. Well I'm most certainly no expert but I tried your little example. The line that worked for me ... ffmpeg -threads 3 -i in.h264 -vcodec mpeg4 -r 25 -sameq outfile.mp4 I didn't need to " -f h264 " but I did need to force the codec which, running the way you expressed it, complained about the frame rates being incompatible (and quite rightfully ?? so because MP4 is only 25 fps ???) FWIW, I tried your lines and it "sort of" worked. The '-f h264' was needed. There's sound? The quality was crude and the file size truncated at about 4 Mb but refused to play in Windows Media beyond 8 secs of the file's max duration. Bar "flipping/tilting" = 1 tilt/sec. In contrast, my working line rendered a bar sway of about 2 tilts/sec. Perhaps ffprobe -show_streams your_original_infile.xxx ? ---------------------------------------------- As reported C:\Users\Administrator>ffmpeg -threads 3 -i in.h264 -vcodec mpeg4 -r 25 -sameq outfile.mp4 ffmpeg version git-N-29638-g95f163b, Copyright (c) 2000-2011 the FFmpeg developers [ etc stuff skipped ] libavutil 51. 2. 1 / 51. 2. 1 libavcodec 53. 3. 0 / 53. 3. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 0. 14. 0 / 0. 14. 0 [h264 @ 018AAE60] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (50/2) Input #0, h264, from 'e:\in.h264': Duration: N/A, bitrate: N/A Stream #0.0: Video: h264 (High), yuv420p, 1280x960, 25 fps, 25 tbr, 1200k tbn, 50 tbc [buffer @ 03E6FD20] w:1280 h:960 pixfmt:yuv420p Output #0, mp4, to 'e:\outfile.mp4': Metadata: encoder : Lavf53.0.3 Stream #0.0: Video: mpeg4, yuv420p, 1280x960, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 85 fps= 15 q=0.0 Lsize= 8933kB time=3.40 bitrate=21523.8kbits/s dup=0 drop=172 video:8932kB audio:0kB global headers:0kB muxing overhead 0.016171% ---------------------------------------------- From hardik.sharma22 at yahoo.com Wed Nov 2 03:06:25 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Tue, 1 Nov 2011 19:06:25 -0700 (PDT) Subject: [FFmpeg-user] where to look for missing frame? Message-ID: <1320199585.66183.YahooMailNeo@web46213.mail.sp1.yahoo.com> Hi Can anyone tell that where and how we can check if we are missing some frame after decoding in a GOP in the code? So that at the missing point I can use current_picture_ptr and previous_picture_ptr to replace previous frame for frame copy concealment. Thanks. From vandung.tran at vn.panasonic.com Wed Nov 2 05:08:59 2011 From: vandung.tran at vn.panasonic.com (vandung.tran at vn.panasonic.com) Date: Wed, 2 Nov 2011 11:08:59 +0700 Subject: [FFmpeg-user] can ffplay show how many frames are decoded per sencond when running? Message-ID: I am playing video by ffplay on simulator, and the speed is quite slow. Is there anyway to know how many frames are decoded per sencond when playing video by ffplay? Thank ! "The information in this e-mail (including attachments) is confidential and is only intended for use by the addressee. If you are not the intended recipient or addressee, please notify us immediately. Any unauthorized disclosure, use or dissemination either in whole or in part is prohibited. Opinions, conclusions and other information contained in this message are personal opinions of the sender and do not necessarily represent the views of the Panasonic Group of companies." From kadin320 at gmail.com Wed Nov 2 06:18:27 2011 From: kadin320 at gmail.com (kadin 320) Date: Tue, 1 Nov 2011 22:18:27 -0700 Subject: [FFmpeg-user] How to grab a jpg thumb image? Message-ID: I am trying to grab a frame of video and convert it to a jpg thumb image. I am only seccessful at coverting it to a png thumb. I tried putting the word jpg and jpeg in the exec() and making sure the file ext. is jpg or jpeg, but I can't get it to work. What am I doing wrong? Thanks. exec("$ffmpegPath -y -i $srcFile -vframes 1 -ss 00:00:03 -an -vcodec png -f rawvideo -s 110x90 $destThumb"); From akshar_tank at yahoo.com Wed Nov 2 06:27:24 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Tue, 1 Nov 2011 22:27:24 -0700 (PDT) Subject: [FFmpeg-user] How to grab a jpg thumb image? In-Reply-To: References: Message-ID: <1320211644.66436.YahooMailNeo@web122520.mail.ne1.yahoo.com> check this link. http://blog.prashanthellina.com/2008/03/29/creating-video-thumbnails-using-ffmpeg/ regards, pranav. ________________________________ From: kadin 320 To: ffmpeg-user at ffmpeg.org Sent: Wednesday, November 2, 2011 10:48 AM Subject: [FFmpeg-user] How to grab a jpg thumb image? I am trying to grab a frame of video and convert it to a jpg thumb image. I am only seccessful at coverting it to a png thumb. I tried putting the word jpg and jpeg in the exec() and making sure the file ext. is jpg or jpeg, but I can't get it to work. What am I doing wrong? Thanks. exec("$ffmpegPath -y -i $srcFile -vframes 1 -ss 00:00:03 -an -vcodec png -f rawvideo -s 110x90 $destThumb"); _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From kadin320 at gmail.com Wed Nov 2 06:38:16 2011 From: kadin320 at gmail.com (kadin 320) Date: Tue, 1 Nov 2011 22:38:16 -0700 Subject: [FFmpeg-user] How to grab a jpg thumb image? In-Reply-To: <1320211644.66436.YahooMailNeo@web122520.mail.ne1.yahoo.com> References: <1320211644.66436.YahooMailNeo@web122520.mail.ne1.yahoo.com> Message-ID: It worked. Thank you. On Tue, Nov 1, 2011 at 10:27 PM, tank pranav wrote: > check this link. > > > http://blog.prashanthellina.com/2008/03/29/creating-video-thumbnails-using-ffmpeg/ > > > regards, > pranav. > > > ________________________________ > From: kadin 320 > To: ffmpeg-user at ffmpeg.org > Sent: Wednesday, November 2, 2011 10:48 AM > Subject: [FFmpeg-user] How to grab a jpg thumb image? > > I am trying to grab a frame of video and convert it to a jpg thumb image. > I am only seccessful at coverting it to a png thumb. > I tried putting the word jpg and jpeg in the exec() > and making sure the file ext. is jpg or jpeg, but I can't get it to work. > What am I doing wrong? Thanks. > > exec("$ffmpegPath -y -i $srcFile -vframes 1 -ss 00:00:03 -an -vcodec png -f > rawvideo -s 110x90 $destThumb"); > _______________________________________________ > 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 ashish at oliveindesign.com Wed Nov 2 06:42:19 2011 From: ashish at oliveindesign.com (ashish) Date: Wed, 2 Nov 2011 11:12:19 +0530 Subject: [FFmpeg-user] How to grab a jpg thumb image? In-Reply-To: <1320211644.66436.YahooMailNeo@web122520.mail.ne1.yahoo.com> References: <1320211644.66436.YahooMailNeo@web122520.mail.ne1.yahoo.com> Message-ID: <002b01cc9922$33401fc0$99c05f40$@com> Hello pranaw, You can try this one exec("ffmpeg -i $ename.mp4 -f image2 -vframes 1 -s 320x240 $ename.png"); this is working fine Thanks & Regards Ashish -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of tank pranav Sent: Wednesday, November 02, 2011 10:57 AM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] How to grab a jpg thumb image? check this link. http://blog.prashanthellina.com/2008/03/29/creating-video-thumbnails-using-f fmpeg/ regards, pranav. ________________________________ From: kadin 320 To: ffmpeg-user at ffmpeg.org Sent: Wednesday, November 2, 2011 10:48 AM Subject: [FFmpeg-user] How to grab a jpg thumb image? I am trying to grab a frame of video and convert it to a jpg thumb image. I am only seccessful at coverting it to a png thumb. I tried putting the word jpg and jpeg in the exec() and making sure the file ext. is jpg or jpeg, but I can't get it to work. What am I doing wrong? Thanks. exec("$ffmpegPath -y -i $srcFile -vframes 1 -ss 00:00:03 -an -vcodec png -f rawvideo -s 110x90 $destThumb"); _______________________________________________ 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 vlad.ion at gmail.com Wed Nov 2 10:26:58 2011 From: vlad.ion at gmail.com (oblivion) Date: Wed, 2 Nov 2011 02:26:58 -0700 (PDT) Subject: [FFmpeg-user] Muxing raw H264 to container format (MP4, MOV) In-Reply-To: References: <1319124086069-3922255.post@n4.nabble.com> <1319125913860-3922333.post@n4.nabble.com> <1319623314753-3940025.post@n4.nabble.com> <4EA7E4B2.6070304@mdsh.com> <1319703002850-3943546.post@n4.nabble.com> <1320158190449-3963745.post@n4.nabble.com> Message-ID: <1320226018194-3972300.post@n4.nabble.com> Hi, > ffmpeg -threads 3 -i in.h264 -vcodec mpeg4 -r 25 -sameq outfile.mp4 Thanks for the help, but as I said I don't want to recompress it, I just want to mux the original encoded video stream in a container to get seeking information (hence the "-vcodec copy"). I know it works because avc2avi does it, but it outputs AVI 1.0 and I need AVI 2.0 (or mkv or mp4). I know the video content is ok because I can play it in vlc and I can read the h264 packets in code. > I didn't need to " -f h264 " but I did need to force the codec > which, running the way you expressed it, complained about the > frame rates being incompatible (and quite rightfully ?? so because MP4 > is only 25 fps ???) The stream is at about 24fps but that is not the problem really. I can tweak the frame rate if I get the file to actually mux in the container. > FWIW, I tried your lines and it "sort of" worked. The '-f h264' was > needed. There's sound? There is no sound. The original file is just a raw h264 stream from a hardware device. I don't need to mux sound in the container at the moment anyway. > Perhaps ffprobe -show_streams your_original_infile.xxx ? The output is one stream, as expected: Input #0, h264, from 'Anonymous-3-recording.h264': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (High), yuv420p, 1280x960, 25 fps, 25 tbr, 1200k tb n, 50 tbc [STREAM] index=0 codec_name=h264 codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 codec_type=video codec_time_base=1/50 codec_tag_string=[0][0][0][0] codec_tag=0x0000 width=1280 height=960 has_b_frames=0 pix_fmt=yuv420p level=40 r_frame_rate=50/2 avg_frame_rate=25/1 time_base=1/1200000 start_time=N/A duration=N/A [/STREAM] -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Muxing-raw-H264-to-container-format-MP4-MOV-tp3922255p3972300.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From hema.angamuthu at gmail.com Wed Nov 2 11:48:34 2011 From: hema.angamuthu at gmail.com (Hema A) Date: Wed, 2 Nov 2011 16:18:34 +0530 Subject: [FFmpeg-user] Streaming video over wifi network Message-ID: Hi, I am trying to encode and stream mpeg4 video using RTP to a vlc player. When tested on X86 platform, the streaming is successful over wired network (LAN). But when ported the same application on Android and tried to stream over wifi, i get the following error in line; url_fopen(&rtpctx->pb, rtpctx->filename, URL_WRONLY); The function return -2 which seems to be ENOENT [possible call for AVERROR(ENOENT)] The inputs to url_fopen are; rtpctx = This is the rtp format context [AVFormatContext] rtpctx->filename = url address Eg: rtp://192.168.1.133:5004 Please suggest if any changes are required for url_fopen() especially over the wifi network. Thanks, Hema From archil1983 at gmail.com Wed Nov 2 12:07:26 2011 From: archil1983 at gmail.com (Archil Matchavariani) Date: Wed, 2 Nov 2011 13:07:26 +0200 Subject: [FFmpeg-user] Approx. 15 seconds delay when re-streaming h.264 to RTMP Message-ID: Hi. I'm trying to re-stream h.264 packets to rtmp server, using the following command: ffmpeg -i rtsp://root:pass at xxx.xxx.xxx.xxx:554/live?tcp -vcodec copy -f flv rtmp://xxx.xxx.xxx.xxx/live/teststream The resulting stream hangs on the first frame for some time, then resumes playback, with approximately 15 seconds of delay. If I let FFMPEG re-code the stream to FLV via following command: ffmpeg -i rtsp://root:pass at xxx.xxx.xxx.xxx:554/live?tcp -f flv rtmp://xxx.xxx.xxx.xxx/live/teststream The stream works fine and has minimal delay, but the resulting quality naturally lower. Any idea how to remove the delay on re-stream? I'm using latest 0.8 (oldapi) of FFMPEG. Thanks! From archil1983 at gmail.com Wed Nov 2 12:11:30 2011 From: archil1983 at gmail.com (Archil Matchavariani) Date: Wed, 2 Nov 2011 13:11:30 +0200 Subject: [FFmpeg-user] Fwd: Approx. 15 seconds delay when re-streaming h.264 to RTMP In-Reply-To: References: Message-ID: Forgot to mention, I have latest librtmp linked in ffmpeg as well. Regards. ---------- Forwarded message ---------- From: Archil Matchavariani Date: Wed, Nov 2, 2011 at 1:07 PM Subject: Approx. 15 seconds delay when re-streaming h.264 to RTMP To: ffmpeg-user at ffmpeg.org Hi. I'm trying to re-stream h.264 packets to rtmp server, using the following command: ffmpeg -i rtsp://root:pass at xxx.xxx.xxx.xxx:554/live?tcp -vcodec copy -f flv rtmp://xxx.xxx.xxx.xxx/live/teststream The resulting stream hangs on the first frame for some time, then resumes playback, with approximately 15 seconds of delay. If I let FFMPEG re-code the stream to FLV via following command: ffmpeg -i rtsp://root:pass at xxx.xxx.xxx.xxx:554/live?tcp -f flv rtmp://xxx.xxx.xxx.xxx/live/teststream The stream works fine and has minimal delay, but the resulting quality naturally lower. Any idea how to remove the delay on re-stream? I'm using latest 0.8 (oldapi) of FFMPEG. Thanks! From archil1983 at gmail.com Wed Nov 2 12:49:43 2011 From: archil1983 at gmail.com (Archil Matchavariani) Date: Wed, 2 Nov 2011 13:49:43 +0200 Subject: [FFmpeg-user] FFmpg the decoder doesn't detects camera resolution changes Message-ID: Good day, in other words, when I change resolution on my cam, (for example increase) I see just a part of the view , and FFmpg doesn't detect the event of resolution change From archil1983 at gmail.com Wed Nov 2 13:20:47 2011 From: archil1983 at gmail.com (Archil Matchavariani) Date: Wed, 2 Nov 2011 14:20:47 +0200 Subject: [FFmpeg-user] FFmpg the decoder doesn't detects camera resolution changes In-Reply-To: References: Message-ID: in other words (to be more correctly) ** ** On older versions of FFpmg we could detect these changes code,it is called on resolution change, and then influence the rest of process:**** libavcodec/mpeg4videodec.c:1624**** ** ** if(width && height && !(s->width && s->codec_tag == AV_RL32("MP4S"))){ ** s->width = width;**** s->height = height;**** }**** ** ** On new version it is not called.**** ** On Wed, Nov 2, 2011 at 1:49 PM, Archil Matchavariani wrote: > Good day, > > in other words, when I change resolution on my cam, (for example > increase) I see just a part of the view , and FFmpg doesn't detect the > event of resolution change > > > From drabner at zoobe.com Wed Nov 2 14:06:15 2011 From: drabner at zoobe.com (Drabner) Date: Wed, 2 Nov 2011 06:06:15 -0700 (PDT) Subject: [FFmpeg-user] Right pixel column is repeated in the left In-Reply-To: <1320237888449-3972841.post@n4.nabble.com> References: <1320237888449-3972841.post@n4.nabble.com> Message-ID: <1320239175462-3972958.post@n4.nabble.com> Sorry, I meant render engine, not video engine ;) -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-pixel-column-is-repeated-in-the-left-tp3972841p3972958.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From drabner at zoobe.com Wed Nov 2 13:44:48 2011 From: drabner at zoobe.com (Drabner) Date: Wed, 2 Nov 2011 05:44:48 -0700 (PDT) Subject: [FFmpeg-user] Right pixel column is repeated in the left Message-ID: <1320237888449-3972841.post@n4.nabble.com> Hey, I have a rather weird problem: I use ffmpeg to convert frames from a video engine to a video. Now in the engine, everything looks fine. And basically, the conversion of the frames into the video also works. But somehow, the rightmost pixel of each line is repeated in the left. This does not happen in the engine, so it must be something within the ffmpeg code. Here is the function that writes the frames: //--------------------------------------------------------- //--------------------------------------------------------- //--------------------------------------------------------- int out_size, ret; AVCodecContext *c; static struct SwsContext *img_convert_ctx = NULL; c = st->codec; /* as we only generate a YUV420P picture, we must convert it to the codec pixel format if needed */ if (ptr->img_convert_ctx == NULL) { ptr->img_convert_ctx = sws_getContext( ptr->srcWidth, ptr->srcHeight, PIX_FMT_ARGB, c->width, c->height, c->pix_fmt, SWS_BICUBIC, NULL, NULL, NULL ); if (ptr->img_convert_ctx == NULL) { av_log(c, AV_LOG_ERROR, "%s","Cannot initialize the conversion context\n"); exit(1); } } sws_scale(ptr->img_convert_ctx, ptr->tmp_picture->data, ptr->tmp_picture->linesize, 0, ptr->srcHeight, ptr->picture->data, ptr->picture->linesize); /* encode the image */ out_size = avcodec_encode_video( c, ptr->video_outbuf, ptr->video_outbuf_size, ptr->picture ); /* if zero size, it means the image was buffered */ if (out_size > 0) { AVPacket pkt; av_init_packet(&pkt); if (c->coded_frame->pts != AV_NOPTS_VALUE) pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base); if(c->coded_frame->key_frame) pkt.flags |= AV_PKT_FLAG_KEY; pkt.stream_index= st->index; pkt.data= ptr->video_outbuf; pkt.size= out_size; /* write the compressed frame in the media file */ ret = av_write_frame(oc, &pkt); } else { ret = 0; } //--------------------------------------------------------- //--------------------------------------------------------- //--------------------------------------------------------- As I said, this generally works, except that glitch. Any ideas what could cause this? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-pixel-column-is-repeated-in-the-left-tp3972841p3972841.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From Hongfeng at Mesa-Engr.com Wed Nov 2 15:45:39 2011 From: Hongfeng at Mesa-Engr.com (Hongfeng Wang) Date: Wed, 2 Nov 2011 09:45:39 -0500 Subject: [FFmpeg-user] problem using FFMPEG -ss to extract segment of h264 video Message-ID: Hi, guys: I got a 10min video is generated by DM368-DVSDK4.02's encode program. The 264 file could be played by Ubuntu movie player. If I convert the whole clip to mp4 using ffmpeg, it works without any problem. But if I try to extract a fraction of the video, it always run into "missing picture in access unit" problem, and only generates a 1 second clip, which is the end of whole video. ffmpeg -r 20.2 -ss 300 -i 20111101_153000.264 -t 10 -vcodec copy 201101153500.mp4 FFmpeg version SVN-r0.5.4-4:0.5.4-1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.4-1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --extra-cflags=-marm -fPIC -DPIC --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Sep 18 2011 15:55:02, gcc: 4.4.5 [NULL @ 0x49170]missing picture in access unit Seems stream 0 codec frame rate differs from container frame rate: 40.40 (202/5) -> 20.20 (202/10) Input #0, h264, from '20111101_153000.264': Duration: N/A, bitrate: N/A Stream #0.0: Video: h264, yuv420p, 1920x1080, 20.20 tbr, 1200k tbn, 40.40 tbc Output #0, mp4, to '201101153500.mp4': Stream #0.0: Video: 0x0000, yuv420p, 1920x1080, q=2-31, 90k tbn, 20.20 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [NULL @ 0x49170]missing picture in access unit frame= 20 fps= 0 q=-1.0 Lsize= 280kB time=0.99 bitrate=2313.3kbits/s video:279kB audio:0kB global headers:0kB muxing overhead 0.314991% the FFMPEG I used is FFmpeg version SVN-r0.5.4-4:0.5.4-1 from Debain ARMEL packages. I also tried FFMPEG0.7.2-4 on my Ubuntu machine, which has the same result. I tried different values for -ss option, from 5 to 300, all the same. The video has an IDR frame for every 20 frames. Thanks! Hongfeng Wang From lou at lrcd.com Wed Nov 2 19:57:17 2011 From: lou at lrcd.com (Lou) Date: Wed, 2 Nov 2011 10:57:17 -0800 Subject: [FFmpeg-user] problem using FFMPEG -ss to extract segment of h264 video In-Reply-To: References: Message-ID: <20111102105717.5b03ddff@lrcd.com> On Wed, 2 Nov 2011 09:45:39 -0500 Hongfeng Wang wrote: > Hi, guys: > > I got a 10min video is generated by DM368-DVSDK4.02's encode program. > The 264 file could be played by Ubuntu movie player. > > If I convert the whole clip to mp4 using ffmpeg, it works without any > problem. But if I try to extract a fraction of the video, it always > run into "missing picture in access unit" problem, and only generates > a 1 second clip, which is the end of whole video. > > ffmpeg -r 20.2 -ss 300 -i 20111101_153000.264 -t 10 -vcodec copy > 201101153500.mp4 > > FFmpeg version SVN-r0.5.4-4:0.5.4-1, Copyright (c) 2000-2009 Fabrice > Bellard, et al. configuration: --extra-version=4:0.5.4-1 > --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau > --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib > --disable-stripping --disable-vhook --enable-runtime-cpudetect > --extra-cflags=-marm -fPIC -DPIC --enable-gpl --enable-postproc > --enable-swscale --enable-x11grab --enable-libfaad --enable-libdc1394 > --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 > libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 > libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 > libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 > built on Sep 18 2011 15:55:02, gcc: 4.4.5 [NULL @ 0x49170]missing > picture in access unit > > Seems stream 0 codec frame rate differs from container frame rate: > 40.40 (202/5) -> 20.20 (202/10) Input #0, h264, from > '20111101_153000.264': Duration: N/A, bitrate: N/A > Stream #0.0: Video: h264, yuv420p, 1920x1080, 20.20 tbr, 1200k > tbn, 40.40 tbc Output #0, mp4, to '201101153500.mp4': > Stream #0.0: Video: 0x0000, yuv420p, 1920x1080, q=2-31, 90k tbn, > 20.20 tbc Stream mapping: > Stream #0.0 -> #0.0 > Press [q] to stop encoding > [NULL @ 0x49170]missing picture in access unit > frame= 20 fps= 0 q=-1.0 Lsize= 280kB time=0.99 > bitrate=2313.3kbits/s video:279kB audio:0kB global headers:0kB muxing > overhead 0.314991% > > > the FFMPEG I used is FFmpeg version SVN-r0.5.4-4:0.5.4-1 from Debain > ARMEL packages. I also tried FFMPEG0.7.2-4 on my Ubuntu machine, > which has the same result. I tried different values for -ss option, > from 5 to 300, all the same. The video has an IDR frame for every 20 > frames. > > Thanks! > > Hongfeng Wang Try moving -ss after -i 20111101_153000.264 so it is applied as an output option. This changes the behavior of -ss: * -ss after -i will decode until your -ss value. This method is slower but is usually more accurate. * -ss before -i will seek before decoding, but will potentially not be frame-accurate depending on your input formats(s). Also, FFmpeg release 0.5 is quite old and is no longer supported by FFmpeg. From cehoyos at ag.or.at Thu Nov 3 00:02:12 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Nov 2011 23:02:12 +0000 (UTC) Subject: [FFmpeg-user] problem using FFMPEG -ss to extract segment of h264 video References: Message-ID: Hongfeng Wang Mesa-Engr.com> writes: > ffmpeg -r 20.2 -ss 300 -i 20111101_153000.264 -t 10 -vcodec copy 0.mp4 > > FFmpeg version SVN-r0.5.4-4:0.5.4-1, Copyright (c) 2000-2009 Please update to current git head (it contains relatively new fixes that apply to your use case), your version is ancient! Carl Eugen From cehoyos at ag.or.at Thu Nov 3 00:03:08 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 2 Nov 2011 23:03:08 +0000 (UTC) Subject: [FFmpeg-user] FFmpg the decoder doesn't detects camera resolution changes References: Message-ID: Archil Matchavariani gmail.com> writes: > in other words, when I change resolution on my cam, (for example increase) > I see just a part of the view , and FFmpg doesn't detect the event of > resolution change (Command line and complete, uncut output missing.) Please provide a failing sample. Carl Eugen From lamia at jeack.com.au Thu Nov 3 01:31:27 2011 From: lamia at jeack.com.au (Bazza) Date: Thu, 03 Nov 2011 11:31:27 +1100 Subject: [FFmpeg-user] Muxing raw H264 to container format (MP4, MOV) References: <1319124086069-3922255.post@n4.nabble.com> <1319125913860-3922333.post@n4.nabble.com> <1319623314753-3940025.post@n4.nabble.com> <4EA7E4B2.6070304@mdsh.com> <1319703002850-3943546.post@n4.nabble.com> <1320158190449-3963745.post@n4.nabble.com> <1320226018194-3972300.post@n4.nabble.com> Message-ID: On Wed, 2 Nov 2011 02:26:58 -0700 (PDT), oblivion wrote: OK From cehoyos at ag.or.at Thu Nov 3 20:23:40 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Nov 2011 19:23:40 +0000 (UTC) Subject: [FFmpeg-user] Issues splitting a 3gp file using -ss and -t options References: <4EAA652B.1000703@cinemacraft.tv> Message-ID: Deepika cinemacraft.tv> writes: > I am trying to split a 3gp file using the command below and I get the > following errors. Any help in solving this issue is highly appreciated. > *[3gp @ 0x2fc67e0] track 1: codec frame size is not set* This specific error should be fixed, but it is possible that you see other problems with your command line... Carl Eugen From alex.zhen.ma at gmail.com Thu Nov 3 22:09:08 2011 From: alex.zhen.ma at gmail.com (Zhen Ma) Date: Thu, 3 Nov 2011 17:09:08 -0400 Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? Message-ID: Hey guys, Is it possible to run one command to output multiple files? for example , I would like to covert one video into 3 videos format ,each video has different video codec, different bitrate and different resolution. Thanks! Alex Zhen Ma From hardik.sharma22 at yahoo.com Thu Nov 3 22:38:20 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Thu, 3 Nov 2011 14:38:20 -0700 (PDT) Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? In-Reply-To: References: Message-ID: <1320356300.25759.YahooMailNeo@web46212.mail.sp1.yahoo.com> As per I know in ffmpeg you can convert different input files into different output file formats for example-? ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg, never tried with different bitrates but I guess you can try that. ? ________________________________ From: Zhen Ma To: ffmpeg-user at ffmpeg.org Sent: Thursday, 3 November 2011 2:09 PM Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? Hey guys, Is it possible to run one command? to output multiple files?? for example , I would like to covert one video into 3 videos format ,each video has different video codec, different bitrate and different resolution. Thanks! Alex Zhen Ma _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From krueger at lesspain.de Thu Nov 3 23:21:34 2011 From: krueger at lesspain.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Thu, 3 Nov 2011 23:21:34 +0100 Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? In-Reply-To: References: Message-ID: On Nov 3, 2011, at 22:09 , Zhen Ma wrote: > Hey guys, > > Is it possible to run one command to output multiple files? for example , > I would like to covert one video into 3 videos format ,each video has > different video codec, different bitrate and different resolution. yes, just specify the output files one after the other, always keeping the arguments before the next output file, i.e. ffmpeg -i ..... e.g. ffmpeg -i input.mpg -vcodec mjpeg -acodec copy out1.mov -vcodec mpeg2video -acodec mp2 out2.avi ....... HTH From cehoyos at ag.or.at Thu Nov 3 23:45:33 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 3 Nov 2011 22:45:33 +0000 (UTC) Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? References: Message-ID: Zhen Ma gmail.com> writes: > Is it possible to run one command to output multiple files? for example , > I would like to covert one video into 3 videos format ,each video has > different video codec, different bitrate yes > and different resolution. but I believe this does not (always?) work as expected. Carl Eugen From deepika at cinemacraft.tv Fri Nov 4 09:17:55 2011 From: deepika at cinemacraft.tv (Deepika) Date: Fri, 04 Nov 2011 16:17:55 +0800 Subject: [FFmpeg-user] Issues splitting a 3gp file using -ss and -t options In-Reply-To: References: <4EAA652B.1000703@cinemacraft.tv> Message-ID: <4EB39FB3.2080404@cinemacraft.tv> Hi Carl, On 11/04/2011 03:23 AM, Carl Eugen Hoyos wrote: > Deepika cinemacraft.tv> writes: > >> I am trying to split a 3gp file using the command below and I get the >> following errors. Any help in solving this issue is highly appreciated. >> *[3gp @ 0x2fc67e0] track 1: codec frame size is not set* > This specific error should be fixed, but it is possible that you see other > problems with your command line... I updated to the latest version and it works now. Thanks. What do you mean by " but it is possible that you see other problems with your command line... " I see this message [mov,mp4,m4a,3gp,3g2,mj2 @ 0x26377c0] multiple edit list entries, a/v desync might occur, patch welcome Last message repeated 1 times But I do not see any av desync Regards, Deepika > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From arunkumar at omeonsolutions.com Fri Nov 4 16:04:51 2011 From: arunkumar at omeonsolutions.com (Arunkumar) Date: Fri, 4 Nov 2011 20:34:51 +0530 Subject: [FFmpeg-user] Hardware accelerated H.264 Decoding in Android with ffmpeg Message-ID: <000001cc9b03$1b970130$52c50390$@com> Hi, Could somebody give us pointers regarding Hardware accelerated H.264 decoding in Android using ffmpeg as with regards to compile script and the wrapper class? Thank you, B.Arunkumar From alex.zhen.ma at gmail.com Fri Nov 4 17:47:08 2011 From: alex.zhen.ma at gmail.com (Zhen Ma) Date: Fri, 4 Nov 2011 12:47:08 -0400 Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? In-Reply-To: References: Message-ID: Thanks Robert ! It works like your mentioned. But It turns out another problem for me . I am trying to do the 2pass encoding. So the first multiple line encoding command would be like that: ffmpeg -y -i youtube_480.flv -pass 1 -stats 1 -f mp4 -s 864x480 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 600k -maxrate 600k -fpre presets/main.ffpreset -an /dev/null -pass 1 -stats 2 -f mp4 -s 432x240 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 300k -maxrate 300k -fpre presets/baseline.ffpreset -an /dev/null -pass 1 -stats 3 -f mp4 -s 320x180 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 200k -maxrate 200k -fpre presets/baseline.ffpreset -an /dev/null But the problem is the x264_2pass.log which was generated with first pass of first sub-command was overwirtten by the second and third sub-command. I was trying to find the way to specify the name of log file. I can only see x264 has parameter stats which allows us to change the name of log file, but by looking at the ffmpeg ,it seems it doesn't implement this parameter in ffmpeg command line. I don't know if you know something about it , how can I change the name of pass log file with ffmpeg command line? Thank you very much! 2011/11/3 Robert Kr?ger > > On Nov 3, 2011, at 22:09 , Zhen Ma wrote: > > > Hey guys, > > > > Is it possible to run one command to output multiple files? for > example , > > I would like to covert one video into 3 videos format ,each video has > > different video codec, different bitrate and different resolution. > > yes, just specify the output files one after the other, always keeping the > arguments before the next output file, i.e. > > ffmpeg -i > ..... > > e.g. > > ffmpeg -i input.mpg -vcodec mjpeg -acodec copy out1.mov -vcodec mpeg2video > -acodec mp2 out2.avi ....... > > > HTH > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Your sincerely ! ------------------------------------------------------- Alex Zhen Ma ? Email: alex.zhen.ma at gmail.com ? Messenger: coolface8 at hotmail.com From alex.zhen.ma at gmail.com Fri Nov 4 19:16:00 2011 From: alex.zhen.ma at gmail.com (Zhen Ma) Date: Fri, 4 Nov 2011 14:16:00 -0400 Subject: [FFmpeg-user] How to change the name of x264_2pass.log file when you are doing 2pass encoding? Message-ID: Guys, I am trying to do the 2pass encoding. So the first multiple line encoding command would be like that: ffmpeg -y -i youtube_480.flv -pass 1 -f mp4 -s 864x480 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 600k -maxrate 600k -fpre presets/main.ffpreset -an /dev/null -pass 1 -stats 2 -f mp4 -s 432x240 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 300k -maxrate 300k -fpre presets/baseline.ffpreset -an /dev/null -pass 1 -stats 3 -f mp4 -s 320x180 -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 200k -maxrate 200k -fpre presets/baseline.ffpreset -an /dev/null But the problem is the x264_2pass.log which was generated with first pass of first sub-command was overwirtten by the second and third sub-command. I was trying to find the way to specify the name of log file. I can only see x264 has parameter "stats" which allows us to change the name of log file, but by looking at the ffmpeg ,it seems it doesn't implement this parameter in ffmpeg command line. I don't know if you guys know something about it , how can I change the name of pass log file with ffmpeg command line? Thank you very much! -- Your sincerely ! ------------------------------------------------------- Alex Zhen Ma ? Email: alex.zhen.ma at gmail.com ? Messenger: coolface8 at hotmail.com From hardik.sharma22 at yahoo.com Fri Nov 4 22:15:42 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Fri, 4 Nov 2011 14:15:42 -0700 (PDT) Subject: [FFmpeg-user] Is there any frame copy concealment available? Message-ID: <1320441342.27468.YahooMailNeo@web46202.mail.sp1.yahoo.com> Is FFMPEG provides any frame copy concealment by any chance? It's not working for me. If someone has any idea regarding this please let me know. Thanks. ? ? From cehoyos at ag.or.at Sat Nov 5 01:18:04 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 5 Nov 2011 00:18:04 +0000 (UTC) Subject: [FFmpeg-user] Issues splitting a 3gp file using -ss and -t options References: <4EAA652B.1000703@cinemacraft.tv> <4EB39FB3.2080404@cinemacraft.tv> Message-ID: Deepika cinemacraft.tv> writes: > On 11/04/2011 03:23 AM, Carl Eugen Hoyos wrote: > > Deepika cinemacraft.tv> writes: > > > >> I am trying to split a 3gp file using the command below and I get the > >> following errors. Any help in solving this issue is highly appreciated. > >> *[3gp @ 0x2fc67e0] track 1: codec frame size is not set* > > This specific error should be fixed, but it is possible that you see other > > problems with your command line... > I updated to the latest version and it works now. Thanks. Great. (With other files, I could see different problems that are less easy to solve.) Carl Eugen From raju2803 at yahoo.com Fri Nov 4 22:46:12 2011 From: raju2803 at yahoo.com (Raju Nanduri) Date: Fri, 4 Nov 2011 14:46:12 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg build - ERROR: libfaac not found Message-ID: <1320443172.61208.YahooMailNeo@web30307.mail.mud.yahoo.com> Hi, I got the latest source file on my ubuntu 10.04 ( lucid) and tried running ./configure and got the following error; ERROR: libfaac not found Please see config.log as attached. Any help will be greatly appreciated.? Pl note that running the common triple ofconfigure (configure,? make and make install) with out any command line options or arugemts ran just fine. Here is more detailed description of commands I ran and the error. I have followed instructions from? the? URL in ubuntu forums? (http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 ) to? get and build ffmpeg? and here are the 3 steps I did: 1.sudo? git clone git://git.videolan.org/ffmpeg 2. cd ffmpeg 3. sudo ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ ??? --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb \ ??? --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab nanduri at rnanduri-ubuntu:/usr/local/opal/plugins/video/ffmpeg$ sudo ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ >???? --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb \ >???? --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab ERROR: libfaac 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. Thanks -Raju -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 124103 bytes Desc: not available URL: From kurtn at 126.com Sat Nov 5 06:52:40 2011 From: kurtn at 126.com (Ethan Wang) Date: Sat, 5 Nov 2011 13:52:40 +0800 (CST) Subject: [FFmpeg-user] ffmpeg build - ERROR: libfaac not found In-Reply-To: <1320443172.61208.YahooMailNeo@web30307.mail.mud.yahoo.com> References: <1320443172.61208.YahooMailNeo@web30307.mail.mud.yahoo.com> Message-ID: <3aae38a5.339c.13372493549.Coremail.kurtn@126.com> Try to install libfaac or just remove '--enable-libfaac' from your configure list. At 2011-11-05 05:46:12,"Raju Nanduri" wrote: >Hi, > >I got the latest source file on my ubuntu 10.04 ( lucid) and tried running ./configure and got the following error; >ERROR: libfaac not found > >Please see config.log as attached. Any help will be greatly appreciated.? Pl note that running the common triple ofconfigure >(configure,? make and make install) with out any command line options or arugemts >ran just fine. Here is more detailed description of commands I ran and the error. > >I have followed instructions from? the? URL in ubuntu forums? (http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 ) to? get and build ffmpeg? and here are the 3 steps I did: > >1.sudo? git clone git://git.videolan.org/ffmpeg >2. cd ffmpeg >3. sudo ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ >??? --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb \ >??? --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab > > >nanduri at rnanduri-ubuntu:/usr/local/opal/plugins/video/ffmpeg$ sudo ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ >>???? --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb \ >>???? --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab >ERROR: libfaac 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. > > >Thanks >-Raju From nomiya at galaxy.dti.ne.jp Sat Nov 5 07:30:43 2011 From: nomiya at galaxy.dti.ne.jp (Masaru Nomiya) Date: Sat, 05 Nov 2011 15:30:43 +0900 Subject: [FFmpeg-user] ffmpeg build - ERROR: libfaac not found In-Reply-To: <1320443172.61208.YahooMailNeo@web30307.mail.mud.yahoo.com> References: <1320443172.61208.YahooMailNeo@web30307.mail.mud.yahoo.com> Message-ID: <87lirvytbg.wl%nomiya@galaxy.dti.ne.jp> Hello, In the Message; Subject : [FFmpeg-user] ffmpeg build - ERROR: libfaac not found Message-ID : <1320443172.61208.YahooMailNeo at web30307.mail.mud.yahoo.com> Date & Time: Fri, 4 Nov 2011 14:46:12 -0700 (PDT) [Raju] == Raju Nanduri has written: Raju> I got the latest source file on my ubuntu 10.04 ( lucid) and Raju> tried running ./configure and got the following error; Raju> ERROR: libfaac not found [...] Raju> [2 config.log] Have you installed faac? --- ????? Masaru Nomiya mail-to: nomiya @ galaxy.dti.ne.jp ???? ???? "Bill! You married with Computers. Not with Me!" "No..., with money." From gabrielsimoes at rocketmail.com Sat Nov 5 18:23:48 2011 From: gabrielsimoes at rocketmail.com (=?iso-8859-1?Q?Gabriel_Sim=F5es?=) Date: Sat, 5 Nov 2011 10:23:48 -0700 (PDT) Subject: [FFmpeg-user] FFMPEG Seeking brings audio artifacts Message-ID: <1320513828.84197.YahooMailNeo@web36706.mail.mud.yahoo.com> Hello! I?m implementing a audio decoder using ffmpeg. While reading audio and even seeking already works, I can?t figure out a way to clear the buffers after seeking so I have no artifacts when the app starts reading audio right after seeking. avcodec_flush_buffers doesn?t seem to have any effect on the internal buffers. This issue happens with all decoders (mp3, aac, wma, ...) but PCM/WAV (which doesn?t use internal buffers to hold data to decode since the audio is not compressed). The code snippet is simple: ? ? av_seek_frame(audioFilePack->avContext, audioFilePack->stream, posInTimeFrame, AVSEEK_FLAG_ANY); ? ? avcodec_flush_buffers(audioFilePack->avContext->streams[audioFilePack->stream]->codec); Explaining: ? ? audioFilePack->avContext = FormatContext ? ? audioFilePack->stream = Stream Position (also used to read audio packets) ? ? audioFilePack->avContext->streams[audioFilePack->stream]->codec = CodecContext for the codec used Any ideas on what I should do so I can seek and get no residual audio? Question was also submitted to StackOverflow but I have got no answers until now: http://stackoverflow.com/questions/7989623/ffmpeg-seeking-brings-audio-artifacts Thanks! From ubitux at gmail.com Sun Nov 6 00:47:59 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Sun, 6 Nov 2011 00:47:59 +0100 Subject: [FFmpeg-user] 8 channels audio to 8 x 1 channel audio In-Reply-To: <4EA582DC.9040902@mdsh.com> References: <4EA582DC.9040902@mdsh.com> Message-ID: <20111105234759.GB6426@leki> On Mon, Oct 24, 2011 at 04:23:08PM +0100, Mark Himsley wrote: > Hi Igor, > > On 24/10/11 16:01, Igor Podobi?ski wrote: > >Hello, > >This is output of ffprobe:-----------------------------------------------------------------------------------------------------------Input #0, mxf, from 'd:\Media\PFR\test.mxf': Duration: 00:00:19.56, start: 0.000000, bitrate: 62570 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p, 720x608 [SAR 608:405DAR 16:9], 50000 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, 8 channels, s16, 6144 kb/s------------------------------------------------------------------------------------------------------------I am trying to demux this mxf file into:- 1 video file (m2v)- 8 mono audio files (wav) or 4 stereo files > >Is it even possible with ffmpeg? > >Thanks,Igor > > I believe that is currently only possible with ffmbc and it's > -map_audio_channel command, but patches are appearing on the > ffmpeg's development email list that suggest it'll be possible in > ffmpeg soon. > It's now possible with FFmpeg git/master through the -map_channel option (see http://ffmpeg.org/ffmpeg.html#Advanced-options for more information). Testing and suggestions are welcome. -- 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 dev at rarevision.com Sun Nov 6 01:53:23 2011 From: dev at rarevision.com (Thomas Worth) Date: Sat, 5 Nov 2011 17:53:23 -0700 Subject: [FFmpeg-user] Is there a CodecID for stream copy? What if codec support isn't compiled in? Message-ID: I want to take a file and do a stream copy, that is bypass decoding and encoding. This may need to be done without the actual codec compiled into avcodec. Since no decoding or encoding takes place, what would I specify as the CodecID? Since avformat_write_header() requires an AVCodec argument, what would the codec be if it's not compiled in? Can I just set the AVCodec fields manually? From master at io.ua Sun Nov 6 18:24:57 2011 From: master at io.ua (Anderw Gora) Date: Sun, 6 Nov 2011 19:24:57 +0200 Subject: [FFmpeg-user] Error when creating jpeg from video frame in ffmpeg 0.8.5 Message-ID: <14913891031.20111106192457@io.ua> ffmpeg 0.8.5 cant create jpeg from SOME videos (2-3% of total). Just black frames. All other videos are OK Also, ffmpeg 0.5 is OK too with all videos. There are two erors|warnings while jpeg creating: 1.Estimating duration from bitrate, this may be inaccurate 2.Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' Seems, we are not alone with this issue: http://stackoverflow.com/questions/7677398/ffmpeg-generating-screenshots-drop-frames Would you please clarify the situation? Thanks a lot! OK: /usr/local/bin/ffmpeg -ss 6 -i video.flv -vframes 1 -an -s 720x400 -f image2 -vcodec mjpeg -y pic.jpg FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-shared --enable-pthreads --enable-libamr-nb --enable-libamr-wb --enable-nonfree --enable-gpl --enable-libmp3lame --enable-libxvid --enable-libfaad --enable-libfaac --enable-libvorbis --enable-libx264 --enable-avfilter --enable-postproc libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libpostproc 51. 2. 0 / 51. 2. 0 built on May 18 2009 14:07:19, gcc: 4.1.2 20071124 (Red Hat 4.1.2-42) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1) Input #0, flv, from 'video.flv': Duration: 00:28:06.88, start: 0.040000, bitrate: 384 kb/s Stream #0.0: Video: h264, yuv420p, 720x400 [PAR 1:1 DAR 9:5], 320 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: mp3, 44100 Hz, mono, s16, 64 kb/s Output #0, image2, to 'pic.jpg': Stream #0.0: Video: mjpeg, yuvj420p, 720x400 [PAR 1:1 DAR 9:5], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 1 fps= 1 q=5.0 Lsize= -0kB time=0.04 bitrate= -4.4kbits/s video:23kB audio:0kB global headers:0kB muxing overhead -100.094971% ERROR: /usr/local/ffmpeg-0.8.5/bin/ffmpeg -ss 6 -i video.flv -vframes 1 -an -s 720x400 -f image2 -vcodec mjpeg -y pic.jpg ffmpeg version 0.8.5, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 17 2011 15:54:16 with gcc 4.4.4 20100726 (Red Hat 4.4.4-13) configuration: --prefix=/usr/local/ffmpeg-0.8.5 --enable-shared --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-gpl --enable-libmp3lame --enable-libxvid --enable-libvorbis --enable-libfaac --enable-libx264 --enable-avfilter --enable-postproc --enable-version3 libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [flv @ 0x152c400] Estimating duration from bitrate, this may be inaccurate Input #0, flv, from 'video.flv': Metadata: audiodatarate : 63 audiosize : 14270641 audiocodecid : 2 audiosamplerate : 3 height : 400 hasMetadata : true duration : 1687 canSeekToEnd : true stereo : false videosize : 68095165 hasVideo : true hasCuePoints : false metadatacreator : Yet Another Metadata Injector for FLV - Version 1.8 hasAudio : true hasKeyframes : true width : 720 lastkeyframetimestamp: 1687 lasttimestamp : 1687 filesize : 82810085 videocodecid : 7 audiosamplesize : 1 totalframes : 42175 videodatarate : 313 framerate : 25 datasize : 82792822 lastkeyframelocation: 82810065 Duration: 00:28:06.88, start: 0.040000, bitrate: 384 kb/s Stream #0.0: Video: h264 (High), yuv420p, 720x400 [PAR 1:1 DAR 9:5], 320 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: mp3, 44100 Hz, mono, s16, 64 kb/s Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x152c320] w:720 h:400 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [buffersink @ 0x1531120] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x153e760] w:720 h:400 fmt:yuv420p -> w:720 h:400 fmt:yuvj420p flags:0x4 Output #0, image2, to 'pic.jpg': Metadata: audiodatarate : 63 audiosize : 14270641 audiocodecid : 2 audiosamplerate : 3 height : 400 hasMetadata : true duration : 1687 canSeekToEnd : true stereo : false videosize : 68095165 hasVideo : true hasCuePoints : false metadatacreator : Yet Another Metadata Injector for FLV - Version 1.8 hasAudio : true hasKeyframes : true width : 720 lastkeyframetimestamp: 1687 lasttimestamp : 1687 filesize : 82810085 videocodecid : 7 audiosamplesize : 1 totalframes : 42175 videodatarate : 313 framerate : 25 datasize : 82792822 lastkeyframelocation: 82810065 encoder : Lavf53.4.0 Stream #0.0: Video: mjpeg, yuvj420p, 720x400 [PAR 1:1 DAR 9:5], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 Lsize= -0kB time=00:00:00.00 bitrate= 0.0kbits/s video:0kB audio:0kB global headers:0kB muxing overhead -inf% From cehoyos at ag.or.at Sun Nov 6 22:55:22 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 6 Nov 2011 21:55:22 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Error_when_creating_jpeg_from_video_frame?= =?utf-8?q?_in_ffmpeg=090=2E8=2E5?= References: <14913891031.20111106192457@io.ua> Message-ID: Anderw Gora io.ua> writes: > ffmpeg 0.8.5 cant create jpeg from SOME videos (2-3% of total). Please upload a sample to http://www.datafilehost.com/ and post the download link. Carl Eugen From master at io.ua Mon Nov 7 00:15:54 2011 From: master at io.ua (Anderw Gora) Date: Mon, 7 Nov 2011 01:15:54 +0200 Subject: [FFmpeg-user] Error when creating jpeg from video frame in ffmpeg 0.8.5 In-Reply-To: References: <14913891031.20111106192457@io.ua> Message-ID: <14434948703.20111107011554@io.ua> >> ffmpeg 0.8.5 cant create jpeg from SOME videos (2-3% of total). CEH> Please upload a sample to http://www.datafilehost.com/ and post the download link. CEH> Carl Eugen Here it is: http://www.datafilehost.com/download-de5f253a.html Andrew Gora From amit.dor.shifer at gmail.com Mon Nov 7 03:07:33 2011 From: amit.dor.shifer at gmail.com (Amit Dor-Shifer) Date: Mon, 7 Nov 2011 13:07:33 +1100 Subject: [FFmpeg-user] decoding raw RTP? In-Reply-To: References: Message-ID: On Wed, Nov 2, 2011 at 10:33 AM, Amit Dor-Shifer wrote: > Attached is a sample of mulaw-encoded audio, packeted as RTP. Is there a > way for ffmpeg to decode this? > > $ ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample > /tmp/out.wav > ffmpeg version N-34304-gc0dbab9, Copyright (c) 2000-2011 the FFmpeg > developers > built on Oct 31 2011 17:54:58 with gcc 4.4.5 > configuration: --disable-vaapi --enable-libtheora --enable-libvorbis > --enable-nonfree --enable-postproc --enable-version3 --enable-libspeex > libavutil 51. 22. 0 / 51. 22. 0 > libavcodec 53. 26. 0 / 53. 26. 0 > libavformat 53. 18. 0 / 53. 18. 0 > libavdevice 53. 4. 0 / 53. 4. 0 > libavfilter 2. 45. 2 / 2. 45. 2 > libswscale 2. 1. 0 / 2. 1. 0 > [rtp @ 0x1a2c8c0] Guessing on RTP content - if not received properly you > need an SDP file describing it > [rtp @ 0x1a2c8c0] Estimating duration from bitrate, this may be inaccurate > Input #0, rtp, from 'testing-123.mulaw.rtp.sample': > Duration: N/A, bitrate: N/A > Stream #0:0: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s > Output #0, wav, to '/tmp/out.wav': > Metadata: > encoder : Lavf53.18.0 > Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1 > channels, s16, 128 kb/s > Stream mapping: > Stream #0.0 -> #0.0 (pcm_mulaw -> pcm_s16le) > Press [q] to stop, [?] for help > size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s > video:0kB audio:0kB global headers:0kB muxing overhead inf% > > ffmpeg -f rtp -acodec pcm_mulaw -i testing-123.mulaw.rtp.sample > re-posting. Anyone? From krueger at lesspain.de Mon Nov 7 11:06:27 2011 From: krueger at lesspain.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Mon, 7 Nov 2011 11:06:27 +0100 Subject: [FFmpeg-user] Can FFMPEG output multiple files with one command runs? In-Reply-To: References: Message-ID: <0E43D865-1DA7-422F-B4C5-B760656BC4E7@lesspain.de> Hi, On Nov 4, 2011, at 17:47 , Zhen Ma wrote: > Thanks Robert ! It works like your mentioned. But It turns out another > problem for me . I am trying to do the 2pass encoding. So the first > multiple line encoding command would be like that: > > ffmpeg -y -i youtube_480.flv -pass 1 -stats 1 -f mp4 -s 864x480 -aspect 1.8 > -r 25 -threads 0 -vcodec libx264 -b 600k -maxrate 600k -fpre > presets/main.ffpreset -an /dev/null -pass 1 -stats 2 -f mp4 -s 432x240 > -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 300k -maxrate 300k -fpre > presets/baseline.ffpreset -an /dev/null -pass 1 -stats 3 -f mp4 -s 320x180 > -aspect 1.8 -r 25 -threads 0 -vcodec libx264 -b 200k -maxrate 200k -fpre > presets/baseline.ffpreset -an /dev/null > > But the problem is the x264_2pass.log which was generated with first pass > of first sub-command was overwirtten by the second and third sub-command. > I was trying to find the way to specify the name of log file. I can only > see x264 has parameter stats which allows us to change the name of log > file, but by looking at the ffmpeg ,it seems it doesn't implement this > parameter in ffmpeg command line. > > I don't know if you know something about it , how can I change the name of > pass log file with ffmpeg command line? > > > Thank you very much! > > > 2011/11/3 Robert Kr?ger > >> >> On Nov 3, 2011, at 22:09 , Zhen Ma wrote: >> >>> Hey guys, >>> >>> Is it possible to run one command to output multiple files? for >> example , >>> I would like to covert one video into 3 videos format ,each video has >>> different video codec, different bitrate and different resolution. >> >> yes, just specify the output files one after the other, always keeping the >> arguments before the next output file, i.e. >> >> ffmpeg -i >> ..... >> >> e.g. >> >> ffmpeg -i input.mpg -vcodec mjpeg -acodec copy out1.mov -vcodec mpeg2video >> -acodec mp2 out2.avi ....... >> sorry but I don't know the answer to that question but I would probably look at the argument -passlogfile documented at http://ffmpeg.org/ffmpeg.html and if anything does not behave as documented there, file a bug report as described here: http://ffmpeg.org/bugreports.html. And please do not top post (http://en.wikipedia.org/wiki/Posting_style#Top-posting). It's more or less a rule on this list not to. HTH, Robert From gabrielsimoes at rocketmail.com Mon Nov 7 12:51:19 2011 From: gabrielsimoes at rocketmail.com (=?iso-8859-1?Q?Gabriel_Sim=F5es?=) Date: Mon, 7 Nov 2011 03:51:19 -0800 (PST) Subject: [FFmpeg-user] FFMPEG Seeking brings audio artifacts In-Reply-To: <1320513828.84197.YahooMailNeo@web36706.mail.mud.yahoo.com> References: <1320513828.84197.YahooMailNeo@web36706.mail.mud.yahoo.com> Message-ID: <1320666679.28424.YahooMailNeo@web36707.mail.mud.yahoo.com> Hello, No ideas on this one? I?m trying to figure it out for more than 2 weeks now and can?t even imagine what is wrong. At least any material I should go look for? Thanks! ________________________________ From: Gabriel Sim?es To: "ffmpeg-user at ffmpeg.org" Sent: Saturday, November 5, 2011 3:23 PM Subject: FFMPEG Seeking brings audio artifacts Hello! I?m implementing a audio decoder using ffmpeg. While reading audio and even seeking already works, I can?t figure out a way to clear the buffers after seeking so I have no artifacts when the app starts reading audio right after seeking. avcodec_flush_buffers doesn?t seem to have any effect on the internal buffers. This issue happens with all decoders (mp3, aac, wma, ...) but PCM/WAV (which doesn?t use internal buffers to hold data to decode since the audio is not compressed). The code snippet is simple: ? ? av_seek_frame(audioFilePack->avContext, audioFilePack->stream, posInTimeFrame, AVSEEK_FLAG_ANY); ? ? avcodec_flush_buffers(audioFilePack->avContext->streams[audioFilePack->stream]->codec); Explaining: ? ? audioFilePack->avContext = FormatContext ? ? audioFilePack->stream = Stream Position (also used to read audio packets) ? ? audioFilePack->avContext->streams[audioFilePack->stream]->codec = CodecContext for the codec used Any ideas on what I should do so I can seek and get no residual audio? Question was also submitted to StackOverflow but I have got no answers until now: http://stackoverflow.com/questions/7989623/ffmpeg-seeking-brings-audio-artifacts Thanks! From rogerdpack2 at gmail.com Mon Nov 7 12:59:49 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 7 Nov 2011 04:59:49 -0700 Subject: [FFmpeg-user] Thanks! In-Reply-To: References: Message-ID: > Just wanted to thank the ffmpeg team. > I went to a conference the other day and one of the presenters showed > us how to use ffmpeg in windows to remux videos. ?Then I realized just > how wide spread and useful it is. Not to mention it being the foundation for popular things like VLC, which gets like 6M downloads a week. http://sourceforge.net/top/ -r From fox at orbitalfox.com Mon Nov 7 15:15:32 2011 From: fox at orbitalfox.com (fox) Date: Mon, 7 Nov 2011 14:15:32 +0000 Subject: [FFmpeg-user] H.264 decoder VUI overread Message-ID: <20111107141532.4e500a6a@apollon> Im dealing with what I believe is a malformed TS stream. I am getting "Overread VUI by x bits" error. What are the consequences to the decoding process when the VUI has been over-read? Does it recover? Is data permanently dropped/lost? Fox From Olivier.Camus at ate-group.com Mon Nov 7 15:30:40 2011 From: Olivier.Camus at ate-group.com (Olivier Camus) Date: Mon, 7 Nov 2011 14:30:40 +0000 Subject: [FFmpeg-user] video 2160p50 Message-ID: Hi, I cannot read a large video (format 3840*2160) using ffplay, neither on Linux nor on Windows. The frame rate is not correct: to slow. I'm using o computer with 4Go of RAM, and the video is located on a SSD disk. I'm wondering if it could be a problem of disk access. Did some of you use to manage large video format? Regards, Olivier. From ravi.kumar at zenverge.com Mon Nov 7 02:04:48 2011 From: ravi.kumar at zenverge.com (ravi.kumar) Date: Sun, 6 Nov 2011 17:04:48 -0800 (PST) Subject: [FFmpeg-user] Problem compiling ffmpeg - compilation gets stuck at dsputil.c Message-ID: Hi, I am trying to compile ffmpeg for a Marvell platform and I am facing an issue. Here are the details, Platform - Linux debian 2.6.31.8 running on armv5tel Compiler - gcc (Debian 4.4.5-8) 4.4.5 The problem I am facing is, after configure, when I run make, it compiles few files but gets stuck while compiling dsputil.c (CC libavcodec/dsputil.o) When the problem happens I see that the compiler has taken the 100% of the cpu but still does not proceed further. Please note that I have even configured swap partition to make sure compiler does not run out of memory. Let me know if any of you have faced any such issue and resolved the same. Thanks in advance. Regards, -Ravi -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Problem-compiling-ffmpeg-compilation-gets-stuck-at-dsputil-c-tp3997175p3997175.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From aghaezat at hotmail.com Mon Nov 7 07:34:49 2011 From: aghaezat at hotmail.com (Ezat Ezati) Date: Sun, 6 Nov 2011 22:34:49 -0800 Subject: [FFmpeg-user] ffmpeg ffmpeg-php Message-ID: Hello. I am trying to install ffmpeg and ffmpeg-php on a Centos 6.0 32bit but I keep encountering error after error. Here is the code. It says Libvorbis not found but I have installed libogg and libvorbis using yum install libogg libvorbis. Any idea what the problem would be? Please help me on this. I need to do to install; Video Module FFmpeg FFmpeg-PHP Mplayer + Mencoder flv2tool Libogg + Libvorbis LAME MP3 Encoder MP4Box [root at php Downloads]# wget http://ffmpeg.org/releases/ffmpeg-0.7.6.tar.gz --2011-11-05 23:12:58-- http://ffmpeg.org/releases/ffmpeg-0.7.6.tar.gz Resolving ffmpeg.org... 192.190.173.45 Connecting to ffmpeg.org|192.190.173.45|:80... connected. HTTP request sent, awaiting response... 200 OK Length: 5493937 (5.2M) [application/x-gzip] Saving to: ???ffmpeg-0.7.6.tar.gz??? 100%[======================================>] 5,493,937 249K/s in 27s 2011-11-05 23:13:27 (196 KB/s) - ???ffmpeg-0.7.6.tar.gz??? saved [5493937/5493937] [root at php Downloads]# tar xzf ffmpeg-0.7.6.tar.gz [root at php Downloads]# ls ffmpeg-0.7.6 ffmpeg-php-0.6.0.tar ffmpeg-0.7.6.tar.gz mplayer-codecs-20061022-1.i386.rpm ffmpeg-php-0.6.0 mplayer-codecs-extra-20061022-1.i386.rpm [root at php Downloads]# cd ffmpeg-0.7.6 [root at php ffmpeg-0.7.6]# make distclean Makefile:1: config.mak: No such file or directory libavutil/Makefile:1: libavutil/../config.mak: No such file or directory libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' libavutil/../subdir.mak:26: warning: ignoring old commands for target `libavutil/' libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' libavutil/../subdir.mak:96: warning: ignoring old commands for target `libavutil/' Makefile:239: /tests/fate.mak: No such file or directory Makefile:240: /tests/fate2.mak: No such file or directory Makefile:242: /tests/fate/aac.mak: No such file or directory Makefile:243: /tests/fate/als.mak: No such file or directory Makefile:244: /tests/fate/fft.mak: No such file or directory Makefile:245: /tests/fate/h264.mak: No such file or directory Makefile:246: /tests/fate/mp3.mak: No such file or directory Makefile:247: /tests/fate/vorbis.mak: No such file or directory Makefile:248: /tests/fate/vp8.mak: No such file or directory make: *** No rule to make target `/tests/fate/vp8.mak'. Stop. [root at php ffmpeg-0.7.6]# ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-shared --enable-libmp3lame --enable-libx264 --enable-libfaac --enable-libvorbis --enable-libopencore-amrnb --enable-libopencore-amrwb ERROR: libvorbis 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. [root at php ffmpeg-0.7.6]# yum install libvorbis Loaded plugins: fastestmirror, presto, refresh-packagekit Loading mirror speeds from cached hostfile * base: mirror.chpc.utah.edu * extras: mirrors.200p-sf.sonic.net * updates: centos.hostrack.net Setting up Install Process Package 1:libvorbis-1.2.3-4.el6.i686 already installed and latest version Nothing to do [root at php ffmpeg-0.7.6]# yum install libogg Loaded plugins: fastestmirror, presto, refresh-packagekit Loading mirror speeds from cached hostfile * base: mirror.chpc.utah.edu * extras: mirrors.200p-sf.sonic.net * updates: centos.hostrack.net Setting up Install Process Package 2:libogg-1.1.4-2.1.el6.i686 already installed and latest version Nothing to do [root at php ffmpeg-0.7.6]# ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-shared --enable-libmp3lame --enable-libx264 --enable-libfaac --enable-libvorbis --enable-libopencore-amrnb --enable-libopencore-amrwb ERROR: libvorbis 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. [root at php ffmpeg-0.7.6]# From linuzzz at yahoo.it Mon Nov 7 10:51:07 2011 From: linuzzz at yahoo.it (Paolo) Date: Mon, 7 Nov 2011 10:51:07 +0100 Subject: [FFmpeg-user] forcing x264 baseline Message-ID: dear all, for a lot of time I used this command line to convert my video to a format compatible with my android device: ffmpeg -i input.mp4 -strict experimental -s 320x240 -b 384k -vcodec libx264 -flags +loop+mv4 -cmp 256 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -subq 7 -trellis 1 -refs 5 -bf 0 -flags2 +mixed_refs -coder 0 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -qmin 10 -qmax 51 -qdiff 4 -acodec aac -ac 2 -ab 128k -ar 44100 output.mp4 using mediainfo I see that Format profile : Baseline at L1.3 and the result file will play on android without problems Now, I suppose after an ffmpeg update, the same command line produce a file with Format profile : Main at L1.3 by this way the file mp4 will not play on my android device. I see in the man ffmpeg that "-profile" is deprecated, so, what will be the best way to force Baseline profile? many thanks Paolo From csavtche at gmail.com Mon Nov 7 18:20:42 2011 From: csavtche at gmail.com (Constantin Savtchenko) Date: Mon, 7 Nov 2011 12:20:42 -0500 Subject: [FFmpeg-user] Incomplete libavcodec in Ubuntu 10.04? Message-ID: Hello all, I am attempting to compile a program which uses the avcodec and avutil libraries. The program fails to compile because AVMEDIA_TYPE_VIDEO is undeclared and AVPacket is not defined. I see that the libraries on ffmpeg.org are complete with the necessary declaration and definition though. I am unable to remove avcodec using my package manager because an incredibly large package (ROS) is reliant on it and will be removed as well. Thus my question is two fold. Is it a known problem for Ubuntu 10.04 to have outdated/incomplete ffmpeg libraries? Is there a way to use ubuntu's package manager and update my avcodec libraries? Thank you all for any help. Constantin S PS - A comment in my libraries makes mention of AVPacket, yet AVPacket is not defined. What's going on? From h.reindl at thelounge.net Mon Nov 7 21:58:09 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 07 Nov 2011 21:58:09 +0100 Subject: [FFmpeg-user] ffmpeg ffmpeg-php In-Reply-To: References: Message-ID: <4EB84661.8080305@thelounge.net> Am 07.11.2011 07:34, schrieb Ezat Ezati: > I am trying to install ffmpeg and ffmpeg-php on a Centos 6.0 32bit this package is dead since a really long time ago it also did nevr provide any encoding interface only read of metainfos -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From belcampo at zonnet.nl Mon Nov 7 22:12:34 2011 From: belcampo at zonnet.nl (belcampo) Date: Mon, 07 Nov 2011 22:12:34 +0100 Subject: [FFmpeg-user] ffmpeg ffmpeg-php In-Reply-To: References: Message-ID: <4EB849C2.90705@zonnet.nl> On 11/07/2011 07:34 AM, Ezat Ezati wrote: > Hello. > > I am trying to install ffmpeg and ffmpeg-php on a Centos 6.0 32bit but I keep encountering error after error. Here is the code. It says Libvorbis not found but I have installed libogg and libvorbis using yum install libogg libvorbis. Any idea what the problem would be? Please help me on this. I need to do to install; > > Video Module > > FFmpeg > FFmpeg-PHP > Mplayer + Mencoder > flv2tool > Libogg + Libvorbis > LAME MP3 Encoder > MP4Box > > > > [root at php Downloads]# wget http://ffmpeg.org/releases/ffmpeg-0.7.6.tar.gz > --2011-11-05 23:12:58-- http://ffmpeg.org/releases/ffmpeg-0.7.6.tar.gz > Resolving ffmpeg.org... 192.190.173.45 > Connecting to ffmpeg.org|192.190.173.45|:80... connected. > HTTP request sent, awaiting response... 200 OK > Length: 5493937 (5.2M) [application/x-gzip] > Saving to: ???ffmpeg-0.7.6.tar.gz??? > > 100%[======================================>] 5,493,937 249K/s in 27s > > 2011-11-05 23:13:27 (196 KB/s) - ???ffmpeg-0.7.6.tar.gz??? saved [5493937/5493937] > > [root at php Downloads]# tar xzf ffmpeg-0.7.6.tar.gz > [root at php Downloads]# ls > ffmpeg-0.7.6 ffmpeg-php-0.6.0.tar > ffmpeg-0.7.6.tar.gz mplayer-codecs-20061022-1.i386.rpm > ffmpeg-php-0.6.0 mplayer-codecs-extra-20061022-1.i386.rpm > [root at php Downloads]# cd ffmpeg-0.7.6 > [root at php ffmpeg-0.7.6]# make distclean > Makefile:1: config.mak: No such file or directory > libavutil/Makefile:1: libavutil/../config.mak: No such file or directory > libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' > libavutil/../subdir.mak:26: warning: ignoring old commands for target `libavutil/' > libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' > libavutil/../subdir.mak:96: warning: ignoring old commands for target `libavutil/' > Makefile:239: /tests/fate.mak: No such file or directory > Makefile:240: /tests/fate2.mak: No such file or directory > Makefile:242: /tests/fate/aac.mak: No such file or directory > Makefile:243: /tests/fate/als.mak: No such file or directory > Makefile:244: /tests/fate/fft.mak: No such file or directory > Makefile:245: /tests/fate/h264.mak: No such file or directory > Makefile:246: /tests/fate/mp3.mak: No such file or directory > Makefile:247: /tests/fate/vorbis.mak: No such file or directory > Makefile:248: /tests/fate/vp8.mak: No such file or directory > make: *** No rule to make target `/tests/fate/vp8.mak'. Stop. > [root at php ffmpeg-0.7.6]# ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-shared --enable-libmp3lame --enable-libx264 --enable-libfaac --enable-libvorbis --enable-libopencore-amrnb --enable-libopencore-amrwb > ERROR: libvorbis not found You need the devel packages of libogg and libvorbis and probably also faac lame and x264 > > 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. > [root at php ffmpeg-0.7.6]# yum install libvorbis > Loaded plugins: fastestmirror, presto, refresh-packagekit > Loading mirror speeds from cached hostfile > * base: mirror.chpc.utah.edu > * extras: mirrors.200p-sf.sonic.net > * updates: centos.hostrack.net > Setting up Install Process > Package 1:libvorbis-1.2.3-4.el6.i686 already installed and latest version > Nothing to do > [root at php ffmpeg-0.7.6]# yum install libogg > Loaded plugins: fastestmirror, presto, refresh-packagekit > Loading mirror speeds from cached hostfile > * base: mirror.chpc.utah.edu > * extras: mirrors.200p-sf.sonic.net > * updates: centos.hostrack.net > Setting up Install Process > Package 2:libogg-1.1.4-2.1.el6.i686 already installed and latest version > Nothing to do > [root at php ffmpeg-0.7.6]# ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-shared --enable-libmp3lame --enable-libx264 --enable-libfaac --enable-libvorbis --enable-libopencore-amrnb --enable-libopencore-amrwb > ERROR: libvorbis 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. > [root at php ffmpeg-0.7.6]# > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jeisom at gmail.com Mon Nov 7 22:19:12 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Mon, 7 Nov 2011 15:19:12 -0600 Subject: [FFmpeg-user] forcing x264 baseline In-Reply-To: References: Message-ID: On Mon, Nov 7, 2011 at 3:51 AM, Paolo wrote: > dear all, > > for a lot of time I used this command line to convert my video to a > format compatible with my android device: > > ffmpeg -i input.mp4 -strict experimental -s 320x240 -b 384k -vcodec > libx264 -flags +loop+mv4 -cmp 256 -partitions > +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -subq 7 -trellis 1 -refs > 5 -bf 0 -flags2 +mixed_refs -coder 0 -me_range 16 -g 250 -keyint_min > 25 -sc_threshold 40 -i_qfactor 0.71 -qmin 10 -qmax 51 -qdiff 4 -acodec > aac -ac 2 -ab 128k -ar 44100 output.mp4 > > using mediainfo I see that > > Format profile ? ? ? ? ? ? ? ? ? ? ? ? ? : Baseline at L1.3 > > and the result file will play on android without problems > > Now, I suppose after an ffmpeg update, the same command line produce a file with > > Format profile ? ? ? ? ? ? ? ? ? ? ? ? ? : Main at L1.3 > > by this way the file mp4 will not play on my android device. > > I see in the man ffmpeg that "-profile" is deprecated, so, what will > be the best way to force Baseline profile? I believe you are looking for "-vprofile baseline" HTH Jonathan > many thanks > Paolo > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From krueger at lesspain.de Mon Nov 7 22:23:51 2011 From: krueger at lesspain.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Mon, 7 Nov 2011 22:23:51 +0100 Subject: [FFmpeg-user] forcing x264 baseline In-Reply-To: References: Message-ID: <25752AA8-218D-4D88-AD4F-ECE5E565648D@lesspain.de> On Nov 7, 2011, at 10:51 , Paolo wrote: > dear all, > > for a lot of time I used this command line to convert my video to a > format compatible with my android device: > > ffmpeg -i input.mp4 -strict experimental -s 320x240 -b 384k -vcodec > libx264 -flags +loop+mv4 -cmp 256 -partitions > +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -subq 7 -trellis 1 -refs > 5 -bf 0 -flags2 +mixed_refs -coder 0 -me_range 16 -g 250 -keyint_min > 25 -sc_threshold 40 -i_qfactor 0.71 -qmin 10 -qmax 51 -qdiff 4 -acodec > aac -ac 2 -ab 128k -ar 44100 output.mp4 > > using mediainfo I see that > > Format profile : Baseline at L1.3 > > and the result file will play on android without problems > > Now, I suppose after an ffmpeg update, the same command line produce a file with > > Format profile : Main at L1.3 > > by this way the file mp4 will not play on my android device. > > I see in the man ffmpeg that "-profile" is deprecated, so, what will http://ffmpeg.org/ffmpeg.html#Options this doesn't look like it's deprecated. Have you tried -profile baseline? And btw. it is recommended to use presets (i.e. -preset) instead of specifying all the encoder options. try x264 --fullhelp for docs. HTH, Robert From lou at lrcd.com Mon Nov 7 22:33:58 2011 From: lou at lrcd.com (Lou) Date: Mon, 7 Nov 2011 12:33:58 -0900 Subject: [FFmpeg-user] forcing x264 baseline In-Reply-To: <25752AA8-218D-4D88-AD4F-ECE5E565648D@lesspain.de> References: <25752AA8-218D-4D88-AD4F-ECE5E565648D@lesspain.de> Message-ID: <20111107123358.34cebab5@lrcd.com> On Mon, 7 Nov 2011 22:23:51 +0100 Robert Kr?ger wrote: > > On Nov 7, 2011, at 10:51 , Paolo wrote: > > > dear all, > > > > for a lot of time I used this command line to convert my video to a > > format compatible with my android device: > > > > ffmpeg -i input.mp4 -strict experimental -s 320x240 -b 384k -vcodec > > libx264 -flags +loop+mv4 -cmp 256 -partitions > > +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -subq 7 -trellis 1 > > -refs 5 -bf 0 -flags2 +mixed_refs -coder 0 -me_range 16 -g 250 > > -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -qmin 10 -qmax 51 > > -qdiff 4 -acodec aac -ac 2 -ab 128k -ar 44100 output.mp4 > > > > using mediainfo I see that > > > > Format profile : Baseline at L1.3 > > > > and the result file will play on android without problems > > > > Now, I suppose after an ffmpeg update, the same command line > > produce a file with > > > > Format profile : Main at L1.3 > > > > by this way the file mp4 will not play on my android device. > > > > I see in the man ffmpeg that "-profile" is deprecated, so, what will > > http://ffmpeg.org/ffmpeg.html#Options > > this doesn't look like it's deprecated. Have you tried -profile > baseline? -vprofile is required if audio is also being encoded otherwise ffmpeg will output errors. > And btw. it is recommended to use presets (i.e. -preset) > instead of specifying all the encoder options. try x264 --fullhelp > for docs. I second this suggestion. Using a preset, a close approximation of your command is: ffmpeg -i input.mp4 -vcodec libx264 -preset medium -vprofile baseline \ -s 320x240 -b 384k -acodec aac -ac 2 -ab 128k -ar 44100 output.mp4 Using the presets ensures that your command will more likely continue to work through any updates and API changes. > HTH, > > Robert From dashing.meng at gmail.com Tue Nov 8 03:26:10 2011 From: dashing.meng at gmail.com (littlebat) Date: Tue, 8 Nov 2011 10:26:10 +0800 Subject: [FFmpeg-user] Incomplete libavcodec in Ubuntu 10.04? In-Reply-To: References: Message-ID: <20111108102610.430f4775.dashing.meng@gmail.com> On Mon, 7 Nov 2011 12:20:42 -0500 Constantin Savtchenko wrote: > Hello all, > I am attempting to compile a program which uses the avcodec and > avutil libraries. The program fails to compile because > AVMEDIA_TYPE_VIDEO is undeclared and AVPacket is not defined. I see > that the libraries on ffmpeg.org are complete with the necessary > declaration and definition though. I am unable to remove avcodec > using my package manager because an incredibly large package (ROS) is > reliant on it and will be removed as well. Thus my question is two > fold. Is it a known problem for Ubuntu 10.04 to have > outdated/incomplete ffmpeg libraries? Is there a way to use ubuntu's > package manager and update my avcodec libraries? Thank you all for > any help. > > Constantin S > > PS - A comment in my libraries makes mention of AVPacket, yet AVPacket > is not defined. What's going on? I have used a way to install a ffmpeg with shared libaries enabled and keep the original ffmpeg in the Debian. 1, first uninstall ffmpeg and its libraries(such as libavcodec, etc.), and some packages(such as gnome-desktop-environment, gnome-desktop-environment includes totem, gnash, sound-juicer, etc..). I recorded the packages be purged under Ubuntu 10.10: sudo apt-get purge libavutil-dev libavutil50 libavutil-extra-50 \ libavcodec-dev libavcodec52 libavcodec-extra-52 libavformat-dev \ libavformat52 libavformat-extra-52 libavdevice-dev libavdevice52 \ libavdevice-extra-52 libavfilter-dev libavfilter1 libavfilter-extra-1 \ libswscale-dev libswscale0 libswscale-extra-0 libpostproc-dev \ libpostproc51 libpostproc-extra-51 gnome-desktop-environment 2, Compile and install ffmpeg into /usr/local from source. 3, Install original Debian ffmpeg and the packages to be purged in the first setup. It works for me, only for reference. From cehoyos at ag.or.at Tue Nov 8 10:19:29 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 09:19:29 +0000 (UTC) Subject: [FFmpeg-user] Error when creating jpeg from video frame in ffmpeg 0.8.5 References: <14913891031.20111106192457@io.ua> <14434948703.20111107011554@io.ua> Message-ID: Anderw Gora io.ua> writes: > Here it is: http://www.datafilehost.com/download-de5f253a.html Fixed in git head, thank you for the sample! Carl Eugen From cehoyos at ag.or.at Tue Nov 8 10:53:10 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 09:53:10 +0000 (UTC) Subject: [FFmpeg-user] Incomplete libavcodec in Ubuntu 10.04? References: Message-ID: Constantin Savtchenko gmail.com> writes: > I am attempting to compile a program which uses the avcodec and > avutil libraries. The program fails to compile because > AVMEDIA_TYPE_VIDEO is undeclared and AVPacket is not defined. I see > that the libraries on ffmpeg.org are complete with the necessary > declaration and definition though. I am unable to remove avcodec > using my package manager because an incredibly large package (ROS) is > reliant on it and will be removed as well. Thus my question is two > fold. Is it a known problem for Ubuntu 10.04 to have > outdated/incomplete ffmpeg libraries? There are two unrelated problems with FFmpeg in Ubuntu: The FFmpeg version in 10.04 is simply old (because 10.04 is old) and does not contain (for example) the definition for AVMEDIA_TYPE_VIDEO. Adapting your program to (also) work with FFmpeg branch oldabi possibly fixes this problem. An alternative is to link against static libav*. The more severe problem is that current Ubuntu contains an intentionally broken version of FFmpeg that contains >100 bugs, missing features and security issues not present in FFmpeg. I believe there are repositories that contain a working version of FFmpeg. Carl Eugen From cehoyos at ag.or.at Tue Nov 8 10:56:04 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 09:56:04 +0000 (UTC) Subject: [FFmpeg-user] video 2160p50 References: Message-ID: Olivier Camus ate-group.com> writes: > I cannot read a large video (format 3840*2160) using ffplay, neither on Linux > nor on Windows. The frame rate is not correct: to slow. FFplay depends on external libraries, and while I did not test, I don't find it surprising that it has performance issues. Does it work correctly with mplayer (speed 0.2)? Carl Eugen From fox at orbitalfox.com Tue Nov 8 11:58:34 2011 From: fox at orbitalfox.com (fox) Date: Tue, 8 Nov 2011 10:58:34 +0000 Subject: [FFmpeg-user] Problem compiling ffmpeg - compilation gets stuck at dsputil.c In-Reply-To: References: Message-ID: <20111108105834.3a372ac3@apollon> On Sun, 6 Nov 2011 17:04:48 -0800 (PST) "ravi.kumar" wrote: > When the problem happens I see that the compiler has taken the 100% > of the cpu but still does not proceed further. Please note that I > have even configured swap partition to make sure compiler does not > run out of memory. I dont know much about the particular source nor architecture, but even if you have swap setup and some compilation takes a lot of time, the thrashing will make the process take a lot longer. But you shouldnt be seeing 100% utilisation at that point. Fox From fox at orbitalfox.com Tue Nov 8 12:04:19 2011 From: fox at orbitalfox.com (fox) Date: Tue, 8 Nov 2011 11:04:19 +0000 Subject: [FFmpeg-user] Incomplete libavcodec in Ubuntu 10.04? In-Reply-To: References: Message-ID: <20111108110419.56d9ff5f@apollon> On Tue, 8 Nov 2011 09:53:10 +0000 (UTC) Carl Eugen Hoyos wrote: > The more severe problem is that current Ubuntu contains an > intentionally broken version of FFmpeg that contains >100 bugs, > missing features and security issues not present in FFmpeg. > I believe there are repositories that contain a working version of > FFmpeg. Carl what do you mean by intentionally broken? What is the intention? From fox at orbitalfox.com Tue Nov 8 12:05:16 2011 From: fox at orbitalfox.com (fox) Date: Tue, 8 Nov 2011 11:05:16 +0000 Subject: [FFmpeg-user] Incomplete libavcodec in Ubuntu 10.04? In-Reply-To: References: Message-ID: <20111108110516.101ceb57@apollon> Look into using a PPA. I think there is a ffmpeg-daily one. Fox From cehoyos at ag.or.at Tue Nov 8 14:29:35 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 13:29:35 +0000 (UTC) Subject: [FFmpeg-user] Blue Cherry PV-143 (BT878 Chipset) not working! References: <1320158162.42207.YahooMailNeo@web113314.mail.gq1.yahoo.com> Message-ID: Mike Carambat rocketmail.com> writes: > I keep getting ioctl errors when issuing the following simplified command: > > ffmpeg -an -f video4linux2 -i /dev/video0 test.avi While a lot of necessary information is missing (complete, uncut output), it is possible that this has been fixed in current git head. Carl Eugen From cehoyos at ag.or.at Tue Nov 8 18:32:37 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 17:32:37 +0000 (UTC) Subject: [FFmpeg-user] Problem compiling ffmpeg - compilation gets stuck at dsputil.c References: Message-ID: ravi.kumar zenverge.com> writes: > The problem I am facing is, after configure, when I run make, it compiles few > files but gets stuck while compiling dsputil.c (CC libavcodec/dsputil.o) How many hours did you wait? Carl Eugen From tuan.dn at anlab.vn Tue Nov 8 08:39:48 2011 From: tuan.dn at anlab.vn (Tuan DN) Date: Tue, 8 Nov 2011 14:39:48 +0700 Subject: [FFmpeg-user] Decode h264 width no start code error Message-ID: Hi everyone, I am a newbie with ffmpeg, I would like to ask a question. === I use ffmpeg.exe as external program to generate thumbnail by following command: -itsoffset -7 -i ..\outputdata\output.mp4 -vcodec mjpeg -vframes 1 -an -f image2 ..\outputdata\ouput.jpg File MP4 is h264 standard My problem is that - If file MP4 have start code at the begin of frame: 00 00 00 01 then above command runs ok and generates output.jpg - If file MP4 have no start code then above command will faill to generate output.jpg, please see the attached file + [h264 @ 01eb1fe0] no frame! + [h264 @ 01eb1fe0] mmco: unref short failure === Could someone give me some advice on how to generate image Forgive my English. -------------- next part -------------- A non-text attachment was scrubbed... Name: noStartCode.mp4 Type: video/mp4 Size: 4168910 bytes Desc: not available URL: From adismsc at gmail.com Tue Nov 8 11:42:46 2011 From: adismsc at gmail.com (e) Date: Tue, 8 Nov 2011 11:42:46 +0100 Subject: [FFmpeg-user] FFmpeg MPEG-TS encoding Message-ID: Hi, is there anybody who can help me with video encoding? I want to encode video to MPEG-TS with null packet (PID 0x1fff). Anybody know how to do this? Is it possible? My current command: ffmpeg -i audio.mp2 -i video.mpg -s 720x576 -minrate 3200k -maxrate 3200k -bufsize 3200k -vcodec libx264 -vpre libx264-iptv -acodec copy -r 25 -f mpegts out.mpg From cehoyos at ag.or.at Tue Nov 8 19:30:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 8 Nov 2011 18:30:36 +0000 (UTC) Subject: [FFmpeg-user] Decode h264 width no start code error References: Message-ID: Tuan DN anlab.vn> writes: > Attachment (noStartCode.mp4): video/mp4, 4071 KiB Which program can play this file? Carl Eugen From tuan.dn at anlab.vn Wed Nov 9 03:24:46 2011 From: tuan.dn at anlab.vn (Tuan DN) Date: Wed, 9 Nov 2011 09:24:46 +0700 Subject: [FFmpeg-user] Decode h264 width no start code error In-Reply-To: References: Message-ID: It can be played by Windows Media Player (I use win7) By your suggestion, I check it more width VLC and it can play but not available to see anything. On Wed, Nov 9, 2011 at 1:30 AM, Carl Eugen Hoyos wrote: > Tuan DN anlab.vn> writes: > > > Attachment (noStartCode.mp4): video/mp4, 4071 KiB > > Which program can play this file? > > 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 Nov 9 11:10:58 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 10:10:58 +0000 (UTC) Subject: [FFmpeg-user] Decode h264 width no start code error References: Message-ID: Tuan DN anlab.vn> writes: > It can be played by Windows Media Player (I use win7) This is now ticket #631. Thank you for the sample, Carl Eugen From ssoni at lifesize.com Wed Nov 9 12:12:37 2011 From: ssoni at lifesize.com (Sanjay Soni) Date: Wed, 9 Nov 2011 05:12:37 -0600 Subject: [FFmpeg-user] Chroma buffer offset Message-ID: Hi, I am using FFMPEG to decode H264 streams. My question is why in the output yuv AVFrame why data[1] is not equal to data[0] + (linesize[0] * height). The chroma pointer which I print is little further than that. Can somebody tell how can I fix this? I am feeding this yuv to other encoder which does not have facility to take Y, U and V plane pointer separately. It just takes Y starting address and then calculate U = Y + stride * height !!! Thanks SS From rickcorteza at gmail.com Wed Nov 9 13:21:08 2011 From: rickcorteza at gmail.com (Rick C.) Date: Wed, 9 Nov 2011 20:21:08 +0800 Subject: [FFmpeg-user] audio streams Message-ID: Hi, I have noticed that taking .vob files from a DVD and converting them to let's say .mp4 (ffmpeg -i input.vob output.mp4) will often grab the wrong audio track and the converted .mp4 will end up with a narrative track or something but not the expected default audio track. What is the best way to address this issue? Thanks! rc From cehoyos at ag.or.at Wed Nov 9 13:55:19 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 12:55:19 +0000 (UTC) Subject: [FFmpeg-user] audio streams References: Message-ID: Rick C. gmail.com> writes: > I have noticed that taking .vob files from a DVD and converting them to let's > say .mp4 (ffmpeg -i input.vob output.mp4) will often grab the wrong audio > track and the converted .mp4 will end up with a narrative track or something > but not the expected default audio track. What is the best way to address > this issue? (Complete, uncut output missing.) The -map option allows you to choose which stream(s) of the input file should be encoded. Carl Eugen From dwalker0044 at gmail.com Wed Nov 9 15:20:00 2011 From: dwalker0044 at gmail.com (David Walker) Date: Wed, 9 Nov 2011 14:20:00 +0000 Subject: [FFmpeg-user] ffmpeg marker error mpeg4videocodec.c Message-ID: Hi everyone, hope I'm posting to the correct place (should this be posted to devel?). When playing mpeg4 video I receive the message: marker does not match f_code. I have traced this back to line 361 in mpeg4videocodec.c. There are two things I would like to understand: the first is what this piece of code is doing? I.e. what does it expect / trying to find. Is it related to the video packet header? The second is what is the GetBitContext struct? I see it used to determine len but what is its significance in the grander scheme of things. I'd be very grateful if someone could shed some light on this! Thanks, From cehoyos at ag.or.at Wed Nov 9 16:10:00 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 15:10:00 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg marker error mpeg4videocodec.c References: Message-ID: David Walker gmail.com> writes: > Hi everyone, hope I'm posting to the correct place (should this be posted > to devel?). Only if you intend to send a patch fixing this problem (?). > When playing mpeg4 video I receive the message: marker does not match > f_code. Command line and complete, uncut output (and a sample) missing. Carl Eugen From mediastream at gmail.com Wed Nov 9 16:31:40 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 9 Nov 2011 10:31:40 -0500 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. Message-ID: A long story, but will try to keep it short. Was converting MPEG2 to H,.264 noticed that video has faint but persistent vertical lines top to bottom. It's not extremely noticeable, but gets very annoying as lines stay in one place and 'burn in' after a while. Playback: QuickTime, VLC and FFplay media players. Encoding: FFmpeg 0.8.6 release with latest libx264 (Nov 5th tar ball). Source file:http://www.mediafire.com/?khrzaps202b1ksq FFmpeg & libx264: http://www.mediafire.com/?3fkyzo1oh5d6l72 X264 encode(y4m to H264): http://www.mediafire.com/?g8x2ww1rcwhm08k y4m file (scaled to 800x448 by ffmpeg): http://www.mediafire.com/?y8cvvk07r1vegna Any advice on removing these lines, perhaps a video filter? Or other ways control the reduction of chroma spread, so that dark grays are encoded as black? Thank you all. From cehoyos at ag.or.at Wed Nov 9 17:06:52 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 16:06:52 +0000 (UTC) Subject: [FFmpeg-user] vertical lines noise in scaled raw video. References: Message-ID: Dennis gmail.com> writes: > Was converting MPEG2 to H,.264 noticed that video has faint but persistent > vertical lines top to bottom. It's not extremely noticeable, but gets very > annoying as lines stay in one place and 'burn in' after a while. > > Playback: QuickTime, VLC and FFplay media players. > Encoding: FFmpeg 0.8.6 release with latest libx264 (Nov 5th tar ball). > > Source file:http://www.mediafire.com/?khrzaps202b1ksq > FFmpeg & libx264: http://www.mediafire.com/?3fkyzo1oh5d6l72 > X264 encode(y4m to H264): http://www.mediafire.com/?g8x2ww1rcwhm08k > y4m file (scaled to 800x448 by ffmpeg): > http://www.mediafire.com/?y8cvvk07r1vegna Since I don't see any obvious red lines: Which files show red lines with FFplay/MPlayer (can QuickTime really play raw H264?), which do not? Are the red lines also reproducible with ffmpeg -i file -ss 35 out.png / out.jpg? Carl Eugen From sheen.andy at googlemail.com Wed Nov 9 17:05:59 2011 From: sheen.andy at googlemail.com (Andy Sheen) Date: Wed, 09 Nov 2011 16:05:59 +0000 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. In-Reply-To: References: Message-ID: <4EBAA4E7.9070706@googlemail.com> I've downloaded your ffmpeg recode. As far as I can tell I can't see any vertical lines on a calibrated monitor (playing with mpc-hc). Do you have the latest drivers for your graphics card? Tried it on a different PC/output device? It may be a rendering problem. Dennis wrote on Wed 09 Nov at 15:31 UK time > A long story, but will try to keep it short. > > Was converting MPEG2 to H,.264 noticed that video has faint but persistent > vertical lines top to bottom. It's not extremely noticeable, but gets very > annoying as lines stay in one place and 'burn in' after a while. > > Playback: QuickTime, VLC and FFplay media players. > Encoding: FFmpeg 0.8.6 release with latest libx264 (Nov 5th tar ball). > > Source file:http://www.mediafire.com/?khrzaps202b1ksq > FFmpeg & libx264: http://www.mediafire.com/?3fkyzo1oh5d6l72 > X264 encode(y4m to H264): http://www.mediafire.com/?g8x2ww1rcwhm08k > y4m file (scaled to 800x448 by ffmpeg): > http://www.mediafire.com/?y8cvvk07r1vegna > > Any advice on removing these lines, perhaps a video filter? Or other ways > control the reduction of chroma spread, so that dark grays are encoded as > black? > > Thank you all. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mediastream at gmail.com Wed Nov 9 17:45:47 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 9 Nov 2011 11:45:47 -0500 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. In-Reply-To: <4EBAA4E7.9070706@googlemail.com> References: <4EBAA4E7.9070706@googlemail.com> Message-ID: On Wed, Nov 9, 2011 at 11:05 AM, Andy Sheen wrote: > I've downloaded your ffmpeg recode. As far as I can tell I can't see any > vertical lines on a calibrated monitor (playing with mpc-hc). > > Do you have the latest drivers for your graphics card? Tried it on a > different PC/output device? It may be a rendering problem. > > Dennis wrote on Wed 09 Nov at 15:31 UK time > > A long story, but will try to keep it short. > > > > Was converting MPEG2 to H,.264 noticed that video has faint but > persistent > > vertical lines top to bottom. It's not extremely noticeable, but gets > very > > annoying as lines stay in one place and 'burn in' after a while. > > > > Playback: QuickTime, VLC and FFplay media players. > > Encoding: FFmpeg 0.8.6 release with latest libx264 (Nov 5th tar ball). > > > > Source file:http://www.mediafire.com/?khrzaps202b1ksq > > FFmpeg & libx264: http://www.mediafire.com/?3fkyzo1oh5d6l72 > > X264 encode(y4m to H264): http://www.mediafire.com/?g8x2ww1rcwhm08k > > y4m file (scaled to 800x448 by ffmpeg): > > http://www.mediafire.com/?y8cvvk07r1vegna > > > > Any advice on removing these lines, perhaps a video filter? Or other ways > > control the reduction of chroma spread, so that dark grays are encoded as > > black? > Here's the screenshot with "find edges" filter applied by ImageJ : http://www.mediafire.com/?0hesdf21e6a2ok2 these lines get amplified when encoded by x264, if I can suppress the amplification it'd be kinda cool. From rjwse at aol.com Wed Nov 9 18:05:16 2011 From: rjwse at aol.com (pop) Date: Wed, 09 Nov 2011 11:05:16 -0600 Subject: [FFmpeg-user] "screencastor" French GUI Message-ID: <4EBAB2CC.7040104@aol.com> Would like to use this delightful-looking program in Ubuntu, but it does not seem to work after installation. I have been making YouTube 'how to' videos about Ubuntu using ffmpeg through a terminal command (which works quite well). I would like to use 'screencastor' but do not know French. Would like to help with this project in English. Best regards, Pop From mediastream at gmail.com Wed Nov 9 18:12:39 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 9 Nov 2011 12:12:39 -0500 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. In-Reply-To: References: <4EBAA4E7.9070706@googlemail.com> Message-ID: > > On Wed, Nov 9, 2011 at 11:05 AM, Andy Sheen wrote: > >> I've downloaded your ffmpeg recode. As far as I can tell I can't see any >> vertical lines on a calibrated monitor (playing with mpc-hc). >> >> Do you have the latest drivers for your graphics card? Tried it on a >> different PC/output device? It may be a rendering problem. >> >> Dennis wrote on Wed 09 Nov at 15:31 UK time >> > A long story, but will try to keep it short. >> > >> > Was converting MPEG2 to H,.264 noticed that video has faint but >> persistent >> > vertical lines top to bottom. It's not extremely noticeable, but gets >> very >> > annoying as lines stay in one place and 'burn in' after a while. >> > >> > Playback: QuickTime, VLC and FFplay media players. >> > Encoding: FFmpeg 0.8.6 release with latest libx264 (Nov 5th tar ball). >> > >> > Source file:http://www.mediafire.com/?khrzaps202b1ksq >> > FFmpeg & libx264: http://www.mediafire.com/?3fkyzo1oh5d6l72 >> > X264 encode(y4m to H264): http://www.mediafire.com/?g8x2ww1rcwhm08k >> > y4m file (scaled to 800x448 by ffmpeg): >> > http://www.mediafire.com/?y8cvvk07r1vegna >> > >> > Any advice on removing these lines, perhaps a video filter? Or other >> ways >> > control the reduction of chroma spread, so that dark grays are encoded >> as >> > black? >> > > Here's the screenshot of the scaled y4m with "find edges" filter applied > by ImageJ : http://www.mediafire.com/?0hesdf21e6a2ok2 > these lines get amplified when encoded by x264, if I can suppress the > amplification it'd be kinda cool. > > Versus unscaled y4m screenshot with the same filter applied - its just > solid black: http://www.mediafire.com/?5m3seip1z8zb4mf > From lists at arlomedia.com Wed Nov 9 18:29:44 2011 From: lists at arlomedia.com (Arlo Leach) Date: Wed, 9 Nov 2011 09:29:44 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy Message-ID: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> Hello, I'm converting user-uploaded videos to H.264/MP4 files with ffmpeg 0.6.1 (obtained from yum on CentOS): ffmpeg -i input.mov -f mp4 -vcodec libx264 -vpre medium -acodec libfaac -r 15 -b 360k -ab 48k -ar 22050 -s 480x320 fullclip.mp4 Then I want to extract sample clips from those videos. This should give me the first 30 seconds: ffmpeg -i fullclip.mp4 -vcodec copy -acodec copy -ss 0 -t 30 sampleclip.mp4 But when I do that, the sampleclip.mp4 file has no video track. I don't see an error message from ffmpeg; it just shows a final video size of 0kb in its output. If I apply the second command to the H.264/MP4 demo clip that came with JWPlayer, I have no problems. But if I re-encode that clip with my first command, then apply the second command, the video conversion fails again. So it looks like my second command is OK but my first command is missing something to make the file complete. Unfortunately I've tried adding lots of additional options gathered from various tutorial sites and none have fixed the problem. If I change -vcodec copy to -vcodec libx264 in my second command, I do get a video track, but I would rather not re-encode the video when I extract the sample. I'm pasting full examples (using shorter videos) below. Can anyone see what's wrong? Thanks, -Arlo _______________________________ Encoding command: /usr/bin/ffmpeg -i /tmp/php5GkSrD -f mp4 -vcodec libx264 -vpre medium -acodec libfaac -r 15 -b 360k -ab 48k -ar 22050 -s 512x288 -t 900 175.mp4 Encoding output: FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Dec 4 2010 15:35:31 with gcc 4.1.2 20080704 (Red Hat 4.1.2-48) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab 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 Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/php5GkSrD': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : HandBrake 0.9.5 2011010300 Duration: 00:00:03.26, start: 0.000000, bitrate: 5407 kb/s Stream #0.0(): Video: h264, yuv420p, 1024x576 [PAR 1:1 DAR 16:9], 5223 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 171 kb/s [libx264 @ 0x282e0e0]using SAR=1/1 [libx264 @ 0x282e0e0]using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0x282e0e0]profile High, level 2.1 [libx264 @ 0x282e0e0]264 - core 107 - H.264/MPEG-4 AVC codec - Copyleft 2003-2010 - 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=1 sliced_threads=0 nr=0 decimate=1 interlaced=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=360 ratetol=11.1 qcomp=0.60 qpmin=10 qpmax=51 qpstep=4 ip_ratio=1.41 aq=1:1.00 Output #0, mp4, to '175.mp4': Metadata: encoder : Lavf52.64.2 Stream #0.0(): Video: libx264, yuv420p, 512x288 [PAR 1:1 DAR 16:9], q=10-51, 360 kb/s, 15 tbn, 15 tbc Stream #0.1(eng): Audio: libfaac, 22050 Hz, stereo, s16, 48 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 19 fps= 0 q=-1.0 size= 0kB time=1.86 bitrate= 0.2kbits/s dup=0 drop=18 frame= 41 fps= 39 q=-1.0 size= 0kB time=3.11 bitrate= 0.1kbits/s dup=0 drop=40 frame= 49 fps= 21 q=-1.0 Lsize= 149kB time=3.13 bitrate= 390.4kbits/s dup=0 drop=49 video:128kB audio:19kB global headers:0kB muxing overhead 1.678923% [libx264 @ 0x282e0e0]frame I:1 Avg QP:27.47 size: 34806 [libx264 @ 0x282e0e0]frame P:27 Avg QP:34.61 size: 3092 [libx264 @ 0x282e0e0]frame B:21 Avg QP:39.38 size: 581 [libx264 @ 0x282e0e0]consecutive B-frames: 12.5% 87.5% 0.0% 0.0% [libx264 @ 0x282e0e0]mb I I16..4: 1.6% 29.2% 69.3% [libx264 @ 0x282e0e0]mb P I16..4: 0.2% 0.3% 0.3% P16..4: 24.7% 20.2% 14.9% 0.0% 0.0% skip:39.5% [libx264 @ 0x282e0e0]mb B I16..4: 0.0% 0.0% 0.0% B16..8: 30.2% 6.1% 1.4% direct: 1.6% skip:60.7% L0:31.0% L1:60.4% BI: 8.6% [libx264 @ 0x282e0e0]final ratefactor: 27.48 [libx264 @ 0x282e0e0]8x8 transform intra:30.9% inter:43.7% [libx264 @ 0x282e0e0]coded y,uvDC,uvAC intra: 86.6% 90.9% 75.6% inter: 14.6% 8.6% 1.7% [libx264 @ 0x282e0e0]i16 v,h,dc,p: 44% 12% 26% 18% [libx264 @ 0x282e0e0]i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 11% 17% 8% 8% 7% 8% 11% 11% [libx264 @ 0x282e0e0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 13% 15% 8% 9% 8% 9% 10% 10% [libx264 @ 0x282e0e0]i8c dc,h,v,p: 55% 15% 21% 8% [libx264 @ 0x282e0e0]Weighted P-Frames: Y:0.0% [libx264 @ 0x282e0e0]ref P L0: 72.9% 13.0% 8.2% 6.0% [libx264 @ 0x282e0e0]ref B L0: 89.4% 10.6% [libx264 @ 0x282e0e0]kb/s:319.57 Excerpt command: /usr/bin/ffmpeg -i 175.mp4 -f mp4 -vcodec copy -acodec copy -ss 1 -t 2 -y 175_sample.mp4 Excerpt output: FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Dec 4 2010 15:35:31 with gcc 4.1.2 20080704 (Red Hat 4.1.2-48) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab 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 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '175.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf52.64.2 Duration: 00:00:03.26, start: 0.000000, bitrate: 374 kb/s Stream #0.0(): Video: h264, yuv420p, 512x288 [PAR 1:1 DAR 16:9], 321 kb/s, 15 fps, 15 tbr, 15 tbn, 30 tbc Stream #0.1(eng): Audio: aac, 22050 Hz, stereo, s16, 48 kb/s Output #0, mp4, to '175_sample.mp4': Metadata: encoder : Lavf52.64.2 Stream #0.0(): Video: libx264, yuv420p, 512x288 [PAR 1:1 DAR 16:9], q=2-31, 321 kb/s, 15 tbn, 15 tbc Stream #0.1(eng): Audio: libfaac, 22050 Hz, stereo, 48 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 0 fps= 0 q=-1.0 Lsize= 13kB time=2.02 bitrate= 51.2kbits/s video:0kB audio:12kB global headers:0kB muxing overhead 8.727150% From cehoyos at ag.or.at Wed Nov 9 18:42:14 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 17:42:14 +0000 (UTC) Subject: [FFmpeg-user] trouble with -vcodec copy References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> Message-ID: Arlo Leach arlomedia.com> writes: > I'm converting user-uploaded videos to H.264/MP4 files with ffmpeg 0.6.1 > (obtained from yum on CentOS): This version is very old and there is not enough manpower to support old FFmpeg versions. I you cannot use current git head (this is recommended), please use at least 0.7.7 which is ABI/API compatible with 0.6 Carl Eugen From tuan.dn at anlab.vn Wed Nov 9 10:03:31 2011 From: tuan.dn at anlab.vn (Tuan DN) Date: Wed, 9 Nov 2011 16:03:31 +0700 Subject: [FFmpeg-user] Decode h264 width no start code error In-Reply-To: References: Message-ID: Other MP4 file (h264) with no start code 00 00 00 01, width same command: ffmpeg.exe -itsoffset -7 -i ..\outputdata\output.mp4 -vcodec mjpeg -vframes 1 -an -f image2 ..\outputdata\ouput.jpg Error is: [h264 @ 012d1fe0] error while decoding MB 7 0 (bytestream -58) === Could someone give me some advice, did I miss something On Wed, Nov 9, 2011 at 9:24 AM, Tuan DN wrote: > It can be played by Windows Media Player (I use win7) > > By your suggestion, I check it more width VLC and it can play but not > available to see anything. > > > > > > On Wed, Nov 9, 2011 at 1:30 AM, Carl Eugen Hoyos wrote: > >> Tuan DN anlab.vn> writes: >> >> > Attachment (noStartCode.mp4): video/mp4, 4071 KiB >> >> Which program can play this file? >> >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > -------------- next part -------------- A non-text attachment was scrubbed... Name: output.mp4 Type: video/mp4 Size: 264810 bytes Desc: not available URL: From cehoyos at ag.or.at Wed Nov 9 18:47:14 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 9 Nov 2011 17:47:14 +0000 (UTC) Subject: [FFmpeg-user] vertical lines noise in scaled raw video. References: <4EBAA4E7.9070706@googlemail.com> Message-ID: Dennis gmail.com> writes: > Here's the screenshot with "find edges" filter applied by ImageJ : > http://www.mediafire.com/?0hesdf21e6a2ok2 I can see the artefacts now (don't know why I searched for "red" lines). Do you think they are in the original video and amplified by the encoding (and you are searching for a filter that suppresses them), or is FFmpeg producing them (which may be a serious bug)? Carl Eugen From mediastream at gmail.com Wed Nov 9 19:52:41 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 9 Nov 2011 13:52:41 -0500 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. In-Reply-To: References: <4EBAA4E7.9070706@googlemail.com> Message-ID: On Wed, Nov 9, 2011 at 12:47 PM, Carl Eugen Hoyos wrote: > Dennis gmail.com> writes: > > > Here's the screenshot with "find edges" filter applied by ImageJ : > > http://www.mediafire.com/?0hesdf21e6a2ok2 > > I can see the artefacts now (don't know why I searched for "red" lines). > Do you think they are in the original video and amplified by the encoding > (and > you are searching for a filter that suppresses them), or is FFmpeg > producing > them (which may be a serious bug)? > > Carl Eugen_______________________________________________ > well, look at the un-scaled y4m output screenshot here: http://www.mediafire.com/i/?5m3seip1z8zb4mf its solid black, i'd expect the same from the scaled screenshot. From lists at arlomedia.com Wed Nov 9 23:03:51 2011 From: lists at arlomedia.com (Arlo Leach) Date: Wed, 9 Nov 2011 14:03:51 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> Message-ID: <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> Hello, > This version is very old and there is not enough manpower to support old FFmpeg > versions. > > I you cannot use current git head (this is recommended), please use at least > 0.7.7 which is ABI/API compatible with 0.6 I'm not real good at installing Linux software, but I managed to uninstall libx286 and ffmpeg from yum, reinstall manually from the latest git versions, and find a new x264 preset name to use with this version. Unfortunately I'm still getting the same result -- no video track if I use -vcodec copy. Actually, I've been testing with a short source video (15 seconds) and extracted sample (3-4 seconds). I just tried with a longer source video (180 seconds) and sample (30 seconds) and my sample clip has a video track, but the first six seconds of it are blank (I just see the background color of the video player). The audio plays correctly from the start, but the video doesn't begin until six seconds into the clip. I'm pasting the updated commands and output below. Can anyone see what's wrong now? Thanks, -Arlo _______________________________ Encoding command: /usr/bin/ffmpeg -i /tmp/phpfLnsEm -f mp4 -vcodec libx264 -vpre libx264-baseline -acodec libfaac -r 15 -b 360k -ab 48k -ar 22050 -s 480x320 -t 900 184.mp4 Encoding output: ffmpeg version N-34622-g701e534, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 9 2011 13:09:49 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 29. 0 / 53. 29. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 47. 0 / 2. 47. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/phpfLnsEm': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-27 05:21:54 encoder : HandBrake 0.9.5 2011010300 Duration: 00:00:03.26, start: 0.000000, bitrate: 5407 kb/s Stream #0:0(): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1024x576 [SAR 1:1 DAR 16:9], 5223 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc Metadata: creation_time : 2011-10-27 05:21:54 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 171 kb/s Metadata: creation_time : 2011-10-27 05:21:54 handler_name : Please use -b:a or -b:v, -b is ambiguous [buffer @ 0x1ac413e0] w:1024 h:576 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x1ac41680] w:1024 h:576 fmt:yuv420p -> w:480 h:320 fmt:yuv420p flags:0x4 [libx264 @ 0x1ac3a220] using SAR=32/27 [libx264 @ 0x1ac3a220] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0x1ac3a220] profile High, level 2.1 [libx264 @ 0x1ac3a220] 264 - core 119 r2106 07efeb4 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 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 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=15 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=360 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to '184.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-27 05:21:54 encoder : Lavf53.20.0 Stream #0:0(): Video: h264 (![0][0][0] / 0x0021), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=-1--1, 360 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 2011-10-27 05:21:54 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: aac (@[0][0][0] / 0x0040), 22050 Hz, stereo, s16, 48 kb/s Metadata: creation_time : 2011-10-27 05:21:54 handler_name : Stream mapping: Stream #0:0 -> #0:0 (h264 -> libx264) Stream #0:1 -> #0:1 (aac -> libfaac) Press [q] to stop, [?] for help frame= 18 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s dup=0 drop=15 frame= 38 fps= 38 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s dup=0 drop=35 frame= 51 fps= 29 q=-1.0 Lsize= 173kB time=00:00:03.34 bitrate= 423.3kbits/s dup=0 drop=47 video:151kB audio:19kB global headers:0kB muxing overhead 1.248162% [libx264 @ 0x1ac3a220] frame I:1 Avg QP:25.95 size: 41153 [libx264 @ 0x1ac3a220] frame P:50 Avg QP:36.81 size: 2259 [libx264 @ 0x1ac3a220] mb I I16..4: 3.0% 20.3% 76.7% [libx264 @ 0x1ac3a220] mb P I16..4: 0.1% 0.2% 0.2% P16..4: 27.0% 16.2% 6.6% 0.0% 0.0% skip:49.7% [libx264 @ 0x1ac3a220] final ratefactor: 30.53 [libx264 @ 0x1ac3a220] 8x8 transform intra:23.0% inter:41.0% [libx264 @ 0x1ac3a220] coded y,uvDC,uvAC intra: 86.7% 91.9% 81.2% inter: 15.3% 11.3% 1.8% [libx264 @ 0x1ac3a220] i16 v,h,dc,p: 56% 17% 26% 2% [libx264 @ 0x1ac3a220] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 10% 21% 7% 9% 8% 8% 9% 12% [libx264 @ 0x1ac3a220] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 11% 14% 9% 9% 9% 8% 10% 9% [libx264 @ 0x1ac3a220] i8c dc,h,v,p: 53% 14% 23% 10% [libx264 @ 0x1ac3a220] ref P L0: 87.0% 6.7% 6.3% [libx264 @ 0x1ac3a220] kb/s:362.60 Extract command: /usr/bin/ffmpeg -i 184.mp4 -f mp4 -vcodec copy -acodec copy -ss 0 -t 4 -y 184_sample.mp4 Extract output: ffmpeg version N-34622-g701e534, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 9 2011 13:09:49 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 29. 0 / 53. 29. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 47. 0 / 2. 47. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '184.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2011-10-27 08:00:00 encoder : Lavf53.20.0 Duration: 00:00:03.40, start: 0.000000, bitrate: 416 kb/s Stream #0:0(): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 480x320 [SAR 32:27 DAR 16:9], 364 kb/s, 15 fps, 15 tbr, 15 tbn, 30 tbc Metadata: creation_time : 2011-10-27 08:00:00 handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 22050 Hz, stereo, s16, 47 kb/s Metadata: creation_time : 2011-10-27 08:00:00 handler_name : Output #0, mp4, to '184_sample.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 2011-10-27 08:00:00 encoder : Lavf53.20.0 Stream #0:0(): Video: h264 (![0][0][0] / 0x0021), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=2-31, 364 kb/s, 15 fps, 15 tbn, 15 tbc Metadata: creation_time : 2011-10-27 08:00:00 handler_name : VideoHandler Stream #0:1(eng): Audio: aac (@[0][0][0] / 0x0040), 22050 Hz, stereo, 47 kb/s Metadata: creation_time : 2011-10-27 08:00:00 handler_name : Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 51 fps= 0 q=-1.0 Lsize= 173kB time=00:00:03.34 bitrate= 423.3kbits/s video:151kB audio:19kB global headers:0kB muxing overhead 1.273083% _______________________________ Arlo Leach 773.769.6106 http://arlomedia.com From jsd at cluttered.com Thu Nov 10 01:00:58 2011 From: jsd at cluttered.com (Jon Drukman) Date: Thu, 10 Nov 2011 00:00:58 +0000 (UTC) Subject: [FFmpeg-user] Transcoding FLV with no/broken audio Message-ID: I'm using Red5 (open source Flash Media Server) to record videos from webcams. Sometimes the files come through with no audio. I want to transcode these files to H.264/AAC, but if there's no audio, I'd like to just leave it out. I can identify these bad input files by looking at ffmpeg's output: Input #0, flv, from 'webcam.flv': Metadata: audiocodecid : -1 duration : 5 videocodecid : 2 canSeekToEnd : true Duration: 00:00:04.55, start: 0.000000, bitrate: N/A Stream #0.0: Video: flv, yuv420p, 320x240, 1k tbr, 1k tbn, 1k tbc Stream #0.1: Audio: [0][0][0][0] / 0x0000, 0 channels No problem. Audio [0][0][0][0] means no audio. But how do I specify that I don't want any audio in the output? My normal command line is: ffmpeg -v 0 -y -i webcam.flv -s 320x240 -acodec libfaac -ab 64k -ac 2 -ar 44100 -vcodec libx264 -preset fast -b 300k -bt 300k -threads 0 -vsync 0 -async 1 webcam.flv.mp4 This fails with: Decoder (codec id 0) not found for input stream #0.1 I tried leaving off all the -acodec, -ab, -ac, -ar options. This fails with: [libfaac @ 0x7fa28c01b600] encoding 0 channel(s) is not allowed Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height I tried -acodec copy [mp4 @ 0x7fe5fb818c00] sample rate not set Could not write header for output file #0 (incorrect codec parameters ?) I tried -acodec null, -acodec none... That just gives me "unknown encoder". Any ideas how I can tell it to just ignore the audio entirely? From lou at lrcd.com Thu Nov 10 01:12:50 2011 From: lou at lrcd.com (Lou) Date: Wed, 9 Nov 2011 15:12:50 -0900 Subject: [FFmpeg-user] Transcoding FLV with no/broken audio In-Reply-To: References: Message-ID: <20111109151250.22157dbd@lrcd.com> On Thu, 10 Nov 2011 00:00:58 +0000 (UTC) Jon Drukman wrote: > I'm using Red5 (open source Flash Media Server) to record videos from > webcams. Sometimes the files come through with no audio. I want to > transcode these files to H.264/AAC, but if there's no audio, I'd like > to just leave it out. > > I can identify these bad input files by looking at ffmpeg's output: > > Input #0, flv, from 'webcam.flv': > Metadata: > audiocodecid : -1 > duration : 5 > videocodecid : 2 > canSeekToEnd : true > Duration: 00:00:04.55, start: 0.000000, bitrate: N/A > Stream #0.0: Video: flv, yuv420p, 320x240, 1k tbr, 1k tbn, 1k tbc > Stream #0.1: Audio: [0][0][0][0] / 0x0000, 0 channels > > No problem. Audio [0][0][0][0] means no audio. But how do I specify > that I don't want any audio in the output? My normal command line is: > > ffmpeg -v 0 -y -i webcam.flv -s 320x240 -acodec libfaac -ab 64k -ac 2 > -ar 44100 -vcodec libx264 -preset fast -b 300k -bt 300k -threads 0 > -vsync 0 -async 1 webcam.flv.mp4 > > This fails with: > > Decoder (codec id 0) not found for input stream #0.1 > > I tried leaving off all the -acodec, -ab, -ac, -ar options. This > fails with: > > [libfaac @ 0x7fa28c01b600] encoding 0 channel(s) is not allowed > Error while opening encoder for output stream #0.1 - maybe incorrect > parameters such as bit_rate, rate, width or height > > I tried -acodec copy > > [mp4 @ 0x7fe5fb818c00] sample rate not set > Could not write header for output file #0 (incorrect codec > parameters ?) > > I tried -acodec null, -acodec none... That just gives me "unknown > encoder". > > Any ideas how I can tell it to just ignore the audio entirely? Try the -an option. From jsd at cluttered.com Thu Nov 10 01:13:58 2011 From: jsd at cluttered.com (Jon Drukman) Date: Thu, 10 Nov 2011 00:13:58 +0000 (UTC) Subject: [FFmpeg-user] Transcoding FLV with no/broken audio References: Message-ID: Jon Drukman cluttered.com> writes: > No problem. Audio [0][0][0][0] means no audio. But how do I specify that I > don't want any audio in the output? My normal command line is: Of course 10 seconds after posting I figure it out, it's -an From rickcorteza at gmail.com Thu Nov 10 01:43:58 2011 From: rickcorteza at gmail.com (Rick C.) Date: Thu, 10 Nov 2011 08:43:58 +0800 Subject: [FFmpeg-user] audio streams In-Reply-To: References: Message-ID: <61BE060D-F128-48F0-B654-EA8D2576620C@gmail.com> On Nov 9, 2011, at 8:55 PM, Carl Eugen Hoyos wrote: > Rick C. gmail.com> writes: > >> I have noticed that taking .vob files from a DVD and converting them to let's >> say .mp4 (ffmpeg -i input.vob output.mp4) will often grab the wrong audio >> track and the converted .mp4 will end up with a narrative track or something >> but not the expected default audio track. What is the best way to address >> this issue? > > (Complete, uncut output missing.) > > The -map option allows you to choose which stream(s) of the input file should be > encoded. > > Carl Eugen > Sorry here's the output below. I guess there's no choice but to use the -map option in this case? Last login: Thu Nov 10 08:37:18 on ttys001 mahalkos-MacBook:~ mahalko$ /Users/mahalko/Desktop/ffmpeg -i /Users/mahalko/Desktop/VTS_04_1.VOB -t 180 -qscale 10 /Users/mahalko/Desktop/test.mp4 ffmpeg version 0.8.6, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 5 2011 10:59:17 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) 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 --arch=x86_64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0x101807c00] max_analyze_duration 5000000 reached at 5000000 Input #0, mpeg, from '/Users/mahalko/Desktop/VTS_04_1.VOB': Duration: 00:23:56.44, start: 0.203422, bitrate: 5979 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 64:45 DAR 16:9], 9800 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x20]: Subtitle: dvdsub Stream #0.2[0x21]: Subtitle: dvdsub Stream #0.3[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s Stream #0.4[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s [buffer @ 0x101413b20] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:64/45 sws_param: Output #0, mp4, to '/Users/mahalko/Desktop/test.mp4': Metadata: encoder : Lavf53.4.0 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: aac, 48000 Hz, 5.1, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.4 -> #0.1 Press [q] to stop, [?] for help frame= 4502 fps=116 q=10.0 Lsize= 13450kB time=00:03:00.01 bitrate= 612.1kbits/s video:12043kB audio:1303kB global headers:0kB muxing overhead 0.777514% mahalkos-MacBook:~ mahalko$ From lytithwyn at gmail.com Thu Nov 10 15:57:03 2011 From: lytithwyn at gmail.com (Matthew Morgan) Date: Thu, 10 Nov 2011 09:57:03 -0500 Subject: [FFmpeg-user] fixing videos that don't start with keyframes Message-ID: <4EBBE63F.8010804@gmail.com> I have some videos that were generated by a Sony Handicam. These videos don't start with keyframes and that *really* messes with the output of programs like DeVeDe. Is there an easy way to re-encode these files such that the first frame is forced to be a keyframe? I tried using -force-key-frames 00:00:00 (and I also tried just 0) but that just causes ffmpeg to fail with a failed assertion. If it helps, here's the info on one of the files: Input #0, mpeg, from 'M2U00107.MPG': Duration: 00:30:19.79, start: 0.113344, bitrate: 9316 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [PAR 8:9 DAR 4:3], 9100 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s At least one output file must be specified Also, here's my ffmpeg version info in case that makes a difference: ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav developers built on Oct 2 2011 15:12:32 with gcc 4.6.1 From cehoyos at ag.or.at Thu Nov 10 16:02:59 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Nov 2011 15:02:59 +0000 (UTC) Subject: [FFmpeg-user] Decode h264 width no start code error References: Message-ID: Tuan DN anlab.vn> writes: > Attachment (output.mp4): video/mp4, 258 KiB Which software plays this file? (It does not play with WMP12, your original sample does.) Carl Eugen From cehoyos at ag.or.at Thu Nov 10 16:11:49 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Nov 2011 15:11:49 +0000 (UTC) Subject: [FFmpeg-user] trouble with -vcodec copy References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> Message-ID: Arlo Leach arlomedia.com> writes: > Unfortunately I'm still getting the same result -- no video track if I use > -vcodec copy. > > Actually, I've been testing with a short source video (15 seconds) and > extracted sample (3-4 seconds). I just tried with a longer source video (180 > seconds) and sample (30 seconds) and my sample clip has a video track, but > the first six seconds of it are blank (I just see the background color of the > video player). The audio plays correctly from the start, but the video > doesn't begin until six seconds into the clip. Could you upload a sample to http://www.datafilehost.com/ and post the download link here? You may have found a new feature that allows libavcodec not to output initial H264 frames that cannot be decoded (because the stream does not start at gop boundary), but that cannot be the problem you have seen with the old FFmpeg version. Carl Eugen From cehoyos at ag.or.at Thu Nov 10 17:12:03 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Nov 2011 16:12:03 +0000 (UTC) Subject: [FFmpeg-user] fixing videos that don't start with keyframes References: <4EBBE63F.8010804@gmail.com> Message-ID: Matthew Morgan gmail.com> writes: > ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav > developers built on Oct 2 2011 That is an intentionally broken (and unsupported) version. Please see http://ffmpeg.org/download.html Carl Eugen From lytithwyn at gmail.com Thu Nov 10 17:17:13 2011 From: lytithwyn at gmail.com (Matthew Morgan) Date: Thu, 10 Nov 2011 11:17:13 -0500 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> Message-ID: <4EBBF909.9040508@gmail.com> On Thu 10 Nov 2011 11:12:03 AM EST, Carl Eugen Hoyos wrote: > Matthew Morgan gmail.com> writes: > >> ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav >> developers built on Oct 2 2011 > > That is an intentionally broken (and unsupported) version. > Please see http://ffmpeg.org/download.html > > Carl Eugen I understand the vendor-customized repo versions can't be supported, but I don't even know what command line options would be required to repair the files. I actually already have a set of scripts for quickly building and setting up the source version of ffmpeg if the repo version doesn't do what I need. I mainly just need to know if it is possible using ffmpeg (and if so, how) to trim everything off the beginning of a video file up to the first keyframe. From cehoyos at ag.or.at Thu Nov 10 17:40:03 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Nov 2011 16:40:03 +0000 (UTC) Subject: [FFmpeg-user] fixing videos that don't start with keyframes References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> Message-ID: Matthew Morgan gmail.com> writes: > I mainly just need to know if it is possible using ffmpeg (and if so, > how) to trim everything off the beginning of a video file up to the > first keyframe. This is nowadays FFmpeg's default. The reason this was changed where many reports like yours, I am not sure if the discussion is completely over, but for the moment FFmpeg tries to skip the initial non-key frames. There probably are videos for which this does not work, but testing (and reporting) about this is very welcome! Please consider opening a report on http://avcodec.org/trac/ffmpeg if your problem is reproducible with current git head. Carl Eugen From lists at arlomedia.com Thu Nov 10 18:00:57 2011 From: lists at arlomedia.com (Arlo Leach) Date: Thu, 10 Nov 2011 09:00:57 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> Message-ID: Hello, > Could you upload a sample to http://www.datafilehost.com/ and post the download > link here? > > You may have found a new feature that allows libavcodec not to output initial > H264 frames that cannot be decoded (because the stream does not start at gop > boundary), but that cannot be the problem you have seen with the old FFmpeg > version. I just put them on my site because one of the files is larger than 100 MB. I also did some more tests and found that the ability to extract good a sample clip depends on the video source and is not affected by my first command that reformats the video. 1a) This is a video I encoded using Handbrake: http://www.arlomedia.com/projects/ffmpeg/band.mp4 1b) This is the same clip reformatted with ffmpeg: http://www.arlomedia.com/projects/ffmpeg/band_formatted.mp4 1c) And a shorter clip extracted from that (missing first six seconds): http://www.arlomedia.com/projects/ffmpeg/band_formatted_sample.mp4 1d) However, I also lose some of the video if I extract from the original clip, so apparently my reformatting is not causing the problem: http://www.arlomedia.com/projects/ffmpeg/band_sample.mp4 2a) As another test, this is a trailer downloaded from apple.com/trailers: http://www.arlomedia.com/projects/ffmpeg/trailer.mov 2b) I can extract a sample clip from that with no problems: http://www.arlomedia.com/projects/ffmpeg/trailer_sample.mp4 2c) Or reformat it and then extract a sample clip with no problems: http://www.arlomedia.com/projects/ffmpeg/trailer_formatted_sample.mp4 So now the questions are, what is it about the first clip that causes some of its video to be lost when extracting a sample, and can I fix that when I reformat the original clip? Thanks, -Arlo _______________________________ Arlo Leach http://arlomedia.com From lytithwyn at gmail.com Thu Nov 10 18:15:09 2011 From: lytithwyn at gmail.com (Matthew Morgan) Date: Thu, 10 Nov 2011 12:15:09 -0500 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> Message-ID: <4EBC069D.6000602@gmail.com> On Thu 10 Nov 2011 11:40:03 AM EST, Carl Eugen Hoyos wrote: > Matthew Morgan gmail.com> writes: > >> I mainly just need to know if it is possible using ffmpeg (and if so, >> how) to trim everything off the beginning of a video file up to the >> first keyframe. > > This is nowadays FFmpeg's default. > The reason this was changed where many reports like yours, I am not sure if the > discussion is completely over, but for the moment FFmpeg tries to skip the > initial non-key frames. > There probably are videos for which this does not work, but testing (and > reporting) about this is very welcome! Please consider opening a report on > http://avcodec.org/trac/ffmpeg if your problem is reproducible with current git > head. > > Carl Eugen So ffmpeg skips them by default? That's actually really great news for me. Does this behavior happen with '-acodec copy -vcodec' copy as well? My idea is to just run the file through an ffmpeg conversion and let it trim the initial non-key frames off. I'll going ahead and build from git-head and file a report if it doesn't work. From cehoyos at ag.or.at Thu Nov 10 18:34:53 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 10 Nov 2011 17:34:53 +0000 (UTC) Subject: [FFmpeg-user] trouble with -vcodec copy References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> Message-ID: Arlo Leach arlomedia.com> writes: > 1a) This is a video I encoded using Handbrake: > http://www.arlomedia.com/projects/ffmpeg/band.mp4 I can reproduce the problem with -vcodec copy -acodec copy -ss 1 you reported. Is there a sample "before" this one? I cannot reproduce the problem with band_formatted.mp4 Carl Eugen From lists at arlomedia.com Thu Nov 10 18:52:36 2011 From: lists at arlomedia.com (Arlo Leach) Date: Thu, 10 Nov 2011 09:52:36 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> Message-ID: <4ED92E2E-0A29-43F9-9954-762F95151217@arlomedia.com> Hello, > I can reproduce the problem with -vcodec copy -acodec copy -ss 1 you reported. > I cannot reproduce the problem with band_formatted.mp4 Hmm ... I just ran this command: ffmpeg -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -ss 30 -t 30 -y band_formatted_sample_again.mp4 And got this result: http://www.arlomedia.com/projects/ffmpeg/band_formatted_sample_again.mp4 In QuickTime Player, I see no video for the first three seconds. This is with ffmpeg version N-34622-g701e534, downloaded yesterday. > Is there a sample "before" this one? Do you mean the source clip that I had originally encoded with Handbrake? That would be a .mov file exported from Final Cut Express. I can dig that up if you'd like to see it. Thanks, -Arlo _______________________________ Arlo Leach http://arlomedia.com From alex.zhen.ma at gmail.com Thu Nov 10 21:04:21 2011 From: alex.zhen.ma at gmail.com (Zhen Ma) Date: Thu, 10 Nov 2011 15:04:21 -0500 Subject: [FFmpeg-user] How do I fix the warning "Invalid and inefficient vfw-avi packed B frames detected" while converting an avi into 3gp by using h263 Message-ID: Hey guys, Is there someone knows how to fix that problem? ffmpeg -y -i sample1.avi -f 3gp -s 176x144 -aspect 1.8 -r 15 -vcodec h263 -b 150k -bt 100k -an ph__140.3gp Thanks! -- Your sincerely ! ------------------------------------------------------- Alex Zhen Ma ? Email: alex.zhen.ma at gmail.com ? Messenger: coolface8 at hotmail.com From redneb8888 at gmail.com Fri Nov 11 03:12:04 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Thu, 10 Nov 2011 21:12:04 -0500 Subject: [FFmpeg-user] -map and matroska files Message-ID: Hi all, I have a video file with multiple audio streams and I want to select one of them. I did ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.avi to select streams 0 (the video) and 2 (the desired audio stream) and this works fine. But if I try to output to matroska, ie ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.mkv this fails with the following error: Number of stream maps must match number of output streams Why is this happening? From tuan.dn at anlab.vn Fri Nov 11 03:15:55 2011 From: tuan.dn at anlab.vn (Tuan DN) Date: Fri, 11 Nov 2011 09:15:55 +0700 Subject: [FFmpeg-user] Decode h264 width no start code error In-Reply-To: References: Message-ID: Hi Carl, It can be played with both WMP and VLC But its duration is too small Tuan On Thu, Nov 10, 2011 at 10:02 PM, Carl Eugen Hoyos wrote: > Tuan DN anlab.vn> writes: > > > Attachment (output.mp4): video/mp4, 258 KiB > > Which software plays this file? > (It does not play with WMP12, your original sample does.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From james.darnley at gmail.com Fri Nov 11 03:19:53 2011 From: james.darnley at gmail.com (James Darnley) Date: Fri, 11 Nov 2011 03:19:53 +0100 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: Message-ID: <4EBC8649.80407@gmail.com> On 2011-11-11 03:12, Marios Titas wrote: > Hi all, > > I have a video file with multiple audio streams and I want to select > one of them. I did > > ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.avi > > to select streams 0 (the video) and 2 (the desired audio stream) and > this works fine. But if I try to output to matroska, ie > > ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.mkv > > this fails with the following error: > > Number of stream maps must match number of output streams > > Why is this happening? Without the uncut output, we can never really know, but a guess leads me to think that you are not accounting for subtitles. From redneb8888 at gmail.com Fri Nov 11 03:54:25 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Thu, 10 Nov 2011 21:54:25 -0500 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: <4EBC8649.80407@gmail.com> References: <4EBC8649.80407@gmail.com> Message-ID: You are right, adding the -sn option solved the problem. This doesn't make much sense to me though. I think it would be more reasonable for ffmpeg to assume that I don't want the subtitles since I did not explicitly included them in my stream mappings. It does do that for the other audio streams, why not for the subtitles too? On Thu, Nov 10, 2011 at 21:19, James Darnley wrote: > On 2011-11-11 03:12, Marios Titas wrote: >> >> Hi all, >> >> I have a video file with multiple audio streams and I want to select >> one of them. I did >> >> ? ? ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.avi >> >> to select streams 0 (the video) and 2 (the desired audio stream) and >> this works fine. But if I try to output to matroska, ie >> >> ? ? ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.mkv >> >> this fails with the following error: >> >> ? ? Number of stream maps must match number of output streams >> >> Why is this happening? > > Without the uncut output, we can never really know, but a guess leads me to > think that you are not accounting for subtitles. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From belcampo at zonnet.nl Fri Nov 11 09:34:20 2011 From: belcampo at zonnet.nl (belcampo) Date: Fri, 11 Nov 2011 09:34:20 +0100 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: <4EBC8649.80407@gmail.com> Message-ID: <4EBCDE0C.8050802@zonnet.nl> On 11/11/2011 03:54 AM, Marios Titas wrote: > You are right, adding the -sn option solved the problem. This doesn't > make much sense to me though. I think it would be more reasonable for > ffmpeg to assume that I don't want the subtitles since I did not > explicitly included them in my stream mappings. It does do that for > the other audio streams, why not for the subtitles too? As .avi cannot contain subs it's automatically excluded. But other containers that can contain subs you have to explicitely define or exclude them. > > > On Thu, Nov 10, 2011 at 21:19, James Darnley wrote: >> On 2011-11-11 03:12, Marios Titas wrote: >>> Hi all, >>> >>> I have a video file with multiple audio streams and I want to select >>> one of them. I did >>> >>> ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.avi >>> >>> to select streams 0 (the video) and 2 (the desired audio stream) and >>> this works fine. But if I try to output to matroska, ie >>> >>> ffmpeg -i input.mkv -map 0.0 -map 0.2 -y output.mkv >>> >>> this fails with the following error: >>> >>> Number of stream maps must match number of output streams >>> >>> Why is this happening? >> Without the uncut output, we can never really know, but a guess leads me to >> think that you are not accounting for subtitles. >> _______________________________________________ >> 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 cehoyos at ag.or.at Fri Nov 11 11:12:23 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Nov 2011 10:12:23 +0000 (UTC) Subject: [FFmpeg-user] -map and matroska files References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: belcampo zonnet.nl> writes: > On 11/11/2011 03:54 AM, Marios Titas wrote: > > You are right, adding the -sn option solved the problem. This doesn't > > make much sense to me though. I think it would be more reasonable for > > ffmpeg to assume that I don't want the subtitles since I did not > > explicitly included them in my stream mappings. And this is exactly how (current) ffmpeg works. > As .avi cannot contain subs it's automatically excluded. I don't think that is correct (XSUB). Carl Eugen From cehoyos at ag.or.at Fri Nov 11 11:13:51 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Nov 2011 10:13:51 +0000 (UTC) Subject: [FFmpeg-user] How do I fix the warning "Invalid and inefficient vfw-avi packed B frames detected" while converting an avi into 3gp by using h263 References: Message-ID: Zhen Ma gmail.com> writes: > Is there someone knows how to fix that problem? (Complete, uncut output missing.) You can fix (=change) your encoder, the warning should inform you that your input file was encoded invalid and inefficiently. Carl Eugen From marc at hallmarcwebsites.com Fri Nov 11 14:24:54 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 11 Nov 2011 08:24:54 -0500 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg Message-ID: Anyone know why QuickTime would report a different width for an .mov than ffmpeg does? From dave.bevan at bbc.co.uk Fri Nov 11 16:12:51 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Fri, 11 Nov 2011 15:12:51 -0000 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg References: Message-ID: >Anyone know why QuickTime would report a different width for an .mov than ffmpeg does? Check aspect ratio flags (DAR/PAR etc). If I remember correctly, QT reports, for example, 1024x576 for anamorphic 720x576. http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3100 bytes Desc: not available URL: From marc at hallmarcwebsites.com Fri Nov 11 16:31:48 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 11 Nov 2011 10:31:48 -0500 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg In-Reply-To: References: Message-ID: > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Dave Bevan > Sent: Friday, November 11, 2011 10:13 AM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg > > >Anyone know why QuickTime would report a different width for an .mov > than ffmpeg does? > > Check aspect ratio flags (DAR/PAR etc). If I remember correctly, QT reports, > for example, 1024x576 for anamorphic 720x576. [>] Stream #0.0(eng): Video: svq3, yuvj420p, 648x486, 1577 kb/s, SAR 43185:32768 DAR 14395:8192, 29.91 fps, 29.95 tbr, 29954 tbn, 29954 tbc the wxh shows ~4:3 SAR is also ~4:3 yet DAR is ~16:9 the actual video plays at ~16:9 or 854x486 the only dimension reported here evenly div by 4 is 648 which makes me think ffmpeg isn't so much reading the dimensions as it is calculating them? Still learning From cehoyos at ag.or.at Fri Nov 11 17:39:22 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Nov 2011 16:39:22 +0000 (UTC) Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > Anyone know why QuickTime would report a different width for an .mov than > ffmpeg does? (Complete, uncut console output missing.) QuickTime reports the dimension corrected by cropping set in the container (mov), the line you are referring to in FFmpeg's output reports the codec dimensions that may be different. Additionally, FFmpeg does not support cropping set in the mov container;-( In the case of DAR/SAR, mentioned line still reports the encoded dimensions that may have to be corrected when displayed (if you find bugs, please report them), I suppose QuickTime always reports display dimensions. Carl Eugen From dave.bevan at bbc.co.uk Fri Nov 11 18:09:04 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Fri, 11 Nov 2011 17:09:04 -0000 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg References: Message-ID: >> >Anyone know why QuickTime would report a different width for an .mov >> than ffmpeg does? >> >> Check aspect ratio flags (DAR/PAR etc). If I remember correctly, QT reports, > for example, 1024x576 for anamorphic 720x576. >Stream #0.0(eng): Video: svq3, yuvj420p, 648x486, 1577 kb/s, SAR 43185:32768 >DAR 14395:8192, 29.91 fps, 29.95 tbr, 29954 tbn, 29954 tbc >the wxh shows ~4:3 SAR is also ~4:3 yet DAR is ~16:9 >the actual video plays at ~16:9 or 854x486 >the only dimension reported here evenly div by 4 is 648 which makes me think >ffmpeg isn't so much reading the dimensions as it is calculating them? >Still learning So your DAR (display aspect ratio) is 14395:8192. Take that with your HEIGHT, and you get: (14395 / 8192) * 486 = 854.000244.... So, it's working exactly as expected, with QT doing exactly what's asked of it - to take 648x486 video and display using the DAR defined in the wrapper, giving you 854x486. Basically, video is stored in square pixels - in your case 648x486 - so 314928 pixels per frame. Pixels can have DAR applied and it's down to display devices to STRETCH the original video WIDTH at the DAR ratio to give the final output. In European TV land, all [digital] video of Standard Definition is transported down video pipes that are 720x576 in size. Further, there is padding applied which actually means the "active picture" is contained in 702x576 pixels, so 9px padding on left/right. When WIDESCREEN was introduced, none of the physical equipment changed. Simplistically, all that occurred was flagging of video with a 16:9 flag. It meant that when a camera captured video, it took in a "viewport" that was 16:9 aspect ratio and scaled/squashed the horizontal pixels seen into the 702x576 box. Then the video travelled around all it's pipes, vision mixers etc, eventually to TV transmitters into your house and into your TV, all squashed into the 702x576 box. A side-band flag tells your TV to stretch the video picture back to be shown to you as widescreen. That "display" 4:3, 16:9 button on your TV remote is simply overriding the signal from the TV station saying the picture should be shown in widescreen, vs the old 4:3 aspect ratio. Many people thought that the introduction of 16:9 meant better pixels, but in fact it means less quality, because the width of what you see is being stretched more, rather than less. In HD, things are different of course - the signal [usually] transported to your TV is 1920x1080, though it can be 1440x1080, 1280x720, or even 960x720. --Dave. http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4788 bytes Desc: not available URL: From marc at hallmarcwebsites.com Fri Nov 11 18:38:22 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 11 Nov 2011 12:38:22 -0500 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg In-Reply-To: References: Message-ID: > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos > Sent: Friday, November 11, 2011 11:39 AM > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg > > HallMarc Websites hallmarcwebsites.com> writes: > > > Anyone know why QuickTime would report a different width for an .mov > > than ffmpeg does? > > (Complete, uncut console output missing.) > > QuickTime reports the dimension corrected by cropping set in the container > (mov), the line you are referring to in FFmpeg's output reports the codec > dimensions that may be different. > Additionally, FFmpeg does not support cropping set in the mov container;-( > > In the case of DAR/SAR, mentioned line still reports the encoded dimensions > that may have to be corrected when displayed (if you find bugs, please > report them), I suppose QuickTime always reports display dimensions. > > Carl Eugen > Um could you put that into human readable form? Where are you seeing cropping? QuickTime plays this at the correct ratio 16:9. It is ffmpeg that is going by the SAR instead of the DAR. So I guess my next question is should we always follow the ar of the DAR because it is the actual ar of the video or...? the whole output : /$ ffmpeg -i /home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/open30_ctm1.mov -vstats 2>&1 ffmpeg version N-31911-g1a34478, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 15 2011 17:30:02 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50) configuration: --prefix=/usr/local/hgffmpeg --enable-shared --enable-nonfree --enable-avfilter --enable-filter=movie --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libxvid --extra-cflags=-I/usr/local/hgffmpeg/include/ --extra-ldflags=-L/usr/local/hgffmpeg/lib --enable-decoder=ac3 --enable-decoder=asv1 --enable-decoder=asv2 --enable-decoder=flac --enable-decoder=wmv1 --enable-decoder=wmv2 --enable-decoder=wmv3 --enable-decoder=mpeg1video --enable-decoder=mpeg2video --enable-decoder=flv --enable-decoder=fraps --enable-decoder=h263 --enable-decoder=h264 --enable-decoder=libgsm --enable-decoder=mjpeg --enable-decoder=mpeg4 --enable-decoder=mpeg4aac --enable-decoder=mpegvideo --enable-decoder=mpeg4aac --enable-decoder=msmpeg4v1 --enable-decoder=msmpeg4v2 --enable-decoder=msmpeg4v3 --enable-decoder=pcm_alaw --enable-decoder=pcm_mulaw --enable-encoder=ac3 --enable-encoder=asv1 --enable-encoder=asv2 --enable-encoder=flac --enable-encoder=h263 --enable-encoder=flashsv --enable-encoder=flv --enable-encoder=libgsm --enable-encoder=mjpeg --enable-encoder=msmpeg4v3 --enable-encoder=pcm_alaw --enable-encoder=pcm_mulaw --enable-encoder=mpeg1video --enable-encoder=mpeg2video --enable-encoder=mpeg4 --enable-encoder=msmpeg4v1 --enable-encoder=msmpeg4v2 --enable-encoder=rv10 --enable-encoder=rv20 --enable-encoder=vorbis --enable-encoder=wmav1 --enable-encoder=wmav2 --enable-encoder=wmv1 --enable-encoder=wmv2 --disable-demuxer=v4l --disable-demuxer=v4l2 --enable-version3 --disable-decoder=vp8 libavutil 51. 12. 0 / 51. 12. 0 libavcodec 53. 10. 0 / 53. 10. 0 libavformat 53. 7. 0 / 53. 7. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 31. 1 / 2. 31. 1 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 29954.00 (29954/1) -> 29.95 (14977/500) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/open30_ctm1.mov' : Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2008-03-11 17:40:41 encoder : Sorenson Squeeze encoder-eng : Sorenson Squeeze Duration: 00:00:30.45, start: 0.000000, bitrate: 1709 kb/s Stream #0.0(eng): Video: svq3, yuvj420p, 648x486, 1577 kb/s, SAR 43185:32768 DAR 14395:8192, 29.91 fps, 29.95 tbr, 29954 tbn, 29954 tbc Metadata: creation_time : 2008-03-11 17:40:41 Stream #0.1(eng): Audio: mp3, 44100 Hz, 1 channels, s16, 128 kb/s Metadata: creation_time : 2008-03-11 17:40:41 At least one output file must be specified /$ ffmpeg -i /home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/open30_ctm1.mov -vstats 2>&1 ffmpeg version N-31911-g1a34478, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 15 2011 17:30:02 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50) configuration: --prefix=/usr/local/hgffmpeg --enable-shared --enable-nonfree --enable-avfilter --enable-filter=movie --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libxvid --extra-cflags=-I/usr/local/hgffmpeg/include/ --extra-ldflags=-L/usr/local/hgffmpeg/lib --enable-decoder=ac3 --enable-decoder=asv1 --enable-decoder=asv2 --enable-decoder=flac --enable-decoder=wmv1 --enable-decoder=wmv2 --enable-decoder=wmv3 --enable-decoder=mpeg1video --enable-decoder=mpeg2video --enable-decoder=flv --enable-decoder=fraps --enable-decoder=h263 --enable-decoder=h264 --enable-decoder=libgsm --enable-decoder=mjpeg --enable-decoder=mpeg4 --enable-decoder=mpeg4aac --enable-decoder=mpegvideo --enable-decoder=mpeg4aac --enable-decoder=msmpeg4v1 --enable-decoder=msmpeg4v2 --enable-decoder=msmpeg4v3 --enable-decoder=pcm_alaw --enable-decoder=pcm_mulaw --enable-encoder=ac3 --enable-encoder=asv1 --enable-encoder=asv2 --enable-encoder=flac --enable-encoder=h263 --enable-encoder=flashsv --enable-encoder=flv --enable-encoder=libgsm --enable-encoder=mjpeg --enable-encoder=msmpeg4v3 --enable-encoder=pcm_alaw --enable-encoder=pcm_mulaw --enable-encoder=mpeg1video --enable-encoder=mpeg2video --enable-encoder=mpeg4 --enable-encoder=msmpeg4v1 --enable-encoder=msmpeg4v2 --enable-encoder=rv10 --enable-encoder=rv20 --enable-encoder=vorbis --enable-encoder=wmav1 --enable-encoder=wmav2 --enable-encoder=wmv1 --enable-encoder=wmv2 --disable-demuxer=v4l --disable-demuxer=v4l2 --enable-version3 --disable-decoder=vp8 libavutil 51. 12. 0 / 51. 12. 0 libavcodec 53. 10. 0 / 53. 10. 0 libavformat 53. 7. 0 / 53. 7. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 31. 1 / 2. 31. 1 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 29954.00 (29954/1) -> 29.95 (14977/500) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/open30_ctm1.mov' : Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2008-03-11 17:40:41 encoder : Sorenson Squeeze encoder-eng : Sorenson Squeeze Duration: 00:00:30.45, start: 0.000000, bitrate: 1709 kb/s Stream #0.0(eng): Video: svq3, yuvj420p, 648x486, 1577 kb/s, SAR 43185:32768 DAR 14395:8192, 29.91 fps, 29.95 tbr, 29954 tbn, 29954 tbc Metadata: creation_time : 2008-03-11 17:40:41 Stream #0.1(eng): Audio: mp3, 44100 Hz, 1 channels, s16, 128 kb/s Metadata: creation_time : 2008-03-11 17:40:41 From marc at hallmarcwebsites.com Fri Nov 11 18:48:59 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 11 Nov 2011 12:48:59 -0500 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg In-Reply-To: References: Message-ID: > So your DAR (display aspect ratio) is 14395:8192. Take that with your HEIGHT, > and you get: > > (14395 / 8192) * 486 = 854.000244.... > > So, it's working exactly as expected, with QT doing exactly what's asked of it - > to take 648x486 video and display using the DAR defined in the wrapper, > giving you 854x486. > > Basically, video is stored in square pixels - in your case 648x486 - so 314928 > pixels per frame. Pixels can have DAR applied and it's down to display devices > to STRETCH the original video WIDTH at the DAR ratio to give the final output. > > In European TV land, all [digital] video of Standard Definition is transported > down video pipes that are 720x576 in size. Further, there is padding applied > which actually means the "active picture" is contained in 702x576 pixels, so > 9px padding on left/right. When WIDESCREEN was introduced, none of the > physical equipment changed. Simplistically, all that occurred was flagging of > video with a 16:9 flag. It meant that when a camera captured video, it took in > a "viewport" that was 16:9 aspect ratio and scaled/squashed the horizontal > pixels seen into the 702x576 box. Then the video travelled around all it's > pipes, vision mixers etc, eventually to TV transmitters into your house and > into your TV, all squashed into the 702x576 box. A side-band flag tells your TV > to stretch the video picture back to be shown to you as widescreen. That > "display" 4:3, 16:9 button on your TV remote is simply overriding the signal > from the TV station saying the picture should be shown in widescreen, vs the > old 4:3 aspect ratio. > > Many people thought that the introduction of 16:9 meant better pixels, but > in fact it means less quality, because the width of what you see is being > stretched more, rather than less. > > In HD, things are different of course - the signal [usually] transported to your > TV is 1920x1080, though it can be 1440x1080, 1280x720, or even 960x720. > > --Dave. > > [>] Leave it to an actual Englishman to speak English. Thank you for that! So I guess my question now is thus; why is ffmpeg not applying this when getting the video dimensions needed for playback? Why is it giving me dimensions that squish the video if I were to use them? And yes I will write the PHP code to check and correct. Thank you for the explanation. From marc at hallmarcwebsites.com Fri Nov 11 19:17:35 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 11 Nov 2011 13:17:35 -0500 Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg In-Reply-To: References: Message-ID: > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Dave Bevan > Sent: Friday, November 11, 2011 12:09 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg > > >> >Anyone know why QuickTime would report a different width for an .mov > >> than ffmpeg does? > >> > >> Check aspect ratio flags (DAR/PAR etc). If I remember correctly, QT > >> reports, > > for example, 1024x576 for anamorphic 720x576. > >Stream #0.0(eng): Video: svq3, yuvj420p, 648x486, 1577 kb/s, SAR > >43185:32768 DAR 14395:8192, 29.91 fps, 29.95 tbr, 29954 tbn, 29954 tbc > >the wxh shows ~4:3 SAR is also ~4:3 yet DAR is ~16:9 the actual video > >plays at ~16:9 or 854x486 the only dimension reported here evenly div > >by 4 is 648 which makes me think ffmpeg isn't so much reading the > >dimensions as it is calculating them? > >Still learning > > So your DAR (display aspect ratio) is 14395:8192. Take that with your HEIGHT, > and you get: > > (14395 / 8192) * 486 = 854.000244.... > > So, it's working exactly as expected, with QT doing exactly what's asked of it - > to take 648x486 video and display using the DAR defined in the wrapper, > giving you 854x486. > > Basically, video is stored in square pixels - in your case 648x486 - so 314928 > pixels per frame. Pixels can have DAR applied and it's down to display devices > to STRETCH the original video WIDTH at the DAR ratio to give the final output. > > In European TV land, all [digital] video of Standard Definition is transported > down video pipes that are 720x576 in size. Further, there is padding applied > which actually means the "active picture" is contained in 702x576 pixels, so > 9px padding on left/right. When WIDESCREEN was introduced, none of the > physical equipment changed. Simplistically, all that occurred was flagging of > video with a 16:9 flag. It meant that when a camera captured video, it took in > a "viewport" that was 16:9 aspect ratio and scaled/squashed the horizontal > pixels seen into the 702x576 box. Then the video travelled around all it's > pipes, vision mixers etc, eventually to TV transmitters into your house and > into your TV, all squashed into the 702x576 box. A side-band flag tells your TV > to stretch the video picture back to be shown to you as widescreen. That > "display" 4:3, 16:9 button on your TV remote is simply overriding the signal > from the TV station saying the picture should be shown in widescreen, vs the > old 4:3 aspect ratio. > > Many people thought that the introduction of 16:9 meant better pixels, but > in fact it means less quality, because the width of what you see is being > stretched more, rather than less. > > In HD, things are different of course - the signal [usually] transported to your > TV is 1920x1080, though it can be 1440x1080, 1280x720, or even 960x720. > > --Dave. [>] Am I correct in assuming this is true for any container not just mov? From cehoyos at ag.or.at Fri Nov 11 19:51:24 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 11 Nov 2011 18:51:24 +0000 (UTC) Subject: [FFmpeg-user] Quicktime reports diiferent width than ffmpeg References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > > QuickTime reports the dimension corrected by cropping set in the container > > (mov), the line you are referring to in FFmpeg's output reports the codec > > dimensions that may be different. > > Additionally, FFmpeg does not support cropping set in the mov container;-( > > > > In the case of DAR/SAR, mentioned line still reports the encoded dimensions > > that may have to be corrected when displayed (if you find bugs, please > > report them), I suppose QuickTime always reports display dimensions. > > Um could you put that into human readable form? Where are you seeing > cropping? QuickTime plays this at the correct ratio 16:9. Sorry, I just wanted to enumerate some possible reasons why QuickTime shows another resolution than FFmpeg. Can you make your sample available? Carl Eugen From developer at noknok.net Fri Nov 11 19:52:45 2011 From: developer at noknok.net (NokNok Developer) Date: Fri, 11 Nov 2011 13:52:45 -0500 Subject: [FFmpeg-user] -RE option broken? Message-ID: <4EBD6EFD.5040405@noknok.net> Im using the latest version pull from git, and when using the -RE in my command line, its no longer processing at REALTIME speed? it used to be that if the movie was 20fps, the transcode with the -re option would transcode at 20fps... but here, it just runs wild.... Has something changed? -v info -y -i "http://1245:12345 at somedomain.com/11949/23571937" -threads 0 -re -acodec aac -ac 2 -ab 128000 -ar 44100 -vsync 2 -strict experimental -vcodec libx264 -r 18 -b 1024k -bt 1024k -bufsize 512k -maxrate 1229k -fpre "E:\Transcoder\tools\ffpresets\libx264-noknokfast.ffpreset" -fpre "E:\Transcoder\tools\ffpresets\libx264-main.ffpreset" -g 36 -keyint_min 18 -crf 18 -f mpegts -y "E:\Transcoder\work\d886eba0-9aa3-4f87-958c-9f5e1a972a4a.TS" From lytithwyn at gmail.com Fri Nov 11 22:05:15 2011 From: lytithwyn at gmail.com (Matthew Morgan) Date: Fri, 11 Nov 2011 16:05:15 -0500 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: <4EBC069D.6000602@gmail.com> References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> Message-ID: <4EBD8E0B.400@gmail.com> On Thu 10 Nov 2011 12:15:09 PM EST, Matthew Morgan wrote: > On Thu 10 Nov 2011 11:40:03 AM EST, Carl Eugen Hoyos wrote: >> Matthew Morgan gmail.com> writes: >> >>> I mainly just need to know if it is possible using ffmpeg (and if so, >>> how) to trim everything off the beginning of a video file up to the >>> first keyframe. >> >> This is nowadays FFmpeg's default. >> The reason this was changed where many reports like yours, I am not >> sure if the >> discussion is completely over, but for the moment FFmpeg tries to >> skip the >> initial non-key frames. >> There probably are videos for which this does not work, but testing (and >> reporting) about this is very welcome! Please consider opening a >> report on >> http://avcodec.org/trac/ffmpeg if your problem is reproducible with >> current git >> head. >> >> Carl Eugen > So ffmpeg skips them by default? That's actually really great news for > me. Does this behavior happen with '-acodec copy -vcodec' copy as well? > > My idea is to just run the file through an ffmpeg conversion and let > it trim the initial non-key frames off. I'll going ahead and build > from git-head and file a report if it doesn't work. Just to follow up on this, ffmpeg *is* successfully removing the leading non-key frames. This does *not* seem to be the case when using '-acodec copy -vcodec copy' (which doesn't surprise me). I just ran something like 'ffmpeg -i my_video.mpg -target ntsc-dvd my_fixed_video.mpg' and the leading non-key frames were gone. Thanks! From redneb8888 at gmail.com Sat Nov 12 02:19:11 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Fri, 11 Nov 2011 20:19:11 -0500 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: I understand how ffmpeg works. I was just complaining that it doesn't make sense. I think it would have been preferable if it worked like this: once a map has been specified, no streams that are not explicitly referenced by a -map should not be included in the output file. On Fri, Nov 11, 2011 at 05:12, Carl Eugen Hoyos wrote: > belcampo zonnet.nl> writes: > >> On 11/11/2011 03:54 AM, Marios Titas wrote: >> > You are right, adding the -sn option solved the problem. This doesn't >> > make much sense to me though. I think it would be more reasonable for >> > ffmpeg to assume that I don't want the subtitles since I did not >> > explicitly included them in my stream mappings. > > And this is exactly how (current) ffmpeg works. > >> As .avi cannot contain subs it's automatically excluded. > > I don't think that is correct (XSUB). > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From longyemail at gmail.com Sat Nov 12 10:20:24 2011 From: longyemail at gmail.com (yan dragon) Date: Sat, 12 Nov 2011 17:20:24 +0800 Subject: [FFmpeg-user] quicktime cannot play the mov file generated by ffmpeg? Message-ID: I generated a test.mov file using ffmpeg, with the AVFMT_FLAG_RTP_HINT flag enabled. When I play the test.mov with QuickTime player, the player show error like this: error 2002, one invalid pubilc atom found in the movie! Anybody else has the same experience? From ivano.arrighetta at gmail.com Sun Nov 13 00:43:01 2011 From: ivano.arrighetta at gmail.com (Ivano Arrighetta) Date: Sun, 13 Nov 2011 00:43:01 +0100 Subject: [FFmpeg-user] Please help the big project Message-ID: Hello everyone. Please help my open source project named "the big project". This just requires you to send an email to me sharing your knowledge. Visit: http://igpgames.altervista.org to contribute. From cehoyos at ag.or.at Sun Nov 13 03:27:09 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Nov 2011 02:27:09 +0000 (UTC) Subject: [FFmpeg-user] -map and matroska files References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: Marios Titas gmail.com> writes: > once a map has been specified, no streams that are not > explicitly referenced by a -map should not be included in the output > file. This is exactly how (current) FFmpeg works. Carl Eugen From cehoyos at ag.or.at Sun Nov 13 03:29:08 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Nov 2011 02:29:08 +0000 (UTC) Subject: [FFmpeg-user] quicktime cannot play the mov file generated by ffmpeg? References: Message-ID: yan dragon gmail.com> writes: > I generated a test.mov file using ffmpeg, with the AVFMT_FLAG_RTP_HINT flag > enabled. > When I play the test.mov with QuickTime player, the player show error like > this: > error 2002, one invalid pubilc atom found in the movie! Anybody else has the > same experience? Command line and complete, uncut console output of ffmpeg missing. Carl Eugen From redneb8888 at gmail.com Sun Nov 13 10:43:23 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Sun, 13 Nov 2011 04:43:23 -0500 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: On Sat, Nov 12, 2011 at 21:27, Carl Eugen Hoyos wrote: > Marios Titas gmail.com> writes: > >> once a map has been specified, no streams that are not >> explicitly referenced by a -map should not be included in the output >> file. > > This is exactly how (current) FFmpeg works. No, it's not. If there is a subtitle stream in the input file it will want to include a subtitle stream in the output file. So you must either select such a stream with -map or specifiy the -sn option. Otherwise it will faill. From cehoyos at ag.or.at Sun Nov 13 12:57:15 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Nov 2011 11:57:15 +0000 (UTC) Subject: [FFmpeg-user] -map and matroska files References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: Marios Titas gmail.com> writes: > On Sat, Nov 12, 2011 at 21:27, Carl Eugen Hoyos ag.or.at> wrote: > > Marios Titas gmail.com> writes: > > > >> once a map has been specified, no streams that are not > >> explicitly referenced by a -map should not be included in the output > >> file. > > > > This is exactly how (current) FFmpeg works. > > No, it's not. If there is a subtitle stream in the input file it will > want to include a subtitle stream in the output file. So you must > either select such a stream with -map or specifiy the -sn option. > Otherwise it will faill. Command line and complete, uncut console output missing. (It works for me exactly as you describe it above but I may be missing something. To fix it, I have to be able to reproduce your problem.) Carl Eugen From cehoyos at ag.or.at Sun Nov 13 13:02:43 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Nov 2011 12:02:43 +0000 (UTC) Subject: [FFmpeg-user] audio streams References: <61BE060D-F128-48F0-B654-EA8D2576620C@gmail.com> Message-ID: Rick C. gmail.com> writes: > > The -map option allows you to choose which stream(s) of the input file > > should be encoded. > > Sorry here's the output below. I guess there's no choice but to use the -map > option in this case? [...] > Input #0, mpeg, from '/Users/mahalko/Desktop/VTS_04_1.VOB': > Duration: 00:23:56.44, start: 0.203422, bitrate: 5979 kb/s > Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 64:45 DAR 16:9], 9800 kb/s, 25 fps, > 25 tbr, 90k tbn, 50 tbc > Stream #0.1[0x20]: Subtitle: dvdsub > Stream #0.2[0x21]: Subtitle: dvdsub > Stream #0.3[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s > Stream #0.4[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s I agree with you that there are several signs that indicate 0x80 is the track most users want to encode. Unfortunately, I don't think it is so easy in the general case, so yes, you will have to provide -map afaict. (My guess is that 0x81 starts earlier with non-silent sound and this is why FFmpeg believes it is the more useful audio track. Just a guess.) Carl Eugen From redneb8888 at gmail.com Sun Nov 13 19:51:32 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Sun, 13 Nov 2011 13:51:32 -0500 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: Marios Titas gmail.com> writes: > once a map has been specified, no streams that are not> explicitly referenced by a -map should not be included in the output> file. OK, so here's a complete example. In this case I have a good ol' avi file with 1 video stream and 1 audio stream and I want to get rid of the audio stream. If the above statement about -map was true, I should be able to do that using only -map. But here's what I get: 8<-------------------------------------------------------------------------------------- marios at box13 ~ $ ffmpeg -i eggs.avi -map 0.0 -vcodec copy eggs-no_audio.avi ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 11 2011 04:30:53 with gcc 4.5.3 configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-network --disable-vaapi --disable-vdpau --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --disable-indev=v4l --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-libfreetype --enable-pthreads --enable-libvpx --enable-libopenjpeg --disable-altivec --disable-avx --cpu=host --enable-hardcoded-tables libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'eggs.avi': Duration: 00:03:39.08, start: 0.000000, bitrate: 4798 kb/s Stream #0.0: Video: mpeg4, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 192 kb/s Number of stream maps must match number of output streams 8<-------------------------------------------------------------------------------------- Just to be clear, here's what I am claiming: -vn/-an/-sn should not be needed if at least one -map is specified. From cehoyos at ag.or.at Sun Nov 13 20:15:06 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 13 Nov 2011 19:15:06 +0000 (UTC) Subject: [FFmpeg-user] -map and matroska files References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: Marios Titas gmail.com> writes: > ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers Please test current git head. Carl Eugen From timpsa at kotinet.com Sun Nov 13 21:30:26 2011 From: timpsa at kotinet.com (timo orava) Date: Sun, 13 Nov 2011 22:30:26 +0200 (EET) Subject: [FFmpeg-user] MOV to MP4 for PS3 In-Reply-To: <217060352.180305.1321215882091.JavaMail.root@zmbs2.ppohosted.fi> Message-ID: <922620079.180326.1321216226099.JavaMail.root@zmbs2.ppohosted.fi> I have been trying to get my .MOV-files to work on PS3. I used ffmpeg to convert them to MP4 but they still don't work. I tried the following: ffmpeg.exe -i "%e" -vprofile baseline -aq 224k -b 4000k "%e.mp4 What are the parameters for PS3 as it seems to be quite picky? I have tried both PS3 Media Server and TVersity without any luck. Thanks! From h.reindl at thelounge.net Sun Nov 13 22:29:13 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 13 Nov 2011 22:29:13 +0100 Subject: [FFmpeg-user] current old-abi-snapshot is really slow Message-ID: <4EC036A9.1090708@thelounge.net> something seems to went wrong in the current shnapshot independent of the target-format with the same input-file see the differences while the faster one is an older machine from 2008 and the slow on an Intel(R) Core(TM) i7-2600 CPU @ 3.40GHz 0.7.7 directly from ffmpeg.org: OK: mp4 => mp4-x264 (10 sec / 1.27 MB / dur: 00:00:20 / flvtool: 0 / faststart: 1 / 2p: 0) 0.7.7 svn snapshot from today: OK: m4v => mp4-x264 (48 sec / 18.12 MB / dur: 00:04:37 / flvtool: 0 / faststart: 1 / 2p: 0) BTW: why are there currently no svn-version after cloning the snapshot? it says only 0.7.7 in the version but there re dfiierences because my x264-requires patch had to be modified [builduser at buildserver64:/rpmbuild/SOURCES]$ cat ffmpeg-snapshot.sh #!/bin/bash set -e tmp=$(mktemp -d) trap cleanup EXIT cleanup() { set +e [ -z "$tmp" -o ! -d "$tmp" ] || rm -rf "$tmp" } unset CDPATH pwd=$(pwd) date=$(date +%Y%m%d) echo "$tmp" cd "$tmp" git clone -b oldabi git://git.videolan.org/ffmpeg.git mv ffmpeg ffmpeg-$date cd ffmpeg-$date pushd libswscale popd ./version.sh . version.h find . -type d -name .git -print0 | xargs -0r rm -rf sed -i -e '/^\.PHONY: version\.h$/d' Makefile cat version.h cd .. tar jcf "$pwd"/ffmpeg-$date.tar.bz2 ffmpeg-$date cd - >/dev/null -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From h.reindl at thelounge.net Sun Nov 13 22:32:16 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 13 Nov 2011 22:32:16 +0100 Subject: [FFmpeg-user] SORRY Fwd: current old-abi-snapshot is really slow In-Reply-To: <4EC036A9.1090708@thelounge.net> References: <4EC036A9.1090708@thelounge.net> Message-ID: <4EC03760.10709@thelounge.net> sorry for the noise, i am an idiot on the local machine was another test-file configured only one question remains: why is there no svn-version in the snapshots? -------- Original-Nachricht -------- Betreff: [FFmpeg-user] current old-abi-snapshot is really slow Datum: Sun, 13 Nov 2011 22:29:13 +0100 Von: Reindl Harald Antwort an: FFmpeg user questions and RTFMs Organisation: the lounge interactive design An: Mailing-List ffmpeg something seems to went wrong in the current shnapshot independent of the target-format with the same input-file see the differences while the faster one is an older machine from 2008 and the slow on an Intel(R) Core(TM) i7-2600 CPU @ 3.40GHz 0.7.7 directly from ffmpeg.org: OK: mp4 => mp4-x264 (10 sec / 1.27 MB / dur: 00:00:20 / flvtool: 0 / faststart: 1 / 2p: 0) 0.7.7 svn snapshot from today: OK: m4v => mp4-x264 (48 sec / 18.12 MB / dur: 00:04:37 / flvtool: 0 / faststart: 1 / 2p: 0) BTW: why are there currently no svn-version after cloning the snapshot? it says only 0.7.7 in the version but there re dfiierences because my x264-requires patch had to be modified [builduser at buildserver64:/rpmbuild/SOURCES]$ cat ffmpeg-snapshot.sh #!/bin/bash set -e tmp=$(mktemp -d) trap cleanup EXIT cleanup() { set +e [ -z "$tmp" -o ! -d "$tmp" ] || rm -rf "$tmp" } unset CDPATH pwd=$(pwd) date=$(date +%Y%m%d) echo "$tmp" cd "$tmp" git clone -b oldabi git://git.videolan.org/ffmpeg.git mv ffmpeg ffmpeg-$date cd ffmpeg-$date pushd libswscale popd ./version.sh . version.h find . -type d -name .git -print0 | xargs -0r rm -rf sed -i -e '/^\.PHONY: version\.h$/d' Makefile cat version.h cd .. tar jcf "$pwd"/ffmpeg-$date.tar.bz2 ffmpeg-$date cd - >/dev/null -- Mit besten Gr??en, Reindl Harald the lounge interactive design GmbH A-1060 Vienna, Hofm?hlgasse 17 CTO / software-development / cms-solutions p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 icq: 154546673, http://www.thelounge.net/ http://www.thelounge.net/signature.asc.what.htm -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: Nachrichtenteil als Anhang URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From redneb8888 at gmail.com Sun Nov 13 23:54:18 2011 From: redneb8888 at gmail.com (Marios Titas) Date: Sun, 13 Nov 2011 17:54:18 -0500 Subject: [FFmpeg-user] -map and matroska files In-Reply-To: References: <4EBC8649.80407@gmail.com> <4EBCDE0C.8050802@zonnet.nl> Message-ID: On Sun, Nov 13, 2011 at 14:15, Carl Eugen Hoyos wrote: > Marios Titas gmail.com> writes: > >> ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers > > Please test current git head. Yes thank you, the current git HEAD works just fine: marios at box13 ~ $ /tmp/ffmpeg/ffmpeg -i eggs.avi -map 0:0 -vcodec copy eggs-no_audio.avi ffmpeg version N-34819-g3a9f2f1, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 13 2011 16:58:53 with gcc 4.5.3 configuration: libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 Input #0, avi, from 'eggs.avi': Duration: 00:03:39.08, start: 0.000000, bitrate: 4798 kb/s Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16, 192 kb/s Output #0, avi, to 'eggs-no_audio.avi': Metadata: ISFT : Lavf53.20.0 Stream #0:0: Video: mpeg4 (XVID / 0x44495658), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 29.97 tbn, 29.97 tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help frame= 6566 fps=1540 q=-1.0 Lsize= 123045kB time=00:03:39.08 bitrate=4600.9kbits/s video:122882kB audio:0kB global headers:0kB muxing overhead 0.132441% From longyemail at gmail.com Mon Nov 14 02:26:08 2011 From: longyemail at gmail.com (yan dragon) Date: Mon, 14 Nov 2011 09:26:08 +0800 Subject: [FFmpeg-user] Fwd: quicktime cannot play the mov file generated by ffmpeg? In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: yan dragon Date: 2011/11/13 Subject: Re: [FFmpeg-user] quicktime cannot play the mov file generated by ffmpeg? To: FFmpeg user questions and RTFMs First of all thank you for your reply. The command line and complete output: [root at as5 ffmpeg_bin]# ./ffmpeg -i test.asf -acodec copy -movflags rtphint test.mov ffmpeg version 0.8.5.git, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 12 2011 17:36:44 with gcc 4.1.1 20070105 (Red Hat 4.1.1-52) configuration: --enable-memalign-hack --disable-shared --enable-static --enable-debug --enable-libmp3lame libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 0 / 2. 48. 0 libswscale 2. 1. 0 / 2. 1. 0 [asf @ 0x8f61aa0] parser not found for codec pcm_s16le, packets or times may be invalid. Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 25.00 (25/1) Input #0, asf, from 'test.asf': Metadata: encoder : Lavf52.93.0 Duration: 00:00:20.04, start: 0.000000, bitrate: 659 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1 channels, s16, 128 kb/s Stream #0:1: Video: msmpeg4 (MP43 / 0x3334504D), yuv420p, 352x288, 25 tbr, 1k tbn, 1k tbc [buffer @ 0x8f69220] w:352 h:288 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: Output #0, mov, to 'test.mov': Metadata: encoder : Lavf53.20.0 Stream #0:0: Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 352x288, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le (sowt / 0x74776F73), 8000 Hz, 1 channels, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (msmpeg4 -> mpeg4) Stream #0:0 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 501 fps= 0 q=23.7 Lsize= 979kB time=00:00:20.01 bitrate= 400.7kbits/s video:610kB audio:313kB global headers:0kB muxing overhead 6.044626% [root at as5 ffmpeg_bin]# I tried to play the test.mov with the quicktime player (version 7.2) in the windows system, and the player show error like this: error 2002, invlalid public atom found in 'test.mov' file. Thanks again, I'm looking forward to your help. 2011/11/13 Carl Eugen Hoyos > yan dragon gmail.com> writes: > > > I generated a test.mov file using ffmpeg, with the AVFMT_FLAG_RTP_HINT > flag > > enabled. > > When I play the test.mov with QuickTime player, the player show error > like > > this: > > error 2002, one invalid pubilc atom found in the movie! Anybody else > has the > > same experience? > > Command line and complete, uncut console output of ffmpeg missing. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rickcorteza at gmail.com Mon Nov 14 04:29:33 2011 From: rickcorteza at gmail.com (Rick C.) Date: Mon, 14 Nov 2011 11:29:33 +0800 Subject: [FFmpeg-user] audio streams In-Reply-To: References: <61BE060D-F128-48F0-B654-EA8D2576620C@gmail.com> Message-ID: On Nov 13, 2011, at 8:02 PM, Carl Eugen Hoyos wrote: > Rick C. gmail.com> writes: > >>> The -map option allows you to choose which stream(s) of the input file >>> should be encoded. >> >> Sorry here's the output below. I guess there's no choice but to use the -map >> option in this case? > > [...] > >> Input #0, mpeg, from '/Users/mahalko/Desktop/VTS_04_1.VOB': >> Duration: 00:23:56.44, start: 0.203422, bitrate: 5979 kb/s >> Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 64:45 > DAR 16:9], 9800 kb/s, 25 fps, >> 25 tbr, 90k tbn, 50 tbc >> Stream #0.1[0x20]: Subtitle: dvdsub >> Stream #0.2[0x21]: Subtitle: dvdsub >> Stream #0.3[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s >> Stream #0.4[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s > > I agree with you that there are several signs that indicate 0x80 is the track > most users want to encode. Unfortunately, I don't think it is so easy in the > general case, so yes, you will have to provide -map afaict. > (My guess is that 0x81 starts earlier with non-silent sound and this is why > FFmpeg believes it is the more useful audio track. Just a guess.) > > Carl Eugen > Thank you for the reply Carl. Yes 0x81 has a narrative and I believe it starts speaking before the "normal" sounds would come in with 0x80. Just for me to learn what is the general rule FFmpeg uses to decide what audio track to use? Thanks again! From cehoyos at ag.or.at Mon Nov 14 09:37:15 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Nov 2011 08:37:15 +0000 (UTC) Subject: [FFmpeg-user] Fwd: quicktime cannot play the mov file generated by ffmpeg? References: Message-ID: yan dragon gmail.com> writes: > # ./ffmpeg -i test.asf -acodec copy -movflags rtphint test.mov Why are you using rtphint? Does the file work with DSS? Carl Eugen From longyemail at gmail.com Mon Nov 14 10:27:25 2011 From: longyemail at gmail.com (yan dragon) Date: Mon, 14 Nov 2011 17:27:25 +0800 Subject: [FFmpeg-user] Fwd: quicktime cannot play the mov file generated by ffmpeg? In-Reply-To: References: Message-ID: Not tested in DSS yet, I just play the mov file width quicktimie in the local system. I want to place the mov file in the web server, and remote users can play it immediately, needn't wait all the mov file downloaded to local system. Thanks. 2011/11/14 Carl Eugen Hoyos > yan dragon gmail.com> writes: > > > # ./ffmpeg -i test.asf -acodec copy -movflags rtphint test.mov > > Why are you using rtphint? Does the file work with DSS? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Mon Nov 14 11:35:37 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Nov 2011 10:35:37 +0000 (UTC) Subject: [FFmpeg-user] audio streams References: <61BE060D-F128-48F0-B654-EA8D2576620C@gmail.com> Message-ID: Rick C. gmail.com> writes: > > (My guess is that 0x81 starts earlier with non-silent sound and this is why > > FFmpeg believes it is the more useful audio track. Just a guess.) > > Thank you for the reply Carl. Yes 0x81 has a narrative and I believe it > starts speaking before the "normal" sounds would come in with 0x80. Just for > me to learn what is the general rule FFmpeg uses to decide what audio > track to use? I don't know exactly myself, but I believe your sample gives good indication on how it works;-) Carl Eugen From dmonlist at gmail.com Mon Nov 14 12:07:47 2011 From: dmonlist at gmail.com (Dmitry Monakhov) Date: Mon, 14 Nov 2011 15:07:47 +0400 Subject: [FFmpeg-user] Android encoded video Demux and Mux again Message-ID: <87bosfc67w.fsf@dmbot.sw.ru> Hi, I want grab video stream from android device There are two ways are possible 1) Stream container with a single stream(only video) to socket, and later parse it, and grab raw h264 stream similar to sipdroid(parse h263) or http://code.google.com/p/spydroid-ipcamera/ 2) Use Camera.PreviewCallback to grab raw YUV images and later encode it via ffmpeg (bambuser) or to mjpeg similar to https://github.com/vanevery/Android-MJPEG-Test Second one is simpler, but not optimized and has horrible performance. That's why i want use first alternative. So i've done simple proof of concept test demux and mux container again: Save several samples for different containers (mp4,3gp with) with one stream (video only) to regular files, so all container's data on it's paces(See links at the end of the letter). Now i want to demux my container in order to get raw h264 stream, #ffmpeg -i h264sample.mp4 -vcodec copy 264sample.h264 raw h264 stream is that i will have after my runtime stream parser. And now i want to check that resulted stream are usable and it is possible to mux it again to container #ffmpeg -i h264sample.h264 -vcodec copy 264sample-out.mp4 But i always get an error. Question #1 How i can mux raw h264 stream to container. Question #2 I'm not good in encoding standards so can you please describe me. Does individual frame of h263's or h264's raw stream contains any timings information pts or dts? https://docs.google.com/open?id=0B9rKdymGDIaoYWE3ODA0OWQtMjk1MC00MDEwLThhMmQtZjJhNjgzOGRiOTlj https://docs.google.com/open?id=0B9rKdymGDIaoNGIzZGJlZTMtYWI0MS00NzJhLWEwZWEtZjBjODJmZTExNTAy From cehoyos at ag.or.at Mon Nov 14 12:51:30 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Nov 2011 11:51:30 +0000 (UTC) Subject: [FFmpeg-user] Android encoded video Demux and Mux again References: <87bosfc67w.fsf@dmbot.sw.ru> Message-ID: Dmitry Monakhov gmail.com> writes: > And now i want to check that resulted stream are usable and it is > possible to mux it again to container > #ffmpeg -i h264sample.h264 -vcodec copy 264sample-out.mp4 > But i always get an error. Complete, uncut console output missing. Carl Eugen From dmonlist at gmail.com Mon Nov 14 13:30:39 2011 From: dmonlist at gmail.com (Dmitry Monakhov) Date: Mon, 14 Nov 2011 16:30:39 +0400 Subject: [FFmpeg-user] Android encoded video Demux and Mux again In-Reply-To: References: <87bosfc67w.fsf@dmbot.sw.ru> Message-ID: <87wrb2998w.fsf@dmbot.sw.ru> On Mon, 14 Nov 2011 11:51:30 +0000 (UTC), Carl Eugen Hoyos wrote: > Dmitry Monakhov gmail.com> writes: > > > And now i want to check that resulted stream are usable and it is > > possible to mux it again to container > > #ffmpeg -i h264sample.h264 -vcodec copy 264sample-out.mp4 > > But i always get an error. > > Complete, uncut console output missing. ffmpeg version N-34800-g46eae15, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 13 2011 15:23:31 with gcc 4.4.3 configuration: --disable-yasm --enable-gpl libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 0 / 2. 48. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [h264 @ 0xa339e20] missing picture in access unit with size 1693527 [h264 @ 0xa339e20] no frame! [h264 @ 0xa32bac0] Could not find codec parameters (Video: h264) [h264 @ 0xa32bac0] Estimating duration from bitrate, this may be inaccurate h264sample-1.h264: could not find codec parameters i've tried to demux it with following command ffmpeg -i h264sample.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 and now i'm able to mux it again, but i've issues with timings Seems like android encode it with variable framerate. So in order to mux it correctly i have to know pts or dts. Is it possible to determine this info from the stream? I'm asking because sipdroid and spydroid-ipcamera does strange things: Basically it parse h263, or 264 stream and paketize it to srtp 1) Camera encoder write three_gp container (with video only ) to socket 2) receiver read data from socket, parse each nal unit one by one, and paketize it to SRTP stream and tag each packet with the current time (the time when this unit was transfered to srtp unit), but it seems to be wrong, and it should be tagged with the time then each nal was grabbed from camera. My theory that this is because it is hard or impossible to determine real time. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Mon Nov 14 14:13:41 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Nov 2011 13:13:41 +0000 (UTC) Subject: [FFmpeg-user] Android encoded video Demux and Mux again References: <87bosfc67w.fsf@dmbot.sw.ru> <87wrb2998w.fsf@dmbot.sw.ru> Message-ID: Dmitry Monakhov gmail.com> writes: > > > And now i want to check that resulted stream are usable and it is > > > possible to mux it again to container > > > #ffmpeg -i h264sample.h264 -vcodec copy 264sample-out.mp4 > > > But i always get an error. > > > > Complete, uncut console output missing. > ffmpeg version N-34800-g46eae15, Copyright (c) 2000-2011 the FFmpeg > developers > built on Nov 13 2011 15:23:31 with gcc 4.4.3 > [h264 @ 0xa339e20] missing picture in access unit with size 1693527 > [h264 @ 0xa339e20] no frame! > [h264 @ 0xa32bac0] Could not find codec parameters (Video: h264) > [h264 @ 0xa32bac0] Estimating duration from bitrate, this may be > inaccurate > h264sample-1.h264: could not find codec parameters Is there any software that understands your file? (If you have posted complete, uncut output, FFmpeg completely fails to read h264sample.h264, this is not a problem of remuxing or pts.) Carl Eugen From peace at AleksandrSolzhenitsyn.net Mon Nov 14 17:19:28 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Mon, 14 Nov 2011 11:19:28 -0500 Subject: [FFmpeg-user] Convert an Mp4 file to a smaller size? In-Reply-To: References: Message-ID: <4EC13F90.5020401@AleksandrSolzhenitsyn.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I want to convert a 10 mg mp4 (video) file down to a smaller file size. What "switches" should I use? Keeping audio quality reasonably close to the original would be nice and I don't want any of the sides or top of the video cropped. The file will be put on my iPod Classic. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJOwT+LAAoJEPBpZNn4grcjwfgH/3rxY8tEg0kO7DMzL5F5b+2Y lS1VeoWJ2RN9tptJVvidmpbl8Xr5VrZivoX3qLXJs5k2mekAvOOGy5UPCeRQrd3T XqHLGhX1D8YMgYu7+PFLtKn04oEFjnHAzICerY3OSqMMjBY3kHRhPL/eSiWVOa1A nrFOMjLjv4E68bTrVjKLZrgr5VCYjnalYgomloP1uLLOK0Q/lObv7iedNFiE5KdV 98LojdZNkW6ZsbC/srf9PPjYSRFPr9ezSN+mpyIkkSiza65QnKW/I06LWwHKZ/xu dovqzRnejNzn5e0AvLlbnXmrDV4fxFYWptWngRHm565uCtheyEWMeTimrHQ/2W8= =3D5p -----END PGP SIGNATURE----- From dmonlist at gmail.com Mon Nov 14 17:58:58 2011 From: dmonlist at gmail.com (Dmitry Monakhov) Date: Mon, 14 Nov 2011 20:58:58 +0400 Subject: [FFmpeg-user] Android encoded video Demux and Mux again In-Reply-To: References: <87bosfc67w.fsf@dmbot.sw.ru> <87wrb2998w.fsf@dmbot.sw.ru> Message-ID: <87boser67h.fsf@dmbot.sw.ru> On Mon, 14 Nov 2011 13:13:41 +0000 (UTC), Carl Eugen Hoyos wrote: > Dmitry Monakhov gmail.com> writes: > > > > > And now i want to check that resulted stream are usable and it is > > > > possible to mux it again to container > > > > #ffmpeg -i h264sample.h264 -vcodec copy 264sample-out.mp4 > > > > But i always get an error. > > > > > > Complete, uncut console output missing. > > ffmpeg version N-34800-g46eae15, Copyright (c) 2000-2011 the FFmpeg > > developers > > built on Nov 13 2011 15:23:31 with gcc 4.4.3 > > > [h264 @ 0xa339e20] missing picture in access unit with size 1693527 > > [h264 @ 0xa339e20] no frame! > > [h264 @ 0xa32bac0] Could not find codec parameters (Video: h264) > > [h264 @ 0xa32bac0] Estimating duration from bitrate, this may be > > inaccurate > > h264sample-1.h264: could not find codec parameters > > Is there any software that understands your file? No, i dont know any. > (If you have posted complete, uncut output, FFmpeg completely fails to read > h264sample.h264, this is not a problem of remuxing or pts.) Yes i know, and my question, what should i do to provide such options? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From 13760746839 at 163.com Mon Nov 14 03:08:06 2011 From: 13760746839 at 163.com (=?GBK?B?0KSyqA==?=) Date: Mon, 14 Nov 2011 10:08:06 +0800 (CST) Subject: [FFmpeg-user] Problems with lxf demuxer Message-ID: <7913e1af.b337.1339fd4d781.Coremail.13760746839@163.com> Hi, As noted in ffmpeg changelog, the current ffmpeg support demux of lxf format, but I noticed that for Leitch Version 1,when use ffplay, it will resulted in the following error message: [lxf @ 02F91840] checksum error [lxf @ 02F91840] expected 120 B size header, got 0 ... :Invalid data found when processing input Now Leitch Verison 1 is common used in Harris NEXIO Video Server, and I found the MediaInfo now also support this format, is anybody available to update the dumuxer code of it? XiaoBo From cehoyos at ag.or.at Mon Nov 14 23:34:56 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 14 Nov 2011 22:34:56 +0000 (UTC) Subject: [FFmpeg-user] Problems with lxf demuxer References: <7913e1af.b337.1339fd4d781.Coremail.13760746839@163.com> Message-ID: Hi! ?? <13760746839 163.com> writes: > I noticed that for Leitch Version 1,when use ffplay, it will resulted in the > following error message: > [lxf @ 02F91840] checksum error > [lxf @ 02F91840] expected 120 B size header, got 0 > ... :Invalid data found when processing input Please provide a sample. Carl Eugen From rickcorteza at gmail.com Tue Nov 15 03:58:26 2011 From: rickcorteza at gmail.com (Rick C.) Date: Tue, 15 Nov 2011 10:58:26 +0800 Subject: [FFmpeg-user] audio streams In-Reply-To: References: <61BE060D-F128-48F0-B654-EA8D2576620C@gmail.com> Message-ID: <89309F79-8113-464C-95E2-9AD051182585@gmail.com> On Nov 14, 2011, at 6:35 PM, Carl Eugen Hoyos wrote: > Rick C. gmail.com> writes: > >>> (My guess is that 0x81 starts earlier with non-silent sound and this is why >>> FFmpeg believes it is the more useful audio track. Just a guess.) >> >> Thank you for the reply Carl. Yes 0x81 has a narrative and I believe it >> starts speaking before the "normal" sounds would come in with 0x80. Just for >> me to learn what is the general rule FFmpeg uses to decide what audio >> track to use? > > I don't know exactly myself, but I believe your sample gives good indication on > how it works;-) > > Carl Eugen > Great thanks Carl as always you're most helpful! :-) From tim.nicholson at bbc.co.uk Tue Nov 15 09:46:46 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 08:46:46 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue Message-ID: <4EC226F6.8010403@bbc.co.uk> According to http://ffmpeg.org/ffmpeg.html#Main-options "ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT encodes all video streams with libx264 and copies all audio streams" However I have an mxf file with 1 video and 4 audio streams I am trying to rewrap as a mov where I seem unable to get more than a single audio stream copied. I have tried the following command lines:- ffmpeg -i in.mxf -vcodec copy -acodec copy tim2.mov ffmpeg -i in.mxf -vcodec copy -acodec copy -ac 4 tim2.mov ffmpeg -i in.mxf -vcodec copy -c:a copy tim2.mov ffmpeg -i in.mxf -vcodec copy -c:a:0 copy -c:a:1 copy -c:a:2 copy -c:a:3 copy tim2.mov fmpeg -i in.mxf -vcodec copy -ac 4 -c:a:0 copy -c:a:1 copy -c:a:2 copy -c:a:3 copy tim2.mov But all produce the same result:- ffmpeg version N-34849-g07c7ffc-by_Tim, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 14 2011 12:24:11 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --extra-version=by_Tim --enable-static --disable-shared --enable-gpl --enable-nonfree --enable-version3 --prefix=/mnt/store-0/tims/ffmpeg-tux/usr/local --enable-runtime-cpudetect --extra-cflags='-static -I/mnt/store-0/tims/ffmpeg-tux/usr/local/include' --extra-ldflags='-static -L/mnt/store-0/tims/ffmpeg-tux/usr/local/lib' --progs-suffix=STATIC --enable-libfaac --enable-libx264 libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (50/2) Input #0, mxf, from 'JH3-AVCi100.mxf': Duration: 00:00:31.96, start: 0.000000, bitrate: 117147 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:3: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:4: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Output #0, mov, to 'tim2.mov': Metadata: encoder : Lavf53.20.0 Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 1 channels, 768 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Tue Nov 15 10:42:09 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 09:42:09 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC226F6.8010403@bbc.co.uk> References: <4EC226F6.8010403@bbc.co.uk> Message-ID: <4EC233F1.4000800@bbc.co.uk> On 15/11/11 08:46, Tim Nicholson wrote: > According to http://ffmpeg.org/ffmpeg.html#Main-options > "ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT > encodes all video streams with libx264 and copies all audio streams" > > However I have an mxf file with 1 video and 4 audio streams I am trying > to rewrap as a mov where I seem unable to get more than a single audio > stream copied. > > I have tried the following command lines:- > > ffmpeg -i in.mxf -vcodec copy -acodec copy tim2.mov > > ffmpeg -i in.mxf -vcodec copy -acodec copy -ac 4 tim2.mov > > ffmpeg -i in.mxf -vcodec copy -c:a copy tim2.mov > > ffmpeg -i in.mxf -vcodec copy -c:a:0 copy -c:a:1 copy -c:a:2 copy -c:a:3 > copy tim2.mov > > fmpeg -i in.mxf -vcodec copy -ac 4 -c:a:0 copy -c:a:1 copy -c:a:2 copy > -c:a:3 copy tim2.mov > > But all produce the same result:- > > [..] > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Stream #0:1 -> #0:1 (copy) So to answer my own point it seems that the "-map 0" is essential to get all the tracks*. However according to:- http://ffmpeg.org/ffmpeg.html#Stream-selection "By default ffmpeg tries to pick the "best" stream of each type present in input files and add them to each output file. For video, this means the highest resolution, for audio the highest channel count. For subtitle it?s simply the first subtitle stream. [...] For full manual control, use the -map option, which disables the defaults just described. " In my case with only 1 video and 4 audios I would expect the "default" of "highest resolution" and "highest channel count" to give me what I wanted, but it would appear that the audio default is *not* the highest channel count but the first channel as per the subtitle spec. *just for the record I used:- ffmpeg -i in.mxf -map 0 -vcodec copy -acodec copy tim2.mov -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From hughht5 at gmail.com Tue Nov 15 10:54:05 2011 From: hughht5 at gmail.com (Hugh Halford-Thompson) Date: Tue, 15 Nov 2011 09:54:05 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC233F1.4000800@bbc.co.uk> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> Message-ID: I believe you may have to use Sox (or another program) until ffmpeg developers add support for this. Please correct me if im wrong. From hughht5 at gmail.com Tue Nov 15 11:05:35 2011 From: hughht5 at gmail.com (hughht5) Date: Tue, 15 Nov 2011 02:05:35 -0800 (PST) Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> Message-ID: <1321351535353-4042416.post@n4.nabble.com> I would use the -map_channel option, however the documentation says it is limited at the moment: (From http://ffmpeg.org/ffmpeg.html) Note that "-map_channel" is currently limited to the scope of one input for each output; you can?t for example use it to pick multiple input audio files and mix them into one single output. I hope this is being implemented. Hugh -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Audio-Stream-Mapping-issue-tp4042261p4042416.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ingemar.s.johansson at ericsson.com Tue Nov 15 11:11:34 2011 From: ingemar.s.johansson at ericsson.com (Ingemar Johansson S) Date: Tue, 15 Nov 2011 11:11:34 +0100 Subject: [FFmpeg-user] ffmpeg for rate adaptive experiments Message-ID: Hi I am looking for a simple way to visualize an encoding scenario where the target bitrate varies with time. The typical use case is a rate adaptive video for a realtime interactive service where the bitrate is determined by channel conditions. Typically it should be possible to change the bitrate once per second or possibly more often. I would like to visualize packet loss and delay as well, in short I want to demo the cases 1) No rate adaptation e.g 384kbps 2) Rate adaptation e.g between 64 and 384kpbs The demo should give answers to the questions how is video quality affect by the packet delay packet drops and the bitrate as a reult of the two approaches above, I get the the bitrate and delay traces from a proprietary system simulator. Is there a possiblity to use ffmpeg for this purpose ?, alternatively is there an experimental platform out there that does something like the above ?. /Ingemar ================================= INGEMAR JOHANSSON M.Sc. Senior Researcher Ericsson AB Wireless Access Networks Labratoriegr?nd 11 971 28, Lule?, Sweden Phone +46-1071 43042 SMS/MMS +46-73 078 3289 ingemar.s.johansson at ericsson.com www.ericsson.com ================================= From cehoyos at ag.or.at Tue Nov 15 11:14:29 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Nov 2011 10:14:29 +0000 (UTC) Subject: [FFmpeg-user] Audio Stream Mapping issue References: <4EC226F6.8010403@bbc.co.uk> Message-ID: HI! Tim Nicholson bbc.co.uk> writes: > According to http://ffmpeg.org/ffmpeg.html#Main-options > "ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT > encodes all video streams with libx264 and copies all audio streams" > > However I have an mxf file with 1 video and 4 audio streams I am trying > to rewrap as a mov where I seem unable to get more than a single audio > stream copied. > > I have tried the following command lines:- > > ffmpeg -i in.mxf -vcodec copy -acodec copy tim2.mov > > ffmpeg -i in.mxf -vcodec copy -acodec copy -ac 4 tim2.mov > > ffmpeg -i in.mxf -vcodec copy -c:a copy tim2.mov > > ffmpeg -i in.mxf -vcodec copy -c:a:0 copy -c:a:1 copy -c:a:2 copy -c:a:3 > copy tim2.mov > > fmpeg -i in.mxf -vcodec copy -ac 4 -c:a:0 copy -c:a:1 copy -c:a:2 copy > -c:a:3 copy tim2.mov Afaict, all command lines you list omit "-map 0" meaning you ask ffmpeg NOT to map all streams into the output file (but exactly one video and one audio stream). Or do I miss something? Carl Eugen From ubitux at gmail.com Tue Nov 15 11:19:31 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 15 Nov 2011 11:19:31 +0100 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <1321351535353-4042416.post@n4.nabble.com> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> Message-ID: <20111115101931.GC26053@leki> On Tue, Nov 15, 2011 at 02:05:35AM -0800, hughht5 wrote: > I would use the -map_channel option, however the documentation says it is > limited at the moment: > > (From http://ffmpeg.org/ffmpeg.html) > Note that "-map_channel" is currently limited to the scope of one input for > each output; you can?t for example use it to pick multiple input audio files > and mix them into one single output. > AFAIU, Tim only has one file so it doesn't apply here. Tim, I'm not sure to understand what you really want from your 4 stereo audio streams. I can't tell if you want to pick only some (then -map) or actually merge the channels of each streams into one single audio stream (then what you need is indeed the -map_channel feature). > I hope this is being implemented. > The multiple files limitation could be bypassed with an audio merge filter; a WIP was sent on the mailing list by Nicolas. We lack man-power, and patches are always welcome when it is a feature request :) Regards, -- 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 rogerdpack2 at gmail.com Tue Nov 15 15:06:04 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 15 Nov 2011 07:06:04 -0700 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: <4EBD8E0B.400@gmail.com> References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> Message-ID: > Just to follow up on this, ffmpeg *is* successfully removing the leading > non-key frames. ?This does *not* seem to be the case when using '-acodec > copy -vcodec copy' (which doesn't surprise me). ?I just ran something like > 'ffmpeg -i my_video.mpg -target ntsc-dvd my_fixed_video.mpg' and the leading > non-key frames were gone. with -ss 0 it "might" skip to the first keyframe with vcodec copy. Maybe? -r From rogerdpack2 at gmail.com Tue Nov 15 15:30:59 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 15 Nov 2011 07:30:59 -0700 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 Message-ID: http://ffmpeg.org/trac/ffmpeg/ticket/437 It is reproducible in linux I believe. $150 bounty! -roger- From lytithwyn at gmail.com Tue Nov 15 15:40:14 2011 From: lytithwyn at gmail.com (Matthew Morgan) Date: Tue, 15 Nov 2011 09:40:14 -0500 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> Message-ID: <4EC279CE.2070800@gmail.com> >> Just to follow up on this, ffmpeg *is* successfully removing the leading >> non-key frames. This does *not* seem to be the case when using '-acodec >> copy -vcodec copy' (which doesn't surprise me). I just ran something like >> 'ffmpeg -i my_video.mpg -target ntsc-dvd my_fixed_video.mpg' and the leading >> non-key frames were gone. > with -ss 0 it "might" skip to the first keyframe with vcodec copy. Maybe? > -r Alas but no, that doesn't work either. The skip happens, but it's not seeking to the first keyframe after the skip like it usually would. I don't know whether it's because keyframe-seeking gets disabled with codec copy or if it's the fact that these input files are a little weird. I found that if I get rid of the codec copy commands and try to do a skip of 60 seconds (long enough to make it obvious), I get: ffmpeg -i M2U00108.MPG -ss 60 -target ntsc-dvd -y m2u00108_copy.mpg ... [mpeg2video @ 0x8cd99c0] warning: first frame is no keyframe Last message repeated 1 times [buffer @ 0x8cd9980] Buffering several frames is not supported. Please consume all available frames before adding a new one. Last message repeated 1791 times 0kB time=10000000000.00 bitrate= 0.0kbits/s And after that it starts encoding. I also find that with these input files if I don't specify "-target ntsc-dvd" the output file has no audio. If I run "ffmpeg -i" on that output file it *says* there's audio, but mplayer says there isn't and I only get video during playback. From dave.bevan at bbc.co.uk Tue Nov 15 15:41:13 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Tue, 15 Nov 2011 14:41:13 -0000 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > http://ffmpeg.org/trac/ffmpeg/ticket/437 > It is reproducible in linux I believe. > $150 bounty! I'm not actually sure this is a bug. RGB & YUV have differing colour space, and differing gamut curves - conversion is NOT linear. On your ffplay cmd output, the scale filter is inserting [scale @ 03C06280] w:1680 h:1050 fmt:bgr24 -> w:1680 h:1050 fmt:yuv420p flags:0x4 which WILL change colours, because yuv is a compressed colour space (pure black is 16,16,16 and pure white is 235,235,235), always has been, always will be. Further, within the YUV colour space, there are different gamut curves - rec601 & rec709 for example. Please read http://en.wikipedia.org/wiki/YUV for a more indepth explaination. --D. http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Tue Nov 15 16:27:48 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 15:27:48 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <20111115101931.GC26053@leki> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> Message-ID: <4EC284F4.3010109@bbc.co.uk> On 15/11/11 10:19, Cl?ment B?sch wrote: > On Tue, Nov 15, 2011 at 02:05:35AM -0800, hughht5 wrote: >> I would use the -map_channel option, however the documentation says it is >> limited at the moment: >> >> (From http://ffmpeg.org/ffmpeg.html) >> Note that "-map_channel" is currently limited to the scope of one input for >> each output; you can?t for example use it to pick multiple input audio files >> and mix them into one single output. >> > > AFAIU, Tim only has one file so it doesn't apply here. > Quite so. > Tim, I'm not sure to understand what you really want from your 4 stereo > audio streams. I can't tell if you want to pick only some (then -map) or > actually merge the channels of each streams into one single audio stream > (then what you need is indeed the -map_channel feature). > All I actually wanted to do was rewrap mxf->mov with no other changes. (They were in fact 4 mono streams mxf does not support stereo streams afaik) ffmpeg -i in.mxf -map 0 -vcodec copy -acodec copy tim2.mov or more succinctly:- ffmpeg -i in.mxf -map 0 -c copy tim2.mov performs this. However according to the documentation the -map 0 *should* be unnecessary as the default *should* be "for audio the highest channel count" wheeras it seems to be the first audio stream only. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Tue Nov 15 16:34:36 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 15:34:36 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: References: <4EC226F6.8010403@bbc.co.uk> Message-ID: <4EC2868C.4080305@bbc.co.uk> On 15/11/11 10:14, Carl Eugen Hoyos wrote: > HI! > > Tim Nicholson bbc.co.uk> writes: > >> According to http://ffmpeg.org/ffmpeg.html#Main-options >> "ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT >> encodes all video streams with libx264 and copies all audio streams" >> >> However I have an mxf file with 1 video and 4 audio streams I am trying >> to rewrap as a mov where I seem unable to get more than a single audio >> stream copied. >> >> I have tried the following command lines:- >> >> ffmpeg -i in.mxf -vcodec copy -acodec copy tim2.mov >> >> ffmpeg -i in.mxf -vcodec copy -acodec copy -ac 4 tim2.mov >> >> ffmpeg -i in.mxf -vcodec copy -c:a copy tim2.mov >> >> ffmpeg -i in.mxf -vcodec copy -c:a:0 copy -c:a:1 copy -c:a:2 copy -c:a:3 >> copy tim2.mov >> >> fmpeg -i in.mxf -vcodec copy -ac 4 -c:a:0 copy -c:a:1 copy -c:a:2 copy >> -c:a:3 copy tim2.mov > > Afaict, all command lines you list omit "-map 0" meaning you ask ffmpeg NOT to > map all streams into the output file (but exactly one video and one audio > stream). > Or do I miss something? You miss the documentation that states that the default mapping is:- "For video, this means the highest resolution, for audio the highest channel count" which says to me that one video of the highest resolution is mapped and all audio (The highest count). If this is not what it means then the documentation is ambiguous. since for subtitles it specifically says "the first subtitle stream", and in my tests as above it is the first audio as well. Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From ubitux at gmail.com Tue Nov 15 17:08:51 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 15 Nov 2011 17:08:51 +0100 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC284F4.3010109@bbc.co.uk> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> Message-ID: <20111115160851.GD26053@leki> On Tue, Nov 15, 2011 at 03:27:48PM +0000, Tim Nicholson wrote: > On 15/11/11 10:19, Cl?ment B?sch wrote: > >On Tue, Nov 15, 2011 at 02:05:35AM -0800, hughht5 wrote: > >>I would use the -map_channel option, however the documentation says it is > >>limited at the moment: > >> > >>(From http://ffmpeg.org/ffmpeg.html) > >>Note that "-map_channel" is currently limited to the scope of one input for > >>each output; you can?t for example use it to pick multiple input audio files > >>and mix them into one single output. > >> > > > >AFAIU, Tim only has one file so it doesn't apply here. > > > > Quite so. > > >Tim, I'm not sure to understand what you really want from your 4 stereo > >audio streams. I can't tell if you want to pick only some (then -map) or > >actually merge the channels of each streams into one single audio stream > >(then what you need is indeed the -map_channel feature). > > > > All I actually wanted to do was rewrap mxf->mov with no other > changes. (They were in fact 4 mono streams mxf does not support > stereo streams afaik) > My bad, I misread it was mono. > ffmpeg -i in.mxf -map 0 -vcodec copy -acodec copy tim2.mov > > or more succinctly:- > > ffmpeg -i in.mxf -map 0 -c copy tim2.mov > > performs this. > > However according to the documentation the -map 0 *should* be > unnecessary as the default *should* be "for audio the highest > channel count" wheeras it seems to be the first audio stream only. > The documenation sounds quite obvious to me; the selection pick "the best", and in case of score equality, only the first one is picked. Maybe it should indeed be explicited, or the behaviour changed. Sorry for the first misunderstanding, I'll do some checks and send a patch for this in a while if no one does. -- 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 tim.nicholson at bbc.co.uk Tue Nov 15 17:23:36 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 16:23:36 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <20111115160851.GD26053@leki> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> Message-ID: <4EC29208.5000802@bbc.co.uk> On 15/11/11 16:08, Cl?ment B?sch wrote: > On Tue, Nov 15, 2011 at 03:27:48PM +0000, Tim Nicholson wrote: >> On 15/11/11 10:19, Cl?ment B?sch wrote: >>> On Tue, Nov 15, 2011 at 02:05:35AM -0800, hughht5 wrote: >>>> I would use the -map_channel option, however the documentation says it is >>>> limited at the moment: >>>> >>>> (From http://ffmpeg.org/ffmpeg.html) >>>> Note that "-map_channel" is currently limited to the scope of one input for >>>> each output; you can?t for example use it to pick multiple input audio files >>>> and mix them into one single output. >>>> >>> >>> AFAIU, Tim only has one file so it doesn't apply here. >>> >> >> Quite so. >> >>> Tim, I'm not sure to understand what you really want from your 4 stereo >>> audio streams. I can't tell if you want to pick only some (then -map) or >>> actually merge the channels of each streams into one single audio stream >>> (then what you need is indeed the -map_channel feature). >>> >> >> All I actually wanted to do was rewrap mxf->mov with no other >> changes. (They were in fact 4 mono streams mxf does not support >> stereo streams afaik) >> > > My bad, I misread it was mono. > >> ffmpeg -i in.mxf -map 0 -vcodec copy -acodec copy tim2.mov >> >> or more succinctly:- >> >> ffmpeg -i in.mxf -map 0 -c copy tim2.mov >> >> performs this. >> >> However according to the documentation the -map 0 *should* be >> unnecessary as the default *should* be "for audio the highest >> channel count" wheeras it seems to be the first audio stream only. >> > > The documenation sounds quite obvious to me; the selection pick "the > best", and in case of score equality, only the first one is picked. Maybe > it should indeed be explicited, or the behaviour changed. > In my experience, of much documentation writing and editing of others write ups, something that is abundantly clear to the writer can be completely misinterpreted by end users approaching things from a different starting point, it seems to be the way of things. That is why I first rasied the issue here to try and get a consensus of what the expected behaviour should be, and then try and understand how the documentation can be read to mean that... I am not sure where you get the 'pick "the best"' from the documentation, it only says this for video, not audio! I am happy to contribute to better wording, once I understand what the "correct" behaviour is. > Sorry for the first misunderstanding, I'll do some checks and send a patch > for this in a while if no one does. > -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From ubitux at gmail.com Tue Nov 15 17:33:52 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 15 Nov 2011 17:33:52 +0100 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC29208.5000802@bbc.co.uk> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> Message-ID: <20111115163352.GE26053@leki> On Tue, Nov 15, 2011 at 04:23:36PM +0000, Tim Nicholson wrote: > On 15/11/11 16:08, Cl?ment B?sch wrote: > >On Tue, Nov 15, 2011 at 03:27:48PM +0000, Tim Nicholson wrote: > >>On 15/11/11 10:19, Cl?ment B?sch wrote: > >>>On Tue, Nov 15, 2011 at 02:05:35AM -0800, hughht5 wrote: > >>>>I would use the -map_channel option, however the documentation says it is > >>>>limited at the moment: > >>>> > >>>>(From http://ffmpeg.org/ffmpeg.html) > >>>>Note that "-map_channel" is currently limited to the scope of one input for > >>>>each output; you can?t for example use it to pick multiple input audio files > >>>>and mix them into one single output. > >>>> > >>> > >>>AFAIU, Tim only has one file so it doesn't apply here. > >>> > >> > >>Quite so. > >> > >>>Tim, I'm not sure to understand what you really want from your 4 stereo > >>>audio streams. I can't tell if you want to pick only some (then -map) or > >>>actually merge the channels of each streams into one single audio stream > >>>(then what you need is indeed the -map_channel feature). > >>> > >> > >>All I actually wanted to do was rewrap mxf->mov with no other > >>changes. (They were in fact 4 mono streams mxf does not support > >>stereo streams afaik) > >> > > > >My bad, I misread it was mono. > > > >>ffmpeg -i in.mxf -map 0 -vcodec copy -acodec copy tim2.mov > >> > >>or more succinctly:- > >> > >>ffmpeg -i in.mxf -map 0 -c copy tim2.mov > >> > >>performs this. > >> > >>However according to the documentation the -map 0 *should* be > >>unnecessary as the default *should* be "for audio the highest > >>channel count" wheeras it seems to be the first audio stream only. > >> > > > >The documenation sounds quite obvious to me; the selection pick "the > >best", and in case of score equality, only the first one is picked. Maybe > >it should indeed be explicited, or the behaviour changed. > > > > In my experience, of much documentation writing and editing of > others write ups, something that is abundantly clear to the writer > can be completely misinterpreted by end users approaching things > from a different starting point, it seems to be the way of things. Sure, I think we all agree with that. > That is why I first rasied the issue here to try and get a consensus > of what the expected behaviour should be, and then try and > understand how the documentation can be read to mean that... > > I am not sure where you get the 'pick "the best"' from the > documentation, it only says this for video, not audio! > OK, let's quote this: By default ffmpeg tries to pick the "best" stream of each type present in input files and add them to each output file. For video, this means the highest resolution, for audio the highest channel count. For subtitle it's simply the first subtitle stream. "pick the best stream of each type [...] For video, [...], for audio [...]" I'm not sure where the confusion comes from? > I am happy to contribute to better wording, once I understand what > the "correct" behaviour is. > Quickly looking at ffmpeg.c (around L3934), the audio stream picking is what we already suggested: it loops against all stream, pick the one with the most number of channels (and thus *ignore* "score" equality) and add it. So if you want to fix the documentation, you should IMO just reword in a sense to explicit the "only one pick per stream type". I'm not a native English speaker so I'm sure you will be able to propose a better wording than me; feel free to send a patch on ffmpeg-devel. If you don't, I'll take the time to of course. -- 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 phil_rhodes at rocketmail.com Tue Nov 15 17:50:50 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Tue, 15 Nov 2011 16:50:50 -0000 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > which WILL change colours, because yuv is a compressed colour space > (pure black is 16,16,16 and pure white is 235,235,235), Well, careful there Quickdraw - there are, unfortunately, RGB files out there in studio swing, and YUV files out there in full swing, even if most of them were created by software like ffmpeg that doesn't often give you much say in how things are handled. P From tim.nicholson at bbc.co.uk Tue Nov 15 18:03:53 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 15 Nov 2011 17:03:53 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <20111115163352.GE26053@leki> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> <20111115163352.GE26053@leki> Message-ID: <4EC29B79.5080605@bbc.co.uk> On 15/11/11 16:33, Cl?ment B?sch wrote: >[...] > > OK, let's quote this: > > By default ffmpeg tries to pick the "best" stream of each type present > in input files and add them to each output file. For video, this means > the highest resolution, for audio the highest channel count. For > subtitle it's simply the first subtitle stream. > > "pick the best stream of each type [...] For video, [...], for audio > [...]" > > I'm not sure where the confusion comes from? > I think the fact that, for audio a single channel, can be packaged either in its own stream, or be combined with others into a multi channel stream leads sometimes to thinking of one audio "item" as a single channel, and at other times as a single stream. The transition between these ways of thinking occurring subliminally and leading to confusion (That's my excuse anyway). >> I am happy to contribute to better wording, once I understand what >> the "correct" behaviour is. >> > > Quickly looking at ffmpeg.c (around L3934), the audio stream picking is > what we already suggested: it loops against all stream, pick the one with > the most number of channels (and thus *ignore* "score" equality) and add > it. > With your help I think I have it sorted in my own mind, now. But I think the doc could be improved. > So if you want to fix the documentation, you should IMO just reword in a > sense to explicit the "only one pick per stream type". I'm not a native > English speaker so I'm sure you will be able to propose a better wording > than me; feel free to send a patch on ffmpeg-devel. If you don't, I'll > take the time to of course. Never tried submitting a patch so will haver to bone up on how to do it... -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From ubitux at gmail.com Tue Nov 15 18:20:03 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 15 Nov 2011 18:20:03 +0100 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC29B79.5080605@bbc.co.uk> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> <20111115163352.GE26053@leki> <4EC29B79.5080605@bbc.co.uk> Message-ID: <20111115172003.GF26053@leki> On Tue, Nov 15, 2011 at 05:03:53PM +0000, Tim Nicholson wrote: > On 15/11/11 16:33, Cl?ment B?sch wrote: > >[...] > > > >OK, let's quote this: > > > > By default ffmpeg tries to pick the "best" stream of each type present > > in input files and add them to each output file. For video, this means > > the highest resolution, for audio the highest channel count. For > > subtitle it's simply the first subtitle stream. > > > >"pick the best stream of each type [...] For video, [...], for audio > >[...]" > > > >I'm not sure where the confusion comes from? > > > > I think the fact that, for audio a single channel, can be packaged > either in its own stream, or be combined with others into a multi > channel stream leads sometimes to thinking of one audio "item" as a > single channel, and at other times as a single stream. The > transition between these ways of thinking occurring subliminally and > leading to confusion (That's my excuse anyway). > The N mono channel to 1 stream can't really be autodetected, this is why the -map_channel option now exists btw :) > >>I am happy to contribute to better wording, once I understand what > >>the "correct" behaviour is. > >> > > > >Quickly looking at ffmpeg.c (around L3934), the audio stream picking is > >what we already suggested: it loops against all stream, pick the one with > >the most number of channels (and thus *ignore* "score" equality) and add > >it. > > > > With your help I think I have it sorted in my own mind, now. But I > think the doc could be improved. > Certainly, and such changes are always welcome. > >So if you want to fix the documentation, you should IMO just reword in a > >sense to explicit the "only one pick per stream type". I'm not a native > >English speaker so I'm sure you will be able to propose a better wording > >than me; feel free to send a patch on ffmpeg-devel. If you don't, I'll > >take the time to of course. > > Never tried submitting a patch so will haver to bone up on how to do it... > For such a trivial patch you can just write here the new paragraph, I'll write the patch and put your name in the credits. You can also try the "hard way": http://ffmpeg.org/developer.html#Submitting-patches-1 -- 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 danielbradham at gmail.com Tue Nov 15 19:57:39 2011 From: danielbradham at gmail.com (Dan Bradham) Date: Tue, 15 Nov 2011 13:57:39 -0500 Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 Message-ID: Hey I'm having some trouble converting IFF sequences to mpeg4. My input and output are as follows. ffmpeg -f image2 -r 24 -s 960x540 -i 'shot1.%04d.iff' test.mp4 ffmpeg version 0.7.6-rpmfusion, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 23 2011 17:45:03 with gcc 4.6.1 20110908 (Red Hat 4.6.1-9) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --extra-version=rpmfusion --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --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/lib64 --enable-runtime-cpudetect libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from 'shot1.%04d.iff': Duration: 00:00:11.00, start: 0.000000, bitrate: N/A Stream #0.0: Video: [0][0][0][0] / 0x0000, 960x540, 24 fps, 24 tbr, 24 tbn, 24 tbc File 'test.mp4' already exists. Overwrite ? [y/N] y [buffer @ 0x19403e0] Invalid pixel format string '-1' Error opening filters! Any ideas? From cehoyos at ag.or.at Tue Nov 15 22:34:22 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Nov 2011 21:34:22 +0000 (UTC) Subject: [FFmpeg-user] Audio Stream Mapping issue References: <4EC226F6.8010403@bbc.co.uk> <4EC2868C.4080305@bbc.co.uk> Message-ID: Tim Nicholson bbc.co.uk> writes: > You miss the documentation that states that the default mapping is:- > "For video, this means the highest resolution, for audio the highest > channel count" which says to me that one video of the highest resolution > is mapped and all audio (The highest count). The default is (and always was) to map one audio stream to the destination file, iirc there is a heuristic to choose the right stream which often works fine but sometimes chooses the wrong stream, see "audio streams" a few days ago. (If you believe the documentation is wrong - I did not check - please send a patch to -devel.) Carl Eugen From cehoyos at ag.or.at Tue Nov 15 22:37:30 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Nov 2011 21:37:30 +0000 (UTC) Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 References: Message-ID: Dan Bradham gmail.com> writes: > Hey I'm having some trouble converting IFF sequences to mpeg4. My input and > output are as follows. > > ffmpeg -f image2 -r 24 -s 960x540 -i 'shot1.%04d.iff' test.mp4 iff is not "image2" but - surprise - "IFF". It should not be necessary to specify the format, there is a probe function. (Without testing I would suspect that you should not specify the input resolution either.) Carl Eugen From cehoyos at ag.or.at Tue Nov 15 22:58:57 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 15 Nov 2011 21:58:57 +0000 (UTC) Subject: [FFmpeg-user] Audio Stream Mapping issue References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> <20111115163352.GE26053@leki> <4EC29B79.5080605@bbc.co.uk> <20111115172003.GF26053@leki> Message-ID: Cl?ment B?sch gmail.com> writes: > > Never tried submitting a patch so will haver to bone up on how to do it... > > For such a trivial patch you can just write here the new paragraph, I'll > write the patch and put your name in the credits. You can also try the > "hard way": http://ffmpeg.org/developer.html#Submitting-patches-1 As the old patch application monkey, I still accept patches made with git diff >patchfile,diff attached as plain/text. Carl Eugen From lists at arlomedia.com Tue Nov 15 23:03:22 2011 From: lists at arlomedia.com (Arlo Leach) Date: Tue, 15 Nov 2011 14:03:22 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: <4ED92E2E-0A29-43F9-9954-762F95151217@arlomedia.com> References: <9F9CBEC1-3466-4D4E-B271-61778B079731@arlomedia.com> <98A96D62-9E28-42E5-AE0D-57D3235012D1@arlomedia.com> <4ED92E2E-0A29-43F9-9954-762F95151217@arlomedia.com> Message-ID: Does anyone have any further insight on this? > I just ran this command: > > ffmpeg -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -ss 30 -t 30 -y band_formatted_sample_again.mp4 > > And got this result: > > http://www.arlomedia.com/projects/ffmpeg/band_formatted_sample_again.mp4 > > In QuickTime Player, I see no video for the first three seconds. I'm still re-encoding my videos when I just need to extract a sample clip, because I'm still losing the first view seconds of video with -vcodec copy. Thanks, -Arlo _______________________________ Arlo Leach http://arlomedia.com From dev at rarevision.com Tue Nov 15 23:55:19 2011 From: dev at rarevision.com (Thomas Worth) Date: Tue, 15 Nov 2011 14:55:19 -0800 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > which WILL change colours, because yuv is a compressed colour space > (pure black is 16,16,16 and pure white is 235,235,235), always has been, > always will be. This is only true for Rec. 709. Rec. 601 can be 16-235 or 0-255 (per JFIF spec), although SD broadcast is typically 16-235. In either case, anyone can force Rec. 709 to be full range, although encountering this would be unlikely. It is possible to see 601 material as full range, even in HD. Canon compresses their HDSLR footage with Rec. 601/full range. And yes, it's HD and not SD. And yes, this is very unusual. But it's a fact. From marc at hallmarcwebsites.com Wed Nov 16 00:02:56 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Tue, 15 Nov 2011 18:02:56 -0500 Subject: [FFmpeg-user] ProRes to x264 Message-ID: I have used the following command ffmpeg -y -i source.mov -sameq new.mp4 I have tried a few different flags in the command line and yet I still get a video that when played through QT plugin well, the audio plays fine yet the video seems to be in slow motion. Complete output /$ ffmpeg -y -i /home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/Hamilton-Beach-Sc oop-30.mov -sameq /home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/atest.mp4 ffmpeg version N-34855-gc8136eb, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 14 2011 16:03:31 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr/local/hgffmpeg --enable-shared --enable-nonfree --enable-avfilter --enable-filter=movie --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libxvid --extra-cflags=-I/usr/local/hgffmpeg/include/ --extra-ldflags=-L/usr/local/hgffmpeg/lib --enable-decoder=ac3 --enable-decoder=asv1 --enable-decoder=asv2 --enable-decoder=flac --enable-decoder=wmv1 --enable-decoder=wmv2 --enable-decoder=wmv3 --enable-decoder=mpeg1video --enable-decoder=mpeg2video --enable-decoder=flv --enable-decoder=fraps --enable-decoder=h263 --enable-decoder=h264 --enable-decoder=libgsm --enable-decoder=mjpeg --enable-decoder=mpeg4 --enable-decoder=mpeg4aac --enable-decoder=mpegvideo --enable-decoder=mpeg4aac --enable-decoder=msmpeg4v1 --enable-decoder=msmpeg4v2 --enable-decoder=msmpeg4v3 --enable-decoder=pcm_alaw --enable-decoder=pcm_mulaw --enable- libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 23976.00 (23976/1) -> 23.98 (2997/125) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/Hamilton-Beach-S coop-30.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-05 14:26:42 Duration: 00:00:29.98, start: 0.000000, bitrate: 104608 kb/s Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le, 1920x1080, 102281 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 23976 tbn, 23976 tbc Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 2 channels, s32, 2304 kb/s Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2011-10-05 14:27:11 handler_name : ?Apple Alias Data Handler Incompatible pixel format 'yuv422p10le' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x97e8080] w:1920 h:1080 pixfmt:yuv422p10le tb:1/1000000 sar:1/1 sws_param: [buffersink @ 0x97e82e0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 0x97d3680] w:1920 h:1080 fmt:yuv422p10le -> w:1920 h:1080 fmt:yuv420p flags:0x4 Incompatible sample format 's32' for codec 'libfaac', auto-selecting format 's16' [libx264 @ 0x97f7180] using SAR=1/1 [libx264 @ 0x97f7180] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x97f7180] profile High, level 4.0 [libx264 @ 0x97f7180] 264 - core 119 r2106 07efeb4 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - 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=12 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 Output #0, mp4, to '/home/hallmarc/public_html/wrdp/wp-content/uploads/2011/11/atest.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-05 14:26:42 encoder : Lavf53.20.0 Stream #0:0(eng): Video: h264 (![0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 2997 tbn, 23.98 tbc Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: aac (@[0][0][0] / 0x0040), 48000 Hz, 2 channels, s16, 128 kb/s Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream mapping: Stream #0:0 -> #0:0 (prores -> libx264) Stream #0:1 -> #0:1 (pcm_s24le -> libfaac) Press [q] to stop, [?] for help frame= 9 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 18 fps= 16 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 26 fps= 16 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 35 fps= 16 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 43 fps= 16 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 47 fps= 14 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 54 fps= 14 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 63 fps= 15 q=28.0 size= 99kB time=00:00:00.20 bitrate=3883.8kbits/s frame= 72 fps= 15 q=28.0 size= 189kB time=00:00:00.58 bitrate=2654.5kbits/s frame= 81 fps= 15 q=28.0 size= 304kB time=00:00:00.95 bitrate=2592.5kbits/s frame= 90 fps= 15 q=28.0 size= 394kB time=00:00:01.33 bitrate=2421.2kbits/s frame= 98 fps= 15 q=28.0 size= 515kB time=00:00:01.66 bitrate=2529.7kbits/s frame= 107 fps= 15 q=28.0 size= 615kB time=00:00:02.04 bitrate=2464.1kbits/s frame= 116 fps= 16 q=28.0 size= 747kB time=00:00:02.41 bitrate=2529.1kbits/s frame= 124 fps= 15 q=28.0 size= 858kB time=00:00:02.75 bitrate=2552.1kbits/s frame= 132 fps= 15 q=28.0 size= 992kB time=00:00:03.08 bitrate=2631.9kbits/s frame= 141 fps= 16 q=28.0 size= 1183kB time=00:00:03.46 bitrate=2800.4kbits/s frame= 150 fps= 16 q=28.0 size= 1315kB time=00:00:03.83 bitrate=2806.8kbits/s frame= 159 fps= 16 q=28.0 size= 1430kB time=00:00:04.21 bitrate=2781.5kbits/s frame= 168 fps= 16 q=28.0 size= 1533kB time=00:00:04.58 bitrate=2737.4kbits/s frame= 177 fps= 16 q=28.0 size= 1600kB time=00:00:04.96 bitrate=2640.7kbits/s frame= 186 fps= 16 q=28.0 size= 1712kB time=00:00:05.33 bitrate=2626.2kbits/s frame= 195 fps= 16 q=28.0 size= 1805kB time=00:00:05.71 bitrate=2587.1kbits/s frame= 204 fps= 16 q=28.0 size= 1908kB time=00:00:06.08 bitrate=2566.7kbits/s frame= 213 fps= 16 q=28.0 size= 2012kB time=00:00:06.46 bitrate=2549.9kbits/s frame= 222 fps= 16 q=28.0 size= 2080kB time=00:00:06.84 bitrate=2490.9kbits/s frame= 231 fps= 16 q=28.0 size= 2181kB time=00:00:07.21 bitrate=2476.0kbits/s frame= 240 fps= 16 q=28.0 size= 2258kB time=00:00:07.59 bitrate=2436.4kbits/s frame= 249 fps= 16 q=28.0 size= 2343kB time=00:00:07.96 bitrate=2409.5kbits/s frame= 258 fps= 16 q=28.0 size= 2425kB time=00:00:08.34 bitrate=2381.3kbits/s frame= 267 fps= 16 q=28.0 size= 2531kB time=00:00:08.71 bitrate=2378.4kbits/s frame= 275 fps= 16 q=28.0 size= 2606kB time=00:00:09.05 bitrate=2358.9kbits/s frame= 285 fps= 16 q=28.0 size= 2686kB time=00:00:09.46 bitrate=2324.0kbits/s frame= 294 fps= 16 q=28.0 size= 2804kB time=00:00:09.84 bitrate=2333.8kbits/s frame= 304 fps= 16 q=28.0 size= 2983kB time=00:00:10.26 bitrate=2381.9kbits/s frame= 314 fps= 16 q=28.0 size= 3054kB time=00:00:10.67 bitrate=2342.8kbits/s frame= 323 fps= 16 q=28.0 size= 3116kB time=00:00:11.05 bitrate=2309.7kbits/s frame= 332 fps= 16 q=28.0 size= 3179kB time=00:00:11.42 bitrate=2278.7kbits/s frame= 341 fps= 16 q=28.0 size= 3212kB time=00:00:11.80 bitrate=2229.5kbits/s frame= 350 fps= 16 q=28.0 size= 3246kB time=00:00:12.17 bitrate=2183.2kbits/s frame= 359 fps= 16 q=28.0 size= 3275kB time=00:00:12.55 bitrate=2137.3kbits/s frame= 368 fps= 16 q=28.0 size= 3329kB time=00:00:12.92 bitrate=2109.1kbits/s frame= 377 fps= 17 q=28.0 size= 3369kB time=00:00:13.30 bitrate=2074.4kbits/s frame= 384 fps= 16 q=28.0 size= 3398kB time=00:00:13.59 bitrate=2047.0kbits/s frame= 393 fps= 16 q=28.0 size= 3438kB time=00:00:13.97 bitrate=2015.7kbits/s frame= 402 fps= 16 q=28.0 size= 3469kB time=00:00:14.34 bitrate=1980.8kbits/s frame= 410 fps= 16 q=28.0 size= 3490kB time=00:00:14.68 bitrate=1947.1kbits/s frame= 419 fps= 16 q=28.0 size= 3521kB time=00:00:15.05 bitrate=1915.5kbits/s frame= 428 fps= 16 q=28.0 size= 3555kB time=00:00:15.43 bitrate=1887.0kbits/s frame= 437 fps= 16 q=28.0 size= 3645kB time=00:00:15.80 bitrate=1889.0kbits/s frame= 446 fps= 16 q=28.0 size= 3717kB time=00:00:16.18 bitrate=1881.6kbits/s frame= 455 fps= 16 q=28.0 size= 3784kB time=00:00:16.55 bitrate=1871.9kbits/s frame= 463 fps= 16 q=28.0 size= 3837kB time=00:00:16.89 bitrate=1861.0kbits/s frame= 473 fps= 17 q=28.0 size= 3960kB time=00:00:17.30 bitrate=1874.0kbits/s frame= 483 fps= 17 q=28.0 size= 4074kB time=00:00:17.72 bitrate=1882.7kbits/s frame= 490 fps= 16 q=28.0 size= 4139kB time=00:00:18.01 bitrate=1881.8kbits/s frame= 498 fps= 16 q=28.0 size= 4200kB time=00:00:18.35 bitrate=1874.8kbits/s frame= 507 fps= 16 q=28.0 size= 4297kB time=00:00:18.72 bitrate=1879.5kbits/s frame= 517 fps= 17 q=28.0 size= 4360kB time=00:00:19.14 bitrate=1865.7kbits/s frame= 528 fps= 17 q=28.0 size= 4490kB time=00:00:19.60 bitrate=1876.5kbits/s frame= 540 fps= 17 q=28.0 size= 4589kB time=00:00:20.10 bitrate=1869.9kbits/s frame= 552 fps= 17 q=28.0 size= 4679kB time=00:00:20.60 bitrate=1860.5kbits/s frame= 564 fps= 17 q=28.0 size= 4766kB time=00:00:21.10 bitrate=1850.1kbits/s frame= 575 fps= 17 q=28.0 size= 4839kB time=00:00:21.56 bitrate=1838.6kbits/s frame= 586 fps= 17 q=28.0 size= 4969kB time=00:00:22.02 bitrate=1848.4kbits/s frame= 596 fps= 17 q=28.0 size= 5044kB time=00:00:22.43 bitrate=1841.3kbits/s frame= 606 fps= 17 q=28.0 size= 5158kB time=00:00:22.85 bitrate=1848.6kbits/s frame= 616 fps= 17 q=28.0 size= 5225kB time=00:00:23.27 bitrate=1839.1kbits/s frame= 624 fps= 17 q=28.0 size= 5280kB time=00:00:23.60 bitrate=1832.3kbits/s frame= 634 fps= 17 q=28.0 size= 5360kB time=00:00:24.02 bitrate=1827.7kbits/s frame= 643 fps= 17 q=28.0 size= 5401kB time=00:00:24.39 bitrate=1813.2kbits/s frame= 652 fps= 17 q=28.0 size= 5490kB time=00:00:24.77 bitrate=1815.4kbits/s frame= 660 fps= 17 q=28.0 size= 5595kB time=00:00:25.10 bitrate=1825.4kbits/s frame= 668 fps= 17 q=28.0 size= 5785kB time=00:00:25.44 bitrate=1862.5kbits/s frame= 677 fps= 17 q=28.0 size= 6004kB time=00:00:25.81 bitrate=1905.1kbits/s frame= 688 fps= 17 q=28.0 size= 6183kB time=00:00:26.27 bitrate=1927.8kbits/s frame= 696 fps= 17 q=28.0 size= 6318kB time=00:00:26.60 bitrate=1944.9kbits/s frame= 705 fps= 17 q=28.0 size= 6461kB time=00:00:26.98 bitrate=1961.3kbits/s frame= 715 fps= 17 q=28.0 size= 6657kB time=00:00:27.40 bitrate=1990.1kbits/s frame= 719 fps= 16 q=-1.0 Lsize= 7202kB time=00:00:29.90 bitrate=1973.0kbits/s video:6896kB audio:287kB global headers:0kB muxing overhead 0.275473% [libx264 @ 0x97f7180] frame I:19 Avg QP:19.25 size: 50514 [libx264 @ 0x97f7180] frame P:560 Avg QP:23.44 size: 9863 [libx264 @ 0x97f7180] frame B:140 Avg QP:27.04 size: 4124 [libx264 @ 0x97f7180] consecutive B-frames: 67.7% 18.6% 0.8% 12.8% [libx264 @ 0x97f7180] mb I I16..4: 38.8% 55.8% 5.4% [libx264 @ 0x97f7180] mb P I16..4: 2.9% 4.3% 0.4% P16..4: 19.9% 3.4% 2.1% 0.0% 0.0% skip:67.0% [libx264 @ 0x97f7180] mb B I16..4: 0.3% 0.3% 0.1% B16..8: 26.3% 1.8% 0.1% direct: 0.3% skip:70.9% L0:47.4% L1:50.7% BI: 1.9% [libx264 @ 0x97f7180] 8x8 transform intra:55.9% inter:86.5% [libx264 @ 0x97f7180] coded y,uvDC,uvAC intra: 36.6% 30.5% 10.4% inter: 5.9% 4.1% 0.1% [libx264 @ 0x97f7180] i16 v,h,dc,p: 50% 24% 5% 21% [libx264 @ 0x97f7180] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 16% 40% 3% 5% 4% 7% 4% 4% [libx264 @ 0x97f7180] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 25% 15% 4% 7% 5% 7% 4% 3% [libx264 @ 0x97f7180] i8c dc,h,v,p: 77% 11% 10% 2% [libx264 @ 0x97f7180] Weighted P-Frames: Y:0.9% UV:0.2% [libx264 @ 0x97f7180] ref P L0: 62.5% 14.6% 14.4% 8.6% 0.0% [libx264 @ 0x97f7180] ref B L0: 65.3% 27.3% 7.4% [libx264 @ 0x97f7180] ref B L1: 89.3% 10.7% [libx264 @ 0x97f7180] kb/s:1883.56 From danielbradham at gmail.com Wed Nov 16 00:09:46 2011 From: danielbradham at gmail.com (Dan Bradham) Date: Tue, 15 Nov 2011 18:09:46 -0500 Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 In-Reply-To: References: Message-ID: Thanks for pointing me toward the IFF format Carl, I had overlooked that. However, my attempts at inputting an IFF sequence using no format instead of image2 yields: "no such file or directory," leading me to believe that the syntax %0Nd is only enabled under the format image2. It's likely that I'm wrong and perhaps you could point me in the right direction with an example command line. On Tue, Nov 15, 2011 at 4:37 PM, Carl Eugen Hoyos wrote: > Dan Bradham gmail.com> writes: > > > Hey I'm having some trouble converting IFF sequences to mpeg4. My input > and > > output are as follows. > > > > ffmpeg -f image2 -r 24 -s 960x540 -i 'shot1.%04d.iff' test.mp4 > > iff is not "image2" but - surprise - "IFF". > It should not be necessary to specify the format, there is a probe > function. > (Without testing I would suspect that you should not specify the input > resolution either.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From 13760746839 at 163.com Wed Nov 16 10:26:33 2011 From: 13760746839 at 163.com (=?GBK?B?0KSyqA==?=) Date: Wed, 16 Nov 2011 17:26:33 +0800 (CST) Subject: [FFmpeg-user] Problems with lxf demuxer In-Reply-To: References: <7913e1af.b337.1339fd4d781.Coremail.13760746839@163.com> Message-ID: <2f713414.20187.133abb2f97d.Coremail.13760746839@163.com> Please see the following link,it's a sample LXF clip of Leitch Version 1,And the pdf document is format of it.who are interested please have a look and update the code. Sample Video Clip:http://www.datafilehost.com/download-9748c4b9.html New format document:http://www.datafilehost.com/download-730294fd.html ? 2011-11-15 06:34:56?"Carl Eugen Hoyos" ??? >Hi! > >?? <13760746839 163.com> writes: > >> I noticed that for Leitch Version 1,when use ffplay, it will resulted in the >> following error message: >> [lxf @ 02F91840] checksum error >> [lxf @ 02F91840] expected 120 B size header, got 0 >> ... :Invalid data found when processing input > >Please provide a sample. > >Carl Eugen > >_______________________________________________ >ffmpeg-user mailing list >ffmpeg-user at ffmpeg.org >http://ffmpeg.org/mailman/listinfo/ffmpeg-user ???163????????? ZB09_108_LXF_V1.lxf (10.7M, 2011?11?30? 8:57 ??) ?? From cehoyos at ag.or.at Wed Nov 16 10:50:27 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 09:50:27 +0000 (UTC) Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 References: Message-ID: Dan Bradham gmail.com> writes: > However, my attempts at inputting an IFF sequence using no format instead > of image2 yields: "no such file or directory," leading me to believe that > the syntax %0Nd is only enabled under the format image2. (Command line and complete, uncut output missing.) How does (the beginning of) ls -l *.iff look like? Carl Eugen From cehoyos at ag.or.at Wed Nov 16 10:54:38 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 09:54:38 +0000 (UTC) Subject: [FFmpeg-user] ProRes to x264 References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > I have used the following command > ffmpeg -y -i source.mov -sameq new.mp4 > I have tried a few different flags in the command line and yet I still get a > video that when played through QT plugin well, the audio plays fine yet the > video seems to be in slow motion. Since it is unclear to me if you have a problem that is both reproducible with the original and the new file or only the new file: Does the original file play fine with ffplay? with mplayer? With mplayer -speed 0.3? (=Is it a performance problem you see?) What about ffmpeg -i source.mov -s 480x260 out.avi? Or ffmpeg -i source.mov -s 480x260 -vcodec mpeg4 -strict experimental -acodec aac out.mov Does the resulting file play ok? Carl Eugen From fox at orbitalfox.com Wed Nov 16 11:36:26 2011 From: fox at orbitalfox.com (fox) Date: Wed, 16 Nov 2011 10:36:26 +0000 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> Message-ID: <20111116103626.1ab10e1f@apollon> On Thu, 10 Nov 2011 16:12:03 +0000 (UTC) Carl Eugen Hoyos wrote: > That is an intentionally broken (and unsupported) version. > Please see http://ffmpeg.org/download.html > > Carl Eugen Carl, you have said that in the past, and once more I had asked -- what is the intention? Why is it intentionally broken? I never received a reply on this. Fox From marc at hallmarcwebsites.com Wed Nov 16 13:04:35 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 16 Nov 2011 07:04:35 -0500 Subject: [FFmpeg-user] ProRes to x264 In-Reply-To: References: Message-ID: Don't want an .mov or .avi or anything but .mp4 It doesn't play in Quicktime as expected and as I described when I posed the question. I realize that ProRes codec is still new to ffmpeg. What I am looking for is a general answer along the lines of "You need to -enable-version-2" or you need to include these flags in the command line. That kind of answer. From danielbradham at gmail.com Wed Nov 16 13:34:36 2011 From: danielbradham at gmail.com (Dan Bradham) Date: Wed, 16 Nov 2011 07:34:36 -0500 Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 In-Reply-To: References: Message-ID: It appears that IFF sequences are unsupported, so I took an extra step and just converted to png. This allowed me to simply use the image2 format. Thanks for your help. On Wed, Nov 16, 2011 at 4:50 AM, Carl Eugen Hoyos wrote: > Dan Bradham gmail.com> writes: > > > However, my attempts at inputting an IFF sequence using no format instead > > of image2 yields: "no such file or directory," leading me to believe that > > the syntax %0Nd is only enabled under the format image2. > > (Command line and complete, uncut output missing.) > How does (the beginning of) ls -l *.iff look like? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rogerdpack2 at gmail.com Wed Nov 16 16:18:25 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 16 Nov 2011 08:18:25 -0700 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: <20111116103626.1ab10e1f@apollon> References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> Message-ID: > Carl Eugen Hoyos wrote: >> That is an intentionally broken (and unsupported) version. >> Please see http://ffmpeg.org/download.html >> >> Carl Eugen > > Carl, you have said that in the past, and once more I had asked -- what > is the intention? Why is it intentionally broken? > > I never received a reply on this. I think he's saying the version you were using was old and unsupported, and to download a newer one. -r From marc at hallmarcwebsites.com Wed Nov 16 17:25:21 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 16 Nov 2011 11:25:21 -0500 Subject: [FFmpeg-user] x264 and anamophic play back Message-ID: Does x264 have playback issues if you try to play it back at a viewing size that is different than the actual size? Say displaying a 1920x1080 at 983x553 From fox at orbitalfox.com Wed Nov 16 17:25:53 2011 From: fox at orbitalfox.com (fox) Date: Wed, 16 Nov 2011 16:25:53 +0000 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> Message-ID: <20111116162553.216aea7e@apollon> On Wed, 16 Nov 2011 08:18:25 -0700 Roger Pack wrote: > I think he's saying the version you were using was old and > unsupported, and to download a newer one. Personally Im using an SVN version. But Ive heard stuff regarding the split of ffmpeg dev team and backings by distros, I would like to know what he means by "intentionally". I dont think he just meant to get the latest version. For the record, I am not involved with any of the distros mentioned other than being a user. Fox From cehoyos at ag.or.at Wed Nov 16 17:30:56 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 16:30:56 +0000 (UTC) Subject: [FFmpeg-user] ProRes to x264 References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > Don't want an .mov or .avi or anything but .mp4 I know, but I won't be able to help you if cannot provide a sample as long as I don't understand where your problem comes from. There is nothing "new" about the ProRes decoder that has any implications on usage. Carl Eugen From cehoyos at ag.or.at Wed Nov 16 17:36:58 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 16:36:58 +0000 (UTC) Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 References: Message-ID: Dan Bradham gmail.com> writes: > It appears that IFF sequences are unsupported, You are right;-( I created ticket #661. Carl Eugen From cehoyos at ag.or.at Wed Nov 16 17:44:40 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 16:44:40 +0000 (UTC) Subject: [FFmpeg-user] fixing videos that don't start with keyframes References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> <20111116162553.216aea7e@apollon> Message-ID: fox orbitalfox.com> writes: > On Wed, 16 Nov 2011 08:18:25 -0700 > Roger Pack gmail.com> wrote: > > > I think he's saying the version you were using was old and > > unsupported, and to download a newer one. > > Personally Im using an SVN version. That is definitely unsupported. > But Ive heard stuff regarding the split of ffmpeg dev team and backings by > distros, Only one packager is supporting the traitors who had tried to steal FFmpeg. > I would like to know what he means by "intentionally". I dont think he just > meant to get the latest version. It is very strongly recommended that you use current git head, all bugs (etc.) are first fixed in git head (and only later in release branches), there is nothing "stable" in the releases that brings any advantage to users. (Packagers claim they need releases, that is why we provide them, users should not download them.) If you look at our bug tracker, you will find fixed bugs (and enhancements) there. Some of the fixes were made in the fork, some bugs may never have been reproducible with the fork. But most fixes are missing from the fork, making it an intentionally broken version of FFmpeg that - due to insufficient manpower - we cannot support because it contains >100 known bugs not present in FFmpeg. Carl Eugen From cehoyos at ag.or.at Wed Nov 16 17:54:56 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 16:54:56 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg for rate adaptive experiments References: Message-ID: Ingemar Johansson S ericsson.com> writes: > I am looking for a simple way to visualize an encoding scenario where the > target bitrate varies with time. The typical use case is a rate adaptive video > for a realtime interactive service where the bitrate is determined by channel > conditions. Typically it should be possible to change the bitrate once per > second or possibly more often. > I would like to visualize packet loss and delay as well, in short I want to > demo the cases > 1) No rate adaptation e.g 384kbps > 2) Rate adaptation e.g between 64 and 384kpbs > The demo should give answers to the questions how is video quality affect by > the packet delay packet drops > and the bitrate as a reult of the two approaches above, I get the the bitrate > and delay traces from a proprietary system simulator. > > Is there a possiblity to use ffmpeg for this purpose ? Why not? Afaik, you can set the target bitrate for every frame (that may not make much sense, but at least for every gop should be no problem), it should be sufficient to program your encoder (that uses libavcodec) to feed the requested bitrates from the system simulator to the rate control (that may or may not work sufficiently exact for your requirements). Carl Eugen From cehoyos at ag.or.at Wed Nov 16 17:57:24 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 16:57:24 +0000 (UTC) Subject: [FFmpeg-user] ProRes to x264 References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > Don't want an .mov or .avi or anything but .mp4 > It doesn't play in Quicktime as expected and as I described when I posed the > question. What I completely missed: QuickTime generally fails to play generic mp4 files, so I strongly believe you should use mov. But in any case, you can substitute "mov" in my command line with "mp4" and the only difference in the resulting file is that it is not explicitly marked as QuickTime-compatible iirc. Carl Eugen From phil_rhodes at rocketmail.com Wed Nov 16 18:03:28 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Wed, 16 Nov 2011 17:03:28 -0000 Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 In-Reply-To: References: Message-ID: >> It appears that IFF sequences are unsupported So are Amiga ANIMs. And CDXL. Both of which are a shame. I think they both used simple delta compression. I have some ANIMs I'd love to get at. P From cehoyos at ag.or.at Wed Nov 16 18:23:35 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 17:23:35 +0000 (UTC) Subject: [FFmpeg-user] Converting IFF image sequence to mpeg4 References: Message-ID: Phil Rhodes rocketmail.com> writes: > >> It appears that IFF sequences are unsupported You may misunderstand (or I have cut the quote too savagely): IFF is not unsupported, reading IFF pictures as a range as it is possible with general images is not implemented. > So are Amiga ANIMs. And CDXL. Both of which are a shame. I think they both > used simple delta compression. Please upload sample and provide failing command lines and complete, uncut console output. Thank you for your continuing support, Carl Eugen From marc at hallmarcwebsites.com Wed Nov 16 18:27:34 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 16 Nov 2011 12:27:34 -0500 Subject: [FFmpeg-user] ProRes to x264 In-Reply-To: References: Message-ID: > > What I completely missed: > QuickTime generally fails to play generic mp4 files, so I strongly believe you > should use mov. > But in any case, you can substitute "mov" in my command line with "mp4" > and the only difference in the resulting file is that it is not explicitly marked as > QuickTime-compatible iirc. > > Carl Eugen > Bull frogs! I have been playing plenty of mp4 files with Quicktime without issue. mp4 is just the dang container anyway! Please stop muddying the list with nonsense. The other huge difference between mp4 and mov is that I can set the mp4 container to start playing while the file is downloading. AFAIK this cannot be done with mov. Not to put too fine of a point on it but I have had more issues with mov and Quicktime than I care to count! [>] Now, what I have found is that using the x264 codec in the mp4 container, and I cannot speak on any other containers suited to x264 as I have not tested them, is that it doesn't play well when you "squish" it into a smaller viewport than it is coded for. From rogerdpack2 at gmail.com Wed Nov 16 19:00:44 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 16 Nov 2011 11:00:44 -0700 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > which WILL change colours, because yuv is a compressed colour space > (pure black is 16,16,16 and pure white is 235,235,235), always has been, > always will be. Further, within the YUV colour space, there are > different gamut curves - rec601 & rec709 for example. > > Please read http://en.wikipedia.org/wiki/YUV for a more indepth > explaination. So my first thought is "does this mean that ffplay should stretch its pixel values before displaying them, from [16,235] to [0, 255]" (which is the same as converting them to full swing...I guess?) At least then the bug is only in ffplay, which is...better than having it elsewhere I suppose. -roger- From xuyuand at 163.com Wed Nov 16 11:37:39 2011 From: xuyuand at 163.com (xuyuand) Date: Wed, 16 Nov 2011 18:37:39 +0800 (CST) Subject: [FFmpeg-user] can't set x264 profile Message-ID: <25fd83d2.19c27.133abf413d0.Coremail.xuyuand@163.com> hi, I use lastest ffmpeg:ffmpeg-git-985e768-win32-shared.7z,I use this command: ffmpeg -threads 4 -i "E:\clipts\111.rmvb" -r 25.00 -profile main -level 31 -vcodec libx264 -s 720x576 -x264opts keyint=123:min-keyint=20:vbv-maxrate=2000:vbv-bufsize=300:vbv-init=0.9:nal-hrd=cbr:bitrate=2000:bframes=3:videoformat=pal -acodec mp2 -ab 128k -ar 48000 -ac 2 -y d:\h264.ts It will appear next error message, Input #0, rm, from 'E:\clipts\1111.rmvb': Metadata: title : author : copyright : comment : Duration: 00:42:47.19, start: 0.000000, bitrate: 815 kb/s Stream #0:0: Audio: cook (cook / 0x6B6F6F63), 44100 Hz, stereo, flt, 44 kb/s Stream #0:1: Video: rv40 (RV40 / 0x30345652), yuv420p, 1024x576, 717 kb/s, 25 fps, 25 tbr, 1k tb n, 25 tbc Stream #0:2: Data: none [buffer @ 00BB3C20] w:1024 h:576 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [scale @ 003EA580] w:1024 h:576 fmt:yuv420p -> w:720 h:576 fmt:yuv420p flags:0x4 Incompatible sample format 'flt' for codec 'mp2', auto-selecting format 's16' [libx264 @ 00B98020] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 00B98020] profile Main, level 3.1 [NULL @ 00BBFB00] [Eval @ 0022DB28] Undefined constant or missing '(' in 'main' [NULL @ 00BBFB00] Unable to parse option value "main" [NULL @ 00BBFB00] Error setting option profile to value main. Output #0, mpegts, to 'd:\h264.ts': Metadata: title : author : copyright : comment : Stream #0:0: Video: h264, yuv420p, 720x576, q=-1--1, 2000 kb/s, 90k tbn, 25 tbc Stream #0:1: Audio: none, 48000 Hz, 2 channels, s16, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (rv40 -> libx264) Stream #0:0 -> #0:1 (cook -> mp2) Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height please tell me what to do? From rhodri at kynesim.co.uk Wed Nov 16 19:38:28 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Wed, 16 Nov 2011 18:38:28 -0000 Subject: [FFmpeg-user] can't set x264 profile In-Reply-To: <25fd83d2.19c27.133abf413d0.Coremail.xuyuand@163.com> References: <25fd83d2.19c27.133abf413d0.Coremail.xuyuand@163.com> Message-ID: On Wed, 16 Nov 2011 10:37:39 -0000, xuyuand wrote: > ffmpeg -threads 4 -i "E:\clipts\111.rmvb" -r 25.00 -profile main -level ^^^^^^^^^^^^^ > 31 -vcodec libx264 -s 720x576 -x264opts > keyint=123:min-keyint=20:vbv-maxrate=2000:vbv-bufsize=300:vbv-init=0.9:nal-hrd=cbr:bitrate=2000:bframes=3:videoformat=pal > -acodec mp2 -ab 128k -ar 48000 -ac 2 -y d:\h264.ts I believe you need "-vprofile main" these days, not "-profile main". -- Rhodri James Kynesim Ltd From marc at hallmarcwebsites.com Wed Nov 16 22:28:44 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 16 Nov 2011 16:28:44 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality Message-ID: Really beginning to hate transcoding. I have been able to successfully keep the video quality and deal with the resizing issue of x264 by using mpeg4 codec instead. Now, the issue I am dealing with is keeping the audio quality the same! libfaac just doesn't seem to be able to match the audio quality of pcm_s24le. So I am trying use it as the encoder and ffmpeg (yes the latest version) keeps reporting this error: Could not write header for output file #0 (incorrect codec parameters ?) the command lines I have tried are: ffmpeg -y -i INPUT.mov -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec pcm_s24le -ar 44100 -ab 2304 -ac 2 OUTPUT.mp4 ffmpeg -y -i INPUT.mov -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec pcm_s24le 2 OUTPUT.mp4 etc etc And the same error comes up. Complete output below: ffmpeg version N-34855-gc8136eb, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 14 2011 16:03:31 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51) configuration: --prefix=/usr/local/hgffmpeg --enable-shared --enable-nonfree --enable-avfilter --enable-filter=movie --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libxvid --extra-cflags=-I/usr/local/hgffmpeg/include/ --extra-ldflags=-L/usr/local/hgffmpeg/lib --enable-decoder=ac3 --enable-decoder=asv1 --enable-decoder=asv2 --enable-decoder=flac --enable-decoder=wmv1 --enable-decoder=wmv2 --enable-decoder=wmv3 --enable-decoder=mpeg1video --enable-decoder=mpeg2video --enable-decoder=flv --enable-decoder=fraps --enable-decoder=h263 --enable-decoder=h264 --enable-decoder=libgsm --enable-decoder=mjpeg --enable-decoder=mpeg4 --enable-decoder=mpeg4aac --enable-decoder=mpegvideo --enable-decoder=mpeg4aac --enable-decoder=msmpeg4v1 --enable-decoder=msmpeg4v2 --enable-decoder=msmpeg4v3 --enable-decoder=pcm_alaw --enable-decoder=pcm_mulaw --enable- libavutil 51. 24. 1 / 51. 24. 1 libavcodec 53. 33. 0 / 53. 33. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 23976.00 (23976/1) -> 23.98 (2997/125) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Hamilton-Beach-Scoop-30.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-05 14:26:42 Duration: 00:00:29.98, start: 0.000000, bitrate: 104608 kb/s Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le, 1920x1080, 102281 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 23976 tbn, 23976 tbc Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 2 channels, s32, 2304 kb/s Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74) Metadata: creation_time : 2011-10-05 14:27:11 handler_name : ?Apple Alias Data Handler [mp4 @ 0x8e5da00] track 0: could not find tag, codec not currently supported in container Output #0, mp4, to 'atest.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-10-05 14:26:42 encoder : Lavf53.20.0 Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 102281 kb/s, 23.98 fps, 90k tbn, 23976 tbc Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 2 channels, 2304 kb/s Metadata: creation_time : 2011-10-05 14:26:42 handler_name : ?Apple Alias Data Handler 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 ?) From cehoyos at ag.or.at Wed Nov 16 23:36:32 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 16 Nov 2011 22:36:32 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > ffmpeg -y -i INPUT.mov -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec pcm_s24le > -ar 44100 -ab 2304 -ac 2 OUTPUT.mp4 > ffmpeg -y -i INPUT.mov -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec pcm_s24le > 2 OUTPUT.mp4 mp4 does not support pcm audio (mov does). [...] > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Stream #0:1 -> #0:1 (copy) For future reports, please try to match command line and output. If you still see A/V desync when reading ProRes, please upload a sample, the problem is not known. Carl Eugen From marc at hallmarcwebsites.com Thu Nov 17 00:58:41 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 16 Nov 2011 18:58:41 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > HallMarc Websites hallmarcwebsites.com> writes: > > > ffmpeg -y -i INPUT.mov -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec > > pcm_s24le -ar 44100 -ab 2304 -ac 2 OUTPUT.mp4 ffmpeg -y -i INPUT.mov > > -vcodec mpeg4 -b:v 1883k -r 23.976 -acodec pcm_s24le > > 2 OUTPUT.mp4 > > mp4 does not support pcm audio (mov does). > Um I beg to differ. I have an mp4 container with pcm-s24be and it plays beautifully. However, it was done with a piece of software for Windows. While it uses ffmpeg and almost all of the codecs; I have no way of seeing what and how it is going about this. So, I know it can be done. The question is how. Stop asking me for the file. I cannot give it to you. It is a commercial and is copyright protected. Not a fan of getting sued. It is the only one I have where ProRes was used in the post. Look, in all honesty, I have asked for help on this list a number of times. Not once have I been given a working solution. The documentation for ffmpeg is sparse at best. Can any of you point me in the direction of good books to learn ffmpeg? Marc Hall HallMarc Websites www.HallMarcWebsites.com 610-446-3346 From cehoyos at ag.or.at Thu Nov 17 01:16:12 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Nov 2011 00:16:12 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > > mp4 does not support pcm audio (mov does). > > Um I beg to differ. I have an mp4 container with pcm-s24be and it plays > beautifully. However, it was done with a piece of software for Windows. > While it uses ffmpeg and almost all of the codecs; I have no way of seeing > what and how it is going about this. > > So, I know it can be done. The question is how. Just ignore the mp4 specification that enumerates the allowed codecs and use mov tags. Carl Eugen From phil_rhodes at rocketmail.com Thu Nov 17 01:37:14 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Thu, 17 Nov 2011 00:37:14 -0000 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > So my first thought is "does this mean that ffplay should stretch its > pixel values before displaying them, from [16,235] to [0, 255]" No, because you can't know what range the input is using. There is often no way to indicate this in a file, and even if there is, the contents of some files are not what the file states them to be - usually because of exactly this situation occurring in some previous processing step. The only way to fix this is to allow the user to specify what the input and output is. Last time I looked at this - maybe up to a year ago? - there was the facility in ffmpeg's codebase to ask for this sort of luma scaling, but no way to specify it from the command line. I'm not sure if this is still the case. > (which is the same as converting them to full swing...I guess?) At > least then the bug is only in ffplay, which is...better than having it > elsewhere I suppose. It's not really a bug, it's an unknowable fact about the input and output video. The only way to fix it is to ask the user what they think the input is and what the output should be. Computer software has traditionally shown an almost wilful ignorance of the fact that 0 isn't black to some (most?) video and because of this a very difficult situation has been allowed to arise. Bummer, eh? P From etienne.buira.lists at free.fr Thu Nov 17 09:26:49 2011 From: etienne.buira.lists at free.fr (Etienne Buira) Date: Thu, 17 Nov 2011 09:26:49 +0100 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: <4EC279CE.2070800@gmail.com> References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> <4EC279CE.2070800@gmail.com> Message-ID: <20111117082649.GB3363@epicure.lazyet.homelinux.net> Hi. On Tue, Nov 15, 2011 at 09:40:14AM -0500, Matthew Morgan wrote: > Alas but no, that doesn't work either. The skip happens, but it's not seeking to the first keyframe after the skip like it usually would. I don't know whether it's because keyframe-seeking gets disabled with codec copy or if it's the fact that these input files are a little weird. I found that if I get rid of the codec copy commands and try to do a skip of 60 seconds (long enough to make it obvious), I get: > > ffmpeg -i M2U00108.MPG -ss 60 -target ntsc-dvd -y m2u00108_copy.mpg Keyframe seeking can occur if you ask it (ie you put -ss before -i). From fox at orbitalfox.com Thu Nov 17 11:47:30 2011 From: fox at orbitalfox.com (fox) Date: Thu, 17 Nov 2011 10:47:30 +0000 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> <20111116162553.216aea7e@apollon> Message-ID: <20111117104730.0e4fd37d@apollon> On Wed, 16 Nov 2011 16:44:40 +0000 (UTC) Carl Eugen Hoyos wrote: > Only one packager is supporting the traitors who had tried to steal > FFmpeg. Which distro is using those packages? > It is very strongly recommended that you use current git head, all > bugs (etc.) are first fixed in git head (and only later in release > branches), there is nothing "stable" in the releases that brings any > advantage to users. (Packagers claim they need releases, that is why > we provide them, users should not download them.) > > If you look at our bug tracker, you will find fixed bugs (and > enhancements) there. Some of the fixes were made in the fork, some > bugs may never have been reproducible with the fork. But most fixes > are missing from the fork, making it an intentionally broken version > of FFmpeg that - due to insufficient manpower - we cannot support > because it contains >100 known bugs not present in FFmpeg. That makes sense -- good to know. Fox From tim.nicholson at bbc.co.uk Thu Nov 17 12:19:48 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Thu, 17 Nov 2011 11:19:48 +0000 Subject: [FFmpeg-user] Using the "select" filter Message-ID: <4EC4EDD4.5000102@bbc.co.uk> The examples of using the select filter given at:- http://ffmpeg.org/ffmpeg.html#select suggest, for example, # select only I-frames select='eq(pict_type\,I)' However when I tried this I got an error. ffmpeg -i in.mp4 -vf "select='eq(pict_type\,I)', showinfo" -f mp4 -y /dev/null [..] Missing ')' or too many args in 'eq(pict_type\,I)' Removing the "\" which I assume is there to "escape" the comma solved the problem, thus:- ffmpeg -i in.mp4 -vf "select='eq(pict_type,I)', showinfo" -f mp4 -y /dev/null This looks like a small error in the docs. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From rhodri at kynesim.co.uk Thu Nov 17 14:14:50 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Thu, 17 Nov 2011 13:14:50 -0000 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: On Wed, 16 Nov 2011 23:58:41 -0000, HallMarc Websites wrote: [Carl said:] >> mp4 does not support pcm audio (mov does). > > Um I beg to differ. I have an mp4 container with pcm-s24be and it plays > beautifully. However, it was done with a piece of software for Windows. > While it uses ffmpeg and almost all of the codecs; I have no way of > seeing what and how it is going about this. > So, I know it can be done. The question is how. Both of you are right, and the answer is "by lying." There are no official MP4 ids identifying PCM audio, so you have to make one up. Probably the MOV ids work. Probably. The disadvantage of using unofficial ids is that you have *NO* guarantee that anything else will read your file. > Stop asking me for the file. I cannot give it to you. It is a commercial > and is copyright protected. Not a fan of getting sued. It is the only > one I have where ProRes was used in the post. Without a clip that provokes the problem you are seeing, there's pretty much nothing anyone else can do other than take wild guesses. Unless you are prepared to go through and debug the source yourself -- not an experience for the faint of heart! -- there simply isn't enough information to work on. -- Rhodri James Kynesim Ltd From cehoyos at ag.or.at Thu Nov 17 14:43:53 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Nov 2011 13:43:53 +0000 (UTC) Subject: [FFmpeg-user] fixing videos that don't start with keyframes References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> <20111116162553.216aea7e@apollon> <20111117104730.0e4fd37d@apollon> Message-ID: fox orbitalfox.com> writes: > On Wed, 16 Nov 2011 16:44:40 +0000 (UTC) > Carl Eugen Hoyos ag.or.at> wrote: > > Only one packager is supporting the traitors who had tried to steal > > FFmpeg. > > Which distro is using those packages? Debian and Ubuntu, afaik (they share the same packager who is part of the group mentioned above). Iirc, two Debian and one Ubuntu bugs opened against the fork are fixed in FFmpeg... Carl Eugen From marc at hallmarcwebsites.com Thu Nov 17 14:43:58 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Thu, 17 Nov 2011 08:43:58 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > [Carl said:] > >> mp4 does not support pcm audio (mov does). > > > > Um I beg to differ. I have an mp4 container with pcm-s24be and it > > plays beautifully. However, it was done with a piece of software for > Windows. > > While it uses ffmpeg and almost all of the codecs; I have no way of > > seeing what and how it is going about this. > > So, I know it can be done. The question is how. > > Both of you are right, and the answer is "by lying." There are no official MP4 > ids identifying PCM audio, so you have to make one up. > Probably the MOV ids work. Probably. The disadvantage of using unofficial > ids is that you have *NO* guarantee that anything else will read your file. > > > Stop asking me for the file. I cannot give it to you. It is a > > commercial and is copyright protected. Not a fan of getting sued. It > > is the only one I have where ProRes was used in the post. > > Without a clip that provokes the problem you are seeing, there's pretty much > nothing anyone else can do other than take wild guesses. Unless you are > prepared to go through and debug the source yourself -- not an experience > for the faint of heart! -- there simply isn't enough information to work on. > > -- I understand about the need of the file. Sorry, I can't provide it. I'm not looking for absolute answers; if I were, I would have one of you sign a NDA and hire you to write the correct command line(s). What I hope for is a general idea as to how something might be done or why it might have failed. i.e. I have read many references to the supposed necessity to compile ffmpeg with certain flags enabled in order to use the ProRes codec. I have also read where this isn't necessary anymore. Seems awfully fast for a codec I could swear ffmpeg only recently started including in the stable releases(?). So when I posted the output and the sample command lines I used I expected someone to spot a glaring mistake. Didn't happen. [>] Such is life. As far as not being playable here or there. Not my concern here as the ONLY place this file will be played is from a web page and that through an older version of Floatbox. It plays it happily. I know, I know, "You have a way of making the file you want why don't you just use that and be done with it?" Because, it is done from my PC with a 3rd party piece of software called SUPER (c) which is in the same genre as Handbrake, etc. and I need my server side ffmpeg to do this. Not faint of heart here, I started myself in web development 10 years ago. I am completely self-taught and because of the nature of this beast, still learning. If I need to get down to the source code and figure it out, I will. Can you provide a sample command line that would do what you are thinking concerning the mov IDs? Next, what is the best way to view the errors? Is the error I see in the output, is that it? I asked this before, please recommend some reading material/books that I can use to learn about ffmpeg. From fox at orbitalfox.com Thu Nov 17 16:37:03 2011 From: fox at orbitalfox.com (fox) Date: Thu, 17 Nov 2011 15:37:03 +0000 Subject: [FFmpeg-user] fixing videos that don't start with keyframes In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <20111116103626.1ab10e1f@apollon> <20111116162553.216aea7e@apollon> <20111117104730.0e4fd37d@apollon> Message-ID: <20111117153703.749dc00c@apollon> On Thu, 17 Nov 2011 13:43:53 +0000 (UTC) Carl Eugen Hoyos wrote: > Debian and Ubuntu, afaik (they share the same packager who is part of > the group mentioned above). > > Iirc, two Debian and one Ubuntu bugs opened against the fork are > fixed in FFmpeg... Good to know thanks. I have switched to the git repo, and checkinstall-ed it. From tim.nicholson at bbc.co.uk Thu Nov 17 17:16:39 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Thu, 17 Nov 2011 16:16:39 +0000 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <20111115172003.GF26053@leki> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> <20111115163352.GE26053@leki> <4EC29B79.5080605@bbc.co.uk> <20111115172003.GF26053@leki> Message-ID: <4EC53367.4040305@bbc.co.uk> On 15/11/11 17:20, Cl?ment B?sch wrote: > On Tue, Nov 15, 2011 at 05:03:53PM +0000, Tim Nicholson wrote: [...] >> With your help I think I have it sorted in my own mind, now. But I >> think the doc could be improved. >> > > Certainly, and such changes are always welcome. > >>> So if you want to fix the documentation, you should IMO just reword in a >>> sense to explicit the "only one pick per stream type". I'm not a native >>> English speaker so I'm sure you will be able to propose a better wording >>> than me; feel free to send a patch on ffmpeg-devel. If you don't, I'll >>> take the time to of course. >> >> Never tried submitting a patch so will haver to bone up on how to do it... >> > > For such a trivial patch you can just write here the new paragraph, I'll > write the patch and put your name in the credits. You can also try the > "hard way": http://ffmpeg.org/developer.html#Submitting-patches-1 Here is my suggestion -------------cut-------------------- By default ffmpeg includes only one stream of each type (video, audio, subtitle) present in input files and adds them to each output file. It picks the "best" of each based upon the following criteria; for video it is the stream with the highest resolution, for audio the stream with the most channels, for subtitle it's the first subtitle stream. In the case where several streams of the same type rate equally, the lowest numbered stream is chosen. -----------------cut------------------------- If you are happy to submit it then meanwhile I will have a play at all the git stuff required for doing it the "hard way" so hopefully I will be better placed to be more useful next time... -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From ubitux at gmail.com Thu Nov 17 18:10:45 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 17 Nov 2011 18:10:45 +0100 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <4EC53367.4040305@bbc.co.uk> References: <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> <4EC284F4.3010109@bbc.co.uk> <20111115160851.GD26053@leki> <4EC29208.5000802@bbc.co.uk> <20111115163352.GE26053@leki> <4EC29B79.5080605@bbc.co.uk> <20111115172003.GF26053@leki> <4EC53367.4040305@bbc.co.uk> Message-ID: <20111117171045.GE13899@leki> On Thu, Nov 17, 2011 at 04:16:39PM +0000, Tim Nicholson wrote: > On 15/11/11 17:20, Cl?ment B?sch wrote: > >On Tue, Nov 15, 2011 at 05:03:53PM +0000, Tim Nicholson wrote: > [...] > >>With your help I think I have it sorted in my own mind, now. But I > >>think the doc could be improved. > >> > > > >Certainly, and such changes are always welcome. > > > >>>So if you want to fix the documentation, you should IMO just reword in a > >>>sense to explicit the "only one pick per stream type". I'm not a native > >>>English speaker so I'm sure you will be able to propose a better wording > >>>than me; feel free to send a patch on ffmpeg-devel. If you don't, I'll > >>>take the time to of course. > >> > >>Never tried submitting a patch so will haver to bone up on how to do it... > >> > > > >For such a trivial patch you can just write here the new paragraph, I'll > >write the patch and put your name in the credits. You can also try the > >"hard way": http://ffmpeg.org/developer.html#Submitting-patches-1 > > Here is my suggestion > > -------------cut-------------------- > By default ffmpeg includes only one stream of each type (video, > audio, subtitle) present in input files and adds them to each output > file. > It picks the "best" of each based upon the following criteria; > for video it is the stream with the highest resolution, for audio > the stream with the most channels, for subtitle it's the first > subtitle stream. > In the case where several streams of the same type rate equally, the > lowest numbered stream is chosen. > > -----------------cut------------------------- Thank you, just submitted this version to ffmpeg-devel (you should have received a copy of the patch). It will be discussed if necessary and should be applied soon. [...] -- 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 rhodri at kynesim.co.uk Thu Nov 17 19:40:04 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Thu, 17 Nov 2011 18:40:04 -0000 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: On Thu, 17 Nov 2011 13:43:58 -0000, HallMarc Websites wrote: > Can you provide a sample command line that would do what you are thinking > concerning the mov IDs? Sorry, I mostly live in embedded-land, so commands lines are a thing of mystery and terror to me. I'm coming at this from a knowledge of the standards rather than ffmpeg :-( -- Rhodri James Kynesim Ltd From marc at hallmarcwebsites.com Thu Nov 17 19:56:50 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Thu, 17 Nov 2011 13:56:50 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > > Can you provide a sample command line that would do what you are > > thinking concerning the mov IDs? > > Sorry, I mostly live in embedded-land, so commands lines are a thing of > mystery and terror to me. I'm coming at this from a knowledge of the > standards rather than ffmpeg :-( > > -- > Rhodri James > Kynesim Ltd No problem. I found out a way to retain the audio quality of the pcm_s24le codec from a mov container when pulling it into a mp4 container by using aac and -strict experimental in the command line. Still s16 instead of s24 but I couldn't hear any diff in the sound quality like I could when using libfaac. From ubitux at gmail.com Thu Nov 17 20:09:40 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 17 Nov 2011 20:09:40 +0100 Subject: [FFmpeg-user] Using the "select" filter In-Reply-To: <4EC4EDD4.5000102@bbc.co.uk> References: <4EC4EDD4.5000102@bbc.co.uk> Message-ID: <20111117190940.GC14546@leki> On Thu, Nov 17, 2011 at 11:19:48AM +0000, Tim Nicholson wrote: > The examples of using the select filter given at:- > > http://ffmpeg.org/ffmpeg.html#select > > suggest, for example, > > # select only I-frames > select='eq(pict_type\,I)' > > > However when I tried this I got an error. > > ffmpeg -i in.mp4 -vf "select='eq(pict_type\,I)', showinfo" -f mp4 -y > /dev/null > [..] > Missing ')' or too many args in 'eq(pict_type\,I)' > > > Removing the "\" which I assume is there to "escape" the comma > solved the problem, thus:- > > ffmpeg -i in.mp4 -vf "select='eq(pict_type,I)', showinfo" -f mp4 -y > /dev/null > > This looks like a small error in the docs. > You are not supposed to add quotes around the select. ffmpeg -i in.mp4 -vf select='eq(pict_type,I)',showinfo -f ... should work. -- 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 ubitux at gmail.com Thu Nov 17 20:10:46 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 17 Nov 2011 20:10:46 +0100 Subject: [FFmpeg-user] Using the "select" filter In-Reply-To: <20111117190940.GC14546@leki> References: <4EC4EDD4.5000102@bbc.co.uk> <20111117190940.GC14546@leki> Message-ID: <20111117191046.GD14546@leki> On Thu, Nov 17, 2011 at 08:09:40PM +0100, Cl?ment B?sch wrote: > On Thu, Nov 17, 2011 at 11:19:48AM +0000, Tim Nicholson wrote: > > The examples of using the select filter given at:- > > > > http://ffmpeg.org/ffmpeg.html#select > > > > suggest, for example, > > > > # select only I-frames > > select='eq(pict_type\,I)' > > > > > > However when I tried this I got an error. > > > > ffmpeg -i in.mp4 -vf "select='eq(pict_type\,I)', showinfo" -f mp4 -y > > /dev/null > > [..] > > Missing ')' or too many args in 'eq(pict_type\,I)' > > > > > > Removing the "\" which I assume is there to "escape" the comma > > solved the problem, thus:- > > > > ffmpeg -i in.mp4 -vf "select='eq(pict_type,I)', showinfo" -f mp4 -y > > /dev/null > > > > This looks like a small error in the docs. > > > > You are not supposed to add quotes around the select. > > ffmpeg -i in.mp4 -vf select='eq(pict_type,I)',showinfo -f ... > Erh. I meant: ffmpeg -i in.mp4 -vf select='eq(pict_type,\I)',showinfo -f ... -- 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 lists at arlomedia.com Thu Nov 17 20:53:40 2011 From: lists at arlomedia.com (Arlo Leach) Date: Thu, 17 Nov 2011 11:53:40 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: <20111117082649.GB3363@epicure.lazyet.homelinux.net> References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> <4EC279CE.2070800@gmail.com> <20111117082649.GB3363@epicure.lazyet.homelinux.net> Message-ID: <61E12515-25B4-46C5-9183-FAE9353CB830@arlomedia.com> Hello, From Re: [FFmpeg-user] fixing videos that don't start with keyframes: > Keyframe seeking can occur if you ask it (ie you put -ss before -i). I think this helps answer my question about the few seconds of blank video when using -vcodec copy. When I changed my sample extraction command from ffmpeg -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -ss 30 -t 30 -y band_formatted_sample.mp4 to ffmpeg -ss 30 -t 30 -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -y band_formatted_sample.mp4 I started getting more than 30 seconds of video, but no blank video. It looks like the start of the extraction moves forward to the next earlier keyframe and starts from there instead of the requested location. So I might get a 34-second sample instead of a 30-second sample. I also found that if I added the -g flag in my original conversion command, I could write more keyframes into the full video and increase the accuracy of my sample timing. That only created a slight increase in the file size of my full video, which in my application is a good tradeoff. So I think those two changes are my solution. Does that sound right? Thanks, -Arlo _______________________________ Arlo Leach http://arlomedia.com From cehoyos at ag.or.at Thu Nov 17 22:55:13 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 17 Nov 2011 21:55:13 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > No problem. I found out a way to retain the audio quality of the pcm_s24le > codec from a mov container when pulling it into a mp4 container by using aac > and -strict experimental in the command line. Please understand that the reason -strict experimental is required for -acodec aac is that its quality is known to be significantly worse than libfaac. Carl Eugen From jshupert at pps-inc.com Fri Nov 18 00:16:41 2011 From: jshupert at pps-inc.com (Jim Shupert) Date: Thu, 17 Nov 2011 18:16:41 -0500 Subject: [FFmpeg-user] to crop the bottom Message-ID: <4EC595D9.1020901@pps-inc.com> friends, I wish to crop off the bottom 40 lines of a 720x480 video and have the output 720x480 actually i wish to simply crop off the bottom 5 or 8 lines - but used 40 so the effect would be bold. i think my command is not doing anything. I do not wish to reencode - but simply trim off the bottom. I found the "old way( -cropbottom NN " simple - but i seem to not be able to make the -vf way work Thanks ffmpeg -i 1.mp4 -vf crop=720:480:0:40 -vcodec copy -acodec copy 2.mp D:\VidC_05_shr\vmp4\j>ffmpeg -i 1.mp4 -vf crop=720:480:0:40 -vcodec copy -acodec copy 2.mp4 ffmpeg version N-34549-g13b7781, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 6 2011 22:02:08 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --en able-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 24. 0 / 51. 24. 0 libavcodec 53. 28. 0 / 53. 28. 0 libavformat 53. 19. 0 / 53. 19. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 47. 0 / 2. 47. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [aac @ 021B8860] channel element 3.0 is not allocated Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: isomiso2avc1mp41mp423gp5 creation_time : 2011-11-03 15:46:00 encoder : FFmbc 0.6 Duration: 00:04:22.82, start: 0.000000, bitrate: 3437 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 3060 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2011-11-03 15:46:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 4.0, s16, 223 kb/s Metadata: creation_time : 2011-11-03 15:46:00 handler_name : SoundHandler Stream #0:2(und): Data: none (mp4s / 0x7334706D) Metadata: creation_time : 2011-11-16 21:38:21 handler_name : GPAC MPEG-4 OD Handler Stream #0:3(und): Data: none (mp4s / 0x7334706D) Metadata: creation_time : 2011-11-16 21:38:21 handler_name : GPAC MPEG-4 BIFS Handler Stream #0:4(und): Data: none (rtp / 0x20707472) Metadata: creation_time : 2011-11-16 21:38:21 handler_name : GPAC ISO Hint Handler Stream #0:5(und): Data: none (rtp / 0x20707472) Metadata: creation_time : 2011-11-16 21:38:22 handler_name : strptime() unavailable on this system, cannot convert the date string. Output #0, mp4, to '2.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: isomiso2avc1mp41mp423gp5 creation_time : 2011-11-03 15:46:00 encoder : Lavf53.19.0 Stream #0:0(und): Video: h264 (![0][0][0] / 0x0021), yuv420p, 720x480 [SAR 8:9 DAR 4:3], q=2-31, 3060 kb/s, 29.97 fps, 30k tbn, 29.97 tbc Metadata: creation_time : 2011-11-03 15:46:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (@[0][0][0] / 0x0040), 48000 Hz, 4.0, 223 kb/s Metadata: creation_time : 2011-11-03 15:46:00 handler_name : SoundHandler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 7875 fps=3111 q=-1.0 Lsize= 105551kB time=00:04:22.76 bitrate=3290.7kbits/s video:98173kB audio:7184kB global headers:0kB muxing overhead 0.183468% thanks j From stefasab at gmail.com Fri Nov 18 00:44:29 2011 From: stefasab at gmail.com (Stefano Sabatini) Date: Fri, 18 Nov 2011 00:44:29 +0100 Subject: [FFmpeg-user] to crop the bottom In-Reply-To: <4EC595D9.1020901@pps-inc.com> References: <4EC595D9.1020901@pps-inc.com> Message-ID: <20111117234429.GA7035@arborea> On date Thursday 2011-11-17 18:16:41 -0500, Jim Shupert encoded: > friends, > > I wish to crop off the bottom 40 lines of a 720x480 video and have > the output 720x480 > actually i wish to simply crop off the bottom 5 or 8 lines - but > used 40 so the effect would be bold. > i think my command is not doing anything. > > I do not wish to reencode - but simply trim off the bottom. > > I found the "old way( -cropbottom NN " simple - but i seem to not be > able to make the -vf way work > Thanks > > > ffmpeg -i 1.mp4 -vf crop=720:480:0:40 -vcodec copy -acodec copy 2.mp filtering is disabled with -vcodec copy, we should really make ffmpeg more verbose about this. -- ffmpeg-user random tip #9 One minute of audio silence with ffmpeg: ffmpeg -ar 48000 -t 60 -f s16le -acodec pcm_s16le -i /dev/zero \ -ab 64K -f mp2 -acodec mp2 -y silence.mp2 From mahu_berehat at yahoo.co.jp Fri Nov 18 09:06:10 2011 From: mahu_berehat at yahoo.co.jp (orang Aumori Jepun) Date: Fri, 18 Nov 2011 17:06:10 +0900 (JST) Subject: [FFmpeg-user] to crop the bottom In-Reply-To: <4EC595D9.1020901@pps-inc.com> Message-ID: <338583.51613.qm@web3514.mail.bbt.yahoo.co.jp> > I wish to crop off the bottom 40 lines of a 720x480 video? and have the output 720x480 > ffmpeg -i 1.mp4 -vf crop=720:480:0:40 -vcodec copy -acodec copy 2.mp Hi. Sorry if I misunderstand you , but here's what I got from this list about using -vf crop. As I understand, -vf crop=out_w:out_h:x:y out_w is the width in pixels of the final output you want out_h is the height in pixels of the final output you want and these are based on the ORIGINAL frame: x is the pixel position, counting from the left side of the original frame, of the left side of the frame you want to keep y is the pixel position, counting from the top of the original frame, of the top side of the frame you want to keep So, for example, -vf crop=640:480:60:40 will find the point that is 60 pixels from the left side of the original and 40 pixels from the top of the original, and then save a frame that is 640x480 from there, with that point as the upper left corner. If x and y are not given, it defaults to centering the final frame in the original frame. ( from Jim Worrall) Hope this helps. From mark at mdsh.com Fri Nov 18 11:46:09 2011 From: mark at mdsh.com (Mark Himsley) Date: Fri, 18 Nov 2011 10:46:09 +0000 Subject: [FFmpeg-user] to crop the bottom In-Reply-To: <4EC595D9.1020901@pps-inc.com> References: <4EC595D9.1020901@pps-inc.com> Message-ID: <4EC63771.5000201@mdsh.com> On 17/11/2011 23:16, Jim Shupert wrote: > friends, > > I wish to crop off the bottom 40 lines of a 720x480 video and have the > output 720x480 > actually i wish to simply crop off the bottom 5 or 8 lines - but used 40 > so the effect would be bold. > i think my command is not doing anything. > > I do not wish to reencode - but simply trim off the bottom. That is not possible. Filtering is applied to decoded raw video frames, so decoding and reencoding is required. -- Mark From tim.nicholson at bbc.co.uk Fri Nov 18 12:06:12 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 18 Nov 2011 11:06:12 +0000 Subject: [FFmpeg-user] Using the "select" filter In-Reply-To: <20111117191046.GD14546@leki> References: <4EC4EDD4.5000102@bbc.co.uk> <20111117190940.GC14546@leki> <20111117191046.GD14546@leki> Message-ID: <4EC63C24.3080400@bbc.co.uk> On 17/11/11 19:10, Cl?ment B?sch wrote: > On Thu, Nov 17, 2011 at 08:09:40PM +0100, Cl?ment B?sch wrote: >> On Thu, Nov 17, 2011 at 11:19:48AM +0000, Tim Nicholson wrote: >>> The examples of using the select filter given at:- >>> >>> http://ffmpeg.org/ffmpeg.html#select >>> >>> suggest, for example, >>> >>> # select only I-frames >>> select='eq(pict_type\,I)' >>> >>> >>> However when I tried this I got an error. >>> >>> ffmpeg -i in.mp4 -vf "select='eq(pict_type\,I)', showinfo" -f mp4 -y >>> /dev/null >>> [..] >>> Missing ')' or too many args in 'eq(pict_type\,I)' >>> >>> >>> Removing the "\" which I assume is there to "escape" the comma >>> solved the problem, thus:- >>> >>> ffmpeg -i in.mp4 -vf "select='eq(pict_type,I)', showinfo" -f mp4 -y >>> /dev/null >>> >>> This looks like a small error in the docs. >>> >> >> You are not supposed to add quotes around the select. The quotes were around the whole filtergraph, not just the select. >> >> ffmpeg -i in.mp4 -vf select='eq(pict_type,I)',showinfo -f ... >> > > Erh. I meant: > ffmpeg -i in.mp4 -vf select='eq(pict_type,\I)',showinfo -f ... Actually you meant:- ffmpeg -i in.mp4 -vf select='eq(pict_type\,I)', showinfo -f ... Why is select different to any other filter? Looking through the list of examples provided in:- http://ffmpeg.org/ffmpeg.html#Video-Filters shows the following:- 23.11 fieldorder ffmpeg -i in.vob -vf "fieldorder=bff" out.dv 23.16 hflip ffmpeg -i in.avi -vf "hflip" out.avi 23.34 slicify ffmpeg -i in.avi -vf "slicify=32" out.avi 23.37 unsharp ffmpeg -i in.avi -vf "unsharp" out.mp4 23.38 vflip ffmpeg -i in.avi -vf "vflip" out.avi From which one might reasonably deduce that is is normal, and recommended to enclose the whole filtergraph within " ". It certainly aids clarity in seeing what elements on the command line are part of the whole filter arrangement, and I think its more readable when you don't end up with \ escapes littering a filter that already has ample use of other punctuation characters such as = , : it reminds me a bit of perl. ;P Of course its not actually that select is any different. Its just that when building a filtergraph you have the choice of either escaping individual characters as required, or enclosing the whole filtergraph in quotes, and it just happens that the select example is given using the alternative method, as are 4 other filters so in terms of examples its a dead heat! However the inconsistency can be confusing, especially as the either/or nature of fully quoting or escaping is not mentioned in the general filtergraph description, with the -vf " " syntax only appearing in the above listed examples. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From marc at hallmarcwebsites.com Fri Nov 18 15:39:54 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 18 Nov 2011 09:39:54 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality Message-ID: AAC is not limited to -strict experimental because of any quality issue. My experience has shown that AAC produced a much better quality audio than libfaac. Even audiocoding.com states that currently AAC is much better than libfaac! I see some postings, maybe someone here that KNOWS can expound on it, that say libfaac/faac were removed from ffmpeg builds due to licensing issues. I've asked you once before and now I plead with you, stop posting garbage in this list. You are not helping anyone by misinforming them. Marc Hall HallMarc Websites www.HallMarcWebsites.com 610-446-3346 From tim.nicholson at bbc.co.uk Fri Nov 18 15:56:03 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 18 Nov 2011 14:56:03 +0000 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: <4EC67203.6@bbc.co.uk> On 18/11/11 14:39, HallMarc Websites wrote: > AAC is not limited to -strict experimental because of any quality issue. My > experience has shown that AAC produced a much better quality audio than > libfaac. Even audiocoding.com states that currently AAC is much better than > libfaac! > I could find no reference to ffmpegs AAC on the audiocoding.com website. Could you provide the link? > > > I see some postings, maybe someone here that KNOWS can expound on it, that > say libfaac/faac were removed from ffmpeg builds due to licensing issues. > According to:- http://www.audiocoding.com/faac.html "FAAC is based on the original ISO MPEG reference code. The changes to this code are licensed under the LGPL license. The original license is not compatible with the LGP..." Since ffmpeg is LGPL I think you can see the issue. > > > I've asked you once before and now I plead with you, stop posting garbage in > this list. You are not helping anyone by misinforming them. > I find that comments by ffmpeg maintainers are rarely garbage, it is usually the most technically accurate and they are under no obligation to offer advice, so publicly ridiculing them is not likely to work in your favour. > > > Marc Hall > -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From rogerdpack2 at gmail.com Fri Nov 18 16:41:19 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 18 Nov 2011 08:41:19 -0700 Subject: [FFmpeg-user] Audio Stream Mapping issue In-Reply-To: <20111115101931.GC26053@leki> References: <4EC226F6.8010403@bbc.co.uk> <4EC233F1.4000800@bbc.co.uk> <1321351535353-4042416.post@n4.nabble.com> <20111115101931.GC26053@leki> Message-ID: > The multiple files limitation could be bypassed with an audio merge > filter; a WIP was sent on the mailing list by Nicolas. We lack man-power, > and patches are always welcome when it is a feature request :) Do you have a link you could provide for Nicolas' WIP? (quick google search seemed unfruitful). Thanks! -r From marc at hallmarcwebsites.com Fri Nov 18 17:11:59 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 18 Nov 2011 11:11:59 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: <4EC67203.6@bbc.co.uk> References: <4EC67203.6@bbc.co.uk> Message-ID: http://www.audiocoding.com/faac.html [>] Took me 2 secs to find it I don't see the person I'm referring to in the list of ffmpeg developers.... And he has posted a few responses that are incorrect. If this is one of your developers.... From tim.nicholson at bbc.co.uk Fri Nov 18 17:52:38 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 18 Nov 2011 16:52:38 +0000 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: <4EC67203.6@bbc.co.uk> Message-ID: <4EC68D56.6090903@bbc.co.uk> On 18/11/11 16:11, HallMarc Websites wrote: > http://www.audiocoding.com/faac.html > [>] Took me 2 secs to find it Oh, I think I may have misunderstood you. When in an earlier post you said:- "AAC is not limited to -strict experimental because of any quality issue. My experience has shown that AAC produced a much better quality audio than libfaac. Even audiocoding.com states that currently AAC is much better than libfaac!" your reference to the "-strict experimental" flag of ffmpeg led me to believe you were referring to its internal version of aac and claiming that audiocoding.com were stating that *it* was better than theirs, not that:- "FAAC is not up to par with the currently *best* AAC encoders available" In the interests of avoiding confusion, could you perhaps clarify which AAC encoder you consider better than FAAC since the above quote is non specific? > I don't see the person I'm referring to in the list of ffmpeg developers.... > And he has posted a few responses that are incorrect. If this is one of your > developers.... I claim no ownership, but I am grateful whenever someone takes the time to try and respond to my query, which is often down to my own lack of understanding, and which leads to the sorts of questions that those with a much greater knowledge of the system than me must at times consider to show a feeble grasp of its intricacies, and even if their answers miss my point, they usually spark enough of an idea to let me work my way to a solution which makes belittling them seem churlish. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From marc at hallmarcwebsites.com Fri Nov 18 18:05:30 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 18 Nov 2011 12:05:30 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: <4EC68D56.6090903@bbc.co.uk> References: <4EC67203.6@bbc.co.uk> <4EC68D56.6090903@bbc.co.uk> Message-ID: > On 18/11/11 16:11, HallMarc Websites wrote: > > http://www.audiocoding.com/faac.html > > [>] Took me 2 secs to find it > > Oh, I think I may have misunderstood you. When in an earlier post you said:- > > "AAC is not limited to -strict experimental because of any quality issue. My > experience has shown that AAC produced a much better quality audio than > libfaac. Even audiocoding.com states that currently AAC is much better than > libfaac!" > > your reference to the "-strict experimental" flag of ffmpeg led me to believe > you were referring to its internal version of aac and claiming that > audiocoding.com were stating that *it* was better than theirs, not > that:- > > "FAAC is not up to par with the currently *best* AAC encoders available" > > In the interests of avoiding confusion, could you perhaps clarify which AAC > encoder you consider better than FAAC since the above quote is non > specific? > > > > I don't see the person I'm referring to in the list of ffmpeg developers.... > > And he has posted a few responses that are incorrect. If this is one > > of your developers.... > > I claim no ownership, but I am grateful whenever someone takes the time to > try and respond to my query, which is often down to my own lack of > understanding, and which leads to the sorts of questions that those with a > much greater knowledge of the system than me must at times consider to > show a feeble grasp of its intricacies, and even if their answers miss my point, > they usually spark enough of an idea to let me work my way to a solution > which makes belittling them seem churlish. > > -- > Tim [>] Direct reference to my statement that using the AAC codec produced a much better audio stream than using libfaac. I know, I know libfaac and faac are not one in the same. Close enough though. I read that FAAC and libfaac are only slightly different. I will test faac later. When I mentioned that I needed to use -strict_experimental flag in order to use AAC I was told that the reason is because AAC is a poorer quality codec than libfaac even though my own ears heard the exact opposite. From diesql at googlemail.com Fri Nov 18 18:06:31 2011 From: diesql at googlemail.com (Jamie Tufnell) Date: Fri, 18 Nov 2011 12:06:31 -0500 Subject: [FFmpeg-user] Command line option to convert to square pixels? Message-ID: How can I convert to square pixels IF necessary, AND maintain height based the input file's display aspect ratio? I want to force the width of the output file, but set height based on the input file's display aspect ratio and convert to square pixels if necessary. Right now I use -vf "scale=640:-1" and I want to extend that to convert to square pixels as well. e.g. the same command line option would have the following effects: (no square pixel conversion necessary) in: 1440x810 PAR 1:1 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 in: 1440x1080 PAR 1:1 DAR 4:3 => out: 640x480 PAR 1:1 DAR 4:3 (convert to square pixels) in: 1440x1080 PAR 4:3 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 Is there a way to do this? Thanks! From rogerdpack2 at gmail.com Fri Nov 18 21:06:51 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 18 Nov 2011 13:06:51 -0700 Subject: [FFmpeg-user] Command line option to convert to square pixels? In-Reply-To: References: Message-ID: > How can I convert to square pixels IF necessary, AND maintain height > based the input file's display aspect ratio? square pixels? -roger- From cehoyos at ag.or.at Fri Nov 18 21:49:20 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Nov 2011 20:49:20 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > AAC is not limited to -strict experimental because of any quality issue. My > experience has shown that AAC produced a much better quality audio than > libfaac. Since other people are reading this list: The native FFmpeg AAC encoder can only be used with -strict experimental because it has significantly worse quality than all external AAC encoders (libfaac is not the only one), meaning audio quality is worse for a given bitrate. I usually test the native FFmpeg AAC encoder with slightly increased bitrates (>64k), it is absolutely usable imo, dts for example is definitely not usable (to name another audio encoder that needs -strict experimental). But there are many reports and requests both on this list and the developer mailing list about the relatively poor quality of the native AAC encoder compared with other AAC encoders. [...] > I see some postings, maybe someone here that KNOWS can expound on it, that > say libfaac/faac were removed from ffmpeg builds due to licensing issues. You cannot fulfill the requirements of the GPL for libfaac, meaning you cannot legally distribute a binary of FFmpeg linked against both libfaac and a GPL'd library, for example x264. > I've asked you once before and now I plead with you, stop posting garbage in > this list. You are not helping anyone by misinforming them. Hm. lol. Carl Eugen From krueger at lesspain.de Fri Nov 18 21:56:48 2011 From: krueger at lesspain.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Fri, 18 Nov 2011 21:56:48 +0100 Subject: [FFmpeg-user] Command line option to convert to square pixels? In-Reply-To: References: Message-ID: <90D645E5-1E58-4A2F-96F6-96593F2A9B61@lesspain.de> On Nov 18, 2011, at 18:06 , Jamie Tufnell wrote: > How can I convert to square pixels IF necessary, AND maintain height > based the input file's display aspect ratio? > > I want to force the width of the output file, but set height based on > the input file's display aspect ratio and convert to square pixels if > necessary. > > Right now I use -vf "scale=640:-1" and I want to extend that to > convert to square pixels as well. > > e.g. the same command line option would have the following effects: > > (no square pixel conversion necessary) > in: 1440x810 PAR 1:1 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 > in: 1440x1080 PAR 1:1 DAR 4:3 => out: 640x480 PAR 1:1 DAR 4:3 > > (convert to square pixels) > in: 1440x1080 PAR 4:3 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 > > Is there a way to do this? > have you tried using an expression in the scale filter that uses the sar constant (http://ffmpeg.org/ffmpeg.html#scale) to compute width correctly and then use the setsar filter with a value of 1:1 to force square pixels? I would guess that should do the job. From cehoyos at ag.or.at Fri Nov 18 22:03:59 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Nov 2011 21:03:59 +0000 (UTC) Subject: [FFmpeg-user] trouble with -vcodec copy References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> <4EC279CE.2070800@gmail.com> <20111117082649.GB3363@epicure.lazyet.homelinux.net> <61E12515-25B4-46C5-9183-FAE9353CB830@arlomedia.com> Message-ID: Arlo Leach arlomedia.com> writes: > ffmpeg -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -ss 30 -t 30 -y band_formatted_sample.mp4 > > to > > ffmpeg -ss 30 -t 30 -i band_formatted.mp4 -f mp4 -vcodec copy -acodec copy -y band_formatted_sample.mp4 > > I started getting more than 30 seconds of video, but no blank video. It looks > like the start of the extraction moves forward to the next earlier keyframe > and starts from there instead of the requested location. Yes, but I suspect you don't see the "blank video" with MPlayer (or ffplay), but QuickTime does not want to display the initial non-key frames that are copied with the first of your command lines (that asks ffmpeg to seek to exactly 30 seconds, no matter if there is a key frame or not). The second command line asks to seek to approximately 30 seconds (depending on where a keyframe - or whatever FFmpeg believes is a keyframe - can be found). > So I might get a 34-second sample instead of a 30-second sample. > > I also found that if I added the -g flag in my original conversion command, I > could write more keyframes into the full video and increase the accuracy of > my sample timing. That only created a slight increase in the > file size of my full video, which in my application is a good tradeoff. I don't think -g has any effect with -vcodec copy. (But if you mean you first encode the sample and cut it later, than more keyframes of course increase the seeking accuracy.) Carl Eugen From lists at arlomedia.com Fri Nov 18 22:20:12 2011 From: lists at arlomedia.com (Arlo Leach) Date: Fri, 18 Nov 2011 13:20:12 -0800 Subject: [FFmpeg-user] trouble with -vcodec copy In-Reply-To: References: <4EBBE63F.8010804@gmail.com> <4EBBF909.9040508@gmail.com> <4EBC069D.6000602@gmail.com> <4EBD8E0B.400@gmail.com> <4EC279CE.2070800@gmail.com> <20111117082649.GB3363@epicure.lazyet.homelinux.net> <61E12515-25B4-46C5-9183-FAE9353CB830@arlomedia.com> Message-ID: > Yes, but I suspect you don't see the "blank video" with MPlayer (or ffplay), but > QuickTime does not want to display the initial non-key frames that are copied > with the first of your command lines Hmm, that could be. I don't have any media player on my Mac that doesn't seem to be QuickTime-based. If anyone else is curious about this, you can try playing this file in a different player and see if the video starts right away or after about 8 seconds: http://www.arlomedia.com/projects/ffmpeg/band_sample.mp4 > I don't think -g has any effect with -vcodec copy. (But if you mean you first > encode the sample and cut it later, than more keyframes of course increase the > seeking accuracy.) Yes, that's what I meant -- thanks for confirming. Cheers, -Arlo _______________________________ Arlo Leach http://arlomedia.com From lionel.tresaugues at gmail.com Fri Nov 18 22:39:08 2011 From: lionel.tresaugues at gmail.com (=?ISO-8859-1?Q?Lionel_Tr=E9saugues?=) Date: Fri, 18 Nov 2011 22:39:08 +0100 Subject: [FFmpeg-user] Duplicates frames when converting interlaced h264 to png frames and double length when converting interlaced .png to interlaced mp4 Message-ID: Dear FFmpeg users, I am currently?encountering two major problems during my video-editing process (I am running the 64bits edition of Ubuntu Linux 11.10). My camcorder (Panasonic HDC-SD60) shots?1080/50i (AVCHD in a mts container). First problem) I want, in a first step, to convert the .mts into a sequence of .png frames while preserving the interlacing. To do so, I used the following FFmpeg command-line: ffmpeg -i 00030.MTS -an -vcodec png -r 50 -flags +ilme+ildct+mv4+aic -top 1 -qscale 1 -s 1920x1080 -sameq 00030%05d.png Then for a 2min21sec long .mts movie, ffmpeg produces 7080 frames each of them clearly showing signs of interlacing (which was the expected result) (and 7080/50=2min21sec). But, indeed, two consecutive frames are absolutely identical (e.g frames 1 and 2 are identical, 3 and 4 are identical, 5 and 6, etc..) When I try to do the same process using avisynth and VirtualDub (on windows), while using this avisynth script : DirectShowSource("E:\00030.MTS").AssumeTFF().DoubleWeave() I, then obtain once again 7080 interlaced png frames but this time they all are different. So, first question, how would I be able to produce the same results with ffmpeg ? Second problem) Then, I want to convert the interlaced png frames (ideally after processing in Blender) and merge them with an audio track into a mp4 container and still want to preserve the interlacing. I used the following command-line : ffmpeg -threads 2 -i %05d.png -i audio.aac -qscale 1 -vcodec mpeg4 -acodec copy -minrate 0 -b 50000k -mbd 2 -trellis 2 -cmp 2 -subcmp 2 -g 300 -r 50 -flags +ilme+ildct+mv4+aic -top 1 -f mp4 00030.mp4 And I obtain a slow-motion interlaced movie which lasts twice the duration of the initial footage. The problem is the same whether I use the 7080 frames produced by ffmpeg (containing duplicates) or the 7080 frames (without duplicates) produced by VirtualDub. How would I be able to use the sequence of .png frames to produce an interlaced movie (50i) in a mp4 container correctly ? Thanks a lot for your suggestions /lionel These are some informations about the version of FFmpeg I am using : ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav developers ? built on Oct ?2 2011 15:13:26 with gcc 4.6.1 ? configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static ? WARNING: library configuration mismatch ? avutil ? ? ?configuration: --extra-version='4:0.7.2.1ubuntu1+medibuntu1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --enable-shared --disable-static ? avcodec ? ? configuration: --extra-version='4:0.7.2.1ubuntu1+medibuntu1' --arch=amd64 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --enable-shared --disable-static ? libavutil ? ?51. ?7. 0 / 51. ?7. 0 ? libavcodec ? 53. ?5. 0 / 53. ?5. 0 ? libavformat ?53. ?2. 0 / 53. ?2. 0 ? libavdevice ?53. ?0. 0 / 53. ?0. 0 ? libavfilter ? 2. ?4. 0 / ?2. ?4. 0 ? libswscale ? ?2. ?0. 0 / ?2. ?0. 0 ? libpostproc ?52. ?0. 0 / 52. ?0. 0 From cehoyos at ag.or.at Fri Nov 18 23:48:18 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Nov 2011 22:48:18 +0000 (UTC) Subject: [FFmpeg-user] Duplicates frames when converting interlaced h264 to png frames and double length when converting interlaced .png to interlaced mp4 References: Message-ID: Lionel Tr?saugues gmail.com> writes: > ffmpeg -i 00030.MTS -an -vcodec png -r 50 -flags +ilme+ildct+mv4+aic > -top 1 -qscale 1 -s 1920x1080 -sameq 00030%05d.png (Complete, uncut output missing.) -an, -vcodec, -flags, -qscale (?) and -s should be unneeded, I am not sure about -top (does it make a difference?), -r 50 should be responsible for the duplicated frames (and removing it may fix your second problem). [...] > ffmpeg -threads 2 -i %05d.png -i audio.aac -qscale 1 -vcodec mpeg4 > -acodec copy -minrate 0 -b 50000k -mbd 2 -trellis 2 -cmp 2 -subcmp 2 > -g 300 -r 50 -flags +ilme+ildct+mv4+aic -top 1 -f mp4 00030.mp4 I don't think -qscale and -b (now -vb) make sense in the same command line... > ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the > Libav developers This is known to be _very_ buggy and is therefore unsupported. Carl Eugen From marc at hallmarcwebsites.com Fri Nov 18 23:48:09 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 18 Nov 2011 17:48:09 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > > AAC is not limited to -strict experimental because of any quality > > issue. My experience has shown that AAC produced a much better quality > > audio than libfaac. > > Since other people are reading this list: > The native FFmpeg AAC encoder can only be used with -strict experimental > because it has significantly worse quality than all external AAC encoders > (libfaac is not the only one), meaning audio quality is worse for a given > bitrate. > > I usually test the native FFmpeg AAC encoder with slightly increased bitrates > (>64k), it is absolutely usable imo, dts for example is definitely not usable (to > name another audio encoder that needs -strict experimental). > But there are many reports and requests both on this list and the developer > mailing list about the relatively poor quality of the native AAC encoder > compared with other AAC encoders. > > [...] > > > I see some postings, maybe someone here that KNOWS can expound on it, > > that say libfaac/faac were removed from ffmpeg builds due to licensing > issues. > > You cannot fulfill the requirements of the GPL for libfaac, meaning you > cannot legally distribute a binary of FFmpeg linked against both libfaac and a > GPL'd library, for example x264. > > > I've asked you once before and now I plead with you, stop posting > > garbage in this list. You are not helping anyone by misinforming them. > > Hm. > lol. > > Carl Eugen > [>] Look, I don't need an argument, I need answers and or suggestions. I did not realize until the last few days that there were so many aac libs out there and that they can be quite different in quality. Again, anybody on this list that can recommend good reading material regarding this. I've been in web dev for 9 years so I'm not ascared. From cehoyos at ag.or.at Fri Nov 18 23:59:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Nov 2011 22:59:36 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > Look, I don't need an argument, I need answers and or suggestions. I am still trying my best;-) > I did not realize until the last few days that there were so many aac libs out > there and that they can be quite different in quality. Both http://ffmpeg.org/general.html#Audio-Codecs and ./configure --help try to answer this, but if you want to improve our documentation, patches are very welcome! > Again, anybody on this list that can recommend good reading material > regarding this. I've been in web dev for 9 years so I'm not ascared. If you want general information about AAC encoders, this list may not be the best place (I suspect the hydrogen forum is), if you have questions regarding AAC encoding with FFmpeg, I suggest you ask here, I would be surprised if you find more information elsewhere. Carl Eugen From marc at hallmarcwebsites.com Sat Nov 19 00:28:11 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Fri, 18 Nov 2011 18:28:11 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > > Again, anybody on this list that can recommend good reading material > > regarding this. I've been in web dev for 9 years so I'm not ascared. > > If you want general information about AAC encoders, this list may not be the > best place (I suspect the hydrogen forum is), if you have questions regarding > AAC encoding with FFmpeg, I suggest you ask here, I would be surprised if > you find more information elsewhere. > > Carl Eugen [>] I mean in general. I ordered a book from Amazon (Real World Video Compression-Andy Beach) and I have read the ffmpeg documentation, like reading an ingredient list more than anything, I have read posts on different forums. My task is this, to learn the ins and outs of a/v transcoding and conversion. I aim to create another CDN that can accept any of the major video formats and auto convert them into something web friendly; desktops, tablets and smart phones. Yep, that's how I've always been jump right into the deep end with both feet. [>] What can you say about the faac from here: http://sourceforge.net/projects/faac/ This kind of reminds me how I felt when I first dove into website building, Holy Data Turds but there's a lot to learn! From cehoyos at ag.or.at Sat Nov 19 00:37:13 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 18 Nov 2011 23:37:13 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?mov_=5BProRes/pcm=5Fs24le=5D_-=3E_mp4_sam?= =?utf-8?q?e_quality?= References: Message-ID: HallMarc Websites hallmarcwebsites.com> writes: > What can you say about the faac from here: > http://sourceforge.net/projects/faac/ I believe it was repeatedly said in this thread that while it is much better than FFmpeg's native AAC encoder, it "is not up to par with the currently *best* AAC encoders available". (It is simply old, iirc, the author has written one or more of the "best AAC encoders" after writing faac.) Additionally, it has the explained license problems if you want to distribute FFmpeg-based binaries. Carl Eugen From zongyao.qu at gmail.com Fri Nov 18 15:40:45 2011 From: zongyao.qu at gmail.com (Zongyao Qu) Date: Fri, 18 Nov 2011 14:40:45 +0000 (UTC) Subject: [FFmpeg-user] A/V are out of sync Message-ID: I used to post it to the mplayer.user board. And as I tested further, I think it related to the thread issue. So I am not sure whether it is on mplayer side or ffmpeg side. The issue is that when I set thread to 8, the video is always behind of audio about 0.3s but when I set thread to 1, I could not see the out-of-sync. when set to 6, it is seeable, but less than 8. the log and the original post is at http://article.gmane.org/gmane.comp.video.mplayer.user/67197 Thank you very much. From cehoyos at ag.or.at Sat Nov 19 12:30:32 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 19 Nov 2011 11:30:32 +0000 (UTC) Subject: [FFmpeg-user] A/V are out of sync References: Message-ID: Zongyao Qu gmail.com> writes: > The issue is that when I set thread to 8, the video is always behind > of audio about 0.3s > but when I set thread to 1, I could not see the out-of-sync. > when set to 6, it is seeable, but less than 8. ffmpeg -threads 8 -i input -qscale 2 -ab 256k out.avi produces a file for you that is out-of-sync? If yes, please provide complete, uncut console output. Please do not point to external information, but provide everything here on the list that is necessary to reproduce the problem. Carl Eugen From marc at hallmarcwebsites.com Sat Nov 19 14:15:36 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Sat, 19 Nov 2011 08:15:36 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: > > What can you say about the faac from here: > > http://sourceforge.net/projects/faac/ > > I believe it was repeatedly said in this thread that while it is much better than > FFmpeg's native AAC encoder, it "is not up to par with the currently *best* > AAC encoders available". (It is simply old, iirc, the author has written one or > more of the "best AAC encoders" after writing faac.) Additionally, it has the > explained license problems if you want to distribute FFmpeg-based binaries. > > Carl Eugen > And I have said repeatedly that IN MY CASE you are quite simply wrong! I used that libfaac and I pushed the settings as high as they can go and got an audio stream that sounded like my speakers were covered by a wet blanket. When I use the AAC that comes with the current ffmpeg from GIT then the audio is clear and crisp as it should be. [>] So, again, in my case I get the exact opposite result you repeatedly insist I should be getting. From krueger at lesspain.de Sat Nov 19 15:33:17 2011 From: krueger at lesspain.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Sat, 19 Nov 2011 15:33:17 +0100 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: References: Message-ID: <28CD1CB9-CAFF-430A-AECE-1181D261DFF6@lesspain.de> On Nov 19, 2011, at 14:15 , HallMarc Websites wrote: >>> What can you say about the faac from here: >>> http://sourceforge.net/projects/faac/ >> >> I believe it was repeatedly said in this thread that while it is much > better than >> FFmpeg's native AAC encoder, it "is not up to par with the currently > *best* >> AAC encoders available". (It is simply old, iirc, the author has written > one or >> more of the "best AAC encoders" after writing faac.) Additionally, it has > the >> explained license problems if you want to distribute FFmpeg-based > binaries. >> >> Carl Eugen >> > And I have said repeatedly that IN MY CASE you are quite simply wrong! I > used that libfaac and I pushed the settings as high as they can go and got > an audio stream that sounded like my speakers were covered by a wet blanket. > When I use the AAC that comes with the current ffmpeg from GIT then the > audio is clear and crisp as it should be. > [>] > So, again, in my case I get the exact opposite result you repeatedly insist > I should be getting. The this is at least _very_ unusual and AFAIK the first time someone has reported this to this list. My personal tests support what carl eugen and countless other competent people have said on this list. So either you go to the trouble of building a reproducible test case that you can share without an NDA or you are probably on your own and you should at least stop being rude to people who to the best of their (undisputed) knowledge are trying to help you with a patience I don't think anyone could ask them to have. You are a beginner and it's fine to ask questions but you attitude is like pissing on the table of the family who has just invited you for a free lunch and then insulting the cook without ever having cooked yourself in your entire life. Peace From blacktrash at gmx.net Sat Nov 19 17:01:25 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Sat, 19 Nov 2011 16:01:25 +0000 Subject: [FFmpeg-user] build failure on MacOS 10.5.8 Message-ID: <20111119160125.GX835@krille.blacktrash.org> Hi, git head fails to build on MacOS 10.5.8 and --enable-shared: LD libavformat/libavformat.53.dylib Undefined symbols: "_ff_httpproxy_protocol", referenced from: _ff_httpproxy_protocol$non_lazy_ptr in allformats.o ld: symbol(s) not found collect2: ld returned 1 exit status make: *** [libavformat/libavformat.53.dylib] Error 1 Let me know if I should provide more info. TIA c -- 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 marc at hallmarcwebsites.com Sat Nov 19 17:25:27 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Sat, 19 Nov 2011 11:25:27 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: <28CD1CB9-CAFF-430A-AECE-1181D261DFF6@lesspain.de> References: <28CD1CB9-CAFF-430A-AECE-1181D261DFF6@lesspain.de> Message-ID: > The this is at least _very_ unusual and AFAIK the first time someone has > reported this to this list. My personal tests support what carl eugen and > countless other competent people have said on this list. So either you go to > the trouble of building a reproducible test case that you can share without an > NDA or you are probably on your own and you should at least stop being > rude to people who to the best of their (undisputed) knowledge are trying to > help you with a patience I don't think anyone could ask them to have. You > are a beginner and it's fine to ask questions but you attitude is like pissing on > the table of the family who has just invited you for a free lunch and then > insulting the cook without ever having cooked yourself in your entire life. > [>] Yeah sure. Watch what you add to the text you read. I'm not being rude. I'm telling the list and mainly Carl, what my experience has been thus far. I may be a beginner with ffmpeg but I am not a beginner when it comes to programming and how things should be. Not like this where we have polar opposite results. So how about scratching your head like I am and ask "Why is he seeing this result?" Obviously something is happening that shouldn't be. From dflett at bigpond.net.au Sun Nov 20 05:43:17 2011 From: dflett at bigpond.net.au (Dan Flett) Date: Sun, 20 Nov 2011 15:43:17 +1100 Subject: [FFmpeg-user] Command line option to convert to square pixels? In-Reply-To: <90D645E5-1E58-4A2F-96F6-96593F2A9B61@lesspain.de> References: <90D645E5-1E58-4A2F-96F6-96593F2A9B61@lesspain.de> Message-ID: <18F2304FDF7A49479E988E5DEF565261@gtv.local> > On Nov 18, 2011, at 18:06 , Jamie Tufnell wrote: > > > How can I convert to square pixels IF necessary, AND > maintain height > > based the input file's display aspect ratio? > > > > I want to force the width of the output file, but set > height based on > > the input file's display aspect ratio and convert to square > pixels if > > necessary. > > > > Right now I use -vf "scale=640:-1" and I want to extend that to > > convert to square pixels as well. > > > > e.g. the same command line option would have the following effects: > > > > (no square pixel conversion necessary) > > in: 1440x810 PAR 1:1 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 > > in: 1440x1080 PAR 1:1 DAR 4:3 => out: 640x480 PAR 1:1 DAR 4:3 > > > > (convert to square pixels) > > in: 1440x1080 PAR 4:3 DAR 16:9 => out: 640x360 PAR 1:1 DAR 16:9 > > > > Is there a way to do this? > > > > have you tried using an expression in the scale filter that > uses the sar constant (http://ffmpeg.org/ffmpeg.html#scale) > to compute width correctly and then use the setsar filter > with a value of 1:1 to force square pixels? > > I would guess that should do the job. When doing any sort of scaling, the first thing in my -vf sequence is this: -vf scale=iw*sar:ih This scales the width to make square pixels. Any subsequent transformations are then done on square pixel video. From ib at wupperonline.de Sun Nov 20 11:20:26 2011 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Sun, 20 Nov 2011 11:20:26 +0100 Subject: [FFmpeg-user] mp3 sound bug Message-ID: <4ec8d4e3.2056206a.bm000@wupperonline.de> Commit 22e25c002e103e52ace35703423e896b08b51aef introduced some bug that makes ffmp3float produce no sound but only occasional crackles. MPlayer selects this codec by default for me and suddenly has sound issues. git bisect says: 22e25c002e103e52ace35703423e896b08b51aef is the first bad commit commit 22e25c002e103e52ace35703423e896b08b51aef Author: Vitor Sessak Date: Mon Nov 7 21:54:50 2011 +0100 mpegaudiodec: add SSE-optimized imdct36() Signed-off-by: Michael Niedermayer :040000 040000 9d50414f50017cf16dfddac36daf603f8a1555c4 7c0ff2c3d17485bda7538d239253eda6d8b39cc7 M libavcodec :040000 040000 0c6094238f52dcd7b18cb160322caed54d7d9408 4dd510e7a6e81e78c7d12bc605f360d45eaa02b3 M libavutil Ingo From cehoyos at ag.or.at Sun Nov 20 16:56:22 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 20 Nov 2011 15:56:22 +0000 (UTC) Subject: [FFmpeg-user] mp3 sound bug References: <4ec8d4e3.2056206a.bm000@wupperonline.de> Message-ID: Ingo Br?ckl wupperonline.de> writes: > Commit 22e25c002e103e52ace35703423e896b08b51aef introduced some bug that > makes ffmp3float produce no sound but only occasional crackles. Complete, uncut output missing and please point to a sample on samples.mplayerhq.hu that allows to reproduce the problem. Carl Eugen From koxaniy at mail.ru Sun Nov 20 17:09:01 2011 From: koxaniy at mail.ru (Tuuls) Date: Sun, 20 Nov 2011 08:09:01 -0800 (PST) Subject: [FFmpeg-user] XDCAM HD 35 In-Reply-To: <1287999477306-3009996.post@n4.nabble.com> References: <1287999477306-3009996.post@n4.nabble.com> Message-ID: <1321805341715-4088892.post@n4.nabble.com> ffmpeg -i D:\input.avi -pix_fmt yuv420p *-vtag xdv3* *-vcodec mpeg2video -s 1440:1080 -b 35 000k* -bf 2 -r 25 *-copyts* -acodec pcm_s16*be* -y d:\output.mov -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/XDCAM-HD-35-tp3009996p4088892.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ballenato at gmail.com Sun Nov 20 17:31:02 2011 From: ballenato at gmail.com (=?ISO-8859-1?Q?Fernando_S=E1nchez?=) Date: Sun, 20 Nov 2011 18:31:02 +0200 Subject: [FFmpeg-user] Remuxed MP4 (H.264) plays audio but no video in QuickTime Message-ID: <4EC92B46.1080800@gmail.com> Hi all, I am remuxing an FLV file (H.264 + AAC encoded) into MP4. The resulting file plays correctly in every other player. However, in QuickTime 7.7.1 only audio is played, there's no video at all, just a black (in Windows) or green (in Mac) screen. This is the command I'm using for conversion: ffmpeg -i problem_video_qt.flv -acodec copy -vcodec copy problem_video_qt.mp4 The only "warning" I see in the output is "Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1)". Could that be the problem? How can I fix it? To be honest, it may be a problem in the encoded data and not in ffmpeg. Could anyone take a look at the file and tell me what's wrong with it? Or any "magical" ffmpeg parameter that makes it work in QuickTime (without reecoding)? Video files are attached and also the output of ffmpeg command line. The original FLV is recorded from a live stream using rtmpdump and it can be potentially really long, that's why I need to remux it and can't retranscode (although retranscoding with ffmpeg works fine) Thank you very much in advance, any comment will be really appreciated! Command line output: attached Original FLV: http://www.datafilehost.com/download-e037b1c8.html Resulting MP4: http://www.datafilehost.com/download-9226ac9a.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: ffmpeg-output.txt URL: From cehoyos at ag.or.at Sun Nov 20 17:50:48 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 20 Nov 2011 16:50:48 +0000 (UTC) Subject: [FFmpeg-user] Remuxed MP4 (H.264) plays audio but no video in QuickTime References: <4EC92B46.1080800@gmail.com> Message-ID: Fernando S?nchez gmail.com> writes: > FFmpeg version SVN-r21566-xuggle-3.4.843 This is ancient, please test current git head (a snapshot is fine): http://ffmpeg.org/download.html Does out.mov work? Carl Eugen From spam at crcw.mb.ca Sun Nov 20 20:30:10 2011 From: spam at crcw.mb.ca (SPAM) Date: Sun, 20 Nov 2011 13:30:10 -0600 Subject: [FFmpeg-user] adding audio to hundreds of images Message-ID: <4EC95542.1090602@crcw.mb.ca> Hi. Iam trying to add audio to a movie created with between 500 -1000 jpegs. The audio iam trying to add will be 1 of 30 mp3s. I would also like to keep to quality of the images in the movie. Can anyone help? thx. -Ian From lou at lrcd.com Sun Nov 20 20:45:54 2011 From: lou at lrcd.com (Lou) Date: Sun, 20 Nov 2011 10:45:54 -0900 Subject: [FFmpeg-user] adding audio to hundreds of images In-Reply-To: <4EC95542.1090602@crcw.mb.ca> References: <4EC95542.1090602@crcw.mb.ca> Message-ID: <20111120104554.5a51a21e@lrcd.com> On Sun, 20 Nov 2011 13:30:10 -0600 SPAM wrote: > Hi. > > Iam trying to add audio to a movie created with between 500 -1000 > jpegs. The audio iam trying to add will be 1 of 30 mp3s. > I would also like to keep to quality of the images in the movie. > > Can anyone help? > thx. > -Ian Basic example: ffmpeg -i video -i audio -c copy -shortest output or if you're using older FFmpeg: ffmpeg -i video -i audio -vcodec copy -acodec copy -shortest output These commands will copy your video and audio inputs instead of re-encoding them. You may have to re-encode if your desired output format does not support your input streams. From mimeini at gmail.com Sun Nov 20 21:21:13 2011 From: mimeini at gmail.com (mikkel meinike) Date: Sun, 20 Nov 2011 21:21:13 +0100 Subject: [FFmpeg-user] adding audio to hundreds of images In-Reply-To: <20111120104554.5a51a21e@lrcd.com> References: <4EC95542.1090602@crcw.mb.ca> <20111120104554.5a51a21e@lrcd.com> Message-ID: And if you re-encode you should use -sameq to maintain quality mikkel From lou at lrcd.com Sun Nov 20 21:50:47 2011 From: lou at lrcd.com (Lou) Date: Sun, 20 Nov 2011 11:50:47 -0900 Subject: [FFmpeg-user] adding audio to hundreds of images In-Reply-To: References: <4EC95542.1090602@crcw.mb.ca> <20111120104554.5a51a21e@lrcd.com> Message-ID: <20111120115047.41eb9e82@lrcd.com> On Sun, 20 Nov 2011 21:21:13 +0100 mikkel meinike wrote: > And if you re-encode you should use -sameq to maintain quality > > mikkel Contrary to what the documentation used to imply, the -sameq option does not necessarily mean same quality. This option only works in some situations where the input and output share the same quantizer scale in the same range (such as linear mpeg quantizer). Most users will probably benefit more from adjusting qscale instead of using sameq when encoding mpeg*. A value of 3-5 is a good balance between quality and file size. A lower value is a higher quality. From koxaniy at mail.ru Sun Nov 20 22:21:53 2011 From: koxaniy at mail.ru (Tuuls) Date: Sun, 20 Nov 2011 13:21:53 -0800 (PST) Subject: [FFmpeg-user] How do I add a new audio stream? Message-ID: <1321824113329-4089600.post@n4.nabble.com> Hi all. How, after all the new changes, you can add a new audio stream to the .mov file? Previously, it was a option -newaudio , after the output file name. Now any attempt to do so will generate an error "Missing argument for option 'newaudio'" G:\ffmpeg\bin>ffmpeg -threads 4 -i E:\IN.avi -pix_fmt yuv420p -codec:v mpeg2video -b:v 25000k -r 25 -copyts -codec:a pcm_s16le -ar 48000 -ac 2 -y d:\out.mov -map 0:1 -codec:a pcm_s16le -ar 48000 -ac 2 -newaudio -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-do-I-add-a-new-audio-stream-tp4089600p4089600.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From blacktrash at gmx.net Sun Nov 20 22:22:43 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Sun, 20 Nov 2011 21:22:43 +0000 Subject: [FFmpeg-user] [SOLVED] build failure on MacOS 10.5.8 In-Reply-To: <20111119160125.GX835@krille.blacktrash.org> References: <20111119160125.GX835@krille.blacktrash.org> Message-ID: <20111120212243.GE88875@krille.blacktrash.org> * Christian Ebert on Saturday, November 19, 2011 at 16:01:25 +0000 > git head fails to build on MacOS 10.5.8 and --enable-shared: > > LD libavformat/libavformat.53.dylib > Undefined symbols: > "_ff_httpproxy_protocol", referenced from: > _ff_httpproxy_protocol$non_lazy_ptr in allformats.o > ld: symbol(s) not found > collect2: ld returned 1 exit status > make: *** [libavformat/libavformat.53.dylib] Error 1 Solved, most probably by 66e9c0b6ab5360f144cf8f9912ab003d5cbed1fc. Thank you. c -- \black\trash movie _SAME TIME SAME PLACE_ New York, in the summer of 2001 --->> http://www.blacktrash.org/underdogma/stsp.php From blacktrash at gmx.net Sun Nov 20 22:22:43 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Sun, 20 Nov 2011 21:22:43 +0000 Subject: [FFmpeg-user] [SOLVED] build failure on MacOS 10.5.8 In-Reply-To: <20111119160125.GX835@krille.blacktrash.org> References: <20111119160125.GX835@krille.blacktrash.org> Message-ID: <20111120212243.GE88875@krille.blacktrash.org> * Christian Ebert on Saturday, November 19, 2011 at 16:01:25 +0000 > git head fails to build on MacOS 10.5.8 and --enable-shared: > > LD libavformat/libavformat.53.dylib > Undefined symbols: > "_ff_httpproxy_protocol", referenced from: > _ff_httpproxy_protocol$non_lazy_ptr in allformats.o > ld: symbol(s) not found > collect2: ld returned 1 exit status > make: *** [libavformat/libavformat.53.dylib] Error 1 Solved, most probably by 66e9c0b6ab5360f144cf8f9912ab003d5cbed1fc. Thank you. c -- \black\trash movie _SAME TIME SAME PLACE_ New York, in the summer of 2001 --->> http://www.blacktrash.org/underdogma/stsp.php From cehoyos at ag.or.at Sun Nov 20 22:51:12 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 20 Nov 2011 21:51:12 +0000 (UTC) Subject: [FFmpeg-user] How do I add a new audio stream? References: <1321824113329-4089600.post@n4.nabble.com> Message-ID: Tuuls mail.ru> writes: > Hi all. How, after all the new changes, you can add a new audio stream to the > .mov file? > G:\ffmpeg\bin>ffmpeg -threads 4 -i E:\IN.avi -pix_fmt yuv420p -codec:v > mpeg2video -b:v 25000k -r 25 -copyts -codec:a pcm_s16le -ar 48000 -ac 2 -y > d:\out.mov -map 0:1 -codec:a pcm_s16le -ar 48000 -ac 2 -newaudio (Complete, uncut output console output missing, therefore I have to guess.) ffmpeg -i i.avi -map 0:0 -map 0:1 -map 0:1 -c:v mpeg2video -c:a pcm_s16le out.mov Carl Eugen From mark at mdsh.com Sun Nov 20 22:56:33 2011 From: mark at mdsh.com (Mark Himsley) Date: Sun, 20 Nov 2011 21:56:33 +0000 Subject: [FFmpeg-user] Duplicates frames when converting interlaced h264 to png frames and double length when converting interlaced .png to interlaced mp4 In-Reply-To: References: Message-ID: <4EC97791.1090500@mdsh.com> On 18/11/2011 21:39, Lionel Tr?saugues wrote: > Dear FFmpeg users, > > I am currently encountering two major problems during my video-editing > process (I am running the 64bits edition of Ubuntu Linux 11.10). My > camcorder (Panasonic HDC-SD60) shots 1080/50i (AVCHD in a mts > container). > > First problem) > > I want, in a first step, to convert the .mts into a sequence of .png > frames while preserving the interlacing. > > To do so, I used the following FFmpeg command-line: > > ffmpeg -i 00030.MTS -an -vcodec png -r 50 -flags +ilme+ildct+mv4+aic > -top 1 -qscale 1 -s 1920x1080 -sameq 00030%05d.png > > Then for a 2min21sec long .mts movie, ffmpeg produces 7080 frames each > of them clearly showing signs of interlacing (which was the expected > result) (and 7080/50=2min21sec). > > But, indeed, two consecutive frames are absolutely identical (e.g > frames 1 and 2 are identical, 3 and 4 are identical, 5 and 6, etc..) I suspect you are mistaken in your thinking of what 1080/50i means. Unless I am very much mistaken, and your description backs me up on this, your camera is actually shooting 25 interlaced frames per second. That is 50 interlaced fields per second, which is why some manufacturers write 50i. And since 50 is twice as many as 25 it's got to be better. This really gets my goat. To quote Wikipedia, so it must be right: The European Broadcasting Union (EBU) prefers to use the resolution and frame rate (not field rate) separated by a slash, as in 1080i/30 and 1080i/25, likewise 480i/30 and 576i/25.[1] Resolutions of 1080i60 or 1080i50 often refers to 1080i/30 or 1080i/25 in EBU notation. http://en.wikipedia.org/wiki/1080i So, try -r 25. I think that will give you 25 interlaced frames, each being unique. HTH. > When I try to do the same process using avisynth and VirtualDub (on > windows), while using this avisynth script : > DirectShowSource("E:\00030.MTS").AssumeTFF().DoubleWeave() > I, then obtain once again 7080 interlaced png frames but this time > they all are different. > > So, first question, how would I be able to produce the same results > with ffmpeg ? > > Second problem) > > Then, I want to convert the interlaced png frames (ideally after > processing in Blender) and merge them with an audio track into a mp4 > container and still want to preserve the interlacing. > > I used the following command-line : > > ffmpeg -threads 2 -i %05d.png -i audio.aac -qscale 1 -vcodec mpeg4 > -acodec copy -minrate 0 -b 50000k -mbd 2 -trellis 2 -cmp 2 -subcmp 2 > -g 300 -r 50 -flags +ilme+ildct+mv4+aic -top 1 -f mp4 00030.mp4 > > And I obtain a slow-motion interlaced movie which lasts twice the > duration of the initial footage. The problem is the same whether I use > the 7080 frames produced by ffmpeg (containing duplicates) or the 7080 > frames (without duplicates) produced by VirtualDub. > > How would I be able to use the sequence of .png frames to produce an > interlaced movie (50i) in a mp4 container correctly ? > > Thanks a lot for your suggestions > [...] -- Mark From koxaniy at mail.ru Sun Nov 20 23:05:31 2011 From: koxaniy at mail.ru (Tuuls) Date: Sun, 20 Nov 2011 14:05:31 -0800 (PST) Subject: [FFmpeg-user] How do I add a new audio stream? In-Reply-To: References: <1321824113329-4089600.post@n4.nabble.com> Message-ID: <1321826731491-4089741.post@n4.nabble.com> Thank you so much! The option -newaudio is no longer used? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-do-I-add-a-new-audio-stream-tp4089600p4089741.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From koxaniy at mail.ru Sun Nov 20 23:38:54 2011 From: koxaniy at mail.ru (Tuuls) Date: Sun, 20 Nov 2011 14:38:54 -0800 (PST) Subject: [FFmpeg-user] How do I add a new audio stream? In-Reply-To: <1321826731491-4089741.post@n4.nabble.com> References: <1321824113329-4089600.post@n4.nabble.com> <1321826731491-4089741.post@n4.nabble.com> Message-ID: <1321828734389-4089839.post@n4.nabble.com> Another question! How different channels of a stereo, expanded in different streams ? If I use this option: -map 0:1 -map_channel 0.1.0 -map 0:1 -map_channel 0.1.1 -codec:a pcm_s16le -ar 48000 -ac 1 the output I get two stream, but the same channels in it, and not different, the left and right! How to get different channels in different stream? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-do-I-add-a-new-audio-stream-tp4089600p4089839.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From zeoxyvid at gmail.com Sun Nov 20 22:33:51 2011 From: zeoxyvid at gmail.com (zeo) Date: Sun, 20 Nov 2011 13:33:51 -0800 (PST) Subject: [FFmpeg-user] frame grabbing from MP4 file takes too long (says: muxing overhead -inf%) Message-ID: <1321824831877-4089662.post@n4.nabble.com> Hi everybody, I am trying to grab a frame from a MP4 video, which is recorded on my Samsung Galaxy S android phone. But it takes too long to grab a frame, ffmpeg produces the following output. If I use a smaller starting time like -ss 00:00.04, then it performs faster without saying "muxing overhead -inf%". Any help is appreciated, thank you very much. Zeo ---------------------------------------------------------------------------------------------------- $ ./ffmpeg.exe -i input.mp4 -ss 00:01:34 -vframes 1 -y out.jpg Seems stream 1 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 30.00 (30/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'utkunamik.mp4': Metadata: major_brand : 3gp4 minor_version : 768 compatible_brands: 3gp43gp6 Duration: 00:01:21.57, start: 0.000000, bitrate: 11099 kb/s Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, s16, 62 kb/s Metadata: handler_name : SoundHandler Stream #0:1(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 11032 kb/s, 29.70 fps, 30 tbr, 30k tbn, 30k tbc Metadata: handler_name : VideoHandler Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p' [buffer @ 0x10464ae0] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [buffersink @ 0x1046e3e0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out' [scale @ 0x1046e140] w:1280 h:720 fmt:yuv420p -> w:1280 h:720 fmt:yuvj420p flags:0x4 Output #0, image2, to 'out.jpg': Metadata: major_brand : 3gp4 minor_version : 768 compatible_brands: 3gp43gp6 encoder : Lavf53.21.0 Stream #0:0(eng): Video: mjpeg, yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 30 tbc Metadata: handler_name : VideoHandler Stream mapping: Stream #0:1 -> #0:0 (mpeg4 -> mjpeg) Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 Lsize= -0kB time=00:00:00.00 bitrate= 0.0kbits/s video:0kB audio:0kB global headers:0kB muxing overhead -inf% --------------------------------------------------------------------------------------------------- -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/frame-grabbing-from-MP4-file-takes-too-long-says-muxing-overhead-inf-tp4089662p4089662.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From lou at lrcd.com Mon Nov 21 00:03:13 2011 From: lou at lrcd.com (Lou) Date: Sun, 20 Nov 2011 14:03:13 -0900 Subject: [FFmpeg-user] frame grabbing from MP4 file takes too long (says: muxing overhead -inf%) In-Reply-To: <1321824831877-4089662.post@n4.nabble.com> References: <1321824831877-4089662.post@n4.nabble.com> Message-ID: <20111120140313.77f86b6e@lrcd.com> On Sun, 20 Nov 2011 13:33:51 -0800 (PST) zeo wrote: > Hi everybody, > I am trying to grab a frame from a MP4 video, which is recorded on my > Samsung Galaxy S android phone. But it takes too long to grab a frame, > ffmpeg produces the following output. If I use a smaller starting > time like -ss 00:00.04, then it performs faster without saying > "muxing overhead -inf%". Any help is appreciated, thank you very much. > Zeo > > > ---------------------------------------------------------------------------------------------------- > $ ./ffmpeg.exe -i input.mp4 -ss 00:01:34 -vframes 1 -y out.jpg > > Seems stream 1 codec frame rate differs from container frame rate: > 30000.00 (30000/1) -> 30.00 (30/1) > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'utkunamik.mp4': > Metadata: > major_brand : 3gp4 > minor_version : 768 > compatible_brands: 3gp43gp6 > Duration: 00:01:21.57, start: 0.000000, bitrate: 11099 kb/s > Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, > s16, 62 kb/s > Metadata: > handler_name : SoundHandler > Stream #0:1(eng): Video: mpeg4 (Simple Profile) (mp4v / > 0x7634706D), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 11032 kb/s, 29.70 > fps, 30 tbr, 30k tbn, 30k tbc > Metadata: > handler_name : VideoHandler > Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting > format 'yuvj420p' > [buffer @ 0x10464ae0] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 > sws_param: > [buffersink @ 0x1046e3e0] auto-inserting filter 'auto-inserted scale > 0' between the filter 'src' and the filter 'out' > [scale @ 0x1046e140] w:1280 h:720 fmt:yuv420p -> w:1280 h:720 > fmt:yuvj420p flags:0x4 > Output #0, image2, to 'out.jpg': > Metadata: > major_brand : 3gp4 > minor_version : 768 > compatible_brands: 3gp43gp6 > encoder : Lavf53.21.0 > Stream #0:0(eng): Video: mjpeg, yuvj420p, 1280x720 [SAR 1:1 DAR > 16:9], q=2-31, 200 kb/s, 90k tbn, 30 tbc > Metadata: > handler_name : VideoHandler > Stream mapping: > Stream #0:1 -> #0:0 (mpeg4 -> mjpeg) > Press [q] to stop, [?] for help > frame= 0 fps= 0 q=0.0 Lsize= -0kB time=00:00:00.00 bitrate= > 0.0kbits/s > video:0kB audio:0kB global headers:0kB muxing overhead -inf% Try moving -ss as an input option: ffmpeg -ss 00:01:34 -i input.mp4 -vframes 1 -y out.jpg This can be faster than using ss as an output option, but has the potential of not being frame accurate and may not provide satisfactory results (depending on your input format). To generalize: with -ss as an output option, the video will decode until 1:34 and then ffmpeg will start encoding. As an input option ffmpeg will attempt to immediately seek to 1:34 and then start decoding. If you must use -ss as an output option then you can try to decrease decoding time with -threads as an input option: ffmpeg -threads 4 -i input.mp4 -ss 00:01:34 -vframes 1 -y out.jpg However not all decoders can utilize -ss, if I recall correctly, so it may not provide any speed increase depending on your input format. From cehoyos at ag.or.at Mon Nov 21 00:56:09 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 20 Nov 2011 23:56:09 +0000 (UTC) Subject: [FFmpeg-user] frame grabbing from MP4 file takes too long (says: muxing overhead -inf%) References: <1321824831877-4089662.post@n4.nabble.com> Message-ID: zeo gmail.com> writes: > $ ./ffmpeg.exe -i input.mp4 -ss 00:01:34 -vframes 1 -y out.jpg > Duration: 00:01:21.57 Are you trying to seek over 90 seconds into a video that is less than 82 seconds long and ask why the output file is empty? (You could try ffmpeg -i input.mp4 -f null - to find out how long the sample really is, above line is not necessarily correct.) Carl Eugen From zeoxyvid at gmail.com Mon Nov 21 01:08:13 2011 From: zeoxyvid at gmail.com (zeo) Date: Sun, 20 Nov 2011 16:08:13 -0800 (PST) Subject: [FFmpeg-user] frame grabbing from MP4 file takes too long (says: muxing overhead -inf%) In-Reply-To: <1321824831877-4089662.post@n4.nabble.com> References: <1321824831877-4089662.post@n4.nabble.com> Message-ID: <1321834093058-4090032.post@n4.nabble.com> Actually for the above sample, Carl is right, I was trying a frame beyond the end file. However, I got the same result for 00:01:04 for example. Also I've tried with longer videos for different time values. It seems like Lou-2's suggestion solved my problem, I've moved -ss option to the beginning. "ffmpeg -ss 00:01:04 -i input.mp4 -vframes 1 -y out.jpg" Thank you very much... -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/frame-grabbing-from-MP4-file-takes-too-long-says-muxing-overhead-inf-tp4089662p4090032.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ballenato at gmail.com Mon Nov 21 11:26:03 2011 From: ballenato at gmail.com (=?ISO-8859-1?Q?Fernando_S=E1nchez?=) Date: Mon, 21 Nov 2011 12:26:03 +0200 Subject: [FFmpeg-user] Remuxed MP4 (H.264) plays audio but no video in QuickTime In-Reply-To: References: <4EC92B46.1080800@gmail.com> Message-ID: 2011/11/20 Carl Eugen Hoyos > Fernando S?nchez gmail.com> writes: > > > FFmpeg version SVN-r21566-xuggle-3.4.843 > > This is ancient, please test current git head (a snapshot is fine): > http://ffmpeg.org/download.html > > Does out.mov work? > > Carl Eugen > > Hi Carl, I tried with the latest snapshot but the problem is still there: no video in QuickTime, just audio. Here's the new output attached. I also remuxed into MOV and the same problem happens. Please let me know if you need any other information, and thanks for your time. -------------- next part -------------- $ ffmpeg -i problem_video_qt.flv -acodec copy -vcodec copy problem_video_qt .mp4 ffmpeg version N-35057-g2c44aed, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 21 2011 02:41:21 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-ru ntime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libope ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --en able-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwben c --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable- libxvid --enable-zlib libavutil 51. 26. 0 / 51. 26. 0 libavcodec 53. 36. 0 / 53. 36. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [h264 @ 0000000001BE0280] Increasing reorder buffer to 1 [flv @ 0000000001D0DB50] Estimating duration from bitrate, this may be inaccurat e Input #0, flv, from 'problem_video_qt.flv': Duration: N/A, start: 0.001000, bitrate: N/A Stream #0:0: Video: h264 (Main), yuv420p, 640x360, 25 tbr, 1k tbn, 50 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, s16 Output #0, mp4, to 'problem_video_qt.mp4': Metadata: encoder : Lavf53.21.0 Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 640x360, q=2-31, 25 tbn, 25 tbc Stream #0:1: Audio: aac (@[0][0][0] / 0x0040), 44100 Hz, stereo Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 301 fps= 0 q=-1.0 Lsize= 1107kB time=00:00:08.73 bitrate=1038.6kbits /s video:1032kB audio:67kB global headers:0kB muxing overhead 0.737801% From seba.wagner at gmail.com Mon Nov 21 11:46:58 2011 From: seba.wagner at gmail.com (seba.wagner at gmail.com) Date: Mon, 21 Nov 2011 11:46:58 +0100 Subject: [FFmpeg-user] Invalid stream specifier: .0. / Stream map '0.0' matches no streams - Changes in FFMPEG -map option ? Message-ID: Hi, I have a problem with latest FFMPEG build, it seems like the problem did not exist with previous version of FFMPEG. The Command is (paths shortened, long version at the bottom): ffmpeg -i file1.flv -i wave1.wav -ar 22050 -acodec libmp3lame -ab 32k -s 1664x1328 -vcodec flashsv -map 0.0 -map 1.0 result.flv Error message: [flv @ 020BA040] Invalid stream specifier: .0. Last message repeated 1 times Stream map '0.0' matches no streams. *Is there anything wrong in my command params? Have there been changes on how FFMPEG handles the "-map" option?* Full trace of the command looks like that: Command: ffmpeg -i D:\work\workspaces\indigo_om\ROOT_branch_AV\dist\red5\webapps\openmeetings\streams\2\rec_2_stream_bbd17adb95120c585a7c5dc84a842be9_21_11_2011_11_29_21.flv -i D:\work\workspaces\indigo_om\ROOT_branch_AV\dist\red5\webapps\openmeetings\streams\2\rec_2_stream_bbd17adb95120c585a7c5dc84a842be9_21_11_2011_11_29_21_FINAL_WAVE.wav -ar 22050 -acodec libmp3lame -ab 32k -s 1664x1328 -vcodec flashsv -map 0.0 -map 1.0 D:\work\workspaces\indigo_om\ROOT_branch_AV\dist\red5\webapps\openmeetings\streams\hibernate\flvRecording_2.flv Output: ffmpeg version N-35057-g2c44aed, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 21 2011 02:36:31 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enab le-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 26. 0 / 51. 26. 0 libavcodec 53. 36. 0 / 53. 36. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [flv @ 020BA040] Could not find codec parameters (Audio: none, 0 channels) Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 1.00 (1/1) Input #0, flv, from 'D:\work\workspaces\indigo_om\ROOT_branch_AV\dist\red5\webapps\openmeetings\streams\2\rec_2_stream_bbd17adb95120c585a7c5dc84a842be9_21_11_2011_11_29_21.flv': Metadata: server : Red5 Server 1.0.0 RC1 $Rev: 4193 $ canSeekToEnd : true Duration: 00:00:10.00, start: 0.000000, bitrate: 944 kb/s Stream #0:0: Video: flashsv, bgr24, 1280x1024, 1 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: none, 0 channels [wav @ 021B6020] parser not found for codec pcm_s16le, packets or times may be invalid. [wav @ 021B6020] max_analyze_duration 5000000 reached at 5015510 Input #1, wav, from 'D:\work\workspaces\indigo_om\ROOT_branch_AV\dist\red5\webapps\openmeetings\streams\2\rec_2_stream_bbd17adb95120c585a7c5dc84a842be9_21_11_2011_11_29_21_FINAL_WAVE.wav': Duration: 00:00:10.89, bitrate: 705 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s [flv @ 020BA040] Invalid stream specifier: .0. Last message repeated 1 times Stream map '0.0' matches no streams. Thanks Sebastian -- Sebastian Wagner http://www.openmeetings.de http://www.webbase-design.de http://www.wagner-sebastian.com seba.wagner at gmail.com From cehoyos at ag.or.at Mon Nov 21 11:49:57 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 10:49:57 +0000 (UTC) Subject: [FFmpeg-user] Invalid stream specifier: .0. / Stream map '0.0' matches no streams - Changes in FFMPEG -map option ? References: Message-ID: seba.wagner gmail.com gmail.com> writes: > The Command is (paths shortened, long version at the bottom): > ffmpeg -i file1.flv -i wave1.wav -ar 22050 -acodec libmp3lame -ab 32k -s > 1664x1328 -vcodec flashsv -map 0.0 -map 1.0 result.flv It's now -map 0:0 and -map 1:0 _^_ _^_ Carl Eugen From seba.wagner at gmail.com Mon Nov 21 11:58:50 2011 From: seba.wagner at gmail.com (seba.wagner at gmail.com) Date: Mon, 21 Nov 2011 11:58:50 +0100 Subject: [FFmpeg-user] Invalid stream specifier: .0. / Stream map '0.0' matches no streams - Changes in FFMPEG -map option ? In-Reply-To: References: Message-ID: Thanks Carl, working fine! Since what revision do we need to change that map param syntax? Thanks, Sebastian 2011/11/21 Carl Eugen Hoyos > seba.wagner gmail.com gmail.com> writes: > > > The Command is (paths shortened, long version at the bottom): > > ffmpeg -i file1.flv -i wave1.wav -ar 22050 -acodec libmp3lame -ab 32k -s > > 1664x1328 -vcodec flashsv -map 0.0 -map 1.0 result.flv > > It's now -map 0:0 and -map 1:0 > _^_ _^_ > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Sebastian Wagner http://www.openmeetings.de http://www.webbase-design.de http://www.wagner-sebastian.com seba.wagner at gmail.com From pb at das-werkstatt.com Mon Nov 21 12:56:41 2011 From: pb at das-werkstatt.com (Peter B.) Date: Mon, 21 Nov 2011 12:56:41 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? Message-ID: <4ECA3C79.5040102@das-werkstatt.com> Hello, I am evaluating Dirac's for video archiving, but I cannot figure out how to enable its lossless mode. According to a thread on Doom10 forums (from June 2011) [1], the parameter "qscale" should be used for this, but currently only supports lossy mode. Is this still the case? I would be very grateful if someone could point me in the right direction. Thanks in advance, Pb == References: [1] http://doom10.org/index.php?topic=366.0 From cehoyos at ag.or.at Mon Nov 21 13:18:40 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 12:18:40 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > I am evaluating Dirac's for video archiving, but I cannot figure out how > to enable its lossless mode. According to a thread on Doom10 forums > (from June 2011) [1], the parameter "qscale" should be used for this, > but currently only supports lossy mode. Why do you think so? If it fails, please provide command line and complete, uncut console output. Carl Eugen From pb at das-werkstatt.com Mon Nov 21 14:39:22 2011 From: pb at das-werkstatt.com (Peter B.) Date: Mon, 21 Nov 2011 14:39:22 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: References: <4ECA3C79.5040102@das-werkstatt.com> Message-ID: <4ECA548A.8020900@das-werkstatt.com> Carl Eugen Hoyos wrote: >> but currently only supports lossy mode. >> > > Why do you think so? > If it fails, please provide command line and complete, uncut console output. > Sure. (Sorry for not posting it in my earlier message) [code] ffmpeg -i input_ffv1.avi -an -vcodec libschroedinger -qscale 255 test_dirac.mkv [/code] I've also tried "-qscale 0.0001" as suggested in an earlier thread on the ffmpeg-user list in July 2011 [1] by the user "charvak", since it's still unclear to me whether a high or low qscale value triggers lossless mode. Then I've compared the output of "framemd5" of both Dirac encodings with its FFv1 original (using the current git-version of ffmpeg): [code] ffmpeg -i dirac-x.mkv -an -f framemd5 dirac-x-checksums.txt [/code] The results were: *) MD5 checksums of both Dirac encoded videos had no match with the FFv1 checksums *) Surprisingly, 'some' of the checksums of the Dirac encodings (q=0.0001 and q=255) matched - which would mean that they decode to exactly same image. Although I still wouldn't consider any of these encodings to be lossless. Pb == References: [1] http://ffmpeg.org/pipermail/ffmpeg-user/2011-July/001745.html From cehoyos at ag.or.at Mon Nov 21 18:41:40 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 17:41:40 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > [code] > ffmpeg -i input_ffv1.avi -an -vcodec libschroedinger -qscale 255 > test_dirac.mkv > [/code] 255 requests maximum compression (worst quality), does 0 work? Carl Eugen From adamklobukowski at gmail.com Mon Nov 21 18:51:35 2011 From: adamklobukowski at gmail.com (=?UTF-8?B?QWRhbSBLxYJvYnVrb3dza2k=?=) Date: Mon, 21 Nov 2011 18:51:35 +0100 Subject: [FFmpeg-user] 5.1 sound aac Message-ID: <4ECA8FA7.4040900@gmail.com> Hello I'm reencoding 5.1 sound from ac3 to aac. I have it scripted to reencode a lot of different videos. I use ffprobe to get source bitrate, and use it for output bitrate. So, with source audio bitrete reported as 384k I reencode it like this: ffmpeg -threads 4 -i video.mkv -vcodec copy -acodec aac -ab 384k -strict experimental -f mp4 video.mp4 But, it is wrong. I'n fact, with 5.1 sound bitrate is divided per subchannel, so effective bitrate is much lower (and it can be heard). There are 2 possibilities: 1. ffprobe reports wrong bitrate 2. ffmpeg uses wrong bitrate 3. I'm wrong and do not understand what it all means. Which one is true? AdamK From pb at das-werkstatt.com Mon Nov 21 18:55:06 2011 From: pb at das-werkstatt.com (Peter B.) Date: Mon, 21 Nov 2011 18:55:06 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> Message-ID: <4ECA907A.2060706@das-werkstatt.com> Carl Eugen Hoyos wrote: > 255 requests maximum compression (worst quality), does 0 work? > No, because when trying "0" it says: [quote] qscale must be > 0.0 and <= 255 [/quote] Which is why I've tried a value < 1.0, hoping that if it's smaller than 1, it enables lossless ;) I must admit, I could take a peek at the code to see how qfactor is handled, but I was hoping that, since Dirac could be used by others out there, someone could give me an example of how to do it. Or is there any documentation for Dirac's codec parameters, which I just couldn't find? Thanks, Pb From ian at crcw.mb.ca Mon Nov 21 16:41:41 2011 From: ian at crcw.mb.ca (Ian) Date: Mon, 21 Nov 2011 09:41:41 -0600 Subject: [FFmpeg-user] stop motion with synced audio Message-ID: <4ECA7135.6020507@crcw.mb.ca> Hi. Iam trying to create stop motion video (mp4), with an audio track(mp3). The video source is jpegs. I would like to add mp3 audio to this and sync the audio with the video (dropping/delaying video frames accordingly) Can you help? From yohan at sim-indonesia.com Mon Nov 21 19:11:19 2011 From: yohan at sim-indonesia.com (yohan Romadhoni) Date: Tue, 22 Nov 2011 01:11:19 +0700 Subject: [FFmpeg-user] Fwd: question please about ffmpeg In-Reply-To: References: Message-ID: Dear, Sir Sir, Im work in industries broadcast indonesia i have a question for you. ffmpeg how to produce the best results with good image quality but small size (bytes) in the format mp4, mov? thanks.. From rogerdpack2 at gmail.com Mon Nov 21 20:07:20 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 21 Nov 2011 12:07:20 -0700 Subject: [FFmpeg-user] Fwd: question please about ffmpeg In-Reply-To: References: Message-ID: > ffmpeg how to produce the best results with good image quality but small > size (bytes) in the format mp4, mov? Maybe x264 preset's? In general, dual pass encoding is higher quality. GL. -r From startx at plentyfact.org Mon Nov 21 21:16:04 2011 From: startx at plentyfact.org (startx) Date: Mon, 21 Nov 2011 20:16:04 +0000 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <4ECA907A.2060706@das-werkstatt.com> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> Message-ID: <20111121201604.6228bc4b@worthil> On Mon, 21 Nov 2011 18:55:06 +0100 "Peter B." wrote: > Carl Eugen Hoyos wrote: > > 255 requests maximum compression (worst quality), does 0 work? > > > No, because when trying "0" it says: > > [quote] > qscale must be > 0.0 and <= 255 > [/quote] "0" didnt use to work due to a bug in libschroedinger, as far as i recall it. check out this thread: http://doom10.org/index.php?topic=366.0 startx From ib at wupperonline.de Mon Nov 21 22:04:20 2011 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Mon, 21 Nov 2011 22:04:20 +0100 Subject: [FFmpeg-user] mp3 sound bug In-Reply-To: Message-ID: <4ecabfac.271cc6c1.bm000@wupperonline.de> Carl Eugen Hoyos wrote on Sun, 20 Nov 2011 15:56:22 +0000 (UTC): > Ingo Br?ckl wupperonline.de> writes: >> Commit 22e25c002e103e52ace35703423e896b08b51aef introduced some bug that >> makes ffmp3float produce no sound but only occasional crackles. > Complete, uncut output missing and please point to a sample on > samples.mplayerhq.hu that allows to reproduce the problem. Sample (including both the mp3 and output of './ffmpeg -i 4210.mp3 outf.wav') can be found at http://www.load.to/gFAEGFe4Wy/outf.zip. ffmepg has been compiled with --disable-decoder=mp2,mp3 to force usage of mp3float. Output attached, CPU info follows: processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 6 model name : AMD Athlon(tm) XP 1700+ stepping : 2 cpu MHz : 1465.933 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 mmx fxsr sse syscall mp mmxext 3dnowext 3dnow bogomips : 2931.86 clflush size : 32 power management: ts ffmpeg's mp3float produces only 00 80 patterns with any mp3. Ingo -------------- next part -------------- [mp3 @ 0x9daba80] max_analyze_duration 5000000 reached at 5015510 [mp3 @ 0x9daba80] Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from '4210.mp3': Metadata: comment : genre : Other Duration: 00:01:04.87, start: 0.000000, bitrate: 95 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, flt, 96 kb/s Incompatible sample format 'flt' for codec 'pcm_s16le', auto-selecting format 's16' Output #0, wav, to 'outf.wav': Metadata: comment : genre : Other encoder : Lavf53.21.0 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Stream mapping: Stream #0:0 -> #0:0 (mp3float -> pcm_s16le) Press [q] to stop, [?] for help size= 9792kB time=00:00:57.02 bitrate=1406.7kbits/s size= 11115kB time=00:01:04.52 bitrate=1411.2kbits/s video:0kB audio:11115kB global headers:0kB muxing overhead 0.000404% From cehoyos at ag.or.at Mon Nov 21 23:06:17 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 22:06:17 +0000 (UTC) Subject: [FFmpeg-user] 5.1 sound aac References: <4ECA8FA7.4040900@gmail.com> Message-ID: Adam K?obukowski gmail.com> writes: > ffmpeg -threads 4 -i video.mkv -vcodec copy -acodec aac -ab 384k -strict > experimental -f mp4 video.mp4 > But, it is wrong. I'n fact, with 5.1 sound bitrate is divided per > subchannel, I don't think this is correct. > so effective bitrate is much lower (and it can be heard). > There are 2 possibilities: > 1. ffprobe reports wrong bitrate Very unlikely > 2. ffmpeg uses wrong bitrate Very possible, but please note that ffmpeg tells you the effective bitrate, so it should be easy for you to tell > 3. I'm wrong and do not understand what it all means. What is most important is that you realize the reason you have to use -strict experimental is that the native AAC encoder (as opposed to the three alternative AAC encoders) is experimental and does not necessarily produce good results, this may be more noticeable for multichannel. I always use high bitrates when I test the native AAC encoder, for multi-channel, you may test 0.8.7, it may be slightly better than current git head. (Please report if you can reproduce!) What I forgot when this was discussed last week is that work on the native AAC encoder is as welcome as sponsoring work on the native AAC encoder! Carl Eugen From lionel.tresaugues at gmail.com Mon Nov 21 23:12:53 2011 From: lionel.tresaugues at gmail.com (=?ISO-8859-1?Q?Lionel_Tr=E9saugues?=) Date: Mon, 21 Nov 2011 23:12:53 +0100 Subject: [FFmpeg-user] Duplicates frames when converting interlaced h264 to png frames and double length when converting interlaced .png to interlaced mp4 In-Reply-To: <4EC97791.1090500@mdsh.com> References: <4EC97791.1090500@mdsh.com> Message-ID: <4ECACCE5.5050503@gmail.com> On 11/20/2011 10:56 PM, Mark Himsley wrote: > On 18/11/2011 21:39, Lionel Tr?saugues wrote: >> Dear FFmpeg users, >> >> I am currently encountering two major problems during my video-editing >> process (I am running the 64bits edition of Ubuntu Linux 11.10). My >> camcorder (Panasonic HDC-SD60) shots 1080/50i (AVCHD in a mts >> container). >> >> First problem) >> >> I want, in a first step, to convert the .mts into a sequence of .png >> frames while preserving the interlacing. >> >> To do so, I used the following FFmpeg command-line: >> >> ffmpeg -i 00030.MTS -an -vcodec png -r 50 -flags +ilme+ildct+mv4+aic >> -top 1 -qscale 1 -s 1920x1080 -sameq 00030%05d.png >> >> Then for a 2min21sec long .mts movie, ffmpeg produces 7080 frames each >> of them clearly showing signs of interlacing (which was the expected >> result) (and 7080/50=2min21sec). >> >> But, indeed, two consecutive frames are absolutely identical (e.g >> frames 1 and 2 are identical, 3 and 4 are identical, 5 and 6, etc..) > > I suspect you are mistaken in your thinking of what 1080/50i means. > Unless I am very much mistaken, and your description backs me up on > this, your camera is actually shooting 25 interlaced frames per > second. That is 50 interlaced fields per second, which is why some > manufacturers write 50i. And since 50 is twice as many as 25 it's got > to be better. And you were right ! My camera effectively shots in 1080i/25 (I was mistaken by the manual provided with the camcorder that was mentioning 50i). I didn't get until your message the fact that the two interlaced fields that compose a frame have the same timestamp. > > This really gets my goat. To quote Wikipedia, so it must be right: > > The European Broadcasting Union (EBU) prefers to use the resolution > and frame rate (not field rate) separated by a slash, as in 1080i/30 > and 1080i/25, likewise 480i/30 and 576i/25.[1] Resolutions of 1080i60 > or 1080i50 often refers to 1080i/30 or 1080i/25 in EBU notation. > > > http://en.wikipedia.org/wiki/1080i > > So, try -r 25. I think that will give you 25 interlaced frames, each > being unique. > > HTH. > I tried to replace -r 50 by -r 25 in the command-line I used to generate the .png frames and I didn't obtain any duplicate. Using these generated png as an input to produce an mp4 while, once again replacing -r 50 by -25, give now a fluid movie (with the right length and framerate) whose quality is similar to the one used to generated the png. Thank you a lot for your help. /lionel > >> When I try to do the same process using avisynth and VirtualDub (on >> windows), while using this avisynth script : >> DirectShowSource("E:\00030.MTS").AssumeTFF().DoubleWeave() >> I, then obtain once again 7080 interlaced png frames but this time >> they all are different. >> >> So, first question, how would I be able to produce the same results >> with ffmpeg ? >> >> Second problem) >> >> Then, I want to convert the interlaced png frames (ideally after >> processing in Blender) and merge them with an audio track into a mp4 >> container and still want to preserve the interlacing. >> >> I used the following command-line : >> >> ffmpeg -threads 2 -i %05d.png -i audio.aac -qscale 1 -vcodec mpeg4 >> -acodec copy -minrate 0 -b 50000k -mbd 2 -trellis 2 -cmp 2 -subcmp 2 >> -g 300 -r 50 -flags +ilme+ildct+mv4+aic -top 1 -f mp4 00030.mp4 >> >> And I obtain a slow-motion interlaced movie which lasts twice the >> duration of the initial footage. The problem is the same whether I use >> the 7080 frames produced by ffmpeg (containing duplicates) or the 7080 >> frames (without duplicates) produced by VirtualDub. >> >> How would I be able to use the sequence of .png frames to produce an >> interlaced movie (50i) in a mp4 container correctly ? >> >> Thanks a lot for your suggestions >> > [...] > From cehoyos at ag.or.at Mon Nov 21 23:16:47 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 22:16:47 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > Carl Eugen Hoyos wrote: > > 255 requests maximum compression (worst quality), does 0 work? > > > No, because when trying "0" it says: > > [quote] > qscale must be > 0.0 and <= 255 > [/quote] Command line and complete, uncut console output missing. Carl Eugen From lionel.tresaugues at gmail.com Mon Nov 21 23:20:08 2011 From: lionel.tresaugues at gmail.com (=?UTF-8?B?TGlvbmVsIFRyw6lzYXVndWVz?=) Date: Mon, 21 Nov 2011 23:20:08 +0100 Subject: [FFmpeg-user] Duplicates frames when converting interlaced h264 to png frames and double length when converting interlaced .png to interlaced mp4 In-Reply-To: References: Message-ID: <4ECACE98.9090003@gmail.com> On 11/18/2011 11:48 PM, Carl Eugen Hoyos wrote: > Lionel Tr?saugues gmail.com> writes: > >> ffmpeg -i 00030.MTS -an -vcodec png -r 50 -flags +ilme+ildct+mv4+aic >> -top 1 -qscale 1 -s 1920x1080 -sameq 00030%05d.png > (Complete, uncut output missing.) > -an, -vcodec, -flags, -qscale (?) and -s should be unneeded, I am not sure about > -top (does it make a difference?), -r 50 should be responsible for the > duplicated frames (and removing it may fix your second problem). > > [...] This problem came from a confusion in the format used by my camcorder. Actually, replacing -r 50 by -r 25 solved the problem. No more duplicates. >> ffmpeg -threads 2 -i %05d.png -i audio.aac -qscale 1 -vcodec mpeg4 >> -acodec copy -minrate 0 -b 50000k -mbd 2 -trellis 2 -cmp 2 -subcmp 2 >> -g 300 -r 50 -flags +ilme+ildct+mv4+aic -top 1 -f mp4 00030.mp4 > I don't think -qscale and -b (now -vb) make sense in the same command line... For this step, replacing -r 50 by -r 25 solved the problem too. >> ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the >> Libav developers > This is known to be _very_ buggy and is therefore unsupported. > > Carl Eugen > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Thank you for your advice, I have now update to version 0.8.6-4:0.8.6-0ubuntu1~jon1. Regards /lionel From cehoyos at ag.or.at Mon Nov 21 23:26:46 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 21 Nov 2011 22:26:46 +0000 (UTC) Subject: [FFmpeg-user] Invalid stream specifier: .0. / Stream map '0.0' matches no streams - Changes in FFMPEG -map option ? References: Message-ID: seba.wagner gmail.com gmail.com> writes: > working fine! > Since what revision do we need to change that map param syntax? Possibly since 88bfe4518. Carl Eugen From lou at lrcd.com Tue Nov 22 01:46:40 2011 From: lou at lrcd.com (Lou) Date: Mon, 21 Nov 2011 15:46:40 -0900 Subject: [FFmpeg-user] Fwd: question please about ffmpeg In-Reply-To: References: Message-ID: <20111121154640.20a5671b@lrcd.com> On Mon, 21 Nov 2011 12:07:20 -0700 Roger Pack wrote: > > On Tue, 22 Nov 2011 01:11:19 +0700 > > yohan Romadhoni wrote: > > ffmpeg how to produce the best results with good image quality but > > small size (bytes) in the format mp4, mov? The MP4 container format can utilize several video formats. Can you describe your project and tell us what format(s) you want? We can then give some example commands based on the additional information you provide. > Maybe x264 preset's? > In general, dual pass encoding is higher quality. > GL. > -r Are you comparing -crf vs two-pass -b, or single-pass -b vs two-pass -b? From pb at das-werkstatt.com Tue Nov 22 09:48:51 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 09:48:51 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <20111121201604.6228bc4b@worthil> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <20111121201604.6228bc4b@worthil> Message-ID: <4ECB61F3.3050602@das-werkstatt.com> startx wrote: > "0" didnt use to work due to a bug in libschroedinger, as far as i > recall it. > > check out this thread: > > http://doom10.org/index.php?topic=366.0 > That's exactly the thread I've posted in my first message ;) But still, thanks for the hint. Anyway, if I read "eetommyj"'s comment in that thread correctly, it says it's not a bug in libschroedinger, but "-qscale 0" gets intercepted by FFmpeg, because it only allows values > 0.0 Has noone else ever used FFmpeg to encode Dirac losslessly before? :( Pb From startx at plentyfact.org Tue Nov 22 09:55:31 2011 From: startx at plentyfact.org (startx) Date: Tue, 22 Nov 2011 08:55:31 +0000 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <4ECB61F3.3050602@das-werkstatt.com> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <20111121201604.6228bc4b@worthil> <4ECB61F3.3050602@das-werkstatt.com> Message-ID: <20111122085531.49994b51@worthil> On Tue, 22 Nov 2011 09:48:51 +0100 "Peter B." wrote: > startx wrote: > > "0" didnt use to work due to a bug in libschroedinger, as far as i > > recall it. > > > > check out this thread: > > > > http://doom10.org/index.php?topic=366.0 > > > That's exactly the thread I've posted in my first message ;) > But still, thanks for the hint. ooops, my fault :/ > Anyway, if I read "eetommyj"'s comment in that thread correctly, it > says it's not a bug in libschroedinger, but "-qscale 0" gets > intercepted by FFmpeg, because it only allows values > 0.0 > > Has noone else ever used FFmpeg to encode Dirac losslessly before? :( > i tried but failed and switched to ffmpeg2dirac ( which has other shortcomings though ). startx From pb at das-werkstatt.com Tue Nov 22 10:11:08 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 10:11:08 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> Message-ID: <4ECB672C.8010305@das-werkstatt.com> Carl Eugen Hoyos wrote: >> No, because when trying "0" it says: >> >> [quote] >> qscale must be > 0.0 and <= 255 >> [/quote] >> > > Command line and complete, uncut console output missing. > Sorry... [quote] user at hostname:~/test/dirac$ ffmpeg -i v-10922_01-071.avi -an -vcodec libdirac -qscale 0 test.mkv FFmpeg version 0.6, Copyright (c) 2000-2010 the FFmpeg developers built on Jul 30 2010 12:08:53 with gcc 4.3.2 configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-pthreads --enable-bzlib --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libschroedinger --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-libdirac 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 Input #0, avi, from 'v-10922_01-071.avi': Metadata: ISFT : Lavf52.31.0 Duration: 00:00:41.88, start: 0.000000, bitrate: 52834 kb/s Stream #0.0: Video: ffv1, yuv422p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s qscale must be > 0.0 and <= 255 [/quote] Hm... Now that I see this, I figure that for this test, I've used the wrong FFmpeg version! The git-version *does* allow "0" as qscale factor. Darn... I'll do some new testing... Thanks, Pb From pb at das-werkstatt.com Tue Nov 22 10:30:42 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 10:30:42 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <4ECB672C.8010305@das-werkstatt.com> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> Message-ID: <4ECB6BC2.9050005@das-werkstatt.com> Unfortunately, I'm having problems compiling the current git-version of FFmpeg with libschroedinger on my Debian Lenny machine: Using the following command: [code] ./configure --prefix=/usr/local --enable-gpl --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-decoder=png --enable-encoder=png --enable-libdirac --enable-libschroedinger && make clean && make [/code] Throws the following error: [quote] CC libavcodec/libschroedinger.o CC libavcodec/libschroedingerdec.o CC libavcodec/libschroedingerenc.o libavcodec/libschroedingerenc.c: In function ?libschroedinger_encode_init?: libavcodec/libschroedingerenc.c:191: error: ?SCHRO_ENCODER_RATE_CONTROL_CONSTANT_QUALITY? undeclared (first use in this function) libavcodec/libschroedingerenc.c:191: error: (Each undeclared identifier is reported only once libavcodec/libschroedingerenc.c:191: error: for each function it appears in.) make: *** [libavcodec/libschroedingerenc.o] Error 1 [/quote] I'm using "libschroedinger-dev" package from Lenny-backports: 1.0.8-2~bpo50+2 Do you think that getting the most recent libschroedinger (v1.0.10) will fix this? On its website [1] it says: New in 1.0.10 ============= ? Build fixes on various platforms ? Speed increases for low-delay syntax ? Fix unaligned access in orc code, which was uncovered by a recent Orc bug fix. Bump orc requirement to 0.4.10, which makes sure everyone works right. ? No encoder changes Regards, Pb == References: [1] http://diracvideo.org/ From cehoyos at ag.or.at Tue Nov 22 11:41:38 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Nov 2011 10:41:38 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > CC libavcodec/libschroedingerenc.o > libavcodec/libschroedingerenc.c: In function ?libschroedinger_encode_init?: > libavcodec/libschroedingerenc.c:191: error: > ?SCHRO_ENCODER_RATE_CONTROL_CONSTANT_QUALITY? undeclared (first use in > this function) Afaik, this was defined in 1.0.7, what does "pkg-config --modversion schroedinger-1.0" report? (I suspect you - also - have to backport the development package.) See also http://thread.gmane.org/gmane.comp.video.ffmpeg.cvs/30001/focus=30052 to understand why this wasn't fixed properly... Carl Eugen From pb at das-werkstatt.com Tue Nov 22 11:55:49 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 11:55:49 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> Message-ID: <4ECB7FB5.7070008@das-werkstatt.com> Carl Eugen Hoyos wrote: > Afaik, this was defined in 1.0.7, what does > "pkg-config --modversion schroedinger-1.0" report? > (I suspect you - also - have to backport the development package.) > running: [code] pkg-config --modversion schroedinger-1.0 [/code] returns: [quote] 1.0.5 [/quote] I've now compiled ffmpeg's current git-version with libschroedinger 1.0.9 on Debian Squeeze, because libschroedinger v1.0.10 on Lenny required ORC > 0.4.10 (so I gave up). Interestingly, with libdirac (pkgconfig --modversion: 1.0.0), "-qscale 0" works, but on Squeeze with libdirac (pkgconfig --modversion: 1.0.2), it doesn't work anymore. The command working fine on Lenny: [code] ffmpeg -i input-ffv1.avi -an -vcodec libdirac -qscale 0 dirac-0.mkv [/code] Again throws the error: [quote] qscale must be > 0.0 and <= 255" [/quote] Same for: [code] ffmpeg -i input-ffv1.avi -an -vcodec libschroedinger -qscale 0 dirac-0.mkv [/code] Now, I'm really confused... :( Pb From pb at das-werkstatt.com Tue Nov 22 11:59:17 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 11:59:17 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <4ECB7FB5.7070008@das-werkstatt.com> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> Message-ID: <4ECB8085.4010807@das-werkstatt.com> Peter B. wrote: > The command working fine on Lenny: > [code] > ffmpeg -i input-ffv1.avi -an -vcodec libdirac -qscale 0 dirac-0.mkv > [/code] > > Again throws the error: > [quote] > qscale must be > 0.0 and <= 255" > [/quote] > Forget that. My fault! Forgot to add the right path to the git-version of ffmpeg. Now it's working with "-qscale 0". Sorry, sorry... Pb From cehoyos at ag.or.at Tue Nov 22 11:58:39 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Nov 2011 10:58:39 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > [quote] > qscale must be > 0.0 and <= 255" > [/quote] > > Same for: > [code] > ffmpeg -i input-ffv1.avi -an -vcodec libschroedinger -qscale 0 dirac-0.mkv > [/code] Complete, uncut console output missing. Please do not post bug reports on -devel, they are not welcome there. If you believe http://ffmpeg.org/contact.html please tell us! Carl Eugen From cehoyos at ag.or.at Tue Nov 22 12:03:52 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Nov 2011 11:03:52 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > [quote] > 1.0.5 > [/quote] > > I've now compiled ffmpeg's current git-version with libschroedinger > 1.0.9 on Debian Squeeze, because libschroedinger v1.0.10 on Lenny > required ORC > 0.4.10 (so I gave up). As said, 1.0.7 is sufficient. > Interestingly, with libdirac (pkgconfig --modversion: 1.0.0), "-qscale > 0" works, but on Squeeze with libdirac (pkgconfig --modversion: 1.0.2), > it doesn't work anymore. I don't know much about dirac, but I believe the recommended encoder is libschroedinger (isn't libdirac deprecated?). Carl Eugen From pb at das-werkstatt.com Tue Nov 22 12:08:39 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 12:08:39 +0100 Subject: [FFmpeg-user] "--enable-avfilter-lavf" switch not working for v0.8.7 Message-ID: <4ECB82B7.7080509@das-werkstatt.com> I've been successfully compiling FFmpeg versions 0.5 and 0.6, as well as the current git-version, with the following commandline: [code] ./configure --prefix=/usr/local --enable-gpl --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-decoder=png --enable-encoder=png --enable-avfilter-lavf --enable-libfaad && make clean && make [/code] However, version 0.8.7 (from the tar.bz2 archive) throws the following error when running configure: [quote] Unknown option "--enable-avfilter-lavf". See ./configure --help for available options. [/quote] If that switch would be deprecated, why does it work for the current git-version? Thanks in advance, Pb From cehoyos at ag.or.at Tue Nov 22 12:05:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Nov 2011 11:05:36 +0000 (UTC) Subject: [FFmpeg-user] Encode lossless Dirac? References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> Message-ID: Carl Eugen Hoyos ag.or.at> writes: > Please do not post bug reports on -devel, they are not welcome there. > If you believe http://ffmpeg.org/contact.html please tell us! Should be: If you believe http://ffmpeg.org/contact.html is unclear, please tell us how we can improve it! Carl Eugen From pb at das-werkstatt.com Tue Nov 22 12:12:00 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 12:12:00 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> Message-ID: <4ECB8380.6000505@das-werkstatt.com> Carl Eugen Hoyos wrote: > Peter B. das-werkstatt.com> writes: > > >> [quote] >> qscale must be > 0.0 and <= 255" >> [/quote] >> >> Same for: >> [code] >> ffmpeg -i input-ffv1.avi -an -vcodec libschroedinger -qscale 0 dirac-0.mkv >> [/code] >> > > Complete, uncut console output missing. > Sorry, I just always have the feeling that I'd be unnecessarily flooding the message here with that output. But I'll just suppress that feeling and post the output. You're absolutely right: If I had posted it, you would have seen that I've been using the wrong FFmpeg version in that case (my bad) :( > Please do not post bug reports on -devel, they are not welcome there. > If you believe http://ffmpeg.org/contact.html please tell us! > I thought that compile errors might rather go on -devel than on -user, but now I know for the future. I've resent the message to -user now. Hope that's ok. Thank you very much for your time, Pb From pb at das-werkstatt.com Tue Nov 22 12:27:28 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 12:27:28 +0100 Subject: [FFmpeg-user] Encode lossless Dirac? In-Reply-To: <4ECB8380.6000505@das-werkstatt.com> References: <4ECA3C79.5040102@das-werkstatt.com> <4ECA548A.8020900@das-werkstatt.com> <4ECA907A.2060706@das-werkstatt.com> <4ECB672C.8010305@das-werkstatt.com> <4ECB6BC2.9050005@das-werkstatt.com> <4ECB7FB5.7070008@das-werkstatt.com> <4ECB8380.6000505@das-werkstatt.com> Message-ID: <4ECB8720.6030807@das-werkstatt.com> Success! The current git-version of ffmpeg (N-35077-g7876f14) encoded a Dirac video losslessly, with "libschroedinger" v1.0.9, using the following command: [code] ffmpeg -i input-ffv1.avi -an -vcodec libschroedinger -qscale 0 dirac-0.mkv [/code] I've verified that its lossless, using "-f framemd5" and "vimdiff" like this: [code] ffmpeg -i dirac-0.mkv -an -f framemd5 dirac-0-checksums.txt ffmpeg -i input-ffv1.avi -an -f framemd5 input-ffv1-checksums.txt vimdiff *-checksums.txt [/code] All frames match perfectly with the original. Thank you very much, Pb PS: Now I'm *not* posting the full uncut console output, since I just wanted to close this thread with the information which commands were successful. I hope it's fine like that? From pb at das-werkstatt.com Tue Nov 22 13:21:57 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 13:21:57 +0100 Subject: [FFmpeg-user] parser not found for codec pcm_s16le, packets or times may be invalid. Message-ID: <4ECB93E5.8060907@das-werkstatt.com> Sorry for spamming the list, but I've encountered another issue which could indicate silent problems, I'd like to avoid in a production system: When transcoding a file with the current git-version of ffmpeg, I receive warnings that do not appear in earlier versions: [quote] [avi @ 0x96ad220] parser not found for codec pcm_s16le, packets or times may be invalid. [avi @ 0x96ad220] parser not found for codec ffv1, packets or times may be invalid. [/quote] Running the following command with the old version of ffmpeg (v0.5.2 from Debian Squeeze repositories): [code] ffmpeg -i input-ffv1.avi -an -vcodec ffv1 delme.avi [/code] Works fine, without warnings: [quote] FFmpeg version SVN-r0.5.2-4:0.5.2-6, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.2-6 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Oct 5 2010 08:33:07, gcc: 4.4.5 Input #0, avi, from 'input-ffv1.avi': Duration: 00:00:41.88, start: 0.000000, bitrate: 52834 kb/s Stream #0.0: Video: ffv1, yuv422p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Output #0, avi, to 'delme.avi': Stream #0.0: Video: ffv1, yuv422p, 720x576, q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 1047 fps= 17 q=0.0 Lsize= 269237kB time=41.88 bitrate=52664.5kbits/s video:269206kB audio:0kB global headers:0kB muxing overhead 0.011362% [/quote] However, running the same command with the current git-version (N-35077-g7876f14), contains warnings: [quote] ffmpeg version N-35077-g7876f14, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 22 2011 11:38:22 with gcc 4.4.5 configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-zlib --enable-libopenjpeg --enable-decoder=png --enable-encoder=png --enable-libdirac --enable-libschroedinger libavutil 51. 26. 0 / 51. 26. 0 libavcodec 53. 36. 0 / 53. 36. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [avi @ 0x96ad220] parser not found for codec pcm_s16le, packets or times may be invalid. [avi @ 0x96ad220] parser not found for codec ffv1, packets or times may be invalid. Input #0, avi, from 'input-ffv1.avi': Metadata: encoder : Lavf52.31.0 Duration: 00:00:41.88, start: 0.000000, bitrate: 52834 kb/s Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s [buffer @ 0x96c2520] w:720 h:576 pixfmt:yuv422p tb:1/1000000 sar:0/1 sws_param: Output #0, avi, to 'delme.avi': Metadata: ISFT : Lavf53.21.0 Stream #0:0: Video: ffv1 (FFV1 / 0x31564646), yuv422p, 720x576, q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (ffv1 -> ffv1) Press [q] to stop, [?] for help frame= 1047 fps= 17 q=0.0 Lsize= 269237kB time=00:00:41.88 bitrate=52664.5kbits/s video:269206kB audio:0kB global headers:0kB muxing overhead 0.011362% [/quote] The transcoding itself runs fine, in both cases. I've tried searching for these warnings on the web, but none of them referred to as where those warnings might come from. Thank you for any information, Pb From marc at hallmarcwebsites.com Tue Nov 22 15:46:35 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Tue, 22 Nov 2011 09:46:35 -0500 Subject: [FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality In-Reply-To: <28CD1CB9-CAFF-430A-AECE-1181D261DFF6@lesspain.de> References: <28CD1CB9-CAFF-430A-AECE-1181D261DFF6@lesspain.de> Message-ID: > The this is at least _very_ unusual and AFAIK the first time someone has > reported this to this list. My personal tests support what carl eugen and > countless other competent people have said on this list. So either you go to > the trouble of building a reproducible test case that you can share without an > NDA or you are probably on your own and you should at least stop being > rude to people who to the best of their (undisputed) knowledge are trying to > help you with a patience I don't think anyone could ask them to have. You > are a beginner and it's fine to ask questions but you attitude is like pissing on > the table of the family who has just invited you for a free lunch and then > insulting the cook without ever having cooked yourself in your entire life. > > Peace > And now for the final say, I never argued with anyone about what the current release of ffmpeg is supposed to do, what I have asked for through this list is to point me in the right direction, tell me why my audio is sounding so muddy when it shouldn't. That's it. A very simple and basic question and it took a Multimedia Engineer from another company to show me the missing flag from my command line!!! I'm done beating my head against this brick wall of a list. Maybe the people like Carl are just incapable of helping when it comes to the basics. Maybe they are only truly helpful if it is a complicated query. I don't know. What I do know is that I now have my answer that NO ONE HERE provided and that what I did get from this list was reprimands and hints that I should hand over private client files or get lost. The person who finally helped me didn't need the file. They looked into the #faac --long-help (no one mentioned this, either) and from there was able to see the flag that needed to be included. -aq 128. I know it doesn't need to be set quite this high but he prefers powers of 2. w/e This person not only provided the answer but gave me a decent explanation as well. Funny how he could do what the devs that actually wrote ffmpeg just simply didn't. "other competent people" While I have no doubt that the folks that frequent this list may be competent and I know that I have a good amount of respect for anyone that can work with compression algorithms it still bewilders me that I couldn't get the answer I was looking for. You can claim anything you wish, actions do the speaking. " but you attitude is like pissing on the table of the family who has just invited you for a free lunch and then insulting the cook without ever having cooked yourself in your entire life" - really? My frustration at not being able to get a meaningful answer from this list is fuel for an attack on me? Good grief... If any of you feel I was being rude, I apologize. Was never my intent. My intent has been to learn from this list not to make enemies. Now, for the knowledge dump: # faac --long-help Freeware Advanced Audio Coder FAAC 1.28 Usage: faac [options] infiles ... Quality-related options: -q Set default variable bitrate (VBR) quantizer quality in percent. (default: 100, averages at approx. 120 kbps VBR for a normal stereo input file with 16 bit and 44.1 kHz sample rate; max. value 500, min. 10). -b Set average bitrate (ABR) to approximately kbps. (max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised with a higher -c setting). -c Set the bandwidth in Hz (default: automatic, i.e. adapts maximum value to input sample rate). Input/output options: - : If you simply use a hyphen/minus sign instead of an input file name, FAAC can encode directly from stdin, thus enabling piping from other applications and utilities. The same works for stdout as well, so FAAC can pipe its output to other apps such as a server. -o X Set output file to X (only for one input file) only for one input file; you can use *.aac, *.mp4, *.m4a or *.m4b as file extension, and the file format will be set automatically to ADTS or MP4). -P Raw PCM input mode (default: off, i.e. expecting a WAV header; necessary for input files or bitstreams without a header; using only -P assumes the default values for -R, -B and -C in the input file). -R Raw PCM input sample rate in Hz (default: 44100 Hz, max. 96 kHz) -B Raw PCM input sample size (default: 16, also possible 8, 24, 32 bit fixed or float input). -C Raw PCM input channels (default: 2, max. 33 + 1 LFE). -X Raw PCM swap input bytes (default: bigendian). -I Input multichannel configuration (default: 3,4 which means Center is third and LFE is fourth like in 5.1 WAV, so you only have to specify a different position of these two mono channels in your multichannel input files if they haven't been reordered already). MP4 specific options: MP4 support unavailable. Expert options, only for testing purposes: --tns Enable coding of TNS, temporal noise shaping. --no-midside Don't use mid/side coding. --mpeg-vers X Force AAC MPEG version, X can be 2 or 4 --obj-type X AAC object type. (LC (Low Complexity, default), Main or LTP (Long Term Prediction) --shortctl X Enforce block type (0 = both (default); 1 = no short; 2 = no long). -r Generate raw AAC bitstream (i.e. without any headers). Not advised!!!, RAW AAC files are practically useless!!! Documentation: --license Show the FAAC license. --help Show this abbreviated help. --long-help Show complete help. More tips can be found in the audiocoding.com Knowledge Base at As for the -aq setting; FAAC is a variable bit rate codec so the 128 doesn't translate directly. The range is 0-100 with 100 reported as capable of getting 120kbps average. From cehoyos at ag.or.at Tue Nov 22 16:11:38 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 22 Nov 2011 15:11:38 +0000 (UTC) Subject: [FFmpeg-user] (no subject) References: <4ECB82B7.7080509@das-werkstatt.com> Message-ID: Peter B. das-werkstatt.com> writes: > I've been successfully compiling FFmpeg versions 0.5 and 0.6, as well as > the current git-version, with the following commandline: > > [code] > ./configure --prefix=/usr/local --enable-gpl --enable-nonfree > --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads > --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid > --enable-zlib --enable-libopenjpeg --enable-decoder=png > --enable-encoder=png --enable-avfilter-lavf --enable-libfaad && make > clean && make > [/code] Sorry, but what you write is completely unreproducible... (libfaad has been removed years ago, and configure does not accept unknown options.) Carl Eugen From pb at das-werkstatt.com Tue Nov 22 16:54:17 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 22 Nov 2011 16:54:17 +0100 Subject: [FFmpeg-user] (no subject) In-Reply-To: References: <4ECB82B7.7080509@das-werkstatt.com> Message-ID: <4ECBC5A9.1030309@das-werkstatt.com> Carl Eugen Hoyos wrote: > Peter B. das-werkstatt.com> writes: > > >> I've been successfully compiling FFmpeg versions 0.5 and 0.6, as well as >> the current git-version, with the following commandline: >> >> [code] >> ./configure --prefix=/usr/local --enable-gpl --enable-nonfree >> --enable-postproc --enable-swscale --enable-avfilter --enable-pthreads >> --enable-bzlib --enable-libmp3lame --enable-libvorbis --enable-libxvid >> --enable-zlib --enable-libopenjpeg --enable-decoder=png >> --enable-encoder=png --enable-avfilter-lavf --enable-libfaad && make >> clean && make >> [/code] >> > > Sorry, but what you write is completely unreproducible... > (libfaad has been removed years ago, and configure does not accept unknown > options.) > Damn! It's not my day today. After having tons of different versions of FFmpeg lying around on our conversion server, I've overlooked that my compile script is appending those parameters when building older versions. Absolutely my fault. Sorry, sorry, sorry... Too many screens, too many shells, too much coffee... Next time I promise to triple-check before I post. Promise! :) Thanks and sorry again, Pb From jshupert at pps-inc.com Tue Nov 22 16:54:44 2011 From: jshupert at pps-inc.com (Jim Shupert) Date: Tue, 22 Nov 2011 10:54:44 -0500 Subject: [FFmpeg-user] to crop the bottom In-Reply-To: <20111117234429.GA7035@arborea> References: <4EC595D9.1020901@pps-inc.com> <20111117234429.GA7035@arborea> Message-ID: <4ECBC5C4.6010202@pps-inc.com> > filtering is disabled with -vcodec copy, we should really make ffmpeg > more verbose about this. Much thanks to all - of course my difficulties all stemmed from the -vcodec copy my goal was to only trim a few lines from the bottom without reencoding i suppose that cannot be done -- exactly From jshupert at pps-inc.com Tue Nov 22 17:40:22 2011 From: jshupert at pps-inc.com (Jim Shupert) Date: Tue, 22 Nov 2011 11:40:22 -0500 Subject: [FFmpeg-user] means to inverse telecine Message-ID: <4ECBD076.8020704@pps-inc.com> I have looked and i do not think that there is a video filter for inverse telecine ( to remove 3:2 pull down from 29.97 vid to 24 fps [ film org ] ) But i thought i would ask is there a vf filter to do this ? thanks j From daverice at mac.com Tue Nov 22 18:23:19 2011 From: daverice at mac.com (Dave Rice) Date: Tue, 22 Nov 2011 12:23:19 -0500 Subject: [FFmpeg-user] means to inverse telecine In-Reply-To: <4ECBD076.8020704@pps-inc.com> References: <4ECBD076.8020704@pps-inc.com> Message-ID: <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> On Nov 22, 2011, at 11:40 AM, Jim Shupert wrote: > I have looked and i do not think that there is a video filter for inverse telecine > ( to remove 3:2 pull down from 29.97 vid to 24 fps [ film org ] ) > > But i thought i would ask > > is there a vf filter to do this ? http://ffmpeg.org/libavfilter.html#mp lists a few of them (i.e. -vf mp=pullup,softskip), but I don't think they work. For inverse telecine I use mplayer|mencoder. Dave Rice > thanks > > j > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jshupert at pps-inc.com Tue Nov 22 20:01:31 2011 From: jshupert at pps-inc.com (Jim Shupert) Date: Tue, 22 Nov 2011 14:01:31 -0500 Subject: [FFmpeg-user] means to inverse telecine In-Reply-To: <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> References: <4ECBD076.8020704@pps-inc.com> <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> Message-ID: <4ECBF18B.9010306@pps-inc.com> >> is there a vf filter to do this ? > http://ffmpeg.org/libavfilter.html#mp lists a few of them (i.e. -vf mp=pullup,softskip), but I don't think they work. For inverse telecine I use mplayer|mencoder. yes i have done the old -vf pullup,softskip but was hoping/thinking/wishing that such capability find itz way into the FFworld From ubitux at gmail.com Tue Nov 22 21:30:44 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 22 Nov 2011 21:30:44 +0100 Subject: [FFmpeg-user] Using the "select" filter In-Reply-To: <4EC63C24.3080400@bbc.co.uk> References: <4EC4EDD4.5000102@bbc.co.uk> <20111117190940.GC14546@leki> <20111117191046.GD14546@leki> <4EC63C24.3080400@bbc.co.uk> Message-ID: <20111122203044.GL29124@leki> On Fri, Nov 18, 2011 at 11:06:12AM +0000, Tim Nicholson wrote: > On 17/11/11 19:10, Cl?ment B?sch wrote: > >On Thu, Nov 17, 2011 at 08:09:40PM +0100, Cl?ment B?sch wrote: > >>On Thu, Nov 17, 2011 at 11:19:48AM +0000, Tim Nicholson wrote: > >>>The examples of using the select filter given at:- > >>> > >>>http://ffmpeg.org/ffmpeg.html#select > >>> > >>>suggest, for example, > >>> > >>># select only I-frames > >>>select='eq(pict_type\,I)' > >>> > >>> > >>>However when I tried this I got an error. > >>> > >>>ffmpeg -i in.mp4 -vf "select='eq(pict_type\,I)', showinfo" -f mp4 -y > >>>/dev/null > >>>[..] > >>>Missing ')' or too many args in 'eq(pict_type\,I)' > >>> > >>> > >>>Removing the "\" which I assume is there to "escape" the comma > >>>solved the problem, thus:- > >>> > >>>ffmpeg -i in.mp4 -vf "select='eq(pict_type,I)', showinfo" -f mp4 -y > >>>/dev/null > >>> > >>>This looks like a small error in the docs. > >>> > >> > >>You are not supposed to add quotes around the select. > > The quotes were around the whole filtergraph, not just the select. > > >> > >> ffmpeg -i in.mp4 -vf select='eq(pict_type,I)',showinfo -f ... > >> > > > >Erh. I meant: > > ffmpeg -i in.mp4 -vf select='eq(pict_type,\I)',showinfo -f ... > > Actually you meant:- > > ffmpeg -i in.mp4 -vf select='eq(pict_type\,I)', showinfo -f ... > Yes, without the space before showinfo since there is no surrounding quotes. > Why is select different to any other filter? > The filter isn't, the documentation might be though. > Looking through the list of examples provided in:- > > http://ffmpeg.org/ffmpeg.html#Video-Filters > > shows the following:- > > 23.11 fieldorder > ffmpeg -i in.vob -vf "fieldorder=bff" out.dv > > 23.16 hflip > ffmpeg -i in.avi -vf "hflip" out.avi > > 23.34 slicify > ffmpeg -i in.avi -vf "slicify=32" out.avi > > 23.37 unsharp > ffmpeg -i in.avi -vf "unsharp" out.mp4 > > 23.38 vflip > ffmpeg -i in.avi -vf "vflip" out.avi > > From which one might reasonably deduce that is is normal, and > recommended to enclose the whole filtergraph within " ". It > certainly aids clarity in seeing what elements on the command line > are part of the whole filter arrangement, and I think its more > readable when you don't end up with \ escapes littering a filter > that already has ample use of other punctuation characters such as = > , : it reminds me a bit of perl. ;P > The documentation is a bit inconsistent I agree, but I think the examples should have the '\' in it: it's more obvious when the failure occurs on the '\' instead of the comma: "why does it fails on the comma? I don't see why I need to escape it...", while in case of the an extra escape the most common thought is something like "herp derp yet another shell escaping issue i need to double escape, remove escape, ..." which is more likely to lead to a solution. Also note the shell escaping varies between shells (I know tcsh has for instance a totally insane way of escaping). I think examples with the full command line should include the " ", or at least a command that obviously work out of the box. On the other hand, if it's just for the -vf part, I don't think the surrounding quotes should appear: it is obvious you have to send that part of the command line "verbatim" to FFmpeg, so deal with your shell. Of course the documentation could be improved, but I don't feel like doing it :) If you want to discuss such topic, you are welcome on ffmpeg-devel. PS: sorry for the late reply. -- 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 ubitux at gmail.com Tue Nov 22 21:47:47 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Tue, 22 Nov 2011 21:47:47 +0100 Subject: [FFmpeg-user] How do I add a new audio stream? In-Reply-To: <1321828734389-4089839.post@n4.nabble.com> References: <1321824113329-4089600.post@n4.nabble.com> <1321826731491-4089741.post@n4.nabble.com> <1321828734389-4089839.post@n4.nabble.com> Message-ID: <20111122204747.GN29124@leki> On Sun, Nov 20, 2011 at 02:38:54PM -0800, Tuuls wrote: > Another question! > How different channels of a stereo, expanded in different streams ? > You mean something like this?: ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg I'll add a similar example to the documentation soon. [...] -- 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 koxaniy at mail.ru Tue Nov 22 22:00:20 2011 From: koxaniy at mail.ru (Tuuls) Date: Tue, 22 Nov 2011 13:00:20 -0800 (PST) Subject: [FFmpeg-user] How do I add a new audio stream? In-Reply-To: <20111122204747.GN29124@leki> References: <1321824113329-4089600.post@n4.nabble.com> <1321826731491-4089741.post@n4.nabble.com> <1321828734389-4089839.post@n4.nabble.com> <20111122204747.GN29124@leki> Message-ID: <1321995620847-4097328.post@n4.nabble.com> Very very big thank you for this! This is very useful! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-do-I-add-a-new-audio-stream-tp4089600p4097328.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From daad at libero.it Tue Nov 22 22:13:21 2011 From: daad at libero.it (daad at libero.it) Date: Tue, 22 Nov 2011 22:13:21 +0100 (CET) Subject: [FFmpeg-user] Use ffmpeg to watermark and scale an image on video Message-ID: <6241778.3292761321996401137.JavaMail.root@wmail74> I want to be able to watermark videos with a logo image, which contains a website url. The videos can be of different formats and dimension. I'm trying to figure out a generic ffmpeg command to achieve it, so that i don't have to tweak the command depending on the video i have to process. So far i got: ffmpeg -i sample.mov -sameq -acodec copy -vf 'movie=logo.png [watermark]; [in] [watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' sample2.mov In this way though the logo will look too big or too small with video of different size. I've seen there is a scale option for avfilter but that will scale the video, not the watermarked image (i guess?), but I haven't figure out whether it's possible to resize the image logo based on the dimension of the input video, so that I can say to scale the logo width to 1/3 of the video width for example, and keep the image ratio. Any idea? doesn't need to be done in a single command, could even be a script. thanks in advance. From alex.zhen.ma at gmail.com Tue Nov 22 22:57:21 2011 From: alex.zhen.ma at gmail.com (Zhen Ma) Date: Tue, 22 Nov 2011 16:57:21 -0500 Subject: [FFmpeg-user] Use ffmpeg to watermark and scale an image on video In-Reply-To: <6241778.3292761321996401137.JavaMail.root@wmail74> References: <6241778.3292761321996401137.JavaMail.root@wmail74> Message-ID: It should work with this way. ffmpeg -i sample.mov -sameq -acodec copy -vf 'movie=logo.png,*scale=60:30* [watermark]; [in] [watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' sample2.mov On Tue, Nov 22, 2011 at 4:13 PM, daad at libero.it wrote: > I want to be able to watermark videos with a logo image, which contains a > website url. The videos can be of different formats and dimension. I'm > trying > to figure out a generic ffmpeg command to achieve it, so that i don't have > to > tweak the command depending on the video i have to process. So far i got: > > ffmpeg -i sample.mov -sameq -acodec copy -vf 'movie=logo.png [watermark]; > [in] > [watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' > sample2.mov > > In this way though the logo will look too big or too small with video of > different size. I've seen there is a scale option for avfilter but that > will > scale the video, not the watermarked image (i guess?), but I haven't > figure out > whether it's possible to resize the image logo based on the dimension of > the > input video, so that I can say to scale the logo width to 1/3 of the video > width for example, and keep the image ratio. > > Any idea? doesn't need to be done in a single command, could even be a > script. > thanks in advance. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Your sincerely ! ------------------------------------------------------- Alex Zhen Ma ? Email: alex.zhen.ma at gmail.com From daad at libero.it Tue Nov 22 23:18:51 2011 From: daad at libero.it (daad at libero.it) Date: Tue, 22 Nov 2011 23:18:51 +0100 (CET) Subject: [FFmpeg-user] R: Re: Use ffmpeg to watermark and scale an image on video Message-ID: <11116834.3313731322000331967.JavaMail.root@wmail74> doesn't it scale the video instead? i need the image to be scaled. in the meantime i came up with the following script: #!/bin/bash VIDEO=$1 LOGO=$2 VIDEO_WATERMARKED=w_${VIDEO} VIDEO_WIDTH=`ffprobe -show_streams $VIDEO 2>&1 | grep ^width | sed s/width=//` echo The video width is $VIDEO_WIDTH cp $LOGO logo.png IMAGE_WIDTH=$((VIDEO_WIDTH/3)) echo The image width will be $IMAGE_WIDTH mogrify -resize $IMAGE_WIDTH logo.png echo logo.png resized echo Starting watermarking ffmpeg -i $VIDEO -sameq -acodec copy -vf 'movie=logo.png [watermark]; [in] [watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' $VIDEO_WATERMARKED echo Video watermarked The last thing i'm not sure about is, why the watermarked video is smaller in size compared to the original. I thought that using "-sameq" would just keep the original video quality. If i specify "-vcodec copy" then the watermark bit is not applied, hence I thought to use "-sameq". INPUT Duration: 00:01:25.53, start: 0.000000, bitrate: 307 kb/s Stream #0:0(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 261 kb/s, 10 fps, 10 tbr, 3k tbn, 25 tbc OUTPUT encoder : Lavf53.20.0 Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1: 1 DAR 4:3], q=-1--1, 10 tbn, 10 tbc whereas the audio info information are identical. any advice on how to use the same video quality as well? thanks >----Messaggio originale---- >Da: alex.zhen.ma at gmail.com >Data: 22/11/2011 22.57 >A: "FFmpeg user questions and RTFMs" >Ogg: Re: [FFmpeg-user] Use ffmpeg to watermark and scale an image on video > >It should work with this way. > >ffmpeg -i sample.mov -sameq -acodec copy -vf >'movie=logo.png,*scale=60:30* [watermark]; >[in] >[watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' >sample2.mov > >On Tue, Nov 22, 2011 at 4:13 PM, daad at libero.it wrote: > >> I want to be able to watermark videos with a logo image, which contains a >> website url. The videos can be of different formats and dimension. I'm >> trying >> to figure out a generic ffmpeg command to achieve it, so that i don't have >> to >> tweak the command depending on the video i have to process. So far i got: >> >> ffmpeg -i sample.mov -sameq -acodec copy -vf 'movie=logo.png [watermark]; >> [in] >> [watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' >> sample2.mov >> >> In this way though the logo will look too big or too small with video of >> different size. I've seen there is a scale option for avfilter but that >> will >> scale the video, not the watermarked image (i guess?), but I haven't >> figure out >> whether it's possible to resize the image logo based on the dimension of >> the >> input video, so that I can say to scale the logo width to 1/3 of the video >> width for example, and keep the image ratio. >> >> Any idea? doesn't need to be done in a single command, could even be a >> script. >> thanks in advance. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > > >-- >Your sincerely ! >------------------------------------------------------- >Alex Zhen Ma >? Email: alex.zhen.ma at gmail.com >_______________________________________________ >ffmpeg-user mailing list >ffmpeg-user at ffmpeg.org >http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Wed Nov 23 01:50:31 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 23 Nov 2011 00:50:31 +0000 (UTC) Subject: [FFmpeg-user] means to inverse telecine References: <4ECBD076.8020704@pps-inc.com> <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> <4ECBF18B.9010306@pps-inc.com> Message-ID: Jim Shupert pps-inc.com> writes: > >> is there a vf filter to do this ? > > http://ffmpeg.org/libavfilter.html#mp lists a few of them (i.e. > > -vf mp=pullup,softskip), but I don't think they work. For inverse telecine I > > use mplayer|mencoder. > > yes i have done the old -vf pullup,softskip > > but was hoping/thinking/wishing that such capability find itz way into the > FFworld A reproducible bug-report that shows that -vf mp=pullup is not working would be very welcome! Carl Eugen From tim.nicholson at bbc.co.uk Wed Nov 23 11:19:15 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Wed, 23 Nov 2011 10:19:15 +0000 Subject: [FFmpeg-user] Using the "select" filter In-Reply-To: <20111122203044.GL29124@leki> References: <4EC4EDD4.5000102@bbc.co.uk> <20111117190940.GC14546@leki> <20111117191046.GD14546@leki> <4EC63C24.3080400@bbc.co.uk> <20111122203044.GL29124@leki> Message-ID: <4ECCC8A3.1090504@bbc.co.uk> On 22/11/11 20:30, Cl?ment B?sch wrote: > On Fri, Nov 18, 2011 at 11:06:12AM +0000, Tim Nicholson wrote: >> On 17/11/11 19:10, Cl?ment B?sch wrote: >>> On Thu, Nov 17, 2011 at 08:09:40PM +0100, Cl?ment B?sch wrote: [..] >>> Erh. I meant: >>> ffmpeg -i in.mp4 -vf select='eq(pict_type,\I)',showinfo -f ... >> >> Actually you meant:- >> >> ffmpeg -i in.mp4 -vf select='eq(pict_type\,I)', showinfo -f ... >> > > Yes, without the space before showinfo since there is no surrounding > quotes. > AGGGH, its so much easier with quotes.... >>[..] >> > > The documentation is a bit inconsistent I agree, but I think the examples > should have the '\' in it: it's more obvious when the failure occurs on > the '\' instead of the comma: "why does it fails on the comma? I don't see > why I need to escape it...", while in case of the an extra escape the most > common thought is something like "herp derp yet another shell escaping > issue i need to double escape, remove escape, ..." which is more likely to > lead to a solution. > I think it doesn't help that all examples that give the full command line and include quotes around the filtergraph are simple syntax ones where the quotes aren't actually needed... A more complex example shown in both forms would probably help. Also some more complex examples showing the development of a graph from a simple filter to chains, to chains with branches would, I think, make the whole thing more accessible to newcomers. Filter syntax questions do seem to come up quite often. > Also note the shell escaping varies between shells (I know tcsh has for > instance a totally insane way of escaping). > > I think examples with the full command line should include the " ", or at > least a command that obviously work out of the box. > > On the other hand, if it's just for the -vf part, I don't think the > surrounding quotes should appear: it is obvious you have to send that part > of the command line "verbatim" to FFmpeg, so deal with your shell. I think I agree with both of those > > Of course the documentation could be improved, but I don't feel like doing > it :) > :P > If you want to discuss such topic, you are welcome on ffmpeg-devel. > When I have some space I might propose something there, and even submit a patch. I have a growing "cheat sheet" of examples I use myself. > PS: sorry for the late reply. > no problems -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From daverice at mac.com Wed Nov 23 13:24:23 2011 From: daverice at mac.com (David Rice) Date: Wed, 23 Nov 2011 07:24:23 -0500 Subject: [FFmpeg-user] means to inverse telecine In-Reply-To: References: <4ECBD076.8020704@pps-inc.com> <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> <4ECBF18B.9010306@pps-inc.com> Message-ID: On Nov 22, 2011, at 7:50 PM, Carl Eugen Hoyos wrote: > Jim Shupert pps-inc.com> writes: > >>>> is there a vf filter to do this ? >>> http://ffmpeg.org/libavfilter.html#mp lists a few of them (i.e. >>> -vf mp=pullup,softskip), but I don't think they work. For inverse telecine I >>> use mplayer|mencoder. >> >> yes i have done the old -vf pullup,softskip >> >> but was hoping/thinking/wishing that such capability find itz way into the >> FFworld > > A reproducible bug-report that shows that -vf mp=pullup is not working would be > very welcome! I started one here https://ffmpeg.org/trac/ffmpeg/ticket/681. As Jim says, it would be great to have this in ffmpeg rather than having to pipe back and forth to mencoder. Dave Rice > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ib at wupperonline.de Wed Nov 23 13:25:34 2011 From: ib at wupperonline.de (=?ISO-8859-1?Q?Ingo=20Br=FCckl?=) Date: Wed, 23 Nov 2011 13:25:34 +0100 Subject: [FFmpeg-user] mp3 sound bug In-Reply-To: <4ec8d4e3.2056206a.bm000@wupperonline.de> Message-ID: <4ecce694.24da6db5.bm000@wupperonline.de> I wrote on Sun, 20 Nov 2011 11:20:26 +0100: > Commit 22e25c002e103e52ace35703423e896b08b51aef introduced some bug that > makes ffmp3float produce no sound but only occasional crackles. MPlayer > selects this codec by default for me and suddenly has sound issues. In case somebody is interested, Vitor (the author) found the bug. Apparently there were SSE2 instructions where they should not be. Ingo From peace at AleksandrSolzhenitsyn.net Wed Nov 23 14:41:27 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Wed, 23 Nov 2011 08:41:27 -0500 Subject: [FFmpeg-user] Covnert for iPod Message-ID: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I wrote before but didn't get any replies. Here's the situation; My hard drive wrecked and was unrecoverable. The machine is running XP. I used CopyTrans to successfully (believe it or not) to move ALL of my iPod's music and videos over to the new hard drive with a new install on iTunes. For whatever reason my iPod no longer plays certain videos completely or can't be transferred from iTunes to the iPod. But, the same videos will play perfectly when played from the hard drive. The same thing happens when I download a video from Youtube (360- mp4) and try to put it on the iPod- it plays locally but won't transfer to the iPod (iTunes says the format isn't compatible or some such nonsense). Another problem is that the videos on the iPod will play part way through and then will appear in fast motion until the end of the video. I did have a backup of all the music and videos on a separate drive and have tried using those videos in case there was some corruption which occurred during the transfer from the iPod back to the new hard drive. The same thing happens as stated above. Does anyone have a solution to this problem? Is the following assumption correct? 1. When a video is downloaded from Youtube in mp4 format that video has a specific audio format, bit rate, aspect ratio which unless converted to a format that iPod's will "read" it can't be played successfully on an iPod's. If the assumption is correct, what is a FFMPEG code line to convert the video so it'll play on an iPod? Which do you use? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJOzPgCAAoJEPBpZNn4grcjduYIAJdDlZ91aZM03jLHSveIjKnr 53RKSV1lo5a2cShIAeiMGO02foictavEtdhOz3Nx5ft55CEcfGCCNttQtmdOe3+q GhoR22dtPy0bJW+iMTyDnfTuUM2JLi4aRrUhHtnpvDVfVcgXTxidLchv0lj/GWBc z2xY0O0uWsNp4HyW8X/Rn7kzIrQUZxvKD1yjHo3Qo9ssbbi23+/OC5pI9fjk+afc RjhjUiAdPXNQBVfeDDSjIDPfvlSkB/4R4oNVweTOuDtBbjQzl1LukbTyXWXLka/2 dLAzKz+SA+qDKqXC6aGNW9yT7e3rGEu5liS0sCfHTrgxAowdBPjVxExxdE0y2zk= =zE1O -----END PGP SIGNATURE----- From jeisom at gmail.com Wed Nov 23 16:31:09 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Wed, 23 Nov 2011 09:31:09 -0600 Subject: [FFmpeg-user] Covnert for iPod In-Reply-To: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> References: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> Message-ID: On Wed, Nov 23, 2011 at 7:41 AM, . wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I wrote before but didn't get any replies. > > Here's the situation; > > My hard drive wrecked and was unrecoverable. > > The machine is running XP. > > I used CopyTrans to successfully (believe it or not) to move ALL of my > iPod's music and videos over to the new hard drive with a new install on > iTunes. > > For whatever reason my iPod no longer plays certain videos completely or > can't be transferred from iTunes to the iPod. ?But, the same videos will > play perfectly when played from the hard drive. ?The same thing happens > when I download a video from Youtube (360- mp4) and try to put it on the > iPod- it plays locally but won't transfer to the iPod (iTunes says the > format isn't compatible or some such nonsense). ?Another problem is that > the videos on the iPod will play part way through and then will appear > in fast motion until the end of the video. > > I did have a backup of all the music and videos on a separate drive and > have tried using those videos in case there was some corruption which > occurred during the transfer from the iPod back to the new hard drive. > The same thing happens as stated above. > > Does anyone have a solution to this problem? > > Is the following assumption correct? > > 1. ?When a video is downloaded from Youtube in mp4 format that video has > a specific audio format, bit rate, aspect ratio which unless converted > to a format that iPod's will "read" it can't be played successfully on > an iPod's. > > > If the assumption is correct, what is a FFMPEG code line to convert the > video so it'll play on an iPod? ?Which do you use? I use the following for mine. You may be able, depending on generation/type of ipod, to raise the resolution. 1500000 is the max bitrate for video on early ipod touches. Adjust to to your needs or wants. ffmpeg -i -y -vcodec libx264 -vprofile baseline -preset medium -level 30 \ -b:v 1500000 -acodec libfaac -ac 2 -ab 128000 -f mp4 -s 640x480 HTH Jonathan > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.10 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iQEcBAEBAgAGBQJOzPgCAAoJEPBpZNn4grcjduYIAJdDlZ91aZM03jLHSveIjKnr > 53RKSV1lo5a2cShIAeiMGO02foictavEtdhOz3Nx5ft55CEcfGCCNttQtmdOe3+q > GhoR22dtPy0bJW+iMTyDnfTuUM2JLi4aRrUhHtnpvDVfVcgXTxidLchv0lj/GWBc > z2xY0O0uWsNp4HyW8X/Rn7kzIrQUZxvKD1yjHo3Qo9ssbbi23+/OC5pI9fjk+afc > RjhjUiAdPXNQBVfeDDSjIDPfvlSkB/4R4oNVweTOuDtBbjQzl1LukbTyXWXLka/2 > dLAzKz+SA+qDKqXC6aGNW9yT7e3rGEu5liS0sCfHTrgxAowdBPjVxExxdE0y2zk= > =zE1O > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jeisom at gmail.com Wed Nov 23 16:31:09 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Wed, 23 Nov 2011 09:31:09 -0600 Subject: [FFmpeg-user] Covnert for iPod In-Reply-To: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> References: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> Message-ID: On Wed, Nov 23, 2011 at 7:41 AM, . wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I wrote before but didn't get any replies. > > Here's the situation; > > My hard drive wrecked and was unrecoverable. > > The machine is running XP. > > I used CopyTrans to successfully (believe it or not) to move ALL of my > iPod's music and videos over to the new hard drive with a new install on > iTunes. > > For whatever reason my iPod no longer plays certain videos completely or > can't be transferred from iTunes to the iPod. ?But, the same videos will > play perfectly when played from the hard drive. ?The same thing happens > when I download a video from Youtube (360- mp4) and try to put it on the > iPod- it plays locally but won't transfer to the iPod (iTunes says the > format isn't compatible or some such nonsense). ?Another problem is that > the videos on the iPod will play part way through and then will appear > in fast motion until the end of the video. > > I did have a backup of all the music and videos on a separate drive and > have tried using those videos in case there was some corruption which > occurred during the transfer from the iPod back to the new hard drive. > The same thing happens as stated above. > > Does anyone have a solution to this problem? > > Is the following assumption correct? > > 1. ?When a video is downloaded from Youtube in mp4 format that video has > a specific audio format, bit rate, aspect ratio which unless converted > to a format that iPod's will "read" it can't be played successfully on > an iPod's. > > > If the assumption is correct, what is a FFMPEG code line to convert the > video so it'll play on an iPod? ?Which do you use? I use the following for mine. You may be able, depending on generation/type of ipod, to raise the resolution. 1500000 is the max bitrate for video on early ipod touches. Adjust to to your needs or wants. ffmpeg -i -y -vcodec libx264 -vprofile baseline -preset medium -level 30 \ -b:v 1500000 -acodec libfaac -ac 2 -ab 128000 -f mp4 -s 640x480 HTH Jonathan > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.10 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iQEcBAEBAgAGBQJOzPgCAAoJEPBpZNn4grcjduYIAJdDlZ91aZM03jLHSveIjKnr > 53RKSV1lo5a2cShIAeiMGO02foictavEtdhOz3Nx5ft55CEcfGCCNttQtmdOe3+q > GhoR22dtPy0bJW+iMTyDnfTuUM2JLi4aRrUhHtnpvDVfVcgXTxidLchv0lj/GWBc > z2xY0O0uWsNp4HyW8X/Rn7kzIrQUZxvKD1yjHo3Qo9ssbbi23+/OC5pI9fjk+afc > RjhjUiAdPXNQBVfeDDSjIDPfvlSkB/4R4oNVweTOuDtBbjQzl1LukbTyXWXLka/2 > dLAzKz+SA+qDKqXC6aGNW9yT7e3rGEu5liS0sCfHTrgxAowdBPjVxExxdE0y2zk= > =zE1O > -----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 Wed Nov 23 17:19:29 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Wed, 23 Nov 2011 17:19:29 +0100 Subject: [FFmpeg-user] parser not found for codec pcm_s16le, packets or times may be invalid. In-Reply-To: <4ECB93E5.8060907@das-werkstatt.com> References: <4ECB93E5.8060907@das-werkstatt.com> Message-ID: <20111123161929.GA26105@phare.normalesup.org> Le duodi 2 frimaire, an CCXX, Peter B. a ?crit?: > When transcoding a file with the current git-version of ffmpeg, I > receive warnings that do not appear in earlier versions: > > [avi @ 0x96ad220] parser not found for codec pcm_s16le, packets or times > may be invalid. > [avi @ 0x96ad220] parser not found for codec ffv1, packets or times may > be invalid. > The transcoding itself runs fine, in both cases. The warning says that the packets or time _may_ be invalid, not that they actually will be. It seems that your setting does not actually need the information found by the parser. For PCM, that could be expected. For ffv1, I guess there are no B-frames. This warning seemed like a good idea at the time, but now I realize that there are lots of codecs that work fine without a parser even if the format thinks it needs one. I will think about it some more. In the meantime, just ignore the warning: if it works, then it means it was harmless. 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 peace at AleksandrSolzhenitsyn.net Wed Nov 23 17:59:49 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Wed, 23 Nov 2011 11:59:49 -0500 Subject: [FFmpeg-user] Convert for iPod In-Reply-To: References: <4ECCF807.7060309@AleksandrSolzhenitsyn.net> Message-ID: <4ECD2685.5070304@AleksandrSolzhenitsyn.net> A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 554 bytes Desc: OpenPGP digital signature URL: From tjg at soe.ucsc.edu Wed Nov 23 18:19:59 2011 From: tjg at soe.ucsc.edu (Tim Gustafson) Date: Wed, 23 Nov 2011 09:19:59 -0800 (PST) Subject: [FFmpeg-user] Problems with IE9 In-Reply-To: <248456329.95898.1322068770837.JavaMail.root@mail-01.cse.ucsc.edu> Message-ID: <2102774300.95902.1322068799130.JavaMail.root@mail-01.cse.ucsc.edu> Hi, I've got a video service that's been up and running for a while that uses FFMPEG to convert all its videos. It works very well, except in IE9 (of course). Here's a link to the video player page: http://slugtube.soe.ucsc.edu/slugtube/video/1321566272 I default to using HTML5 video (with MP4 and WEBM versions of each file), if available, and then fall back to a flash player if that doesn't work. Here are the command lines that I use to create the MP4 video: ffmpeg -y -i source.video -f mp4 -threads 4 -vcodec libx264 -vpre libx264-lossless_slow -b 768K -acodec libfaac -ar 22050 -r 30 -g 150 -s 800x600 ideo.mp4 It would appear that IE9 is attempting to use the MP4 video, but it chokes on it. The player controls show up for a minute and then go away and don't come back, and even if you right-click the video and hit "play", it does not play. Is there some better options that I should be choosing for MP4 video to work properly in IE9? -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Tim Gustafson tjg at soe.ucsc.edu Baskin School of Engineering 831-459-5354 UC Santa Cruz Baskin Engineering 317B -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- From lou at lrcd.com Wed Nov 23 18:42:08 2011 From: lou at lrcd.com (Lou) Date: Wed, 23 Nov 2011 08:42:08 -0900 Subject: [FFmpeg-user] Problems with IE9 In-Reply-To: <2102774300.95902.1322068799130.JavaMail.root@mail-01.cse.ucsc.edu> References: <248456329.95898.1322068770837.JavaMail.root@mail-01.cse.ucsc.edu> <2102774300.95902.1322068799130.JavaMail.root@mail-01.cse.ucsc.edu> Message-ID: <20111123084208.585644f8@lrcd.com> On Wed, 23 Nov 2011 09:19:59 -0800 (PST) Tim Gustafson wrote: > Hi, > > I've got a video service that's been up and running for a while that > uses FFMPEG to convert all its videos. It works very well, except in > IE9 (of course). > > Here's a link to the video player page: > > http://slugtube.soe.ucsc.edu/slugtube/video/1321566272 > > I default to using HTML5 video (with MP4 and WEBM versions of each > file), if available, and then fall back to a flash player if that > doesn't work. Here are the command lines that I use to create the > MP4 video: > > ffmpeg -y -i source.video -f mp4 -threads 4 -vcodec libx264 -vpre > libx264-lossless_slow -b 768K -acodec libfaac -ar 22050 -r 30 -g 150 > -s 800x600 ideo.mp4 Are you absolutely sure you want or need lossless? I'm assuming you probably want to use the "slow" preset instead since you are attempting to apply the -b option which is probably ignored with lossless presets. > It would appear that IE9 is attempting to use the MP4 video, but it > chokes on it. The player controls show up for a minute and then go > away and don't come back, and even if you right-click the video and > hit "play", it does not play. Is there some better options that I > should be choosing for MP4 video to work properly in IE9? > > -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- > Tim Gustafson > tjg at soe.ucsc.edu Baskin School of > Engineering 831-459-5354 UC Santa > Cruz Baskin Engineering 317B > -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- From h.reindl at thelounge.net Wed Nov 23 18:54:51 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 23 Nov 2011 18:54:51 +0100 Subject: [FFmpeg-user] Problems with IE9 In-Reply-To: <2102774300.95902.1322068799130.JavaMail.root@mail-01.cse.ucsc.edu> References: <2102774300.95902.1322068799130.JavaMail.root@mail-01.cse.ucsc.edu> Message-ID: <4ECD336B.5020104@thelounge.net> Am 23.11.2011 18:19, schrieb Tim Gustafson: > Hi, > > I've got a video service that's been up and running for a while that uses FFMPEG to convert all its videos. It works very well, except in IE9 (of course). > > Here's a link to the video player page: > > http://slugtube.soe.ucsc.edu/slugtube/video/1321566272 > > I default to using HTML5 video (with MP4 and WEBM versions of each file), if available, and then fall back to a flash player if that doesn't work. Here are the command lines that I use to create the MP4 video: > ffmpeg -y -i source.video -f mp4 -threads 4 -vcodec libx264 -vpre libx264-lossless_slow -b 768K -acodec libfaac -ar 22050 -r 30 -g 150 -s 800x600 ideo.mp4 > It would appear that IE9 is attempting to use the MP4 video, but it chokes on it. The player controls show up for a minute and then go away and don't come back, and even if you right-click the video and hit "play", it does not play. Is there some better options that I should be choosing for MP4 video to work properly in IE9? most clients needing H264 BASELINE for HTML5 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From pb at das-werkstatt.com Wed Nov 23 19:21:57 2011 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 23 Nov 2011 19:21:57 +0100 Subject: [FFmpeg-user] parser not found for codec pcm_s16le, packets or times may be invalid. In-Reply-To: <20111123161929.GA26105@phare.normalesup.org> References: <4ECB93E5.8060907@das-werkstatt.com> <20111123161929.GA26105@phare.normalesup.org> Message-ID: <4ECD39C5.6090404@das-werkstatt.com> Nicolas George wrote: > The warning says that the packets or time _may_ be invalid, not that they > actually will be. > If packets or timing "may" be invalid, is there any way for me to check this? > It seems that your setting does not actually need the information found by > the parser. For PCM, that could be expected. For ffv1, I guess there are no > B-frames. > FFv1 is I-frames only. > This warning seemed like a good idea at the time, but now I realize that > there are lots of codecs that work fine without a parser even if the format > thinks it needs one. I will think about it some more. > What is this parser checking for? > In the meantime, just ignore the warning: if it works, then it means it was > harmless. > But you know how it is: If you're used to ignoring warnings, you'll ignore other warnings in the future, too... ;) Wouldn't that defeat the purpose of the warnings in the first place? Thanks for the information, Pb From mediastream at gmail.com Wed Nov 23 19:32:33 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 23 Nov 2011 13:32:33 -0500 Subject: [FFmpeg-user] FFplay monitoring gig. Message-ID: Can someone build this for me? I'm ok with publishing it under GPL. Let me know the time for and cost of work. Payment per module, upon completion. *in brackets are (required API/config parameters) 1. Arrange multiple FFplay players side by side on a screen in xWindows. (window position) 2. While streaming, each player can generate a screenshot with a timestamp overlay. (encoding parameters/time interval/time stamp format/font/size/color) 3. Entire layout encoded live. (encoding parameters) 4. Audio meter overlay on top of video e.g. Green/yellow/red VU meter. (API for VU meter template) 5. Freeze frame detection. (tolerance to % change and duration). 6. Silence/"too loud" detection (tolerance to db levels and duration). 7. Monitor network traffic, and if session traffic drops to 0 kbps Tx/Rx, then tear down and attempt to re-establish the session. (number of tries) 8. If session is terminated - attempt to re-establish the session. (number of tries) 9. Cycle multiple players in the same xWindows position. (layout config) 10. Allow monitoring only for freeze frame/silence/RxTx without displaying. (save events to log, notify on err) 11. Change configuration without restart. (config file) 12. Notify on err with a flashing red frame over the video / email / log. Thanks. Dennis. From tjg at soe.ucsc.edu Wed Nov 23 19:37:13 2011 From: tjg at soe.ucsc.edu (Tim Gustafson) Date: Wed, 23 Nov 2011 10:37:13 -0800 (PST) Subject: [FFmpeg-user] Problems with IE9 In-Reply-To: <4ECD336B.5020104@thelounge.net> Message-ID: <1282095223.96301.1322073433363.JavaMail.root@mail-01.cse.ucsc.edu> > Are you absolutely sure you want or need lossless? I'm assuming > you probably want to use the "slow" preset instead since you are > attempting to apply the -b option which is probably ignored with > lossless presets. My distribution (FreeBSD's ports installation) does not come with a "slow" preset; here's the list of what I have: libvpx-1080p libvpx-1080p50_60 libvpx-360p libvpx-720p libvpx-720p50_60 libx264-baseline libx264-ipod320 libx264-ipod640 libx264-lossless_fast libx264-lossless_max libx264-lossless_medium libx264-lossless_slow libx264-lossless_slower libx264-lossless_ultrafast Would you suggest a different preset? Or perhaps can you furnish the contents of the "slow" preset file so that I can try that? > most clients needing H264 BASELINE for HTML5 I re-ran the conversion like this: ffmpeg -y -i '1321566272.uploaded' -f mp4 -threads 4 -vcodec libx264 -vpre libx264-lossless_slow -vpre libx264-baseline -b '391104' -acodec libfaac -ar 22050 -r 30 -g 150 -s '640x480' '1321566272.mp4' That did not fix the problem; IE is still not playing the video. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Tim Gustafson tjg at soe.ucsc.edu Baskin School of Engineering 831-459-5354 UC Santa Cruz Baskin Engineering 317B -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- From rogerdpack2 at gmail.com Wed Nov 23 19:42:42 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 23 Nov 2011 11:42:42 -0700 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: > It's not really a bug, it's an unknowable fact about the input and output > video. The only way to fix it is to ask the user what they think the input > is and what the output should be. I suppose it just seems odd that, given a "known" full swing RGB bitmap, ffplay is unable to display it in full swing (since it has control of the conversions from beginning to end). From sweetthdevil at gmail.com Wed Nov 23 20:51:33 2011 From: sweetthdevil at gmail.com (Sw@g) Date: Wed, 23 Nov 2011 19:51:33 +0000 Subject: [FFmpeg-user] New version - date in name Message-ID: <4ECD4EC5.2040304@gmail.com> Hi all, I am using ffmepg on arch - the update from today return an error because of me using date in the name of the file recorded (see below for the command) the error return no file or directory of cam_`date +%#F_%T`.mkv (being the current date/time) Do I must change the format of the command or else? Looking forward from your replies, Regards, => ffmpeg -f alsa -i pulse -acodec ac3 -f video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset medium -crf 26 -threads 0 cam_`date +%#F_%T`.mkv From rhodri at kynesim.co.uk Wed Nov 23 21:19:28 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Wed, 23 Nov 2011 20:19:28 -0000 Subject: [FFmpeg-user] New version - date in name In-Reply-To: <4ECD4EC5.2040304@gmail.com> References: <4ECD4EC5.2040304@gmail.com> Message-ID: On Wed, 23 Nov 2011 19:51:33 -0000, Sw at g wrote: > I am using ffmepg on arch - the update from today return an error > because of me using date in the name of the file recorded (see below for > the command) > the error return no file or directory of cam_`date +%#F_%T`.mkv (being > the current date/time) If that is the error that you got, rather than an approximation of it, then whatever you used to run the ffmpeg command didn't interpret the backticks as meaning "execute this command". I suspect you have shell escaping issues. Showing us the exact, uncut error message would help. -- Rhodri James Kynesim Ltd From phil_rhodes at rocketmail.com Wed Nov 23 23:04:53 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Wed, 23 Nov 2011 22:04:53 -0000 Subject: [FFmpeg-user] bounty: fix this bug #437 $150 In-Reply-To: References: Message-ID: On Wed, 23 Nov 2011 18:42:42 -0000, Roger Pack wrote: >> It's not really a bug, it's an unknowable fact about the input and >> output >> video. The only way to fix it is to ask the user what they think the >> input >> is and what the output should be. > > I suppose it just seems odd that, given a "known" full swing RGB > bitmap, ffplay is unable to display it in full swing (since it has > control of the conversions from beginning to end). Yes. Thomas Worth knows more about this than I do, because he spent some time characterising ffmpeg's exact behaviour and writing his way around it using the source code. The features are in there to solve the problem, they're just not exposed to the user. I suspect this is another case of some software engineers assuming that anything they don't understand isn't important... P From mail2ashi.86 at gmail.com Thu Nov 24 05:12:27 2011 From: mail2ashi.86 at gmail.com (Ashish Mathur) Date: Thu, 24 Nov 2011 09:42:27 +0530 Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 Message-ID: Hi, I am new to ffmpeg and am using ffmpeg 0.8.5. I want to utilize rtsp demuxer of ffmpeg to receive rtsp stream and play it. However, I am facing the following issue. Please help me in solving it: Issue: How do we enable 'rtsp' demuxer in ffmpeg 0.8.5 stack? In av_register_all() there is a call to REGISTER_MUXDEMUX(RTSP, rtsp) but when i run configure in the list of enabled demuxers there is no 'RTSP' demuxer. I saw the dynamically generated config.h and config.mak and in both, rtsp demuxer is defined with value 0 or disabled. What exactly should be done to enable rtsp demuxer? Thanks in advance for any dort of help. Regards, Ashish From vganesh2 at ncsu.edu Thu Nov 24 06:02:59 2011 From: vganesh2 at ncsu.edu (Varun Ganesh) Date: Thu, 24 Nov 2011 00:02:59 -0500 Subject: [FFmpeg-user] FFMPEG unable to receive UDP stream Message-ID: Hi, I want to receive a UDP stream, transcode it and send it out to another host. I have setup a streaming media server using vlc. My machine IP is 192.168.1.108. I setup the vlc to stream on udp:// 192.168.1.108 port 1234. I tried receiving the stream on the same machine using the command ffmpeg -i udp://192.168.1.108:1234 From vganesh2 at ncsu.edu Thu Nov 24 06:07:56 2011 From: vganesh2 at ncsu.edu (Varun Ganesh) Date: Thu, 24 Nov 2011 00:07:56 -0500 Subject: [FFmpeg-user] FFMPEG unable to receive UDP stream In-Reply-To: References: Message-ID: Hi, I am sorry for sending the mail incorrectly. I am resending it properly. I want to receive a UDP stream, transcode it and send it out to another host. I have setup a streaming media server using vlc. My machine IP is 192.168.1.108. I setup the vlc to stream on udp:// > 192.168.1.108 port 1234. I tried receiving the stream on the same machine > using the command > > ffmpeg -i udp://192.168.1.108:1234 -f mpegts -vcodec mpeg4 -acodec mp2 > -ac 2 -ab 128k -s 720x432 -r 30 -re -b 2000k -threads 2 udp:// > 192.168.1.107:8005 > I have configured a client to listen on port 8005 to receive the stream on the second machine. But the ffmpeg doesn't accept the first connection. I was able to use a separate vlc client to receive the same on the host machine. I also tried it to send a file locally saved on the machine using ffmpeg -i filename -f mpegts -vcodec mpeg4 -acodec mp2 -ac 2 -ab 128k -s 720x432 -r 30 -re -b 2000k -threads 2 udp://192.168.1.107:8005 and was able to successfully receive it on the second machine. How can I get FFMPEG to listen and receive the UDP stream from the VLC server? Thanks, Varun Ganesh From cehoyos at ag.or.at Thu Nov 24 11:02:27 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:02:27 +0000 (UTC) Subject: [FFmpeg-user] FFMPEG unable to receive UDP stream References: Message-ID: Varun Ganesh ncsu.edu> writes: > > ffmpeg -i udp://192.168.1.108:1234 -f mpegts -vcodec mpeg4 -acodec mp2 > > -ac 2 -ab 128k -s 720x432 -r 30 -re -b 2000k -threads 2 udp:// > > 192.168.1.107:8005 Does encoding to a file work? Complete, uncut console output missing. Carl Eugen From cehoyos at ag.or.at Thu Nov 24 11:03:13 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:03:13 +0000 (UTC) Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 References: Message-ID: Ashish Mathur gmail.com> writes: > I saw the dynamically generated config.h and > config.mak and in both, rtsp demuxer is defined with value 0 or > disabled. What exactly should be done to enable rtsp demuxer? How does your configure command look like? Carl Eugen From sweetthdevil at gmail.com Thu Nov 24 11:20:55 2011 From: sweetthdevil at gmail.com (Sw@g) Date: Thu, 24 Nov 2011 10:20:55 +0000 Subject: [FFmpeg-user] New version - date in name In-Reply-To: References: <4ECD4EC5.2040304@gmail.com> Message-ID: <4ECE1A87.1090501@gmail.com> On Wed 23 Nov 2011 20:19:28 GMT, Rhodri James wrote: > On Wed, 23 Nov 2011 19:51:33 -0000, Sw at g wrote: > >> I am using ffmepg on arch - the update from today return an error >> because of me using date in the name of the file recorded (see below >> for the command) > >> the error return no file or directory of cam_`date +%#F_%T`.mkv >> (being the current date/time) > > If that is the error that you got, rather than an approximation of it, > then whatever you used to run the ffmpeg command didn't interpret the > backticks as meaning "execute this command". I suspect you have shell > escaping issues. Showing us the exact, uncut error message would help. > Hi Many thanks for your quick reply, see below the complete terminal output. Looking forward for your reply, [sweetth at myhost ~]$ ffmpeg -f alsa -i pulse -acodec ac3 -f video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset medium -crf 26 -threads 0 cam_`date +%#F_%T`.mkv ffmpeg version N-35110-g0b9a69f, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 23 2011 12:51:56 with gcc 4.6.2 configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-libgsm --enable-libspeex --enable-postproc --enable-shared --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libschroedinger --enable-libopenjpeg --enable-librtmp --enable-gpl --enable-version3 --enable-runtime-cpudetect --disable-debug --disable-static libavutil 51. 26. 0 / 51. 26. 0 libavcodec 53. 37. 0 / 53. 37. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [alsa @ 0x9d7faa0] Estimating duration from bitrate, this may be inaccurate Input #0, alsa, from 'pulse': Duration: N/A, start: 1322129967.230191, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [video4linux2,v4l2 @ 0x9d84cc0] Estimating duration from bitrate, this may be inaccurate Input #1, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 92.040540, bitrate: 73728 kb/s Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 73728 kb/s, 15 tbr, 1000k tbn, 15 tbc cam_2011-11-24_10:19:27.mkv: No such file or directory From cehoyos at ag.or.at Thu Nov 24 11:21:54 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:21:54 +0000 (UTC) Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 References: Message-ID: Ashish Mathur gmail.com> writes: > I am new to ffmpeg and am using ffmpeg 0.8.5. I forgot: If you are a user (and not an operation system distributor), please use current git head, it has less bugs and more features than any released version. (I believe it also contains networking fixes.) Carl Eugen From cehoyos at ag.or.at Thu Nov 24 11:29:23 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:29:23 +0000 (UTC) Subject: [FFmpeg-user] bounty: fix this bug #437 $150 References: Message-ID: Roger Pack gmail.com> writes: > http://ffmpeg.org/trac/ffmpeg/ticket/437 > It is reproducible in linux I believe. What is unfortunately really missing - and makes fixing this "bug" (I can imagine the misunderstanding is simply about how colours are represented in different standards, I believe there are several relevant threads on doom9) extremely unlikely, no matter how much you offer - is a way to reproduce your issue. Is it only reproducible with ffplay? Note that SDL (the library ffplay depends on) is known for different bugs on different platforms. Apart from that, I would not really recommend ffplay as a playback application (it is very slow), there are alternatives like MPLayer (that allows you to select the colour representation at least for VDPAU). If it is reproducible with ffmpeg (that does not depend on external libraries the same way as ffplay), please add a sample and a command line together with complete, uncut console output. Carl Eugen From nicolas.george at normalesup.org Thu Nov 24 11:31:47 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 24 Nov 2011 11:31:47 +0100 Subject: [FFmpeg-user] New version - date in name In-Reply-To: <4ECE1A87.1090501@gmail.com> References: <4ECD4EC5.2040304@gmail.com> <4ECE1A87.1090501@gmail.com> Message-ID: <20111124103147.GA6488@phare.normalesup.org> Le quartidi 4 frimaire, an CCXX, Sw at g a ?crit?: > [sweetth at myhost ~]$ ffmpeg -f alsa -i pulse -acodec ac3 -f > video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset > medium -crf 26 -threads 0 cam_`date +%#F_%T`.mkv > cam_2011-11-24_10:19:27.mkv: No such file or directory Strange. You should try the following tests: * Directly write "cam_2011-11-24_10:19:27.mkv" on ffmpeg command-line, without using the shell substitution. * The same with a more normal name: "cam_20111124_101927.mkv". * Opening the file from the shell instead of ffmpeg: ffmpeg ... -f mkv - > cam...mkv * Just creating the file from the shell: touch cam_2011-11-24_10:19:27.mkv Those tests would help find out what is going wrong. 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 Thu Nov 24 11:35:35 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:35:35 +0000 (UTC) Subject: [FFmpeg-user] New version - date in name References: <4ECD4EC5.2040304@gmail.com> Message-ID: Sw g gmail.com> writes: > I am using ffmepg on arch - the update from today return an error > because of me using date in the name of the file recorded (see below for > the command) Do you know which version still worked with your command line? Carl Eugen From cehoyos at ag.or.at Thu Nov 24 11:37:33 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 10:37:33 +0000 (UTC) Subject: [FFmpeg-user] New version - date in name References: <4ECD4EC5.2040304@gmail.com> <4ECE1A87.1090501@gmail.com> <20111124103147.GA6488@phare.normalesup.org> Message-ID: Nicolas George normalesup.org> writes: > * Directly write "cam_2011-11-24_10:19:27.mkv" on ffmpeg command-line, > without using the shell substitution. FFmpeg doesn't like colons in file names (anymore), this is a regression, afaict. Carl Eugen From sweetthdevil at gmail.com Thu Nov 24 11:43:31 2011 From: sweetthdevil at gmail.com (Sw@g) Date: Thu, 24 Nov 2011 10:43:31 +0000 Subject: [FFmpeg-user] New version - date in name In-Reply-To: <4ECE1A87.1090501@gmail.com> References: <4ECD4EC5.2040304@gmail.com> <4ECE1A87.1090501@gmail.com> Message-ID: <4ECE1FD3.4030107@gmail.com> On 24/11/11 10:20, Sw at g wrote: > > > On Wed 23 Nov 2011 20:19:28 GMT, Rhodri James wrote: >> On Wed, 23 Nov 2011 19:51:33 -0000, Sw at g wrote: >> >>> I am using ffmepg on arch - the update from today return an error >>> because of me using date in the name of the file recorded (see below >>> for the command) >> >>> the error return no file or directory of cam_`date +%#F_%T`.mkv >>> (being the current date/time) >> >> If that is the error that you got, rather than an approximation of it, >> then whatever you used to run the ffmpeg command didn't interpret the >> backticks as meaning "execute this command". I suspect you have shell >> escaping issues. Showing us the exact, uncut error message would help. >> > > Hi Many thanks for your quick reply, see below the complete terminal > output. > > Looking forward for your reply, > > [sweetth at myhost ~]$ ffmpeg -f alsa -i pulse -acodec ac3 -f > video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset > medium -crf 26 -threads 0 cam_`date +%#F_%T`.mkv > ffmpeg version N-35110-g0b9a69f, Copyright (c) 2000-2011 the FFmpeg > developers > built on Nov 23 2011 12:51:56 with gcc 4.6.2 > configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis > --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora > --enable-libgsm --enable-libspeex --enable-postproc --enable-shared > --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb > --enable-libschroedinger --enable-libopenjpeg --enable-librtmp > --enable-gpl --enable-version3 --enable-runtime-cpudetect > --disable-debug --disable-static > libavutil 51. 26. 0 / 51. 26. 0 > libavcodec 53. 37. 0 / 53. 37. 0 > libavformat 53. 21. 0 / 53. 21. 0 > libavdevice 53. 4. 0 / 53. 4. 0 > libavfilter 2. 49. 0 / 2. 49. 0 > libswscale 2. 1. 0 / 2. 1. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [alsa @ 0x9d7faa0] Estimating duration from bitrate, this may be > inaccurate > Input #0, alsa, from 'pulse': > Duration: N/A, start: 1322129967.230191, bitrate: N/A > Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s > [video4linux2,v4l2 @ 0x9d84cc0] Estimating duration from bitrate, this > may be inaccurate > Input #1, video4linux2,v4l2, from '/dev/video0': > Duration: N/A, start: 92.040540, bitrate: 73728 kb/s > Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, > 73728 kb/s, 15 tbr, 1000k tbn, 15 tbc > cam_2011-11-24_10:19:27.mkv: No such file or directory Right, As requested => cam_2011-11-24_10:19:27.mkv ffmpeg -f alsa -i pulse -acodec ac3 -f video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset medium -crf 26 -threads 0 cam_2011-11-24_10:19:27.mkv ffmpeg version N-35110-g0b9a69f, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 23 2011 12:51:56 with gcc 4.6.2 configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-libgsm --enable-libspeex --enable-postproc --enable-shared --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libschroedinger --enable-libopenjpeg --enable-librtmp --enable-gpl --enable-version3 --enable-runtime-cpudetect --disable-debug --disable-static libavutil 51. 26. 0 / 51. 26. 0 libavcodec 53. 37. 0 / 53. 37. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [alsa @ 0x9a6caa0] Estimating duration from bitrate, this may be inaccurate Input #0, alsa, from 'pulse': Duration: N/A, start: 1322130866.877732, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [video4linux2,v4l2 @ 0x9a71cc0] Estimating duration from bitrate, this may be inaccurate Input #1, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 991.762297, bitrate: 73728 kb/s Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 73728 kb/s, 15 tbr, 1000k tbn, 15 tbc cam_2011-11-24_10:19:27.mkv: No such file or directory => cam_20111124_101927.mkv This work To answer Carl the last version on arch was => ffmpeg-20111108-1-i686 Right so if it doesn't like the colons anymore, how can I amend my command to give a similar date/time? Regards From sweetthdevil at gmail.com Thu Nov 24 11:51:08 2011 From: sweetthdevil at gmail.com (Sw@g) Date: Thu, 24 Nov 2011 10:51:08 +0000 Subject: [FFmpeg-user] New version - date in name In-Reply-To: <4ECE1FD3.4030107@gmail.com> References: <4ECD4EC5.2040304@gmail.com> <4ECE1A87.1090501@gmail.com> <4ECE1FD3.4030107@gmail.com> Message-ID: <4ECE219C.7070000@gmail.com> On Thu 24 Nov 2011 10:43:31 GMT, Sw at g wrote: > > > On 24/11/11 10:20, Sw at g wrote: >> >> >> On Wed 23 Nov 2011 20:19:28 GMT, Rhodri James wrote: >>> On Wed, 23 Nov 2011 19:51:33 -0000, Sw at g >>> wrote: >>> >>>> I am using ffmepg on arch - the update from today return an error >>>> because of me using date in the name of the file recorded (see >>>> below for the command) >>> >>>> the error return no file or directory of cam_`date +%#F_%T`.mkv >>>> (being the current date/time) >>> >>> If that is the error that you got, rather than an approximation of it, >>> then whatever you used to run the ffmpeg command didn't interpret the >>> backticks as meaning "execute this command". I suspect you have shell >>> escaping issues. Showing us the exact, uncut error message would help. >>> >> >> Hi Many thanks for your quick reply, see below the complete terminal >> output. >> >> Looking forward for your reply, >> >> [sweetth at myhost ~]$ ffmpeg -f alsa -i pulse -acodec ac3 -f >> video4linux2 -s 640x480 -r 15 -i /dev/video0 -vcodec libx264 -preset >> medium -crf 26 -threads 0 cam_`date +%#F_%T`.mkv >> ffmpeg version N-35110-g0b9a69f, Copyright (c) 2000-2011 the FFmpeg >> developers >> built on Nov 23 2011 12:51:56 with gcc 4.6.2 >> configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis >> --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora >> --enable-libgsm --enable-libspeex --enable-postproc --enable-shared >> --enable-x11grab --enable-libopencore_amrnb >> --enable-libopencore_amrwb --enable-libschroedinger >> --enable-libopenjpeg --enable-librtmp --enable-gpl --enable-version3 >> --enable-runtime-cpudetect --disable-debug --disable-static >> libavutil 51. 26. 0 / 51. 26. 0 >> libavcodec 53. 37. 0 / 53. 37. 0 >> libavformat 53. 21. 0 / 53. 21. 0 >> libavdevice 53. 4. 0 / 53. 4. 0 >> libavfilter 2. 49. 0 / 2. 49. 0 >> libswscale 2. 1. 0 / 2. 1. 0 >> libpostproc 51. 2. 0 / 51. 2. 0 >> [alsa @ 0x9d7faa0] Estimating duration from bitrate, this may be >> inaccurate >> Input #0, alsa, from 'pulse': >> Duration: N/A, start: 1322129967.230191, bitrate: N/A >> Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s >> [video4linux2,v4l2 @ 0x9d84cc0] Estimating duration from bitrate, >> this may be inaccurate >> Input #1, video4linux2,v4l2, from '/dev/video0': >> Duration: N/A, start: 92.040540, bitrate: 73728 kb/s >> Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, >> 73728 kb/s, 15 tbr, 1000k tbn, 15 tbc >> cam_2011-11-24_10:19:27.mkv: No such file or directory > > Right, > > As requested > > => cam_2011-11-24_10:19:27.mkv > ffmpeg -f alsa -i pulse -acodec ac3 -f video4linux2 -s 640x480 -r 15 > -i /dev/video0 -vcodec libx264 -preset medium -crf 26 -threads 0 > cam_2011-11-24_10:19:27.mkv > ffmpeg version N-35110-g0b9a69f, Copyright (c) 2000-2011 the FFmpeg > developers > built on Nov 23 2011 12:51:56 with gcc 4.6.2 > configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis > --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora > --enable-libgsm --enable-libspeex --enable-postproc --enable-shared > --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb > --enable-libschroedinger --enable-libopenjpeg --enable-librtmp > --enable-gpl --enable-version3 --enable-runtime-cpudetect > --disable-debug --disable-static > libavutil 51. 26. 0 / 51. 26. 0 > libavcodec 53. 37. 0 / 53. 37. 0 > libavformat 53. 21. 0 / 53. 21. 0 > libavdevice 53. 4. 0 / 53. 4. 0 > libavfilter 2. 49. 0 / 2. 49. 0 > libswscale 2. 1. 0 / 2. 1. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [alsa @ 0x9a6caa0] Estimating duration from bitrate, this may be > inaccurate > Input #0, alsa, from 'pulse': > Duration: N/A, start: 1322130866.877732, bitrate: N/A > Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s > [video4linux2,v4l2 @ 0x9a71cc0] Estimating duration from bitrate, this > may be inaccurate > Input #1, video4linux2,v4l2, from '/dev/video0': > Duration: N/A, start: 991.762297, bitrate: 73728 kb/s > Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, > 73728 kb/s, 15 tbr, 1000k tbn, 15 tbc > cam_2011-11-24_10:19:27.mkv: No such file or directory > > => cam_20111124_101927.mkv > This work > > To answer Carl the last version on arch was => ffmpeg-20111108-1-i686 > > Right so if it doesn't like the colons anymore, how can I amend my > command to give a similar date/time? > > Regards Hi again, I am amended the file name to "cam_`date +%#F_%H.%M.%S`.mkv" that remove the colons and works now given a similar result. I however think it's far from being an improvement. Regards, From cehoyos at ag.or.at Thu Nov 24 14:25:47 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 13:25:47 +0000 (UTC) Subject: [FFmpeg-user] New version - date in name References: <4ECD4EC5.2040304@gmail.com> Message-ID: Sw g gmail.com> writes: > I am using ffmepg on arch - the update from today return an error > because of me using date in the name of the file recorded (see below for > the command) > > the error return no file or directory of cam_`date +%#F_%T`.mkv (being > the current date/time) This is a regression since c12e1bd1, go back to 2bb1c713 if you need an immediate work-around until fixed. Thank you for the report, Carl Eugen From cehoyos at ag.or.at Thu Nov 24 20:06:04 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 19:06:04 +0000 (UTC) Subject: [FFmpeg-user] New version - date in name References: <4ECD4EC5.2040304@gmail.com> Message-ID: Sw g gmail.com> writes: > I am using ffmepg on arch - the update from today return an error > because of me using date in the name of the file recorded (see below for > the command) > > the error return no file or directory of cam_`date +%#F_%T`.mkv (being > the current date/time) Fixed in current git head, thanks again for the report! Carl Eugen From bartoszx at gmail.com Thu Nov 24 20:47:50 2011 From: bartoszx at gmail.com (bartoszx) Date: Thu, 24 Nov 2011 20:47:50 +0100 Subject: [FFmpeg-user] Can I add audio to avi with audio Message-ID: <8D2433E0-B7C1-436C-A295-B59AD7CFDDA3@gmail.com> Can I add audio to file which already has audio. I have avi with couple of sound fx and now I want to add loop in the background. I've tried ffmpeg -i loop.mp3 -i test.avi -acodec copy -vcodec copy out.avi it adds loop instead existing audio in test.avi From laurent.bellegarde at free.fr Thu Nov 24 09:47:35 2011 From: laurent.bellegarde at free.fr (laurent.bellegarde) Date: Thu, 24 Nov 2011 09:47:35 +0100 Subject: [FFmpeg-user] list of audio codecs only ? Message-ID: <4ECE04A7.6010503@free.fr> Hi all, i'm trying to improve a new program which use ffmpeg. I need to get back from a terminal the complete list of audio codecs. the command ffmpeg -codecs doesn't work and told : laurent at laurent-laptop:~$ ffmpeg -codecs FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1.2 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Sep 16 2011 17:04:18, gcc: 4.4.3 ffmpeg: missing argument for option '-codecs' the command ffmpeg -formats works, but all the codecs are listed. Any suggestions ? Laurent lprod.org From ckutay at cse.unsw.edu.au Thu Nov 24 11:44:23 2011 From: ckutay at cse.unsw.edu.au (Cat Kutay) Date: Thu, 24 Nov 2011 21:44:23 +1100 Subject: [FFmpeg-user] uploaded file Message-ID: <44D6F35E-E632-4086-8CF2-4E4CB0EA80CE@cse.unsw.edu.au> Using ffmpeg2theora_0.24-2build2_i386.deb on ubuntu get message Using network protocols without global network initialization. Please use avformat_network_init(), this will become mandatory later. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9326460] Unimplemented container channel layout. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9326460] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list. So i did...... But host not existing and The resultant .ogv file is not readable by Safari, but is by Chrome and Firefox Cheers ---- )\._.,--....,'``. Cat Kutay _/, _.. \! _\ ;`._ ,.__' Room 206 CSE `._.-(,_..'--(,_..'`-.;.' UNSW, Australia, 2052. Email: ckutay at cse.unsw.edu.au Skype: cat.kutay Ph: 61 (0)2 9385 5257 Fax: 61 (0)2 9385 5995 Mob: 0418 455 089 http://www.cse.unsw.edu.au/~ckutay Languages are precious storehouses of history, experience and culture; a crucial link between the past and the future - Jeanie Bell, Aboriginal Linguist From nacho.torronteras at asesoresnt.com Thu Nov 24 13:26:45 2011 From: nacho.torronteras at asesoresnt.com (Nacho Torronteras) Date: Thu, 24 Nov 2011 13:26:45 +0100 Subject: [FFmpeg-user] ffmpeg for rtsp Message-ID: ffmpeg can be used to send rtsp data to Youtube live? From ubitux at gmail.com Thu Nov 24 21:05:40 2011 From: ubitux at gmail.com (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Thu, 24 Nov 2011 21:05:40 +0100 Subject: [FFmpeg-user] list of audio codecs only ? In-Reply-To: <4ECE04A7.6010503@free.fr> References: <4ECE04A7.6010503@free.fr> Message-ID: <20111124200540.GI1935@leki> On Thu, Nov 24, 2011 at 09:47:35AM +0100, laurent.bellegarde wrote: > Hi all, > > i'm trying to improve a new program which use ffmpeg. > > I need to get back from a terminal the complete list of audio codecs. > > the command > > ffmpeg -codecs doesn't work and told : > Works here with a recent ffmpeg. > laurent at laurent-laptop:~$ ffmpeg -codecs > FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2, Copyright (c) > 2000-2009 Fabrice Bellard, et al. 0.5 was released in march 2009, just upgrade your ffmpeg copy. [...] -- 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 Nov 24 21:20:46 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 24 Nov 2011 20:20:46 +0000 (UTC) Subject: [FFmpeg-user] uploaded file References: <44D6F35E-E632-4086-8CF2-4E4CB0EA80CE@cse.unsw.edu.au> Message-ID: Cat Kutay cse.unsw.edu.au> writes: > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9326460] Unimplemented container channel layout. > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x9326460] If you want to help, upload a sample of > this file to > ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing > list. > So i did...... > But host not existing $ host upload.ffmpeg.org upload.ffmpeg.org is an alias for streams.videolan.org. streams.videolan.org is an alias for jones.videolan.org. jones.videolan.org has address 138.195.131.196 (The directory "MPlayer" does not exist anymore, just "cd incoming") Alternatively, you can upload to http://www.datafilehost.com/ and post the download link here. Carl Eugen From mail2ashi.86 at gmail.com Fri Nov 25 04:58:41 2011 From: mail2ashi.86 at gmail.com (Ashish Mathur) Date: Fri, 25 Nov 2011 09:28:41 +0530 Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 In-Reply-To: References: Message-ID: Hi Carl, Thanks for the response. After some experiments i was able to figure out a way of enabling 'rtsp' demuxer but my findings are worth a discussion. My configure command looked like: ./configure --disable-demux=asf --disable-decoders Actually in my application, i don't want 'asf' demuxing to happen through ffmpeg and therefore the following line in allformats.c, inside av_register_all() is also commented : //REGISTER_MUXDEMUX (ASF, asf); With the above command i don't know why RTSP demuxer was getting disabled. As an experiment i removed --disable-demux=asf, and ran configure again. Now, in the list of enabled demuxers i was able to see 'RTSP' and the corresponding 'RTSP' files compiled. Also, new config.h and config.mak have RTSP demuxer enabled. This behavior was a little strange as removing --disable-demux=asf actually sorted out the issue. I think --disable-demux=asf was unrequired once the REGISTER_MUXDEMUX(ASF,asf) has been commented in allformats.c. Could you help in justifying this behavior by ffmpeg 0.8.5 configuration? Regards, Ashish On 11/24/11, Carl Eugen Hoyos wrote: > Ashish Mathur gmail.com> writes: > >> I saw the dynamically generated config.h and >> config.mak and in both, rtsp demuxer is defined with value 0 or >> disabled. What exactly should be done to enable rtsp demuxer? > > How does your configure command look like? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mail2ashi.86 at gmail.com Fri Nov 25 05:11:06 2011 From: mail2ashi.86 at gmail.com (Ashish Mathur) Date: Fri, 25 Nov 2011 09:41:06 +0530 Subject: [FFmpeg-user] FFMPEG 0.8.5 : AC3 (a52) codec unrecognized while receving RTSP stream Message-ID: Hi, I am using FFMPEG 0.8.5 to receive RTSP streams. Just to be sure that it works fine for all kind of elementary streams, i used ffprobe application to probe and recognize the codecs of the incoming RTSP streams. Whenever i stream H.264 video stream and AC3(aka a52) audio stream through VLC( using its RTSP server) , and try probing it using ffprobe command on the client side, I get : [rtsp @ 0x1165c400] audio codec set to: (null) [rtsp @ 0x1165c400] audio samplerate set to: 48000 [rtsp @ 0x1165c400] audio channels set to: 6 [rtsp @ 0x1165c400] video codec set to: h264 [NULL @ 0x11662720] RTP Packetization Mode: 1 [NULL @ 0x11662720] RTP Profile IDC: 64 Profile IOP: 0 Level: 29 FFMPEG 0.8.5 RTSP client is not able to recognize AC3(aka a52) audio codec. Hence, even when i play these RTSP streams using FFMPEG 0.8.5 only the video H.264 stream gets played. Is this a known issue with FFMPEG 0.8.5? Or is there a fix to it in the latest FFMPEG versions? Regards, Ashish From cehoyos at ag.or.at Fri Nov 25 11:35:50 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 25 Nov 2011 10:35:50 +0000 (UTC) Subject: [FFmpeg-user] FFMPEG 0.8.5 : AC3 (a52) codec unrecognized while receving RTSP stream References: Message-ID: Ashish Mathur gmail.com> writes: > Is this a known issue with FFMPEG 0.8.5? > Or is there a fix to it in the latest FFMPEG versions? Please test and report back. Carl Eugen From cehoyos at ag.or.at Fri Nov 25 11:38:41 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 25 Nov 2011 10:38:41 +0000 (UTC) Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 References: Message-ID: Ashish Mathur gmail.com> writes: > ./configure --disable-demux=asf --disable-decoders > With the above command i don't know why RTSP demuxer was getting > disabled. As an experiment i removed --disable-demux=asf, and ran > configure again. Now, in the list of enabled demuxers i was able to > see 'RTSP' and the corresponding 'RTSP' files compiled. Also, new > config.h and config.mak have RTSP demuxer enabled. Yes, the rtsp demuxer depends on the asf demuxer. Carl Eugen From mail2ashi.86 at gmail.com Fri Nov 25 14:10:44 2011 From: mail2ashi.86 at gmail.com (Ashish Mathur) Date: Fri, 25 Nov 2011 18:40:44 +0530 Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 In-Reply-To: References: Message-ID: So this means that commenting REGISTER_MUXDEMUX(ASF,asf) in allformats.c is enough to disable asf demuxer in ffmpeg? There is no need of using --disable-demux=asf?? Regards, Ashish Mathur On 11/25/11, Carl Eugen Hoyos wrote: > Ashish Mathur gmail.com> writes: > >> ./configure --disable-demux=asf --disable-decoders > >> With the above command i don't know why RTSP demuxer was getting >> disabled. As an experiment i removed --disable-demux=asf, and ran >> configure again. Now, in the list of enabled demuxers i was able to >> see 'RTSP' and the corresponding 'RTSP' files compiled. Also, new >> config.h and config.mak have RTSP demuxer enabled. > > Yes, the rtsp demuxer depends on the asf demuxer. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Fri Nov 25 14:57:44 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 25 Nov 2011 13:57:44 +0000 (UTC) Subject: [FFmpeg-user] means to inverse telecine References: <4ECBD076.8020704@pps-inc.com> <5B45A63B-CBC6-42D7-9B59-D354027E041D@mac.com> <4ECBF18B.9010306@pps-inc.com> Message-ID: David Rice mac.com> writes: > > A reproducible bug-report that shows that -vf mp=pullup is not working would > > be very welcome! > > I started one here https://ffmpeg.org/trac/ffmpeg/ticket/681 Should be "reproducible" now. Carl Eugen From cehoyos at ag.or.at Fri Nov 25 15:34:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 25 Nov 2011 14:34:36 +0000 (UTC) Subject: [FFmpeg-user] Enabling 'rtsp' demuxer in FFMPEG 0.8.5 References: Message-ID: Ashish Mathur gmail.com> writes: > So this means that commenting REGISTER_MUXDEMUX(ASF,asf) in > allformats.c is enough to disable asf demuxer in ffmpeg? I suspect that only stops the demuxer from being registered (and used), but the asf demuxer will still be compiled if that is your question. > There is no need of using --disable-demux=asf?? If you do not want to compile the asf demuxer (and all components that depend on the asf demuxer like the rtsp demuxer) you have to pass --disable-demuxer=asf to configure. Carl Eugen From pb at das-werkstatt.com Fri Nov 25 15:45:16 2011 From: pb at das-werkstatt.com (Peter B.) Date: Fri, 25 Nov 2011 15:45:16 +0100 Subject: [FFmpeg-user] Can I add audio to avi with audio In-Reply-To: <8D2433E0-B7C1-436C-A295-B59AD7CFDDA3@gmail.com> References: <8D2433E0-B7C1-436C-A295-B59AD7CFDDA3@gmail.com> Message-ID: <4ECFA9FC.1050205@das-werkstatt.com> bartoszx wrote: > Can I add audio to file which already has audio. I have avi with couple of sound fx and now I want to add loop in the background. > I've tried ffmpeg -i loop.mp3 -i test.avi -acodec copy -vcodec copy out.avi > it adds loop instead existing audio in test.avi > > For what you are trying to do, you will need to mix the audio in an editing application and then bounce it as one audio track into the final AVI. Pb From mark at mdsh.com Fri Nov 25 19:25:29 2011 From: mark at mdsh.com (Mark Himsley) Date: Fri, 25 Nov 2011 18:25:29 +0000 Subject: [FFmpeg-user] static link including opencv Message-ID: <4ECFDD99.9060303@mdsh.com> I'd like a hint on including opencv in a static FFmpeg. This is using FFmpeg git head and OpenCV-2.3.1a. I built opencv with these settings: cmake \ -D CMAKE_BUILD_TYPE=RELEASE \ -D BUILD_SHARED_LIBS=OFF \ -D WITH_FFMPEG=OFF \ -D BUILD_TESTS=OFF \ -D CMAKE_INSTALL_PREFIX=$PREFIX \ ../OpenCV-2.3.1/ make sudo make install I'm configuring FFmbc with this ($PREFIX is /usr/local) /configure \ --arch=x86 \ --cpu=i686 \ --enable-static \ --extra-cflags='--static -I$PREFIX/include' \ --extra-libs='-static -L$PREFIX/lib' \ --prefix=$PREFIX \ --disable-ffplay \ --disable-ffprobe \ --enable-gpl \ --enable-version3 \ --enable-nonfree \ --enable-pthreads \ --enable-x11grab \ --enable-libdirac \ --enable-libfaac \ --enable-libgsm \ --enable-libmp3lame \ --enable-libopenjpeg \ --enable-libspeex \ --enable-libtheora \ --enable-libvorbis \ --enable-libvpx \ --enable-libxvid \ --enable-libopencv In order for ./configure to complete I had to edit /usr/local/lib/pkgconfig/opencv.pc to add -ltr to the end of the the Libs: line. But now making FFmpeg, all goes well until the link: $ make V=1 [...] gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil -o ffmpeg_g ffmpeg.o cmdutils.o -lavdevice -lavfilter -lavformat -lavcodec -lpostproc -lswresample -lswscale -lavutil -ldl -lxvidcore -lvpx -lvpx -lvorbisenc -lvorbis -logg -ltheoraenc -ltheoradec -logg -lspeex -lopenjpeg -L/usr/local/lib -lopencv_core -lopencv_imgproc -lopencv_highgui -lopencv_ml -lopencv_video -lopencv_features2d -lopencv_calib3d -lopencv_objdetect -lopencv_contrib -lopencv_legacy -lopencv_flann -lrt -lmp3lame -lgsm -lfaac -ldirac_encoder -ldirac_decoder -lm -lstdc++ -lm -pthread -lbz2 -lz -static -L/usr/local/lib /usr/local/lib/libopencv_core.a(system.o): In function `cv::tempfile(char const*)': system.cpp:(.text._ZN2cv8tempfileEPKc+0x3a): warning: the use of `tmpnam' is dangerous, better use `mkstemp' libavformat/libavformat.a(rtsp.o): In function `get_sockaddr': /home/himslm01/src/ffmpeg/libavformat/rtsp.c:160: warning: Using 'getaddrinfo' in statically linked applications requires at runtime the shared libraries from the glibc version used for linking /usr/local/lib/libopencv_imgproc.a(templmatch.o): In function `cv::crossCorr(cv::Mat const&, cv::Mat const&, cv::Mat&, cv::Size_, int, cv::Point_, double, int)': templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0x412): undefined reference to `cv::getOptimalDFTSize(int)' templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0x445): undefined reference to `cv::getOptimalDFTSize(int)' templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0xded): undefined reference to `cv::dft(cv::_InputArray const&, cv::_OutputArray const&, int, int)' templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0x1b2e): undefined reference to `cv::dft(cv::_InputArray const&, cv::_OutputArray const&, int, int)' templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0x1bf6): undefined reference to `cv::mulSpectrums(cv::_InputArray const&, cv::_InputArray const&, cv::_OutputArray const&, int, bool)' templmatch.cpp:(.text._ZN2cv9crossCorrERKNS_3MatES2_RS0_NS_5Size_IiEEiNS_6Point_IiEEdi+0x1c38): undefined reference to `cv::dft(cv::_InputArray const&, cv::_OutputArray const&, int, int)' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 Any ideas? -- Mark From peace at AleksandrSolzhenitsyn.net Fri Nov 25 21:30:48 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Fri, 25 Nov 2011 15:30:48 -0500 Subject: [FFmpeg-user] Why doesn't this play on iPod? Message-ID: <4ECFFAF8.2080008@AleksandrSolzhenitsyn.net> Here's the ffmpeg- i information from a video I downloaded. It transfers and plays on the iPod BUT the width of the video is partially cut off on the iPod screen. Any idea why? Please let me know how to fix it. Soter at Soter-laptop:~/Downloads$ ffmpeg -i rp.mp4 ffmpeg version git-2011-11-24-957867a, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 24 2011 16:13:53 with gcc 4.4.3 configuration: --enable-libmp3lame --enable-libvpx --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab libavutil 51. 27. 0 / 51. 27. 0 libavcodec 53. 37. 0 / 53. 37. 0 libavformat 53. 21. 0 / 53. 21. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 49. 0 / 2. 49. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'rp.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-09-11 20:34:31 Duration: 00:09:58.73, start: 0.000000, bitrate: 610 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360, 508 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 95 kb/s Metadata: creation_time : 2011-09-11 20:34:32 handler_name : At least one output file must be specified From lou at lrcd.com Fri Nov 25 22:00:11 2011 From: lou at lrcd.com (Lou) Date: Fri, 25 Nov 2011 12:00:11 -0900 Subject: [FFmpeg-user] list of audio codecs only ? In-Reply-To: <20111124200540.GI1935@leki> References: <4ECE04A7.6010503@free.fr> <20111124200540.GI1935@leki> Message-ID: <20111125120011.476595f2@lrcd.com> On Thu, 24 Nov 2011 21:05:40 +0100 Cl?ment B?sch wrote: > On Thu, Nov 24, 2011 at 09:47:35AM +0100, laurent.bellegarde wrote: > > Hi all, > > > > i'm trying to improve a new program which use ffmpeg. > > > > I need to get back from a terminal the complete list of audio > > codecs. > > > > the command > > > > ffmpeg -codecs doesn't work and told : > > > > Works here with a recent ffmpeg. > > > laurent at laurent-laptop:~$ ffmpeg -codecs > > FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2, Copyright (c) > > 2000-2009 Fabrice Bellard, et al. > > 0.5 was released in march 2009, just upgrade your ffmpeg copy. For instructions see: HOWTO: Install and use the latest FFmpeg and x264 on Ubuntu http://ubuntuforums.org/showthread.php?t=786095 From adamklobukowski at gmail.com Sat Nov 26 11:29:14 2011 From: adamklobukowski at gmail.com (=?UTF-8?B?QWRhbSBLxYJvYnVrb3dza2k=?=) Date: Sat, 26 Nov 2011 11:29:14 +0100 Subject: [FFmpeg-user] Getting bitrate of a stream Message-ID: <4ED0BF7A.2020704@gmail.com> Hello I need to find out the bitrate of audio and video stream in file (they have only one video and audio stream), using commandline tool (in a script) I find ffprobe unreliable: first, returned string is not easy to parse because it depends on codec used, sometimes it does not show bitrate information at all (XviD), or reports bitrate unreliable (mp3). Is there a better tool for such job? AdamK From hazersha at gmail.com Sat Nov 26 11:40:37 2011 From: hazersha at gmail.com (hazer) Date: Sat, 26 Nov 2011 16:10:37 +0530 Subject: [FFmpeg-user] ffmpeg.exe not working with mov Message-ID: Hi I tried to run ffmpeg.exe with option *-i sample_iTunes.mov -sameq -f swf -y -s 640x360 mySlides.swf* But it is showing an error [swf @ 02D07360] swf does not support that sample rate, choose from (44100, 22050, 11025). But the same mov file works if I put ??an? option. I got the mov file from http://support.apple.com/kb/ht1425 Can anyone help me out. Thanks CodeNimoI From james.darnley at gmail.com Sat Nov 26 11:49:26 2011 From: james.darnley at gmail.com (James Darnley) Date: Sat, 26 Nov 2011 11:49:26 +0100 Subject: [FFmpeg-user] ffmpeg.exe not working with mov In-Reply-To: References: Message-ID: <4ED0C436.7050809@gmail.com> On 2011-11-26 11:40, hazer wrote: > [swf @ 02D07360] swf does not support that sample rate, choose from (44100, 22050, 11025). What is ambiguous about this? SWF only supports those three audio sample rates. You need to choose one of those and then tell ffmpeg to convert to it by using -ar From hazersha at gmail.com Sat Nov 26 13:30:28 2011 From: hazersha at gmail.com (hazer) Date: Sat, 26 Nov 2011 18:00:28 +0530 Subject: [FFmpeg-user] ffmpeg png not working Message-ID: Hi I have a tmp folder which have two files Img000.jpg, Img001.jpg and Img002.png, the below argument is not working. ffmpeg.exe -r 0.2 -f image2 -i tmp\Img%%03d.jpg -r 24 -sameq -vframes 375 -f swf -y -s 640x360 MySlides.swf I am getting error: [mjpeg @ 01A7F260] mjpeg: unsupported coding type (c6) [mjpeg @ 01A7F260] mjpeg: unsupported coding type (c8) [mjpeg @ 01A7F260] mjpeg: unsupported coding type (ca) Can anyone help me out. Thanks CodeNimo1 From cehoyos at ag.or.at Sat Nov 26 15:48:32 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 26 Nov 2011 14:48:32 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg png not working References: Message-ID: hazer gmail.com> writes: > ffmpeg.exe -r 0.2 -f image2 -i tmp\Img%%03d.jpg -r 24 -sameq -vframes 375 > -f swf -y -s 640x360 MySlides.swf > > I am getting error: > > [mjpeg @ 01A7F260] mjpeg: unsupported coding type (c6) (Complete, uncut console output missing.) Please provide a sample. Carl Eugen From de.techno at gmail.com Sat Nov 26 17:02:59 2011 From: de.techno at gmail.com (dE .) Date: Sat, 26 Nov 2011 21:32:59 +0530 Subject: [FFmpeg-user] Solution to xvid: invalid pixel aspect ratio. Message-ID: <4ED10DB3.1080206@gmail.com> Hi. Many times when converting videos, for a specific aspect ratio, xvid pops up with this error(xvid: invalid pixel aspect ratio). What I've not seen so far, is exactly which aspect ratio is considered invalid by xvid? I know the mpeg4 codec, but I prefer xvid for better compression and better multi threading with the scale filter on and ability to do more than 16 threads and still consume less memory. Thanks! From joseluis at eserre.com Sat Nov 26 19:10:54 2011 From: joseluis at eserre.com (Jose Luis Rivas) Date: Sat, 26 Nov 2011 13:40:54 -0430 Subject: [FFmpeg-user] Overwriting JPEG at output In-Reply-To: <4ED123FF.9040605@rivco.net> References: <4ED123FF.9040605@rivco.net> Message-ID: <4ED12BAE.60405@eserre.com> Hi, I'm trying to get an AVI trough RTSP (or any input, really, be it a static video, by e.g.) to output a still JPEG, and this been overwritten. This is a line of what I'm trying and what I've already tried: snip:start 8< ------ ffmpeg -i test.webm -r 1 -f image2 -s qvga what.jpg ffmpeg -loop_output 1 -i test.webm -r 1 -f image2 -s qvga what.jpg ffmpeg -loop_output 1 -i test.webm -r 1 -f image2 -s qvga -vframes 1 what.jpg ------ >8 snip:end None of them work for what I want, not even using '-y' flag. I also tried using the '%d', and I was suggested on IRC to make it rewrite if it were working on a limit of digits, but it doesn't, it just goes to infinite. Someone could help me out here? I need it to write a single JPEG file, and not an mjpeg. I want to pass it on base64 and need only a JPEG each time. Regards, From raster at rastersoft.com Sat Nov 26 21:09:23 2011 From: raster at rastersoft.com (rastersoft) Date: Sat, 26 Nov 2011 21:09:23 +0100 Subject: [FFmpeg-user] keep all audio tracks when transcoding videos Message-ID: <4ED14773.8000906@rastersoft.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all: I have several videos with two audio tracks, and I want to create video DVDs from them. Is it possible to keep all the audio tracks in the final .mpg file? How? Thanks. - -- Nos leemos RASTER (Linux user #228804) raster at rastersoft.com http://www.rastersoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk7RR3MACgkQXEZvyfy1ha/d0wCgkBt7GbNN43wjJ9URyBtZ5E+z hlEAoJdsgW5z3YVyQf09SGAvD9QNtLYO =XV23 -----END PGP SIGNATURE----- From raster at rastersoft.com Sat Nov 26 21:30:14 2011 From: raster at rastersoft.com (rastersoft) Date: Sat, 26 Nov 2011 21:30:14 +0100 Subject: [FFmpeg-user] keep all audio tracks when transcoding videos In-Reply-To: <4ED14773.8000906@rastersoft.com> References: <4ED14773.8000906@rastersoft.com> Message-ID: <4ED14C56.4000802@rastersoft.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Just to be more specific: I already know how to use "-newaudio"; the problem is how to assign IDs to the new audio tracks (for DVDs, the IDs seems to must be from 128 up; instead, FFMpeg asigns ID 128 to the first audio track, and ID 0 to the second audio track). El 26/11/11 21:09, rastersoft escribi?: > > Hi all: > > I have several videos with two audio tracks, and I want to create video > DVDs from them. Is it possible to keep all the audio tracks in the final > .mpg file? How? > > Thanks. > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > - -- Nos leemos RASTER (Linux user #228804) raster at rastersoft.com http://www.rastersoft.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk7RTFUACgkQXEZvyfy1ha8m1QCfel4kpbjVev23QLwIMPvrYjDj VtkAoNgEZluJTMFhHEyZuC8hagrT2NXw =7XhD -----END PGP SIGNATURE----- From jono at redowl.ca Sat Nov 26 22:10:26 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Sat, 26 Nov 2011 16:10:26 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy Message-ID: <4ED155C2.7050807@redowl.ca> Hello, I'm trying to fix a video with an out-of-sync audio track. You can see the file in its current state at youtube - http://www.youtube.com/watch?v=F1jv6V5aYGk Here's the command line I've tried: ffmpeg -g 1 -i F1jv6V5aYGk.mp4 -itsoffset 0.5 -i F1jv6V5aYGk.mp4 -map 1:0 -map 0:1 -vcodec copy -acodec copy fixed.mp4 Unfortunately, the itsoffset command is completely ignored in this case - the 'fixed' video has the same out-of-sync audio as the original. However, if I re-encode the video by removing '-vcodec copy' from the commandline, the itsoffset works as expected. Of course, I don't want to re-encode the video, since it reduces the quality too much. I've also tried adding -async 1, but did not see any effect. Is this a bug in ffmpeg? Or am I misunderstanding how this command should work? Thanks! -- Jon-o Addleman - http://www.redowl.ca From cehoyos at ag.or.at Sun Nov 27 03:18:03 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 02:18:03 +0000 (UTC) Subject: [FFmpeg-user] -istoffset and -vcodec copy References: <4ED155C2.7050807@redowl.ca> Message-ID: Jonathan Addleman redowl.ca> writes: > However, if I re-encode the video by removing '-vcodec copy' from the > commandline, the itsoffset works as expected. Of course, I don't want to > re-encode the video, since it reduces the quality too much. You do realize that can re-encode both with no visible and with absolutely no quality loss? Carl Eugen From cehoyos at ag.or.at Sun Nov 27 03:18:57 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 02:18:57 +0000 (UTC) Subject: [FFmpeg-user] keep all audio tracks when transcoding videos References: <4ED14773.8000906@rastersoft.com> Message-ID: rastersoft rastersoft.com> writes: > I have several videos with two audio tracks, and I want to create video > DVDs from them. Is it possible to keep all the audio tracks in the final > .mpg file? How? (Command line and complete, uncut console output missing.) Use -map 0 Carl Eugen From jono at redowl.ca Sun Nov 27 04:10:01 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Sat, 26 Nov 2011 22:10:01 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: References: <4ED155C2.7050807@redowl.ca> Message-ID: <4ED1AA09.5050507@redowl.ca> On 11-11-26 09:18 PM, Carl Eugen Hoyos wrote: > Jonathan Addleman redowl.ca> writes: > >> However, if I re-encode the video by removing '-vcodec copy' from the >> commandline, the itsoffset works as expected. Of course, I don't want to >> re-encode the video, since it reduces the quality too much. > > You do realize that can re-encode both with no visible and with absolutely no > quality loss? Hmm.. apparently not... everything I've tried was very visible. But I'm new to this! I must be doing *something* wrong, since even when I tried the various -vpre lossless presets, it still looked horrible... There must be something I'm missing... -- Jon-o Addleman - http://www.redowl.ca From cehoyos at ag.or.at Sun Nov 27 04:16:47 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 03:16:47 +0000 (UTC) Subject: [FFmpeg-user] -istoffset and -vcodec copy References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> Message-ID: Jonathan Addleman redowl.ca> writes: > >> However, if I re-encode the video by removing '-vcodec copy' from the > >> commandline, the itsoffset works as expected. Of course, I don't want to > >> re-encode the video, since it reduces the quality too much. > > > > You do realize that can re-encode both with no visible and with absolutely no > > quality loss? What I tried to write was: 1: You can re-encode with no visible quality loss (often without significantly increasing the file size) 2: You can re-encode with absolutely no quality loss (identical output), that usually means increased file size. > Hmm.. apparently not... everything I've tried was very visible. But I'm > new to this! I must be doing *something* wrong, since even when I tried > the various -vpre lossless presets, it still looked horrible... Command line and complete, uncut console output missing. Carl Eugen From peace at AleksandrSolzhenitsyn.net Sun Nov 27 06:22:07 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Sun, 27 Nov 2011 00:22:07 -0500 Subject: [FFmpeg-user] Delete certain metadata? Message-ID: <4ED1C8FF.9000402@AleksandrSolzhenitsyn.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Using the program MediaInfo a video I looked at had the following information- gsst : 0 gstd : 598731 gssd : BCDC235A3MH1922217440563422 gshh : o-o.preferred.iad07s12.v8.lscache8.c.youtube.com Title : IsoMedia File Produced by Google, 5-11-2011 Encoded date : UTC 2011-09-11 20:34:32 Tagged date : UTC 2011-09-11 20:34:34 Is there a way to delete this information using ffmpeg? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJO0cj3AAoJEPBpZNn4grcjDzMH/As4LsDyJeuec5ft47Ww6xOP cOy2ZDjiKuVevtUPkfIwA4kRogL5LXrjwZlmTUMlmfweFCIpNY3RdcKk0XZEIVrv ANZSWtWy2XsqXtB7tcYVuEAR5BdNp5brcEbLxoe9TbW0qAwxV4ou5GL574z9zMF/ nlhwrWtVM/7wh4VyAiF6+9ldY8N5iOPBkhbVpkAtGL12tMF71ZqoRGg4hMpa4Ri7 LZcp4+8DpnZiA7HrKLbEA6GLzI6POAUJghwY+ZoKY+3i0enq0tyiFZY6+La4uLo+ 0ymMYStckNvad6heJaPCboKUPnmpzYhgOrjPbU/x7PvTrg8ZNpYg0nuvU7rcE0s= =s+Hv -----END PGP SIGNATURE----- From ffmpeg-user at herveybayaustralia.com.au Sun Nov 27 12:50:19 2011 From: ffmpeg-user at herveybayaustralia.com.au (Da Rock) Date: Sun, 27 Nov 2011 21:50:19 +1000 Subject: [FFmpeg-user] ffserver network issues Message-ID: <4ED223FB.7080000@herveybayaustralia.com.au> I got back to an old project I've been fiddling with for some time now, but ffserver doesn't appear to be doing so well now. I've run some updates, new platform, and new platform versions and I've done ok until now. When I first kicked ffserver off it refused to bind to the network at all, after some retries, searching through confs, it worked- not sure what the failure was after all that. When I started sending data to the server using ffmpeg I get the following error: av_interleaved_write_frame(): Connection reset by peer My data is gstreamer pipelined dvb stream with ffmpeg splitting and processing to mpegts output, although I have another method which essentially has the same result anyway- gstreamer drops the pipe when the buffers fill because ffmpeg doesn't take the data from the pipe. Cmd is as follows: gst-launch dvbsrc adapter=1 frequency=xxxxxxxx modulation="QAM 64" bandwidth=7 guard=16 ! queue ! fdsink | ffmpeg -i - -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1601 http://127.0.0.1:8090/stream1.ffm -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1605 http://127.0.0.1:8090/stream2.ffm -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1608 http://127.0.0.1:8090/stream3.ffm If I output to file it all goes beautifully. Does anyone have any clues on this? Any help would be much appreciated. I also could use some help as to where the logs are going as well? Cheers My platform is this: FreeBSD 8.1-RELEASE-p1 FreeBSD 8.1-RELEASE-p1 ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 26 2011 12:45:06 with gcc 4.2.1 20070719 [FreeBSD] configuration: --prefix=/usr/local --mandir=/usr/local/man --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-memalign-hack --enable-runtime-cpudetect --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 --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libcelt --enable-libdirac --enable-libfaac --enable-libfreetype --enable-frei0r --enable-libgsm --enable-libmp3lame --enable-libopencv --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-ffplay --enable-libspeex --enable-libtheora --disable-vaapi --disable-vdpau --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-nonfree --enable-version3 libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 ffserver version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 26 2011 12:45:06 with gcc 4.2.1 20070719 [FreeBSD] configuration: --prefix=/usr/local --mandir=/usr/local/man --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-memalign-hack --enable-runtime-cpudetect --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 --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libcelt --enable-libdirac --enable-libfaac --enable-libfreetype --enable-frei0r --enable-libgsm --enable-libmp3lame --enable-libopencv --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-ffplay --enable-libspeex --enable-libtheora --disable-vaapi --disable-vdpau --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-nonfree --enable-version3 libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Config: # Port on which the server is listening. You must select a different # port from your standard HTTP web server if it is running on the same # computer. Port 8090 # Address on which the server is bound. Only useful if you have # several network interfaces. BindAddress 0.0.0.0 # Number of simultaneous HTTP connections that can be handled. It has # to be defined *before* the MaxClients parameter, since it defines the # MaxClients maximum limit. MaxHTTPConnections 2000 # Number of simultaneous requests that can be handled. Since FFServer # is very fast, it is more likely that you will want to leave this high # and use MaxBandwidth, below. MaxClients 1000 # This the maximum amount of kbit/sec that you are prepared to # consume when streaming to clients. MaxBandwidth 500000 # Access log file (uses standard Apache log file format) # '-' is the standard output. CustomLog - # Suppress that if you want to launch ffserver as a daemon. NoDaemon ################################################################## # Definition of the live feeds. Each live feed contains one video # and/or audio sequence coming from an ffmpeg encoder or another # ffserver. This sequence may be encoded simultaneously with several # codecs at several resolutions. # You must use 'ffmpeg' to send a live feed to ffserver. In this # example, you can type: # # ffmpeg http://localhost:8090/feed1.ffm # ffserver can also do time shifting. It means that it can stream any # previously recorded live stream. The request should contain: # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify # a path where the feed is stored on disk. You also specify the # maximum size of the feed, where zero means unlimited. Default: # File=/tmp/feed_name.ffm FileMaxSize=5M File /tmp/feed1.ffm FileMaxSize 512M # You could specify # ReadOnlyFile /saved/specialvideo.ffm # This marks the file as readonly and it will not be deleted or updated. # Specify launch in order to start ffmpeg automatically. # First ffmpeg must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field #Launch ffmpeg # Only allow connections from localhost to the feed. ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 2.25G File /home/share/stream2.ffm FileMaxSize 2304M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 2.25G File /home/share/stream3.ffm FileMaxSize 2304M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 6.5G File /home/share/stream1.ffm FileMaxSize 6656M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 ################################################################## # Now you can define each stream which will be generated from the # original audio and video stream. Each format has a filename (here # 'test1.mpg'). FFServer will send this stream when answering a # request containing this filename. # coming from live feed 'feed1' Feed feed1.ffm # Format of the stream : you can choose among: # mpeg : MPEG-1 multiplexed video and audio # mpegvideo : only MPEG-1 video # mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) # ogg : Ogg format (Vorbis audio codec) # rm : RealNetworks-compatible stream. Multiplexed audio and video. # ra : RealNetworks-compatible stream. Audio only. # mpjpeg : Multipart JPEG (works with Netscape without any plugin) # jpeg : Generate a single JPEG image. # asf : ASF compatible streaming (Windows Media Player format). # swf : Macromedia Flash compatible stream # avi : AVI format (MPEG-4 video, MPEG audio sound) Format mpeg # Bitrate for the audio stream. Codecs usually support only a few # different bitrates. AudioBitRate 192 # Number of audio channels: 1 = mono, 2 = stereo AudioChannels 2 # Sampling frequency for audio. When using low bitrates, you should # lower this frequency to 22050 or 11025. The supported frequencies # depend on the selected audio codec. AudioSampleRate 44100 # Bitrate for the video stream VideoBitRate 1000 # Ratecontrol buffer size VideoBufferSize 40 # Number of frames per second VideoFrameRate 25 # Size of the video frame: WxH (default: 160x128) # The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga, # qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga, # wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720, # hd1080 VideoSize 160x128 # Transmit only intra frames (useful for low bitrates, but kills frame rate). #VideoIntraOnly # If non-intra only, an intra frame is transmitted every VideoGopSize # frames. Video synchronization can only begin at an intra frame. VideoGopSize 12 # More MPEG-4 parameters # VideoHighQuality # Video4MotionVector # Choose your codecs: #AudioCodec mp2 #VideoCodec mpeg1video # Suppress audio #NoAudio # Suppress video #NoVideo #VideoQMin 3 #VideoQMax 31 # Set this to the number of seconds backwards in time to start. Note that # most players will buffer 5-10 seconds of video, and also you need to allow # for a keyframe to appear in the data stream. #Preroll 15 # ACL: # You can allow ranges of addresses (or single addresses) #ACL ALLOW # You can deny ranges of addresses (or single addresses) #ACL DENY # You can repeat the ACL allow/deny as often as you like. It is on a per # stream basis. The first match defines the action. If there are no matches, # then the default is the inverse of the last ACL statement. # # Thus 'ACL allow localhost' only allows access from localhost. # 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and # allow everybody else. ################################################################## # Example streams # Multipart JPEG # #Feed feed1.ffm #Format mpjpeg #VideoFrameRate 2 #VideoIntraOnly #NoAudio #Strict -1 # # Single JPEG # #Feed feed1.ffm #Format jpeg #VideoFrameRate 2 #VideoIntraOnly ##VideoSize 352x240 #NoAudio #Strict -1 # # Flash Feed feed1.ffm Format swf VideoFrameRate 25 #VideoIntraOnly VideoCodec libx264 AudioCodec libmp3lame VideoSize hd720 VideoBitRate 1200 #NoAudio ACL allow 192.168.0.0 255.255.255.0 # ASF compatible # #Feed feed1.ffm #Format asf #VideoFrameRate 15 #VideoSize 352x240 #VideoBitRate 256 #VideoBufferSize 40 #VideoGopSize 30 #AudioBitRate 64 #StartSendOnKey # # MP3 audio # #Feed feed1.ffm #Format mp2 #AudioCodec mp3 #AudioBitRate 64 #AudioChannels 1 #AudioSampleRate 44100 #NoVideo # # Ogg Vorbis audio # #Feed feed1.ffm #Title "Stream title" #AudioBitRate 64 #AudioChannels 2 #AudioSampleRate 44100 #NoVideo # # Real with audio only at 32 kbits # #Feed feed1.ffm #Format rm #AudioBitRate 32 #NoVideo #NoAudio # # Real with audio and video at 64 kbits # #Feed feed1.ffm #Format rm #AudioBitRate 32 #VideoBitRate 128 #VideoFrameRate 25 #VideoGopSize 25 #NoAudio # ################################################################## # Local Streams Feed ten.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 Feed eleven.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 Feed one.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 ################################################################## # A stream coming from a file: you only need to set the input # filename and optionally a new format. Supported conversions: # AVI -> ASF # #File "/usr/local/httpd/htdocs/tlive.rm" #NoAudio # # #File "/usr/local/httpd/htdocs/test.asf" #NoAudio #Author "Me" #Copyright "Super MegaCorp" #Title "Test stream from disk" #Comment "Test comment" # ################################################################## # RTSP examples # # You can access this stream with the RTSP URL: # rtsp://localhost:5454/test1-rtsp.mpg # # A non-standard RTSP redirector is also created. Its URL is: # http://localhost:8090/test1-rtsp.rtsp # #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" # ################################################################## # SDP/multicast examples # # If you want to send your stream in multicast, you must set the # multicast address with MulticastAddress. The port and the TTL can # also be set. # # An SDP file is automatically generated by ffserver by adding the # 'sdp' extension to the stream name (here # http://localhost:8090/test1-sdp.sdp). You should usually give this # file to your player to play the stream. # # The 'NoLoop' option can be used to avoid looping when the stream is # terminated. # #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" #MulticastAddress 224.124.0.1 #MulticastPort 5000 #MulticastTTL 16 #NoLoop # ################################################################## # Special streams # Server status Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico # Redirect index.html to the appropriate site URL http:/// From mail2ashi.86 at gmail.com Sun Nov 27 15:52:02 2011 From: mail2ashi.86 at gmail.com (Ashish Mathur) Date: Sun, 27 Nov 2011 20:22:02 +0530 Subject: [FFmpeg-user] FFMPEG 0.8.5 : AC3 (a52) codec unrecognized while receving RTSP stream In-Reply-To: References: Message-ID: Hi Carl, It seems to be a bug with FFMPEG 0.8.5. I can't play any AC3 stream through rtsp demuxer. Can someone look into this issue? Regards, Ashish On Fri, Nov 25, 2011 at 4:05 PM, Carl Eugen Hoyos wrote: > Ashish Mathur gmail.com> writes: > > > Is this a known issue with FFMPEG 0.8.5? > > > Or is there a fix to it in the latest FFMPEG versions? > > Please test and report back. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lazarusportugalwebmasters at gmail.com Sun Nov 27 16:29:50 2011 From: lazarusportugalwebmasters at gmail.com (Lazarus Portugal Lazarus Portugal) Date: Sun, 27 Nov 2011 15:29:50 +0000 Subject: [FFmpeg-user] Really completly front-end for ffmpeg editing video and audio. Message-ID: Hello. I try to fou really completly front-end to ffmpeg, for video and audio creation and editon and I can't found anything. It needs be opensource because is for i study and include video recording. This applications need to be for windows, and have video upload directlyfor youtube Anyone can give me one application, or list of applications for this? From cehoyos at ag.or.at Sun Nov 27 16:56:36 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 15:56:36 +0000 (UTC) Subject: [FFmpeg-user] Overwriting JPEG at output References: <4ED123FF.9040605@rivco.net> <4ED12BAE.60405@eserre.com> Message-ID: Jose Luis Rivas eserre.com> writes: > I'm trying to get an AVI trough RTSP (or any input, really, be it a > static video, by e.g.) to output a still JPEG, and this been > overwritten. This is a line of what I'm trying and what I've already tried: > ffmpeg -i test.webm -r 1 -f image2 -s qvga what.jpg This works now with -updatefirst 1 what.jpg Carl Eugen From cehoyos at ag.or.at Sun Nov 27 17:13:03 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 16:13:03 +0000 (UTC) Subject: [FFmpeg-user] FFMPEG 0.8.5 : AC3 (a52) codec unrecognized while receving RTSP stream References: Message-ID: Ashish Mathur gmail.com> writes: > It seems to be a bug with FFMPEG 0.8.5. I can't play any AC3 stream through > rtsp demuxer. > > Can someone look into this issue? I am not sure I understand correctly: Do you mean it works fine with current git head, but does not work with 0.8.7 (unfortunately, 0.8.5 is not a supported version anymore)? If this is the case, you should be able to find the commit that introduced the fix, and I can backport it to the next 0.8 release. Carl Eugen From lou at lrcd.com Sun Nov 27 19:51:54 2011 From: lou at lrcd.com (Lou) Date: Sun, 27 Nov 2011 09:51:54 -0900 Subject: [FFmpeg-user] Delete certain metadata? In-Reply-To: <4ED1C8FF.9000402@AleksandrSolzhenitsyn.net> References: <4ED1C8FF.9000402@AleksandrSolzhenitsyn.net> Message-ID: <20111127095154.0be5ed4c@lrcd.com> On Sun, 27 Nov 2011 00:22:07 -0500 "." wrote: > Using the program MediaInfo a video I looked at had the following > information- > > gsst : 0 > gstd : 598731 > gssd : BCDC235A3MH1922217440563422 > gshh : > o-o.preferred.iad07s12.v8.lscache8.c.youtube.com > > > Title : IsoMedia File Produced by > Google, 5-11-2011 > Encoded date : UTC 2011-09-11 20:34:32 > Tagged date : UTC 2011-09-11 20:34:34 > > > Is there a way to delete this information using ffmpeg? One method is to remove all non-ffmpeg metadata: ffmpeg -i input -c copy -map_metadata -1 output From jono at redowl.ca Sun Nov 27 22:34:33 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Sun, 27 Nov 2011 16:34:33 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> Message-ID: <4ED2ACE9.8020800@redowl.ca> On 11-11-26 10:16 PM, Carl Eugen Hoyos wrote: > Jonathan Addleman redowl.ca> writes: > >> Hmm.. apparently not... everything I've tried was very visible. But I'm >> new to this! I must be doing *something* wrong, since even when I tried >> the various -vpre lossless presets, it still looked horrible... > > Command line and complete, uncut console output missing. Hmm.. Maybe its just a problem with one of the other lossless presets - I just tried the lossless_max, and is sort of worked: ffmpeg -i F1jv6V5aYGk.mp4 -itsoffset 0.5 -i F1jv6V5aYGk.mp4 -map 0:0 -map 1:1 -vcodec libx264 -vpre lossless_max -acodec copy fixed.mp4 The output looks the same as the original, though the file is over 10x as big, and my computer can't play it back at full speed. With the 'veryslow' preset, the results are abysmal - nasty artifacts everywhere. Utterly unusable. ffmpeg -i short.mp4 -itsoffset 0.5 -i short.mp4 -map 0:0 -map 1:1 -vcodec libx264 -vpre veryslow -acodec copy fixed.mp4 The sound is at least in sync... Here's the console output from that command: scaph:~/bla$ ffmpeg -i short.mp4 -itsoffset 0.5 -i short.mp4 -map 0:0 -map 1:1 -vcodec libx264 -vpre veryslow -acodec copy fixed.mp4 ffmpeg version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav developers built on Oct 2 2011 15:12:32 with gcc 4.6.1 configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch avutil configuration: --extra-version='4:0.7.2.1ubuntu1+medibuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avcodec configuration: --extra-version='4:0.7.2.1ubuntu1+medibuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdirac --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libopencore-amrnb --enable-version3 --enable-libopencore-amrwb --enable-version3 --enable-libvo-aacenc --enable-version3 --enable-libvo-amrwbenc --enable-version3 --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avformat configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avdevice configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay avfilter configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay swscale configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay postproc configuration: --extra-version='4:0.7.2-1ubuntu1' --arch=i386 --prefix=/usr --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --enable-runtime-cpudetect --enable-vaapi --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavutil 51. 7. 0 / 51. 7. 0 libavcodec 53. 5. 0 / 53. 5. 0 libavformat 53. 2. 0 / 53. 2. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 4. 0 / 2. 4. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 52. 0. 0 / 52. 0. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'short.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 Duration: 00:03:15.06, start: 0.000000, bitrate: 430 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080, 5958 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'short.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 Duration: 00:03:15.06, start: 0.000000, bitrate: 430 kb/s Stream #1.0(und): Video: h264 (High), yuv420p, 1920x1080, 5958 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #1.1(und): Audio: aac, 44100 Hz, stereo, s16, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 File 'fixed.mp4' already exists. Overwrite ? [y/N] y [buffer @ 0x87dfec0] w:1920 h:1080 pixfmt:yuv420p [libx264 @ 0x87dfae0] using cpu capabilities: MMX2 Cache64 [libx264 @ 0x87dfae0] profile High, level 5.1 [libx264 @ 0x87dfae0] 264 - core 116 r2042 178455c - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 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=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=200 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.41 aq=1:1.00 Output #0, mp4, to 'fixed.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 encoder : Lavf53.2.0 Stream #0.0(und): Video: libx264, yuv420p, 1920x1080, q=0-69, 200 kb/s, 30k tbn, 29.97 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: libfaac, 44100 Hz, stereo, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 Stream mapping: Stream #0.0 -> #0.0 Stream #1.1 -> #0.1 Press ctrl-c to stop encoding [h264 @ 0x87e7100] AVC: nal size 4805564kB time=11.65 bitrate= 326.1kbits/s its/s [h264 @ 0x87e7100] no frame! Error while decoding stream #0.0 frame= 413 fps= 1 q=46.0 Lsize= 574kB time=13.71 bitrate= 343.0kbits/s video:298kB audio:262kB global headers:0kB muxing overhead 2.381960% frame I:2 Avg QP:43.47 size: 21430 [libx264 @ 0x87dfae0] frame P:383 Avg QP:47.08 size: 669 [libx264 @ 0x87dfae0] frame B:28 Avg QP:48.66 size: 208 [libx264 @ 0x87dfae0] consecutive B-frames: 89.6% 3.9% 1.5% 3.9% 1.2% 0.0% 0.0% 0.0% 0.0% [libx264 @ 0x87dfae0] mb I I16..4: 42.2% 55.2% 2.6% [libx264 @ 0x87dfae0] mb P I16..4: 0.6% 0.2% 0.0% P16..4: 3.7% 0.1% 0.0% 0.0% 0.0% skip:95.4% [libx264 @ 0x87dfae0] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 2.1% 0.0% 0.0% direct: 0.0% skip:97.8% L0: 5.8% L1:94.2% BI: 0.0% [libx264 @ 0x87dfae0] final ratefactor: 42.30 [libx264 @ 0x87dfae0] 8x8 transform intra:36.4% inter:91.5% [libx264 @ 0x87dfae0] direct mvs spatial:89.3% temporal:10.7% [libx264 @ 0x87dfae0] coded y,uvDC,uvAC intra: 11.1% 10.7% 0.0% inter: 0.1% 0.1% 0.0% [libx264 @ 0x87dfae0] i16 v,h,dc,p: 55% 33% 3% 9% [libx264 @ 0x87dfae0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 11% 46% 3% 4% 4% 5% 4% 3% [libx264 @ 0x87dfae0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 13% 5% 6% 6% 9% 9% 7% 4% [libx264 @ 0x87dfae0] i8c dc,h,v,p: 84% 12% 3% 1% [libx264 @ 0x87dfae0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x87dfae0] ref P L0: 73.8% 6.8% 10.3% 1.7% 1.6% 1.5% 1.6% 0.2% 0.3% 0.3% 0.3% 0.3% 0.3% 0.4% 0.4% 0.1% [libx264 @ 0x87dfae0] ref B L0: 93.0% 3.5% 0.3% 0.0% 1.0% 0.7% 0.7% 0.0% 0.3% 0.3% [libx264 @ 0x87dfae0] ref B L1: 90.2% 9.8% [libx264 @ 0x87dfae0] kb/s:176.99 -- Jon-o Addleman - http://www.redowl.ca From joseluis at eserre.com Sun Nov 27 22:37:57 2011 From: joseluis at eserre.com (Jose Luis Rivas) Date: Sun, 27 Nov 2011 17:07:57 -0430 Subject: [FFmpeg-user] Overwriting JPEG at output In-Reply-To: References: <4ED123FF.9040605@rivco.net> <4ED12BAE.60405@eserre.com> Message-ID: <4ED2ADB5.3070604@eserre.com> On 11/27/2011 11:26 AM, Carl Eugen Hoyos wrote: >> ffmpeg -i test.webm -r 1 -f image2 -s qvga what.jpg > > This works now with -updatefirst 1 what.jpg > > Carl Eugen /usr/bin/ffmpeg -i shell-20110908-1.webm -r 1 -f image2 -updatefirst 1 what.jpg Unrecognized option 'updatefirst' :( Just compiled git master branch, is there any library required to get updatefirst recognized? ghostbar at chaos:~$ /usr/bin/ffmpeg -version ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 22 2011 07:59:05 with gcc 4.6.2 configuration: --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --ena ble-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libdirac - -disable-decoder=libdirac --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-lib theora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 -- enable-libspeex --enable-nonfree --disable-stripping --enable-libschroedinger -- disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable- libvpx --enable-librtmp --enable-avfilter --enable-frei0r --enable-libopencv --e nable-libfreetype --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-a mrwbenc --enable-libaacplus --libdir=/usr/lib/i386-linux-gnu --enable-libdc1394 --disable-altivec --disable-armv5te --disable-armv6 --disable-vis --shlibdir=/us r/lib/i386-linux-gnu libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 ffmpeg 0.8.7 libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 -- Jose Luis Rivas, Web Builder Eserre -- http://www.eserre.com/ Venezuela - +58 (424) 781 2565 GPGs: 7C4DF50D B9AC8C43 From cehoyos at ag.or.at Mon Nov 28 00:04:41 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 23:04:41 +0000 (UTC) Subject: [FFmpeg-user] -istoffset and -vcodec copy References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> <4ED2ACE9.8020800@redowl.ca> Message-ID: Jonathan Addleman redowl.ca> writes: > With the 'veryslow' preset, the results are abysmal - nasty artifacts > everywhere. Utterly unusable. You have to either define a low constant quantiser or a target bitrate. [...] > version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav developers This is known to be severely broken, please see http://ffmpeg.org/download.html to know how to get supported FFmpeg versions, preferable current git head. Carl Eugen From cehoyos at ag.or.at Mon Nov 28 00:07:07 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 27 Nov 2011 23:07:07 +0000 (UTC) Subject: [FFmpeg-user] Overwriting JPEG at output References: <4ED123FF.9040605@rivco.net> <4ED12BAE.60405@eserre.com> <4ED2ADB5.3070604@eserre.com> Message-ID: Jose Luis Rivas eserre.com> writes: > /usr/bin/ffmpeg -i shell-20110908-1.webm -r 1 -f image2 -updatefirst 1 > what.jpg This is the correct syntax. > Unrecognized option 'updatefirst' > > :( > > Just compiled git master branch, is there any library required to get > updatefirst recognized? > > ghostbar chaos:~$ /usr/bin/ffmpeg -version > ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers This does not look like git master... Carl Eugen From jono at redowl.ca Mon Nov 28 01:10:32 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Sun, 27 Nov 2011 19:10:32 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> <4ED2ACE9.8020800@redowl.ca> Message-ID: <4ED2D178.1020308@redowl.ca> On 11-11-27 06:04 PM, Carl Eugen Hoyos wrote: > Jonathan Addleman redowl.ca> writes: > >> With the 'veryslow' preset, the results are abysmal - nasty artifacts >> everywhere. Utterly unusable. > > You have to either define a low constant quantiser or a target bitrate. Even with -qscale 1, it looked awful. (I assume that's what you mean?) > [...] > >> version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2000-2011 the Libav developers > > This is known to be severely broken, please see http://ffmpeg.org/download.html > to know how to get supported FFmpeg versions, preferable current git head. I feared it might be something like that.. I'll try getting/compiling an updated version in the next few days and will let you know if it helps! thanks for your help. -- Jon-o Addleman - http://www.redowl.ca From cehoyos at ag.or.at Mon Nov 28 11:16:51 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 28 Nov 2011 10:16:51 +0000 (UTC) Subject: [FFmpeg-user] -istoffset and -vcodec copy References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> <4ED2ACE9.8020800@redowl.ca> <4ED2D178.1020308@redowl.ca> Message-ID: Jonathan Addleman redowl.ca> writes: > >> With the 'veryslow' preset, the results are abysmal - nasty artifacts > >> everywhere. Utterly unusable. > > > > You have to either define a low constant quantiser or a target bitrate. > > Even with -qscale 1, it looked awful. (I assume that's what you mean?) (Command line and complete, uncut output missing.) -qscale 2 works fine with -vcodec mpeg2video and -vcodec mpeg4, I suspect you have to use a x264 option for -vcodec libx264. Carl Eugen From hcano at mebcn.com Mon Nov 28 11:50:59 2011 From: hcano at mebcn.com (Hector Cano) Date: Mon, 28 Nov 2011 11:50:59 +0100 Subject: [FFmpeg-user] Recording a live stream changing output file every x seconds Message-ID: <4ED36793.7030009@mebcn.com> Hi, I am recording live a stream from an Axis camera. It works perfect. I'm required to record for maybe a couple of hours at a time, and I'd prefer to have several shorter files, maybe a new file every minute, instead of a big file (to minimize risk, and to be able to start post processing without having to wait till the end). Is there any option for ffmepg to do the job automatically? I've tried using -t 60, and restarting the capture, but the jump is too noticeable. I am using an Ubuntu 11.10 server for the job, launching ffmpeg from a bash shell. I tried to find the answer in the archives, but found nothing but an old unanswered question "Recording and splitting at the same time". Any help would be much appreciated. Thank you, -- Hector // From iliana.in.france at free.fr Mon Nov 28 14:24:20 2011 From: iliana.in.france at free.fr (iliana.in.france at free.fr) Date: Mon, 28 Nov 2011 14:24:20 +0100 Subject: [FFmpeg-user] Timecode problem when capturing capturing live multicast/udp mpegvideo stream Message-ID: <1322486660.4ed38b84ab465@webmail.free.fr> Hello, I have a problem when capturing a live multicast udp mpeg2 stream using ffmpeg which concerns timestamps and I hope some of this forum of experts will be able to help me. The simple command: ffmpeg.exe -y -i udp://225.1.1.53:1111 -vcodec copy -acodec copy output.mpg Gives the error (see details at the bottom of the message): [mpeg @ 01cb9760] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 9049 >= 9049 av_interleaved_write_frame(): Invalid argument the number varies, but it is always XXX >= XXX, and the capture aborts. I manage to somewhat solving the problem by adding the option: -dts_delta_threshold 0 But in this case, the audio and video recordings go out of sync in the recorded file by approximately 1 or 2 seconds every hour (99.95% stretching of the audio) I have also tried to add the option ?copyts (this proved to be effective in fixing video/audio synchronization issues in file-to-file re-encoding) but in this case (live mpeg-ts multicast) I get the same error [mpeg @ 021e7ac0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1130 >= 1130 av_interleaved_write_frame(): Invalid argument Can someone enlight me on how I can record this multicast live transmission with audio and video in sync ? (Worth mentioning that when watching the live video with VLC, audio and video are always in sync even after several hours of watching. Many thanks in advance to anyone who could give me even a simple hint. Ileana. Here is the exact output: > ffmpeg.exe -y -i udp://225.1.1.53:1111 -vcodec copy -acodec copy output.mpg ffmpeg version N-32754-g936d4d4-Sherpya, Copyright (c) 2000-2011 the FFmpeg deve lopers built on Sep 21 2011 13:48:46 with gcc 4.2.5 20090330 (prerelease) [Sherpya] libavutil 51. 16. 1 / 51. 16. 1 libavcodec 53. 16. 0 / 53. 16. 0 libavformat 53. 12. 0 / 53. 12. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 43. 2 / 2. 43. 2 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 01cb7c40] parser not found for codec dvb_teletext, packets or times ma y be invalid. [mpeg2video @ 01cb2320] mpeg_decode_postinit() failure Last message repeated 10 times [mpegts @ 01cb7c40] max_analyze_duration 5000000 reached at 5016000 [mpegts @ 01cb7c40] Estimating duration from bitrate, this may be inaccurate Input #0, mpegts, from 'udp://225.1.1.53:1111': Duration: N/A, start: 16541.257933, bitrate: 15192 kb/s Program 2001 Metadata: service_name : Direct 8 service_provider: TCOAX Stream #0.0[0x42]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p , 720x576 [SAR 64:45 DAR 16:9], 15000 kb/s, 26.96 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x44](fra): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo , s16, 192 kb/s Stream #0.2[0x45](fra): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006) Output #0, mpeg, to 'D:\data\MyMovies\TvRecordings\Recordings\2011-11-28 TEST.mp g': Metadata: encoder : Lavf53.12.0 Stream #0.0: Video: mpeg2video ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SA R 64:45 DAR 16:9], q=2-31, 15000 kb/s, 90k tbn, 25 tbc Stream #0.1(fra): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 (copy) Stream #0.1 -> #0.1 (copy) Press [q] to stop, [?] for help [mpeg @ 01cb9760] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 9049 >= 9049 av_interleaved_write_frame(): Invalid argument From peace at aleksandrsolzhenitsyn.net Mon Nov 28 15:57:25 2011 From: peace at aleksandrsolzhenitsyn.net (.) Date: Mon, 28 Nov 2011 09:57:25 -0500 Subject: [FFmpeg-user] Plays too fast on iPod? Message-ID: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I put a video named 'dotdmeta.mp4' onto my iPod 160 G (Classic). Below is the FFMPEG -i information. The video plays fine until about the 17 minute mark and then playback speeds up to twice its normal speed. Once the video speeds up the audio is out of sync with the video too. However, if I convert the video using the following FFMPEG code line the playback on the iPod is fine (but the conversion takes a long, long time). ffmpeg -i dotdmeta.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf 24 -acodec libfaac -aq 100 -vf scale="640:trunc(ow/a/2)*2" DOD.mp4 Any idea how to easily fix it? Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'dotdmeta.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.20.0 Duration: 00:48:42.44, start: 0.000000, bitrate: 433 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 480x360 [SAR 1:1 DAR 4:3], 331 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 95 kb/s Metadata: creation_time : 1970-01-01 00:00:00 handler_name : -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJO06FRAAoJEPBpZNn4grcjPMAH/0UKLjIdrvK8W/ptjVfoUOW5 NTz2AH8eykcC8kU8bjpMTOg8CteIlvLMN2VuQ5k3y6LOeAfKeMIakCM9fkS8qDVS ueiBlkAkmcOOfX/HzlsKJVUmuc1SfuznAKF2dNcOuP6pfqhaqD8i++sG0IPtgMbe FwHk2gX2HlAhTI4aN3OV+UOYPBmUWGA7Al7wMNiKne5GLaJjPuco2xowAzDYM/ZO Lk+655IJPKRWYmm3Hn6ivjEJQnZdYpQOy2hcNhUFNli0sInTV6eU307qvHljZ0nB etDxk21oWTw3hPoWwdDbeu1iXVIVJJr54UQlvaBocnA788jO9sNV42URTrgsxcU= =5lAM -----END PGP SIGNATURE----- From peace at AleksandrSolzhenitsyn.net Mon Nov 28 15:58:30 2011 From: peace at AleksandrSolzhenitsyn.net (.) Date: Mon, 28 Nov 2011 09:58:30 -0500 Subject: [FFmpeg-user] Delete certain metadata? In-Reply-To: <20111127095154.0be5ed4c@lrcd.com> References: <4ED1C8FF.9000402@AleksandrSolzhenitsyn.net> <20111127095154.0be5ed4c@lrcd.com> Message-ID: <4ED3A196.1020702@AleksandrSolzhenitsyn.net> On 11/27/2011 01:51 PM, Lou wrote: > On Sun, 27 Nov 2011 00:22:07 -0500 > "." wrote: > >> Using the program MediaInfo a video I looked at had the following >> information- >> >> gsst : 0 >> gstd : 598731 >> gssd : BCDC235A3MH1922217440563422 >> gshh : >> o-o.preferred.iad07s12.v8.lscache8.c.youtube.com >> >> >> Title : IsoMedia File Produced by >> Google, 5-11-2011 >> Encoded date : UTC 2011-09-11 20:34:32 >> Tagged date : UTC 2011-09-11 20:34:34 >> >> >> Is there a way to delete this information using ffmpeg? > One method is to remove all non-ffmpeg metadata: > ffmpeg -i input -c copy -map_metadata -1 output Thanks, Lou. It worked perfectly and quickly. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 554 bytes Desc: OpenPGP digital signature URL: From de.techno at gmail.com Mon Nov 28 16:18:27 2011 From: de.techno at gmail.com (dE .) Date: Mon, 28 Nov 2011 20:48:27 +0530 Subject: [FFmpeg-user] Recording a live stream changing output file every x seconds In-Reply-To: <4ED36793.7030009@mebcn.com> References: <4ED36793.7030009@mebcn.com> Message-ID: <4ED3A643.2080200@gmail.com> On 11/28/11 16:20, Hector Cano wrote: > Hi, > I am recording live a stream from an Axis camera. It works perfect. > > I'm required to record for maybe a couple of hours at a time, and I'd > prefer to have several shorter files, maybe a new file every minute, > instead of a big file (to minimize risk, and to be able to start post > processing without having to wait till the end). > > Is there any option for ffmepg to do the job automatically? > I've tried using -t 60, and restarting the capture, but the jump is > too noticeable. > > I am using an Ubuntu 11.10 server for the job, launching ffmpeg from a > bash shell. > > I tried to find the answer in the archives, but found nothing but an > old unanswered question "Recording and splitting at the same time". > > Any help would be much appreciated. > > > Thank you, I think some bash workarounds will do, you may start the other ffmpeg instances before the first one finishes using bash's '&' e.g. - ffmpeg -t 00:01:00 -f x11grab -preset ultrafast -vcodec libx264 mp4.mp4 & sleep 55; ffmpeg -t 00:01:00 -f x11grab -preset ultrafast -vcodec libx264 mp4.mp4 & And repeat this using while loops. I don't understand the post processing part. But as an alternative, for backups, you may try RAID mirroring using mdadm. From de.techno at gmail.com Mon Nov 28 16:27:55 2011 From: de.techno at gmail.com (dE .) Date: Mon, 28 Nov 2011 20:57:55 +0530 Subject: [FFmpeg-user] Plays too fast on iPod? In-Reply-To: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> References: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> Message-ID: <4ED3A87B.3080601@gmail.com> On 11/28/11 20:27, . wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I put a video named 'dotdmeta.mp4' onto my iPod 160 G (Classic). Below > is the FFMPEG -i information. > > The video plays fine until about the 17 minute mark and then playback > speeds up to twice its normal speed. Once the video speeds up the audio > is out of sync with the video too. > > However, if I convert the video using the following FFMPEG code line the > playback on the iPod is fine (but the conversion takes a long, long time). > > ffmpeg -i dotdmeta.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf > 24 -acodec libfaac -aq 100 -vf scale="640:trunc(ow/a/2)*2" DOD.mp4 > > > Any idea how to easily fix it? > > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'dotdmeta.mp4': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > creation_time : 1970-01-01 00:00:00 > encoder : Lavf53.20.0 > > Duration: 00:48:42.44, start: 0.000000, bitrate: 433 kb/s > Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / > 0x31637661), yuv420p, 480x360 [SAR 1:1 DAR 4:3], 331 kb/s, 25 fps, 25 > tbr, 25 tbn, 50 tbc > Metadata: > creation_time : 1970-01-01 00:00:00 > handler_name : VideoHandler > > Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, > 95 kb/s > Metadata: > creation_time : 1970-01-01 00:00:00 > handler_name : > > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.10 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iQEcBAEBAgAGBQJO06FRAAoJEPBpZNn4grcjPMAH/0UKLjIdrvK8W/ptjVfoUOW5 > NTz2AH8eykcC8kU8bjpMTOg8CteIlvLMN2VuQ5k3y6LOeAfKeMIakCM9fkS8qDVS > ueiBlkAkmcOOfX/HzlsKJVUmuc1SfuznAKF2dNcOuP6pfqhaqD8i++sG0IPtgMbe > FwHk2gX2HlAhTI4aN3OV+UOYPBmUWGA7Al7wMNiKne5GLaJjPuco2xowAzDYM/ZO > Lk+655IJPKRWYmm3Hn6ivjEJQnZdYpQOy2hcNhUFNli0sInTV6eU307qvHljZ0nB > etDxk21oWTw3hPoWwdDbeu1iXVIVJJr54UQlvaBocnA788jO9sNV42URTrgsxcU= > =5lAM > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Have you tried any other codecs like xvid? And have you tried playing any other 17+ minute video (like download from youtube?) I've read this's a problem with ipod. From blacktrash at gmx.net Mon Nov 28 16:30:29 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Mon, 28 Nov 2011 15:30:29 +0000 Subject: [FFmpeg-user] Plays too fast on iPod? In-Reply-To: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> References: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> Message-ID: <20111128153029.GK88337@krille.blacktrash.org> * . on Monday, November 28, 2011 at 09:57:25 -0500 > I put a video named 'dotdmeta.mp4' onto my iPod 160 G (Classic). Below > is the FFMPEG -i information. > > The video plays fine until about the 17 minute mark and then playback > speeds up to twice its normal speed. Once the video speeds up the audio > is out of sync with the video too. > > However, if I convert the video using the following FFMPEG code line the > playback on the iPod is fine (but the conversion takes a long, long time). > > ffmpeg -i dotdmeta.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf > 24 -acodec libfaac -aq 100 -vf scale="640:trunc(ow/a/2)*2" DOD.mp4 > > > Any idea how to easily fix it? For ipod classic you need -vpre ipod320 c -- 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 blacktrash at gmx.net Mon Nov 28 16:30:29 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Mon, 28 Nov 2011 15:30:29 +0000 Subject: [FFmpeg-user] Plays too fast on iPod? In-Reply-To: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> References: <4ED3A155.5000907@aleksandrsolzhenitsyn.net> Message-ID: <20111128153029.GK88337@krille.blacktrash.org> * . on Monday, November 28, 2011 at 09:57:25 -0500 > I put a video named 'dotdmeta.mp4' onto my iPod 160 G (Classic). Below > is the FFMPEG -i information. > > The video plays fine until about the 17 minute mark and then playback > speeds up to twice its normal speed. Once the video speeds up the audio > is out of sync with the video too. > > However, if I convert the video using the following FFMPEG code line the > playback on the iPod is fine (but the conversion takes a long, long time). > > ffmpeg -i dotdmeta.mp4 -vcodec libx264 -preset medium -vpre ipod640 -crf > 24 -acodec libfaac -aq 100 -vf scale="640:trunc(ow/a/2)*2" DOD.mp4 > > > Any idea how to easily fix it? For ipod classic you need -vpre ipod320 c -- 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 hcano at mebcn.com Mon Nov 28 16:39:58 2011 From: hcano at mebcn.com (Hector Cano) Date: Mon, 28 Nov 2011 16:39:58 +0100 Subject: [FFmpeg-user] Recording a live stream changing output file every x seconds In-Reply-To: <4ED3A643.2080200@gmail.com> References: <4ED36793.7030009@mebcn.com> <4ED3A643.2080200@gmail.com> Message-ID: <4ED3AB4E.5040801@mebcn.com> On 28/11/11 16:18 , dE . wrote: > On 11/28/11 16:20, Hector Cano wrote: >> Hi, >> I am recording live a stream from an Axis camera. It works perfect. >> >> I'm required to record for maybe a couple of hours at a time, and I'd >> prefer to have several shorter files, maybe a new file every minute, >> instead of a big file (to minimize risk, and to be able to start post >> processing without having to wait till the end). >> >> Is there any option for ffmepg to do the job automatically? >> I've tried using -t 60, and restarting the capture, but the jump is >> too noticeable. >> >> I am using an Ubuntu 11.10 server for the job, launching ffmpeg from >> a bash shell. >> >> I tried to find the answer in the archives, but found nothing but an >> old unanswered question "Recording and splitting at the same time". >> >> Any help would be much appreciated. >> >> >> Thank you, > I think some bash workarounds will do, you may start the other ffmpeg > instances before the first one finishes using bash's '&' e.g. - > > ffmpeg -t 00:01:00 -f x11grab -preset ultrafast -vcodec libx264 mp4.mp4 & > sleep 55; ffmpeg -t 00:01:00 -f x11grab -preset ultrafast -vcodec > libx264 mp4.mp4 & > > And repeat this using while loops. > > I don't understand the post processing part. But as an alternative, > for backups, you may try RAID mirroring using mdadm. Thanks for the idea. I'll give it a try. The "post processing" in this case is some human interaction. I have to be able to supply on demand a stream with part of the recording, for someone to review it, without stopping the recording. Therefore having the video splitted makes it quite easy to get just the demanded part of the video and forward it to the user. From jono at redowl.ca Mon Nov 28 16:56:58 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Mon, 28 Nov 2011 10:56:58 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> <4ED2ACE9.8020800@redowl.ca> <4ED2D178.1020308@redowl.ca> Message-ID: <4ED3AF4A.9020705@redowl.ca> On 11-11-28 05:16 AM, Carl Eugen Hoyos wrote: > Jonathan Addleman redowl.ca> writes: > >> >> With the 'veryslow' preset, the results are abysmal - nasty artifacts >> >> everywhere. Utterly unusable. >> > >> > You have to either define a low constant quantiser or a target bitrate. >> >> Even with -qscale 1, it looked awful. (I assume that's what you mean?) > > (Command line and complete, uncut output missing.) > -qscale 2 works fine with -vcodec mpeg2video and -vcodec mpeg4, I suspect you > have to use a x264 option for -vcodec libx264. Ah, I didn't realize these options didn't apply to the libx264 codec. It's awfully complicated! I'm going to do some more reading, and try to get the newer version of ffmpeg compiled to see if -vcodec copy works there. In the meantime, mpeg4 with qscale 1 seems to give acceptable quality, though the file size is over twice as big. -- Jon-o Addleman - http://www.redowl.ca From jono at redowl.ca Mon Nov 28 17:51:49 2011 From: jono at redowl.ca (Jonathan Addleman) Date: Mon, 28 Nov 2011 11:51:49 -0500 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: <4ED3AF4A.9020705@redowl.ca> References: <4ED155C2.7050807@redowl.ca> <4ED1AA09.5050507@redowl.ca> <4ED2ACE9.8020800@redowl.ca> <4ED2D178.1020308@redowl.ca> <4ED3AF4A.9020705@redowl.ca> Message-ID: <4ED3BC25.2010503@redowl.ca> On 11-11-28 10:56 AM, Jonathan Addleman wrote: > On 11-11-28 05:16 AM, Carl Eugen Hoyos wrote: >> Jonathan Addleman redowl.ca> writes: >> >>> >> With the 'veryslow' preset, the results are abysmal - nasty artifacts >>> >> everywhere. Utterly unusable. >>> > >>> > You have to either define a low constant quantiser or a target bitrate. >>> >>> Even with -qscale 1, it looked awful. (I assume that's what you mean?) >> >> (Command line and complete, uncut output missing.) >> -qscale 2 works fine with -vcodec mpeg2video and -vcodec mpeg4, I suspect you >> have to use a x264 option for -vcodec libx264. > > Ah, I didn't realize these options didn't apply to the libx264 codec. > It's awfully complicated! I'm going to do some more reading, and try to > get the newer version of ffmpeg compiled to see if -vcodec copy works there. No luck with a freshly compiled version. This command line: ffmpeg -i F1jv6V5aYGk.mp4 -itsoffset 0.5 -i F1jv6V5aYGk.mp4 -map 0:0 -map 1:1 -vcodec copy -acodec copy fixed.mp4 resulted in the same out-of-sync audio as the original. I expect that simply copying the raw encoded video results in the itsoffset flag being ignored. Here's the console output: ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 28 2011 11:13:25 with gcc 4.6.1 configuration: --enable-libx264 --enable-gpl libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'F1jv6V5aYGk.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 Duration: 00:03:15.06, start: 0.000000, bitrate: 6116 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080, 5958 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 short.mp4: No such file or directory scaph:~/bla$ ~/Downloads/ffmpeg-0.8.7/ffmpeg -i F1jv6V5aYGk.mp4 -itsoffset 0.5 -i F1jv6V5aYGk.mp4 -map 0:0 -map 1:1 -vcodec copy -acodec copy fixed.mp4 ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 28 2011 11:13:25 with gcc 4.6.1 configuration: --enable-libx264 --enable-gpl libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'F1jv6V5aYGk.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 Duration: 00:03:15.06, start: 0.000000, bitrate: 6116 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080, 5958 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'F1jv6V5aYGk.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 Duration: 00:03:15.06, start: 0.000000, bitrate: 6116 kb/s Stream #1.0(und): Video: h264 (High), yuv420p, 1920x1080, 5958 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #1.1(und): Audio: aac, 44100 Hz, stereo, s16, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 Output #0, mp4, to 'fixed.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-08-19 22:04:25 encoder : Lavf53.5.0 Stream #0.0(und): Video: libx264, yuv420p, 1920x1080, q=2-31, 5958 kb/s, 30k tbn, 29.97 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, 151 kb/s Metadata: creation_time : 2011-08-19 22:04:25 Stream mapping: Stream #0.0 -> #0.0 Stream #1.1 -> #0.1 Press [q] to stop, [?] for help frame= 5846 fps=991 q=-1.0 Lsize= 145698kB time=00:03:15.06 bitrate=6118.9kbits/s video:141890kB audio:3611kB global headers:0kB muxing overhead 0.135492% -- Jonathan Addleman - http://www.redowl.ca From blacktrash at gmx.net Mon Nov 28 18:15:08 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Mon, 28 Nov 2011 17:15:08 +0000 Subject: [FFmpeg-user] -moov_size Message-ID: <20111128171508.GL88337@krille.blacktrash.org> Hi, Are there suggestions with regard to the recommended value to -moov_size? Or just trial and error? c -- \black\trash movie _SAME TIME SAME PLACE_ New York, in the summer of 2001 --->> http://www.blacktrash.org/underdogma/stsp.php From bdkneela at yahoo.com Mon Nov 28 18:28:02 2011 From: bdkneela at yahoo.com (Brian Kneeland) Date: Mon, 28 Nov 2011 09:28:02 -0800 (PST) Subject: [FFmpeg-user] Command line -b:v 20M Bitrate spikes to 65000 Message-ID: <1322501282.78120.YahooMailNeo@web113808.mail.gq1.yahoo.com> I am attempting to re-encode some HD material to playback on my ?Xbox 360.? Source: (according?to mediainfo) Video: AVC,1080p, High at 4.1 profile in an MKV container? Audio: DTS 6 channels at 510 Kbps Here is the command i am running: ffmpeg.exe -i .\test.mkv -vcodec mpeg2video -f mpegts -b:v 20M -r 23.98 -acodec ac3 -ac 6 -ab 448k .\test.ts the resulting file is great and plays fine on my PC but?according?to Bitrate Viewer (http://www.winhoros.de/docs/bitrate-viewer/) there are peaks of 65126 kbps at one point and several in the 45,000-50,000 range.? Using the same program (Bitrate Viewer) on the original MKV file, there are no peaks higher than 46,000 kbps and most more in the 30-35,000 range. Is there a way to modify my command to hard cap the bitrate of the file at 20,000 or 25,000 kbps? Thanks! From tiago.almeida.s at gmail.com Mon Nov 28 18:03:13 2011 From: tiago.almeida.s at gmail.com (Tiago Almeida de Souza) Date: Mon, 28 Nov 2011 15:03:13 -0200 Subject: [FFmpeg-user] Convertendo Sony MXF para XDCAM pelo ffmpeg Message-ID: Hello friend, I am with the following situation, I use ffmpeg to work with Sony MXF files by converting or exporting a quick way to format XDCAM 50Mbits HD 422?? I can work with these files via the command ffmpeg-i input.MXF -vcodec mpeg2video -s HD1080 -b 50000k -vtag xd5b -f mov -acodec pcm_s16le -ac 2 output.mov, how ever the process is very slow and need to integrate this my software wanted to make a quicker process there any way?? It would be possible to use the following command ffmpeg -i input.MP4 - vcodec copy -acodec copy -f mov- vtag xd5b output.mov, because with this command can work with Sony MP4 files without having to convert or export, for what why I understood it just copies the codec nothing else that works without problems both video and audio tracks paras to Sony MP4 file, but already for Sony MXF files and made the process so that the video is waging with delay, however the track audioOK. Is there any solution available?? -- *Atenciosamente,* *Tiago Almeida de Souza* From phil at philrhodes.com Mon Nov 28 18:05:47 2011 From: phil at philrhodes.com (Phil Rhodes) Date: Mon, 28 Nov 2011 17:05:47 -0000 Subject: [FFmpeg-user] Windows builds with Prores Message-ID: Are there any yet? P From lou at lrcd.com Mon Nov 28 20:54:13 2011 From: lou at lrcd.com (Lou) Date: Mon, 28 Nov 2011 10:54:13 -0900 Subject: [FFmpeg-user] Windows builds with Prores In-Reply-To: References: Message-ID: <20111128105413.3c66e6c8@lrcd.com> On Mon, 28 Nov 2011 17:05:47 -0000 "Phil Rhodes" wrote: > Are there any yet? > > P Appears to be: Zeranoe FFmpeg builds http://ffmpeg.zeranoe.com/builds/ ffmpeg -codecs D..... = Decoding supported .E.... = Encoding supported ..V... = Video codec DEV D prores Apple ProRes D V D prores_lgpl Apple ProRes (iCodec Pro) From rothko.fan at gmail.com Mon Nov 28 21:00:39 2011 From: rothko.fan at gmail.com (AlexanderG) Date: Mon, 28 Nov 2011 21:00:39 +0100 Subject: [FFmpeg-user] latest fails of libfaac Message-ID: hi, mac osx 10.8.0 installed libfaac with homebrew /usr/local/lib libfaac.0.0.0.dylib libfaac.a libfaad.2.0.0.dylib libfaad.dylib libfaac.0.dylib libfaac.dylib libfaad.2.dylib Archive : libfaac.a libfaac.a(aacquant.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(bitstream.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 416 0x00002000 libfaac.a(fft.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 576 0x00002000 libfaac.a(frame.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 736 0x00002000 libfaac.a(midside.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(psychkni.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(util.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(backpred.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 416 0x00002000 libfaac.a(channels.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 336 0x00002000 libfaac.a(filtbank.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(huffman.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 576 0x00002000 libfaac.a(ltp.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 libfaac.a(tns.o): Mach header magic cputype cpusubtype caps filetype ncmds sizeofcmds flags 0xfeedfacf 16777223 3 0x00 1 3 496 0x00002000 -- See. Feel. Paint. -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 153381 bytes Desc: not available URL: From Rbigm101 at gmail.com Mon Nov 28 21:56:01 2011 From: Rbigm101 at gmail.com (Mike Rotondo) Date: Mon, 28 Nov 2011 15:56:01 -0500 Subject: [FFmpeg-user] non-redistributable non-free Message-ID: Hello, Can I provide a shell script to someone that compiles and installs a non-redistributable non-free version of ffmpeg? He can't play some mkv's. I'm new to all this non-redistributable stuff but assuming I can't just give him my binary package... Thanks, Mike From h.reindl at thelounge.net Mon Nov 28 22:17:01 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 28 Nov 2011 22:17:01 +0100 Subject: [FFmpeg-user] non-redistributable non-free In-Reply-To: References: Message-ID: <4ED3FA4D.9000604@thelounge.net> Am 28.11.2011 21:56, schrieb Mike Rotondo: > Can I provide a shell script to someone that compiles and installs a > non-redistributable non-free version of ffmpeg? > He can't play some mkv's. sue because your shell-script intracts only with the source code and is technical compareable with a manual > I'm new to all this non-redistributable stuff but assuming I can't just > give him my binary package... i am pretty sure you can provide a binary in a private context because who could make you responsible, the things are changing if you offer binaries on a public platform or doing the compile/install in a business-environment of a paying customer -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From ninive at gmx.at Mon Nov 28 22:23:45 2011 From: ninive at gmx.at (double) Date: Mon, 28 Nov 2011 22:23:45 +0100 Subject: [FFmpeg-user] Negative "start" Message-ID: <4ED3FBE1.7030708@gmx.at> Hello, A dumped stream has a start value "-1696.36". Is there a parameter to start the video from zero? Otherwise ffmpeg will create 1,656 empty seconds. Thanks a lot Marcus Command-line (file has 1MB): ffmpeg -i http://doppelbauer.name/negative-dts.flv test.wmv [flv @ 0x912f260] negative cts, previous timestamps might be wrong Input #0, flv, from 'http://doppelbauer.name/negative-dts.flv': Duration: 00:00:05.30, start: -1696.360000, bitrate: 1625 kb/s Stream #0:0: Video: h264 (Baseline), yuv420p, 720x574 [SAR 1:1 DAR 360:287], 50 tbr, 1k tbn, 50 tbc Stream #0:1: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s From cehoyos at ag.or.at Tue Nov 29 01:13:56 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 00:13:56 +0000 (UTC) Subject: [FFmpeg-user] Negative "start" References: <4ED3FBE1.7030708@gmx.at> Message-ID: double gmx.at> writes: > A dumped stream has a start value "-1696.36". > Is there a parameter to start the video from zero? Otherwise > ffmpeg will create 1,656 empty seconds. > > Thanks a lot > Marcus > > Command-line (file has 1MB): > ffmpeg -i http://doppelbauer.name/negative-dts.flv test.wmv (Complete, uncut console output missing.) Works fine here - there are no empty seconds, the wav file is five seconds long, as is the original. Carl Eugen From cehoyos at ag.or.at Tue Nov 29 01:16:17 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 00:16:17 +0000 (UTC) Subject: [FFmpeg-user] Command line -b:v 20M Bitrate spikes to 65000 References: <1322501282.78120.YahooMailNeo@web113808.mail.gq1.yahoo.com> Message-ID: Brian Kneeland yahoo.com> writes: > ffmpeg.exe -i .\test.mkv -vcodec mpeg2video -f mpegts -b:v 20M -r 23.98 > -acodec ac3 -ac 6 -ab 448k .\test.ts > Is there a way to modify my command to hard cap the bitrate of the file at > 20,000 or 25,000 kbps? Did you try -bufsize and -maxrate as suggested in the manual? Carl Eugen From bdkneela at yahoo.com Tue Nov 29 01:40:28 2011 From: bdkneela at yahoo.com (bdkneela at yahoo.com) Date: Mon, 28 Nov 2011 19:40:28 -0500 Subject: [FFmpeg-user] Command line -b:v 20M Bitrate spikes to 65000 In-Reply-To: References: <1322501282.78120.YahooMailNeo@web113808.mail.gq1.yahoo.com> Message-ID: I want to try that, however i cannot seem to find a good reference as to what values to use for -bufsize. Also if i use maxrate and bufsize, do i still need -b:v ? As you can tell i am a little out of my depth here with command line. Any suggestions? On Nov 28, 2011, at 7:16 PM, Carl Eugen Hoyos wrote: > Brian Kneeland yahoo.com> writes: > >> ffmpeg.exe -i .\test.mkv -vcodec mpeg2video -f mpegts -b:v 20M -r 23.98 >> -acodec ac3 -ac 6 -ab 448k .\test.ts > >> Is there a way to modify my command to hard cap the bitrate of the file at >> 20,000 or 25,000 kbps? > > Did you try -bufsize and -maxrate as suggested in the manual? > > 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 Nov 29 02:23:16 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 01:23:16 +0000 (UTC) Subject: [FFmpeg-user] Command line -b:v 20M Bitrate spikes to 65000 References: <1322501282.78120.YahooMailNeo@web113808.mail.gq1.yahoo.com> Message-ID: yahoo.com> writes: > Also if i use maxrate and bufsize, do i still need -b:v ? I would suspect so, yes. Carl Eugen From arissirajawali at gmail.com Tue Nov 29 07:50:26 2011 From: arissirajawali at gmail.com (aris sirajawali) Date: Tue, 29 Nov 2011 13:50:26 +0700 Subject: [FFmpeg-user] problem encode mov to 3gp Message-ID: how to encode video from mov to 3gp with good quality and small size .. I use the following command: *ffmpeg-i INPUt.MOV-acodec libfaac-ab 12.2-ac 1-ar 8000-vcodec h263-s QCIF-r 10-b 128k OUTPUT.3gp*. and if I make use *acodec amr_nb or amr_wb*, I get an *unknown encoder*, is there an error on my ffmpeg command? Please answer all the friends of friends, .. From dashing.meng at gmail.com Tue Nov 29 08:17:14 2011 From: dashing.meng at gmail.com (littlebat) Date: Tue, 29 Nov 2011 15:17:14 +0800 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: Message-ID: <20111129151714.1556dab3.dashing.meng@gmail.com> On Tue, 29 Nov 2011 13:50:26 +0700 aris sirajawali wrote: > OUTPUT.3gp*. and if I make use *acodec amr_nb or amr_wb*, I get an > *unknown encoder*, is there an error on my ffmpeg command? You need install libopencore-amrnb and compile ffmpeg with supporting it to get amrnb decoding and encoding, see: ./configure --help (.configure --enable-libopencore-amrnb) Installing libvo-amrwbenc and compile ffmpeg with supporting it can get amrwb encoding (./configure --enable-libvo-amrwbenc) To check which codecs your ffmpeg binary has supported, see: ffmpeg -codecs From de.techno at gmail.com Tue Nov 29 09:15:07 2011 From: de.techno at gmail.com (dE .) Date: Tue, 29 Nov 2011 13:45:07 +0530 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: Message-ID: <4ED4948B.1020403@gmail.com> On 11/29/11 12:20, aris sirajawali wrote: > how to encode video from mov to 3gp with good quality and small size .. > I use the following command: *ffmpeg-i INPUt.MOV-acodec libfaac-ab 12.2-ac > 1-ar 8000-vcodec h263-s QCIF-r 10-b 128k OUTPUT.3gp*. > and if I make use *acodec amr_nb or amr_wb*, I get an *unknown encoder*, is > there an error on my ffmpeg command? > Please answer all the friends of friends, .. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user You may also try winff with the presets. If this's for a device I'll make the preset, just name the device with the codecs it supports. From arissirajawali at gmail.com Tue Nov 29 09:30:40 2011 From: arissirajawali at gmail.com (aris sirajawali) Date: Tue, 29 Nov 2011 15:30:40 +0700 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: <20111129151714.1556dab3.dashing.meng@gmail.com> References: <20111129151714.1556dab3.dashing.meng@gmail.com> Message-ID: when I typing. / configure --enable-libopencore-amrnb, return an error message occurs and reads *ERROR: libopencore_amrnb **not found.* please help .. 2011/11/29 littlebat > On Tue, 29 Nov 2011 13:50:26 +0700 > aris sirajawali wrote: > > > OUTPUT.3gp*. and if I make use *acodec amr_nb or amr_wb*, I get an > > *unknown encoder*, is there an error on my ffmpeg command? > > You need install libopencore-amrnb and compile ffmpeg with supporting > it to get amrnb decoding and encoding, see: ./configure --help > (.configure --enable-libopencore-amrnb) > Installing libvo-amrwbenc and compile ffmpeg with supporting it can get > amrwb encoding (./configure --enable-libvo-amrwbenc) > > To check which codecs your ffmpeg binary has supported, see: ffmpeg > -codecs > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jintonglei at gmail.com Tue Nov 29 10:13:51 2011 From: jintonglei at gmail.com (jintonglei) Date: Tue, 29 Nov 2011 17:13:51 +0800 Subject: [FFmpeg-user] problems of compile Android variant Message-ID: <201111291713492187111@gmail.com> Hello, I tried to compile ffmpeg for Android but fail, it said: mips-linux-gnu-gcc is unable to create an executable file. C compiler test failed. And the configuration is: ./configure --arch=mips --cross-prefix=mips-linux-gnu- --extra-ldflags=-static extra-cflags=-static --enable-cross-compile --target-os=linux I tried many means from google, but still stucking... 2011-11-29 jintonglei From sweetthdevil at gmail.com Tue Nov 29 10:29:37 2011 From: sweetthdevil at gmail.com (Sw@g) Date: Tue, 29 Nov 2011 09:29:37 +0000 Subject: [FFmpeg-user] timestamps? Message-ID: <4ED4A601.8080900@gmail.com> Hi all, I was looking at the documentation and to my surprise the timestamps option is mention few time, am I correct assuming we can now add a timestamps to video recorded from a camera using ffmpeg? i.e. we can have the time when the video was recorded like a security cam? Many thanks for your reply, From dashing.meng at gmail.com Tue Nov 29 10:59:45 2011 From: dashing.meng at gmail.com (littlebat) Date: Tue, 29 Nov 2011 17:59:45 +0800 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: <20111129151714.1556dab3.dashing.meng@gmail.com> Message-ID: <20111129175945.bb343702.dashing.meng@gmail.com> On Tue, 29 Nov 2011 15:30:40 +0700 aris sirajawali wrote: > when I typing. / configure --enable-libopencore-amrnb, return an error > message occurs and reads *ERROR: libopencore_amrnb **not found.* > please help .. http://sourceforge.net/projects/opencore-amr/files/ You need install opencore-amr(for amrnb encoding) or vo-amrwbenc(for amrwb encoding) first. From dave.bevan at bbc.co.uk Tue Nov 29 11:38:26 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Tue, 29 Nov 2011 10:38:26 -0000 Subject: [FFmpeg-user] Convertendo Sony MXF para XDCAM pelo ffmpeg In-Reply-To: References: Message-ID: Subject: [FFmpeg-user] Convertendo Sony MXF para XDCAM pelo ffmpeg Try http://code.google.com/p/ffmbc/ - "FFmpeg customized for broadcast and professional usage." http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From cehoyos at ag.or.at Tue Nov 29 13:02:40 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 12:02:40 +0000 (UTC) Subject: [FFmpeg-user] Timecode problem when capturing capturing live multicast/udp mpegvideo stream References: <1322486660.4ed38b84ab465@webmail.free.fr> Message-ID: free.fr> writes: > ffmpeg.exe -y -i udp://225.1.1.53:1111 -vcodec copy -acodec copy output.mpg > > ffmpeg version N-32754-g936d4d4-Sherpya, Copyright (c) 2000-2011 the FFmpeg This is ~2500 versions old and there have been fixes in the udp code. Assuming you also have access to the file (that is streamed): Is the problem also reproducible with ffmpeg -i file -vcodec copy -acodec copy out.mpg? Carl Eugen From gkinsey at ad-holdings.co.uk Tue Nov 29 14:22:51 2011 From: gkinsey at ad-holdings.co.uk (Gavin Kinsey) Date: Tue, 29 Nov 2011 13:22:51 +0000 Subject: [FFmpeg-user] problems of compile Android variant In-Reply-To: <201111291713492187111@gmail.com> References: <201111291713492187111@gmail.com> Message-ID: <201111291322.51669.gkinsey@ad-holdings.co.uk> On Tuesday 29 November 2011 09:13:51 jintonglei wrote: > Hello, > I tried to compile ffmpeg for Android but fail, it said: > mips-linux-gnu-gcc is unable to create an executable file. > C compiler test failed. > > And the configuration is: > ./configure --arch=mips --cross-prefix=mips-linux-gnu- > --extra-ldflags=-static extra-cflags=-static --enable-cross-compile > --target-os=linux I can give you the flags I use to compile for ARM Android, you'll have to modify them for mips. Didn't know there was any Android for mips, presume some third-party fork. --disable-static --enable-shared -- sysroot=${ANDROID_NDK}/platforms/android-8/arch-arm --arch=arm --target- os=linux --cross-prefix=${ANDROID_NDK}/toolchains/arm-linux- androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi- --disable- symver --cc="/usr/bin/ccache ${ANDROID_NDK}/toolchains/arm-linux- androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-gcc" -- enable-small There are also some minor modifications needed to the source, but those are for ARM assembler and so shouldn't be needed for you. -- Gavin Kinsey AD Holdings Plc From Rbigm101 at yahoo.com Mon Nov 28 21:53:26 2011 From: Rbigm101 at yahoo.com (Mike Rotondo) Date: Mon, 28 Nov 2011 15:53:26 -0500 Subject: [FFmpeg-user] non-redistributable non-free Message-ID: Hello, Can I provide a shell script to someone that compiles and installs a non-redistributable non-free version of ffmpeg? He can't play some mkv's. I'm new to all this non-redistributable stuff but assuming I can't just give him my binary package... Thanks, Mike From foofoobedoo at yahoo.co.uk Tue Nov 29 18:26:21 2011 From: foofoobedoo at yahoo.co.uk (Neal Crook) Date: Tue, 29 Nov 2011 17:26:21 +0000 (GMT) Subject: [FFmpeg-user] controlling bitrate for MJPEG Message-ID: <1322587581.93398.YahooMailNeo@web29618.mail.ird.yahoo.com> Hello kind ffmpeg experts, Starting with an uncompressed video clip in an .avi container, I wish to generate videos at various different bit-rates. The video formats I am interested in are MJPEG and H.264 I-frame only (ie H.264 with GOP=1). [The formats are chosen because they both map to relatively low cost/complexity hardware implementations]. My MJPEG incantations looks like this: $ ffmpeg -i foo.avi -b:v 1M -codec mjpeg out_1M.avi $ ffmpeg -i foo.avi -b:v 2M -codec mjpeg out_2M.avi $ ffmpeg -i foo.avi -b:v 4M -codec mjpeg out_4M.avi $ ffmpeg -i foo.avi -b:v 8M -codec mjpeg out_8M.avi $ ffmpeg -i foo.avi -b:v 16M -codec mjpeg out_16M.avi When I examine/analyse the out_* files I observe that the file size/bit rate does not drop below ~8Mbit/sec -- ?the 1M, 2M, 4M files are all comparable sizes. I realise that the .avi container has some small overhead, and I suppose there is some lower quality limit (and associated minimum file size) for the JPEG encoding, but I was surprised to hit it so soon. Can anyone identify an error I have made or explain why I cannot reduce the file size further? thanks, Neal. ----------------------- ffmpeg version 0.8.6.git, Copyright (c) 2000-2011 the FFmpeg developers ? built on Nov 25 2011 16:33:47 with gcc 4.5.1 ? configuration: --prefix=/home/foo/local --enable-libx264 --enable-gpl --extra-cflags=-I/home/foo/local/include --extra-ldflags=-L/home/foo/local/lib ? libavutil ? ?51. 28. 0 / 51. 28. 0 ? libavcodec ? 53. 37. 0 / 53. 37. 0 ? libavformat ?53. 21. 0 / 53. 21. 0 ? libavdevice ?53. ?4. 0 / 53. ?4. 0 ? libavfilter ? 2. 49. 0 / ?2. 49. 0 ? libswscale ? ?2. ?1. 0 / ?2. ?1. 0 ? libpostproc ?51. ?2. 0 / 51. ?2. 0 From cehoyos at ag.or.at Tue Nov 29 19:55:30 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 18:55:30 +0000 (UTC) Subject: [FFmpeg-user] non-redistributable non-free References: Message-ID: Mike Rotondo yahoo.com> writes: > Can I provide a shell script to someone that compiles and installs a > non-redistributable non-free version of ffmpeg? What was wrong with Haralds answer? Note that afaict, no lawyers are posting on this list, and only a lawyer can give you a binding answer! (I am not a lawyer.) Apart from warranty, The GPL and the LGPL primarily restrict the distribution of binaries, so if you distribute shell scripts (that I suppose are not based on FFmpeg source code), I wonder what you are worried about. Carl Eugen From cehoyos at ag.or.at Tue Nov 29 19:58:38 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 18:58:38 +0000 (UTC) Subject: [FFmpeg-user] controlling bitrate for MJPEG References: <1322587581.93398.YahooMailNeo@web29618.mail.ird.yahoo.com> Message-ID: Neal Crook yahoo.co.uk> writes: > $ ffmpeg -i foo.avi -b:v 16M -codec mjpeg out_16M.avi > > When I examine/analyse the out_* files I observe that the file size/bit rate > does not drop below ~8Mbit/sec Did you try -qscale 31? This should result in the smallest possible size. Carl Eugen From Rbigm101 at gmail.com Tue Nov 29 20:23:56 2011 From: Rbigm101 at gmail.com (Mike Rotondo) Date: Tue, 29 Nov 2011 14:23:56 -0500 Subject: [FFmpeg-user] non-redistributable non-free Message-ID: Mike Rotondo yahoo.com> writes: >> Can I provide a shell script to someone that compiles and installs a >> non-redistributable non-free version of ffmpeg? > What was wrong with Haralds answer? > Note that afaict, no lawyers are posting on this list, and only a lawyer can > give you a binding answer! > (I am not a lawyer.) > Apart from warranty, The GPL and the LGPL primarily restrict the distribution of > binaries, so if you distribute shell scripts (that I suppose are not based on > FFmpeg source code), I wonder what you are worried about. > Carl Eugen Sorry it was a double post (i've never posted before). But no it wouldn't have any FFmpeg source in the script, it would just download configure make and install the source. Haralds answer confirmed what I originally thought. - Mike From klaus.kudielka at gmx.net Tue Nov 29 21:38:24 2011 From: klaus.kudielka at gmx.net (Klaus Kudielka) Date: Tue, 29 Nov 2011 21:38:24 +0100 Subject: [FFmpeg-user] Application provided invalid, non monotonically increasing dts to muxer Message-ID: <20111129203824.GA13472@mars> I would like to benefit from the new command line syntax, but at the moment, with git master, I cannot re-multiplex any MKV with vc1 or h264 in it. On the other hand, - git master seems to work work with MKV's containing mpeg2video - 0.8.7 works perfectly even with vc1 and h264. This might be related to FFmpeg tickets 222 & 415, but is actually somebody working on it? Thanks, Klaus ---- Two examples for NOT WORKING git masker: [0]$ /opt/ffmpeg-git/bin/ffmpeg -i x.mkv -map 0 -c copy -y /tmp/x.mkv ffmpeg version 0.8.7.git-7076967, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 29 2011 20:31:15 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libxvid --enable-libvpx --enable-librtmp --extra-cflags=-I/opt/x264/include --extra-ldflags=-L/opt/x264/lib --prefix=/opt/ffmpeg-git libavutil 51. 29. 1 / 51. 29. 1 libavcodec 53. 39. 1 / 53. 39. 1 libavformat 53. 22. 0 / 53. 22. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 50. 0 / 2. 50. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, matroska,webm, from 'x.mkv': Metadata: ENCODER : Lavf53.4.0 Duration: 02:28:51.68, start: 0.000000, bitrate: 14874 kb/s Stream #0:0: Video: h264 (High), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s (default) Stream #0:2(deu): Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s Stream #0:3(eng): Subtitle: hdmv_pgs_subtitle Stream #0:4(eng): Subtitle: hdmv_pgs_subtitle Stream #0:5(ger): Subtitle: hdmv_pgs_subtitle Stream #0:6(ger): Subtitle: hdmv_pgs_subtitle Output #0, matroska, to '/tmp/x.mkv': Metadata: encoder : Lavf53.22.0 Stream #0:0: Video: h264, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 23.98 fps, 1k tbn, 1k tbc (default) Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s (default) Stream #0:2(deu): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s Stream #0:3(eng): Subtitle: hdmv_pgs_subtitle Stream #0:4(eng): Subtitle: hdmv_pgs_subtitle Stream #0:5(ger): Subtitle: hdmv_pgs_subtitle Stream #0:6(ger): Subtitle: hdmv_pgs_subtitle Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Stream #0:5 -> #0:5 (copy) Stream #0:6 -> #0:6 (copy) Press [q] to stop, [?] for help [matroska @ 0x27ad6c0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: -83 >= -83 av_interleaved_write_frame(): Invalid argument [1]$ /opt/ffmpeg-git/bin/ffmpeg -i y.mkv -map 0 -c copy -y /tmp/x.mkv ffmpeg version 0.8.7.git-7076967, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 29 2011 20:31:15 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libxvid --enable-libvpx --enable-librtmp --extra-cflags=-I/opt/x264/include --extra-ldflags=-L/opt/x264/lib --prefix=/opt/ffmpeg-git libavutil 51. 29. 1 / 51. 29. 1 libavcodec 53. 39. 1 / 53. 39. 1 libavformat 53. 22. 0 / 53. 22. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 50. 0 / 2. 50. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 47.95 (48000/1001) -> 23.98 (48000/2002) Input #0, matroska,webm, from 'y.mkv': Duration: 02:39:01.53, start: 0.000000, bitrate: 18114 kb/s Stream #0:0: Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s (default) Stream #0:2(ger): Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s Stream #0:3(eng): Subtitle: hdmv_pgs_subtitle Stream #0:4(eng): Subtitle: hdmv_pgs_subtitle Stream #0:5(ger): Subtitle: hdmv_pgs_subtitle Stream #0:6(ger): Subtitle: hdmv_pgs_subtitle Output #0, matroska, to '/tmp/x.mkv': Metadata: encoder : Lavf53.22.0 Stream #0:0: Video: vc1 (WVC1 / 0x31435657), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 23.98 fps, 1k tbn, 1k tbc (default) Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s (default) Stream #0:2(ger): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s Stream #0:3(eng): Subtitle: hdmv_pgs_subtitle Stream #0:4(eng): Subtitle: hdmv_pgs_subtitle Stream #0:5(ger): Subtitle: hdmv_pgs_subtitle Stream #0:6(ger): Subtitle: hdmv_pgs_subtitle Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Stream #0:5 -> #0:5 (copy) Stream #0:6 -> #0:6 (copy) Press [q] to stop, [?] for help [matroska @ 0x17e8b60] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 125 >= 42 av_interleaved_write_frame(): Invalid argument [1]$ ---- Counter-example of working 0.8.7: [1]$ /opt/ffmpeg/bin/ffmpeg -i x.mkv" -map 0.0 -vcodec copy -an -sn -y /tmp/x.mkv -map 0.1 -acodec copy -newaudio -map 0.2 -acodec copy -newaudio -map 0.3 -scodec copy -newsubtitle -map 0.4 -scodec copy -newsubtitle -map 0.5 -scodec copy -newsubtitle -map 0.6 -scodec copy -newsubtitle ffmpeg version 0.8.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 28 2011 21:14:00 with gcc 4.6.1 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libxvid --enable-libvpx --enable-librtmp --enable-shared --extra-cflags=-I/opt/x264/include --extra-ldflags=-L/opt/x264/lib --prefix=/opt/ffmpeg libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [matroska,webm @ 0x7083e0] max_analyze_duration 5000000 reached at 5024000 [matroska,webm @ 0x7083e0] Estimating duration from bitrate, this may be inaccurate Input #0, matroska,webm, from 'x.mkv': Metadata: ENCODER : Lavf53.4.0 Duration: 02:28:51.68, start: 0.000000, bitrate: 896 kb/s Stream #0.0: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s (default) Stream #0.2(deu): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s Stream #0.3(eng): Subtitle: pgssub Stream #0.4(eng): Subtitle: pgssub Stream #0.5(ger): Subtitle: pgssub Stream #0.6(ger): Subtitle: pgssub Output #0, matroska, to '/tmp/x.mkv': Metadata: encoder : Lavf53.5.0 Stream #0.0: Video: libx264, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 1k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, 448 kb/s (default) Stream #0.2(deu): Audio: ac3, 48000 Hz, 5.1, 448 kb/s Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 Stream #0.4(eng): Subtitle: [0][0][0][0] / 0x0000 Stream #0.5(ger): Subtitle: [0][0][0][0] / 0x0000 Stream #0.6(ger): Subtitle: [0][0][0][0] / 0x0000 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Stream #0.2 -> #0.2 Stream #0.3 -> #0.3 Stream #0.4 -> #0.4 Stream #0.5 -> #0.5 Stream #0.6 -> #0.6 Press [q] to stop, [?] for help frame=214146 fps=738 q=-1.0 Lsize=16255928kB time=02:24:26.57 bitrate=15365.8kbits/s video:15210771kB audio:976902kB global headers:0kB muxing overhead 0.421647% [0]$ From cehoyos at ag.or.at Tue Nov 29 23:48:58 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 29 Nov 2011 22:48:58 +0000 (UTC) Subject: [FFmpeg-user] Application provided invalid, non monotonically increasing dts to muxer References: <20111129203824.GA13472@mars> Message-ID: Klaus Kudielka gmx.net> writes: > ---- Two examples for NOT WORKING git masker: > > [0]$ /opt/ffmpeg-git/bin/ffmpeg -i x.mkv -map 0 -c copy -y /tmp/x.mkv [...] > ---- Counter-example of working 0.8.7: > > [1]$ /opt/ffmpeg/bin/ffmpeg -i x.mkv" -map 0.0 -vcodec copy -an -sn -y > /tmp/x.mkv -map 0.1 -acodec copy -newaudio -map 0.2 -acodec copy -newaudio > -map 0.3 -scodec copy -newsubtitle -map 0.4 -scodec copy -newsubtitle -map 0.5 > -scodec copy -newsubtitle -map 0.6 -scodec copy -newsubtitle The command lines look different... Could you provide a sample and / or find the version introducing the problem? (I don't think ticket 222 or ticket 415 are regressions.) Carl Eugen From phil_rhodes at rocketmail.com Wed Nov 30 00:39:56 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Tue, 29 Nov 2011 23:39:56 -0000 Subject: [FFmpeg-user] non-redistributable non-free In-Reply-To: References: Message-ID: > Haralds answer confirmed what I originally thought. But is in no way binding on anyone other than Harald. This is the GPL, sadly. Nobody can give you a straight answer, and if they could, the straight answer wouldn't make much objective sense. I suspect that you will probably get away with what you are proposing to do, as most people would usually get away with doing more or less anything with GPL code as long as you don't shout about it. I have long worked on the basis that the GPL, like most sorts of duplication control law, is only tolerated because it is very rarely enforced. P From drurowin at gmail.com Wed Nov 30 00:40:55 2011 From: drurowin at gmail.com (Lucien Pullen) Date: Tue, 29 Nov 2011 16:40:55 -0700 Subject: [FFmpeg-user] Default audio bitrate changed? Message-ID: <4ED56D87.7030707@gmail.com> The manual states that the default audio bitrate is 64k. However, it seems to have changed to 128k silently. Is this change going to stick (should the manual be updated), or is this just temporary? % ffmpeg -i 01\ --\ Musique_Des_Machines_1.flac 01\ --\ Musique_Des_Machines_1.m4a ffmpeg version N-35231-g5c15b78, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 26 2011 17:49:37 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --enable-gpl --enable-libfaac --enable-libvorbis --enable-libx264 --enable-nonfree --enable-x11grab --extra-cflags=-I/opt/local/include --extra-cflags=-I/opt/local/include/X11 --extra-cflags=-I/usr/local/include --extra-cflags='-I/usr/local/include/SDL -D_GNU_SOURCE=1 -D_THREAD_SAFE' --extra-ldflags=-L/opt/local/lib --extra-ldflags=-L/usr/local/lib --extra-ldflags='-L/usr/local/lib /usr/local/lib/libSDLmain.a /usr/local/lib/libSDL.a -Wl,-framework,OpenGL -Wl,-framework,Cocoa -Wl,-framework,ApplicationServices -Wl,-framework,Carbon -Wl,-framework,AudioToolbox -Wl,-framework,AudioUnit -Wl,-framework,IOKit' libavutil 51. 29. 1 / 51. 29. 1 libavcodec 53. 38. 1 / 53. 38. 1 libavformat 53. 22. 0 / 53. 22. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 50. 0 / 2. 50. 0 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 [flac @ 0x101809800] max_analyze_duration 5000000 reached at 5015510 Input #0, flac, from '01 -- Musique_Des_Machines_1.flac': Metadata: TITLE : Musique Des Machines 1 track : 1 TRACKOF : 2 ARTIST : Broekhuis, Keller & Sch?nw?lder ALBUM : Musique Des Machines GENRE : Electronic SUBGENRE1 : Berlin School SUBGENRE2 : Trance DATE : 2005 DATE-RECORDED : 2003 ENCODER : Lavf53.4.0 Duration: 00:26:22.56, bitrate: 818 kb/s Stream #0:0: Audio: flac, 44100 Hz, stereo, s16 Output #0, ipod, to '01 -- Musique_Des_Machines_1.m4a': Metadata: TITLE : Musique Des Machines 1 track : 1 TRACKOF : 2 ARTIST : Broekhuis, Keller & Sch?nw?lder ALBUM : Musique Des Machines GENRE : Electronic SUBGENRE1 : Berlin School SUBGENRE2 : Trance DATE : 2005 DATE-RECORDED : 2003 encoder : Lavf53.22.0 Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (flac -> libfaac) Press [q] to stop, [?] for help sample/frame number mismatch in adjacent frameskbits/s size= 25229kB time=00:26:22.60 bitrate= 130.6kbits/s video:0kB audio:24695kB global headers:0kB muxing overhead 2.159891% From ivano.arrighetta at gmail.com Wed Nov 30 01:12:18 2011 From: ivano.arrighetta at gmail.com (Ivano Arrighetta) Date: Wed, 30 Nov 2011 01:12:18 +0100 Subject: [FFmpeg-user] FFMpeg with other software Message-ID: Hello everyone. I'm trying to realize a small linux distro, I'm choosing the packages with which the distro will come. I'm just missing some packages for some tasks, please see http://igpgames.altervista.org/question.html and share your experience. For more info on the linux distro I'm going to make, please visit http://sourceforge.net/projects/thebigproject/ (the project is not up to date however). For the records, the distro will be media production oriented, with as little disksize as possible. Thanks in advance for any help. Bye, Ivano Arrighetta. From peace at aleksandrsolzhenitsyn.net Wed Nov 30 01:29:48 2011 From: peace at aleksandrsolzhenitsyn.net (peace at aleksandrsolzhenitsyn.net) Date: Tue, 29 Nov 2011 17:29:48 -0700 Subject: [FFmpeg-user] iPod 640x480 In-Reply-To: References: Message-ID: <4bef86dbd37303a551ca0997cc212225@aleksandrsolzhenitsyn.net> Can someone explain to me why my iPod can't handle videos at 640x480 when the spec's for it say it can? On Apple's website it does say about the display- 320-by-240-pixel resolution at 163 pixels per inch. But, it does say in the spec's it can handle 640x480. Videos at 320 x 240 work perfectly, thanks to all of your help, but for whatever reason the 640's don't. Being that I'm so stupid about this sort of thing I'm sure there's an easy explanation. The spec's for the iPod Classic (160G) are below. H.264 video, up to 1.5 Mbps, 640 by 480 pixels, 30 frames per second, Low-Complexity version of the H.264 Baseline Profile with AAC-LC audio up to 160 Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats; H.264 video, up to 2.5 Mbps, 640 by 480 pixels, 30 frames per second, Baseline Profile up to Level 3.0 with AAC-LC audio up to 160 Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats; MPEG-4 video, up to 2.5 Mbps, 640 by 480 pixels, 30 frames per second, Simple Profile with AAC-LC audio up to 160 Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats From llee040 at sbcglobal.net Wed Nov 30 01:31:57 2011 From: llee040 at sbcglobal.net (L. Lee) Date: Tue, 29 Nov 2011 18:31:57 -0600 Subject: [FFmpeg-user] -istoffset and -vcodec copy In-Reply-To: <4ED3BC25.2010503@redowl.ca> Message-ID: I sent this to the Mencoder list today. Maybe you can use it. > I recently submitted this suggestion for adjusting a/v sync using ffmpeg. I've > discovered that all the included examples caused the audio to be advanced, > although the first example incorrectly claims to provide delay for the audio. > I discovered that to apply delay to the audio, a minus (-) must also precede > the numeric value, so the first example should read: > > "Make audio 3 seconds later: > ffmpeg -i input.flv -itsoffset -00:00:03.0 -i input.flv -vcodec copy -acodec > copy -map 1:0 -map 0:1 output_shift3s-delay.flv" > > I haven't tested to determine whether the order of the "-map" options must be > reversed between the first and second examples as indicated. > > Anybody know? > > Anyway, to summarize, if the minus (-) before the numeric value is omitted, > both the original examples serve to advance the audio for the files I've > tested. > > Thanks. > > Laine > > > On 11/8/11 1:27 PM, "L. Lee" wrote: > >> I occasionally have to deal with a stream for which audio seems to begin >> before video, and MPlayer appears to maintain sync by starting playback at >> the >> point where video enters. When transcoding with MEncoder, I found that I had >> to disable -noskip to allow black frames to be created before entrance of >> the >> video so that sync was maintained. That caused some issues that made me >> change >> other options, because I'm having to perform pullup with encoding, but it >> might be worth trying. >> >> Also, because I have to build ffmpeg to build MEncoder now, it's always >> available. You can use ffmpeg to offset the sync so you can tell whether the >> duration is correct. Here are examples (from >> http://lzone.de/fix+async+video+with+ffmpeg) >> >> Make audio 3 seconds later: >> ffmpeg -i input.flv -itsoffset 00:00:03.0 -i input.flv -vcodec copy -acodec >> copy -map 1:0 -map 0:1 output_shift3s-delay.flv >> >> Make audio 3 seconds earlier: >> ffmpeg -i input.flv -itsoffset 00:00:03.0 -i input.flv -vcodec copy -acodec >> copy -map 0:1 -map 1:0 output_shift3s-advance.flv >> >> Here's an example for making an iPad (or iPhone) movie's audio about 300 >> milliseconds earlier: >> >> ffmpeg -i inputmovie.iPad.mp4 -itsoffset 00:00:00.3 -i inputmovie.iPad.mp4 >> -vcodec copy -acodec copy -map 1:0 -map 0:1 >> outputmovie.iPad.300ms-advance.mp4 >> >> Laine Lee From h.reindl at thelounge.net Wed Nov 30 02:00:52 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 30 Nov 2011 02:00:52 +0100 Subject: [FFmpeg-user] non-redistributable non-free In-Reply-To: References: Message-ID: <4ED58044.6050503@thelounge.net> Am 30.11.2011 00:39, schrieb Phil Rhodes: >> Haralds answer confirmed what I originally thought. > > But is in no way binding on anyone other than Harald. > > This is the GPL, sadly. Nobody can give you a straight answer, and if they could, > the straight answer wouldn't make much objective sense. laughable if you distribute a self written SHELL-SCRIPT there is no code nor a license of a single piece of ffmpeg affected in any way -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From arissirajawali at gmail.com Wed Nov 30 04:03:18 2011 From: arissirajawali at gmail.com (aris sirajawali) Date: Wed, 30 Nov 2011 10:03:18 +0700 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: <20111129175945.bb343702.dashing.meng@gmail.com> References: <20111129151714.1556dab3.dashing.meng@gmail.com> <20111129175945.bb343702.dashing.meng@gmail.com> Message-ID: thanks for the response from friends all .. but why do I still get an error message like this * ERROR: not found ** libopencore_amrnb .*.. Please help to send the reference to install ffmpeg is complete .. > http://sourceforge.net/projects/opencore-amr/files/ > > You need install opencore-amr(for amrnb encoding) or vo-amrwbenc(for > amrwb encoding) first. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ffmpeg-user at herveybayaustralia.com.au Wed Nov 30 03:23:27 2011 From: ffmpeg-user at herveybayaustralia.com.au (Da Rock) Date: Wed, 30 Nov 2011 12:23:27 +1000 Subject: [FFmpeg-user] no connections - ffserver rendered useless Message-ID: <4ED5939F.5070508@herveybayaustralia.com.au> I sent this in the other day, but I've had no response at all. I did eventually find the ffserver list, but it seems a ghost town. Further testing shows ffserver is rendered useless as it can't seem to receive input from any source- even if it is a static stream such as outlined at the bottom of the ffserver.conf. Apparently ffmpeg is unable to communicate with ffserver at all. Any attempt to download or connect ends in a 0 byte file. That said the stat.html does work. This is the output: ffserver Status Available Streams Path Served Conns bytes Format Bit rate kbits/s Video kbits/s Codec Audio kbits/s Codec Feed test1.mpg 0 0 mpeg 1192 1000 mpeg1video 192 mp2 feed1.ffm test.swf 0 0 swf 1264 1200 libx264 64 libmp3lame feed1.ffm stream2.mpg 0 0 mp2 64 0 64 mp2 stream2.ffm stream3.mpg 0 0 mp2 64 0 64 mp2 stream3.ffm stream1.mpg 0 0 mp2 64 0 64 mp2 stream1.ffm testfile.flv 1 64 flv 0 0 libx264 0 libvorbis /.mkv stat.html 1 0 - - - - index.html 0 0 - - - - Feed feed1.ffm Stream type kbits/s codec Parameters 0 audio 192 mp2 2 channel(s), 44100 Hz 1 video 1000 mpeg1video 160x128, q=2-31, fps=25 2 audio 64 libmp3lame 1 channel(s), 22050 Hz 3 video 1200 libx264 1280x720, q=2-31, fps=25 Feed stream2.ffm Stream type kbits/s codec Parameters 0 audio 64 mp2 1 channel(s), 22050 Hz Feed stream3.ffm Stream type kbits/s codec Parameters 0 audio 64 mp2 1 channel(s), 22050 Hz Feed stream1.ffm Stream type kbits/s codec Parameters 0 audio 64 mp2 1 channel(s), 22050 Hz Connection Status Number of connections: 1 / 1000 Bandwidth in use: 0k / 500000k # File IP Proto State Target bits/sec Actual bits/sec Bytes transferred *1* stat.html 192.168.0.179 HTTP/1.1 HTTP_WAIT_REQUEST 0 0 0 ------------------------------------------------------------------------ Generated at Wed Nov 30 12:10:54 2011 Any ideas on how to diagnose/fix this problem? I got back to an old project I've been fiddling with for some time now, but ffserver doesn't appear to be doing so well now. I've run some updates, new platform, and new platform versions and I've done ok until now. When I first kicked ffserver off it refused to bind to the network at all, after some retries, searching through confs, it worked- not sure what the failure was after all that. When I started sending data to the server using ffmpeg I get the following error: av_interleaved_write_frame(): Connection reset by peer My data is gstreamer pipelined dvb stream with ffmpeg splitting and processing to mpegts output, although I have another method which essentially has the same result anyway- gstreamer drops the pipe when the buffers fill because ffmpeg doesn't take the data from the pipe. Cmd is as follows: gst-launch dvbsrc adapter=1 frequency=xxxxxxxx modulation="QAM 64" bandwidth=7 guard=16 ! queue ! fdsink | ffmpeg -i - -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1601 http://127.0.0.1:8090/stream1.ffm -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1605 http://127.0.0.1:8090/stream2.ffm -f mpegts -vsync 0 -vcodec copy -acodec copy -map_metadata 0:p:1608 http://127.0.0.1:8090/stream3.ffm If I output to file it all goes beautifully. Does anyone have any clues on this? Any help would be much appreciated. I also could use some help as to where the logs are going as well? Cheers My platform is this: FreeBSD 8.1-RELEASE-p1 FreeBSD 8.1-RELEASE-p1 ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 26 2011 12:45:06 with gcc 4.2.1 20070719 [FreeBSD] configuration: --prefix=/usr/local --mandir=/usr/local/man --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-memalign-hack --enable-runtime-cpudetect --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 --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libcelt --enable-libdirac --enable-libfaac --enable-libfreetype --enable-frei0r --enable-libgsm --enable-libmp3lame --enable-libopencv --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-ffplay --enable-libspeex --enable-libtheora --disable-vaapi --disable-vdpau --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-nonfree --enable-version3 libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 ffserver version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers built on Nov 26 2011 12:45:06 with gcc 4.2.1 20070719 [FreeBSD] configuration: --prefix=/usr/local --mandir=/usr/local/man --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-memalign-hack --enable-runtime-cpudetect --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 --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libcelt --enable-libdirac --enable-libfaac --enable-libfreetype --enable-frei0r --enable-libgsm --enable-libmp3lame --enable-libopencv --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-ffplay --enable-libspeex --enable-libtheora --disable-vaapi --disable-vdpau --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-nonfree --enable-version3 libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Config: # Port on which the server is listening. You must select a different # port from your standard HTTP web server if it is running on the same # computer. Port 8090 # Address on which the server is bound. Only useful if you have # several network interfaces. BindAddress 0.0.0.0 # Number of simultaneous HTTP connections that can be handled. It has # to be defined *before* the MaxClients parameter, since it defines the # MaxClients maximum limit. MaxHTTPConnections 2000 # Number of simultaneous requests that can be handled. Since FFServer # is very fast, it is more likely that you will want to leave this high # and use MaxBandwidth, below. MaxClients 1000 # This the maximum amount of kbit/sec that you are prepared to # consume when streaming to clients. MaxBandwidth 500000 # Access log file (uses standard Apache log file format) # '-' is the standard output. CustomLog - # Suppress that if you want to launch ffserver as a daemon. NoDaemon ################################################################## # Definition of the live feeds. Each live feed contains one video # and/or audio sequence coming from an ffmpeg encoder or another # ffserver. This sequence may be encoded simultaneously with several # codecs at several resolutions. # You must use 'ffmpeg' to send a live feed to ffserver. In this # example, you can type: # # ffmpeg http://localhost:8090/feed1.ffm # ffserver can also do time shifting. It means that it can stream any # previously recorded live stream. The request should contain: # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify # a path where the feed is stored on disk. You also specify the # maximum size of the feed, where zero means unlimited. Default: # File=/tmp/feed_name.ffm FileMaxSize=5M File /tmp/feed1.ffm FileMaxSize 512M # You could specify # ReadOnlyFile /saved/specialvideo.ffm # This marks the file as readonly and it will not be deleted or updated. # Specify launch in order to start ffmpeg automatically. # First ffmpeg must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field #Launch ffmpeg # Only allow connections from localhost to the feed. ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 2.25G File /home/share/stream2.ffm FileMaxSize 2304M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 2.25G File /home/share/stream3.ffm FileMaxSize 2304M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 # Set file size to 6.5G File /home/share/stream1.ffm FileMaxSize 6656M ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 ################################################################## # Now you can define each stream which will be generated from the # original audio and video stream. Each format has a filename (here # 'test1.mpg'). FFServer will send this stream when answering a # request containing this filename. # coming from live feed 'feed1' Feed feed1.ffm # Format of the stream : you can choose among: # mpeg : MPEG-1 multiplexed video and audio # mpegvideo : only MPEG-1 video # mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) # ogg : Ogg format (Vorbis audio codec) # rm : RealNetworks-compatible stream. Multiplexed audio and video. # ra : RealNetworks-compatible stream. Audio only. # mpjpeg : Multipart JPEG (works with Netscape without any plugin) # jpeg : Generate a single JPEG image. # asf : ASF compatible streaming (Windows Media Player format). # swf : Macromedia Flash compatible stream # avi : AVI format (MPEG-4 video, MPEG audio sound) Format mpeg # Bitrate for the audio stream. Codecs usually support only a few # different bitrates. AudioBitRate 192 # Number of audio channels: 1 = mono, 2 = stereo AudioChannels 2 # Sampling frequency for audio. When using low bitrates, you should # lower this frequency to 22050 or 11025. The supported frequencies # depend on the selected audio codec. AudioSampleRate 44100 # Bitrate for the video stream VideoBitRate 1000 # Ratecontrol buffer size VideoBufferSize 40 # Number of frames per second VideoFrameRate 25 # Size of the video frame: WxH (default: 160x128) # The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga, # qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga, # wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720, # hd1080 VideoSize 160x128 # Transmit only intra frames (useful for low bitrates, but kills frame rate). #VideoIntraOnly # If non-intra only, an intra frame is transmitted every VideoGopSize # frames. Video synchronization can only begin at an intra frame. VideoGopSize 12 # More MPEG-4 parameters # VideoHighQuality # Video4MotionVector # Choose your codecs: #AudioCodec mp2 #VideoCodec mpeg1video # Suppress audio #NoAudio # Suppress video #NoVideo #VideoQMin 3 #VideoQMax 31 # Set this to the number of seconds backwards in time to start. Note that # most players will buffer 5-10 seconds of video, and also you need to allow # for a keyframe to appear in the data stream. #Preroll 15 # ACL: # You can allow ranges of addresses (or single addresses) #ACL ALLOW # You can deny ranges of addresses (or single addresses) #ACL DENY # You can repeat the ACL allow/deny as often as you like. It is on a per # stream basis. The first match defines the action. If there are no matches, # then the default is the inverse of the last ACL statement. # # Thus 'ACL allow localhost' only allows access from localhost. # 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and # allow everybody else. ################################################################## # Example streams # Multipart JPEG # #Feed feed1.ffm #Format mpjpeg #VideoFrameRate 2 #VideoIntraOnly #NoAudio #Strict -1 # # Single JPEG # #Feed feed1.ffm #Format jpeg #VideoFrameRate 2 #VideoIntraOnly ##VideoSize 352x240 #NoAudio #Strict -1 # # Flash Feed feed1.ffm Format swf VideoFrameRate 25 #VideoIntraOnly VideoCodec libx264 AudioCodec libmp3lame VideoSize hd720 VideoBitRate 1200 #NoAudio ACL allow 192.168.0.0 255.255.255.0 # ASF compatible # #Feed feed1.ffm #Format asf #VideoFrameRate 15 #VideoSize 352x240 #VideoBitRate 256 #VideoBufferSize 40 #VideoGopSize 30 #AudioBitRate 64 #StartSendOnKey # # MP3 audio # #Feed feed1.ffm #Format mp2 #AudioCodec mp3 #AudioBitRate 64 #AudioChannels 1 #AudioSampleRate 44100 #NoVideo # # Ogg Vorbis audio # #Feed feed1.ffm #Title "Stream title" #AudioBitRate 64 #AudioChannels 2 #AudioSampleRate 44100 #NoVideo # # Real with audio only at 32 kbits # #Feed feed1.ffm #Format rm #AudioBitRate 32 #NoVideo #NoAudio # # Real with audio and video at 64 kbits # #Feed feed1.ffm #Format rm #AudioBitRate 32 #VideoBitRate 128 #VideoFrameRate 25 #VideoGopSize 25 #NoAudio # ################################################################## # Local Streams Feed ten.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 Feed eleven.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 Feed one.ffm Format mp2 ACL allow 192.168.0.0 255.255.255.0 ACL allow 127.0.0.1 ################################################################## # A stream coming from a file: you only need to set the input # filename and optionally a new format. Supported conversions: # AVI -> ASF # #File "/usr/local/httpd/htdocs/tlive.rm" #NoAudio # # #File "/usr/local/httpd/htdocs/test.asf" #NoAudio #Author "Me" #Copyright "Super MegaCorp" #Title "Test stream from disk" #Comment "Test comment" # ################################################################## # RTSP examples # # You can access this stream with the RTSP URL: # rtsp://localhost:5454/test1-rtsp.mpg # # A non-standard RTSP redirector is also created. Its URL is: # http://localhost:8090/test1-rtsp.rtsp # #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" # ################################################################## # SDP/multicast examples # # If you want to send your stream in multicast, you must set the # multicast address with MulticastAddress. The port and the TTL can # also be set. # # An SDP file is automatically generated by ffserver by adding the # 'sdp' extension to the stream name (here # http://localhost:8090/test1-sdp.sdp). You should usually give this # file to your player to play the stream. # # The 'NoLoop' option can be used to avoid looping when the stream is # terminated. # #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" #MulticastAddress 224.124.0.1 #MulticastPort 5000 #MulticastTTL 16 #NoLoop # ################################################################## # Special streams # Server status Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico # Redirect index.html to the appropriate site URL http:/// From stevenliu88 at gmail.com Wed Nov 30 05:51:52 2011 From: stevenliu88 at gmail.com (steven liu) Date: Wed, 30 Nov 2011 12:51:52 +0800 Subject: [FFmpeg-user] web stream video conversion on the fly without downloading Message-ID: Dear All, I am wondering whether FFMPEG has the web stream video conversion on the fly capability. What I want to do is to receive online web video stream from sockets, convert to another format/size/bit rate on the fly without waiting until the whole video file is downloaded completely. For example, I receive 1 sec video and start converting for this 1-sec video content. Then, I go to receive another 1 sec and start converting another 1-sec video content. Is it possible to do this using FFMPEG or other tools? Many thanks. Leo From ffmpeg-user at herveybayaustralia.com.au Wed Nov 30 06:02:49 2011 From: ffmpeg-user at herveybayaustralia.com.au (Da Rock) Date: Wed, 30 Nov 2011 15:02:49 +1000 Subject: [FFmpeg-user] web stream video conversion on the fly without downloading In-Reply-To: References: Message-ID: <4ED5B8F9.2030003@herveybayaustralia.com.au> On 11/30/11 14:51, steven liu wrote: > Dear All, > > I am wondering whether FFMPEG has the web stream video conversion on the > fly capability. What I want to do is to receive online web video stream > from sockets, convert to another format/size/bit rate on the fly without > waiting until the whole video file is downloaded completely. For example, I > receive 1 sec video and start converting for this 1-sec video content. > Then, I go to receive another 1 sec and start converting another 1-sec > video content. Is it possible to do this using FFMPEG or other tools? Many > thanks. > I believe that is the point of it. You can input from many sources (including weburl) convert and view on the fly such as: ffmpeg -i | - Essentially that is what ffserver does- it connects endpoints like this automagically so to speak. One ffmpeg reads the source to ffserver, and ffserver calls another to write it to the remote. HTH From jintonglei at gmail.com Wed Nov 30 06:58:27 2011 From: jintonglei at gmail.com (jintonglei) Date: Wed, 30 Nov 2011 13:58:27 +0800 Subject: [FFmpeg-user] problems of compile Android variant References: <201111291713492187111@gmail.com>, <201111291322.51669.gkinsey@ad-holdings.co.uk> Message-ID: <201111301358241884675@gmail.com> >> Hello, >> I tried to compile ffmpeg for Android but fail, it said: >> mips-linux-gnu-gcc is unable to create an executable file. >> C compiler test failed. >> >> And the configuration is: >> ./configure --arch=mips --cross-prefix=mips-linux-gnu- >> --extra-ldflags=-static extra-cflags=-static --enable-cross-compile >> --target-os=linux > --disable-static --enable-shared -- > sysroot=${ANDROID_NDK}/platforms/android-8/arch-arm --arch=arm --target- > os=linux --cross-prefix=${ANDROID_NDK}/toolchains/arm-linux- > androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi- --disable- > symver --cc="/usr/bin/ccache ${ANDROID_NDK}/toolchains/arm-linux- > androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-gcc" -- > enable-small > There are also some minor modifications needed to the source, but those are > for ARM assembler and so shouldn't be needed for you. thanks a lot! But it still complain "...is unable to create an executable file. C compiler test failed", no matter on arm-abi or mips-abi. Any more tips? ------------------ jintonglei 2011-11-30 From klaus.kudielka at gmx.net Wed Nov 30 07:30:07 2011 From: klaus.kudielka at gmx.net (Klaus Kudielka) Date: Wed, 30 Nov 2011 07:30:07 +0100 Subject: [FFmpeg-user] Application provided invalid, non monotonically increasing dts to muxer In-Reply-To: References: <20111129203824.GA13472@mars> Message-ID: <20111130063007.191380@gmx.net> > > ---- Two examples for NOT WORKING git masker: > > > > [0]$ /opt/ffmpeg-git/bin/ffmpeg -i x.mkv -map 0 -c copy -y /tmp/x.mkv > > [...] > > > ---- Counter-example of working 0.8.7: > > > > [1]$ /opt/ffmpeg/bin/ffmpeg -i x.mkv" -map 0.0 -vcodec copy -an -sn -y > > /tmp/x.mkv -map 0.1 -acodec copy -newaudio -map 0.2 -acodec copy > -newaudio > > -map 0.3 -scodec copy -newsubtitle -map 0.4 -scodec copy -newsubtitle > -map 0.5 > > -scodec copy -newsubtitle -map 0.6 -scodec copy -newsubtitle > > The command lines look different... Well, due to the new command line syntax in git master they HAVE to differ, in order to achieve the same behaviour!? (-map and -codec now handle all streams instead of only one, removal of -newaudio and -newsubtitle options which were needed before) Or am I missing something? Anyway the 'complicated' version of the git master command line (mapping & specifying codec for all streams individually) shows exactly the same behaviour: /opt/ffmpeg-git/bin/ffmpeg -i x.mkv -map 0:v:0 -c:v:0 copy -map 0:a:0 -c:a:0 copy -map 0:a:1 -c:a:1 copy -map 0:s:0 -c:s:0 copy -map 0:s:1 -c:s:1 copy -map 0:s:2 -c:s:2 copy -map 0:s:3 -c:s:3 copy /tmp/x.mkv [...] [matroska @ 0x2abc6c0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: -83 >= -83 av_interleaved_write_frame(): Invalid argument > Could you provide a sample and / or find the version introducing the > problem? I will try to find the version introducing the problem. Will probably take a few evenings. Klaus -- NEU: FreePhone - 0ct/min Handyspartarif mit Geld-zur?ck-Garantie! Jetzt informieren: http://www.gmx.net/de/go/freephone From redtux1 at gmail.com Wed Nov 30 07:59:38 2011 From: redtux1 at gmail.com (Mike Martin) Date: Wed, 30 Nov 2011 06:59:38 +0000 Subject: [FFmpeg-user] Problem creating DVD menu mpeg Message-ID: As people probably know a DVD menu needs to be a DVD compliant mpeg video with an audio stream previous to upgrading to F16 and ffmpeg-0.8.5 this worked ffmpeg -v 2 -i /storage/burn/dvd_tmp/2-%d.png -f s16le -i /dev/zero -s 720x568 -t 15 -shortest -qscale 4 -acodec mp2 -ab 64k -ar 48000 -ac 2 -aspect 4:3 -y test.avi (a group of png files combned with silent audio) however now this is what I get ffmpeg version 0.8.5, Copyright (c) 2000-2011 the FFmpeg developers built on Oct 27 2011 21:33:51 with gcc 4.6.1 20111003 (Red Hat 4.6.1-10) 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' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --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 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [image2 @ 0x8db57c0] max_analyze_duration 5000000 reached at 5000000 Input #0, image2, from '/storage/burn/dvd_tmp/2-%d.png': Duration: 00:00:15.00, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb24, 204x170, 25 fps, 25 tbr, 25 tbn, 25 tbc Ignoring attempt to set invalid timebase for st:0 [s16le @ 0x8db6fe0] Estimating duration from bitrate, this may be inaccurate Input #1, s16le, from '/dev/zero': Duration: N/A, start: 0.000000, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 1 channels, s16 Incompatible pixel format 'rgb24' for codec 'mpeg4', auto-selecting format 'yuv420p' [buffer @ 0x8dd5640] w:204 h:170 pixfmt:rgb24 tb:1/1000000 sar:0/1 sws_param: [scale @ 0x8dd9ea0] w:204 h:170 fmt:rgb24 -> w:720 h:568 fmt:yuv420p flags:0x4 Output #0, avi, to 'test.avi': Metadata: ISFT : Lavf53.4.0 Stream #0.0: Video: mpeg4, yuv420p, 720x568 [PAR 142:135 DAR 4:3], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop, [?] for help Floating point exception (core dumped) any ideas (the avi is a halfway house) From stevenliu88 at gmail.com Wed Nov 30 08:13:49 2011 From: stevenliu88 at gmail.com (steven liu) Date: Wed, 30 Nov 2011 15:13:49 +0800 Subject: [FFmpeg-user] web stream video conversion on the fly without downloading In-Reply-To: <4ED5B8F9.2030003@herveybayaustralia.com.au> References: <4ED5B8F9.2030003@herveybayaustralia.com.au> Message-ID: Thanks for reply. Is it possible to do this programmatically? I am able to receive the video byte stream (from socket) and store it into a tempary memory buffer. I plan to convert the buffer content to another format. The question is how much bytes I need to receive for each buffer before I start calling FFMPEG to convert? The problem is how to cut the received video byte stream to different parts for conversion. Tks. On Wed, Nov 30, 2011 at 1:02 PM, Da Rock < ffmpeg-user at herveybayaustralia.com.au> wrote: > On 11/30/11 14:51, steven liu wrote: > >> Dear All, >> >> I am wondering whether FFMPEG has the web stream video conversion on the >> fly capability. What I want to do is to receive online web video stream >> from sockets, convert to another format/size/bit rate on the fly without >> waiting until the whole video file is downloaded completely. For example, >> I >> receive 1 sec video and start converting for this 1-sec video content. >> Then, I go to receive another 1 sec and start converting another 1-sec >> video content. Is it possible to do this using FFMPEG or other tools? Many >> thanks. >> >> I believe that is the point of it. You can input from many sources > (including weburl) convert and view on the fly such as: > > ffmpeg -i | - > > Essentially that is what ffserver does- it connects endpoints like this > automagically so to speak. One ffmpeg reads the source to ffserver, and > ffserver calls another to write it to the remote. > > HTH > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From dashing.meng at gmail.com Wed Nov 30 10:43:06 2011 From: dashing.meng at gmail.com (littlebat) Date: Wed, 30 Nov 2011 17:43:06 +0800 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: <20111129151714.1556dab3.dashing.meng@gmail.com> <20111129175945.bb343702.dashing.meng@gmail.com> Message-ID: <20111130174306.6e0bce48.dashing.meng@gmail.com> On Wed, 30 Nov 2011 10:03:18 +0700 aris sirajawali wrote: > thanks for the response from friends all .. > but why do I still get an error message like this * ERROR: not found > ** libopencore_amrnb .*.. > Please help to send the reference to install ffmpeg is complete .. > > > > > http://sourceforge.net/projects/opencore-amr/files/ > > > > You need install opencore-amr(for amrnb encoding) or vo-amrwbenc(for > > amrwb encoding) first. You can search google for it. Do you make sure installing libopencore_amrnb? What is the result when execute "ls /usr/lib/*opencore* /usr/local/lib/*opencore*"? Do you compile ffmpeg from source again? Do you compile ffmpeg with configure option "--enable-libopencore-amrnb"? Can you find string such as "DEA libopencore_amrnb OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band" in the result of command "ffmpeg -codecs | grep opencore"? When did you meet the error message you mentioned? Compile ffmpeg? or execute ffmpeg? From ffmpeg-user at herveybayaustralia.com.au Wed Nov 30 10:49:38 2011 From: ffmpeg-user at herveybayaustralia.com.au (Da Rock) Date: Wed, 30 Nov 2011 19:49:38 +1000 Subject: [FFmpeg-user] web stream video conversion on the fly without downloading In-Reply-To: References: <4ED5B8F9.2030003@herveybayaustralia.com.au> Message-ID: <4ED5FC32.2060407@herveybayaustralia.com.au> On 11/30/11 17:13, steven liu wrote: > Thanks for reply. Is it possible to do this programmatically? I am able to > receive the video byte stream (from socket) and store it into a tempary > memory buffer. I plan to convert the buffer content to another format. The > question is how much bytes I need to receive for each buffer before I start > calling FFMPEG to convert? The problem is how to cut the received video > byte stream to different parts for conversion. Tks. I don't think you need to worry about any of that. FFmpeg receives from any input (provided its media) and converts "on the fly"; in other words it will convert as it gets the data. Programmatically shouldn't be a problem either- many apps use ffmpeg in the background. For instance, I'm taking input from gstreamer fdsink or udp:// and I can split and convert (I'm not, but I have tried this) a mp2/mpeg-ts to x264 and output to file or a url in mpeg-ts format. No buffer required and cpu usage is extremely low. You can split the different streams encoded in the format to different files or whatever. Check out -map options. But ffmpeg can encode *all* video streams to a specific codec and *all* audio streams to a specific codec, or independently to different codecs, and output to one file or url/socket or many. All I can suggest is to just play: you'll never know unless you try :) > > On Wed, Nov 30, 2011 at 1:02 PM, Da Rock< > ffmpeg-user at herveybayaustralia.com.au> wrote: > >> On 11/30/11 14:51, steven liu wrote: >> >>> Dear All, >>> >>> I am wondering whether FFMPEG has the web stream video conversion on the >>> fly capability. What I want to do is to receive online web video stream >>> from sockets, convert to another format/size/bit rate on the fly without >>> waiting until the whole video file is downloaded completely. For example, >>> I >>> receive 1 sec video and start converting for this 1-sec video content. >>> Then, I go to receive another 1 sec and start converting another 1-sec >>> video content. Is it possible to do this using FFMPEG or other tools? Many >>> thanks. >>> >>> I believe that is the point of it. You can input from many sources >> (including weburl) convert and view on the fly such as: >> >> ffmpeg -i | - >> >> Essentially that is what ffserver does- it connects endpoints like this >> automagically so to speak. One ffmpeg reads the source to ffserver, and >> ffserver calls another to write it to the remote. >> >> HTH >> ______________________________**_________________ >> 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 cehoyos at ag.or.at Wed Nov 30 11:07:31 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 30 Nov 2011 10:07:31 +0000 (UTC) Subject: [FFmpeg-user] Problem creating DVD menu mpeg References: Message-ID: Mike Martin gmail.com> writes: > ffmpeg -v 2 -i /storage/burn/dvd_tmp/2-%d.png -f s16le -i /dev/zero > -s 720x568 -t 15 -shortest -qscale 4 -acodec mp2 -ab 64k -ar 48000 > -ac 2 -aspect 4:3 -y test.avi > > (a group of png files combned with silent audio) > > however now this is what I get > ffmpeg version 0.8.5, Copyright (c) 2000-2011 the FFmpeg developers This has been fixed in current git head (this is the suggested version if you are a user and not a distributor) and all current releases. Carl Eugen From cehoyos at ag.or.at Wed Nov 30 11:08:50 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 30 Nov 2011 10:08:50 +0000 (UTC) Subject: [FFmpeg-user] no connections - ffserver rendered useless References: <4ED5939F.5070508@herveybayaustralia.com.au> Message-ID: Da Rock herveybayaustralia.com.au> writes: > FreeBSD 8.1-RELEASE-p1 FreeBSD 8.1-RELEASE-p1 > ffmpeg version 0.7.7, Copyright (c) 2000-2011 the FFmpeg developers Please test current git master. Carl Eugen From cehoyos at ag.or.at Wed Nov 30 11:11:31 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 30 Nov 2011 10:11:31 +0000 (UTC) Subject: [FFmpeg-user] problems of compile Android variant References: <201111291713492187111@gmail.com>, <201111291322.51669.gkinsey@ad-holdings.co.uk> <201111301358241884675@gmail.com> Message-ID: jintonglei gmail.com> writes: > But it still complain "...is unable to create an executable file. C compiler > test failed", no matter on arm-abi or mips-abi. > > Any more tips? The last lines of config.log should help you finding the problem. Carl Eugen From cehoyos at ag.or.at Wed Nov 30 11:10:06 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 30 Nov 2011 10:10:06 +0000 (UTC) Subject: [FFmpeg-user] Default audio bitrate changed? References: <4ED56D87.7030707@gmail.com> Message-ID: Lucien Pullen gmail.com> writes: > The manual states that the default audio bitrate is 64k. However, it > seems to have changed to 128k silently. Is this change going to stick > (should the manual be updated), or is this just temporary? The default bitrate has changed, could you point me to the outdated documentation? Thank you, Carl Eugen From mediastream at gmail.com Wed Nov 30 15:24:44 2011 From: mediastream at gmail.com (Dennis) Date: Wed, 30 Nov 2011 09:24:44 -0500 Subject: [FFmpeg-user] vertical lines noise in scaled raw video. In-Reply-To: References: <4EBAA4E7.9070706@googlemail.com> Message-ID: On Wed, Nov 9, 2011 at 1:52 PM, Dennis wrote: > > > On Wed, Nov 9, 2011 at 12:47 PM, Carl Eugen Hoyos wrote: > >> Dennis gmail.com> writes: >> >> > Here's the screenshot with "find edges" filter applied by ImageJ : >> > http://www.mediafire.com/?0hesdf21e6a2ok2 >> >> I can see the artefacts now (don't know why I searched for "red" lines). >> Do you think they are in the original video and amplified by the encoding >> (and >> you are searching for a filter that suppresses them), or is FFmpeg >> producing >> them (which may be a serious bug)? >> >> Carl Eugen_______________________________________________ >> > > well, look at the un-scaled y4m output screenshot here: > http://www.mediafire.com/i/?5m3seip1z8zb4mf > its solid black, i'd expect the same from the scaled screenshot. > It seems that the problem is with FFmpeg 0.8.7(release) scaler - it adds noise. Nov 29 2011 GIT (0.8.7.git-b6ffe44) works fine, no detectable noise. Here are some samples: http://doom10.org/index.php?topic=2034 Thank you all. From de.techno at gmail.com Wed Nov 30 16:43:12 2011 From: de.techno at gmail.com (dE .) Date: Wed, 30 Nov 2011 21:13:12 +0530 Subject: [FFmpeg-user] iPod 640x480 In-Reply-To: <4bef86dbd37303a551ca0997cc212225@aleksandrsolzhenitsyn.net> References: <4bef86dbd37303a551ca0997cc212225@aleksandrsolzhenitsyn.net> Message-ID: <4ED64F10.6000407@gmail.com> On 11/30/11 05:59, peace at aleksandrsolzhenitsyn.net wrote: > Can someone explain to me why my iPod can't handle videos at 640x480 > when the spec's for it say it can? > > On Apple's website it does say about the display- 320-by-240-pixel > resolution at 163 pixels per inch. But, it does say in the spec's it > can handle 640x480. > > Videos at 320 x 240 work perfectly, thanks to all of your help, but > for whatever reason the 640's don't. > > Being that I'm so stupid about this sort of thing I'm sure there's an > easy explanation. > > > The spec's for the iPod Classic (160G) are below. > > > > > > > H.264 video, up to 1.5 Mbps, 640 by 480 pixels, 30 frames per second, > Low-Complexity version of the H.264 Baseline Profile with AAC-LC audio > up to 160 Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file > formats; H.264 video, up to 2.5 Mbps, 640 by 480 pixels, 30 frames per > second, Baseline Profile up to Level 3.0 with AAC-LC audio up to 160 > Kbps, 48kHz, stereo audio in .m4v, .mp4, and .mov file formats; MPEG-4 > video, up to 2.5 Mbps, 640 by 480 pixels, 30 frames per second, Simple > Profile with AAC-LC audio up to 160 Kbps, 48kHz, stereo audio in .m4v, > .mp4, and .mov file formats > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user That's cause it doesn't have a powerful processor to rescale and play the video (one of the may reasons I don't prefer Apple). You may like to read this -- http://delogics.blogspot.com/2011/11/ffmpeg-maintain-aspect-ratio-with-fixed.html If you want, I can make the ffmpeg presets. From de.techno at gmail.com Wed Nov 30 16:53:00 2011 From: de.techno at gmail.com (dE .) Date: Wed, 30 Nov 2011 21:23:00 +0530 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: <20111129151714.1556dab3.dashing.meng@gmail.com> <20111129175945.bb343702.dashing.meng@gmail.com> Message-ID: <4ED6515C.9000203@gmail.com> On 11/30/11 08:33, aris sirajawali wrote: > thanks for the response from friends all .. > but why do I still get an error message like this * ERROR: not found ** > libopencore_amrnb .*.. > Please help to send the reference to install ffmpeg is complete .. > > > >> http://sourceforge.net/projects/opencore-amr/files/ >> >> You need install opencore-amr(for amrnb encoding) or vo-amrwbenc(for >> amrwb encoding) first. >> _______________________________________________ >> 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 Forget compiling. you'll probably never make it. Download from - https://github.com/stvs/ffmpeg-static And no, I've never tried it. What distro are you using? From de.techno at gmail.com Wed Nov 30 16:58:01 2011 From: de.techno at gmail.com (dE .) Date: Wed, 30 Nov 2011 21:28:01 +0530 Subject: [FFmpeg-user] Problem creating DVD menu mpeg In-Reply-To: References: Message-ID: <4ED65289.8050409@gmail.com> On 11/30/11 12:29, Mike Martin wrote: > As people probably know a DVD menu needs to be a DVD compliant mpeg > video with an audio stream > > previous to upgrading to F16 and ffmpeg-0.8.5 this worked > > ffmpeg -v 2 -i /storage/burn/dvd_tmp/2-%d.png -f s16le -i /dev/zero > -s 720x568 -t 15 -shortest -qscale 4 -acodec mp2 -ab 64k -ar 48000 > -ac 2 -aspect 4:3 -y test.avi > > (a group of png files combned with silent audio) > > however now this is what I get > ffmpeg version 0.8.5, Copyright (c) 2000-2011 the FFmpeg developers > built on Oct 27 2011 21:33:51 with gcc 4.6.1 20111003 (Red Hat 4.6.1-10) > 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' --enable-bzlib > --enable-libcelt --enable-libdc1394 --enable-libdirac > --enable-libfreetype --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 51. 9. 1 / 51. 9. 1 > libavcodec 53. 7. 0 / 53. 7. 0 > libavformat 53. 4. 0 / 53. 4. 0 > libavdevice 53. 1. 1 / 53. 1. 1 > libavfilter 2. 23. 0 / 2. 23. 0 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [image2 @ 0x8db57c0] max_analyze_duration 5000000 reached at 5000000 > Input #0, image2, from '/storage/burn/dvd_tmp/2-%d.png': > Duration: 00:00:15.00, start: 0.000000, bitrate: N/A > Stream #0.0: Video: png, rgb24, 204x170, 25 fps, 25 tbr, 25 tbn, 25 tbc > Ignoring attempt to set invalid timebase for st:0 > [s16le @ 0x8db6fe0] Estimating duration from bitrate, this may be inaccurate > Input #1, s16le, from '/dev/zero': > Duration: N/A, start: 0.000000, bitrate: N/A > Stream #1.0: Audio: pcm_s16le, 1 channels, s16 > Incompatible pixel format 'rgb24' for codec 'mpeg4', auto-selecting > format 'yuv420p' > [buffer @ 0x8dd5640] w:204 h:170 pixfmt:rgb24 tb:1/1000000 sar:0/1 sws_param: > [scale @ 0x8dd9ea0] w:204 h:170 fmt:rgb24 -> w:720 h:568 fmt:yuv420p flags:0x4 > Output #0, avi, to 'test.avi': > Metadata: > ISFT : Lavf53.4.0 > Stream #0.0: Video: mpeg4, yuv420p, 720x568 [PAR 142:135 DAR 4:3], > q=2-31, 200 kb/s, 25 tbn, 25 tbc > Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s > Stream mapping: > Stream #0.0 -> #0.0 > Stream #1.0 -> #0.1 > Press [q] to stop, [?] for help > Floating point exception (core dumped) > > > any ideas (the avi is a halfway house) > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Why not use graphical application like devede and dvdstyler? I think even k3b is capable of doing it. From cehoyos at ag.or.at Wed Nov 30 17:35:37 2011 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 30 Nov 2011 16:35:37 +0000 (UTC) Subject: [FFmpeg-user] problem encode mov to 3gp References: <20111129151714.1556dab3.dashing.meng@gmail.com> <20111129175945.bb343702.dashing.meng@gmail.com> <4ED6515C.9000203@gmail.com> Message-ID: dE . gmail.com> writes: > github.com/stvs/ffmpeg-static The script on this side is completely outdated (and uses an old version of FFmpeg with a very high number of known and fixed bugs). Please do not suggest not to compile FFmpeg on this list: It is the supported and suggested way of getting a working version! > And no, I've never tried it. I strongly recommend not trying it. Carl Eugen From phil_rhodes at rocketmail.com Wed Nov 30 18:38:31 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Wed, 30 Nov 2011 17:38:31 -0000 Subject: [FFmpeg-user] non-redistributable non-free In-Reply-To: <4ED58044.6050503@thelounge.net> References: <4ED58044.6050503@thelounge.net> Message-ID: > if you distribute a self written SHELL-SCRIPT there is no code nor > a license of a single piece of ffmpeg affected in any way Neither is there if he just gives the guy the binary, in any even vaguely realistic sense - but he's not allowed to do that. P From rogerdpack2 at gmail.com Wed Nov 30 19:12:21 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 30 Nov 2011 11:12:21 -0700 Subject: [FFmpeg-user] feature request: user editable list of filters Message-ID: Hello. While thinking about writing some filters, the other day, the thought came to me that it might be nice to have some canonical (user editable) "list" of the various filters people have made. Is there anything like this? If not perhaps one could be made. Thank you. -r From lou at lrcd.com Wed Nov 30 19:23:36 2011 From: lou at lrcd.com (Lou) Date: Wed, 30 Nov 2011 09:23:36 -0900 Subject: [FFmpeg-user] iPod 640x480 In-Reply-To: <4bef86dbd37303a551ca0997cc212225@aleksandrsolzhenitsyn.net> References: <4bef86dbd37303a551ca0997cc212225@aleksandrsolzhenitsyn.net> Message-ID: <20111130092336.44cb1c37@lrcd.com> On Tue, 29 Nov 2011 17:29:48 -0700 peace at aleksandrsolzhenitsyn.net wrote: > Can someone explain to me why my iPod can't handle videos at 640x480 > when the spec's for it say it can? Did you encode the videos with ffmpeg? If yes, then show your command and the complete console output. Otherwise we have to make guesses with little information to work with. Also, please make a new message for each question. It appears that you replied to "FFMpeg with other software" and changed the subject. The "In-Reply-To" and "References" headers are preserved when you reply to a message and more capable mail clients will use this info to properly thread the messages. Therefore, your question was still threaded under "FFMpeg with other software". See your message in the archives page if my overly verbose explanation didn't make sense: http://ffmpeg.org/pipermail/ffmpeg-user/2011-November/thread.html From lou at lrcd.com Wed Nov 30 20:03:09 2011 From: lou at lrcd.com (Lou) Date: Wed, 30 Nov 2011 10:03:09 -0900 Subject: [FFmpeg-user] feature request: user editable list of filters In-Reply-To: References: Message-ID: <20111130100309.3097ba37@lrcd.com> On Wed, 30 Nov 2011 11:12:21 -0700 Roger Pack wrote: > Hello. > While thinking about writing some filters, the other day, the thought > came to me that it might be nice to have some canonical (user > editable) "list" of the various filters people have made. > Is there anything like this? > If not perhaps one could be made. > Thank you. > -r Good idea. Here's a start: https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide Feel free to monkey around with my initial formatting and examples if you want to. I'm assuming any registered [1] user can edit the page, but I'm not totally sure. Do you prefer full command examples, or just the -vf part? [1] https://ffmpeg.org/trac/ffmpeg/register From danny.sadai at essexcricket.org.uk Wed Nov 30 11:55:19 2011 From: danny.sadai at essexcricket.org.uk (Danny Sadai) Date: Wed, 30 Nov 2011 10:55:19 +0000 Subject: [FFmpeg-user] Error messages Message-ID: <0aa3ed02-34ec-47fd-909a-0fa6e3f49344@essexcricket.org.uk> Hi there, every time I try to view a folder containing a video file I get an error message: "COM Surrogate has stopped working" (I have attached the full error message and my system details), when I follow the MS windows error link it tells me I have an FFmpeg fault, I did not even know that FFmpeg was installed, can any one PLEASE HELP? Danny Sadai Deputy Safety Officer/Maintenance Assistant Email: danny.sadai at essexcricket.org.uk Telephone: 01245 254037 / Mobile: 07971 451643 / Fax: 01245 254021 Address: The Ford County Ground, New Writtle Street, Chelmsford, Essex, CM2 0PG. -------------- next part -------------- A non-text attachment was scrubbed... Name: image006.jpg Type: image/jpeg Size: 6288 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image005.jpg Type: image/jpeg Size: 12941 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 16205 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 5378 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 4568 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 16193 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: FFmpeg error message.doc Type: application/msword Size: 628224 bytes Desc: not available URL: From nathan.stocks at gmail.com Wed Nov 30 20:05:28 2011 From: nathan.stocks at gmail.com (Nathan) Date: Wed, 30 Nov 2011 12:05:28 -0700 Subject: [FFmpeg-user] Linking help? Message-ID: Can anyone point me in the right direction to figure out how to correct linking errors like the one below? ld: dist/linux-x86-64/libavbin.so.8: version node not found for symbol av_dup_packet LIBAVFORMAT_52 ld: failed to set dynamic section sizes: Bad value I posted full details on the ffmpeg libav-user mailing list...which list seems to be deserted (so I came here): http://permalink.gmane.org/gmane.comp.video.ffmpeg.libav.user/7221 ~ Nathan From rogerdpack2 at gmail.com Wed Nov 30 20:42:34 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 30 Nov 2011 12:42:34 -0700 Subject: [FFmpeg-user] feature request: user editable list of filters In-Reply-To: <20111130100309.3097ba37@lrcd.com> References: <20111130100309.3097ba37@lrcd.com> Message-ID: > > Good idea. Here's a start: > https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide > > Feel free to monkey around with my initial formatting and examples if > you want to. I'm assuming any registered [1] user can edit the page, > but I'm not totally sure. I was able to edit it. > Do you prefer full command examples, or just the -vf part? Full are easier on beginners, I think...(like myself) :) -roger- From nathan.stocks at gmail.com Wed Nov 30 21:41:10 2011 From: nathan.stocks at gmail.com (Nathan) Date: Wed, 30 Nov 2011 13:41:10 -0700 Subject: [FFmpeg-user] Linking help? In-Reply-To: References: Message-ID: Ick, the archive replaced "@" with " ". Here's the actual error, as well as a pastebin of it: ld: dist/linux-x86-64/libavbin.so.8: version node not found for symbol av_dup_packet at LIBAVFORMAT_52 ld: failed to set dynamic section sizes: Bad value http://pastebin.com/xv0GmtSU ~ Nathan On Wed, Nov 30, 2011 at 12:05 PM, Nathan wrote: > Can anyone point me in the right direction to figure out how to > correct linking errors like the one below? > > ld: dist/linux-x86-64/libavbin.so.8: version node not found for symbol > av_dup_packet LIBAVFORMAT_52 > ld: failed to set dynamic section sizes: Bad value > > I posted full details on the ffmpeg libav-user mailing list...which > list seems to be deserted (so I came here): > > http://permalink.gmane.org/gmane.comp.video.ffmpeg.libav.user/7221 > > ~ Nathan From mike at redtux.org.uk Wed Nov 30 22:48:00 2011 From: mike at redtux.org.uk (Mike Martin) Date: Wed, 30 Nov 2011 21:48:00 +0000 Subject: [FFmpeg-user] Problem creating DVD menu mpeg In-Reply-To: <4ED65289.8050409@gmail.com> References: <4ED65289.8050409@gmail.com> Message-ID: Simple its part of my own graphical app which imo is better for what I want ?burn360? On Nov 30, 2011 3:58 PM, "dE ." wrote: > > On 11/30/11 12:29, Mike Martin wrote: >> >> As people probably know a DVD menu needs to be a DVD compliant mpeg >> video with an audio stream >> >> previous to upgrading to F16 and ffmpeg-0.8.5 this worked >> >> ffmpeg -v 2 -i /storage/burn/dvd_tmp/2-%d.png -f s16le -i /dev/zero >> -s 720x568 -t 15 -shortest -qscale 4 -acodec mp2 -ab 64k -ar 48000 >> -ac 2 -aspect 4:3 -y test.avi >> >> (a group of png files combned with silent audio) >> >> however now this is what I get >> ffmpeg version 0.8.5, Copyright (c) 2000-2011 the FFmpeg developers >> built on Oct 27 2011 21:33:51 with gcc 4.6.1 20111003 (Red Hat 4.6.1-10) >> 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' --enable-bzlib >> --enable-libcelt --enable-libdc1394 --enable-libdirac >> --enable-libfreetype --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 51. 9. 1 / 51. 9. 1 >> libavcodec 53. 7. 0 / 53. 7. 0 >> libavformat 53. 4. 0 / 53. 4. 0 >> libavdevice 53. 1. 1 / 53. 1. 1 >> libavfilter 2. 23. 0 / 2. 23. 0 >> libswscale 2. 0. 0 / 2. 0. 0 >> libpostproc 51. 2. 0 / 51. 2. 0 >> [image2 @ 0x8db57c0] max_analyze_duration 5000000 reached at 5000000 >> Input #0, image2, from '/storage/burn/dvd_tmp/2-%d.png': >> Duration: 00:00:15.00, start: 0.000000, bitrate: N/A >> Stream #0.0: Video: png, rgb24, 204x170, 25 fps, 25 tbr, 25 tbn, 25 tbc >> Ignoring attempt to set invalid timebase for st:0 >> [s16le @ 0x8db6fe0] Estimating duration from bitrate, this may be inaccurate >> Input #1, s16le, from '/dev/zero': >> Duration: N/A, start: 0.000000, bitrate: N/A >> Stream #1.0: Audio: pcm_s16le, 1 channels, s16 >> Incompatible pixel format 'rgb24' for codec 'mpeg4', auto-selecting >> format 'yuv420p' >> [buffer @ 0x8dd5640] w:204 h:170 pixfmt:rgb24 tb:1/1000000 sar:0/1 sws_param: >> [scale @ 0x8dd9ea0] w:204 h:170 fmt:rgb24 -> w:720 h:568 fmt:yuv420p flags:0x4 >> Output #0, avi, to 'test.avi': >> Metadata: >> ISFT : Lavf53.4.0 >> Stream #0.0: Video: mpeg4, yuv420p, 720x568 [PAR 142:135 DAR 4:3], >> q=2-31, 200 kb/s, 25 tbn, 25 tbc >> Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s >> Stream mapping: >> Stream #0.0 -> #0.0 >> Stream #1.0 -> #0.1 >> Press [q] to stop, [?] for help >> Floating point exception (core dumped) >> >> >> any ideas (the avi is a halfway house) >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > Why not use graphical application like devede and dvdstyler? I think even k3b is capable of doing it. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lou at lrcd.com Wed Nov 30 22:57:55 2011 From: lou at lrcd.com (Lou) Date: Wed, 30 Nov 2011 12:57:55 -0900 Subject: [FFmpeg-user] problem encode mov to 3gp In-Reply-To: References: Message-ID: <20111130125755.29ece79d@lrcd.com> On Tue, 29 Nov 2011 13:50:26 +0700 aris sirajawali wrote: > how to encode video from mov to 3gp with good quality and small > size .. I use the following command: *ffmpeg-i INPUt.MOV-acodec > libfaac-ab 12.2-ac 1-ar 8000-vcodec h263-s QCIF-r 10-b 128k > OUTPUT.3gp*. Does your command not work? You never said if it worked or not, and if it does not work then show the complete console output. Otherwise how are we to know what is wrong? > and if I make use *acodec amr_nb or amr_wb*, I get an > *unknown encoder*, is there an error on my ffmpeg command? Since I know nothing about the ffmpeg you are using it is hard to determine what the issue is. If you show your complete command and the complete console output then we can give suggestions. 3GP can handle AMR or AAC audio, so if you can encode AAC audio then you won't have to deal with AMR.