From tobias at tonstrom.de Sat Nov 1 09:38:44 2014 From: tobias at tonstrom.de (das.t) Date: Sat, 01 Nov 2014 09:38:44 +0100 Subject: [FFmpeg-user] Radiostream rip, w/o recoding, irregularly fails Message-ID: Hi listpeople, I am trying to swich my private radio recorder from using mplayer to ffmpeg as some tests have resulted in more usable files. Anyway, after having it running for a few weeks now I am facing an issue I have not yet found a solution. As extended internet searches did not help me either I am now seeking help here. The problem: irregularly I get only a few k sized files from one record job or it stops too early. At another time the same job completes as expected. I have searched for a way to increase the buffer cache but did not find it inside the man. Is audio grabbing not yet completely mature? This is the cmd I am using: ffmpeg -i $streamurl -vn -acodec copy -t 01:01:00 filename Any help appreciated. What cmd line are you folks using? I do NOT want to recode because I do not want to reduce the audio quality even further, just dumpstream. tobias -- Sent from my Android device with Squeaky Mail. Please excuse my brevity. From dopelabs at dubstep.fm Sat Nov 1 10:02:26 2014 From: dopelabs at dubstep.fm (DopeLabs) Date: Sat, 1 Nov 2014 02:02:26 -0700 Subject: [FFmpeg-user] Radiostream rip, w/o recoding, irregularly fails In-Reply-To: References: Message-ID: <3FC1FC23-2ABF-4685-BD89-FBE62A5AFAA5@dubstep.fm> can you provide any examples of the complete command line output of when it ?fails? and when is successful? can i ask what the source stream is? server type/audio codec? you might try ffmpeg -ss 0 -i infile -t 01:01:00.000 -c copy outfile > On Nov 1, 2014, at 1:38 AM, das.t wrote: > > Hi listpeople, > > I am trying to swich my private radio recorder from using mplayer to ffmpeg as some tests have resulted in more usable files. Anyway, after having it running for a few weeks now I am facing an issue I have not yet found a solution. As extended internet searches did not help me either I am now seeking help here. > > The problem: irregularly I get only a few k sized files from one record job or it stops too early. At another time the same job completes as expected. I have searched for a way to increase the buffer cache but did not find it inside the man. > > Is audio grabbing not yet completely mature? > > This is the cmd I am using: > > ffmpeg -i $streamurl -vn -acodec copy -t 01:01:00 filename > > Any help appreciated. What cmd line are you folks using? I do NOT want to recode because I do not want to reduce the audio quality even further, just dumpstream. > > tobias > -- > Sent from my Android device with Squeaky Mail. Please excuse my brevity. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From sylvain at lahiette.com Sat Nov 1 10:14:07 2014 From: sylvain at lahiette.com (SF) Date: Sat, 01 Nov 2014 10:14:07 +0100 Subject: [FFmpeg-user] Adding user-defined SEI messages thru libavcodec/libx264 Message-ID: <1420932.BDBbWgqoam@reik.site> Hi, I am searching a way to insert private SEI messages before each picture when encoding with x264. As far as i see from the libx264.c code, it is not supported up to now. Am i correct ? If yes, i am willing to push patch to support this, but do you have a preferred way to do this (ie thru medata field of AVFrame ? thru sidedata of AVFrame ? ...) Thanks for your hints, Regards, SF From cehoyos at ag.or.at Sat Nov 1 11:00:41 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 1 Nov 2014 10:00:41 +0000 (UTC) Subject: [FFmpeg-user] Radiostream rip, w/o recoding, irregularly fails References: Message-ID: das.t tonstrom.de> writes: > ffmpeg -i $streamurl -vn -acodec copy -t 01:01:00 filename (Actual command line and complete, uncut console output missing.) mplayer -dumpstream is the recommended way of recording streams, maybe -re helps. Carl Eugen From cehoyos at ag.or.at Sat Nov 1 11:03:40 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 1 Nov 2014 10:03:40 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg configure error References: <2b1bfeef.8ebbd.14965ed0930.Coremail.xuws@novel-supertv.com> Message-ID: xuws novel-supertv.com> writes: > link -nologo -out:./ffconf.NBRzPjuC.exe ./ffconf.dZgVeOUI.o > LINK : fatal error Did you check "which link.exe" as suggested on https://ffmpeg.org/platform.html ? Carl Eugen From cehoyos at ag.or.at Sat Nov 1 11:05:51 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 1 Nov 2014 10:05:51 +0000 (UTC) Subject: [FFmpeg-user] Automatically ensuring level-conformance for H.264 encodes References: <000001cff51a$84593f70$8d0bbe50$@gmail.com> Message-ID: Francois Visagie gmail.com> writes: > The main aim of the attached utilities is to automatically > ensure the H.264 level-conformance of arbitrary encodes. > In other words, the utilities make it unnecessary to guess > and/or discover the required settings through > trial-and-error test encodes. This sounds to me as if setting the H264 level through FFmpeg does not work. Is that correct? How can I reproduce this? Carl Eugen From cehoyos at ag.or.at Sat Nov 1 11:12:10 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 1 Nov 2014 10:12:10 +0000 (UTC) Subject: [FFmpeg-user] errors: Input buffer exhausted before END element found, error while decoding, Failed to update header with correct duration.. References: Message-ID: LD2K.com Litecoin Mining Pool gmail.com> writes: > [aac 0x317c7e0] Input buffer exhausted before END element found > Error while decoding stream #0:0: Invalid data found when processing input Can you record the stream (with wget) to provide an input file that allows us to reproduce the problem? Carl Eugen From cehoyos at ag.or.at Sat Nov 1 11:10:37 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sat, 1 Nov 2014 10:10:37 +0000 (UTC) Subject: [FFmpeg-user] Errors while muxing into .mkv from ip camera References: <1414764702.199801535@f301.i.mail.ru> Message-ID: Ddfsdf sdfsdf mail.ru> writes: > What's the reason of arising these errors in log? The reference decoder only decodes the first three frames of your input file so I suggest it is broken (reception or encoding errors). > ffmpeg.exe -f h264 -i video.raw video.mkv "-f h264" is unneeded. Carl Eugen From joolzg at btinternet.com Sat Nov 1 11:34:56 2014 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Sat, 1 Nov 2014 10:34:56 +0000 Subject: [FFmpeg-user] Example C code for encoding/muxing/streaming In-Reply-To: <1414589650.42830.YahooMailNeo@web87806.mail.ir2.yahoo.com> References: <1414436816.68983.YahooMailNeo@web161504.mail.bf1.yahoo.com> <201410281143.19964.cehoyos@ag.or.at> <1414589650.42830.YahooMailNeo@web87806.mail.ir2.yahoo.com> Message-ID: <1414838096.50904.YahooMailNeo@web87801.mail.ir2.yahoo.com> >________________________________ > From: JULIAN GARDNER >To: FFmpeg user questions >Sent: Wednesday, 29 October 2014, 14:34 >Subject: [FFmpeg-user] Example C code for encoding/muxing/streaming > > >Ive been playing with the code and have a little test project I want to try, a mosaic generator. > >Now Ive been looking through the examples code and tried a few but none seem to be able to stream out over multicast. > >Has anybody got code/knows of example code which shows how to setup a stream which will output a mpegts to a udp multicast address. > >I think i have most of the other code from the examples. > >The mosaic will work this way > >X streams in, running in threads which will send each I frame to the encoding thread, this thread will resize the incoming frame to the correct size and then put it in the correct place on the output frame. > >The output frame will then be encoded and streamed out, this is the part that i cannot find any code for. > >Ive tried ffmpeg for this but when dealing with 9 streams (3x3) the startup time is way to long and it does not auto restart a broken stream, which Im trying to overcome. > >Any help/involvment would be gratefully received > >joolz OK its working!!!!!, but i need some help on a little problem with my code. snippet below .... ??? /* initialize packet, set data to NULL, let the demuxer fill it */ ??? av_init_packet(&myframe->pkt); ??? inputSource->pkt.data = NULL; ??? inputSource->pkt.size = 0; ??? /* read frames from the file */ ??? while (!inputSource->quit && !stop_all_tasks && av_read_frame(inputSource->fmt_ctx, &inputSource->pkt) >= 0) { ??????? AVPacket orig_pkt = inputSource->pkt; ??????? do { ??????????? ret = decode_packet(&got_frame, 0, inputSource); ??????????? if (ret < 0) ??????????????? break; ??????????? inputSource->pkt.data += ret; ??????????? inputSource->pkt.size -= ret; ??????? } while (inputSource->pkt.size > 0); ??????? av_free_packet(&orig_pkt); ??? } I then added a callback but this never seems to be called!!! static int interrupt_cb(void *ctx) { ??? inputMosaic *inputSource = ctx; ??? printf( "."); fflush( stdout); ??? if (inputSource->quit || stop_all_tasks) ??????? return 1; ??? return 0; } and in the inputThread ??? inputSource->fmt_ctx->interrupt_callback.callback = interrupt_cb; ??? inputSource->fmt_ctx->interrupt_callback.opaque = myframe; Now as this is running in a thread it works well as long as THERE IS data. if no data then this loop just stays here and I cannot kill it when I need to switch the input stream or even close down. Anyone help on this. joolz From francois.visagie at gmail.com Sun Nov 2 11:04:22 2014 From: francois.visagie at gmail.com (Francois Visagie) Date: Sun, 2 Nov 2014 12:04:22 +0200 Subject: [FFmpeg-user] Automatically ensuring level-conformance for H.264 encodes In-Reply-To: References: <000001cff51a$84593f70$8d0bbe50$@gmail.com> Message-ID: <001d01cff684$69bc0740$3d3415c0$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos > Sent: 01 November 2014 12:06 > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] Automatically ensuring level-conformance for > H.264 encodes > > Francois Visagie gmail.com> writes: > > > The main aim of the attached utilities is to automatically ensure the > > H.264 level-conformance of arbitrary encodes. > > In other words, the utilities make it unnecessary to guess and/or > > discover the required settings through trial-and-error test encodes. > > This sounds to me as if setting the H264 level through FFmpeg does not work. > Is that correct? It's a little more subtle than that; I'll try to explain. 1. When using presets, in my testing (lib)x246 does not use '--level m.n'/' -x264opts level=m.n' to govern refs. In this scenario refs are governed by preset alone in accordance with the preset settings listed by x264 --fullhelp. (Although besides the point, (lib)x264 is actually prepared to write specified level metadata that differs from the level encoded to, at least when the latter is lower.) 2. The ffmpeg option '-level x.y' does set refs correctly, even when using libx264 presets. 3. However, '-level x.y' is undocumented, while the '-x264opts' pass-through mechanism is. https://trac.ffmpeg.org/ticket/3947#ticket addresses 1, 2 and 3. None of the above presents a problem when the target device manufacturer specifies conservative individual maxima for _all_ parameters that govern encoding level (e.g. frame size, frame rate, bitrate etc.), as well as for refs. a. Still, adhering to such conservative maxima might lead to sub-optimal encodes. E.g., when using less than the specified maximum frame size, a higher bitrate and/or refs value could have been used within the limits of the specified level. b. In all other cases - even when maximum level is specified - it is up to the user to determine whether desired frame size, frame rate, bitrate etc. will result in a conformant output level, at least. (When not using ffmpeg and '-level x.y' the user will also need to determine the corresponding maximum refs value for the particular combination of parameters.) c. When the user is not concerned with conforming to a _specific_ level but just conformance to whatever output level the encoder decides on, in most cases 1. above will ensure that output is non-level-conformant. This is especially true when using preset veryslow, which always sets refs to 16. This can be overcome by specifying either '-level x.y' or refs directly, but in either case the user needs to know the resulting output level in advance. h264level and ffx264 address a, b and c. > How can I reproduce this? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From gasto5 at hotmail.com Sat Nov 1 18:00:13 2014 From: gasto5 at hotmail.com (carlos gabriel hasbun comandari) Date: Sat, 1 Nov 2014 11:00:13 -0600 Subject: [FFmpeg-user] Tutorial on how to remaster a track with Audacity and FFmpeg. Message-ID: Hello folks, I recently created a screencast tutorial regarding the remastering of a neglected audio track from a multimedia file with Audacity and remuxing and reencoding the audio export back to a multimedia file(video and audio) with FFmpeg. I hope you enjoy it. https://www.youtube.com/watch?v=AR6N105XElw&index=1&list=UUrZxxvZT08MJ63n5MhgnUWQ From tomashnyk at gmail.com Sun Nov 2 20:55:28 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Sun, 02 Nov 2014 20:55:28 +0100 Subject: [FFmpeg-user] Naming an audio track Message-ID: Hello, When a file has more audio tracks, it is handy to name them and at least VLC can show their names - or titles, I am not sure about the terminology - in menu under Audio>Audio Track. How do I set the names using ffmpeg? I know how to set the language using -metadata:s:a:0 language=eng but this is something else. What is the right syntax? Regards, Tomas From dopelabs at dubstep.fm Sun Nov 2 21:06:46 2014 From: dopelabs at dubstep.fm (DopeLabs) Date: Sun, 2 Nov 2014 12:06:46 -0800 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: Message-ID: streams are identified using numbers, starting from 0 https://ffmpeg.org/ffmpeg-all.html#Stream-specifiers-1 > On Nov 2, 2014, at 11:55 AM, Tom?? Hnyk wrote: > > Hello, > When a file has more audio tracks, it is handy to name them and at least VLC can show their names - or titles, I am not sure about the terminology - in menu under Audio>Audio Track. How do I set the names using ffmpeg? I know how to set the language using -metadata:s:a:0 language=eng but this is something else. What is the right syntax? > Regards, > Tomas > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From tomashnyk at gmail.com Sun Nov 2 22:10:06 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Sun, 02 Nov 2014 22:10:06 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: Message-ID: On Sun, 02 Nov 2014 21:06:46 +0100, DopeLabs wrote: > streams are identified using numbers, starting from 0 > > https://ffmpeg.org/ffmpeg-all.html#Stream-specifiers-1 Yes, I know. I could use metadata:s:a:1 language=eng to set the language of the second audios stream, but how would I set its name? Tomas From james.darnley at gmail.com Sun Nov 2 22:15:29 2014 From: james.darnley at gmail.com (James Darnley) Date: Sun, 02 Nov 2014 22:15:29 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: Message-ID: <54569EF1.6060108@gmail.com> On 2014-11-02 22:10, Tom?? Hnyk wrote: > > On Sun, 02 Nov 2014 21:06:46 +0100, DopeLabs wrote: > >> streams are identified using numbers, starting from 0 >> >> https://ffmpeg.org/ffmpeg-all.html#Stream-specifiers-1 > > Yes, I know. I could use metadata:s:a:1 language=eng to set the > language of the second audios stream, but how would I set its name? > Tomas Set the right metadata. "name"? "title"? "title" probably. Look at one that already does the Right Thing (TM) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From tomashnyk at gmail.com Sun Nov 2 22:20:58 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Sun, 02 Nov 2014 22:20:58 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <54569EF1.6060108@gmail.com> References: <54569EF1.6060108@gmail.com> Message-ID: On Sun, 02 Nov 2014 22:15:29 +0100, James Darnley wrote: > On 2014-11-02 22:10, Tom?? Hnyk wrote: >> >> On Sun, 02 Nov 2014 21:06:46 +0100, DopeLabs >> wrote: >> >>> streams are identified using numbers, starting from 0 >>> >>> https://ffmpeg.org/ffmpeg-all.html#Stream-specifiers-1 >> >> Yes, I know. I could use metadata:s:a:1 language=eng to set the >> language of the second audios stream, but how would I set its name? >> Tomas > > Set the right metadata. "name"? "title"? "title" probably. Look at > one that already does the Right Thing (TM) > > I tried both name and title and it does not show in VLC (it shows them in the source file). This does not list per stream metadata: http://wiki.multimedia.cx/index.php?title=FFmpeg_Metadata Tomas From barsnick at gmx.net Sun Nov 2 22:28:22 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Sun, 2 Nov 2014 22:28:22 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: <54569EF1.6060108@gmail.com> Message-ID: <20141102212822.GA8968@sunshine.barsnick.net> On Sun, Nov 02, 2014 at 22:20:58 +0100, Tom?? Hnyk wrote: > I tried both name and title and it does not show in VLC (it shows them in > the source file). Do you happen to have a (short) sample which shows your desired behavior in VLC? Thanks, Moritz From tomashnyk at gmail.com Sun Nov 2 22:39:07 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Sun, 02 Nov 2014 22:39:07 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <20141102212822.GA8968@sunshine.barsnick.net> References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> Message-ID: On Sun, 02 Nov 2014 22:28:22 +0100, Moritz Barsnick wrote: > On Sun, Nov 02, 2014 at 22:20:58 +0100, Tom?? Hnyk wrote: >> I tried both name and title and it does not show in VLC (it shows them >> in >> the source file). > > Do you happen to have a (short) sample which shows your desired > behavior in VLC? > > Thanks, > Moritz Unfortunately, no, the file in question has 7,2 GB. I attach a screenshot of what I mean. Tomas -------------- next part -------------- A non-text attachment was scrubbed... Name: Screenshot from 2014-11-02 22:38:09.png Type: image/png Size: 65445 bytes Desc: not available URL: From james.darnley at gmail.com Sun Nov 2 22:50:22 2014 From: james.darnley at gmail.com (James Darnley) Date: Sun, 02 Nov 2014 22:50:22 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> Message-ID: <5456A71E.2080800@gmail.com> On 2014-11-02 22:39, Tom?? Hnyk wrote: > Unfortunately, no, the file in question has 7,2 GB. I attach a > screenshot of what I mean. > Tomas Can you post what gets printed when you run: ffmpeg -i BIGFILE -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From tomashnyk at gmail.com Sun Nov 2 23:24:58 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Sun, 02 Nov 2014 23:24:58 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <5456A71E.2080800@gmail.com> References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> Message-ID: On Sun, 02 Nov 2014 22:50:22 +0100, James Darnley wrote: > On 2014-11-02 22:39, Tom?? Hnyk wrote: >> Unfortunately, no, the file in question has 7,2 GB. I attach a >> screenshot of what I mean. >> Tomas > > Can you post what gets printed when you run: > ffmpeg -i BIGFILE > > Sure, here it goes: ffmpeg -i BIGFILE fmpeg version 2.3.git-1ace957 Copyright (c) 2000-2014 the FFmpeg developers built on Jul 26 2014 20:25:06 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-x11grab --enable-libpulse --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr libavutil 52. 92.101 / 52. 92.101 libavcodec 55. 69.100 / 55. 69.100 libavformat 55. 49.100 / 55. 49.100 libavdevice 55. 13.102 / 55. 13.102 libavfilter 4. 11.102 / 4. 11.102 libavresample 1. 3. 0 / 1. 3. 0 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 19.100 / 0. 19.100 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'BIGFILE': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2014-10-04 00:16:47 encoder : HandBrake 0.9.9 2013051800 Duration: 00:51:22.63, start: 0.000000, bitrate: 19802 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 18780 kb/s, 24 fps, 24 tbr, 90k tbn, 180k tbc (default) Metadata: creation_time : 2014-10-04 00:16:47 encoder : JVT/AVC Coding Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1(side), fltp, 640 kb/s (default) Metadata: creation_time : 2014-10-04 00:16:47 Stream #0:2(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 197 kb/s Metadata: creation_time : 2014-10-12 23:08:29 Tomas From barsnick at gmx.net Mon Nov 3 00:28:41 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Mon, 3 Nov 2014 00:28:41 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> Message-ID: <20141102232841.GB8968@sunshine.barsnick.net> On Sun, Nov 02, 2014 at 23:24:58 +0100, Tom?? Hnyk wrote: > encoder : HandBrake 0.9.9 2013051800 Okay, now we know how to produce a file in which VLC recognizes the names of audio streams: $ HandBrakeCLI [...] --aname "Name of audio track 1,Name of audio track 2" Only HandBrake and VLC seem to agree on this info, neither ffprobe (with any -show_* option) or mediainfo show this track name (in my experiments with an MP4 container). Too tired to try to disassemble the MP4, and my AtomicParsley is segfaulting. Your turn. :) Good night, Moritz From tomashnyk at gmail.com Mon Nov 3 02:08:31 2014 From: tomashnyk at gmail.com (=?utf-8?B?VG9tw6HFoSBIbnlr?=) Date: Mon, 03 Nov 2014 02:08:31 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <20141102232841.GB8968@sunshine.barsnick.net> References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> <20141102232841.GB8968@sunshine.barsnick.net> Message-ID: On Mon, 03 Nov 2014 00:28:41 +0100, Moritz Barsnick wrote: > On Sun, Nov 02, 2014 at 23:24:58 +0100, Tom?? Hnyk wrote: >> encoder : HandBrake 0.9.9 2013051800 > > Okay, now we know how to produce a file in which VLC recognizes the > names of audio streams: > $ HandBrakeCLI [...] --aname "Name of audio track 1,Name of audio track > 2" > > Only HandBrake and VLC seem to agree on this info, neither ffprobe > (with any -show_* option) or mediainfo show this track name (in my > experiments with an MP4 container). > > Too tired to try to disassemble the MP4, and my AtomicParsley is > segfaulting. Your turn. :) > > Good night, > Moritz Hm, if it is something proprietary to VLC and Handbrake, I can probably let it be... Good night, Tomas From jonathan.viney at gmail.com Mon Nov 3 02:18:50 2014 From: jonathan.viney at gmail.com (Jonathan Viney) Date: Mon, 3 Nov 2014 14:18:50 +1300 Subject: [FFmpeg-user] Problems saving rtsp stream directly as mpegts Message-ID: Hi, I am try to capture an rtsp stream with h264 from an IP camera directly into an mpegts file. ffmpeg -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -f mpegts out.ts This seems to work fine, but the output file is not recognised by ffmpeg/ffprobe. ffmpeg -i out.ts [h264 @ 0x7fa96501d200] decode_slice_header error [h264 @ 0x7fa96501d200] no frame! [h264 @ 0x7fa96501d200] non-existing PPS 0 referenced [mpegts @ 0x7fa964802a00] decoding for stream 0 failed [mpegts @ 0x7fa964802a00] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options out.ts: Operation not permitted Any idea why this is? If I do it in two steps, first saving the stream as a .mp4, it converts to mpegts fine with: ffmpeg -i out.mp4 -c copy -bsf h264_mp4toannexb out.ts Is it possible to do this in one step without the intermediary file? Thanks, -Jonathan. From cehoyos at ag.or.at Mon Nov 3 02:45:59 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 3 Nov 2014 01:45:59 +0000 (UTC) Subject: [FFmpeg-user] Problems saving rtsp stream directly as mpegts References: Message-ID: Jonathan Viney gmail.com> writes: > ffmpeg -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -f mpegts out.ts > > This seems to work fine, but the output file is > not recognised by ffmpeg/ffprobe. Complete, uncut console output missing. (I would have expected above command to return an error message and I would like to know why there is no such message.) > If I do it in two steps, first saving the stream > as a .mp4, it converts to mpegts fine with: > > ffmpeg -i out.mp4 -c copy -bsf h264_mp4toannexb out.ts > > Is it possible to do this in one step without the > intermediary file? What happens if you add the bitstreamfilter to the first command? Carl Eugen From cehoyos at ag.or.at Mon Nov 3 02:47:23 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 3 Nov 2014 01:47:23 +0000 (UTC) Subject: [FFmpeg-user] Naming an audio track References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> <20141102232841.GB8968@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > Okay, now we know how to produce a file in which > VLC recognizes the names of audio streams: > $ HandBrakeCLI [...] --aname "Name of audio track 1,Name of audio track 2" If somebody could now either upload a small sample or provide an actual command line, we could try to implement the missing feature... Carl Eugen From jonathan.viney at gmail.com Mon Nov 3 03:38:12 2014 From: jonathan.viney at gmail.com (Jonathan Viney) Date: Mon, 3 Nov 2014 15:38:12 +1300 Subject: [FFmpeg-user] Problems saving rtsp stream directly as mpegts In-Reply-To: References: Message-ID: On Mon, Nov 3, 2014 at 2:45 PM, Carl Eugen Hoyos wrote: > Jonathan Viney gmail.com> writes: > > > ffmpeg -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -f mpegts > out.ts > > > > This seems to work fine, but the output file is > > not recognised by ffmpeg/ffprobe. > > Complete, uncut console output missing. > (I would have expected above command to return an > error message and I would like to know why there > is no such message.) > Here are the command outputs. ffmpeg -rtsp_transport tcp -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -y -t 5 out.ts ffmpeg version N-67343-gd457478 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx --prefix=/tmp/ffmpeg-test libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.101 / 56. 10.101 libavformat 56. 12.100 / 56. 12.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.101 / 5. 2.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, rtsp, from 'rtsp://10.9.9.3:554/axis-media/media.amp': Metadata: title : Media Presentation Duration: N/A, start: 0.040011, bitrate: N/A Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 180k tbc Output #0, mpegts, to 'out.ts': Metadata: title : Media Presentation encoder : Lavf56.12.100 Stream #0:0: Video: h264, yuvj420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 90k tbn, 90k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [mpegts @ 0x7f9f22826a00] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file. frame= 127 fps= 32 q=-1.0 Lsize= 5774kB time=00:00:04.99 bitrate=9460.9kbits/s video:5341kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.106906% ffprobe out.ts ffprobe out.ts ffprobe version N-67343-gd457478 Copyright (c) 2007-2014 the FFmpeg developers built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx --prefix=/tmp/ffmpeg-test libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.101 / 56. 10.101 libavformat 56. 12.100 / 56. 12.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.101 / 5. 2.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [h264 @ 0x7fe783808000] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x7fe783808000] decode_slice_header error [h264 @ 0x7fe783808000] no frame! [h264 @ 0x7fe783808000] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x7fe783808000] decode_slice_header error [h264 @ 0x7fe783808000] no frame! [h264 @ 0x7fe783808000] non-existing PPS 0 referenced (duplicate messages removed) [mpegts @ 0x7fe78300da00] decoding for stream 0 failed [mpegts @ 0x7fe78300da00] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B), none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options out.ts: Operation not permitted > > > If I do it in two steps, first saving the stream > > as a .mp4, it converts to mpegts fine with: > > > > ffmpeg -i out.mp4 -c copy -bsf h264_mp4toannexb out.ts > > > > Is it possible to do this in one step without the > > intermediary file? > > What happens if you add the bitstreamfilter to the > first command? > ffmpeg -rtsp_transport tcp -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -y -bsf h264_mp4toannexb -t 5 out.ts ffmpeg version N-67343-gd457478 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx --prefix=/tmp/ffmpeg-test libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.101 / 56. 10.101 libavformat 56. 12.100 / 56. 12.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.101 / 5. 2.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, rtsp, from 'rtsp://10.9.9.3:554/axis-media/media.amp': Metadata: title : Media Presentation Duration: N/A, start: 0.040000, bitrate: N/A Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 180k tbc Output #0, mpegts, to 'out.ts': Metadata: title : Media Presentation encoder : Lavf56.12.100 Stream #0:0: Video: h264, yuvj420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 90k tbn, 90k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, corrupted stream or invalid MP4/AVCC bitstream Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, corrupted stream or invalid MP4/AVCC bitstream Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [mpegts @ 0x7fa97a809600] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file. [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, corrupted stream or invalid MP4/AVCC bitstream Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument (duplicate messages removed) frame= 127 fps= 31 q=-1.0 Lsize= 5280kB time=00:00:04.99 bitrate=8651.6kbits/s video:4883kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.132199% The resulting output file gives the same errors as above when passed to ffprobe. Thanks for the help. Regards, -Jonathan. Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From francois.visagie at gmail.com Mon Nov 3 08:32:49 2014 From: francois.visagie at gmail.com (Francois Visagie) Date: Mon, 3 Nov 2014 09:32:49 +0200 Subject: [FFmpeg-user] Getting H264 elementary stream from mp4 In-Reply-To: <545221B6.7020706@einfochips.com> References: <545221B6.7020706@einfochips.com> Message-ID: I experience the same accelerated playback, both with elementary streams demuxed by ffmpeg as well as x264 itself. Something else I noticed in the output file in both cases is that its frame rate gets set to variable, although that patently wasn't the case for input files I tested with, and there were no command line options influencing frame rate. In the case of x264, the --fps option seems to solve the problem, with set to that of the input file. I haven't tested with ffmpeg. On 30 October 2014 13:32, Shashank Pathak wrote: > Hi All, > > We are having one mp4 file with h264 codec. > we want to feed this file to a video decoder but the decoder requires > elementary stream. > So we tried to extract H264 elementary string from mp4 with below ffmpeg > commnd, > > ffmpeg -i input.mp4 -vcodec copy -vbsf h264_mp4toannexb output.h264 > > Now with this command we are getting h264 file but this file is not having > I-frame in it. > We viewed the file in Hex editor and found that it contains, > 00 00 00 01 67 --> SPS frame > 00 00 00 01 68 --> PPS frame > But we couldn't find, > 00 00 00 01 65 --> I-frmae. > > VLC is not able to play the generated h264 file. > ffplay plays the file but very fast (higher fps). > > Can anyone suggest what we are doing wrong? > or Any other method to get H264 file? > > -- > Best Regards, > Shashank Pathak > Senior Engineer | PES > > [cid:part1.05050907.05040709 at einfochips.com] > Product Engineering Services > Software | Embedded | Semiconductor > > Frost & Sullivan Company of the Year 2013-14 > ************************************************************************************************************************************************************* > eInfochips Business Disclaimer: This e-mail message and all attachments > transmitted with it are intended solely for the use of the addressee and may > contain legally privileged and confidential information. If the reader of > this message is not the intended recipient, or an employee or agent > responsible for delivering this message to the intended recipient, you are > hereby notified that any dissemination, distribution, copying, or other use > of this message or its attachments is strictly prohibited. If you have > received this message in error, please notify the sender immediately by > replying to this message and please delete it from your computer. Any views > expressed in this message are those of the individual sender unless > otherwise stated. Company has taken enough precautions to prevent the spread > of viruses. However the co > mpany accepts no liability for any damage caused by any virus transmitted by > this email. > ************************************************************************************************************************************************************* > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From francois.visagie at gmail.com Mon Nov 3 08:33:54 2014 From: francois.visagie at gmail.com (Francois Visagie) Date: Mon, 3 Nov 2014 09:33:54 +0200 Subject: [FFmpeg-user] Getting H264 elementary stream from mp4 In-Reply-To: References: <545221B6.7020706@einfochips.com> Message-ID: On 3 November 2014 09:32, Francois Visagie wrote: << snip >> Apologies for top-posting. From barsnick at gmx.net Mon Nov 3 10:21:39 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Mon, 3 Nov 2014 10:21:39 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> <20141102232841.GB8968@sunshine.barsnick.net> Message-ID: <20141103092139.GB23735@sunshine.barsnick.net> On Mon, Nov 03, 2014 at 01:47:23 +0000, Carl Eugen Hoyos wrote: > > $ HandBrakeCLI [...] --aname "Name of audio track 1,Name of audio track 2" > If somebody could now either upload a small sample or > provide an actual command line, we could try to > implement the missing feature... Sorry, I was quite tired and didn't know what I was thinking. I'm attaching an artificially created file, as it's smaller than most log files. ;-) Created as such: $ ffmpeg -y -f lavfi -i testsrc -f lavfi -i sine=f=880 -f lavfi -i sine=f=1000 -map 0 -map 1 -map 2 -t 0.5 in.mkv $ HandBrakeCLI -i in.mkv -o out.mp4 --audio 1,2 --aname "880 Hz sine wave,1 kHz sine wave" Sorry, my HandBrake is really old and has problems with the video, but the point is moot: VLC shows the two audio tracks of out.mp4 exactly as I tagged them with the HandBrake command line. BTW, using ffmpeg, I notice that when I use '-metadata:s:a:0 name="880 sine"', the file creation process shows me: Stream #0:1: Audio: aac (libfdk_aac) ([64][0][0][0] / 0x0040), 44100 Hz, mono, s16, 96 kb/s (default) Metadata: name : 880 sine encoder : Lavc56.10.100 libfdk_aac yet ffprobe fails to display the "name" metadata in the resulting file. This field - if it is a metadata field - does not seem to be supported by the MP4 muxer, and again ffmpeg is "lying" ;-) about its metadata creation - similar to the thread I started recently. Moritz P.S.: I can actually see the "tags" in a hexdump of the HandBrake created file: 0000-21E0: A6 00 00 00 1C 73 74 73 - 63 00 00 00 00 00 00 00 .....stsc....... 0000-21F0: 01 00 00 00 01 00 00 00 - 17 00 00 00 01 00 00 00 ................ 0000-2200: 14 73 74 63 6F 00 00 00 - 00 00 00 00 01 00 00 05 .stco........... 0000-2210: 56 00 00 00 20 75 64 74 - 61 00 00 00 18 6E 61 6D V... udta....nam 0000-2220: 65 38 38 30 20 48 7A 20 - 73 69 6E 65 20 77 61 76 e880 Hz sine wav 0000-2230: 65 00 00 02 2F 74 72 61 - 6B 00 00 00 5C 74 6B 68 e.../trak...\tkh 0000-2240: 64 00 00 00 02 D0 7C F6 - 53 D0 7C F6 53 00 00 00 d.....|.S.|.S... 0000-2250: 03 00 00 00 00 00 00 5C - 00 00 00 00 00 00 00 00 .......\........ I do not know, though, whether "name" is a header such as "stco" or "udta". -------------- next part -------------- A non-text attachment was scrubbed... Name: out.mp4 Type: application/octet-stream Size: 9421 bytes Desc: not available URL: From pkoshevoy at gmail.com Mon Nov 3 12:00:16 2014 From: pkoshevoy at gmail.com (Pavel Koshevoy) Date: Mon, 03 Nov 2014 04:00:16 -0700 Subject: [FFmpeg-user] gas-preprocessor.pl unknown arch: 'ppc' Message-ID: <54576040.1020201@gmail.com> Hi, I am trying to (re)build recent ffmpeg master (63e62cfbe23de8b362d94f783668620a2cd2b571) on OSX 10.5 ppc. I have installed latest (2014-08-12) gas-preprocessor.pl from http://git.libav.org/?p=gas-preprocessor.git When configuring the build it still complains and asks to install/update gas-preprocessor. I am not sure whether this is a gas-preprocessor problem, or an ffmpeg build configuration problem. This used to work. I don't know when it broke because configuring without gas-preprocessor is non-fatal, so it may have been broken for months and I haven't noticed. One thing I've noticed is that -arch ppc doesn't work, but -arch powerpc does: $ gas-preprocessor.pl -arch ppc -as-type apple-gas -- gcc -v unknown arch: 'ppc' $ gas-preprocessor.pl -arch powerpc -as-type apple-gas -- gcc -v Using built-in specs. Target: powerpc-apple-darwin9 Configured with: /var/tmp/gcc_42/gcc_42-5577~1/src/configure --disable-checking --prefix=/usr --mandir=/usr/share/man --enable-languages=c,objc,c++,obj-c++ --program-transform-name=/^[cg][^.-]*$/s/$/-4.2/ --with-slibdir=/usr/lib --build=i686-apple-darwin9 --with-gxx-include-dir=/usr/include/c++/4.0.0 --program-prefix= --host=powerpc-apple-darwin9 --target=powerpc-apple-darwin9 Thread model: posix gcc version 4.2.1 (Apple Inc. build 5577) Here is an excerpt from config.log: # /nfs/scratch/Developer/ffmpeg-git-src/configure --prefix=/Developer/ppc --disable-debug --disable-shared --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-l ibtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-pthreads --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-am rwb --enable-postproc --enable-libx264 --enable-libxvid --enable-libass --enable-gnutls --enable-runtime-cpudetect --extra-cflags=-I/opt/local/include --extra-ldflags='-headerpad_max_install_names -L/opt/loca l/lib' ... gas-preprocessor.pl -arch ppc -as-type apple-gas -- gcc -v unknown arch: 'ppc' check_gas using 'gcc' as AS check_as BEGIN /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S 1 .macro m n, y:vararg=0 2 \n: .int \y 3 .endm 4 m x END /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -I/opt/local/include -force_cpusubtype_ALL -mdynamic-no-pic -c -o /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.tRd1aGKd.o /var/fold ers/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S:4:Junk character 92 (\). /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S:4:Rest of line ignored. 1st junk character valued 110 (n). check_gas using 'gcc' as AS check_as BEGIN /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S 1 .macro m n, y:vararg=0 2 \n: .int \y 3 .endm 4 m x END /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -I/opt/local/include -force_cpusubtype_ALL -mdynamic-no-pic -c -o /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.tRd1aGKd.o /var/fold ers/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S:4:Junk character 92 (\). /var/folders/zc/zcLuFx8PGxWB0xQUmne1nU+++yU/-Tmp-//ffconf.dZICFXiM.S:4:Rest of line ignored. 1st junk character valued 110 (n). WARNING: GNU assembler not found, install/update gas-preprocessor From barsnick at gmx.net Mon Nov 3 12:12:21 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Mon, 3 Nov 2014 12:12:21 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <20141103092139.GB23735@sunshine.barsnick.net> References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> <20141102232841.GB8968@sunshine.barsnick.net> <20141103092139.GB23735@sunshine.barsnick.net> Message-ID: <20141103111221.GD23735@sunshine.barsnick.net> [Replying to self] On Mon, Nov 03, 2014 at 10:21:39 +0100, Moritz Barsnick wrote: > I do not know, though, whether "name" is a header such as "stco" or > "udta". So apparently we're looking as MPEG4/MOV containers here. That's what Tom?? was using as well, and where I managed to reproduce. Using AtomicParsley, I managed to "see" that the tag in use is an atom trak.udta.name: Atom trak @ 8193 of size: 560, ends @ 8753 Atom tkhd @ 8201 of size: 92, ends @ 8293 Atom mdia @ 8293 of size: 428, ends @ 8721 [...] Atom udta @ 8721 of size: 32, ends @ 8753 Atom name @ 8729 of size: 24, ends @ 8753 The atom is described here: https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-BBCCFFGD I have found a mention that HandBrake and subler use this atom: https://github.com/pjan/osx-dotfiles/blob/55d452fae4037a019c85a6095a5b3a0bd11e3d15/.dotfiles/.encode-handheld.pl#L2907 I couldn't figure how to add arbitrary atoms with AtomicParsley, only how to delete them. I do see ffmpeg has a function named mov_write_track_udta_tag(), called from mov_write_trak_tag(), called from mov_write_moov_tag(), but I don't know if it gets used and whether it does the "right thing" from VLC's point of view, and how to get it to write the "right thing". Getting closer, Moritz P.S.: Interesting, because it follows a question I never dared to ask: Where does my PVR/STB get the names of its audio tracks from? (Albeit MPEG-TS.) From barsnick at gmx.net Mon Nov 3 15:04:42 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Mon, 3 Nov 2014 15:04:42 +0100 Subject: [FFmpeg-user] Naming an audio track In-Reply-To: <20141103111221.GD23735@sunshine.barsnick.net> References: <54569EF1.6060108@gmail.com> <20141102212822.GA8968@sunshine.barsnick.net> <5456A71E.2080800@gmail.com> <20141102232841.GB8968@sunshine.barsnick.net> <20141103092139.GB23735@sunshine.barsnick.net> <20141103111221.GD23735@sunshine.barsnick.net> Message-ID: <20141103140442.GF23735@sunshine.barsnick.net> On Mon, Nov 03, 2014 at 12:12:21 +0100, Moritz Barsnick wrote: > I do see ffmpeg has a function named mov_write_track_udta_tag(), > called from mov_write_trak_tag(), called from mov_write_moov_tag(), > but I don't know if it gets used and whether it does the "right > thing" from VLC's point of view, and how to get it to write the > "right thing". I looked at ffmpeg's source, and figured out that it uses the above functions to write the trak.udta.name atom for its metadata "title": $ ffmpeg -y -f lavfi -i testsrc -f lavfi -i sine=f=880 -f lavfi -i sine=f=1000 -map 0 -map 1 -map 2 -t 0.5 -metadata:s:a:0 title="880 sine" -metadata:s:a:1 title="1000 sine" out_ffmpeg.mp4 AtomicParsley now sees reports the tag, just like it did for HandBrake's file: $ AtomicParsley out_ffmpeg.mp4 -t 1 [...] User data; level: track=2; atom "name" : 880 sine User data; level: track=3; atom "name" : 1000 sine So ffmpeg is basically doing the right thing with "title". Unfortunately, VLC seems picky about this, and still doesn't display it. This may or may not have to do with the differing major_brand, compatible_brands, or the structure of the MOOV atoms. *shrug* At this point, we'd probably need to look at VLC's source code to figure this out. :-P There is some udta parsing in VLC's modules/demux/mp4/mp4.c, in function MP4_TrackCreate(). It doesn't look picky, but what do I know. Moritz From pkoshevoy at gmail.com Mon Nov 3 17:34:19 2014 From: pkoshevoy at gmail.com (Pavel Koshevoy) Date: Mon, 03 Nov 2014 09:34:19 -0700 Subject: [FFmpeg-user] gas-preprocessor.pl unknown arch: 'ppc' In-Reply-To: <54576040.1020201@gmail.com> References: <54576040.1020201@gmail.com> Message-ID: <5457AE8B.4070807@gmail.com> On 2014/11/03 4:00, Pavel Koshevoy wrote: > Hi, > > I am trying to (re)build recent ffmpeg master > (63e62cfbe23de8b362d94f783668620a2cd2b571) on OSX 10.5 ppc. I have > installed latest (2014-08-12) gas-preprocessor.pl from > http://git.libav.org/?p=gas-preprocessor.git > > When configuring the build it still complains and asks to > install/update gas-preprocessor. I am not sure whether this is a > gas-preprocessor problem, or an ffmpeg build configuration problem. > This used to work. I don't know when it broke because configuring > without gas-preprocessor is non-fatal, so it may have been broken for > months and I haven't noticed. > > One thing I've noticed is that -arch ppc doesn't work, but -arch > powerpc does: > > $ gas-preprocessor.pl -arch ppc -as-type apple-gas -- gcc -v > unknown arch: 'ppc' > This happens on line 66 of gas-preprocessor.pl -- it checks the $comments dictionary for 'ppc', doesn't find it and gives up. I've worked around the problem locally by adding 'ppc' => '#' to %comments on line 17. Pavel From aram at ConnectTo.com Tue Nov 4 01:34:58 2014 From: aram at ConnectTo.com (Aram Ter-Martirosyan) Date: Mon, 3 Nov 2014 16:34:58 -0800 Subject: [FFmpeg-user] Looking for consultant Message-ID: <17a401cff7c7$2aba03a0$802e0ae0$@ConnectTo.com> Looking for consultant We are new at ffmpeg and looking for consultant to help us with some advance configuration and understanding and fixing errors. We would prefer someone in US, since we are located in US. Thanks, Aram Ter-Martirosyan ConnectTo Communications http://www.ConnectTo.com 555 Riverdale, Suite A Glendale, CA 91204 aram at ConnectTo.com tel 818.546.4601 fax 818.546.4617 From bluesky at caramail.com Tue Nov 4 10:09:51 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 10:09:51 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? Message-ID: I have some .ts (transport stream) files that contain video and 5.1 audio but XBMC will only play them in stereo. It appears that the only type of file that XBMC will play in full 5.1 is an ac3 file, so my thought was to try converting the files using ffmpeg. My original approach was to use: ffmpeg -i "originalfile.ts" -c:v copy -c:a ac3 "new.ts" This did create an ac3 file but with only two channels. After reading a bit more, I tried adding the -ac 6 switch, so it became: ffmpeg -i "originalfile.ts" -c:v copy -c:a ac3 -ac 6 "new.ts" This created a file that XBMC thinks is a Dolby 5.1 file (according to the on-screen display) but in fact the only audio present is on the stereo channels. The source is definitely in MPEG-TS and it contains 5.1 information, although I'm confused as to how. One of the files is actually a speaker test video and if played in a different program (such as MythTV) it plays the "Left", "Right" and "Center" channels from those speakers, but the "Left Surround" and "Right Surround" audio comes from both the front and rear speakers of the specified channel, with a bit more emphasis on the rear. If played in XBMC it plays from the front speakers only. Yet if I attempt to display the properties in a program like iMediaHUD it only shows two stereo tracks that look like this: Audio #1 Count : 222 Count of stream of this kind : 2 Kind of stream : Audio Stream identifier : 1 StreamOrder : 0-1 ID : 64 (0x40) Menu ID : 137 (0x89) Format : MPEG Audio Commercial name : MPEG Audio Format version : Version 1 Format profile : Layer 2 Internet media type : audio/mpeg Codec ID : 3 Codec : MPEG-1 Audio layer 2 Duration : 00:01:06.624 Bit rate mode : Constant Bit rate : 256 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Samples count : 3197952 Frame count : 2776 Compression mode : Lossy Delay : 00:00:00.268 Delay, origin : Container Delay relative to video : -15 Video0 delay : -15 Stream size : 2.03 MiB (1%) Proportion of this stream : 0.00603 Language : en Audio #2 Count : 222 Count of stream of this kind : 2 Kind of stream : Audio Stream identifier : 2 StreamOrder : 0-2 ID : 65 (0x41) Menu ID : 137 (0x89) Format : MPEG Audio Commercial name : MPEG Audio Format version : Version 1 Format profile : Layer 2 Format settings : Intensity Stereo + MS Stereo Mode extension : Intensity Stereo + MS Stereo Internet media type : audio/mpeg Codec ID : 3 Codec : MPEG-1 Audio layer 2 Duration : 00:01:06.648 Bit rate mode : Constant Bit rate : 256 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Samples count : 3199104 Frame count : 2777 Compression mode : Lossy Delay : 00:00:00.255 Delay, origin : Container Delay relative to video : -28 Video0 delay : -28 Stream size : 2.03 MiB (1%) Proportion of this stream : 0.00603 Language : en I have no idea how they are encoding the 5.1 but I can assure you it's there, and all I want to do is use ffmpeg to convert it and similar files to ac3 so XBMC will play them in 5.1. Can anyone tell me what I am doing wrong? From cehoyos at ag.or.at Tue Nov 4 10:38:55 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 09:38:55 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: Message-ID: caramail.com> writes: > ffmpeg -i "originalfile.ts" -c:v copy -c:a ac3 "new.ts" Complete, uncut console output missing, I don't remember a question where mediainfo output was useful. How do you know that your input file contains surround information? Is it because you have a dolby certified receiver that allows you to play two-channel audio with surround effects? In this case, do not reencode audio: FFmpeg has a Dolby PLII encoder (resampler) but no decoder, just convince xbmc to not mess with audio and allow your receiver to do what is necessary. (Analog input may be required iirc.) Carl Eugen From bluesky at caramail.com Tue Nov 4 11:16:32 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 11:16:32 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , Message-ID: "Carl Eugen Hoyos" wrote: >> ffmpeg -i "originalfile.ts" -c:v copy -c:a ac3 "new.ts" > Complete, uncut console output missing, I don't remember > a question where mediainfo output was useful. Sorry, different forums want to see different types of information, and in some they give you a hard time if you post a message that is too long. I will post console output at the bottom of this message. > How do you know that your input file contains surround > information? Is it because you have a dolby certified > receiver that allows you to play two-channel audio with > surround effects? First off, please don't use too much technical jargon with me, I really don't understand it. My receiver has the Dolby digital logo (and DTS and THX logos) on it. But the reason I know the file contains surround information is very simple: One of the files is an audio test file where someone says "Left, Right, Center, (low hum for subwoofer), Left surround, Right surround" and if I play that file in MythTV it plays with all the sounds coming from the correct speakers. If I play the same file in XBMC only the front speakers are used. >In this case, do not reencode audio: > FFmpeg has a Dolby PLII encoder (resampler) but no > decoder, just convince xbmc to not mess with audio and > allow your receiver to do what is necessary. > (Analog input may be required iirc.) Any suggestions on how to actually do that? I have tried every setting that looks even remotely related. Since I am new to the list I should probably state that I am not a developer or even a very technically-inclined user. When I posted my message I was hoping maybe it was just some setting or switch I was not using in ffmpeg that I should be using, and that maybe some kind person could tell me what I am doing wrong. But if you bury me in technical jargon I'm not going to understand what you are telling me so PLEASE try to keep it simple. Here is what I think you mean by "console output". And before you try to tell me there is no 5.1 in this file, this is the channel test recording I mentioned above and I can assure you that it does play in 5.1, with the channel identifications coming from the correct speakers, if played using MythTV. So I have no idea why ffmpeg doesn't seem to notice the 5.1 any more than a mediainfo type program. This is a mystery to me! $ ffmpeg -i "5.1 Audio Test.ts" -c:v copy -c:a ac3 "new.ts" ffmpeg version 2.4.2- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2014 the FFmpeg developers built on Oct 9 2014 07:24:56 with gcc 4.8 (Debian 4.8.3-11) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable-libass --enable-gnutls --cc=gcc-4.8 libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [mpeg2video @ 0x415d560] Invalid frame dimensions 0x0. Last message repeated 8 times Input #0, mpegts, from '5.1 Audio Test.ts': Duration: 00:01:06.83, start: 0.254678, bitrate: 42317 kb/s Program 137 Stream #0:0[0x30]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 40033 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x40](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 244 kb/s Stream #0:2[0x41](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 244 kb/s Output #0, mpegts, to 'new.ts': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: mpeg2video ([2][0][0][0] / 0x0002), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 40033 kb/s, 29.97 fps, 90k tbn, 29.97 tbc Stream #0:1(eng): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s Metadata: encoder : Lavc56.1.100 ac3 Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (mp2 (native) -> ac3 (native)) Press [q] to stop, [?] for help [mpegts @ 0x41597c0] PES packet size mismatchme=00:00:35.38 bitrate=42685.8kbits/s [mp2 @ 0x415fc20] incomplete frame Error while decoding stream #0:1: Invalid data found when processing input frame= 1993 fps=0.0 q=-1.0 Lsize= 352244kB time=00:01:06.78 bitrate=43209.0kbits/s video:324924kB audio:1562kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 7.889561% I should note that ffmpeg always complains about invalid frames or data at the beginning and end of these recordings, I do not know why but suspect it's because the recordings start and end based on time of day, and simply copy the transport stream to the hard drive without making any attempt to start or stop the recording on a frame boundary. But, that is just a wild guess on my part. From cehoyos at ag.or.at Tue Nov 4 11:41:36 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 10:41:36 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , Message-ID: caramail.com> writes: > One of the files is an audio test file where someone > says "Left, Right, Center, (low hum for subwoofer), > Left surround, Right surround" and if I play that > file in MythTV it plays with all the sounds coming > from the correct speakers. Please provide your input sample. Carl Eugen From cehoyos at ag.or.at Tue Nov 4 13:56:30 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 12:56:30 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , Message-ID: caramail.com> writes: > If I play the same file in XBMC only the front > speakers are used. How is your computer connected to your receiver? I can choose here between analog, hdmi and TOSLINK (optical). If I use hdmi or the optical input, my receiver allows me to switch the "listen mode": As suspected, the audio uses Dolby PLII to encode surround sound via stereo channels. If I select "PLII" on my receiver, I hear the surround sound (with failures because the algorithm cannot be perfect). If I use the analog input, I have to force the receiver to "two-channel" input to make it detect PLII: If I set it to 6-cable input (which is what I normally use), I cannot choose PLII. As said, FFmpeg does not support PLII decoding (only encoding). Are you sure MythTV contains an internal implementation and not does not use your receiver's Dolby decoder? Just to make things less confusing: Above has nothing to do with "mpeg2": You can use any audio codec that supports two channels and use it to store PLII-encoded surround sound. There is also "MPEG-2 5.1" or "MPEG-2 Multichannel". This is very rare and FFmpeg cannot detect it (nor decode it). I don't think your file can use both Dolby PLII and MPEG-2 Multichannel at the same time. Wikipedia articles for Dolby PLII, toslink and MPEG-2 Multichannel exist. Carl Eugen From bluesky at caramail.com Tue Nov 4 17:56:08 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 17:56:08 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , Message-ID: "Carl Eugen Hoyos" wrote: >> If I play the same file in XBMC only the front >> speakers are used. > How is your computer connected to your receiver? > I can choose here between analog, hdmi and > TOSLINK (optical). If I use hdmi or the optical > input, my receiver allows me to switch the > "listen mode": As suspected, the audio uses > Dolby PLII to encode surround sound via stereo > channels. If I select "PLII" on my receiver, I > hear the surround sound (with failures because > the algorithm cannot be perfect). > If I use the analog input, I have to force the > receiver to "two-channel" input to make it > detect PLII: If I set it to 6-cable input > (which is what I normally use), I cannot choose > PLII. Well that is interesting. My computer is connected to my receiver via a S/PDIF (Toslink optical) cable but I know someone else who has encountered the same issue and he is using a HDMI connection, the difference being that I don't think he ever hears audio from his rear speakers when playing in XBMC (but he does in Myth, same as I do). > As said, FFmpeg does not support PLII decoding > (only encoding). Are you sure MythTV contains > an internal implementation and not does not use > your receiver's Dolby decoder? I know nothing about how MythTV handles audio so the short answer to that is "no". > Just to make things less confusing: Above has > nothing to do with "mpeg2": You can use any > audio codec that supports two channels and > use it to store PLII-encoded surround sound. That would explain everything, except why Myth can play this audio and XBMC can't (well actually it does, but it does a poor job of it - see below). > There is also "MPEG-2 5.1" or "MPEG-2 > Multichannel". This is very rare and FFmpeg > cannot detect it (nor decode it). I don't > think your file can use both Dolby PLII and > MPEG-2 Multichannel at the same time. Do you know of any utility that can tell me if any of these lesser-known methods are being used? > Wikipedia articles for Dolby PLII, toslink and > MPEG-2 Multichannel exist. Thank you. I'm aware of Toslink but will look up the other two. It turns out that the reason I thought XBMC was not playing anything from the rear speakers is because I did not have my ear right up next to them. When I play the test video in XBMC, what actually happens is this: Left, Right, Center, and the low frequency hum all play from the correct speakers and NOT from any other speakers. Left Surround plays at significantly reduced volume from BOTH the left and right rear speakers, at normal volume from the left front speaker, and at normal volume but with breakups from the right front speaker! Right Surround plays at significantly reduced volume from BOTH the right and left rear speakers, at normal volume from the right front speaker, and at normal volume but with breakups from the left front speaker! This happens whether I play the original .TS file or the file that I attempted to convert using ffmpeg - it sounds exactly the same; I just never heard the rear speaker audio because the front is so much louder. But as noted, if I play it in MythTV then all the audio comes from the correct spearlers - which is to say, Left, Right, Center, and the low frequency hum all play from the correct speakers and NOT from any other speakers. The left and right surrounds both play from the speakers on their respective sides, but louder from the rear speakers than the front, and not on the opposite side's speakers as they do in XBMC. >From what you are telling me, I'm guessing that ffmpeg cannot decode this signal and break it out to individual channels which can then be properly re-encoded as six channel ac3, but instead can only pass it through. Is there a reason ffmpeg cannot do this or is it simply that no one has figured out a way to do it (yet)? In any case I thank you for the explanation - you are the first person that hasn't in effect tried to tell me I'm crazy for hearing 5.1 audio in this thing when I play it in MythTV. I will read those Wikipedia pages you mentioned. From cehoyos at ag.or.at Tue Nov 4 19:17:23 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 18:17:23 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , Message-ID: caramail.com> writes: > But as noted, if I play it in MythTV then all the > audio comes from the correct spearlers You mean you don't hear the "S" in "Surround" coming (to some degree) from the front speaker while the rest of the word is clearly coming from the rear speaker? I may have a cheap receiver... Don't worry about MPEG-2 Multichannel, your sample is certainly Dolby PLII. If you are interested, ticket #1258 contains source code. Carl Eugen From cehoyos at ag.or.at Tue Nov 4 19:25:23 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 18:25:23 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , Message-ID: caramail.com> writes: > My computer is connected to my receiver via a > S/PDIF (Toslink optical) cable What does your receiver show when you play with MythTV, what does it show for XBMC? Maybe one is "Dolby Digital" (which is unrelated to PLII) and one doesn't show it / shows PLII? Do you have a switch to change between "Stereo" and "PLII": My receiver has such a switch and it helps if I reencode the PLII stream to Dolby Digital with FFmepg: $ ffmpeg -i input -ab 640k out.ac3 I can still here the surround via S/PDIF if I tell my receiver that this is PLII. Carl Eugen From bluesky at caramail.com Tue Nov 4 19:34:38 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 19:34:38 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , Message-ID: "Carl Eugen Hoyos" wrote: >> But as noted, if I play it in MythTV then all the >> audio comes from the correct spearlers > You mean you don't hear the "S" in "Surround" coming > (to some degree) from the front speaker while the > rest of the word is clearly coming from the rear > speaker? I may have a cheap receiver... In MythTV, the surround sound comes from both the front and rear speakers for the specified side, but the volume is higher in the rear channels (might just be the way I have my speakers balanced). Whereas in XBMC, a much lower volume is heard from the rear speakers and the channels aren't properly separated. > Don't worry about MPEG-2 Multichannel, your sample > is certainly Dolby PLII. Thanks. Is there a particular tool you used to determine that? Mediainfo doesn't seem to know anything about it. > If you are interested, ticket #1258 contains source > code. Unfortunately I would not know what to do with that, but thanks anyway. From cehoyos at ag.or.at Tue Nov 4 19:47:11 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 18:47:11 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , Message-ID: caramail.com> writes: > > >> But as noted, if I play it in MythTV then all the > >> audio comes from the correct spearlers > > > You mean you don't hear the "S" in "Surround" coming > > (to some degree) from the front speaker while the > > rest of the word is clearly coming from the rear > > speaker? I may have a cheap receiver... > > In MythTV, the surround sound comes from both the > front and rear speakers for the specified side, but > the volume is higher in the rear channels (might > just be the way I have my speakers balanced). Using MPlayer (and setting my receiver to PLII) I can clearly hear the sound as surround sound, but the "S" of "Surround" comes from the front instead of the rear speaker. This doesn't change if I reencode the audio and send it directly to the receiver. My assumption was that this simply shows the deficits of the PLII system. > Whereas in XBMC, a much lower volume is heard from > the rear speakers and the channels aren't properly > separated. Unfortunately, you can get neither support for XBMC nor MythTV here (the relevant question to them is probably: Are you reencoding my MPEG-2 two-channel audio to AC3 instead of sending it as PCM to give the receiver a better chance to know what to do?), I will hopefully open a ticket for PLII decoding, I have no idea how difficult / trivial this is given that we support encoding... > > Don't worry about MPEG-2 Multichannel, your sample > > is certainly Dolby PLII. > > Thanks. Is there a particular tool you used to > determine that? Iirc (and I think Wikipedia also explains this), there are a few very old DVDs that use MPEG-2 Multichannel, I severely doubt that it was formalized for DVB. You can check the bitrate to be sure: I would expect that 256k are not enough for Multichannel. > Mediainfo doesn't seem to know anything about it. And as explained, Mediainfo output is rarely relevant on this mailing list. Carl Eugen From bluesky at caramail.com Tue Nov 4 19:57:51 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 19:57:51 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , Message-ID: "Carl Eugen Hoyos" wrote: >> My computer is connected to my receiver via a >> S/PDIF (Toslink optical) cable > What does your receiver show when you play with > MythTV, what does it show for XBMC? > Maybe one is "Dolby Digital" (which is unrelated > to PLII) and one doesn't show it / shows PLII? I have never heard of "PLII" prior to this thread. MythTV tells me nothing about the audio, it simply reports it as 1080 and that's all, unless there is some hidden display option I don't know about. XBMC reports it as "1080" "MPEG-2" "2.0" "MP3" "1.78" > Do you have a switch to change between "Stereo" > and "PLII": No. > My receiver has such a switch and > it helps if I reencode the PLII stream to Dolby > Digital with FFmepg: > $ ffmpeg -i input -ab 640k out.ac3 At least on my system, XBMC won't even acknowledge the presence of a file with a .ac3 extension. If I change it to .mp4 then XBMC will play it but in exactly the same way it plays the .ts file - there is no discernable difference. > I can still here the surround via S/PDIF if I > tell my receiver that this is PLII. My receiver has options for THX, Dolby/DTS Surround, Stereo, and Direct. Stereo and Direct play with no audio at all from the back channels, while THX and Dolby both play the file the same way I have described previously. I usually leave it set on THX. From bluesky at caramail.com Tue Nov 4 20:12:56 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 20:12:56 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? Message-ID: "Carl Eugen Hoyos" wrote: > Unfortunately, you can get neither support for XBMC > nor MythTV here (the relevant question to them is > probably: Are you reencoding my MPEG-2 two-channel > audio to AC3 instead of sending it as PCM to give > the receiver a better chance to know what to do?), > I will hopefully open a ticket for PLII decoding, > I have no idea how difficult / trivial this is > given that we support encoding... Thank you. That is all I can hope for. Given my lack of technical understanding I suspect I have taken this as far as I can go for the moment. It would be really great if ffmpeg could detect these channels and reencode them to a more standard (if that is the way to put it) ac3 format. XBMC WILL play Dolby encoded files correctly if the channels are all present and not encoded in some weird format, in fact it is very good at that. Also, it actually indicates that such files are 5.1 rather than 2.0. The problem is that although I have been able to get ffmpeg to create a 5.1 channel ac3 .ts file (using the -ac 6 option, as mentioned in my original post), it does not seem to be decoding the channels from the original file correctly and therefore is not placing them into the output file correctly. So basically I would get like to get the channels extracted from this weird PLII format and into conventional ac3, if that makes any sense (I apologize if I am not stating this correctly but hopefully you understand what I mean). If there is any way you can accomplish that, that would be great! ? ? From cehoyos at ag.or.at Tue Nov 4 20:16:17 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 19:16:17 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , Message-ID: caramail.com> writes: > "Carl Eugen Hoyos" ag.or.at> wrote: > > >> My computer is connected to my receiver via a > >> S/PDIF (Toslink optical) cable > > > What does your receiver show ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ I know (expected) that MythTV and XBMC do not show PLII. My receiver has a display and when digital audio (as via S/PDIF) starts, it shows something on this display. While playing, I can switch the "Input mode" to change between "Stereo" and "PLII". > My receiver has options for THX, Dolby/DTS > Surround, Stereo, and Direct. PLII is (the successor of) Dolby Surround afaict. Since Dolby Surround and DTS Surround are two completely different things afaik, I am surprised that this is the same option... I would be surprised if the file is THX but I don't know much about it, my receiver does not support it (too cheap). (From a very quick look, THX is unrelated: "A common error is that THX is similar to Dolby Surround.") Carl Eugen From dave at dericed.com Tue Nov 4 20:30:01 2014 From: dave at dericed.com (Dave Rice) Date: Tue, 4 Nov 2014 14:30:01 -0500 Subject: [FFmpeg-user] using -f sdl Message-ID: <00D10030-94A6-4B3D-984A-F4BFF840E085@dericed.com> Hi all, I?m having trouble getting sdl output of ffmpeg to work reliably on my Mac. I want to do an encode and display the video simultaneously. I have been doing that by piping ffmpeg to ffplay but would like to drop the ffplay, since if I control-C the process then ffplay quits and the ffmpeg encode doesn?t end properly (no moov atom, etc). So I?m trying to use -f sdl but getting errors. ffmpeg -y -f lavfi -i smptebars -t 60 -c:v ffv1 test.mov -f sdl test_sdl ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 6 2014 23:13:22 with Apple LLVM version 6.0 (clang-600.0.51) (based on LLVM 3.5svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/2.4.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-ffplay --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.1_1/include/openjpeg-1.5 ' libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libavresample 2. 1. 0 / 2. 1. 0 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 Input #0, lavfi, from 'smptebars': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc [mov @ 0x7febb4018000] Using MS style video codec tag, the file may be unplayable! Nov 4 14:22:40 drice.local ffmpeg[98326] : CGSConnectionByID: 0 is not a valid connection ID. Nov 4 14:22:40 drice.local ffmpeg[98326] : Invalid Connection ID 0 2014-11-04 14:22:40.444 ffmpeg[98326:1827165] *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'Error (1000) creating CGSWindow on line 281' *** First throw call stack: ( 0 CoreFoundation 0x00007fff8e5e064c __exceptionPreprocess + 172 1 libobjc.A.dylib 0x00007fff8d2b96de objc_exception_throw + 43 2 CoreFoundation 0x00007fff8e5e04fd +[NSException raise:format:] + 205 3 AppKit 0x00007fff9662d9dd _NSCreateWindowWithOpaqueShape2 + 1417 4 AppKit 0x00007fff9662bd3c -[NSWindow _commonAwake] + 1808 5 AppKit 0x00007fff96539ee0 -[NSWindow _commonInitFrame:styleMask:backing:defer:] + 864 6 AppKit 0x00007fff9653954c -[NSWindow _initContent:styleMask:backing:defer:contentView:] + 1477 7 AppKit 0x00007fff96538f7a -[NSWindow initWithContentRect:styleMask:backing:defer:] + 45 8 libSDL-1.2.0.dylib 0x000000010428f21e -[SDL_QuartzWindow initWithContentRect:styleMask:backing:defer:] + 279 9 libSDL-1.2.0.dylib 0x000000010428cc63 QZ_SetVideoMode + 1417 10 libSDL-1.2.0.dylib 0x0000000104283f71 SDL_SetVideoMode + 906 11 libavdevice.56.dylib 0x0000000102f4ad71 avdevice_free_list_devices + 10775 ) libc++abi.dylib: terminating with uncaught exception of type NSException Abort trap: 6 Any advice? Dave Rice From Laine.Lee at utsa.edu Tue Nov 4 20:46:43 2014 From: Laine.Lee at utsa.edu (Laine Lee) Date: Tue, 4 Nov 2014 19:46:43 +0000 Subject: [FFmpeg-user] Any Mac OS X users interested in ffmpeg AppleScript droplets? - here's one for cat Message-ID: Concatenate similar video files using this AppleScript. How do I use it? Copy the lines containing +ACI-begin script+ACI- through +ACI-end script+ACI- to your Mac's clipboard. Paste the clipboard contents into a Script (or AppleScript, depending on your version of OS X) Edtor window, then save as an application. Name your video files successively, and drop them all onto the AppleScript application's icon. What does it do? Uses your compiled version of the ffmpeg executable to concatenate the files in ascending order, creating the resulting file in the same location as the files you dropped. The ffmpeg executable should be compiled using Homebrew+ADw-http://brew.sh/+AD4- and/or be accessible by your command line shell at /usr/local/bin/ffmpeg. You should be able to change that specification in the AppleScript if you keep ffmpeg somewhere else. I+IBk-ve tested with .m4v files containing 5.1 audio. Please help me test others. Thanks. -----begin script----- property temppath : +ACI-/private/tmp/+ACI- property startnum : 0 property tmpfile : +ACI-/tmp/execme.command+ACI- on open the+AF8-items my build+AF8-archive(the+AF8-items) end open on build+AF8-archive(the+AF8-items) set theshellscript to +ACIAIg- repeat with i from 1 to (count of the+AF8-items) set itemcount to (count of the+AF8-items) set the+AF8-item to item i of the+AF8-items as alias try tell application +ACI-Finder+ACI- set sost to (container of the+AF8-item) as string end tell set pos+AF8-filepath to POSIX path of sost end try set this+AF8-filepath to (the+AF8-item as string) if last character of this+AF8-filepath is +ACI-:+ACI- then tell me to set it+AF8-is+AF8-a+AF8-folder to true else set it+AF8-is+AF8-a+AF8-folder to false end if set thesourcename to (name of (info for the+AF8-item)) set namepart to (name extension of (info for the+AF8-item)) set the+AF8-source+AF8-file to POSIX path of this+AF8-filepath --set newname to replace+AF8-chars(thesourcename, namepart, +ACI-joined.mp4+ACI-) set finalname to replace+AF8-chars(thesourcename, +ACI-.+ACI- +ACY- namepart, +ACIAIg-) try if i +AD0- 1 then set the filelistbody to +ACIAIw- comment+ACI- +ACY- return +ACY- +ACI-file+ACI- +ACY- space +ACY- (quoted form of the+AF8-source+AF8-file) +ACY- return else set the filelistbody to filelistbody +ACY- +ACI-file+ACI- +ACY- space +ACY- (quoted form of the+AF8-source+AF8-file) +ACY- return end if on error onerr activate display dialog onerr end try end repeat set fileData to +ACI-echo+ACI- +ACY- space +ACY- +ACIAXAAiACI- +ACY- filelistbody +ACY- +ACIAXAAiACI- +ACY- space +ACY- +ACIAPg- ffmpegCatList.txt+ACI- set theshellscript to fileData +ACY- +ACIAOw-sleep 3+ADs-/usr/local/bin/ffmpeg -f concat -i ffmpegCatList.txt -c copy+ACI- +ACY- space set shellExec to space +ACY- (quoted form of (pos+AF8-filepath +ACY- finalname +ACY- +ACIAXw-1-+ACI- +ACY- i +ACY- +ACI-.+ACI- +ACY- namepart)) set theshellscript to the theshellscript +ACY- shellExec +ACY- +ACIAOw-sleep 3+ACI- +ACY- return set theshellscript to theshellscript +ACY- +ACIAOw-/bin/echo ' +AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQ- +ACI- +ACY- finalname +ACY- +ACIAXw-1-+ACI- +ACY- i +ACY- +ACI-.+ACI- +ACY- namepart +ACY- space +ACY- +ACI-FINISHED+ACEAIg- +ACY- +ACI- +AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQA9AD0APQ- '+ADs-mv+ACI- +ACY- space +ACY- (quoted form of tmpfile) +ACY- space +ACY- (quoted form of (POSIX path of (path to trash))) +ACY- +ACIAOw-mv ffmpegCatList.txt+ACI- +ACY- space +ACY- (quoted form of (POSIX path of (path to trash))) do shell script +ACI-echo +ACI- +ACY- quoted form of theshellscript +ACY- +ACI- +AD4- +ACI- +ACY- tmpfile repeat try do shell script +ACI-chmod +ACI- +ACY- tmpfile do shell script +ACI-open -a Terminal.app+ACI- +ACY- space +ACY- tmpfile exit repeat on error delay 1 end try end repeat end build+AF8-archive on replace+AF8-chars(this+AF8-text, +AF8-bad, +AF8-good) set AppleScript's text item delimiters to the +AF8-bad set the item+AF8-list to every text item of this+AF8-text set AppleScript's text item delimiters to the +AF8-good as string set this+AF8-text to the item+AF8-list as string set AppleScript's text item delimiters to +ACIAIg- return this+AF8-text end replace+AF8-chars on run --set the+AF8-items to ((choose folder) as list) build+AF8-archive(the+AF8-items) end run -----end script----- From roda at princeton.edu Tue Nov 4 19:09:10 2014 From: roda at princeton.edu (Reid Oda) Date: Tue, 4 Nov 2014 13:09:10 -0500 Subject: [FFmpeg-user] conversion from h.264 to mp3 adds 0.12 seconds of silence to beginning of file Message-ID: Hi list, I'm doing audio/video analysis on videos downloaded from youtube. As part of the process, I need to extract the audio to a form that can be opened by my audio analysis software. I've been using mp3, but I recently discovered that ffmpeg adds 0.12 seconds to the beginning of the .mp3 file. This causes sync problems. Here is the info on the youtube videos: Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s (default) Any thoughts on how to make sure that the extracted audio is exactly in line with the original aac audio (unfortunately, my software can't read aac)? Right now I am simply adding the offset onto my computations, but I would prefer to have a reliable conversion. Thank you very much for any help! From therealhdl at gmail.com Tue Nov 4 13:53:37 2014 From: therealhdl at gmail.com (Adrian Perez) Date: Tue, 4 Nov 2014 08:53:37 -0400 Subject: [FFmpeg-user] AV Foundation screen capture FPS Message-ID: <0BBB8B2B-613F-4562-8E7B-50062D4BFAD0@gmail.com> I?ve had no success trying to screen record past 15 FPS. Here?s my thread with the details: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=1783 I?d greatly appreciate any help with this, thanks. From reid.oda at gmail.com Tue Nov 4 21:54:09 2014 From: reid.oda at gmail.com (Reid Oda) Date: Tue, 4 Nov 2014 15:54:09 -0500 Subject: [FFmpeg-user] conversion from mp4 to mp3 adds 0.12 seconds of silence to beginning of file Message-ID: Hi list, I'm doing audio/video analysis on videos downloaded from youtube. As part of the process, I need to extract the audio to a form that can be opened by my audio analysis software. I've been using mp3, but I recently discovered that ffmpeg adds 0.12 seconds to the beginning of the .mp3 file. This causes sync problems. Here is the info on the youtube videos: Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s (default) Any thoughts on how to make sure that the extracted audio is exactly in line with the original aac audio (unfortunately, my software can't read aac)? Right now I am simply adding the offset onto my computations, but I would prefer to have a reliable conversion. Thank you very much for any help! From cehoyos at ag.or.at Tue Nov 4 23:31:39 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 4 Nov 2014 22:31:39 +0000 (UTC) Subject: [FFmpeg-user] AV Foundation screen capture FPS References: <0BBB8B2B-613F-4562-8E7B-50062D4BFAD0@gmail.com> Message-ID: Adrian Perez gmail.com> writes: > I?ve had no success trying to screen record past 15 FPS. Adrian has kindly opened ticket #4080. Carl Eugen From bluesky at caramail.com Tue Nov 4 23:36:42 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Tue, 4 Nov 2014 23:36:42 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , , Message-ID: "Carl Eugen Hoyos" wrote: > > What does your receiver show > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Sorry, I misread that. When playing the test file, in MythTV it depends on whether I have THX or Dolby selected (it plays the channels correctly in either). It either shows: THX Cinema 3/2.1 Dolby D 3/2.1 But in XBMC, no matter which of the two are selected, it simply says: PCM 48kHz > I know (expected) that MythTV and XBMC do not show > PLII. My receiver has a display and when digital > audio (as via S/PDIF) starts, it shows something > on this display. > While playing, I can switch the "Input mode" to > change between "Stereo" and "PLII". As I mentioned I have never heard of PLII prior to this, and my receiver doesn't have anything about that. >> My receiver has options for THX, Dolby/DTS >> Surround, Stereo, and Direct. > PLII is (the successor of) Dolby Surround afaict. > Since Dolby Surround and DTS Surround are two > completely different things afaik, I am > surprised that this is the same option... The button actually has the Dolby logo and the letters DTS above the button, and the word "SURROUND" below the button. I think that the way this receiver works is that you can select a mode, but if it gets a stream that is in a format it understands it will play it regardless of whether you have selected THX or Dolby/DTS. However if you select Stereo or Direct, a relay clicks inside and I think the rear channels are completely disabled, possibly for those who have this receiver but only stereo speakers connected. > I would be surprised if the file is THX but I > don't know much about it, my receiver does not > support it (too cheap). I think you are correct but as I say it plays exactly the same whether THX or Dolby/DTS is selected (within the same program). > (From a very quick look, THX is unrelated: > "A common error is that THX is similar to > Dolby Surround.") Which is what makes me think that this receiver will let you select one or the other, but actually uses what it receives. Keep in mind that all of this is with a S/PDIF (Toslink audio) connection; it might be different if the receiver had HDMI support, but it's not quite that new. Thank you again and sorry for sending the wrong information the first time. From cehoyos at ag.or.at Wed Nov 5 01:17:49 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 00:17:49 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , , Message-ID: caramail.com> writes: > > > What does your receiver show > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > When playing the test file, in MythTV it depends on > whether I have THX or Dolby selected (it plays the > channels correctly in either). It either shows: > THX Cinema 3/2.1 > Dolby D 3/2.1 > > But in XBMC, no matter which of the two are selected, > it simply says: > PCM 48kHz This indicates that MythTV really has a PLII decoder. [...] > Keep in mind that all of this is with a S/PDIF > (Toslink audio) connection; As said, I also tested with S/PDIF, only analog makes a difference here: I have to force stereo input on the receiver to allow PLII decoding. Carl Eugen From lou at lrcd.com Wed Nov 5 01:51:05 2014 From: lou at lrcd.com (Lou) Date: Tue, 4 Nov 2014 15:51:05 -0900 Subject: [FFmpeg-user] conversion from h.264 to mp3 adds 0.12 seconds of silence to beginning of file In-Reply-To: References: Message-ID: <20141104155105.30642d8d@lrcd.com> On Tue, 4 Nov 2014 13:09:10 -0500 Reid Oda wrote: > Hi list, > > I'm doing audio/video analysis on videos downloaded from youtube. As part > of the process, I need to extract the audio to a form that can be opened by > my audio analysis software. I've been using mp3, but I recently discovered > that ffmpeg adds 0.12 seconds to the beginning of the .mp3 file. This > causes sync problems. Your ffmpeg command and the complete console output are missing. Does this also occur if you use lame? From lou at lrcd.com Wed Nov 5 01:53:15 2014 From: lou at lrcd.com (Lou) Date: Tue, 4 Nov 2014 15:53:15 -0900 Subject: [FFmpeg-user] conversion from h.264 to mp3 adds 0.12 seconds of silence to beginning of file In-Reply-To: <20141104155105.30642d8d@lrcd.com> References: <20141104155105.30642d8d@lrcd.com> Message-ID: <20141104155315.4e73cf72@lrcd.com> On Tue, 4 Nov 2014 15:51:05 -0900 Lou wrote: > Your ffmpeg command and the complete console output are missing. > > Does this also occur if you use lame? I forgot to mention: 2. Why does LAME add silence to the beginning each song? http://lame.sourceforge.net/tech-FAQ.txt From reid.oda at gmail.com Wed Nov 5 02:02:41 2014 From: reid.oda at gmail.com (Reid Oda) Date: Tue, 4 Nov 2014 20:02:41 -0500 Subject: [FFmpeg-user] conversion from h.264 to mp3 adds 0.12 seconds of silence to beginning of file In-Reply-To: <20141104155315.4e73cf72@lrcd.com> References: <20141104155105.30642d8d@lrcd.com> <20141104155315.4e73cf72@lrcd.com> Message-ID: Thank you so much! This answers my question. I'm going to use uncompressed audio from here on out for the analysis. On Tue, Nov 4, 2014 at 7:53 PM, Lou wrote: > On Tue, 4 Nov 2014 15:51:05 -0900 > Lou wrote: > > > Your ffmpeg command and the complete console output are missing. > > > > Does this also occur if you use lame? > > I forgot to mention: > > 2. Why does LAME add silence to the beginning each song? > http://lame.sourceforge.net/tech-FAQ.txt > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From stansp2004 at mail.ru Wed Nov 5 07:53:34 2014 From: stansp2004 at mail.ru (=?UTF-8?B?U3RhbiBTdGFu?=) Date: Wed, 05 Nov 2014 09:53:34 +0300 Subject: [FFmpeg-user] =?utf-8?q?Errors_while_muxing_into_=2Emkv_from_ip_c?= =?utf-8?q?amera?= In-Reply-To: References: <1414764702.199801535@f301.i.mail.ru> Message-ID: <1415170414.473770846@f20.i.mail.ru> Thank for your response! We tried employing ffmpeg without specifying -f option, but it yield us almost the same result. How did you determine that only first three frame are decoded successfully? With best regards, Lev. Sat, 1 Nov 2014 10:10:37 +0000 (UTC) ?? Carl Eugen Hoyos : >Ddfsdf sdfsdf mail.ru> writes: > >> What's the reason of arising these errors in log? > >The reference decoder only decodes the first three >frames of your input file so I suggest it is broken >(reception or encoding errors). > >> ffmpeg.exe -f h264 -i video.raw video.mkv > >"-f h264" is unneeded. > >Carl Eugen > >_______________________________________________ >ffmpeg-user mailing list >ffmpeg-user at ffmpeg.org >http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Stan Stan From cehoyos at ag.or.at Wed Nov 5 10:02:19 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 09:02:19 +0000 (UTC) Subject: [FFmpeg-user] Errors while muxing into .mkv from ip camera References: <1414764702.199801535@f301.i.mail.ru> <1415170414.473770846@f20.i.mail.ru> Message-ID: Stan Stan mail.ru> writes: > >The reference decoder only decodes the first three > >frames of your input file so I suggest it is broken This should have been "suspect". > >(reception or encoding errors). > > How did you determine that only first three frame > are decoded successfully? I tried to decode the file with the reference decoder and it aborts after three frames. Please do not top-post here, Carl Eugen From stansp2004 at mail.ru Wed Nov 5 12:06:09 2014 From: stansp2004 at mail.ru (=?UTF-8?B?U3RhbiBTdGFu?=) Date: Wed, 05 Nov 2014 14:06:09 +0300 Subject: [FFmpeg-user] =?utf-8?q?Errors_while_muxing_into_=2Emkv_from_ip_c?= =?utf-8?q?amera?= In-Reply-To: References: <1414764702.199801535@f301.i.mail.ru> Message-ID: <1415185569.992147561@f223.i.mail.ru> I'm looking at log file produced by ffmpeg with -loglevel debug and can't see that frame errors starts from frame #4. Please promt where I can find this information. How can I guess exactly what's the reason of these errors? Are there any tools for stream analyzing? From cehoyos at ag.or.at Wed Nov 5 12:49:18 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 11:49:18 +0000 (UTC) Subject: [FFmpeg-user] Errors while muxing into .mkv from ip camera References: <1414764702.199801535@f301.i.mail.ru> <1415185569.992147561@f223.i.mail.ru> Message-ID: Stan Stan mail.ru> writes: > I'm looking at log file produced by ffmpeg with -loglevel debug > and can't see that frame errors starts from frame #4. > Please promt where I can find this information. I don't understand this. > How can I guess exactly what's the reason of these errors? > Are there any tools for stream analyzing? I believe the reference decoder can be used to analyze the correctness of H264 streams. Carl Eugen From mjmst74 at gmail.com Wed Nov 5 15:39:48 2014 From: mjmst74 at gmail.com (m. mood) Date: Wed, 5 Nov 2014 09:39:48 -0500 Subject: [FFmpeg-user] Sample files for testing/using ffmpeg? Message-ID: Does the ffmpeg project have any sample/test data available of audio files in the various supported audio formats? I'd like to be able to try out audio format conversions on the various formats that ffmpeg supports, using a known good set of sample audio files. Thanks, Mitch From james.darnley at gmail.com Wed Nov 5 15:47:13 2014 From: james.darnley at gmail.com (James Darnley) Date: Wed, 05 Nov 2014 15:47:13 +0100 Subject: [FFmpeg-user] Sample files for testing/using ffmpeg? In-Reply-To: References: Message-ID: <545A3871.8040105@gmail.com> On 2014-11-05 15:39, m. mood wrote: > Does the ffmpeg project have any sample/test data available of audio files > in the various supported audio formats? I'd like to be able to try out > audio format conversions on the various formats that ffmpeg supports, using > a known good set of sample audio files. There is the FATE testing system which we use for testing assorted components of FFmpeg. You can read about it here: http://ffmpeg.org/fate.html What exactly are you looking to test? On sox-users you said you wanted to test its filters on various inputs and you were told that sox converts to in internal format. ffmpeg (the tool) also automatically converts formats between filters if neither share a common format. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From barsnick at gmx.net Wed Nov 5 16:19:29 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Wed, 5 Nov 2014 16:19:29 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: Message-ID: <20141105151929.GE2966@sunshine.barsnick.net> On Tue, Nov 04, 2014 at 20:12:56 +0100, bluesky at caramail.com wrote: > XBMC WILL play Dolby encoded files correctly if the channels > are all present and not encoded in some weird format, in fact it is > very good at that. That's understood. > It would be really great if ffmpeg could detect these channels and > reencode them to a more standard (if that is the way to put it) ac3 > format. This is the technicality which needs to be understood. If the source is what Carl Eugen has identified, there is no digital indication that this is "Dolby Surround". You will identify two channels which can be played on stereo output, and will sound like a down-mix of surround material on stereo. Yet in these two channels, there is a mixture of four or six audio channels. If you know how to "decode" them, you can separate them. Actually, you should be able to detect this surround encoding if you analyze the audio waveforms, but this may mean actually decoding them and checking whether the "extra" channels contain sound. (My hardware decoder creates pseudo-surround effects from plain stereo material by the way, due to the way it "tries" to decode 6 from 2 channels and the audio's phases.) Actually, a lot of digital TV channels broadcast their movies with surround sound like this, i.e. using two channels, but still containing surround. Mostly this is the case when they don't have 6 channels available. This is very common in broadcast, from my experience. Basically, ffmpeg currently can neither detect nor decode this encoding. I think decoding it would require something like an audio filter. > "Carl Eugen Hoyos" wrote: > > I will hopefully open a ticket for PLII decoding, > > I have no idea how difficult / trivial this is > > given that we support encoding... Since it's easy to find algorithms (i.e. matrices) for encoding but not for decoding, I am guessing that the latter may be a much tougher task. It's not as easy as inverting the matrix. I'm no smarter than Carl Eugen concerning this. Here's the only description I have found of the process, and it looks very DSPish (if not analog): http://www.eetimes.com/document.asp?doc_id=1225389 See figure 5 for a summarizing diagram. No C source code provided though. ;-) Moritz From llee040 at sbcglobal.net Wed Nov 5 16:22:16 2014 From: llee040 at sbcglobal.net (L. Lee) Date: Wed, 05 Nov 2014 09:22:16 -0600 Subject: [FFmpeg-user] Any Mac OS X users interested in ffmpeg AppleScript droplets? - here's one for cat Message-ID: If you+IBk-re interested, here+IBk-s an OS X (Mac) AppleScript droplet source for concatenating multiple video files using existing ffmpeg executable. How do I use it? Copy the lines containing "begin script" through "end script" to your Mac's clipboard. Paste the clipboard contents into a Script (or AppleScript, depending on your version of OS X) Edtor window, then save as an application. Name your video files successively, and drop them all onto the AppleScript application's icon. What does it do? Uses your compiled version of the ffmpeg executable to concatenate the files in ascending order, creating the resulting file in the same location as the files you dropped. The ffmpeg executable should be compiled using Homebrew and/or be accessible by your command line shell at /usr/local/bin/ffmpeg. You should be able to change that specification in the AppleScript if you keep ffmpeg somewhere else. I+IBk-ve tested with .m4v files containing 5.1 audio. Please help me test others. Thanks. -----begin script----- property temppath : "/private/tmp/" property startnum : 0 property tmpfile : "/tmp/execme.command" on open the_items my build_archive(the_items) end open on build_archive(the_items) set theshellscript to "" repeat with i from 1 to (count of the_items) set itemcount to (count of the_items) set the_item to item i of the_items as alias try tell application "Finder" set sost to (container of the_item) as string end tell set pos_filepath to POSIX path of sost end try set this_filepath to (the_item as string) if last character of this_filepath is ":" then tell me to set it_is_a_folder to true else set it_is_a_folder to false end if set thesourcename to (name of (info for the_item)) set namepart to (name extension of (info for the_item)) set the_source_file to POSIX path of this_filepath --set newname to replace_chars(thesourcename, namepart, "joined.mp4") set finalname to replace_chars(thesourcename, "." & namepart, "") try if i = 1 then set the filelistbody to "# comment" & return & "file" & space & (quoted form of the_source_file) & return else set the filelistbody to filelistbody & "file" & space & (quoted form of the_source_file) & return end if on error onerr activate display dialog onerr end try end repeat set fileData to "echo" & space & "+AFw"" & filelistbody & "+AFw"" & space & "> ffmpegCatList.txt" set theshellscript to fileData & ";sleep 3;/usr/local/bin/ffmpeg -f concat -i ffmpegCatList.txt -c copy" & space set shellExec to space & (quoted form of (pos_filepath & finalname & "_1-" & i & "." & namepart)) set theshellscript to the theshellscript & shellExec & ";sleep 3" & return set theshellscript to theshellscript & ";/bin/echo ' ========================== " & finalname & "_1-" & i & "." & namepart & space & "FINISHED!" & " ========================== ';mv" & space & (quoted form of tmpfile) & space & (quoted form of (POSIX path of (path to trash))) & ";mv ffmpegCatList.txt" & space & (quoted form of (POSIX path of (path to trash))) do shell script "echo " & quoted form of theshellscript & " > " & tmpfile repeat try do shell script "chmod " & tmpfile do shell script "open -a Terminal.app" & space & tmpfile exit repeat on error delay 1 end try end repeat end build_archive on replace_chars(this_text, _bad, _good) set AppleScript's text item delimiters to the _bad set the item_list to every text item of this_text set AppleScript's text item delimiters to the _good as string set this_text to the item_list as string set AppleScript's text item delimiters to "" return this_text end replace_chars on run --set the_items to ((choose folder) as list) build_archive(the_items) end run -----end script----- From bluesky at caramail.com Wed Nov 5 18:08:54 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Wed, 5 Nov 2014 18:08:54 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <20141105151929.GE2966@sunshine.barsnick.net> References: , <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: "Moritz Barsnick" wrote: > This is the technicality which needs to be understood. If the source is > what Carl Eugen has identified, there is no digital indication that > this is "Dolby Surround". You will identify two channels which can be > played on stereo output, and will sound like a down-mix of surround > material on stereo. Yet in these two channels, there is a mixture of > four or six audio channels. If you know how to "decode" them, you can > separate them. Actually, you should be able to detect this surround > encoding if you analyze the audio waveforms, but this may mean actually > decoding them and checking whether the "extra" channels contain sound. > (My hardware decoder creates pseudo-surround effects from plain stereo > material by the way, due to the way it "tries" to decode 6 from 2 > channels and the audio's phases.) I might actually believe I was hearing pseudo-surround in MythTV if I did not have the one test recording that clearly and unambiguously states the name of each channel as it plays. Apparently, by luck or by design, the MythTV people have stumbled onto a way to play these types of recordings correctly. If they have figured it out, I would think it's not an impossible thing for someone who understands what they are doing. But that said... > Actually, a lot of digital TV channels broadcast their movies with > surround sound like this, i.e. using two channels, but still containing > surround. Mostly this is the case when they don't have 6 channels > available. This is very common in broadcast, from my experience. In the test file I have, on the video there is a line that says "Dolby E", which I never paid much attention to because none of the audio streams identify as ac3 format and because you could write what I know about Dolby on the head of a pin. There is a page about Dolby E at http://wiki.multimedia.cx/index.php?title=Dolby_E but I do not know if it is applicable in this situation. I also found mentions of Dolby E here https://forum.videolan.org/viewtopic.php?f=18&t=81323 and here http://www.videoredo.net/msgBoard/showthread.php?28982-Dealing-with-Dolby-E&s=9c4d4a18d25bf5b63595673bdaec178b There is a program that supposedly can convert E-AC3 audio tracks at http://www.addictivetips.com/windows-tips/how-to-convert-dolby-truehd-and-e-ac3-audio-tracks-into-other-audio-formats-with-eac3to/ but it appears to be a Windows only program (there is a line in the article that claims that it works in Linux but when you go to this thread http://forum.doom9.org/showthread.php?t=125966 it appears to be a very Windows-centric program). A Windows program does me no good, especially if it in turn depends on the installation of other types of Windows software, because I have no computers running Windows (my desktop is OS X and the system I want to convert these files on is Linux). I had originally thought that because the original file's audio is MPEG2 that perhaps something like mctooLAME (http://sourceforge.net/projects/mctoolame/?source=typ_redirect) would be helpful but the more comments I hear about it the more I suspect we are dealing with E-AC3 (Dolby E) here, though I don't understand how that can be possible if the audio tracks identify as MPEG2. > Basically, ffmpeg currently can neither detect nor decode this > encoding. I think decoding it would require something like an audio > filter. I understand that but it still begs the question of how MythTV pulls it off. In case I didn't mention it, I am not a technically-inclined person and definitely not a programmer, so most of the material in the above links is way over my head. I don't even know for sure that this is E-AC3 and I have not even found a utility that will positively identify E-AC3. And yet it appears that somehow the MythTV people alone have it all figured out, at least as far as being able to play it correctly without the need for Windows or a bunch of additional software. > Since it's easy to find algorithms (i.e. matrices) for encoding but not > for decoding, I am guessing that the latter may be a much tougher task. > It's not as easy as inverting the matrix. I'm no smarter than Carl > Eugen concerning this. Here's the only description I have found of the > process, and it looks very DSPish (if not analog): > http://www.eetimes.com/document.asp?doc_id=1225389 > See figure 5 for a summarizing diagram. No C source code provided > though. ;-) All of that was way beyond my understanding. I don't know if any of the links I have provided above will prove to be any more helpful, but perhaps there will be something in one of those that makes sense to you guys. I'm already in way over my head here - I had no idea that this might be something complicated, because I just assumed that if MythTV can decode these signals then it must not be too difficult. But then I honestly have no idea what is actually involved, nor the slightest clue how MythTV does it. From l1 at newanswertech.com Wed Nov 5 20:01:37 2014 From: l1 at newanswertech.com (Luke Davis) Date: Wed, 5 Nov 2014 14:01:37 -0500 (EST) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: On Wed, 5 Nov 2014, bluesky at caramail.com wrote: > "Moritz Barsnick" wrote: > >> This is the technicality which needs to be understood. If the source is >> what Carl Eugen has identified, there is no digital indication that >> this is "Dolby Surround". You will identify two channels which can be >> played on stereo output, and will sound like a down-mix of surround >> material on stereo. Yet in these two channels, there is a mixture of >> four or six audio channels. If you know how to "decode" them, you can >> separate them. Actually, you should be able to detect this surround >> encoding if you analyze the audio waveforms, but this may mean actually >> decoding them and checking whether the "extra" channels contain sound. > > I might actually believe I was hearing pseudo-surround in MythTV if I did not > have the one test recording that clearly and unambiguously states the name of > each channel as it plays. Apparently, by luck or by design, the MythTV people > have stumbled onto a way to play these types of recordings correctly. If they > have figured it out, I would think it's not an impossible thing for someone > who understands what they are doing. But that said... I know nothing about this subject at all. However the discussion reminded me of something that could possibly be relevant. Steve Harris has one or two LADSPA effects for working with Dolby Surround, which may be what you're talking about here if I understand the most recent post correctly (I have not been following the thread) . Perhaps talking to him about this could be informative? http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.102 Then again, maybe that's the wrong tree to be barking up entirely, but I thought I'd point it out just in case it was good for something. Regards, Luke From cehoyos at ag.or.at Wed Nov 5 23:34:17 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 22:34:17 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > Basically, ffmpeg currently can neither detect nor > decode this encoding. I think decoding it would > require something like an audio filter. The filter and the library are already there, they are just missing this functionality... [...] > http://www.eetimes.com/document.asp?doc_id=1225389 I wonder if the Wikipedia article and the FFmpeg source code don't contain more information... > See figure 5 for a summarizing diagram. No C > source code provided though. MythTV might have it. Carl Eugen From cehoyos at ag.or.at Wed Nov 5 23:37:39 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 22:37:39 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: Luke Davis newanswertech.com> writes: > http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.102 This looks like a Dolby PLII encoder, FFmpeg also contains a PLII encoder. What we don't have is a decoder. Carl Eugen From cehoyos at ag.or.at Wed Nov 5 23:36:34 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 22:36:34 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: caramail.com> writes: > There is a page about Dolby E at > http://wiki.multimedia.cx/index.php?title=Dolby_E but > I do not know if it is applicable in this situation. No, the test file does not contain Dolby E audio. FFmpeg does not support Dolby E. Please understand that Dolby E != E-AC3. E-AC3 decoding and encoding (up to six channels) are supported by FFmpeg. Carl Eugen From barsnick at gmx.net Wed Nov 5 23:45:20 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Wed, 5 Nov 2014 23:45:20 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: <20141105151929.GE2966@sunshine.barsnick.net> Message-ID: <20141105224520.GA10947@sunshine.barsnick.net> On Wed, Nov 05, 2014 at 14:01:37 -0500, Luke Davis wrote: > Steve Harris has one or two LADSPA effects for working with Dolby Surround, > which may be what you're talking about here if I understand the most recent > post correctly (I have not been following the thread) > > http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.102 That's actually the exact opposite: "It allows you to encode four channels of sound into a stereo compatible stream that will be decoded by a Dolby1 Surround/Pro-Logic decoder into Left, Right, Center and Surround signals." ffmpeg already (apparently) supports this, see "-matrix_encoding", https://www.ffmpeg.org/ffmpeg-resampler.html The original poster needs the opposite, a decoder: 2 channels back to 5.1 (or 4.0). I have been at a total loss when googling. The only free software I have found is AC3Filter, http://www.ac3filter.net/ . I actually missed this important sound, but see it now: It's open source, and claims to be GPL2 (plus/minus the licenses of other included code). I'm still skimming through its code though, looking for the code which accomplishes the up-mix. Moritz From cehoyos at ag.or.at Thu Nov 6 00:01:57 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 23:01:57 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: <20141105151929.GE2966@sunshine.barsnick.net> <20141105224520.GA10947@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > I have been at a total loss when googling. el_processor.cpp is not license-compatible but indicates that it is possible to achieve what is needed. Carl Eugen PS: What did you google knowing that MythTV does support PLII? From barsnick at gmx.net Thu Nov 6 00:15:10 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Thu, 6 Nov 2014 00:15:10 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: <20141105151929.GE2966@sunshine.barsnick.net> <20141105224520.GA10947@sunshine.barsnick.net> Message-ID: <20141105231510.GA12499@sunshine.barsnick.net> On Wed, Nov 05, 2014 at 23:01:57 +0000, Carl Eugen Hoyos wrote: > el_processor.cpp is not license-compatible but Because it's GPLv2+, but libavfilter is LGPL? (Just checking, this makes sense. > indicates that it is possible to achieve what > is needed. Reading the code may help to understand the original descriptions of the methods needed to decode the 2-channel input (such as the article and image I referred to, and possibly other whitepapers from Dolby Labs), or find other references which mention Fourier transformation for our digital domain. I didn't come up with that myself, but should've. This could allow for a clean-room implementation. *shrug* > PS: What did you google knowing that MythTV > does support PLII? That's the part of the configuration _I_ didn't understand. I had only jumped onto what you had concluded. ;-) I googled for surround decoders, or dolby decoders, and omitting the MythTV fact didn't lead me to MythTV. Moritz From cehoyos at ag.or.at Thu Nov 6 00:23:19 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 5 Nov 2014 23:23:19 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: <20141105151929.GE2966@sunshine.barsnick.net> <20141105224520.GA10947@sunshine.barsnick.net> <20141105231510.GA12499@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > This could allow for a clean-room implementation. *shrug* Or you could ask the original author if he agrees to relicensing his code. I opened ticket #4085 with the MythTV sample. Carl Eugen From paulo.fidalgo.pt at gmail.com Thu Nov 6 11:26:03 2014 From: paulo.fidalgo.pt at gmail.com (Paulo Fidalgo) Date: Thu, 06 Nov 2014 10:26:03 +0000 Subject: [FFmpeg-user] MP3 enconding bitrate In-Reply-To: <20141030090726.GB12578@sunshine.barsnick.net> References: <5447FD04.80602@gmail.com> <544E9EC5.9070706@gmail.com> <201410281155.40532.cehoyos@ag.or.at> <5451F142.9070903@gmail.com> <20141030090726.GB12578@sunshine.barsnick.net> Message-ID: <545B4CBB.2050409@gmail.com> On 30/10/14 09:07, Moritz Barsnick wrote: > On Thu, Oct 30, 2014 at 08:05:22 +0000, Paulo Fidalgo wrote: >> I've tried with 320k and 256k and both files don't play. > While applying trial and error using ffmpeg, you might also want to > analyze the failing (and the successful) files with an MP3 diagnosis > tool. There are quite a few out there, even open source. I have used > MP3Diags, and recently found a few others such as mp3check and > mp3val.[*] I can only get errors from mp3check: mp3check -ve 2L38_01_96kHz-ffmpeg-256k.mp3 2L38_01_96kHz-ffmpeg-256k.mp3: 813 bytes of junk before first frame header but with lame there's no errors. > > I can't tell though whether these tools would detect your issue. Some > apparently only actually process the _headers_. > > Moritz > > [*] Best link I found, in German though: > http://wiki.ubuntuusers.de/%C3%9Cberpr%C3%BCfung_MP3-Sammlung > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From shacky83 at gmail.com Thu Nov 6 13:21:28 2014 From: shacky83 at gmail.com (shacky) Date: Thu, 6 Nov 2014 13:21:28 +0100 Subject: [FFmpeg-user] FFmpeg record RTMP to FLV Message-ID: Hi. I'm trying to record some RTMP stream with FFmpeg to a FLV file or something else. I'm trying the following command: ffmpeg -i "rtmp://94.47.147.130:1937/live/livestream" -f flv -t 60 test.flv But I receive a 6 byte file which seems to contain only one frame. This is the output of ffmpeg: ffmpeg -i "rtmp://94.47.147.130:1937/live/livestream" -f flv -t 60 test.flv ffmpeg version 1.2.4 Copyright (c) 2000-2013 the FFmpeg developers built on Jan 14 2014 16:31:42 with Apple LLVM version 5.0 (clang-500.2.79) (based on LLVM 3.3svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/1.2.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid libavutil 52. 18.100 / 52. 18.100 libavcodec 54. 92.100 / 54. 92.100 libavformat 54. 63.104 / 54. 63.104 libavdevice 54. 3.103 / 54. 3.103 libavfilter 3. 42.103 / 3. 42.103 libswscale 2. 2.100 / 2. 2.100 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 [flv @ 0x7fd26a81d400] negative cts, previous timestamps might be wrong Last message repeated 2 times [flv @ 0x7fd26a81d400] Estimating duration from bitrate, this may be inaccurate Input #0, flv, from 'rtmp://94.47.147.130:1937/live/livestream': Metadata: author : copyright : description : keywords : rating : title : presetname : Custom creationdate : Thu Sep 18 17:08:34 2014 : videodevice : Osprey-230 Video Device 1 avclevel : 31 avcprofile : 66 videokeyframe_frequency: 5 audiodevice : Osprey-230 Audio Device 1L audiochannels : 2 audioinputvolume: 53 Duration: N/A, start: -5607.794000, bitrate: 456 kb/s Stream #0:0: Video: h264 (Baseline), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 358 kb/s, 25.33 tbr, 1k tbn, 50 tbc Stream #0:1: Audio: mp3, 44100 Hz, stereo, s16p, 98 kb/s Output #0, flv, to 'test.flv': Metadata: author : copyright : description : keywords : rating : title : presetname : Custom creationdate : Thu Sep 18 17:08:34 2014 : videodevice : Osprey-230 Video Device 1 avclevel : 31 avcprofile : 66 videokeyframe_frequency: 5 audiodevice : Osprey-230 Audio Device 1L audiochannels : 2 audioinputvolume: 53 encoder : Lavf54.63.104 Stream #0:0: Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 1k tbn, 25.33 tbc Stream #0:1: Audio: mp3 ([2][0][0][0] / 0x0002), 44100 Hz, stereo, s16p Stream mapping: Stream #0:0 -> #0:0 (h264 -> flv) Stream #0:1 -> #0:1 (mp3 -> libmp3lame) Press [q] to stop, [?] for help DTS 4294972256, next:5607874000 st:0 invalid dropping PTS 4294972256, next:5607874000 invalid dropping st:0 DTS 4294972296, next:5607914000 st:0 invalid dropping PTS 4294972296, next:5607914000 invalid dropping st:0 frame= 1 fps=0.0 q=3.8 Lsize= 10kB time=01:33:27.82 bitrate= 0.0kbits/s dup=0 drop=3 video:9kB audio:1kB subtitle:0 global headers:0kB muxing overhead 7.016300% Could you help me to understand how I can do, please? Thank you very much. Bye From barsnick at gmx.net Thu Nov 6 13:38:59 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Thu, 6 Nov 2014 13:38:59 +0100 Subject: [FFmpeg-user] FFmpeg record RTMP to FLV In-Reply-To: References: Message-ID: <20141106123859.GC15711@sunshine.barsnick.net> On Thu, Nov 06, 2014 at 13:21:28 +0100, shacky wrote: > ffmpeg version 1.2.4 Copyright (c) 2000-2013 the FFmpeg developers This is a quite old version. I don't know if RTMP support was as good back then as it is now. 2.4.x is current. please try to get hold of that, or use latest git if you build your own. > I'm trying to record some RTMP stream with FFmpeg to a FLV file or > something else. The way you're doing it is not optimal. a) For dumping streams to disk, "mplayer -dumpstream" is the recommended path. :-) b) You're re-encoding. That may or may not be your intention. To retain the original quality, use "-c copy". Actually, answer b) may help you even with your old ffmpeg version. Moritz From cehoyos at ag.or.at Thu Nov 6 19:12:40 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 6 Nov 2014 18:12:40 +0000 (UTC) Subject: [FFmpeg-user] MP3 enconding bitrate References: <5447FD04.80602@gmail.com> <544E9EC5.9070706@gmail.com> <201410281155.40532.cehoyos@ag.or.at> <5451F142.9070903@gmail.com> <20141030090726.GB12578@sunshine.barsnick.net> <545B4CBB.2050409@gmail.com> Message-ID: Paulo Fidalgo gmail.com> writes: > mp3check -ve 2L38_01_96kHz-ffmpeg-256k.mp3 > 2L38_01_96kHz-ffmpeg-256k.mp3: > 813 bytes of junk before first frame header > > but with lame there's no errors. Please confirm that you tested with "-write_xing 0" and your results. Please understand that I should have realised this from the beginning but since you insisted on FFmpeg writing frames larger 320k I concentrated on that. Sorry, Carl Eugen From 277893958 at qq.com Thu Nov 6 10:47:45 2014 From: 277893958 at qq.com (cmwu) Date: Thu, 6 Nov 2014 01:47:45 -0800 (PST) Subject: [FFmpeg-user] what's the threads mode when using ffmpeg to transcode Message-ID: <1415267265375-4668036.post@n4.nabble.com> hi, I am confused how ffmpeg allocate it's cpu resource when i am using ffmpeg to transcoding, for example: how much resource to decode, and how much to encode I have set the threads to 1 and do a experiment: ffmpeg2.1 -i /home/mps/chd/out.avi -acodec pcm_s16le -ar 44100 -ac 2 -vcodec libx264 -coder 1 -b:v 250k -x264opts min-keyint=1:keyint=75:no-dct-decimate:partitions=all:8x8dct:merange=24:trellis=1:ratetol=1:qpmin=20:qpmax=45:aq-strength=1.0:psy-rd=1,0:subme=7:qcomp=0.6:chroma-qp-offset=0:me=umh:frameref=6:bframes=6:b-pyramid=1:b-adapt=2:weightp=2:direct=auto:deblock=1,0:stitchable=1, threads=1 -timelimit 6000 -r 15 -pass 1 -passlogfile /home/mps/cmwu/passlog_1414744300_h264_1.log -f null -y /dev/null I found the cpu load use is 140%, it's not 100% Can anybody explain it for me ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/what-s-the-threads-mode-when-using-ffmpeg-to-transcode-tp4668036.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From belcampo at zonnet.nl Thu Nov 6 21:31:05 2014 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Thu, 6 Nov 2014 21:31:05 +0100 Subject: [FFmpeg-user] what's the threads mode when using ffmpeg to transcode In-Reply-To: <1415267265375-4668036.post@n4.nabble.com> References: <1415267265375-4668036.post@n4.nabble.com> Message-ID: <0C24DD1B-58F1-4A47-B186-C38F6065A8EE@zonnet.nl> On 06 Nov 2014, at 10:47, cmwu <277893958 at qq.com> wrote: > hi, > I am confused how ffmpeg allocate it's cpu resource when i am using ffmpeg > to transcoding, > > for example: > how much resource to decode, and how much to encode > > I have set the threads to 1 and do a experiment: > ffmpeg2.1 -i /home/mps/chd/out.avi -acodec pcm_s16le -ar 44100 -ac 2 > -vcodec libx264 -coder 1 -b:v 250k -x264opts > min-keyint=1:keyint=75:no-dct-decimate:partitions=all:8x8dct:merange=24:trellis=1:ratetol=1:qpmin=20:qpmax=45:aq-strength=1.0:psy-rd=1,0:subme=7:qcomp=0.6:chroma-qp-offset=0:me=umh:frameref=6:bframes=6:b-pyramid=1:b-adapt=2:weightp=2:direct=auto:deblock=1,0:stitchable=1, > threads=1 -timelimit 6000 -r 15 -pass 1 -passlogfile > /home/mps/cmwu/passlog_1414744300_h264_1.log -f null -y /dev/null > > I found the cpu load use is 140%, it's not 100% AFAIK threads 1 is meaning 1 Core for Encoding. So the extra 40% is used for decoding, logfile/IO activity and Audio stuff. > > Can anybody explain it for me ? > > > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/what-s-the-threads-mode-when-using-ffmpeg-to-transcode-tp4668036.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: Message signed with OpenPGP using GPGMail URL: From leesiusing at gmail.com Thu Nov 6 22:38:39 2014 From: leesiusing at gmail.com (Aaron Lee) Date: Fri, 7 Nov 2014 05:38:39 +0800 Subject: [FFmpeg-user] going files will slightly different FPS Message-ID: <0ACE867C-7700-4C81-BF07-5D0E246ACD12@gmail.com> Due to some hardware limitation, the my video capture box captured video in mp4 format with slightly different fps such as ranges from 59.21 and 60.01. 1. why would a machine have fluctuating fps like this? 2. how can i join them without reenoding them? From cehoyos at ag.or.at Fri Nov 7 00:22:28 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 6 Nov 2014 23:22:28 +0000 (UTC) Subject: [FFmpeg-user] what's the threads mode when using ffmpeg to transcode References: <1415267265375-4668036.post@n4.nabble.com> Message-ID: cmwu <277893958 qq.com> writes: > ffmpeg2.1 This looks old > -i /home/mps/chd/out.avi > I found the cpu load use is 140%, it's not 100% If you don't want FFmpeg to decode (!) with automatically chosen number of threads (as many cores as you have), tell it with: $ ffmpeg -threads 1 -i file Carl Eugen From morndust at gmail.com Fri Nov 7 03:37:53 2014 From: morndust at gmail.com (Miles Chan) Date: Fri, 7 Nov 2014 10:37:53 +0800 Subject: [FFmpeg-user] Execute several ffmpeg processes concurrently Message-ID: hi all, I'm using ffmpeg command line tool to convert video, (FYI, i'm not just use it in shell, i use a module that execute a ffmpeg command as a subprocess), and when i execute ffmpeg process one by one, everything was all right, but, when i execute 5 and more processes concurrently, things went bad. One of the error is "Invalid data found when processing input", another is ffmpeg process exit with code 1. This is my command. "usr/bin/ffmpeg -i 20130804_100547.mp4 -hls_base_url /test2/ts/ -hls_time 10 -hls_list_size 0 -vf transpose=1 -y small.m3u8" My test hardware is 24 cpu and 49436312k mem From cehoyos at ag.or.at Fri Nov 7 09:42:40 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 7 Nov 2014 08:42:40 +0000 (UTC) Subject: [FFmpeg-user] Execute several ffmpeg processes concurrently References: Message-ID: Miles Chan gmail.com> writes: > One of the error is "Invalid data found when processing input", > another is ffmpeg process exit with code 1. Complete, uncut console output of the failing command missing. And please show how the other commands look like. Carl Eugen From kuban.altan at gmail.com Fri Nov 7 14:27:36 2014 From: kuban.altan at gmail.com (Kuban Altan) Date: Fri, 7 Nov 2014 15:27:36 +0200 Subject: [FFmpeg-user] processing a vumeter (levelmeter) as video for audio inputs Message-ID: Hi, I am testing *ebur128* filter, which can produce a video output. But I need a simple *vumeter* display per audio channel, which does not seem to exist. Any idea to accomplish this? So far, I have only been able to *volumedetect* per frame, mentioning a -ss time and a duration with -t 0.04 (for 25fps)... So I can get a peak sound level as text output using volumedetect. But this is not very efficient. All I need is a video and 2 rectangular bars, which have their lengths connected to input audio level. Any ideas on doing this? kuban altan From dave at dericed.com Fri Nov 7 16:21:50 2014 From: dave at dericed.com (Dave Rice) Date: Fri, 7 Nov 2014 10:21:50 -0500 Subject: [FFmpeg-user] processing a vumeter (levelmeter) as video for audio inputs In-Reply-To: References: Message-ID: <017D58C6-9D15-4F06-AE72-7D09E0D7DDA3@dericed.com> Hi, > On Nov 7, 2014, at 8:27 AM, Kuban Altan wrote: > > Hi, > I am testing *ebur128* filter, which can produce a video output. But I need > a simple *vumeter* display per audio channel, which does not seem to exist. > Any idea to accomplish this? > > So far, I have only been able to *volumedetect* per frame, mentioning a -ss > time and a duration with -t 0.04 (for 25fps)... So I can get a peak sound > level as text output using volumedetect. But this is not very efficient. > All I need is a video and 2 rectangular bars, which have their lengths > connected to input audio level. There is a momentarily ebur128 level in the ebur128 video output so you could crop that out and reuse it noting that the concept that it measures is differently than a typical vu meter. Here?s an example where I split a stereo input and have two side by side momentary ebur128 meters: ffplay -f lavfi "amovie=stereo.mp3,channelsplit[a][b];[a]ebur128=video=1:meter=18[a1][out0];[b]ebur128=video=1:meter=18[b1][out1];[a1]crop=20:432:612:40[a2];[b1]crop=20:432:612:40[b2];[a2][b2]framepack[out2]? From here you could use drawbox and/or drawtext to add labelling. Dave Rice From rogat1y at gmail.com Fri Nov 7 16:54:08 2014 From: rogat1y at gmail.com (Maxim Kozlov) Date: Fri, 7 Nov 2014 18:54:08 +0300 Subject: [FFmpeg-user] -report option and log level Message-ID: Hi all. How i can change the log level of report file? Setting environment variable FFREPORT=level=info throws error. C:\src\ffmpeg-32\bin>set FFREPORT=level=info C:\src\ffmpeg-32\bin>ffmpeg2.exe -threads 0 -rtbufsize 2G -fflags +genpts -report -re -i rtmp://host/app/stream -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -s 640x360 -vb 700k -x264opts level=31:keyint=25:fps=25:ref=3:vbv_maxrate=700:vbv_bufsize=700 -preset:v fast -aspect 16:9 -acodec libvo_aacenc -ab 32k -ar 22050 -ac 2 -bsf:a aac_adtstoasc -map 0:v -map 0:a -flags:a +global_header -flags:v +cgop+global_header -f tee "[f=flv]rtmp://host1/app/stream|[f=flv]rtmp://host2/app/stream" Invalid report file level From kuban.altan at sinefekt.com Fri Nov 7 17:19:40 2014 From: kuban.altan at sinefekt.com (Kuban Altan) Date: Fri, 7 Nov 2014 18:19:40 +0200 Subject: [FFmpeg-user] processing a vumeter (levelmeter) as video for audio inputs In-Reply-To: <017D58C6-9D15-4F06-AE72-7D09E0D7DDA3@dericed.com> References: <017D58C6-9D15-4F06-AE72-7D09E0D7DDA3@dericed.com> Message-ID: Thanks Dave, But there is a minor issue with ebur128 Momentary level. It is defined for a 0.4 second time range. Which is very long compared to 0.04 second for each frame in PAL standard. Do you think that there would be a possible workaround to hijack ebur128 Momentary (0.4s) vumeter, to use 0.04 second? kuban altan On Fri, Nov 7, 2014 at 5:21 PM, Dave Rice wrote: > Hi, > > > On Nov 7, 2014, at 8:27 AM, Kuban Altan wrote: > > > > Hi, > > I am testing *ebur128* filter, which can produce a video output. But I > need > > a simple *vumeter* display per audio channel, which does not seem to > exist. > > Any idea to accomplish this? > > > > So far, I have only been able to *volumedetect* per frame, mentioning a > -ss > > time and a duration with -t 0.04 (for 25fps)... So I can get a peak sound > > level as text output using volumedetect. But this is not very efficient. > > All I need is a video and 2 rectangular bars, which have their lengths > > connected to input audio level. > > There is a momentarily ebur128 level in the ebur128 video output so you > could crop that out and reuse it noting that the concept that it measures > is differently than a typical vu meter. Here?s an example where I split a > stereo input and have two side by side momentary ebur128 meters: > > ffplay -f lavfi > "amovie=stereo.mp3,channelsplit[a][b];[a]ebur128=video=1:meter=18[a1][out0];[b]ebur128=video=1:meter=18[b1][out1];[a1]crop=20:432:612:40[a2];[b1]crop=20:432:612:40[b2];[a2][b2]framepack[out2]? > > From here you could use drawbox and/or drawtext to add labelling. > > Dave Rice > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rogat1y at gmail.com Fri Nov 7 17:20:22 2014 From: rogat1y at gmail.com (Maxim Kozlov) Date: Fri, 7 Nov 2014 19:20:22 +0300 Subject: [FFmpeg-user] -report option and log level In-Reply-To: References: Message-ID: from this post http://ffmpeg-users.933282.n4.nabble.com/FW-ffmpeg-report-option-syntax-error-tp4665954p4665955.html Note that the "level" option is numerical, from what I could make out from the source code. The internally used values are defined as AV_LOG_* in libavutil/log.h. AV_LOG_QUIET -8 AV_LOG_PANIC 0 AV_LOG_FATAL 8 AV_LOG_ERROR 16 AV_LOG_WARNING 24 AV_LOG_INFO 32 AV_LOG_VERBOSE 40 AV_LOG_DEBUG 48 2014-11-07 18:54 GMT+03:00 Maxim Kozlov : > Hi all. > > How i can change the log level of report file? > > Setting environment variable FFREPORT=level=info throws error. > > C:\src\ffmpeg-32\bin>set FFREPORT=level=info > C:\src\ffmpeg-32\bin>ffmpeg2.exe -threads 0 -rtbufsize 2G -fflags +genpts > -report -re -i rtmp://host/app/stream -vcodec libx264 > -pix_fmt yuv420p -profile:v baseline -s 640x360 -vb 700k -x264opts > level=31:keyint=25:fps=25:ref=3:vbv_maxrate=700:vbv_bufsize=700 -preset:v > fast -aspect 16:9 -acodec libvo_aacenc -ab 32k -ar 22050 -ac 2 -bsf:a > aac_adtstoasc -map 0:v -map 0:a -flags:a +global_header -flags:v > +cgop+global_header -f tee > "[f=flv]rtmp://host1/app/stream|[f=flv]rtmp://host2/app/stream" > Invalid report file level > > From adf.lists at gmail.com Sun Nov 9 14:01:07 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 13:01:07 +0000 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , Message-ID: <545F6593.7010000@gmail.com> Carl Eugen Hoyos wrote: > I severely doubt that it was formalized for DVB. > You can check the bitrate to be sure: I would expect > that 256k are not enough for Multichannel. http://www.etsi.org/deliver/etsi_ts/101100_101199/101154/01.11.01_60/ts_101154v011101p.pdf Annex C.4.2.4 From cehoyos at ag.or.at Sun Nov 9 14:15:58 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 13:15:58 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , <545F6593.7010000@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > Carl Eugen Hoyos wrote: This is missing the following: > > > Don't worry about MPEG-2 Multichannel, your sample > > > is certainly Dolby PLII. > > I severely doubt that it was formalized for DVB. (it == MPEG-2 Multichannel) > > You can check the bitrate to be sure: I would expect > > that 256k are not enough for Multichannel. > > http://www.etsi.org/deliver/etsi_ts/101100_101199/101154/01.11.01_60/ts_101154v011101p.pdf Yes, MPEG-2 Multichannel was probably never formalized for DVB, Dolby PLII certainly was. And it is apparently possibly to detect it;-) (I did not yet test if the flag is really set correctly.) Carl Eugen From adf.lists at gmail.com Sun Nov 9 14:53:00 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 13:53:00 +0000 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , <545F6593.7010000@gmail.com> Message-ID: <545F71BC.5000005@gmail.com> Carl Eugen Hoyos wrote: > Andy Furniss gmail.com> writes: > >> Carl Eugen Hoyos wrote: > > This is missing the following: > >>>> Don't worry about MPEG-2 Multichannel, your sample >>>> is certainly Dolby PLII. > >>> I severely doubt that it was formalized for DVB. > > (it == MPEG-2 Multichannel) Oops, sorry, repeats to self mp2 != mpeg2, mp2 != mpeg2 ... From adf.lists at gmail.com Sun Nov 9 15:15:19 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 14:15:19 +0000 Subject: [FFmpeg-user] Is latm decode with fdkaac possible? Message-ID: <545F76F7.5000106@gmail.com> I am asking just in case feeding latm encapsulated aac to fdkaac is in someway possible - in which case I need to try harder. Anyone can say yes or no? Thanks. From cehoyos at ag.or.at Sun Nov 9 15:52:36 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 14:52:36 +0000 (UTC) Subject: [FFmpeg-user] Is latm decode with fdkaac possible? References: <545F76F7.5000106@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > I am asking just in case feeding latm > encapsulated aac to fdkaac is in someway > possible Yes, it is (afair). Why? I believe you know what I am usually (always, in every single email) asking here: No, I don't do it to make user's lifes as hard as possible but because it is needed to provide a useful answer. Carl Eugen (Command line and complete, uncut console output missing.) From cehoyos at ag.or.at Sun Nov 9 15:53:10 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 14:53:10 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , <545F6593.7010000@gmail.com> <545F71BC.5000005@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > >>> I severely doubt that it was formalized for DVB. > > > > (it == MPEG-2 Multichannel) > > Oops, sorry, repeats to self mp2 != mpeg2, mp2 != mpeg2 ... Don't worry, the link you provided looks useful. Carl Eugen From cehoyos at ag.or.at Sun Nov 9 17:52:02 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 16:52:02 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , <545F6593.7010000@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > http://www.etsi.org/deliver/etsi_ts/101100_101199/101154/01.11.01_60/ts_101154v011101p.pdf Several developers agree on irc that the extended ancillary data syntax is typically unused and cannot be used to detect PLII encoding. Carl Eugen From adf.lists at gmail.com Sun Nov 9 18:37:20 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 17:37:20 +0000 Subject: [FFmpeg-user] Is latm decode with fdkaac possible? In-Reply-To: References: <545F76F7.5000106@gmail.com> Message-ID: <545FA650.9080208@gmail.com> Carl Eugen Hoyos wrote: > Andy Furniss gmail.com> writes: > >> I am asking just in case feeding latm >> encapsulated aac to fdkaac is in someway >> possible > > Yes, it is (afair). > > Why? > > I believe you know what I am usually (always, in > every single email) asking here: No, I don't do > it to make user's lifes as hard as possible but > because it is needed to provide a useful answer. > > Carl Eugen > > (Command line and complete, uncut console output > missing.) :-) yea though if the answer was just no it would have saved extended experements. I haven't tried loads of variants yet but for now - first without fdk - file plays ok with players also. ffmpeg -i Bt5th.latm ffmpeg version N-67488-g4e17943 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 9 2014 17:23:50 with gcc 4.8.3 (GCC) configuration: --prefix=/usr --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-x11grab libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 12.100 / 56. 12.100 libavformat 56. 12.103 / 56. 12.103 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, loas, from 'Bt5th.latm': Duration: N/A, bitrate: N/A Stream #0:0: Audio: aac_latm (LC), 48000 Hz, stereo, fltp ffmpeg -c:a libfdk_aac -i Bt5th.latm -f null - ffmpeg version N-67488-g4e17943 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 9 2014 17:23:50 with gcc 4.8.3 (GCC) configuration: --prefix=/usr --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-x11grab libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 12.100 / 56. 12.100 libavformat 56. 12.103 / 56. 12.103 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 4002 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 1001 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 400a [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 1001 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 4002 snip more of the same [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 4002 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 4004 [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 4002 Last message repeated 1 times [libfdk_aac @ 0x1d86b80] aacDecoder_DecodeFrame() failed: 1001 [loas @ 0x1d86220] decoding for stream 0 failed [loas @ 0x1d86220] Estimating duration from bitrate, this may be inaccurate [loas @ 0x1d86220] Could not find codec parameters for stream 0 (Audio: aac, 0 channels, s16, 147 kb/s): unspecified sample rate Consider increasing the value for the 'analyzeduration' and 'probesize' options Bt5th.latm: could not find codec parameters Input #0, loas, from 'Bt5th.latm': Duration: 00:30:12.24, bitrate: 147 kb/s Stream #0:0: Audio: aac, 0 channels, s16, 147 kb/s Output #0, null, to 'pipe:': Output file #0 does not contain any stream From cehoyos at ag.or.at Sun Nov 9 18:49:59 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 17:49:59 +0000 (UTC) Subject: [FFmpeg-user] Is latm decode with fdkaac possible? References: <545F76F7.5000106@gmail.com> <545FA650.9080208@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > ffmpeg -c:a libfdk_aac -i Bt5th.latm -f null - > [libfdk_aac 0x1d86b80] aacDecoder_DecodeFrame() failed: 4002 Sorry, I mixed something up: The fdk encoder supports latm encoding (which doesn't work with any other of the four aac encoders) while the fdk decoder cannot support latm: There should be parser / demuxer that changes latm into aac but when latm decoding was implemented in FFmpeg, it was decided to do it in the native aac decoder which is not was libfdk does / expects. Sorry, Carl Eugen From adf.lists at gmail.com Sun Nov 9 21:08:44 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 20:08:44 +0000 Subject: [FFmpeg-user] Is latm decode with fdkaac possible? In-Reply-To: References: <545F76F7.5000106@gmail.com> <545FA650.9080208@gmail.com> Message-ID: <545FC9CC.9020106@gmail.com> Carl Eugen Hoyos wrote: > Andy Furniss gmail.com> writes: > >> ffmpeg -c:a libfdk_aac -i Bt5th.latm -f null - > >> [libfdk_aac 0x1d86b80] aacDecoder_DecodeFrame() failed: >> 4002 > > Sorry, I mixed something up: The fdk encoder supports latm encoding > (which doesn't work with any other of the four aac encoders) while > the fdk decoder cannot support latm: There should be parser / demuxer > that changes latm into aac but when latm decoding was implemented in > FFmpeg, it was decided to do it in the native aac decoder which is > not was libfdk does / expects. > > Sorry, Carl Eugen Ahh, Ok - thanks for the info. Would be a handy feature. FWIW wanting to use fdk is related to the etsi doc I posted in the other thread. I was going to try and add some debugging to see what, is any, ancillary was broadcast in the UK (like heavy compression and mixdown). I know from adding debugging to ffmpeg aacdec that drc is broadcast, but couldn't find mix down meta. From adf.lists at gmail.com Sun Nov 9 21:19:19 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 20:19:19 +0000 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , <545F6593.7010000@gmail.com> Message-ID: <545FCC47.1050100@gmail.com> Carl Eugen Hoyos wrote: > Andy Furniss gmail.com> writes: > >> > http://www.etsi.org/deliver/etsi_ts/101100_101199/101154/01.11.01_60/ts_101154v011101p.pdf > > Several developers agree on irc that the extended ancillary data > syntax is typically unused and cannot be used to detect PLII > encoding. Fair enough - I was suprised that anyone actually broadcast prologic. It would be interesting to know if the sample was flagged. Maybe MythTV just lucked into passing it out as spdif - I don't have any experience of myth. From cehoyos at ag.or.at Sun Nov 9 21:21:31 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 20:21:31 +0000 (UTC) Subject: [FFmpeg-user] Is latm decode with fdkaac possible? References: <545F76F7.5000106@gmail.com> <545FA650.9080208@gmail.com> <545FC9CC.9020106@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > Ahh, Ok - thanks for the info. > > Would be a handy feature. Do you have a specific problem with the internal decoder? > FWIW wanting to use fdk is related to the etsi > doc I posted in the other thread. The thread was not aac-related and people seem to agree that the dvb feature you pointed at is not actually used. > I was going to try and add some debugging to see > what, is any, ancillary was broadcast in the UK > (like heavy compression and mixdown). The info you pointed to is related to mp2, not aac. Carl Eugen From cehoyos at ag.or.at Sun Nov 9 21:24:57 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 9 Nov 2014 20:24:57 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: , , , <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> Message-ID: Andy Furniss gmail.com> writes: > > Several developers agree on irc that the extended > > ancillary data syntax is typically unused and > > cannot be used to detect PLII encoding. > > Fair enough - I was suprised that anyone actually > broadcast prologic. Where I live all movies are broadcast PLII. > It would be interesting to know if the sample was > flagged. As said, it doesn't help to check the flag because it is mostly unused. (Afaik) > Maybe MythTV just lucked into passing it out as spdif As said MythTV contains a PLII decoder (filter). Carl Eugen From bluesky at caramail.com Sun Nov 9 21:35:34 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Sun, 9 Nov 2014 21:35:34 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <545FCC47.1050100@gmail.com> References: =?UTF-8?Q?, =20=09, =20=09, =20=09=09=09<545F6593.7010000@gmail.com>=20, =20<545FCC47.1050100@gmail.com>?= Message-ID: "Andy Furniss" wrote: > Fair enough - I was suprised that anyone actually broadcast prologic. "Broadcast" is a relative term here. It is very common for use in free-to-air C-band and Ku-band satellite transmissions in the United States. That may or may not be considered "broadcast", depending on how you define the term. From adf.lists at gmail.com Sun Nov 9 21:40:41 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 09 Nov 2014 20:40:41 +0000 Subject: [FFmpeg-user] Is latm decode with fdkaac possible? In-Reply-To: References: <545F76F7.5000106@gmail.com> <545FA650.9080208@gmail.com> <545FC9CC.9020106@gmail.com> Message-ID: <545FD149.5090103@gmail.com> Carl Eugen Hoyos wrote: > Andy Furniss gmail.com> writes: > >> Ahh, Ok - thanks for the info. >> >> Would be a handy feature. > > Do you have a specific problem with the internal decoder? It doesn't do DRC so if I want to use my big old stereo speakers (my TV speakers are crap) to listen to a recorded film it would vary between being too quiet and too loud. As I guess you know the default for 5.1 should be full drc applied and mixdown to 2.0 makes it even more essential as (I believe) the act of mixing down increases dynamic range. >> FWIW wanting to use fdk is related to the etsi doc I posted in the >> other thread. > > The thread was not aac-related and people seem to agree that the dvb > feature you pointed at is not actually used. It's the (unparsed by ffmpeg) ancillary data aspect not the specific codec. > >> I was going to try and add some debugging to see what, is any, >> ancillary was broadcast in the UK (like heavy compression and >> mixdown). > > The info you pointed to is related to mp2, not aac. Yes I know, but right next to it is about aac in fact it was searching for aac stuff that found me that doc. From bluesky at caramail.com Sun Nov 9 21:42:53 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Sun, 9 Nov 2014 21:42:53 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? Message-ID: "Carl Eugen Hoyos" wrote: > As said MythTV contains a PLII decoder (filter). Any possibility that ffmpeg could include something like that, please? From bluesky at caramail.com Sun Nov 9 21:38:42 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Sun, 9 Nov 2014 21:38:42 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: =?UTF-8?Q?, =20=09, =20=09, =20=09=09=09<545F6593.7010000@gmail.com>=20=20<545FCC47.1050100@gmail.com>, =20?= Message-ID: "Carl Eugen Hoyos" wrote: > As said MythTV contains a PLII decoder (filter). Any possibility that ffmpeg could include something like that, please? From bluesky at caramail.com Sun Nov 9 21:47:33 2014 From: bluesky at caramail.com (bluesky at caramail.com) Date: Sun, 9 Nov 2014 21:47:33 +0100 Subject: [FFmpeg-user] Is latm decode with fdkaac possible? In-Reply-To: <545FD149.5090103@gmail.com> References: =?UTF-8?Q?<545F76F7.5000106@gmail.com>=09=09<545FA650.9080208@gmail.com>=09=09<545FC9CC.9020106@gmail.com>=20, =20<545FD149.5090103@gmail.com>?= Message-ID: My apologies to the list for the duplicate post, I'm not exactly sure why that happened. From cehoyos at ag.or.at Mon Nov 10 10:38:40 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 10 Nov 2014 09:38:40 +0000 (UTC) Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS References: Message-ID: C E Macfarlane macfh.co.uk> writes: > ./libavutil/libm.h:88: error: static declaration of > 'fminf' follows non-static declaration The code that was responsible for this compilation problem was removed from the FFmpeg source. Note that afaik, the error indicates that libm.h and libm (the header and the library) have different opinions on what symbols they provide. Carl Eugen From michael.heuberger at binarykitchen.com Mon Nov 10 11:43:24 2014 From: michael.heuberger at binarykitchen.com (Michael Heuberger) Date: Mon, 10 Nov 2014 23:43:24 +1300 Subject: [FFmpeg-user] Confusion about pix_fmt Message-ID: <546096CC.5090304@binarykitchen.com> Hello guys If you look at the bug report at https://trac.ffmpeg.org/ticket/4098 you can see that I am confused about the pix_fmt option. I am not sure what parameter to use for this option. All I want is to make the video compatible on *modern* browsers/players only and I also want to make the warning disappear ffmpeg is generating: deprecated pixel format used, make sure you did set range correctly This warning led me to believe I have to set a different pix_fmt parameter than the default one (yuv420p). But I do not know which one. Any clues? Michael -- Binary Kitchen Michael Heuberger 4c Dunbar Road Mt Eden Auckland 1024 (New Zealand) Mobile (text only) ... +64 21 261 89 81 Email ................ michael at binarykitchen.com Website .............. http://www.binarykitchen.com From james.darnley at gmail.com Mon Nov 10 12:33:37 2014 From: james.darnley at gmail.com (James Darnley) Date: Mon, 10 Nov 2014 12:33:37 +0100 Subject: [FFmpeg-user] Confusion about pix_fmt In-Reply-To: <546096CC.5090304@binarykitchen.com> References: <546096CC.5090304@binarykitchen.com> Message-ID: <5460A291.9090607@gmail.com> On 2014-11-10 11:43, Michael Heuberger wrote: > Hello guys > > If you look at the bug report at https://trac.ffmpeg.org/ticket/4098 you > can see that I am confused about the pix_fmt option. > > I am not sure what parameter to use for this option. All I want is to > make the video compatible on *modern* browsers/players only and I also > want to make the warning disappear ffmpeg is generating: > > deprecated pixel format used, make sure you did set range correctly > > This warning led me to believe I have to set a different pix_fmt > parameter than the default one (yuv420p). But I do not know which one. > > Any clues? > > Michael > yuvj420p and yuv420p are the same. If a video player can play one it can play the other. The "j" refers to the range of values that are output. "j" is full range (or PC range) meaning it uses all 0-255 values rather than the more common 16-235 range (which is known as TV or Studio range). The problem comes when converting to RGB on playback. All good players recognize a flag in the file indicating that it is full range and convert it properly. All bad players do not. They assume it is TV range and expand the range of values causing too high contrast which is noticeable. Another similar symptom is players that do not expand the range when they should leading to too low contrast. I do not know whether browsers fall into the good category. If you want a correct picture on every possible system use yuv420p. If you want to know more about the colour space and this "j" option I suggest you read these: https://en.wikipedia.org/wiki/YCbCr https://en.wikipedia.org/wiki/YCbCr#JPEG_conversion -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From cehoyos at ag.or.at Mon Nov 10 15:37:47 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 10 Nov 2014 14:37:47 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Confusion_about_pix=5Ffmt?= References: <546096CC.5090304@binarykitchen.com> Message-ID: Michael Heuberger binarykitchen.com> writes: > All I want is to make the video compatible on *modern* > browsers/players Then you have to use "-pix_fmt yuv420p" which is different from "yuvj420p" and I believe "good players" are rare. > I also want to make the warning disappear ffmpeg is > generating: As said, this is not possible: The warning appears if either the input or the output pix_fmt are yuvj*- Carl Eugen From paulo.fidalgo.pt at gmail.com Mon Nov 10 20:27:13 2014 From: paulo.fidalgo.pt at gmail.com (Paulo Fidalgo) Date: Mon, 10 Nov 2014 19:27:13 +0000 Subject: [FFmpeg-user] MP3 enconding bitrate In-Reply-To: References: <5447FD04.80602@gmail.com> <544E9EC5.9070706@gmail.com> <201410281155.40532.cehoyos@ag.or.at> <5451F142.9070903@gmail.com> <20141030090726.GB12578@sunshine.barsnick.net> <545B4CBB.2050409@gmail.com> Message-ID: <54611191.4080800@gmail.com> On 06/11/14 18:12, Carl Eugen Hoyos wrote: > Paulo Fidalgo gmail.com> writes: > >> mp3check -ve 2L38_01_96kHz-ffmpeg-256k.mp3 >> 2L38_01_96kHz-ffmpeg-256k.mp3: >> 813 bytes of junk before first frame header >> >> but with lame there's no errors. > Please confirm that you tested with "-write_xing 0" > and your results. This have solved the problem, although since In my system I have an old version where this flag is not available I haven't tested. Now I need to go to the Amarok and see what's can be done to track down this problem. > > Please understand that I should have realised this > from the beginning but since you insisted on FFmpeg > writing frames larger 320k I concentrated on that. Don't worry, I just need to thank you patience and for trying to help me. Best regards, Paulo Fidalgo > > Sorry, Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From michael.heuberger at binarykitchen.com Tue Nov 11 10:03:02 2014 From: michael.heuberger at binarykitchen.com (Michael Heuberger) Date: Tue, 11 Nov 2014 22:03:02 +1300 Subject: [FFmpeg-user] Confusion about pix_fmt In-Reply-To: References: <546096CC.5090304@binarykitchen.com> Message-ID: <5461D0C6.3040608@binarykitchen.com> Thank you guys! - Learning here ... On 11/11/14 03:37, Carl Eugen Hoyos wrote: > Michael Heuberger binarykitchen.com> writes: > >> All I want is to make the video compatible on *modern* >> browsers/players > Then you have to use "-pix_fmt yuv420p" which is > different from "yuvj420p" and I believe "good players" > are rare. > >> I also want to make the warning disappear ffmpeg is >> generating: > As said, this is not possible: The warning appears if > either the input or the output pix_fmt are yuvj*- > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Binary Kitchen Michael Heuberger 4c Dunbar Road Mt Eden Auckland 1024 (New Zealand) Mobile (text only) ... +64 21 261 89 81 Email ................ michael at binarykitchen.com Website .............. http://www.binarykitchen.com From adf.lists at gmail.com Tue Nov 11 13:29:09 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Tue, 11 Nov 2014 12:29:09 +0000 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: References: , , , <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> Message-ID: <54620115.2080205@gmail.com> Carl Eugen Hoyos wrote: > Where I live all movies are broadcast PLII. Ahh, I didn't realise it was still so popular. > As said MythTV contains a PLII decoder (filter). So it does, I couldn't find it when I first looked. Doesn't seem very much compared to the quite complicated looking descriptions of what a "real" decoder supposedly does. float lt = *samples++; float rt = *samples++; bufs->l[ic] = lt; bufs->lfe[ic] = bufs->c[ic] = (lt+rt) * m3db; bufs->r[ic] = rt; // surround channels receive out-of-phase bufs->ls[ic] = (rt-lt) * 0.5; bufs->rs[ic] = (lt-rt) * 0.5; I don't pretend to know what should happen - but it's obvious that that leaves the S in the fronts - but then you said your receiver did too. I wonder if all receivers would. From barsnick at gmx.net Tue Nov 11 14:02:28 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 11 Nov 2014 14:02:28 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <54620115.2080205@gmail.com> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> Message-ID: <20141111130228.GC5455@sunshine.barsnick.net> On Tue, Nov 11, 2014 at 12:29:09 +0000, Andy Furniss wrote: > Carl Eugen Hoyos wrote: > > Where I live all movies are broadcast PLII. > Ahh, I didn't realise it was still so popular. Where I live - Germany - the "minor" stations try to use less bandwidth on the satellite transponders / in their bouquets, and transmit only one AC-3 audio stream with two channels. Nevertheless, I get quite proper surround sound from my sound system during many movies. So my conclusion is that they use the PLII "down-mix". (Unless that statement was a misunderstanding and we're talking about something totally different.) > > As said MythTV contains a PLII decoder (filter). > Doesn't seem very much compared to the quite complicated looking > descriptions of what a "real" decoder supposedly does. > > float lt = *samples++; > float rt = *samples++; > bufs->l[ic] = lt; > bufs->lfe[ic] = bufs->c[ic] = (lt+rt) * m3db; > bufs->r[ic] = rt; > // surround channels receive out-of-phase > bufs->ls[ic] = (rt-lt) * 0.5; > bufs->rs[ic] = (lt-rt) * 0.5; That looks different from what we were looking at earlier in the thread (and far too simple). Carl Eugen had pointed out this file: https://github.com/MythTV/mythtv/blob/master/mythtv/libs/libmythfreesurround/el_processor.cpp and its main worker function is block_decode(). It's far less simple than your example, and sort of corresponds to my understanding of what a decoder needs to do. You can't just add and subtract left and right channels. Since you have to work with phases, you need to do FT when operating in the digital domain. (From what I still know from my studies in electrical engineering.) And that's what this function does. Nevertheless, there's an implemention there! And even open sourced. > I don't pretend to know what should happen Disclaimer: I do honestly mostly pretend to know, but I really don't know much. ;-) Cheers, Moritz From brown at mrvideo.vidiot.com Tue Nov 11 14:11:33 2014 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Tue, 11 Nov 2014 07:11:33 -0600 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <20141111130228.GC5455@sunshine.barsnick.net> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> Message-ID: <20141111131133.GP4678@mrvideo.vidiot.com> On Tue, Nov 11, 2014 at 02:02:28PM +0100, Moritz Barsnick wrote: > Where I live - Germany - the "minor" stations try to use less bandwidth > on the satellite transponders / in their bouquets, and transmit only > one AC-3 audio stream with two channels. Nevertheless, I get quite > proper surround sound from my sound system during many movies. So my > conclusion is that they use the PLII "down-mix". That doesn't make any sense. AC3 5.1 is usually done at 384 kbps, while AC3 2.0 is 192 kbps. The 192 kbps difference is a drop in the bucket compared to all of the other packets that make up the transponder mux. Weird. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email "What do you say Beckett. Wanna have a baby?" - Castle to Det. Beckett "How long have I been gone?" Alexis after seeing Castle and Beckett w/ baby - Castle - 11/25/13 From barsnick at gmx.net Tue Nov 11 14:28:51 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 11 Nov 2014 14:28:51 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <20141111131133.GP4678@mrvideo.vidiot.com> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> <20141111131133.GP4678@mrvideo.vidiot.com> Message-ID: <20141111132851.GD5455@sunshine.barsnick.net> On Tue, Nov 11, 2014 at 07:11:33 -0600, Mike Brown wrote: > That doesn't make any sense. AC3 5.1 is usually done at 384 kbps, while > AC3 2.0 is 192 kbps. The 192 kbps difference is a drop in the bucket > compared to all of the other packets that make up the transponder mux. I think you're right - I was (stupidly) trying to make sense of it. All I do know, e.g.: The "major" channel "RTL" uses various audio stream types in its programming on the SD channel, depending on the type of show and the material. The minor channels, such as "RTL2" only ever have one MP2 audio and one AC-3 audio stream, with two channels each. For whatever reason. (We might need to use other examples than these two channels, but I'm speaking or "relatives" in a family of channels.) BTW, the major (SD) channels also get much more overall bandwidth than the minor ones, resulting in very sharp video quality (or rather quite sucky video quality on the latter channels). I don't know exact numbers, but I believe there may be more than a 2x factor. Audio is probably still said drop in the ocean. Moritz From brown at mrvideo.vidiot.com Tue Nov 11 14:45:40 2014 From: brown at mrvideo.vidiot.com (Mike Brown) Date: Tue, 11 Nov 2014 07:45:40 -0600 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <20141111132851.GD5455@sunshine.barsnick.net> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> <20141111131133.GP4678@mrvideo.vidiot.com> <20141111132851.GD5455@sunshine.barsnick.net> Message-ID: <20141111134540.GQ4678@mrvideo.vidiot.com> On Tue, Nov 11, 2014 at 02:28:51PM +0100, Moritz Barsnick wrote: > BTW, the major (SD) channels also get much more overall bandwidth than > the minor ones, resulting in very sharp video quality (or rather quite > sucky video quality on the latter channels). I don't know exact > numbers, but I believe there may be more than a 2x factor. Audio is > probably still said drop in the ocean. Ah, this piece of info was missing before and that is the video is SD, not HD. It isn't weird to have only AC3 2.0 for SD channels. I was thinking in the world of HD. Silly me. MB -- e-mail: vidiot at vidiot.com | vidiot at vidiot.net /~\ The ASCII 6082066843 at email.uscc.net (140 char limit) \ / Ribbon Campaign Visit - URL: http://vidiot.com/ X Against http://vidiot.net/ / \ HTML Email "What do you say Beckett. Wanna have a baby?" - Castle to Det. Beckett "How long have I been gone?" Alexis after seeing Castle and Beckett w/ baby - Castle - 11/25/13 From adf.lists at gmail.com Tue Nov 11 14:46:26 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Tue, 11 Nov 2014 13:46:26 +0000 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <20141111130228.GC5455@sunshine.barsnick.net> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> Message-ID: <54621332.9000307@gmail.com> Moritz Barsnick wrote: > On Tue, Nov 11, 2014 at 12:29:09 +0000, Andy Furniss wrote: >> Carl Eugen Hoyos wrote: >>> Where I live all movies are broadcast PLII. >> Ahh, I didn't realise it was still so popular. > > Where I live - Germany - the "minor" stations try to use less bandwidth > on the satellite transponders / in their bouquets, and transmit only > one AC-3 audio stream with two channels. Nevertheless, I get quite > proper surround sound from my sound system during many movies. So my > conclusion is that they use the PLII "down-mix". > > (Unless that statement was a misunderstanding and we're talking about > something totally different.) > >>> As said MythTV contains a PLII decoder (filter). >> Doesn't seem very much compared to the quite complicated looking >> descriptions of what a "real" decoder supposedly does. >> >> float lt = *samples++; >> float rt = *samples++; >> bufs->l[ic] = lt; >> bufs->lfe[ic] = bufs->c[ic] = (lt+rt) * m3db; >> bufs->r[ic] = rt; >> // surround channels receive out-of-phase >> bufs->ls[ic] = (rt-lt) * 0.5; >> bufs->rs[ic] = (lt-rt) * 0.5; > > That looks different from what we were looking at earlier in the thread > (and far too simple). Carl Eugen had pointed out this file: > > https://github.com/MythTV/mythtv/blob/master/mythtv/libs/libmythfreesurround/el_processor.cpp Agh, thanks for that - the code I posted is from myth, but obviously I don't know if it's ever used. For some reason this thread has split into two for me and I didn't see the other bit. From barsnick at gmx.net Tue Nov 11 16:54:49 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 11 Nov 2014 16:54:49 +0100 Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? In-Reply-To: <54621332.9000307@gmail.com> References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> <54621332.9000307@gmail.com> Message-ID: <20141111155449.GB8708@sunshine.barsnick.net> On Tue, Nov 11, 2014 at 13:46:26 +0000, Andy Furniss wrote: > For some reason this thread has split into two for me and I didn't see > the other bit. You can see the thread structure and that other "branch" about here: http://thread.gmane.org/gmane.comp.video.ffmpeg.user/54545/focus=54575 [Selfishly poining to own post. ;-)] Moritz From cehoyos at ag.or.at Tue Nov 11 17:57:22 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 11 Nov 2014 16:57:22 +0000 (UTC) Subject: [FFmpeg-user] How can I convert mpeg2 5.1 audio to ac3? References: <545F6593.7010000@gmail.com> <545FCC47.1050100@gmail.com> <54620115.2080205@gmail.com> <20141111130228.GC5455@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > So my conclusion is that they use the PLII "down-mix". It is certainly correct that many German-speaking TV channels use PLII down-mixing. Maybe some of you want to add information that you find in the MythTV source to ticket #4085 to allow a LGPL implementation of the feature? Carl Eugen From c.e.macfarlane at macfh.co.uk Tue Nov 11 22:19:35 2014 From: c.e.macfarlane at macfh.co.uk (C E Macfarlane) Date: Tue, 11 Nov 2014 21:19:35 -0000 Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS In-Reply-To: Message-ID: JFTR, I've finally managed to compile ffmpeg for MIPSEL UCLibc using a third method, the second using cross-compilation, and the results were identical to the previous cross-compilation. This indicates to me that the problem lies in the ffmpeg source code and/or configuration itself. Accordingly, I note and welcome that ... [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos > > The code that was responsible for this compilation > problem was removed from the FFmpeg source. Would that be ffmpeg-2.4.3.tar.bz2? From cehoyos at ag.or.at Tue Nov 11 22:23:38 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 11 Nov 2014 21:23:38 +0000 (UTC) Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS References: Message-ID: C E Macfarlane macfh.co.uk> writes: > I've finally managed to compile ffmpeg for MIPSEL UCLibc > using a third method, the second using cross-compilation, > and the results were identical to the previous > cross-compilation. With identical, do you mean that you still get an error concerning fminf()? In this case, you probably tested an old version. Note that afaict, a system header (libm.h) is broken and you apparently cannot fix it using another library. > > The code that was responsible for this compilation > > problem was removed from the FFmpeg source. > > Would that be ffmpeg-2.4.3.tar.bz2? No. Carl Eugen From c.e.macfarlane at macfh.co.uk Tue Nov 11 22:34:56 2014 From: c.e.macfarlane at macfh.co.uk (C E Macfarlane) Date: Tue, 11 Nov 2014 21:34:56 -0000 Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS In-Reply-To: Message-ID: > With identical, do you mean that you still > get an error concerning fminf()? > In this case, you probably tested an old > version. Yes: ffmpeg-2.4.2$ export CCPREFIX="/home/Embedded/crosstool/mipsel-unknown-linux-uclibc/bin/mipsel-un known-linux-uclibc-" ffmpeg-2.4.2$ ./configure --enable-cross-compile --cross-prefix=${CCPREFIX} --arch=mipsel --target-os=linux --prefix=/opt/share ... ffmpeg-2.4.2$ make CC libavdevice/alldevices.o In file included from ./libavutil/internal.h:168, from ./libavutil/common.h:415, from ./libavutil/avutil.h:289, from ./libavutil/log.h:25, from libavdevice/avdevice.h:46, from libavdevice/alldevices.c:22: ./libavutil/libm.h: > Note that afaict, a system header (libm.h) is broken > and you apparently cannot fix it using another library. > > > > The code that was responsible for this compilation > > > problem was removed from the FFmpeg source. > > > > Would that be ffmpeg-2.4.3.tar.bz2? > > No. As I've just discovered. Where can I download a fixed version? From cehoyos at ag.or.at Tue Nov 11 22:49:10 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 11 Nov 2014 21:49:10 +0000 (UTC) Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS References: Message-ID: C E Macfarlane macfh.co.uk> writes: > Where can I download a fixed version? Please read http://ffmpeg.org/download.html and please us if you find it unclear. (I actually find it unclear but it still says how to download a current version.) Carl Eugen From c.e.macfarlane at macfh.co.uk Wed Nov 12 00:06:09 2014 From: c.e.macfarlane at macfh.co.uk (C E Macfarlane) Date: Tue, 11 Nov 2014 23:06:09 -0000 Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS In-Reply-To: Message-ID: Using same configuration command as above, didn't complete make: CC libavcodec/aacdec.o libavcodec/aacdec.c: In function 'decode_ics': libavcodec/aacdec.c:1987: internal compiler error: in reg_overlap_mentioned_p, at rtlanal.c:1398 Please submit a full bug report, with preprocessed source if appropriate. See for instructions. make: *** [libavcodec/aacdec.o] Error 1 www.macfh.co.uk/CEMH.html UK Residents: If you feel can possibly support it please sign the following ePetition before closing time of 30/03/2015 23:59:- http://epetitions.direct.gov.uk/petitions/71556 > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org > [mailto:ffmpeg-user-bounces at ffmpeg.org]On Behalf Of Carl Eugen Hoyos > > Please read http://ffmpeg.org/download.html and > please us if you find it unclear. > (I actually find it unclear but it still says > how to download a current version.) From f4lconx at gmail.com Wed Nov 12 01:23:06 2014 From: f4lconx at gmail.com (Brandon Lees) Date: Tue, 11 Nov 2014 19:23:06 -0500 Subject: [FFmpeg-user] http timeout not working? Message-ID: I'm using ffmpeg to capture video from an ip camera using: ffmpeg -y -f mjpeg -analyzeduration 0 -probesize 32 -use_wallclock_as_timestamps 1 -timeout 10 -i http://user at 1.2.3.4/video.cgi -vsync passthrough -vcodec libx264 -pix_fmt yuv420p -t 30:00 output.mp4 Everything is working fine, except if the camera is disconnected. For example if I unplug the camera, ffmpeg will just hang, it won't ever exit as far as I can tell. It seems like the timeout value should cause ffmpeg to stop if it hasn't received any data on the socket in this time period, but it doesn't seem to do anything. Am I setting it incorrectly? Thanks for any help, Brandon From Sravan.Kasarla at gtspt.com Tue Nov 11 19:27:30 2014 From: Sravan.Kasarla at gtspt.com (Sravan Kasarla) Date: Tue, 11 Nov 2014 18:27:30 +0000 Subject: [FFmpeg-user] FFMPEG RPM for centos Message-ID: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> Hi there, I am new ffmpeg user. I would like to get an RPM for ffmpeg and its dependencies for latest version. Where can I get those RPM's from? Thanks in advance. Thanks, Sravan. From paulo.fidalgo.pt at gmail.com Wed Nov 12 09:13:01 2014 From: paulo.fidalgo.pt at gmail.com (Paulo Fidalgo) Date: Wed, 12 Nov 2014 08:13:01 +0000 Subject: [FFmpeg-user] FFMPEG RPM for centos In-Reply-To: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> References: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> Message-ID: <5463168D.8090507@gmail.com> I suspect you won't find it, although you can use the static binaries provided by ffmpeg project: http://johnvansickle.com/ffmpeg/ Best regards, Paulo Fidalgo On 11/11/14 18:27, Sravan Kasarla wrote: > Hi there, > I am new ffmpeg user. I would like to get an RPM for ffmpeg and its dependencies for latest version. Where can I get those RPM's from? > > Thanks in advance. > > > > Thanks, > Sravan. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From cehoyos at ag.or.at Wed Nov 12 10:03:16 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 12 Nov 2014 09:03:16 +0000 (UTC) Subject: [FFmpeg-user] FFMPEG RPM for centos References: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> <5463168D.8090507@gmail.com> Message-ID: Paulo Fidalgo gmail.com> writes: > I suspect you won't find it, although you can use the > static binaries provided by ffmpeg project: > > http://johnvansickle.com/ffmpeg/ Just to avoid a misunderstanding: These binaries are not provided by the FFmpeg project. (We are very thankful to relaxed though.) The FFmpeg project does not provide any products. Carl Eugen From cehoyos at ag.or.at Wed Nov 12 10:05:32 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 12 Nov 2014 09:05:32 +0000 (UTC) Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS References: Message-ID: C E Macfarlane macfh.co.uk> writes: > CC libavcodec/aacdec.o > libavcodec/aacdec.c: In function 'decode_ics': > libavcodec/aacdec.c:1987: internal compiler error: in > reg_overlap_mentioned_p, at rtlanal.c:1398 > Please submit a full bug report, > with preprocessed source if appropriate. > See for instructions. > make: *** [libavcodec/aacdec.o] Error 1 Now please update your toolchain, test again and report back. (This is not a bug in FFmpeg but in an ancient broken compiler.) Please do not top-post here, Carl Eugen From cehoyos at ag.or.at Wed Nov 12 10:34:25 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 12 Nov 2014 09:34:25 +0000 (UTC) Subject: [FFmpeg-user] http timeout not working? References: Message-ID: Brandon Lees gmail.com> writes: > I'm using ffmpeg to capture video from an ip camera using: > > ffmpeg -y -f mjpeg -analyzeduration 0 -probesize 32 > -use_wallclock_as_timestamps 1 -timeout 10 -i http:// This malicious option takes ns so you probably want: -timeout 10000000 Carl Eugen From barsnick at gmx.net Wed Nov 12 11:51:43 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Wed, 12 Nov 2014 11:51:43 +0100 Subject: [FFmpeg-user] FFMPEG RPM for centos In-Reply-To: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> References: <853CDA7124D2C946A92FECD0E0E4AAFDD34D03B2@VFLON-XCHMBX1.virtuefusion.corp> Message-ID: <20141112105143.GB25150@sunshine.barsnick.net> Hi Sravan, On Tue, Nov 11, 2014 at 18:27:30 +0000, Sravan Kasarla wrote: > I am new ffmpeg user. I would like to get an RPM for ffmpeg and its > dependencies for latest version. Where can I get those RPM's from? You can get RPMs from RPMFusion: http://ubuntupop.blogspot.de/2012/10/how-to-add-rpm-fusion-repository-on.html But as I pointed out here: http://article.gmane.org/gmane.comp.video.ffmpeg.user/53385 those are very old versions. You are better off building your own. Moritz From f4lconx at gmail.com Wed Nov 12 16:30:20 2014 From: f4lconx at gmail.com (Brandon Lees) Date: Wed, 12 Nov 2014 10:30:20 -0500 Subject: [FFmpeg-user] http timeout not working? In-Reply-To: References: Message-ID: > > > I'm using ffmpeg to capture video from an ip camera using: > > > > ffmpeg -y -f mjpeg -analyzeduration 0 -probesize 32 > > -use_wallclock_as_timestamps 1 -timeout 10 -i http:// > > This malicious option takes ns so you probably want: > -timeout 10000000 > > Carl Eugen I tried your suggestion for various values, 10000, 10000000, etc but do not seem to get any different behavior. I forgot to mention earlier, if it makes any difference, I am running ffmpeg on windows with the binary build from ffmpeg.zeranoe.com. I can attach the console output if it would help, but even with debug output on, everything seems to be running normally then everything abruptly stops when the camera is disconnected. Any idea what might be going on? Thanks, Brandon From jogga at bitfield.se Wed Nov 12 19:53:04 2014 From: jogga at bitfield.se (=?utf-8?Q?Isaksson_J=C3=B6rgen?=) Date: Wed, 12 Nov 2014 19:53:04 +0100 Subject: [FFmpeg-user] Compile without VOB decryption Message-ID: Hi all, How can I compile FFmpeg without VOB decryption? Is there any way to detect if a VOB file is encrypted? Best regards / J?rgen -- J?rgen Isaksson Bitfield AB, http://www.bitfield.se twitter.com/jorgenisaksson From cehoyos at ag.or.at Wed Nov 12 20:43:58 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 12 Nov 2014 19:43:58 +0000 (UTC) Subject: [FFmpeg-user] Compile without VOB decryption References: Message-ID: Isaksson J?rgen bitfield.se> writes: > How can I compile FFmpeg without VOB decryption? FFmpeg does not support VOB decryption. Carl Eugen From c.e.macfarlane at macfh.co.uk Wed Nov 12 21:39:19 2014 From: c.e.macfarlane at macfh.co.uk (C E Macfarlane) Date: Wed, 12 Nov 2014 20:39:19 -0000 Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS In-Reply-To: Message-ID: Works with gcc 4.6.3 Thanks www.macfh.co.uk/CEMH.html UK Residents: If you feel can possibly support it please sign the following ePetition before closing time of 30/03/2015 23:59:- http://epetitions.direct.gov.uk/petitions/71556 From cehoyos at ag.or.at Wed Nov 12 23:10:55 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 12 Nov 2014 22:10:55 +0000 (UTC) Subject: [FFmpeg-user] Errors Compiling FFMPEG on MIPS References: Message-ID: C E Macfarlane macfh.co.uk> writes: > Works with gcc 4.6.3 Great, thank you for the information! Consider running the fate self-test to confirm that your binary is working fine, this is trivial if you compiled natively on your target device, if you want to run fate after cross-compilation, you have to mount one directory on both the host and the target and have a working ssh connection that doesn't need keyboard authentication. Then use the configure options --target-exec and --target-path. I will try to help you if necessary. Carl Eugen From maziar.mehrabi at gmail.com Thu Nov 13 11:34:57 2014 From: maziar.mehrabi at gmail.com (Maziar Mehrabi) Date: Thu, 13 Nov 2014 12:34:57 +0200 Subject: [FFmpeg-user] Linking ffmpeg libraries Message-ID: Hello, I am practising to learn about ffmpeg API by making small applications that use ffmpeg libraries. So far I have successfully made simple applications for codec conversion by simply including libavformat/avformat.h and libswscale/swscale.h libraries into my project. Now I intend to make some simple media streaming applications. I have been reading ffserver.c source code to get ideas from it. The problem is that I cannot include libraries such as network.h into my project. However avformat and avcodec could be easily included. I looked into the usr/local/include/libavformat path and noticed that it only contains three headers of libav into it (namely avformat, avio and version). I manually copied network.h into this folder but now I get the following error : /usr/local/include/libavformat/network.h:27:20: fatal error: config.h: No such file or directory I suppose that error is because network.h has dependencies to config.h. So now I searched for config.h and apparently there are plenty of files named like this and I don't know where to copy them. My question basically is that do I have to copy all headers into usr/local/include/ path? Is it safe? Or how else can I include necessary libraries into my project? I am coding in C++ using Eclipse IDE and I use "extern "C" {}" to include these libraries into my project. Besides this, I have also already set the -l arguments in project properties. The problem is that is works for avformat.h but not network.h. Any help or guide would be highly appreciated. Thanks -- H?lsningar, Maziar Mehrabi, From t.rapp at noa-audio.com Thu Nov 13 11:41:57 2014 From: t.rapp at noa-audio.com (Tobias Rapp) Date: Thu, 13 Nov 2014 11:41:57 +0100 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? Message-ID: <54648AF5.5050904@noa-audio.com> I want to do screen recording with FFmpeg (on Windows). It works fine with "gdigrab" but the output file size is quite high. What video codecs are available that give good compression for screen recordings while maintaining a good picture quality (screen text remains readable)? Maybe anybody has created a nice MP4 profile for it? Regards, Tobias From cehoyos at ag.or.at Thu Nov 13 11:44:01 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 10:44:01 +0000 (UTC) Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? References: <54648AF5.5050904@noa-audio.com> Message-ID: Tobias Rapp noa-audio.com> writes: > I want to do screen recording with FFmpeg (on Windows). > It works fine with "gdigrab" but the output file size > is quite high. Command line and complete, uncut console output missing / which encoders did you already test? > Maybe anybody has created a nice MP4 profile for it? I don't think the container has any relevance for the file size and the best encoders may not work with mov. Carl Eugen From cehoyos at ag.or.at Thu Nov 13 11:56:33 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 10:56:33 +0000 (UTC) Subject: [FFmpeg-user] Linking ffmpeg libraries References: Message-ID: Maziar Mehrabi gmail.com> writes: > So far I have successfully made simple applications > for codec conversion by simply including > libavformat/avformat.h and libswscale/swscale.h > libraries into my project. avformat.h and swscale.h are headers, libavformat and libswscale are libraries. > The problem is that I cannot include libraries > such as network.h into my project. network.h is another header of libavformat but an internal one that you should not need to include: libavformat contains network functionality that you should be able to use without including this internal header. It is a bug that ffserver includes the header, if you want to copy the bug, you will either have to provide a config.h header for your needs or making your project part of the FFmpeg source structure (like ffserver.c). Note that the second solution may mean that you cannot distribute the resulting binary under a different license than the LGPL (or GPL). An alternative would be to send a patch to the developer mailing list to make the needed functionality public but I suspect this will not be easy. Carl Eugen From t.rapp at noa-audio.com Thu Nov 13 12:10:02 2014 From: t.rapp at noa-audio.com (Tobias Rapp) Date: Thu, 13 Nov 2014 12:10:02 +0100 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: References: <54648AF5.5050904@noa-audio.com> Message-ID: <5464918A.9050808@noa-audio.com> On 13.11.2014 11:44, Carl Eugen Hoyos wrote: > Tobias Rapp writes: > >> I want to do screen recording with FFmpeg (on Windows). >> It works fine with "gdigrab" but the output file size >> is quite high. > > Command line and complete, uncut console output missing / > which encoders did you already test? I tested FLV and MP4, see attached log. >> Maybe anybody has created a nice MP4 profile for it? > > I don't think the container has any relevance for the > file size and the best encoders may not work with mov. I meant a x264 profile but now I realize that the binary I use doesn't include this codec. Will get an updated one. Regards, Tobias -------------- next part -------------- ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 16 2014 12:59:03 with gcc 4.8 (GCC) configuration: --disable-static --enable-shared --disable-debug --enable-version3 --disable-w32threads --arch=x86 --target-os=mingw32 --cross-prefix=i686-w64-mingw32- --prefix=/usr/local/win32 --enable-runtime-cpudetect --enable-memalign-hack libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 [gdigrab @ 01dda880] Capturing whole desktop as 1920x1080x32 at (0,0) Input #0, gdigrab, from 'desktop': Duration: N/A, start: 1415876568.549668, bitrate: 398133 kb/s Stream #0:0: Video: bmp, bgra, 1920x1080, 398133 kb/s, 6 tbr, 1000k tbn, 6 tbc Output #0, flv, to 'desktop-rec1.flv': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 1920x1080, q=2-31, 200 kb/s, 6 fps, 1k tbn, 6 tbc Metadata: encoder : Lavc56.1.100 flv Stream mapping: Stream #0:0 -> #0:0 (bmp (native) -> flv1 (flv)) Press [q] to stop, [?] for help frame= 4 fps=0.0 q=6.6 size= 893kB time=00:00:00.66 bitrate=10969.8kbits/s frame= 7 fps=6.9 q=14.6 size= 896kB time=00:00:01.16 bitrate=6291.4kbits/s frame= 10 fps=6.6 q=23.4 size= 899kB time=00:00:01.66 bitrate=4419.4kbits/s frame= 13 fps=6.4 q=24.8 size= 1136kB time=00:00:02.16 bitrate=4293.8kbits/s frame= 16 fps=6.3 q=31.0 size= 1139kB time=00:00:02.66 bitrate=3499.2kbits/s frame= 19 fps=6.3 q=31.0 size= 1142kB time=00:00:03.16 bitrate=2954.7kbits/s frame= 22 fps=6.2 q=31.0 size= 1145kB time=00:00:03.66 bitrate=2558.7kbits/s frame= 25 fps=6.2 q=24.8 size= 1382kB time=00:00:04.16 bitrate=2716.6kbits/s frame= 28 fps=6.2 q=31.0 size= 1385kB time=00:00:04.66 bitrate=2431.5kbits/s frame= 31 fps=6.1 q=31.0 size= 1388kB time=00:00:05.16 bitrate=2201.1kbits/s frame= 34 fps=6.1 q=31.0 size= 1391kB time=00:00:05.66 bitrate=2011.3kbits/s frame= 37 fps=6.1 q=24.8 size= 1628kB time=00:00:06.16 bitrate=2162.4kbits/s frame= 40 fps=6.1 q=31.0 size= 1631kB time=00:00:06.66 bitrate=2004.4kbits/s frame= 43 fps=6.1 q=31.0 size= 1634kB time=00:00:07.16 bitrate=1868.1kbits/s frame= 46 fps=6.1 q=31.0 size= 1637kB time=00:00:07.66 bitrate=1749.5kbits/s frame= 49 fps=6.1 q=24.8 size= 1874kB time=00:00:08.16 bitrate=1879.7kbits/s frame= 53 fps=6.2 q=31.0 size= 1878kB time=00:00:08.83 bitrate=1742.0kbits/s frame= 56 fps=6.2 q=31.0 size= 1881kB time=00:00:09.33 bitrate=1651.4kbits/s frame= 59 fps=6.1 q=31.0 size= 1884kB time=00:00:09.83 bitrate=1570.0kbits/s frame= 60 fps=6.0 q=31.0 Lsize= 1885kB time=00:00:10.00 bitrate=1544.6kbits/s video:1884kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.060688% -------------- next part -------------- ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 16 2014 12:59:03 with gcc 4.8 (GCC) configuration: --disable-static --enable-shared --disable-debug --enable-version3 --disable-w32threads --arch=x86 --target-os=mingw32 --cross-prefix=i686-w64-mingw32- --prefix=/usr/local/win32 --enable-runtime-cpudetect --enable-memalign-hack libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 [gdigrab @ 007ea880] Capturing whole desktop as 1920x1080x32 at (0,0) Input #0, gdigrab, from 'desktop': Duration: N/A, start: 1415876597.363668, bitrate: 398133 kb/s Stream #0:0: Video: bmp, bgra, 1920x1080, 398133 kb/s, 6 tbr, 1000k tbn, 6 tbc Output #0, mp4, to 'desktop-rec2.mp4': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 1920x1080, q=2-31, 200 kb/s, 6 fps, 12288 tbn, 6 tbc Metadata: encoder : Lavc56.1.100 mpeg4 Stream mapping: Stream #0:0 -> #0:0 (bmp (native) -> mpeg4 (native)) Press [q] to stop, [?] for help frame= 4 fps=0.0 q=6.3 size= 827kB time=00:00:00.66 bitrate=10165.1kbits/s frame= 7 fps=6.9 q=14.3 size= 830kB time=00:00:01.16 bitrate=5830.2kbits/s frame= 10 fps=6.6 q=23.1 size= 833kB time=00:00:01.66 bitrate=4096.3kbits/s frame= 13 fps=6.4 q=24.8 size= 1039kB time=00:00:02.16 bitrate=3928.8kbits/s frame= 16 fps=6.3 q=31.0 size= 1043kB time=00:00:02.66 bitrate=3205.6kbits/s frame= 19 fps=6.2 q=31.0 size= 1048kB time=00:00:03.16 bitrate=2709.8kbits/s frame= 22 fps=6.2 q=31.0 size= 1052kB time=00:00:03.66 bitrate=2349.3kbits/s frame= 25 fps=6.2 q=24.8 size= 1258kB time=00:00:04.16 bitrate=2473.1kbits/s frame= 28 fps=6.1 q=31.0 size= 1262kB time=00:00:04.66 bitrate=2215.8kbits/s frame= 31 fps=6.1 q=31.0 size= 1266kB time=00:00:05.16 bitrate=2007.8kbits/s frame= 34 fps=6.1 q=31.0 size= 1270kB time=00:00:05.66 bitrate=1836.4kbits/s frame= 37 fps=6.1 q=24.8 size= 1477kB time=00:00:06.16 bitrate=1961.7kbits/s frame= 40 fps=6.1 q=31.0 size= 1481kB time=00:00:06.66 bitrate=1819.9kbits/s frame= 43 fps=6.1 q=31.0 size= 1485kB time=00:00:07.16 bitrate=1697.5kbits/s frame= 46 fps=6.1 q=31.0 size= 1489kB time=00:00:07.66 bitrate=1591.1kbits/s frame= 50 fps=6.2 q=31.0 size= 1697kB time=00:00:08.33 bitrate=1668.3kbits/s frame= 53 fps=6.2 q=31.0 size= 1701kB time=00:00:08.83 bitrate=1577.6kbits/s frame= 56 fps=6.2 q=31.0 size= 1705kB time=00:00:09.33 bitrate=1496.7kbits/s frame= 59 fps=6.1 q=31.0 size= 1709kB time=00:00:09.83 bitrate=1423.9kbits/s frame= 60 fps=6.0 q=31.0 Lsize= 1712kB time=00:00:10.00 bitrate=1402.2kbits/s video:1711kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.064571% From cehoyos at ag.or.at Thu Nov 13 12:33:00 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 11:33:00 +0000 (UTC) Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> Message-ID: Tobias Rapp noa-audio.com> writes: > > Command line and complete, uncut console output > > missing / which encoders did you already test? > > I tested FLV and MP4, see attached log. Neither are encoders imo but see below. > Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p > Stream #0:0: Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p I am surprised now: You wrote originally "I want to do screen recording [...] but the output file size is quite high". I was assuming that you meant lossless encoding in the rgb colourspace. For lossy encoding (flv1 and mpeg4 only support lossy encoding), you can get a smaller filesize by either specifying a lower bitrate or requesting a higher constant quantizer. As you already mentioned, the best lossy encoder (with acceptable encoding speed) is x264. If you want lossless encoding that keeps the original colourspace (rgb), please test png, ffv1, (ff)huffyuv and flashsv2. Then also test x264rgb with the lossless option. (The filesizes may be very significantly higher in all these cases than what you saw with the lossy encodings you tested, this is not unexpected. The files may be smaller depending on what you record.) Carl Eugen From t.rapp at noa-audio.com Thu Nov 13 12:34:59 2014 From: t.rapp at noa-audio.com (Tobias Rapp) Date: Thu, 13 Nov 2014 12:34:59 +0100 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: <5464918A.9050808@noa-audio.com> References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> Message-ID: <54649763.4060302@noa-audio.com> On 13.11.2014 12:10, Tobias Rapp wrote: > On 13.11.2014 11:44, Carl Eugen Hoyos wrote: >> I don't think the container has any relevance for the >> file size and the best encoders may not work with mov. > > I meant a x264 profile but now I realize that the binary I use doesn't > include this codec. Will get an updated one. When testing with a recent version of FFmpeg with x264 included the output file size decreases significantly (see attached log). Sorry for the noise. Tobias -------------- next part -------------- ffmpeg version N-67586-g3e1ac10 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 12 2014 22:10:14 with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 12.100 / 56. 12.100 libavformat 56. 12.103 / 56. 12.103 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [gdigrab @ 0000000002c4cc40] Capturing whole desktop as 1920x1080x32 at (0,0) Input #0, gdigrab, from 'desktop': Duration: N/A, start: 1415878114.002614, bitrate: 398133 kb/s Stream #0:0: Video: bmp, bgra, 1920x1080, 398133 kb/s, 6 tbr, 1000k tbn, 6 tbc No pixel format specified, yuv444p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players. [libx264 @ 0000000002bec3c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX AVX2 FMA3 LZCNT BMI2 [libx264 @ 0000000002bec3c0] profile High 4:4:4 Predictive, level 4.0, 4:4:4 8-bit [libx264 @ 0000000002bec3c0] 264 - core 142 r2479 dd79a61 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=6 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 'desktop-rec3.mp4': Metadata: encoder : Lavf56.12.103 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv444p, 1920x1080, q=-1--1, 6 fps, 12288 tbn, 6 tbc Metadata: encoder : Lavc56.12.100 libx264 Stream mapping: Stream #0:0 -> #0:0 (bmp (native) -> h264 (libx264)) Press [q] to stop, [?] for help frame= 4 fps=0.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 7 fps=6.6 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 10 fps=6.4 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 13 fps=6.3 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 16 fps=6.2 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 19 fps=6.2 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 22 fps=6.2 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 25 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 28 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 31 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 34 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 37 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 40 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 43 fps=6.1 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 46 fps=6.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 49 fps=6.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A frame= 52 fps=6.0 q=23.0 size= 270kB time=00:00:00.00 bitrate=N/A frame= 55 fps=6.0 q=23.0 size= 271kB time=00:00:00.50 bitrate=4445.4kbits/s frame= 59 fps=6.1 q=23.0 size= 277kB time=00:00:01.16 bitrate=1941.6kbits/s frame= 60 fps=5.6 q=-1.0 Lsize= 291kB time=00:00:09.66 bitrate= 246.6kbits/s video:290kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.525524% [libx264 @ 0000000002bec3c0] frame I:1 Avg QP:16.01 size:265423 [libx264 @ 0000000002bec3c0] frame P:15 Avg QP:18.35 size: 1209 [libx264 @ 0000000002bec3c0] frame B:44 Avg QP:24.27 size: 278 [libx264 @ 0000000002bec3c0] consecutive B-frames: 1.7% 0.0% 5.0% 93.3% [libx264 @ 0000000002bec3c0] mb I I16..4: 27.7% 34.3% 38.0% [libx264 @ 0000000002bec3c0] mb P I16..4: 0.2% 0.0% 0.0% P16..4: 1.4% 0.1% 0.0% 0.0% 0.0% skip:98.2% [libx264 @ 0000000002bec3c0] mb B I16..4: 0.1% 0.2% 0.0% B16..8: 1.6% 0.0% 0.0% direct: 0.0% skip:98.0% L0:42.7% L1:57.3% BI: 0.0% [libx264 @ 0000000002bec3c0] 8x8 transform intra:37.4% inter:33.4% [libx264 @ 0000000002bec3c0] coded y,u,v intra: 23.8% 5.3% 5.1% inter: 0.1% 0.0% 0.0% [libx264 @ 0000000002bec3c0] i16 v,h,dc,p: 38% 62% 0% 0% [libx264 @ 0000000002bec3c0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 57% 9% 33% 0% 0% 0% 0% 0% 0% [libx264 @ 0000000002bec3c0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 34% 12% 2% 2% 2% 3% 2% 3% [libx264 @ 0000000002bec3c0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000002bec3c0] ref P L0: 67.0% 0.7% 28.5% 3.8% [libx264 @ 0000000002bec3c0] ref B L0: 53.0% 44.5% 2.5% [libx264 @ 0000000002bec3c0] ref B L1: 94.7% 5.3% [libx264 @ 0000000002bec3c0] kb/s:236.62 From cehoyos at ag.or.at Thu Nov 13 12:40:22 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 11:40:22 +0000 (UTC) Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> Message-ID: Tobias Rapp noa-audio.com> writes: > Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv444p If your input is rgb and you don't care about compatibility, please don't use yuv444p but -vcodec libx264rgb to get higher output quality. Carl Eugen From t.rapp at noa-audio.com Thu Nov 13 12:52:05 2014 From: t.rapp at noa-audio.com (Tobias Rapp) Date: Thu, 13 Nov 2014 12:52:05 +0100 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> Message-ID: <54649B65.6080904@noa-audio.com> On 13.11.2014 12:40, Carl Eugen Hoyos wrote: > Tobias Rapp writes: > >> Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv444p > > If your input is rgb and you don't care about compatibility, > please don't use yuv444p but -vcodec libx264rgb to get higher > output quality. The recording doesn't need to be loss-less, I just want to make screen-casts to communicate GUI application issues to my colleges. To minimize the network upload time I searched for file size optimizations. Regards, Tobias From kuban.altan at sinefekt.com Thu Nov 13 13:11:38 2014 From: kuban.altan at sinefekt.com (Kuban Altan) Date: Thu, 13 Nov 2014 14:11:38 +0200 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: <54649B65.6080904@noa-audio.com> References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> <54649B65.6080904@noa-audio.com> Message-ID: Hi I suppose that, x264 has a lossless mode when the quantizer scale is set to 0. If that is also applicable to libx264rgb, then it is worth to look at. So your pixels will still stay rgb, and you will have x264 efficiency in lossless mode. kuban altan On Thu, Nov 13, 2014 at 1:52 PM, Tobias Rapp wrote: > On 13.11.2014 12:40, Carl Eugen Hoyos wrote: > >> Tobias Rapp writes: >> >> Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv444p >>> >> >> If your input is rgb and you don't care about compatibility, >> please don't use yuv444p but -vcodec libx264rgb to get higher >> output quality. >> > > The recording doesn't need to be loss-less, I just want to make > screen-casts to communicate GUI application issues to my colleges. To > minimize the network upload time I searched for file size optimizations. > > Regards, > Tobias > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Thu Nov 13 13:18:18 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 12:18:18 +0000 (UTC) Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> <54649B65.6080904@noa-audio.com> Message-ID: Tobias Rapp noa-audio.com> writes: > To minimize the network upload time I searched > for file size optimizations. I understand. If your input is rgb, do not use -vcodec x264 with yuv444p colour space but -vcodec libx264rgb which allows higher quality (or smaller file size). Both (yuv444p and rgb) will not work with all players. Carl Eugen From kuban.altan at gmail.com Thu Nov 13 13:27:49 2014 From: kuban.altan at gmail.com (Kuban Altan) Date: Thu, 13 Nov 2014 14:27:49 +0200 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> <54649B65.6080904@noa-audio.com> Message-ID: I would also give a shot for *quicktime animation* codec *(-vcodec qtrle)*. Which can be lossless. But you have to test for file sizes. Which one gives you better file size and minimum cpu load? kuban altan On Thu, Nov 13, 2014 at 2:18 PM, Carl Eugen Hoyos wrote: > Tobias Rapp noa-audio.com> writes: > > > To minimize the network upload time I searched > > for file size optimizations. > > I understand. > If your input is rgb, do not use -vcodec x264 with > yuv444p colour space but -vcodec libx264rgb which > allows higher quality (or smaller file size). > > Both (yuv444p and rgb) will not work with all players. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From t.rapp at noa-audio.com Thu Nov 13 14:42:31 2014 From: t.rapp at noa-audio.com (Tobias Rapp) Date: Thu, 13 Nov 2014 14:42:31 +0100 Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? In-Reply-To: References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> <54649B65.6080904@noa-audio.com> Message-ID: <5464B547.9030700@noa-audio.com> On 13.11.2014 13:27, Kuban Altan wrote: > I would also give a shot for *quicktime animation* codec *(-vcodec > qtrle)*. Which can be lossless. But you have to test for file sizes. > Which one gives you better file size and minimum cpu load? From a quick test it seems that "-vcodec qtrle -g 60" is about the same size as "-vcodec mpeg4". The video bitrate is around 1200-1400kb/s. With x264 the bitrate is around 400kb/s. Tobias From anilj.mailing at gmail.com Thu Nov 13 16:38:25 2014 From: anilj.mailing at gmail.com (Anil Jangam) Date: Thu, 13 Nov 2014 10:38:25 -0500 Subject: [FFmpeg-user] Decoding continuous stream of I-frame+P/B frames block. Message-ID: Hi, My understanding is that when a new Container object is to be initialized, it always requires an I-frame + subsequent P or B frames to initialize else open fails. My question is, how do we use the same Container where there is a continuous stream of set of I-frame+P/B frames, I-frame spaced at GOP interval (this is a real-time video stream use case), whether it is possible to use the same Container without re-initializing it (i.e. I do not want to initialize/open the Container object each time). I want to know if it is possible to reuse the same Container object while we keep feeding it the received encoded frames into the Container's input buffer. This way, we do not wait for accumulating a set of I-frame+P/B frame block and then give it to the Container for decoding. This approach is to avoid the receive buffering to improve the processing time. We assume some amount of de-jitter buffer to order the received frame. /anil. From cehoyos at ag.or.at Thu Nov 13 18:37:22 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 17:37:22 +0000 (UTC) Subject: [FFmpeg-user] Decoding continuous stream of I-frame+P/B frames block. References: Message-ID: Anil Jangam gmail.com> writes: > My understanding is that when a new Container object > is to be initialized, it always requires an I-frame + > subsequent P or B frames to initialize else open fails. Valid (h264) streams exist that do not contain I-frames, only P- (and B-) frames. I unfortunately don't understand your question, not even if it is decoding- or encoding-related. Could you explain what you are trying to do and what doesn't work? Carl Eugen From cehoyos at ag.or.at Thu Nov 13 18:39:25 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 17:39:25 +0000 (UTC) Subject: [FFmpeg-user] Which videoc codec gives best compression for screen recording? References: <54648AF5.5050904@noa-audio.com> <5464918A.9050808@noa-audio.com> <54649763.4060302@noa-audio.com> <54649B65.6080904@noa-audio.com> Message-ID: Kuban Altan gmail.com> writes: > I would also give a shot for *quicktime animation* > codec *(-vcodec qtrle)*. Which can be lossless. It is only (always) lossless but Tobias wants lossy encoding (which is unusual for screen capturing). Please do not top-post here, Carl Eugen From anilj.mailing at gmail.com Thu Nov 13 22:37:03 2014 From: anilj.mailing at gmail.com (Anil Jangam) Date: Thu, 13 Nov 2014 16:37:03 -0500 Subject: [FFmpeg-user] Decoding continuous stream of I-frame+P/B frames block. In-Reply-To: References: Message-ID: Hello Carl, In my implementation, the encoder is generating the encoded frames and building video chunks, which contains I-frame, P(B)-frames. We are configuring a # of frames per this chunk as well as GOP interval (interval for I-frames). On decoder side, we are decoding this video block to get the pictures and display them to get the video play. As you can observe, block building is a process of accumulating a number of frames, and it takes a time (number of frames x 1/frame rate ms). This is adding to the overall end to end latency. My question is, is it possible to use just one Container and feed it the frame by frame data so that the buffering or block logic can be avoided? Can you also elaborate what do you mean by - "Valid (h264) streams exist that do not contain I-frames, only P- (and B-) frames."? - When does an I-frame is used? - What is the notion of GOP interval and how it is used? Or is this a container level feature? How it is related, if it does? Hope this is more clear. /anil. On Thu, Nov 13, 2014 at 12:37 PM, Carl Eugen Hoyos wrote: > Anil Jangam gmail.com> writes: > > > My understanding is that when a new Container object > > is to be initialized, it always requires an I-frame + > > subsequent P or B frames to initialize else open fails. > > Valid (h264) streams exist that do not contain I-frames, > only P- (and B-) frames. > > I unfortunately don't understand your question, not even > if it is decoding- or encoding-related. > Could you explain what you are trying to do and what > doesn't work? > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From cehoyos at ag.or.at Thu Nov 13 22:45:29 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 13 Nov 2014 21:45:29 +0000 (UTC) Subject: [FFmpeg-user] Decoding continuous stream of I-frame+P/B frames block. References: Message-ID: Anil Jangam gmail.com> writes: > Can you also elaborate what do you mean by - "Valid > (h264) streams exist that do not contain I-frames, > only P- (and B-) frames."? I meant that streams without any I-frames exist and are valid. (I don't know if this has anything to do with your question.) > - When does an I-frame is used? In frames without I-frames, I-frames are never used. Recovery points exist instead, but they can be P-frames. I don't now if GOP intervals have any relevance in such a case (probably not). > Hope this is more clear. No, sorry. Please try not to top-post here, Carl Eugen From suri at baymicrosystems.com Fri Nov 14 17:24:43 2014 From: suri at baymicrosystems.com (Suri Shelvapille) Date: Fri, 14 Nov 2014 16:24:43 +0000 Subject: [FFmpeg-user] ffmpeg play continuous loop! Message-ID: Dear Folks: First of all, this is an awesome program! I am using ffmpeg and ffplay combination to display a mosaic of 12 videos. I am using shortest=1 option to ffmpeg to stop all videos when the shortest video finishes. I need to play all the videos in a loop. Can you please suggest the correct way. Here is the shortened command (for 2 videos) : ----------------------------------------------------------------- #!/bin/bash ffmpeg -i a1.avi -i a2.avi -filter_complex "nullsrc=size=1440x1080 [base];[0:v] setpts=PTS-STARTPTS, scale=480x270 [pos0];[1:v] setpts=PTS-STARTPTS, scale=480x270 [pos1];[base][pos0] overlay=shortest=1 [tmp1];[tmp1][pos1] overlay=shortest=1:x=480" -c:v libx264 -f avi - | ffplay - many thanks, Suri From jeremy at ZeeVee.Com Sat Nov 15 01:04:05 2014 From: jeremy at ZeeVee.Com (Jeremy Greene) Date: Sat, 15 Nov 2014 00:04:05 +0000 Subject: [FFmpeg-user] ts discontinuity indicator Message-ID: (I am not on mailing list, so please cc me!) It appears that ffmpeg does not fully support the discontinuity indicator in the afc. It does ignore discontinuity in the continuity counter, but it should also ignore a discontinuity in the PCR (and PTS/DTS). Is this a known issue? Jeremy From lhcwjy at 163.com Sat Nov 15 13:12:04 2014 From: lhcwjy at 163.com (=?UTF-8?B?5YiY5a6P5bed?=) Date: Sat, 15 Nov 2014 20:12:04 +0800 (CST) Subject: [FFmpeg-user] can not set the defaut stream! Message-ID: <4201ba20.620c.149b35e0823.Coremail.lhcwjy@163.com> if FFmeg.exe's output has more than one stream of the same kind, such as audo or subtitle, there is not the default one. Especially? if there is no the default subtitle, most of the players will not display any subtitle. FFproble.exe can show if there is the deflaut stream of every stream. FFprobe -show_streams INPUT [STREAM] ... DISPOSITION:default=0 ... [/STREAM] From radpopl at gmail.com Sat Nov 15 13:38:55 2014 From: radpopl at gmail.com (radpopl) Date: Sat, 15 Nov 2014 04:38:55 -0800 (PST) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced Message-ID: <1416055135474-4668119.post@n4.nabble.com> Hello, I have a mts file (AVC, 1440x1080i, TFF). First attempt: ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid -flags +qpel -acodec libmp3lame -b:a 320k "%%~dF%%~pF%%~nF.avi" But destination file has an interlace artifacts, unacceptable... I have tried some command but nothing works, I can't even find good solution on web/ffmpeg manual. Can somebody help me which filters/options I should use to get XVID interlaced video? Or maybe I should deinterlace mts file to 1440x540p 50fps file and convert to progressive xvid? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From oinos at web.de Sat Nov 15 22:46:15 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Sat, 15 Nov 2014 22:46:15 +0100 Subject: [FFmpeg-user] screencast in WinXP Message-ID: <5467C9A7.6030401@web.de> Hi there, I need to record some screencasts with audio in a computer using WinXP. I followed the instructions provided at https://trac.ffmpeg.org/wiki/Capture/Desktop#Generalnote, but I?m afraid they didn?t work. Here you have the uncut sample: C:\>ffmpeg -f dshow -i video="screen-capture-recorder":audio="Microphone" -vcode c libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.flv ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [dshow @ 02ed89e0] Could not enumerate video devices. video=screen-capture-recorder:audio=Microphone: Input/output error Listing devices didn?t work either: C:>ffmpeg -list_devices true -f dshow -i dummy ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [dshow @ 02dde020] DirectShow video devices [dshow @ 02dde020] Could not enumerate video devices. [dshow @ 02dde020] DirectShow audio devices [dshow @ 02dde020] "Realtek HD Audio Input" [dshow @ 02dde020] "Realtek HD Digital input" dummy: Immediate exit requested What am I doing wrong? Many thanks for your work, Pablo -- http://www.ousia.tk From james.darnley at gmail.com Sat Nov 15 22:50:16 2014 From: james.darnley at gmail.com (James Darnley) Date: Sat, 15 Nov 2014 22:50:16 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <5467C9A7.6030401@web.de> References: <5467C9A7.6030401@web.de> Message-ID: <5467CA98.8070705@gmail.com> On 2014-11-15 22:46, Pablo Rodr?guez wrote: > [dshow @ 02dde020] DirectShow video devices > [dshow @ 02dde020] Could not enumerate video devices. > [dshow @ 02dde020] DirectShow audio devices > [dshow @ 02dde020] "Realtek HD Audio Input" > [dshow @ 02dde020] "Realtek HD Digital input" > dummy: Immediate exit requested > > What am I doing wrong? You have no video devices so there is no possibility of using one to capture the desktop. Read that wiki link a little better. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From oinos at web.de Sat Nov 15 23:00:34 2014 From: oinos at web.de (=?windows-1252?Q?Pablo_Rodr=EDguez?=) Date: Sat, 15 Nov 2014 23:00:34 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <5467CA98.8070705@gmail.com> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> Message-ID: <5467CD02.5010608@web.de> On 11/15/2014 10:50 PM, James Darnley wrote: > On 2014-11-15 22:46, Pablo Rodr?guez wrote: >> [dshow @ 02dde020] DirectShow video devices >> [dshow @ 02dde020] Could not enumerate video devices. >> [dshow @ 02dde020] DirectShow audio devices >> [dshow @ 02dde020] "Realtek HD Audio Input" >> [dshow @ 02dde020] "Realtek HD Digital input" >> dummy: Immediate exit requested >> >> What am I doing wrong? > > You have no video devices so there is no possibility of using one to > capture the desktop. Read that wiki link a little better. Thanks for the reply, James. Sorry, but I didn?t know that the video card wasn?t a DirectShow video device. Pablo -- http://www.ousia.tk From james.darnley at gmail.com Sat Nov 15 23:23:56 2014 From: james.darnley at gmail.com (James Darnley) Date: Sat, 15 Nov 2014 23:23:56 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <5467CD02.5010608@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> Message-ID: <5467D27C.2000501@gmail.com> On 2014-11-15 23:00, Pablo Rodr?guez wrote: > On 11/15/2014 10:50 PM, James Darnley wrote: >> On 2014-11-15 22:46, Pablo Rodr?guez wrote: >>> [dshow @ 02dde020] DirectShow video devices >>> [dshow @ 02dde020] Could not enumerate video devices. >>> [dshow @ 02dde020] DirectShow audio devices >>> [dshow @ 02dde020] "Realtek HD Audio Input" >>> [dshow @ 02dde020] "Realtek HD Digital input" >>> dummy: Immediate exit requested >>> >>> What am I doing wrong? >> >> You have no video devices so there is no possibility of using one to >> capture the desktop. Read that wiki link a little better. > > Thanks for the reply, James. > > Sorry, but I didn?t know that the video card wasn?t a DirectShow video > device. Sorry, I was probably more terse that I should have been. A "device" to DirectShow has quite a specific meaning. You can see the two inputs to your soundcard there because something has created/installed the two devices in DirectShow. You can do something similar for desktop video capture. The wiki article links to one such utility on github. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From cehoyos at ag.or.at Sun Nov 16 02:13:47 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 01:13:47 +0000 (UTC) Subject: [FFmpeg-user] gas-preprocessor.pl unknown arch: 'ppc' References: <54576040.1020201@gmail.com> <5457AE8B.4070807@gmail.com> Message-ID: Pavel Koshevoy gmail.com> writes: > This happens on line 66 of gas-preprocessor.pl -- it checks > the $comments dictionary for 'ppc', doesn't find it and > gives up. I've worked around the problem locally by adding > 'ppc' => '#' to %comments on line 17. Finally tested and applied: https://github.com/FFmpeg/gas-preprocessor/commit/b5d6b2b7 Sorry for the absurd delay! Thank you, Carl Eugen From pkoshevoy at gmail.com Sun Nov 16 02:33:52 2014 From: pkoshevoy at gmail.com (Pavel Koshevoy) Date: Sat, 15 Nov 2014 18:33:52 -0700 Subject: [FFmpeg-user] gas-preprocessor.pl unknown arch: 'ppc' In-Reply-To: References: <54576040.1020201@gmail.com> <5457AE8B.4070807@gmail.com> Message-ID: <5467FF00.40201@gmail.com> On 11/15/14 18:13, Carl Eugen Hoyos wrote: > Pavel Koshevoy gmail.com> writes: > >> This happens on line 66 of gas-preprocessor.pl -- it checks >> the $comments dictionary for 'ppc', doesn't find it and >> gives up. I've worked around the problem locally by adding >> 'ppc' => '#' to %comments on line 17. > Finally tested and applied: > https://github.com/FFmpeg/gas-preprocessor/commit/b5d6b2b7 That change looks wrong -- "ppc" => '#'); should be "ppc" => '#', Pavel From michaelni at gmx.at Sun Nov 16 03:10:18 2014 From: michaelni at gmx.at (Michael Niedermayer) Date: Sun, 16 Nov 2014 03:10:18 +0100 Subject: [FFmpeg-user] gas-preprocessor.pl unknown arch: 'ppc' In-Reply-To: <5467FF00.40201@gmail.com> References: <54576040.1020201@gmail.com> <5457AE8B.4070807@gmail.com> <5467FF00.40201@gmail.com> Message-ID: <20141116021018.GA2795@nb4> On Sat, Nov 15, 2014 at 06:33:52PM -0700, Pavel Koshevoy wrote: > On 11/15/14 18:13, Carl Eugen Hoyos wrote: > >Pavel Koshevoy gmail.com> writes: > > > >>This happens on line 66 of gas-preprocessor.pl -- it checks > >>the $comments dictionary for 'ppc', doesn't find it and > >>gives up. I've worked around the problem locally by adding > >>'ppc' => '#' to %comments on line 17. > >Finally tested and applied: > >https://github.com/FFmpeg/gas-preprocessor/commit/b5d6b2b7 > > That change looks wrong -- "ppc" => '#'); > should be "ppc" => '#', fixed [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB Breaking DRM is a little like attempting to break through a door even though the window is wide open and the only thing in the house is a bunch of things you dont want and which you would get tomorrow for free anyway -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 181 bytes Desc: Digital signature URL: From oinos at web.de Sun Nov 16 09:14:53 2014 From: oinos at web.de (=?windows-1252?Q?Pablo_Rodr=EDguez?=) Date: Sun, 16 Nov 2014 09:14:53 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <5467D27C.2000501@gmail.com> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> Message-ID: <54685CFD.7090207@web.de> On 11/15/2014 11:23 PM, James Darnley wrote: > On 2014-11-15 23:00, Pablo Rodr?guez wrote: >> [...] >> Sorry, but I didn?t know that the video card wasn?t a DirectShow video >> device. > > Sorry, I was probably more terse that I should have been. A "device" to > DirectShow has quite a specific meaning. You can see the two inputs to > your soundcard there because something has created/installed the two > devices in DirectShow. > > You can do something similar for desktop video capture. The wiki > article links to one such utility on github. Many thanks for your reply, James. I didn?t know that it was a software issue. I don?t have admin rights at the computer in which I would record the screencast. So I adding a DirectShow device is not a option for me. I don?t have much Windows knowledge. But I had recorded screencast with Camstudio before (this seems not to be an option anymore). Is there now way to use ffmpeg with VFW? Sorry if write nonsense, I?m guessing. Many thanks for your help, Pablo -- http://www.ousia.tk From velotiaray at gmail.com Sun Nov 16 11:48:34 2014 From: velotiaray at gmail.com (Velotiaray Toto-Zarasoa) Date: Sun, 16 Nov 2014 11:48:34 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <54685CFD.7090207@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> Message-ID: Hi, Now ffmpeg can be compiled with support to gdigrab device. You can record the whole desktop or any portion of the screen. Velotiaray Toto-Zarasoa ------------------------------- > Le 16 nov. 2014 ? 09:14, Pablo Rodr?guez a ?crit : > >> On 11/15/2014 11:23 PM, James Darnley wrote: >>> On 2014-11-15 23:00, Pablo Rodr?guez wrote: >>> [...] >>> Sorry, but I didn?t know that the video card wasn?t a DirectShow video >>> device. >> >> Sorry, I was probably more terse that I should have been. A "device" to >> DirectShow has quite a specific meaning. You can see the two inputs to >> your soundcard there because something has created/installed the two >> devices in DirectShow. >> >> You can do something similar for desktop video capture. The wiki >> article links to one such utility on github. > > Many thanks for your reply, James. > > I didn?t know that it was a software issue. I don?t have admin rights at > the computer in which I would record the screencast. So I adding a > DirectShow device is not a option for me. > > I don?t have much Windows knowledge. But I had recorded screencast with > Camstudio before (this seems not to be an option anymore). Is there now > way to use ffmpeg with VFW? Sorry if write nonsense, I?m guessing. > > Many thanks for your help, > > > Pablo > -- > http://www.ousia.tk > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rabbit+list at rabbit.us Sun Nov 16 13:36:44 2014 From: rabbit+list at rabbit.us (Peter Rabbitson) Date: Sun, 16 Nov 2014 13:36:44 +0100 Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly Message-ID: <54689A5C.8030200@rabbit.us> Hi! This is not strictly a question on ffmpeg itself, as it may be the fault of vlc. But since vlc uses the same backend it is all somehow related. I am trying to create a video which will display in full RGB color on a random player (in this case vlc). No matter what I do - I end up with color loss (likely yuv420 conversion) somewhere along the road. I have irrefutably verified that my video file contains the full color information, because I can do: source image \ video \ extracted frame and I end up arriving at the same source bytes (described in detail below). However playback of the generated video and original/extracted images show a noticeable color loss when displayed side by side (screen capture in file side_by_side.png). What am I missing? Thank you! ==== Follows description of the testing method All (very small) files described below are attached to this message. The sequence of events: I start with a small image: > ~$ ffmpeg -hide_banner -i miniansi_orig.png > Input #0, png_pipe, from 'miniansi_orig.png': > Duration: N/A, bitrate: N/A > Stream #0:0: Video: png, rgb24, 320x480 [SAR 1:1 DAR 2:3], 25 tbr, 25 tbn, 25 tbc > At least one output file must be specified I generate a short 1fps 10sec video from this source: > ~$ ffmpeg -hide_banner -loop 1 -i miniansi_orig.png -t 10 -r 1 -c:v libx264 -crf 0 -preset veryslow -pix_fmt yuv444p miniansi.mkv > Input #0, png_pipe, from 'miniansi_orig.png': > Duration: N/A, bitrate: N/A > Stream #0:0: Video: png, rgb24, 320x480 [SAR 1:1 DAR 2:3], 25 fps, 25 tbr, 25 tbn, 25 tbc > [libx264 @ 0x1ce5000] using SAR=1/1 > [libx264 @ 0x1ce5000] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 > [libx264 @ 0x1ce5000] profile High 4:4:4 Predictive, level 3.1, 4:4:4 8-bit > [libx264 @ 0x1ce5000] 264 - core 142 r2431 a5831aa - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=9 psy=0 mixed_ref=1 me_range=24 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=2 keyint=250 keyint_min=1 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=0 > Output #0, matroska, to 'miniansi.mkv': > Metadata: > encoder : Lavf56.4.101 > Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv444p, 320x480 [SAR 1:1 DAR 2:3], q=-1--1, 1 fps, 1k tbn, 1 tbc > Metadata: > encoder : Lavc56.1.100 libx264 > Stream mapping: > Stream #0:0 -> #0:0 (png (native) -> h264 (libx264)) > Press [q] to stop, [?] for help > frame= 10 fps=0.0 q=-1.0 Lsize= 36kB time=00:00:10.00 bitrate= 29.1kbits/s dup=0 drop=228 > video:35kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.126225% > [libx264 @ 0x1ce5000] frame I:1 Avg QP: 0.00 size: 34857 > [libx264 @ 0x1ce5000] frame P:9 Avg QP: 0.00 size: 24 > [libx264 @ 0x1ce5000] mb I I16..4: 34.3% 5.5% 60.2% > [libx264 @ 0x1ce5000] mb P I16..4: 0.0% 0.0% 0.0% P16..4: 0.0% 0.0% 0.0% 0.0% 0.0% skip:100.0% > [libx264 @ 0x1ce5000] 8x8 transform intra:5.5% > [libx264 @ 0x1ce5000] coded y,u,v intra: 47.9% 47.8% 47.8% inter: 0.0% 0.0% 0.0% > [libx264 @ 0x1ce5000] i16 v,h,dc,p: 68% 32% 0% 0% > [libx264 @ 0x1ce5000] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 67% 23% 10% 0% 0% 0% 0% 0% 0% > [libx264 @ 0x1ce5000] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 44% 46% 7% 1% 0% 0% 0% 0% 2% > [libx264 @ 0x1ce5000] Weighted P-Frames: Y:0.0% UV:0.0% > [libx264 @ 0x1ce5000] kb/s:28.06 I extract a single frame from the resulting video: > ~$ ffmpeg -hide_banner -i miniansi.mkv -vframes 1 miniansi_frame.png > [matroska,webm @ 0x24700a0] decoding for stream 0 failed > Input #0, matroska,webm, from 'miniansi.mkv': > Metadata: > ENCODER : Lavf56.4.101 > Duration: 00:00:10.00, bitrate: 29 kb/s > Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p, 320x480 [SAR 1:1 DAR 2:3], 1 fps, 1 tbr, 1k tbn, 2 tbc (default) > Metadata: > ENCODER : Lavc56.1.100 libx264 > Output #0, image2, to 'miniansi_frame.png': > Metadata: > encoder : Lavf56.4.101 > Stream #0:0: Video: png, rgb24, 320x480 [SAR 1:1 DAR 2:3], q=2-31, 200 kb/s, 1 fps, 1 tbn, 1 tbc (default) > Metadata: > encoder : Lavc56.1.100 png > Stream mapping: > Stream #0:0 -> #0:0 (h264 (native) -> png (native)) > Press [q] to stop, [?] for help > frame= 1 fps=0.0 q=0.0 Lsize=N/A time=00:00:01.00 bitrate=N/A > video:8kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown This results (as expected) in a file identical to the source, proving that the video contains the full RGB (or equivalent yuv444) set of color information: > ~$ diff miniansi_orig.png miniansi_frame.png; echo $? > 0 Yet the results of playing the generated mkv in vlc (or for that matter any other player) are visibly terrible (see side_by_side.png) -------------- next part -------------- A non-text attachment was scrubbed... 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Name: side_by_side.png Type: image/png Size: 60713 bytes Desc: not available URL: From oinos at web.de Sun Nov 16 13:43:54 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Sun, 16 Nov 2014 13:43:54 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> Message-ID: <54689C0A.7040307@web.de> On 11/16/2014 11:48 AM, Velotiaray Toto-Zarasoa wrote: > Hi, > > Now ffmpeg can be compiled with support to gdigrab device. You can > record the whole desktop or any portion of the screen. Many thanks for your reply, Velotiaray. Which would be the option? Pablo -- http://www.ousia.tk From francois.visagie at gmail.com Sun Nov 16 13:58:22 2014 From: francois.visagie at gmail.com (Francois Visagie) Date: Sun, 16 Nov 2014 14:58:22 +0200 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: <1416055135474-4668119.post@n4.nabble.com> References: <1416055135474-4668119.post@n4.nabble.com> Message-ID: <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of radpopl > Sent: 15 November 2014 14:39 > To: ffmpeg-user at ffmpeg.org > Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced > > Hello, > > I have a mts file (AVC, 1440x1080i, TFF). > > First attempt: > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid -flags +qpel -acodec > libmp3lame -b:a 320k "%%~dF%%~pF%%~nF.avi" > > But destination file has an interlace artifacts, unacceptable... > > I have tried some command but nothing works, I can't even find good > solution on web/ffmpeg manual. > Can somebody help me which filters/options I should use to get XVID > interlaced video? You need to add ' -flags +ildct+ilme'. With -c:v libx264 you also can (should) add the bff/tff option; although personally not familiar with ffmpeg XVID encoding myself you might want to look for something similar. > > Or maybe I should deinterlace mts file to 1440x540p 50fps file and convert to > progressive xvid? > > > > > -- > View this message in context: http://ffmpeg- > users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced- > tp4668119.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From adf.lists at gmail.com Sun Nov 16 14:57:32 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 16 Nov 2014 13:57:32 +0000 Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly In-Reply-To: <54689A5C.8030200@rabbit.us> References: <54689A5C.8030200@rabbit.us> Message-ID: <5468AD4C.9040300@gmail.com> Peter Rabbitson wrote: > This results (as expected) in a file identical to the source, proving > that the video contains the full RGB (or equivalent yuv444) set of > color information: > >> ~$ diff miniansi_orig.png miniansi_frame.png; echo $? 0 Only for this sample, not that it really matters in this case, but rgb -> yuv will loose colour and there is also a conversion from full range to studio and back. > Yet the results of playing the generated mkv in vlc (or for that > matter any other player) are visibly terrible (see side_by_side.png) It's the player or rather the way it works/is setup/is setup for your hardware. Players by default will use GPU for CSC/scale and these often won't avertise 444 as an input option so VLC will likely convert to 420. There is likely some way you could get VLC to do it, with mplayer I think specifying opengl would do. I don't really know what to suggest - 422 would be more likely to work - but then you would still see some loss. From rabbit+list at rabbit.us Sun Nov 16 15:28:40 2014 From: rabbit+list at rabbit.us (Peter Rabbitson) Date: Sun, 16 Nov 2014 15:28:40 +0100 Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly In-Reply-To: <5468AD4C.9040300@gmail.com> References: <54689A5C.8030200@rabbit.us> <5468AD4C.9040300@gmail.com> Message-ID: <5468B498.2060901@rabbit.us> I am replying in-line further down, but wanted to to ask a generic question, just to make sure I fully understand what is actually going on: In 1997 Intel came up with the MMX instruction set which among other things allows in-hardware rgb->422/420 conversion. Since then all players default to this on a virtually hardcoded level, and the GUI subsystem is expected to convert everything back to rgb (using the very same instruction set). As a result of the MMX success, everything in the digital video world *17 years* later is inescapably doing this rgb->yuv->rgb dance, even in the case of 100% correct and complete metadata. This is true for all cases including the situation when both the source and the destination are operating in the RGB24 colorspace: CGI produced under an RGB renderer, to be displayed in a web-browser player, which by definition operates on an RGB device. Did I get the state of the art about right? On 11/16/2014 02:57 PM, Andy Furniss wrote: > Peter Rabbitson wrote: > >> This results (as expected) in a file identical to the source, proving >> that the video contains the full RGB (or equivalent yuv444) set of >> color information: >> >>> ~$ diff miniansi_orig.png miniansi_frame.png; echo $? 0 > > Players by default will use GPU for CSC/scale and these often won't > avertise 444 as an input option so VLC will likely convert to 420. > > There is likely some way you could get VLC to do it, with mplayer I > think specifying opengl would do. > > I don't really know what to suggest - 422 would be more likely to work > - but then you would still see some loss. > You are actually correct - the conversion that took place is 422, not 420 as I originally thought. Verified by: > ~$ ffmpeg -hide_banner -i miniansi_frame.png -f matroska -pix_fmt yuv422p - | ffmpeg -hide_banner -i - -pix_fmt rgb24 minansi_yuv.png and then comparing the way miniansi_yuv.png looks to the actual playback in a vlc window. From velotiaray at gmail.com Sun Nov 16 15:45:28 2014 From: velotiaray at gmail.com (Velotiaray Toto-Zarasoa) Date: Sun, 16 Nov 2014 15:45:28 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <54689C0A.7040307@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> Message-ID: <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> The documentation is very well written, and gives some use examples. I personnally use this as a capture tool http://www.ffmpeg.org/ffmpeg-all.html#gdigrab Velotiaray Toto-Zarasoa ------------------------------- > Many thanks for your reply, Velotiaray. > > Which would be the option? From radpopl at gmail.com Sun Nov 16 18:37:41 2014 From: radpopl at gmail.com (radpopl) Date: Sun, 16 Nov 2014 09:37:41 -0800 (PST) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: I added: ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k "%%~dF%%~pF%%~nF.avi" but ffmpeg crashed. So i tried -c:v libxvid but not worked (libxvid doesn't encode interlaced video :/) so finally I revert to xvid but remove +qpel filter. It seems that the combination of +qpel+ildct+ilme crashes ffmpeg (with xvid encoder). Thanks and regards, rp On Sun, Nov 16, 2014 at 1:53 PM, Francois Visagie [via FFmpeg-users] < ml-node+s933282n4668133h93 at n4.nabble.com> wrote: > > -----Original Message----- > > From: [hidden email] > [mailto: > ffmpeg-user- > > [hidden email] ] > On Behalf Of radpopl > > Sent: 15 November 2014 14:39 > > To: [hidden email] > > > Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced > > > > Hello, > > > > I have a mts file (AVC, 1440x1080i, TFF). > > > > First attempt: > > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid -flags +qpel > -acodec > > libmp3lame -b:a 320k "%%~dF%%~pF%%~nF.avi" > > > > But destination file has an interlace artifacts, unacceptable... > > > > I have tried some command but nothing works, I can't even find good > > solution on web/ffmpeg manual. > > Can somebody help me which filters/options I should use to get XVID > > interlaced video? > > You need to add ' -flags +ildct+ilme'. > > With -c:v libx264 you also can (should) add the bff/tff option; although > personally not familiar with ffmpeg XVID encoding myself you might want to > look for something similar. > > > > > Or maybe I should deinterlace mts file to 1440x540p 50fps file and > convert > to > > > progressive xvid? > > > > > > > > > > -- > > View this message in context: http://ffmpeg- > > users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced- > > tp4668119.html > > Sent from the FFmpeg-users mailing list archive at Nabble.com. > > _______________________________________________ > > ffmpeg-user mailing list > > [hidden email] > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > ffmpeg-user mailing list > [hidden email] > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668133.html > To unsubscribe from H264 (mts) interlaced to XVID interlaced, click here > > . > NAML > > -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668137.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From qshun at live.com Sun Nov 16 07:10:48 2014 From: qshun at live.com (=?gb2312?B?s6PH7Muz?=) Date: Sun, 16 Nov 2014 14:10:48 +0800 Subject: [FFmpeg-user] help me build FFmpeg-Android Message-ID: the error when i run the FFmpeg-Android.sh as follow.Thank you -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 245491 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ???? 2014-11-16 ??2.02.51.png Type: image/png Size: 269076 bytes Desc: not available URL: From qshun at live.com Sun Nov 16 09:27:40 2014 From: qshun at live.com (=?gb2312?B?s6PH7Muz?=) Date: Sun, 16 Nov 2014 16:27:40 +0800 Subject: [FFmpeg-user] help me build FFmpeg-Android In-Reply-To: References: Message-ID: A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 103464 bytes Desc: not available URL: From oinos at web.de Sun Nov 16 21:03:15 2014 From: oinos at web.de (=?windows-1252?Q?Pablo_Rodr=EDguez?=) Date: Sun, 16 Nov 2014 21:03:15 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> Message-ID: <54690303.6030401@web.de> On 11/16/2014 03:45 PM, Velotiaray Toto-Zarasoa wrote: > The documentation is very well written, and gives some use examples. > I personnally use this as a capture tool > > http://www.ffmpeg.org/ffmpeg-all.html#gdigrab Many thanks for your reply, Velotiaray. >From what I read in the documentation I wonder whether this settings would work to capture both video (with mouse) and sound from mic. ffmpeg -f gdigrab -framerate 6 -i desktop -f dshow -i audio="Microphone" -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.flv Sorry for asking basic questions, I only use Windows at work. Many thanks for your help, Pablo -- http://www.ousia.tk From cehoyos at ag.or.at Sun Nov 16 22:41:25 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 21:41:25 +0000 (UTC) Subject: [FFmpeg-user] help me build FFmpeg-Android References: Message-ID: ??? live.com> writes: > the error when i run the FFmpeg-Android.sh I don't think this script is part of the FFmpeg sources. Please understand that only unpatched FFmpeg, running our configure script and make is supported on this mailing list. If you believe that our configure is missing a needed feature, please tell us. Carl Eugen From adf.lists at gmail.com Sun Nov 16 22:41:40 2014 From: adf.lists at gmail.com (Andy Furniss) Date: Sun, 16 Nov 2014 21:41:40 +0000 Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly In-Reply-To: <5468B498.2060901@rabbit.us> References: <54689A5C.8030200@rabbit.us> <5468AD4C.9040300@gmail.com> <5468B498.2060901@rabbit.us> Message-ID: <54691A14.101@gmail.com> Peter Rabbitson wrote: > I am replying in-line further down, but wanted to to ask a generic > question, just to make sure I fully understand what is actually going > on: > > In 1997 Intel came up with the MMX instruction set which among other > things allows in-hardware rgb->422/420 conversion. Since then all > players default to this on a virtually hardcoded level, and the GUI > subsystem is expected to convert everything back to rgb (using the > very same instruction set). Though it was advertised for multimedia and SIMD is handy I don't believe it's why yuv is used. That's from broadcast for bandwidth reasons and given that practically all "real" video is yuv then it needs converting to rgb if that's what the display takes. Using CPU with or without SIMD to do yuv -> rgb is typically last resort of players as for a long time graphics cards/chips have been able to do it. > As a result of the MMX success, everything in the digital video world > *17 years* later is inescapably doing this rgb->yuv->rgb dance, even > in the case of 100% correct and complete metadata. This is true for > all cases including the situation when both the source and the > destination are operating in the RGB24 colorspace: CGI produced under > an RGB renderer, to be displayed in a web-browser player, which by > definition operates on an RGB device. > > Did I get the state of the art about right? I think the MMX bit is a distraction. It's not impossible for players to do RGB direct, but you are using mpeg codecs which were really made for real video/broadcast which is still yuv. The example of CGI straight to browser is probably not typical, but may become more so - it's possible that solutions already exist. Normal CGI as used by broadcast whether overlays or films would be made so as not to expose subsampling artifacts - eg not using 100% colours which suffer most AIUI (something to do with working with gamma corrected vs linear light in addition to the actual res loss). > You are actually correct - the conversion that took place is 422, not > 420 as I originally thought. +1 to vlc some players could go straight for 420 as the lowest common denominator. You can't say that everyone would see the same result though, as it depends on the hardware/set up of the users machine. From cehoyos at ag.or.at Sun Nov 16 22:42:41 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 21:42:41 +0000 (UTC) Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly References: <54689A5C.8030200@rabbit.us> Message-ID: Peter Rabbitson rabbit.us> writes: > However playback of the generated video and original/extracted > images show a noticeable color loss when displayed side by side Please test mplayer -vo gl And please use -vcodec libx264rgb for best quality. Carl Eugen From cehoyos at ag.or.at Sun Nov 16 22:51:48 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 21:51:48 +0000 (UTC) Subject: [FFmpeg-user] Unable to produce a yuv444p video that vlc will play back correctly References: <54689A5C.8030200@rabbit.us> <5468AD4C.9040300@gmail.com> <5468B498.2060901@rabbit.us> Message-ID: Peter Rabbitson rabbit.us> writes: > As a result of the MMX success, everything in the > digital video world *17 years* later is inescapably > doing this rgb->yuv->rgb dance, even in the case of > 100% correct and complete metadata. I don't think your colour issue has anything to do with MMX. All codecs (that have any relevance) only support yuv420p in the variants supported by the "free" (as in beer) multimedia players, so it makes sense for FFmopeg to default to yuv420p if nothing else was specified and it makes sense for video drivers to only support yuv420p. Note that even vlc has / had issues with h264 rgb. Carl Eugen From cehoyos at ag.or.at Sun Nov 16 22:54:30 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 21:54:30 +0000 (UTC) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: radpopl gmail.com> writes: > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > "%%~dF%%~pF%%~nF.avi" > > but ffmpeg crashed. Command line, console output etc. missing. Please understand that crashes are always important but we can only fix them if we are able to reproduce them... > > > But destination file has an interlace artifacts, > > > unacceptable... If you don't want interlacing artefacts in your output file but your input file is interlaced, you need to add a deinterlace filter to your command line, please test -vf yadif. Please don't top-post here, it is considered rude, Carl Eugen From cehoyos at ag.or.at Sun Nov 16 22:56:35 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 16 Nov 2014 21:56:35 +0000 (UTC) Subject: [FFmpeg-user] screencast in WinXP References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> Message-ID: Pablo Rodr?guez web.de> writes: > From what I read in the documentation I wonder whether > this settings would work to capture both video (with > mouse) and sound from mic. The console output will tell you: If the start times are wallclock, audio and video will be (should be) in-sync, if not this is hard / impossible. Carl Eugen From velotiaray at gmail.com Sun Nov 16 23:09:47 2014 From: velotiaray at gmail.com (Velotiaray Toto-Zarasoa) Date: Sun, 16 Nov 2014 23:09:47 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <54690303.6030401@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> Message-ID: <5D39B530-3A81-4723-8408-9E1609A2DD42@gmail.com> It is very basic command but it should work (I have not tested) Velotiaray Toto-Zarasoa ------------------------------- > Le 16 nov. 2014 ? 21:03, Pablo Rodr?guez a ?crit : > >> On 11/16/2014 03:45 PM, Velotiaray Toto-Zarasoa wrote: >> The documentation is very well written, and gives some use examples. >> I personnally use this as a capture tool >> >> http://www.ffmpeg.org/ffmpeg-all.html#gdigrab > > Many thanks for your reply, Velotiaray. > > From what I read in the documentation I wonder whether this settings > would work to capture both video (with mouse) and sound from mic. > > ffmpeg -f gdigrab -framerate 6 -i desktop -f dshow -i audio="Microphone" > -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.flv > > Sorry for asking basic questions, I only use Windows at work. > > Many thanks for your help, > > > Pablo > -- > http://www.ousia.tk > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bart.gopnik at gmail.com Mon Nov 17 06:46:31 2014 From: bart.gopnik at gmail.com (=?UTF-8?B?0JHQsNGA0YIg0JPQvtC/0L3QuNC6?=) Date: Mon, 17 Nov 2014 08:46:31 +0300 Subject: [FFmpeg-user] help me build FFmpeg-Android In-Reply-To: References: Message-ID: 2014-11-17 0:41 GMT+03:00 Carl Eugen Hoyos : > ??? live.com> writes: > >> the error when i run the FFmpeg-Android.sh > > I don't think this script is part of the FFmpeg > sources. > Please understand that only unpatched FFmpeg, > running our configure script and make is > supported on this mailing list. > > If you believe that our configure is missing a > needed feature, please tell us. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user http://vinsol.com/blog/2014/07/30/cross-compiling-ffmpeg-with-x264-for-android/ ;-) From peter_trompeter at hotmail.com Mon Nov 17 12:16:54 2014 From: peter_trompeter at hotmail.com (Peter Trompeter) Date: Sun, 17 Nov 2014 11:16:54 +0000 Subject: [FFmpeg-user] =?iso-8859-1?q?FW=3Ainfoe?= Message-ID: <79C0BFB5A1BA8AAE4833DEBCCEBA7553@radekzdonczyk.com> http://ortakoytarim.com.tr/ybcz/thrxcfyeapvlfllfm.wuvrevjhvsmid Peter Trompeter From maziar.mehrabi at gmail.com Mon Nov 17 11:58:50 2014 From: maziar.mehrabi at gmail.com (Maziar Mehrabi) Date: Mon, 17 Nov 2014 12:58:50 +0200 Subject: [FFmpeg-user] FW:infoe In-Reply-To: <79C0BFB5A1BA8AAE4833DEBCCEBA7553@radekzdonczyk.com> References: <79C0BFB5A1BA8AAE4833DEBCCEBA7553@radekzdonczyk.com> Message-ID: This is very suspicious, please don't open without caution. -- H?lsningar, Maziar Mehrabi, On Mon, Nov 17, 2014 at 1:16 PM, Peter Trompeter < peter_trompeter at hotmail.com> wrote: > http://ortakoytarim.com.tr/ybcz/thrxcfyeapvlfllfm.wuvrevjhvsmid > > > > > > > Peter Trompeter > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From p.rennert at cs.ucl.ac.uk Mon Nov 17 12:51:47 2014 From: p.rennert at cs.ucl.ac.uk (Peter Rennert) Date: Mon, 17 Nov 2014 11:51:47 +0000 Subject: [FFmpeg-user] Compiling ffmpeg to be portable Message-ID: Hi, I am trying to compile ffmpeg on OS X so that it is portable. To be precise, after compilation, all references within the library should be relative, rather than absolute. My current naive approach was to compile ffmpeg with homebrew and try to copy the files over where I need them. This does not work for files like libavformat, which references libavcodec with its absolute, rather than relative path, which then leads to import errors like Library not loaded: /usr/local/Cellar/ffmpeg/2.4.2/lib/libavcodec.56.dylib Referenced from: /Users/peter/anaconda/lib/libavformat.56.4.101.dylib (/Users/peter/anaconda/lib/libavcodec.56.dylib exists. I deleted /usr/local/Cellar/ffmpeg/2.4.2/, so its not a PATH issue) Is there anything I can setup in the configure file to compile ffmpeg in portable mode? Thanks, Peter From velotiaray at gmail.com Mon Nov 17 12:53:53 2014 From: velotiaray at gmail.com (Velotiaray Toto-Zarasoa) Date: Mon, 17 Nov 2014 12:53:53 +0100 Subject: [FFmpeg-user] Compiling ffmpeg to be portable In-Reply-To: References: Message-ID: <49B44A66-B1A3-4198-962B-236FBE65A05B@gmail.com> You should compile every dependencies in static mode (bigger files). Velotiaray Toto-Zarasoa From francois.visagie at gmail.com Mon Nov 17 13:30:38 2014 From: francois.visagie at gmail.com (Francois Visagie) Date: Mon, 17 Nov 2014 14:30:38 +0200 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: On 16 November 2014 23:54, Carl Eugen Hoyos wrote: > radpopl gmail.com> writes: > >> ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid >> -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k >> "%%~dF%%~pF%%~nF.avi" >> >> but ffmpeg crashed. > > Command line, console output etc. missing. > > Please understand that crashes are always > important but we can only fix them if we > are able to reproduce them... > >> > > But destination file has an interlace artifacts, >> > > unacceptable... I also just realised that the advice I gave you was wrong. That was aimed at how to properly make interlaced encodings, which in fact is what you are trying to get away from. Sorry for that, and Carl's advice for adding (only) a deinterlacing filter is therefore spot-on. > > If you don't want interlacing artefacts in your output > file but your input file is interlaced, you need to > add a deinterlace filter to your command line, please > test -vf yadif. > > Please don't top-post here, it is considered rude, > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From oinos at web.de Mon Nov 17 19:22:14 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Mon, 17 Nov 2014 19:22:14 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> Message-ID: <546A3CD6.3030609@web.de> On 11/16/2014 10:56 PM, Carl Eugen Hoyos wrote: > Pablo Rodr?guez web.de> writes: > >> From what I read in the documentation I wonder whether >> this settings would work to capture both video (with >> mouse) and sound from mic. > > The console output will tell you: > If the start times are wallclock, audio and video will > be (should be) in-sync, if not this is hard / impossible. Many thanks for your reply, Carl Eugen. Besides from the fact that audio device wasn?t detected (but I forgot to copy the complete output), from your message I wonder whether I should use_wallclock_as_timestamps as an option (sorry, but this is all Greek to me). Which value should it have in an old and slow computer? (I mean, when the option is required to keep synced audio and video.) Many thanks for your help, Pablo -- http://www.ousia.tk From cehoyos at ag.or.at Mon Nov 17 20:06:01 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 17 Nov 2014 19:06:01 +0000 (UTC) Subject: [FFmpeg-user] screencast in WinXP References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> Message-ID: Pablo Rodr?guez web.de> writes: > but I forgot to copy the complete output Please do so if you ask for support on this mailing list. Carl Eugen From jeremy at ZeeVee.Com Mon Nov 17 21:15:47 2014 From: jeremy at ZeeVee.Com (Jeremy Greene) Date: Mon, 17 Nov 2014 20:15:47 +0000 Subject: [FFmpeg-user] ts discontinuity indicator In-Reply-To: References: Message-ID: It appears that ffmpeg does not fully support the discontinuity indicator (DI) in the afc. In libavformat/mpegts.c::handle_packet(), if the DI is set, it does ignore discontinuity in the continuity counter. But it should also ignore a discontinuity in the PCR (and PTS/DTS). Is this a known issue? I see that the default for ffplay is to ignore the pcr, but if the afc discontinuity indicator is set, then the pts and dts should also be allowed a discontinuity. Jeremy From suri at baymicrosystems.com Mon Nov 17 21:27:23 2014 From: suri at baymicrosystems.com (Suri Shelvapille) Date: Mon, 17 Nov 2014 20:27:23 +0000 Subject: [FFmpeg-user] ffmpeg usage! Message-ID: Dear Folks: I am using ffmpeg to create a video mosaic, the overlay is working fine. I have a question though: If one of the video streams does not exist, how can I tell ffmpeg to use a default video stream? Right now, if the input file does not exist, ffmpeg exits. Thanks, Suri From radpopl at gmail.com Mon Nov 17 22:50:09 2014 From: radpopl at gmail.com (=?UTF-8?B?UmFkb3PFgmF3IFBvcMWCYXdza2k=?=) Date: Mon, 17 Nov 2014 22:50:09 +0100 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: On Sun, Nov 16, 2014 at 10:54 PM, Carl Eugen Hoyos wrote: > radpopl gmail.com> writes: > > > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > > "%%~dF%%~pF%%~nF.avi" > > > > but ffmpeg crashed. > > Command line, console output etc. missing. > > Please understand that crashes are always > important but we can only fix them if we > are able to reproduce them... > There is a screenshot: http://i61.tinypic.com/jfkpkx.jpg Regards, rp From radpopl at gmail.com Mon Nov 17 22:55:49 2014 From: radpopl at gmail.com (=?UTF-8?B?UmFkb3PFgmF3IFBvcMWCYXdza2k=?=) Date: Mon, 17 Nov 2014 22:55:49 +0100 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: On Mon, Nov 17, 2014 at 1:30 PM, Francois Visagie < francois.visagie at gmail.com> wrote: > I also just realised that the advice I gave you was wrong. That was > aimed at how to properly make interlaced encodings, which in fact is > what you are trying to get away from. Sorry for that, and Carl's > advice for adding (only) a deinterlacing filter is therefore spot-on. > No, is was very OK, I want to make interlaced encodings. Thanks :). I prefer to make interlaced encoding, because 25 interlaced frames is better (smooth motion) than 25p, so I prefer to stick with it. Regards, rp From rogerdpack2 at gmail.com Mon Nov 17 23:45:25 2014 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 17 Nov 2014 15:45:25 -0700 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <546A3CD6.3030609@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> Message-ID: On Mon, Nov 17, 2014 at 11:22 AM, Pablo Rodr?guez wrote: > On 11/16/2014 10:56 PM, Carl Eugen Hoyos wrote: > > Pablo Rodr?guez web.de> writes: > > > >> From what I read in the documentation I wonder whether > >> this settings would work to capture both video (with > >> mouse) and sound from mic. > > > > The console output will tell you: > > If the start times are wallclock, audio and video will > > be (should be) in-sync, if not this is hard / impossible. > > Many thanks for your reply, Carl Eugen. > > Besides from the fact that audio device wasn?t detected (but I forgot to > copy the complete output), from your message I wonder whether I should > use_wallclock_as_timestamps as an option (sorry, but this is all Greek > to me). > > Which value should it have in an old and slow computer? (I mean, when > the option is required to keep synced audio and video.) > > AFAIK it shouldn't matter which option you use. > Many thanks for your help, > > > Pablo > -- > http://www.ousia.tk > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From bryanfieldelliot at gmail.com Tue Nov 18 06:05:34 2014 From: bryanfieldelliot at gmail.com (Bryan Field-Elliot) Date: Mon, 17 Nov 2014 21:05:34 -0800 Subject: [FFmpeg-user] h264 file: mplayer plays it, but ffmpeg can't decode it Message-ID: I have a h264 file, which I obtained by capturing bytes streaming (mid-stream) from an IP camera. mplayer has no trouble playing the file on-screen. However, I cannot seem to get ffmpeg to recognize and decode the file (or re-encode it into something else), no matter what combination of command line parameters I try. Example output from ffmpeg is below. As I said, these are h264 bytes captured mid-stream, not necessarily on the key frame boundary. mplayer (with no other options) plays the video just fine, which leads me to believe it is essentially valid. But for someone reason ffmpeg can?t figure it out. Help would be appreciated! Thank you. $ ffmpeg ffmpeg version 2.4.3 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 11 2014 11:21:43 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/2.4.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-ffplay --enable-vda libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libavresample 2. 1. 0 / 2. 1. 0 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... Use -h to get full help or, even better, run 'man ffmpeg' $ ffmpeg -f h264 -i vid.h264 -c:v libx264 vid.mp4 (much omitted...) [h264 @ 0x7ffdb982e200] no frame! [h264 @ 0x7ffdb982e200] Missing reference picture, default is 0 [h264 @ 0x7ffdb982e200] decode_slice_header error [h264 @ 0x7ffdb982e200] Missing reference picture, default is 0 [h264 @ 0x7ffdb982e200] decode_slice_header error [h264 @ 0x7ffdb982e200] Missing reference picture, default is 0 [h264 @ 0x7ffdb982e200] decode_slice_header error [h264 @ 0x7ffdb982e200] Missing reference picture, default is 0 [h264 @ 0x7ffdb982e200] decode_slice_header error [h264 @ 0x7ffdb980da00] decoding for stream 0 failed Input #0, h264, from 'vid.h264': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (High), yuvj420p(pc), 640x480 [SAR 1:1 DAR 4:3], 15.33 fps, 15 tbr, 1200k tbn, 30 tbc No pixel format specified, yuvj420p for H.264 encoding chosen. Use -pix_fmt yuv420p for compatibility with outdated media players. [libx264 @ 0x7ffdba012800] using SAR=1/1 [libx264 @ 0x7ffdba012800] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7ffdba012800] profile High, level 2.2 [libx264 @ 0x7ffdba012800] 264 - core 142 r2455 021c0dc - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=15 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 'vid.mp4': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuvj420p, 640x480 [SAR 1:1 DAR 4:3], q=-1--1, 15 fps, 15360 tbn, 15 tbc Metadata: encoder : Lavc56.1.100 libx264 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Press [q] to stop, [?] for help [h264 @ 0x7ffdba524800] Cannot use next picture in error concealment [h264 @ 0x7ffdba524800] concealing 880 DC, 880 AC, 880 MV errors in P frame frame= 0 fps=0.0 q=0.0 Lsize= 0kB time=00:00:00.00 bitrate=N/A video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown The resulting file, vid.mp4, is only 261 bytes long and (obviously) doesn?t play. From perera.amila at gmail.com Tue Nov 18 06:07:29 2014 From: perera.amila at gmail.com (Amila Perera) Date: Tue, 18 Nov 2014 14:07:29 +0900 Subject: [FFmpeg-user] construct mov from a fragmented mp4 stream. Message-ID: Hi all, I have a fragmented mp4(H.264) stream dumped to file. It has several atoms of the following sequence. moof mfhd free traf thfd trun mdat moof : : How can I create a MOV file out of these fragmented moof atoms. I think that I have to extract H.264 data from moof atoms and dump them in to moov atoms of a MOV file. Is this possible with ffmpeg. Actually I need to achieve this programmatically, but first I want to know if this is possible with ffmpeg. Thank you in advance. -- *Amila Perera.* From xanadu at apost.plala.or.jp Tue Nov 18 10:11:19 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Tue, 18 Nov 2014 18:11:19 +0900 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items Message-ID: Hi list members! I'm trying basic Movie playback with ffplay. When I execute the following command, I got the warning "Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items". What this waring mean and how should it be? I have searched ffmpeg documentation on the net but I can't get the answer! $ ffplay -i ~/Movies/More\ Than\ You\ Know.mp4 -af "pan=stereo:FL Message-ID: Kimio Miyamura apost.plala.or.jp> writes: > When I execute the following command, I got the > warning "Parsed_pan_0 This syntax is deprecated. > Use '|' to separate the list items". > What this waring mean and how should it be? Use "|" instead of ":" to silence the warning. Just to make sure: You do know that this is not the best way to downmix, you are using the pan filter because you know exactly what you are doing? Carl Eugen From cehoyos at ag.or.at Tue Nov 18 10:34:19 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 09:34:19 +0000 (UTC) Subject: [FFmpeg-user] h264 file: mplayer plays it, but ffmpeg can't decode it References: Message-ID: Bryan Field-Elliot gmail.com> writes: > mplayer has no trouble playing the file on-screen. > However, I cannot seem to get ffmpeg to recognize > and decode the file Please provide the sample. Carl Eugen From cehoyos at ag.or.at Tue Nov 18 10:33:33 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 09:33:33 +0000 (UTC) Subject: [FFmpeg-user] construct mov from a fragmented mp4 stream. References: Message-ID: Amila Perera gmail.com> writes: > I have a fragmented mp4(H.264) stream dumped to file. > Is this possible with ffmpeg. > Actually I need to achieve this programmatically Please test if ffmpeg (the application) can read the fragments. Carl Eugen From xanadu at apost.plala.or.jp Tue Nov 18 12:08:46 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Tue, 18 Nov 2014 20:08:46 +0900 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items In-Reply-To: References: Message-ID: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> 2014/11/18 18:32, Carl Eugen Hoyos ag.or.at> wrote: > Kimio Miyamura apost.plala.or.jp> writes: > >> What this waring mean and how should it be? > > Use "|" instead of ":" to silence the warning. Thanks Carl. I have changed the command and it works without waring. ffplay -i ~/Movies/More\ Than\ You\ Know.mp4 -af "pan=stereo|FL Just to make sure: You do know that this is not the > best way to downmix, you are using the pan filter > because you know exactly what you are doing? Well, I have got use pan filter just a result of try and error. https://trac.ffmpeg.org/wiki/AudioChannelManipulation seems to suggest me to use -ac 2 option, but my 5.1ch mp4 does not play full audio with -ac 2 option... Or is there correct way to downmix audio any other? // Miya From cehoyos at ag.or.at Tue Nov 18 12:39:02 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 11:39:02 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Parsed=5Fpan=5F0_This_syntax_is_deprecate?= =?utf-8?q?d=2E_Use_=27=7C=27_to_separate_the_list_items?= References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> Message-ID: Kimio Miyamura apost.plala.or.jp> writes: > I think online documentation should be fixed. Please consider sending a patch. [...] > Well, I have got use pan filter just a result of try and error. > Or is there correct way to downmix audio any other? If "-ac 2" does not work for you, please discuss it here, I don't think the pan filter does downmixing in an acceptable way. Carl Eugen From perera.amila at gmail.com Tue Nov 18 13:00:39 2014 From: perera.amila at gmail.com (Amila Perera) Date: Tue, 18 Nov 2014 21:00:39 +0900 Subject: [FFmpeg-user] construct mov from a fragmented mp4 stream. In-Reply-To: References: Message-ID: > > Please test if ffmpeg (the application) can read > the fragments. > A simple "ffmpeg -i " command fails with the following error. [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000042e1e60] could not find corresponding trex [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000042e1e60] error reading header samples\fragMP4.mp4: Invalid data found when processing input Is there any specific way to achieve this in ffmpeg. On Tue, Nov 18, 2014 at 6:33 PM, Carl Eugen Hoyos wrote: > Amila Perera gmail.com> writes: > > > I have a fragmented mp4(H.264) stream dumped to file. > > > Is this possible with ffmpeg. > > Actually I need to achieve this programmatically > > Please test if ffmpeg (the application) can read > the fragments. > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- *Amila Perera.* From barsnick at gmx.net Tue Nov 18 13:44:46 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 18 Nov 2014 13:44:46 +0100 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items In-Reply-To: References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> Message-ID: <20141118124446.GA22910@sunshine.barsnick.net> On Tue, Nov 18, 2014 at 11:39:02 +0000, Carl Eugen Hoyos wrote: > > I think online documentation should be fixed. > Please consider sending a patch. I have had one lying around here for the last couple of weeks - ever since I got annoyed by that warning. (See some other thread here.) I don't recall why I didn't submit it. I think I was annoyed by the way pan handles arguments, but that can't be fixed without breaking backwards compatibility. Moritz From radpopl at gmail.com Tue Nov 18 14:24:48 2014 From: radpopl at gmail.com (=?UTF-8?B?UmFkb3PFgmF3IFBvcMWCYXdza2k=?=) Date: Tue, 18 Nov 2014 14:24:48 +0100 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: <546B48A0.1020306@gmail.com> W dniu 2014-11-17 22:50, Rados?aw Pop?awski pisze: > On Sun, Nov 16, 2014 at 10:54 PM, Carl Eugen Hoyos > wrote: > > radpopl gmail.com > writes: > > > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > > "%%~dF%%~pF%%~nF.avi" > > > > but ffmpeg crashed. > > Command line, console output etc. missing. > > Please understand that crashes are always > important but we can only fix them if we > are able to reproduce them... > > > There is a screenshot: > http://i61.tinypic.com/jfkpkx.jpg > > Regards, > rp I should add that removing +qpel fixes the problem. From maillist35650 at gmail.com Tue Nov 18 14:54:41 2014 From: maillist35650 at gmail.com (mail list) Date: Tue, 18 Nov 2014 14:54:41 +0100 Subject: [FFmpeg-user] how to use overlay + amix Message-ID: Hello, Is someone can help me? I'm trying to make one short video from an other video by taking 2 extracts with synchro audios and crossfading effect. I tried several ways and finally the best result is as following: ./ffmpeg -loglevel info -y -f lavfi -r 30 -i color=black at 1 \ -ss 30 -i /1024/video/rush.mp4 \ -ss 1220 -i /1024/video/rush.mp4 \ -filter_complex " \ [0:v]scale=640x480[bg]; \ [1:v]scale=640x480,fade=t=in:st=0:d=3:alpha=1,fade=t=out:st=27:d=3:alpha=1,setpts=PTS-STARTPTS[vout1] ; \ [2:v]scale=640x480,fade=t=in:st=0:d=3:alpha=1,setpts=PTS-STARTPTS+27/TB [vout2] ; \ [bg][vout1]overlay[clip2];[clip2][vout2]overlay, trim=duration=60[vclip3] ;\ [1:a]atrim=start=0:end=30, asetpts=PTS-STARTPTS [aout1]; \ [2:a]atrim=start=0:end=33, adelay=27000|27000, asetpts=PTS-STARTPTS+27/TB [aout2]; \ [aout2][aout1]amix=inputs=2:duration=longest, asetpts=PTS-STARTPTS+30/TB[aclip] \ " -strict -2 -c:v h264 -b:a 96k -map [vclip3] -map [aclip] result.mp4 "PTS-STARTPTS+30/TB" is the only way I found to avoid dopping frames. Any idea to get the right command line? rgds From xanadu at apost.plala.or.jp Tue Nov 18 15:34:27 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Tue, 18 Nov 2014 23:34:27 +0900 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items In-Reply-To: References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> Message-ID: 2014/11/18 20:39, Carl Eugen Hoyos ag.or.at> wrote: >> I think online documentation should be fixed. > > Please consider sending a patch. Patch for documentation? Just replace ":" with "|"? If I can write documentation, based on Moritz's comment, there seems to be backward compatibility right? Does it mean the latest documentation can't be provided for ffmpeg? >> Well, I have got use pan filter just a result of try and error. > >> Or is there correct way to downmix audio any other? > > If "-ac 2" does not work for you, please discuss it here, > I don't think the pan filter does downmixing in an > acceptable way. Should I open new discussion about "-ac 2"? Or continue with this thread? Briefly, the source is DVD dumped with mplayer and ffmpeg down-mix audio with "-ac 2", but if didn't down-mix with ffmpeg, ffplay seems to need pan filter to playback. Using "-ac 2" with ffplay dose not help. // Miya From bryanfieldelliot at gmail.com Tue Nov 18 15:44:00 2014 From: bryanfieldelliot at gmail.com (Bryan Field-Elliot) Date: Tue, 18 Nov 2014 06:44:00 -0800 Subject: [FFmpeg-user] h264 file: mplayer plays it, but ffmpeg can't decode it In-Reply-To: References: Message-ID: <378263CD-8DB0-4792-884D-5ACE5F6C014B@gmail.com> Thanks Carl, The h264 file can be downloaded here: https://s3.amazonaws.com/bryanfe/vid.h264 So as to clear any confusion if you watch it ? it is just a camera pointed down at a wood desk, with a red light flashing in the background (not very exciting). On Nov 18, 2014, at 1:34 AM, Carl Eugen Hoyos wrote: Bryan Field-Elliot gmail.com> writes: > mplayer has no trouble playing the file on-screen. > However, I cannot seem to get ffmpeg to recognize > and decode the file Please provide the sample. Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From barsnick at gmx.net Tue Nov 18 16:03:01 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 18 Nov 2014 16:03:01 +0100 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items In-Reply-To: References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> Message-ID: <20141118150301.GH22910@sunshine.barsnick.net> On Tue, Nov 18, 2014 at 23:34:27 +0900, Kimio Miyamura wrote: > > Please consider sending a patch. > Patch for documentation? Just replace ":" with "|"? That's basically it. See attached. Too lazy to format correctly and properly send to the correct list right now, with comments. Might do so later. > If I can write documentation, based on Moritz's comment, there seems > to be backward compatibility right? No, my comment regarding backward compatibility was based on something else. Doc can be fixed to reflect current behavior, period. Attached. :-) > Should I open new discussion about "-ac 2"? Or continue with this thread? > > Briefly, the source is DVD dumped with mplayer and ffmpeg down-mix > audio with "-ac 2", but if didn't down-mix with ffmpeg, ffplay seems > to need pan filter to playback. Using "-ac 2" with ffplay dose not > help. If you ask me: Separate thread. ;-) Moritz -------------- next part -------------- diff --git a/doc/filters.texi b/doc/filters.texi index c70ddf3..661df38 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1714,7 +1714,7 @@ This filter is also designed to remap efficiently the channels of an audio stream. The filter accepts parameters of the form: -"@var{l}:@var{outdef}:@var{outdef}:..." +"@var{l}|@var{outdef}|@var{outdef}|..." @table @option @item l @@ -1724,6 +1724,7 @@ output channel layout or number of channels output channel specification, of the form: "@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]" + at table @option @item out_name output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.) @@ -1735,6 +1736,7 @@ multiplicative coefficient for the channel, 1 leaving the volume unchanged input channel to use, see out_name for details; it is not possible to mix named and numbered input channels @end table + at end table If the `=' in a channel specification is replaced by `<', then the gains for that specification will be renormalized so that the total is 1, thus @@ -1745,13 +1747,13 @@ avoiding clipping noise. For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel: @example -pan=1:c0=0.9*c0+0.1*c1 +pan=1|c0=0.9*c0+0.1*c1 @end example A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround: @example -pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR +pan=stereo | FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR @end example Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system @@ -1774,25 +1776,25 @@ remapping. For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels: @example -pan="stereo: c0=FL : c1=FR" +pan="stereo | c0=FL | c1=FR" @end example Given the same source, you can also switch front left and front right channels and keep the input channel layout: @example -pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5" +pan="5.1 | c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5" @end example If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with: @example -pan="stereo:c1=c1" +pan="stereo|c1=c1" @end example Still with a stereo audio stream input, you can copy the right channel in both front left and right: @example -pan="stereo: c0=FR : c1=FR" +pan="stereo | c0=FR | c1=FR" @end example @section replaygain From xanadu at apost.plala.or.jp Tue Nov 18 16:32:34 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Wed, 19 Nov 2014 00:32:34 +0900 Subject: [FFmpeg-user] Parsed_pan_0 This syntax is deprecated. Use '|' to separate the list items In-Reply-To: <20141118150301.GH22910@sunshine.barsnick.net> References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> <20141118150301.GH22910@sunshine.barsnick.net> Message-ID: <464EFF81-FEA7-4463-A659-68C5FC56B013@apost.plala.or.jp> 2014/11/19 0:03, Moritz Barsnick gmx.net> wrote: > On Tue, Nov 18, 2014 at 23:34:27 +0900, Kimio Miyamura wrote: >>> Please consider sending a patch. >> Patch for documentation? Just replace ":" with "|"? > > That's basically it. See attached. Too lazy to format correctly and > properly send to the correct list right now, with comments. Might do so > later. I didn't know which file should be fixed. That is included in ffmpeg source code!! Oh, by the way thanks for the patch. >> Should I open new discussion about "-ac 2"? Or continue with this thread? > If you ask me: Separate thread. ;-) OK. I'll do so. // Miya From oinos at web.de Tue Nov 18 17:42:12 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Tue, 18 Nov 2014 17:42:12 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> Message-ID: <546B76E4.2020605@web.de> On 11/17/2014 08:06 PM, Carl Eugen Hoyos wrote: > Pablo Rodr?guez web.de> writes: > >> but I forgot to copy the complete output > > Please do so if you ask for support on this > mailing list. This is the complete error message: ffmpeg -f gdigrab -framerate 6 -i desktop -f dshow -i audio="Microphone" -vcodec libx264 -crf 0 -p reset ultrafast -acodec pcm_s16le output.flv ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [gdigrab @ 02ddc020] Capturing whole desktop as 1280x1024x32 at (0,0) Input #0, gdigrab, from 'desktop': Duration: N/A, start: 1416293523.415939, bitrate: 251660 kb/s Stream #0:0: Video: bmp, bgra, 1280x1024, 251660 kb/s, 6 tbr, 1000k tbn, 6 tbc [dshow @ 02ddf800] Could not find audio device. audio=Microphone: Input/output error This is all Greek to me. But I don?t understand why the same machine was able to detect it before (https://ffmpeg.org/pipermail/ffmpeg-user/2014-November/024230.html). Many thanks for your help, Pablo -- http://www.ousia.tk From rogerdpack2 at gmail.com Tue Nov 18 18:28:57 2014 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 18 Nov 2014 10:28:57 -0700 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <546B76E4.2020605@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> <546B76E4.2020605@web.de> Message-ID: On Tue, Nov 18, 2014 at 9:42 AM, Pablo Rodr?guez wrote: > On 11/17/2014 08:06 PM, Carl Eugen Hoyos wrote: > > Pablo Rodr?guez web.de> writes: > > > >> but I forgot to copy the complete output > > > > Please do so if you ask for support on this > > mailing list. > > This is the complete error message: > > ffmpeg -f gdigrab -framerate 6 -i desktop -f dshow -i audio="Microphone" > -vcodec libx264 -crf 0 -p reset ultrafast -acodec pcm_s16le output.flv > ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg > developers > built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r > --enable-gnutls --enable-iconv --enable-libass --enable-libbluray > --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme > --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libopenjpeg --enable-libopus --enable-librtmp > --enable-libschroedinger --enable-libsoxr --enable-libspeex > --enable-libtheora --enable-libtwolame --enable-libvidstab > --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis > --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 > --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink > --enable-zlib > libavutil 54. 7.100 / 54. 7.100 > libavcodec 56. 1.100 / 56. 1.100 > libavformat 56. 4.101 / 56. 4.101 > libavdevice 56. 0.100 / 56. 0.100 > libavfilter 5. 1.100 / 5. 1.100 > libswscale 3. 0.100 / 3. 0.100 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 0.100 / 53. 0.100 > [gdigrab @ 02ddc020] Capturing whole desktop as 1280x1024x32 at (0,0) > Input #0, gdigrab, from 'desktop': > Duration: N/A, start: 1416293523.415939, bitrate: 251660 kb/s > Stream #0:0: Video: bmp, bgra, 1280x1024, 251660 kb/s, 6 tbr, 1000k > tbn, 6 tbc > [dshow @ 02ddf800] Could not find audio device. > audio=Microphone: Input/output error > > what's your output to ffmpeg -list_devices true -f dshow -i dummy From oinos at web.de Tue Nov 18 18:34:02 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Tue, 18 Nov 2014 18:34:02 +0100 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> <546B76E4.2020605@web.de> Message-ID: <546B830A.2020004@web.de> On 11/18/2014 06:28 PM, Roger Pack wrote: > what's your output to > > ffmpeg -list_devices true -f dshow -i dummy Many thanks for your reply, Roger. The weird thing is that here it is detected: C:>ffmpeg -list_devices true -f dshow -i dummy ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [dshow @ 02dde020] DirectShow video devices [dshow @ 02dde020] Could not enumerate video devices. [dshow @ 02dde020] DirectShow audio devices [dshow @ 02dde020] "Realtek HD Audio Input" [dshow @ 02dde020] "Realtek HD Digital input" dummy: Immediate exit requeste -- http://www.ousia.tk From Eric.Lovelace at msdmail.net Tue Nov 18 18:37:46 2014 From: Eric.Lovelace at msdmail.net (Eric Lovelace) Date: Tue, 18 Nov 2014 17:37:46 +0000 Subject: [FFmpeg-user] FFM moving time segment Message-ID: <5A651CA75E25F940A89A85F2A41F388D8C0A76@EXCHANGE-DB2.msdtech.net> Hello, I am trying to send an rtp multicast of a stream with the configuration below. It works; however, when the max file size is reached (or the disk fills up if that parameter is excluded from ffmpeg) the program ends. The desired behavior would be to just keep a small section of the stream as per the docs the ffm file can "store a moving time segment of an infinite movie or a whole movie." The current behavior confuses me as clients are able to play the sdp file and receive the content from the beginning which seems inconsistent with multicast. How do I make ffmpeg create an FFM with a moving time segment of an infinite movie? Thanks, Eric Command line ffmpeg start: Ffmpeg -i "rtmp://192.168.10.10/push/test" -flags:a +global_header -pixel_format yuv420p -acodec libfdk_aac -vcodec libx264 -fs 2097152 /tmp/1.ffm ffserver.cfg: HTTPPort 8090 RTSPPort 554 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 1000 MaxBandwidth 10000000 MaxClients 200 CustomLog - Format rtp MulticastAddress 224.1.1.17 MulticastPort 5000 MulticastTTL 63 NoLoop VideoCodec libx264 AVOptionAudio flags +global_header AudioCodec libfdk_aac File /tmp/1.ffm Format status ACL allow localhost From suri at baymicrosystems.com Tue Nov 18 20:22:18 2014 From: suri at baymicrosystems.com (Suri Shelvapille) Date: Tue, 18 Nov 2014 19:22:18 +0000 Subject: [FFmpeg-user] Playing videos in a loop! Message-ID: Dear Folks: I am creating a Video mosaic as follows. Unfortunately, after the video ends, everything stops. I would like to play all the videos in a loop. I have tried "-loop 0", "-loop 1" to ffmpeg and it has not helped. I have tried the same on ffplay as well and that has not helped either. Any help would be deeply appreciated. ----------------------------------------------------------------------------------- #!/bin/bash ffmpeg -i a1.avi -i a2.avi -filter_complex "nullsrc=size=320x90 [base];[0:v] setpts=PTS-STARTPTS, scale=160X90 [pos0];[1:v] setpts=PTS-STARTPTS, scale=160X90 [pos1];[base][pos0] overlay=shortest=0 [tmp1];[tmp1][pos1] overlay=shortest=0:x=160" -c:v libx264 -f avi - | ffplay - thanks Suri From ikrananka at hotmail.com Tue Nov 18 20:39:28 2014 From: ikrananka at hotmail.com (Andrew Arthur) Date: Tue, 18 Nov 2014 12:39:28 -0700 Subject: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem Message-ID: I am converting a load of WMV files to M2V (i.e. video conversion only) using the following basic command line: ffmpeg -i input.wmv -codec:v mpeg2video -b:v 8000k -maxrate 10000k output.m2v However, it is imperative in my conversion that the number of frames in the output file be identical to the input file. However I consistently get one less frame in the m2v files compared to the wmv files. Is there an alternative command line that will ensure the total number of frames is honored? To provide a little further info, one example is as follows. The input.wmv file has 1037 frames, however the output m2v file only has 1036 frames: D:\ffmpeg -i input.wmv -codec:v mpeg2video -b:v 8000k -maxrate 10000k output.m2v ffmpeg version N-67742-g3f07dd6 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 16 2014 22:01:52 with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 13.100 / 54. 13.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 13.100 / 56. 13.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, asf, from 'input.wmv': Metadata: VBR Peak : 10000000 DeviceConformanceTemplate: MP at HL WM/WMADRCPeakReference: 23859 WM/WMADRCPeakTarget: 23859 WM/WMADRCAverageReference: 4536 WM/WMADRCAverageTarget: 4536 WMFSDKVersion : 10.00.00.3802 WMFSDKNeeded : 0.0.0.0000 IsVBR : 1 Buffer Average : 2743 Duration: 00:00:43.21, start: 0.000000, bitrate: 9064 kb/s Stream #0:0: Audio: wmapro (b[1][0][0] / 0x0162), 48000 Hz, 5.1, fltp, 423 k b/s Stream #0:1: Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 1440x1080, 800 0 kb/s, SAR 1:1 DAR 4:3, 24 fps, 24 tbr, 1k tbn, 1k tbc [mpeg2video @ 041fce80] Automatically choosing VBV buffer size of 224 kbyte Output #0, mpeg2video, to 'output.m2v': Metadata: VBR Peak : 10000000 DeviceConformanceTemplate: MP at HL WM/WMADRCPeakReference: 23859 WM/WMADRCPeakTarget: 23859 WM/WMADRCAverageReference: 4536 WM/WMADRCAverageTarget: 4536 WMFSDKVersion : 10.00.00.3802 WMFSDKNeeded : 0.0.0.0000 IsVBR : 1 Buffer Average : 2743 encoder : Lavf56.13.100 Stream #0:0: Video: mpeg2video, yuv420p, 1440x1080 [SAR 1:1 DAR 4:3], q=2-31 , 8000 kb/s, 24 fps, 24 tbn, 24 tbc Metadata: encoder : Lavc56.12.101 mpeg2video Stream mapping: Stream #0:1 -> #0:0 (wmv3 (native) -> mpeg2video (native)) Press [q] to stop, [?] for help frame= 36 fps=0.0 q=4.3 size= 1531kB time=00:00:01.41 bitrate=8852.0kbits/s frame= 73 fps= 72 q=4.2 size= 2976kB time=00:00:02.95 bitrate=8241.7kbits/s frame= 110 fps= 73 q=4.7 size= 4476kB time=00:00:04.50 bitrate=8148.4kbits/s frame= 146 fps= 73 q=4.7 size= 5934kB time=00:00:06.00 bitrate=8101.6kbits/s frame= 183 fps= 72 q=4.2 size= 7410kB time=00:00:07.54 bitrate=8048.9kbits/s frame= 221 fps= 73 q=4.8 size= 8977kB time=00:00:09.12 bitrate=8059.2kbits/s frame= 259 fps= 73 q=6.5 size= 10510kB time=00:00:10.70 bitrate=8040.6kbits/s frame= 297 fps= 73 q=5.7 size= 12103kB time=00:00:12.29 bitrate=8066.0kbits/s frame= 333 fps= 73 q=7.7 size= 13620kB time=00:00:13.79 bitrate=8090.0kbits/s frame= 370 fps= 73 q=6.0 size= 15050kB time=00:00:15.33 bitrate=8040.7kbits/s frame= 406 fps= 73 q=7.9 size= 16536kB time=00:00:16.83 bitrate=8047.2kbits/s frame= 442 fps= 73 q=6.8 size= 17990kB time=00:00:18.33 bitrate=8038.6kbits/s frame= 479 fps= 73 q=5.4 size= 19394kB time=00:00:19.87 bitrate=7993.6kbits/s frame= 514 fps= 73 q=7.0 size= 20922kB time=00:00:21.33 bitrate=8033.9kbits/s frame= 553 fps= 73 q=10.5 size= 22452kB time=00:00:22.95 bitrate=8011.3kbits/ frame= 591 fps= 73 q=7.6 size= 24004kB time=00:00:24.54 bitrate=8012.5kbits/s frame= 628 fps= 73 q=5.6 size= 25508kB time=00:00:26.08 bitrate=8011.4kbits/s frame= 668 fps= 73 q=4.1 size= 27105kB time=00:00:27.75 bitrate=8001.5kbits/s frame= 705 fps= 73 q=6.0 size= 28663kB time=00:00:29.29 bitrate=8016.1kbits/s frame= 742 fps= 73 q=4.2 size= 30105kB time=00:00:30.83 bitrate=7998.5kbits/s frame= 776 fps= 73 q=10.2 size= 31641kB time=00:00:32.25 bitrate=8037.3kbits/ frame= 811 fps= 73 q=8.8 size= 33015kB time=00:00:33.70 bitrate=8023.6kbits/s frame= 847 fps= 73 q=9.7 size= 34539kB time=00:00:35.20 bitrate=8036.3kbits/s frame= 881 fps= 73 q=9.0 size= 35930kB time=00:00:36.62 bitrate=8036.5kbits/s frame= 916 fps= 72 q=7.9 size= 37290kB time=00:00:38.08 bitrate=8021.3kbits/s frame= 952 fps= 72 q=4.8 size= 38672kB time=00:00:39.58 bitrate=8003.4kbits/s frame= 986 fps= 72 q=3.7 size= 39992kB time=00:00:41.00 bitrate=7990.6kbits/s frame= 1016 fps= 72 q=3.6 size= 41183kB time=00:00:42.25 bitrate=7985.0kbits/s frame= 1037 fps= 71 q=3.3 Lsize= 42093kB time=00:00:43.16 bitrate=7988.2kbits/ s video:42093kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000% Thanks for any help. From info at webdimensions.org Tue Nov 18 21:16:07 2014 From: info at webdimensions.org (Hugh J. Hitchcock) Date: Tue, 18 Nov 2014 15:16:07 -0500 Subject: [FFmpeg-user] -longest rather than -shortest? Message-ID: <07ef01d0036c$7e5a1ad0$7b0e5070$@webdimensions.org> Hi, I am new to ffmpeg and I've been using it lately to create videos from assembling images and mp3s. I'm kind of stumped right now on something and I'm hoping someone in the group can help me! What I want to do is combine two mp3s into a soundtrack and combine them to a video. I don't have any problem doing that, however, the issue is that no matter what I try, the resulting video always stops at the end of the shortest input track. This seems to happen whether or not I use the -shortest argument. What I want to do is find out how to make ffmpeg take the LONGEST piece of media and make the video around that, and I have tried and tried and can't seem to do it! Can anyone help me? Many thanks in advance. Sincerely, Hugh From ikrananka at hotmail.com Tue Nov 18 20:06:51 2014 From: ikrananka at hotmail.com (Andrew Arthur) Date: Tue, 18 Nov 2014 12:06:51 -0700 Subject: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem Message-ID: I am converting a load of WMV files to M2V (i.e. video conversion only) using the following basic command line: ffmpeg -i input.wmv -codec:v mpeg2video -b:v 8000k -maxrate 10000k output.m2v However, it is imperative in my conversion that the number of frames in the output file be identical to the input file. However I consistently get one less frame in the m2v files compared to the wmv files. Is there an alternative command line that will ensure the total number of frames is honored? To provide a little further info, one example is as follows. The input.wmv file has 1037 frames, however the output m2v file only has 1036 frames: D:\ffmpeg -i input.wmv -codec:v mpeg2video -b:v 8000k -maxrate 10000k output.m2v ffmpeg version N-67742-g3f07dd6 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 16 2014 22:01:52 with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 13.100 / 54. 13.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 13.100 / 56. 13.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, asf, from 'input.wmv': Metadata: VBR Peak : 10000000 DeviceConformanceTemplate: MP at HL WM/WMADRCPeakReference: 23859 WM/WMADRCPeakTarget: 23859 WM/WMADRCAverageReference: 4536 WM/WMADRCAverageTarget: 4536 WMFSDKVersion : 10.00.00.3802 WMFSDKNeeded : 0.0.0.0000 IsVBR : 1 Buffer Average : 2743 Duration: 00:00:43.21, start: 0.000000, bitrate: 9064 kb/s Stream #0:0: Audio: wmapro (b[1][0][0] / 0x0162), 48000 Hz, 5.1, fltp, 423 k b/s Stream #0:1: Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 1440x1080, 800 0 kb/s, SAR 1:1 DAR 4:3, 24 fps, 24 tbr, 1k tbn, 1k tbc [mpeg2video @ 041fce80] Automatically choosing VBV buffer size of 224 kbyte Output #0, mpeg2video, to 'output.m2v': Metadata: VBR Peak : 10000000 DeviceConformanceTemplate: MP at HL WM/WMADRCPeakReference: 23859 WM/WMADRCPeakTarget: 23859 WM/WMADRCAverageReference: 4536 WM/WMADRCAverageTarget: 4536 WMFSDKVersion : 10.00.00.3802 WMFSDKNeeded : 0.0.0.0000 IsVBR : 1 Buffer Average : 2743 encoder : Lavf56.13.100 Stream #0:0: Video: mpeg2video, yuv420p, 1440x1080 [SAR 1:1 DAR 4:3], q=2-31 , 8000 kb/s, 24 fps, 24 tbn, 24 tbc Metadata: encoder : Lavc56.12.101 mpeg2video Stream mapping: Stream #0:1 -> #0:0 (wmv3 (native) -> mpeg2video (native)) Press [q] to stop, [?] for help frame= 36 fps=0.0 q=4.3 size= 1531kB time=00:00:01.41 bitrate=8852.0kbits/s frame= 73 fps= 72 q=4.2 size= 2976kB time=00:00:02.95 bitrate=8241.7kbits/s frame= 110 fps= 73 q=4.7 size= 4476kB time=00:00:04.50 bitrate=8148.4kbits/s frame= 146 fps= 73 q=4.7 size= 5934kB time=00:00:06.00 bitrate=8101.6kbits/s frame= 183 fps= 72 q=4.2 size= 7410kB time=00:00:07.54 bitrate=8048.9kbits/s frame= 221 fps= 73 q=4.8 size= 8977kB time=00:00:09.12 bitrate=8059.2kbits/s frame= 259 fps= 73 q=6.5 size= 10510kB time=00:00:10.70 bitrate=8040.6kbits/s frame= 297 fps= 73 q=5.7 size= 12103kB time=00:00:12.29 bitrate=8066.0kbits/s frame= 333 fps= 73 q=7.7 size= 13620kB time=00:00:13.79 bitrate=8090.0kbits/s frame= 370 fps= 73 q=6.0 size= 15050kB time=00:00:15.33 bitrate=8040.7kbits/s frame= 406 fps= 73 q=7.9 size= 16536kB time=00:00:16.83 bitrate=8047.2kbits/s frame= 442 fps= 73 q=6.8 size= 17990kB time=00:00:18.33 bitrate=8038.6kbits/s frame= 479 fps= 73 q=5.4 size= 19394kB time=00:00:19.87 bitrate=7993.6kbits/s frame= 514 fps= 73 q=7.0 size= 20922kB time=00:00:21.33 bitrate=8033.9kbits/s frame= 553 fps= 73 q=10.5 size= 22452kB time=00:00:22.95 bitrate=8011.3kbits/ frame= 591 fps= 73 q=7.6 size= 24004kB time=00:00:24.54 bitrate=8012.5kbits/s frame= 628 fps= 73 q=5.6 size= 25508kB time=00:00:26.08 bitrate=8011.4kbits/s frame= 668 fps= 73 q=4.1 size= 27105kB time=00:00:27.75 bitrate=8001.5kbits/s frame= 705 fps= 73 q=6.0 size= 28663kB time=00:00:29.29 bitrate=8016.1kbits/s frame= 742 fps= 73 q=4.2 size= 30105kB time=00:00:30.83 bitrate=7998.5kbits/s frame= 776 fps= 73 q=10.2 size= 31641kB time=00:00:32.25 bitrate=8037.3kbits/ frame= 811 fps= 73 q=8.8 size= 33015kB time=00:00:33.70 bitrate=8023.6kbits/s frame= 847 fps= 73 q=9.7 size= 34539kB time=00:00:35.20 bitrate=8036.3kbits/s frame= 881 fps= 73 q=9.0 size= 35930kB time=00:00:36.62 bitrate=8036.5kbits/s frame= 916 fps= 72 q=7.9 size= 37290kB time=00:00:38.08 bitrate=8021.3kbits/s frame= 952 fps= 72 q=4.8 size= 38672kB time=00:00:39.58 bitrate=8003.4kbits/s frame= 986 fps= 72 q=3.7 size= 39992kB time=00:00:41.00 bitrate=7990.6kbits/s frame= 1016 fps= 72 q=3.6 size= 41183kB time=00:00:42.25 bitrate=7985.0kbits/s frame= 1037 fps= 71 q=3.3 Lsize= 42093kB time=00:00:43.16 bitrate=7988.2kbits/ s video:42093kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000% Thanks for any help. From radpopl at gmail.com Tue Nov 18 22:38:39 2014 From: radpopl at gmail.com (=?UTF-8?B?UmFkb3PFgmF3IFBvcMWCYXdza2k=?=) Date: Tue, 18 Nov 2014 22:38:39 +0100 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: <546B48A0.1020306@gmail.com> References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> <546B48A0.1020306@gmail.com> Message-ID: Debug log: C:\Users\W?a?ciciel\Documents\2014\a>ffmpeg.exe -v debug -n -i "C:\Users\W?a?cic iel\Documents\2014\a\2014_11_11 15_48_02.mts" -c:v mpeg4 -qscale:v 4 -vtag xvid -flags +qpel+ilme+ildct -acodec libmp3lame -b:a 320k "C:\Users\W?a?ciciel\Docume nts\2014\a\2014_11_11 15_48_02.avi" ffmpeg version N-67742-g3f07dd6 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 16 2014 22:10:05 with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 13.100 / 54. 13.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 13.100 / 56. 13.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Splitting the commandline. Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'. Reading option '-n' ... matched as option 'n' (never overwrite output files) wit h argument '1'. Reading option '-i' ... matched as input file with argument 'C:\Users\W??a??cici el\Documents\2014\a\2014_11_11 15_48_02.mts'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'mpeg 4'. Reading option '-qscale:v' ... matched as option 'qscale' (use fixed quality sca le (VBR)) with argument '4'. Reading option '-vtag' ... matched as option 'vtag' (force video tag/fourcc) wit h argument 'xvid'. Reading option '-flags' ... matched as AVOption 'flags' with argument '+qpel+ilm e+ildct'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('cop y' to copy stream)) with argument 'libmp3lame'. Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v) ) with argument '320k'. Reading option 'C:\Users\W??a??ciciel\Documents\2014\a\2014_11_11 15_48_02.avi' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument debug. Applying option n (never overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file C:\Users\W??a??ciciel\Documents\2014\a\20 14_11_11 15_48_02.mts. Successfully parsed a group of options. Opening an input file: C:\Users\W??a??ciciel\Documents\2014\a\2014_11_11 15_48_0 2.mts. [mpegts @ 0000000000304e40] Format mpegts probed with size=2048 and score=100 [mpegts @ 0000000000304e40] stream=0 stream_type=1b pid=1011 prog_reg_desc=HDMV [mpegts @ 0000000000304e40] stream=1 stream_type=81 pid=1100 prog_reg_desc=HDMV [mpegts @ 0000000000304e40] stream=2 stream_type=90 pid=1200 prog_reg_desc=HDMV [mpegts @ 0000000000304e40] Before avformat_find_stream_info() pos: 0 bytes read :32768 seeks:0 [mpegts @ 0000000000304e40] parser not found for codec hdmv_pgs_subtitle, packet s or times may be invalid. Last message repeated 1 times [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 2 with DTS 87 677, packet 3 with DTS 87677 [h264 @ 000000000032a1a0] unknown SEI type 128 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 5 with DTS 13 4477, packet 6 with DTS 134477 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 6 with DTS 13 4477, packet 7 with DTS 134477 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 7 with DTS 13 4477, packet 8 with DTS 134477 [h264 @ 000000000032a1a0] unknown SEI type 128 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 10 with DTS 1 81277, packet 11 with DTS 181277 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 11 with DTS 1 81277, packet 12 with DTS 181277 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 12 with DTS 1 81277, packet 13 with DTS 181277 [h264 @ 000000000032a1a0] unknown SEI type 128 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 15 with DTS 2 28077, packet 16 with DTS 228077 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 16 with DTS 2 28077, packet 17 with DTS 228077 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 17 with DTS 2 28077, packet 18 with DTS 228077 [h264 @ 000000000032a1a0] unknown SEI type 128 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 20 with DTS 2 74877, packet 21 with DTS 274877 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 21 with DTS 2 74877, packet 22 with DTS 274877 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 22 with DTS 2 74877, packet 23 with DTS 274877 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 25 with DTS 3 21677, packet 26 with DTS 321677 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 26 with DTS 3 21677, packet 27 with DTS 321677 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 27 with DTS 3 21677, packet 28 with DTS 321677 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 30 with DTS 3 68477, packet 31 with DTS 368477 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 31 with DTS 3 68477, packet 32 with DTS 368477 [mpegts @ 0000000000304e40] Non-increasing DTS in stream 2: packet 32 with DTS 3 68477, packet 33 with DTS 368477 [mpegts @ 0000000000304e40] Probe buffer size limit of 5000000 bytes reached [mpegts @ 0000000000304e40] After avformat_find_stream_info() pos: 0 bytes read: 5640336 seeks:2 frames:308 Input #0, mpegts, from 'C:\Users\W??a??ciciel\Documents\2014\a\2014_11_11 15_48_ 02.mts': Duration: 00:01:00.32, start: 1.040000, bitrate: 9569 kb/s Program 1 Stream #0:0[0x1011], 179, 1/90000: Video: h264 (High) (HDMV / 0x564D4448), y uv420p(left), 1440x1080 (1440x1088) [SAR 4:3 DAR 16:9], 1/50, 25 fps, 25 tbr, 90 k tbn, 50 tbc Stream #0:1[0x1100], 94, 1/90000: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s Stream #0:2[0x1200], 35, 1/90000: Subtitle: hdmv_pgs_subtitle ([144][0][0][0 ] / 0x0090), 1920x1080 Successfully opened the file. Parsing a group of options: output file C:\Users\W??a??ciciel\Documents\2014\a\2 014_11_11 15_48_02.avi. Applying option c:v (codec name) with argument mpeg4. Applying option qscale:v (use fixed quality scale (VBR)) with argument 4. Applying option vtag (force video tag/fourcc) with argument xvid. Applying option acodec (force audio codec ('copy' to copy stream)) with argument libmp3lame. Applying option b:a (video bitrate (please use -b:v)) with argument 320k. Successfully parsed a group of options. Opening an output file: C:\Users\W??a??ciciel\Documents\2014\a\2014_11_11 15_48_ 02.avi. Successfully opened the file. detected 2 logical cores [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'video_size' to value '1440x1080' [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'pix_fmt' to value '0 ' [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'time_base' to value '1/90000' [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'pixel_aspect' to val ue '4/3' [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'sws_param' to value 'flags=2' [graph 0 input from stream 0:0 @ 0000000002ce7400] Setting 'frame_rate' to value '25/1' [graph 0 input from stream 0:0 @ 0000000002ce7400] w:1440 h:1080 pixfmt:yuv420p tb:1/90000 fr:25/1 sar:4/3 sws_param:flags=2 [format @ 0000000005cffee0] compat: called with args=[yuv420p] [format @ 0000000005cffee0] Setting 'pix_fmts' to value 'yuv420p' [AVFilterGraph @ 0000000005e373e0] query_formats: 4 queried, 3 merged, 0 already done, 0 delayed [graph 1 input from stream 0:1 @ 0000000002ce79e0] Setting 'time_base' to value '1/48000' [graph 1 input from stream 0:1 @ 0000000002ce79e0] Setting 'sample_rate' to valu e '48000' [graph 1 input from stream 0:1 @ 0000000002ce79e0] Setting 'sample_fmt' to value 'fltp' [graph 1 input from stream 0:1 @ 0000000002ce79e0] Setting 'channel_layout' to v alue '0x3' [graph 1 input from stream 0:1 @ 0000000002ce79e0] tb:1/48000 samplefmt:fltp sam plerate:48000 chlayout:0x3 [audio format for output stream 0:1 @ 00000000003bf520] Setting 'sample_fmts' to value 's32p|fltp|s16p' [audio format for output stream 0:1 @ 00000000003bf520] Setting 'sample_rates' t o value '44100|48000|32000|22050|24000|16000|11025|12000|8000' [audio format for output stream 0:1 @ 00000000003bf520] Setting 'channel_layouts ' to value '0x4|0x3' [AVFilterGraph @ 0000000005e36da0] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed [mpeg4 @ 0000000005900b40] intra_quant_bias = 0 inter_quant_bias = -64 Output #0, avi, to 'C:\Users\W??a??ciciel\Documents\2014\a\2014_11_11 15_48_02.a vi': Metadata: ISFT : Lavf56.13.100 Stream #0:0, 0, 1/25: Video: mpeg4 (xvid / 0x64697678), yuv420p(left), 1440x 1080 [SAR 4:3 DAR 16:9], 1/25, q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc56.12.101 mpeg4 Stream #0:1, 0, 3/125: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 48000 Hz, stereo, fltp, 320 kb/s Metadata: encoder : Lavc56.12.101 libmp3lame Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg4 (native)) Stream #0:1 -> #0:1 (ac3 (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help [NULL @ 000000000032a1a0] unknown SEI type 128 [h264 @ 0000000005e20b40] no picture On Tue, Nov 18, 2014 at 2:24 PM, Rados?aw Pop?awski wrote: > > W dniu 2014-11-17 22:50, Rados?aw Pop?awski pisze: > >> On Sun, Nov 16, 2014 at 10:54 PM, Carl Eugen Hoyos > > wrote: >> >> radpopl gmail.com > writes: >> >> > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid >> > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k >> > "%%~dF%%~pF%%~nF.avi" >> > >> > but ffmpeg crashed. >> >> Command line, console output etc. missing. >> >> Please understand that crashes are always >> important but we can only fix them if we >> are able to reproduce them... >> >> >> There is a screenshot: >> http://i61.tinypic.com/jfkpkx.jpg >> >> Regards, >> rp >> > I should add that removing +qpel fixes the problem. > > From cehoyos at ag.or.at Tue Nov 18 22:57:13 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 21:57:13 +0000 (UTC) Subject: [FFmpeg-user] h264 file: mplayer plays it, but ffmpeg can't decode it References: Message-ID: Bryan Field-Elliot gmail.com> writes: > I have a h264 file, which I obtained by capturing > bytes streaming (mid-stream) from an IP camera. You can convert your sample with: $ ffmpeg -flags2 +showall -i vid.h264 vid.mp4 (MPlayer always sets showall to speed up seeking.) Before I open a ticket: Could you elaborate how (exactly) you captured the stream? Is FFmpeg able to read frames directly from the camera? Or does this also need "-flags2 +showall"? Thank you for the sample, Carl Eugen From cehoyos at ag.or.at Tue Nov 18 23:03:26 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 22:03:26 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Parsed=5Fpan=5F0_This_syntax_is_deprecate?= =?utf-8?q?d=2E_Use_=27=7C=27_to_separate_the_list_items?= References: <1A96BFA0-0A5E-4DA3-8A2D-9E740143865B@apost.plala.or.jp> <20141118150301.GH22910@sunshine.barsnick.net> Message-ID: Moritz Barsnick gmx.net> writes: > See attached. You were beaten today, please consider sending future patches to the ffmpeg-devel mailing list. Thank you, Carl Eugen From cehoyos at ag.or.at Tue Nov 18 23:01:24 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 22:01:24 +0000 (UTC) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: radpopl gmail.com> writes: > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > "%%~dF%%~pF%%~nF.avi" > > but ffmpeg crashed. Please provide the input sample. (I cannot reproduce with a random sample.) Carl Eugen From cehoyos at ag.or.at Tue Nov 18 23:07:02 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 22:07:02 +0000 (UTC) Subject: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem References: Message-ID: Andrew Arthur hotmail.com> writes: > The input.wmv file has 1037 frames, however the output > m2v file only has 1036 frames: > frame= 1037 fps= 71 q=3.3 Lsize= 42093kB time=00:00:43.16 > bitrate=7988.2kbits/ Why do you think that the output file has 1036 frames? The console output you provided suggests it has 1037 frames. Carl Eugen From info at webdimensions.org Tue Nov 18 23:32:15 2014 From: info at webdimensions.org (Hugh J. Hitchcock) Date: Tue, 18 Nov 2014 17:32:15 -0500 Subject: [FFmpeg-user] -longest rather than -shortest? In-Reply-To: <07ef01d0036c$7e5a1ad0$7b0e5070$@webdimensions.org> References: <07ef01d0036c$7e5a1ad0$7b0e5070$@webdimensions.org> Message-ID: <084d01d0037f$82395630$86ac0290$@webdimensions.org> Like for example, I want to say create a video with a 1 min background music track, a 30 second voice over track. I can't figure out how to apply the voice-over track to the longer background track without ffmpeg shortening the output to 30 seconds - I want the entire 60 output with the background music track even when the 30 second voice track stops. Any ideas? Profuse thanks in advance -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Hugh J. Hitchcock Sent: Tuesday, November 18, 2014 3:16 PM To: 'FFmpeg user questions' Subject: [FFmpeg-user] -longest rather than -shortest? Hi, I am new to ffmpeg and I've been using it lately to create videos from assembling images and mp3s. I'm kind of stumped right now on something and I'm hoping someone in the group can help me! What I want to do is combine two mp3s into a soundtrack and combine them to a video. I don't have any problem doing that, however, the issue is that no matter what I try, the resulting video always stops at the end of the shortest input track. This seems to happen whether or not I use the -shortest argument. What I want to do is find out how to make ffmpeg take the LONGEST piece of media and make the video around that, and I have tried and tried and can't seem to do it! Can anyone help me? Many thanks in advance. Sincerely, Hugh _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From ikrananka at hotmail.com Tue Nov 18 23:35:47 2014 From: ikrananka at hotmail.com (Andrew Arthur) Date: Tue, 18 Nov 2014 15:35:47 -0700 Subject: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem In-Reply-To: References: Message-ID: I have been inspecting the mpeg properties using the mpeg plug-in to VirtualDub. If I check the original wmv file it shows it as having 1037 frames and a length of 43.208. If I convert the file using TMPGenc and check it in VirtualDub it shows 1037 frames and a length of gives 43.208, i.e. the same as the wmv file. However, the output from ffmpeg shows in VirtualDub as being 1036 frames and a length of 43.167 (the length is 1036/1037th of the original length). So, it really does seem to be losing a frame in the conversion. Andrew -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: November-18-14 3:07 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem Andrew Arthur hotmail.com> writes: > The input.wmv file has 1037 frames, however the output m2v file only > has 1036 frames: > frame= 1037 fps= 71 q=3.3 Lsize= 42093kB time=00:00:43.16 > bitrate=7988.2kbits/ Why do you think that the output file has 1036 frames? The console output you provided suggests it has 1037 frames. 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 18 23:53:28 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Tue, 18 Nov 2014 22:53:28 +0000 (UTC) Subject: [FFmpeg-user] Conversion from WMV to M2V - Lost Frame Problem References: Message-ID: Andrew Arthur hotmail.com> writes: > However, the output from ffmpeg shows in > VirtualDub as being 1036 frames Please run: $ ffmpeg -i output.m2v -f null - One of the last lines of the console output sill show you the actual number of frames in the m2v file. If you don't believe the number the following command will output every single frame of your video as a raw mpeg2video file. $ ffmpeg -i output.m2v -vcodec copy -f image2 raw%4d.m2v The following will output a jpg for every single frame: $ ffmpeg -i output.m2v out%4d.jpg Please do not top-post here, Carl Eugen From admin at itvc.pl Wed Nov 19 00:22:24 2014 From: admin at itvc.pl (admin) Date: Wed, 19 Nov 2014 00:22:24 +0100 Subject: [FFmpeg-user] Connection reset by peer Message-ID: <5d8e0dff74ac5763f9d14985c3ec8be0@itvc.pl> Hi guys, When i`m trying to send stream from ffmpeg to ffserver i have got Connection reset by peer. It`s not firewall problem. Maybe linux distribution? Its Slackware.. When i`m trying to send stream via udp to server where ffserver is (external IP), i don`t have any error Full debug bellow, i wll be v. glad for help. Maybe anyone can share for some while own ffserver for test. ############# Client cmd: ffmpeg -i udp://@231.1.2.199:1234 -b:v 1500k -pix_fmt yuv420p -vcodec libx264 -tune zerolatency -preset ultrafast -f mpegts "http : / / xx.xx.xx.xx:8090/dupa.ffm" File /home/ffmpeg/dupa.ffm FileMaxSize 2G # This is roughly 24h of media Feed dupa.ffm Format mpegts ############# Server debug: ffserver version N-67244-g1a25c33 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 28 2014 20:12:25 with gcc 4.8.2 (GCC) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --docdir=/usr/doc/ffmpeg-2.4/html --mandir=/usr/man --disable-debug --enable-shared --disable-static --enable-pthreads --enable-libtheora --enable-libvorbis --enable-gpl --enable-version3 --enable-libx264 --enable-postproc --enable-swscale --disable-x11grab --enable-avfilter --enable-gnutls --enable-libcdio --enable-libssh --arch=x86_64 --enable-libmp3lame --enable-libx264 libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.100 / 56. 10.100 libavformat 56. 11.100 / 56. 11.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.100 / 5. 2.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Tue Oct 28 22:46:23 2014 [ffm @ 0xd53ae0]Format ffm probed with size=2048 and score=101 Tue Oct 28 22:46:23 2014 [AVIOContext @ 0xd4f940]Statistics: 4096 bytes read, 0 seeks Tue Oct 28 22:46:23 2014 FFserver started. Tue Oct 28 22:46:40 2014 xx.xx.xx.xx - - [POST] "/dupa.ffm HTTP/1.1" 200 4096[/code] Client debug: [code]ffmpeg version N-67244-g1a25c33 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 28 2014 20:12:25 with gcc 4.8.2 (GCC) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --docdir=/usr/doc/ffmpeg-2.4/html --mandir=/usr/man --disable-debug --enable-shared libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.100 / 56. 10.100 libavformat 56. 11.100 / 56. 11.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.100 / 5. 2.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Splitting the commandline. Reading option '-i' ... matched as input file with argument 'udp://@231.1.2.199:1234'. Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '1500k'. Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument 'yuv420p'. Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'. Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'. Reading option '-preset' ... matched as AVOption 'preset' with argument 'ultrafast'. Reading option '-f' ... matched as option 'f' (force format) with argument 'mpegts'. Reading option 'htt p : / / xx.xx.xx.xx:8090/dupa.ffm' ... matched as output file. Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file udp://@231.1.2.199:1234. Successfully parsed a group of options. Opening an input file: udp://@231.1.2.199:1234. [udp @ 0x10ebea0] end receive buffer size reported is 131072 [mpegts @ 0x10ed240] Format mpegts probed with size=2048 and score=100 [mpegts @ 0x10ed240] stream=0 stream_type=1b pid=7d2 prog_reg_desc= [mpegts @ 0x10ed240] stream=1 stream_type=4 pid=bba prog_reg_desc= [mpegts @ 0x10ed240] Before avformat_find_stream_info() pos: 0 bytes read:25004 seeks:0 [mpegts @ 0x10ed240] All programs have pmt, headers found [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] Current profile doesn't provide more RBSP data in PPS, skipping [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] Increasing reorder buffer to 1 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 1 times [h264 @ 0x1100480] Increasing reorder buffer to 2 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] Increasing reorder buffer to 3 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 5 times [h264 @ 0x1100480] Current profile doesn't provide more RBSP data in PPS, skipping [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 3 times [mpegts @ 0x10ed240] All info found [mpegts @ 0x10ed240] After avformat_find_stream_info() pos: 285572 bytes read:285572 seeks:0 frames:51 Input #0, mpegts, from 'udp://@231.1.2.199:1234': Duration: N/A, start: 86384.517867, bitrate: 247 kb/s Program 5090. Metadata: service_name : TV Disco service_provider:. Stream #0:0[0x7d2], 21, 1/90000: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt470bg, left), 720x576 [SAR 16:11 DAR 20:11], 1/50, 25 fps, 25 Stream #0:1[0xbba], 30, 1/90000: Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 247 kb/s Successfully opened the file. Parsing a group of options: output file htt p : / / xx.xx.xx.xx:8090/dupa.ffm. Applying option b:v (video bitrate (please use -b:v)) with argument 1500k. Applying option pix_fmt (set pixel format) with argument yuv420p. Applying option vcodec (force video codec ('copy' to copy stream)) with argument libx264. Applying option f (force format) with argument mpegts. Successfully parsed a group of options. Opening an output file: htt p :/ / xx.xx.xx.xx:8090/dupa.ffm. [http @ 0x1101c80] request: POST /dupa.ffm HTTP/1.1^M Transfer-Encoding: chunked^M User-Agent: Lavf/56.11.100^M Accept: */*^M Connection: close^M Host: xx.xx.xx.xx:8090^M Icy-MetaData: 1^M ^M Successfully opened the file. detected 8 logical cores [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'video_size' to value '720x576' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'pix_fmt' to value '0' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'time_base' to value '1/90000' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'pixel_aspect' to value '16/11' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'sws_param' to value 'flags=2' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'frame_rate' to value '25/1' [graph 0 input from stream 0:0 @ 0x13778e0] w:720 h:576 pixfmt:yuv420p tb:1/90000 fr:25/1 sar:16/11 sws_param:flags=2 [format @ 0x10ec040] compat: called with args=[yuv420p] [format @ 0x10ec040] Setting 'pix_fmts' to value 'yuv420p' [AVFilterGraph @ 0x1139700] query_formats: 4 queried, 3 merged, 0 already done, 0 delayed [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'time_base' to value '1/48000' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'sample_rate' to value '48000' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'sample_fmt' to value 's16p' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'channel_layout' to value '0x3' [graph 1 input from stream 0:1 @ 0x1377dc0] tb:1/48000 samplefmt:s16p samplerate:48000 chlayout:0x3 [audio format for output stream 0:1 @ 0x113a520] Setting 'sample_fmts' to value 's16' [audio format for output stream 0:1 @ 0x113a520] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000' [audio format for output stream 0:1 @ 0x113a520] Setting 'channel_layouts' to value '0x4|0x3' [audio format for output stream 0:1 @ 0x113a520] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio [AVFilterGraph @ 0x10ecae0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x119e820] ch:2 chl:stereo fmt:s16p r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz [libx264 @ 0x10e4760] using SAR=16/11 [libx264 @ 0x10e4760] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 [libx264 @ 0x10e4760] profile Constrained Baseline, level 3.0 [mpegts @ 0x112b240] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'htt p :/ /xx.xx.xx.xx:8090/dupa.ffm': Metadata: encoder : Lavf56.11.100 Stream #0:0, 0, 1/90000: Video: h264 (libx264), yuv420p(left), 720x576 [SAR 16:11 DAR 20:11], 1/25, q=-1--1, 1500 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc56.10.100 libx264 Stream #0:1, 0, 1/90000: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata: encoder : Lavc56.10.100 mp2 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (mp2 (native) -> mp2 (native)) Press [q] to stop, [?] for help [h264 @ 0x1264d60] Frame num gap 10 5 [h264 @ 0x1264d60] Frame num gap 10 6 [h264 @ 0x1264d60] Frame num gap 10 7 [h264 @ 0x1264d60] Frame num gap 10 8 [h264 @ 0x1264d60] no picture. [h264 @ 0x1265de0] mmco: unref short failure [h264 @ 0x1265de0] number of reference frames (0+5) exceeds max (4; probably corrupt input), discarding one [h264 @ 0x1265de0] no picture. [h264 @ 0x1266660] no picture. *** 18 dup! [libx264 @ 0x10e4760] frame= 0 QP=25.00 NAL=3 Slice:I Poc:0 I:1620 P:0 SKIP:0 size=18259 bytes [libx264 @ 0x10e4760] frame= 1 QP=26.00 NAL=2 Slice:P Poc:2 I:0 P:0 SKIP:1620 size=86 bytes [libx264 @ 0x10e4760] frame= 2 QP=22.00 NAL=2 Slice:P Poc:4 I:5 P:649 SKIP:966 size=2547 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 3 QP=18.00 NAL=2 Slice:P Poc:6 I:27 P:1321 SKIP:272 size=11388 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 4 QP=16.00 NAL=2 Slice:P Poc:8 I:46 P:1416 SKIP:158 size=12070 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 5 QP=15.00 NAL=2 Slice:P Poc:10 I:3 P:883 SKIP:734 size=5210 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 6 QP=14.00 NAL=2 Slice:P Poc:12 I:12 P:1200 SKIP:408 size=7699 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 7 QP=13.00 NAL=2 Slice:P Poc:14 I:8 P:1360 SKIP:252 size=11766 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 8 QP=13.00 NAL=2 Slice:P Poc:16 I:0 P:18 SKIP:1602 size=169 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 9 QP=12.00 NAL=2 Slice:P Poc:18 I:19 P:1342 SKIP:259 size=9715 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 10 QP=12.00 NAL=2 Slice:P Poc:20 I:5 P:617 SKIP:998 size=2314 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 11 QP=12.00 NAL=2 Slice:P Poc:22 I:3 P:580 SKIP:1037 size=2117 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 12 QP=11.00 NAL=2 Slice:P Poc:24 I:3 P:1260 SKIP:357 size=7489 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 13 QP=11.00 NAL=2 Slice:P Poc:26 I:3 P:832 SKIP:785 size=3103 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 14 QP=10.00 NAL=2 Slice:P Poc:28 I:5 P:1398 SKIP:217 size=12018 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 15 QP=11.00 NAL=2 Slice:P Poc:30 I:2 P:386 SKIP:1232 size=1176 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 16 QP=10.00 NAL=2 Slice:P Poc:32 I:2 P:343 SKIP:1275 size=1064 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 17 QP=10.00 NAL=2 Slice:P Poc:34 I:2 P:116 SKIP:1502 size=463 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 18 QP=9.00 NAL=2 Slice:P Poc:36 I:2 P:779 SKIP:839 size=4851 bytes av_interleaved_write_frame(): Connection reset by peer No more output streams to write to, finishing. frame= 19 fps=0.0 q=9.0 Lsize= 92kB time=00:00:00.76 bitrate= 987.5kbits/s dup=18 drop=0 ^M video:111kB audio:22kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Input file #0 (udp://@231.1.2.199:1234): Input stream #0:0 (video): 15 packets read (159198 bytes); 2 frames decoded;. Input stream #0:1 (audio): 20 packets read (15360 bytes); 20 frames decoded (23040 samples);. Total: 35 packets (174558 bytes) demuxed Output file #0 (htt p :/ / xx.xx.xx.xx:8090/dupa.ffm): Output stream #0:0 (video): 19 frames encoded; 19 packets muxed (113504 bytes);. Output stream #0:1 (audio): 20 frames encoded (23040 samples); 20 packets muxed (23040 bytes);. Total: 39 packets (136544 bytes) muxed 36 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x127e240] Statistics: 0 seeks, 16 writeouts [libx264 @ 0x10e4760] frame I:1 Avg QP:25.00 size: 18259 [libx264 @ 0x10e4760] frame P:18 Avg QP:13.61 size: 5291 [libx264 @ 0x10e4760] mb I I16..4: 100.0% 0.0% 0.0% [libx264 @ 0x10e4760] mb P I16..4: 0.5% 0.0% 0.0% P16..4: 49.7% 0.0% 0.0% 0.0% 0.0% skip:49.8% [libx264 @ 0x10e4760] final ratefactor: 20.23 [libx264 @ 0x10e4760] coded y,uvDC,uvAC intra: 45.6% 56.9% 16.4% inter: 19.6% 21.4% 5.9% [libx264 @ 0x10e4760] i16 v,h,dc,p: 29% 34% 14% 23% [libx264 @ 0x10e4760] i8c dc,h,v,p: 47% 21% 24% 8% [libx264 @ 0x10e4760] kb/s:1194.78 [AVIOContext @ 0x10ed980] Statistics: 285572 bytes read, 0 seeks Conversion failed! Thanks for help! Lucas From xanadu at apost.plala.or.jp Wed Nov 19 13:28:01 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Wed, 19 Nov 2014 21:28:01 +0900 Subject: [FFmpeg-user] ffplay does not seems to downmix 5.1ch audio to Stereo with "-ac 2" Message-ID: Hello list members, I'm trying to playback a mp4 video which contains 5.1ch audio with ffplay. The mp4 file was made from DVD stream dump with mplayer, like the following ffmpeg command. $ ffmpeg -i /Volumes/USB\ HD\ 1/stream.dump \ > -map 0:0 -c:v libx264 -preset slow -level 4.1 -profile:v High -crf 18 \ > -map 0:2 -c:a libfdk_aac -filter:a volume=4.0 \ > -ss 00:17:29.500 -t 00:01:49.500 \ > test.mp4 ffmpeg version N-67742-g3f07dd6 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 17 2014 09:27:26 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --prefix=/Volumes/ffmpeg_compile --pkg-config-flags=--static --disable-ffserver --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-filters --enable-runtime-cpudetect --enable-bzlib --enable-zlib --enable-libmp3lame --enable-libfdk-aac --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvorbis --enable-libspeex --enable-libopus --enable-libgsm --enable-libtwolame --enable-libsoxr --enable-libwavpack --enable-libmodplug --enable-libopenjpeg --enable-libwebp --enable-libtheora --enable-libx264 --enable-libx265 --enable-libxvid --enable-libvpx --enable-libxavs --enable-libfreetype --enable-fontconfig --enable-libfribidi --enable-libass --enable-libbluray --enable-libvidstab libavutil 54. 13.100 / 54. 13.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 13.100 / 56. 13.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, mpeg, from '/Volumes/USB HD 1/stream.dump': Duration: 01:53:51.81, start: 0.041500, bitrate: 8136 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x480 [SAR 32:27 DAR 16:9], max. 9800 kb/s, 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x81]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s Stream #0:2[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s [libx264 @ 0x7feffc022400] using SAR=32/27 [libx264 @ 0x7feffc022400] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7feffc022400] profile High, level 4.1 [libx264 @ 0x7feffc022400] 264 - core 142 r2491 24e4fed - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=5 deblock=1:0:0 analyse=0x3:0x113 me=umh subme=8 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 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=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=50 rc=crf mbtree=1 crf=18.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf56.13.100 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 720x480 [SAR 32:27 DAR 16:9], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc Metadata: encoder : Lavc56.12.101 libx264 Stream #0:1: Audio: aac (libfdk_aac) ([64][0][0][0] / 0x0040), 48000 Hz, 5.1, s16, 488 kb/s Metadata: encoder : Lavc56.12.101 libfdk_aac Stream mapping: Stream #0:0 -> #0:0 (mpeg2video (native) -> h264 (libx264)) Stream #0:2 -> #0:1 (ac3 (native) -> aac (libfdk_aac)) Press [q] to stop, [?] for help [mpeg @ 0x7feffc016800] New subtitle stream 0:3 at pos:29865998 and DTS:34.6427s [mpeg @ 0x7feffc016800] New subtitle stream 0:4 at pos:29868046 and DTS:34.6427s [mpeg @ 0x7feffc016800] New subtitle stream 0:5 at pos:146962446 and DTS:152.661s [mpeg @ 0x7feffc016800] New subtitle stream 0:6 at pos:146964494 and DTS:152.661s frame= 69 fps=3.9 q=24.0 size= 80kB time=00:00:02.28 bitrate= 286.4kbits/frame= 115 fps=6.4 q=24.0 size= 585kB time=00:00:03.86 bitrate=1240.7kbits/frame= 157 fps=8.5 q=24.0 size= 1033kB time=00:00:05.16 bitrate=1639.6kbits/frame= 196 fps= 10 q=24.0 size= 1477kB time=00:00:06.48 bitrate=1866.1kbits/frame= 235 fps= 12 q=24.0 size= 1869kB time=00:00:07.85 bitrate=1950.1kbits/frame= 278 fps= 14 q=24.0 size= 2442kB time=00:00:09.25 bitrate=2160.7kbits/frame= 322 fps= 16 q=24.0 size= 2998kB time=00:00:10.70 bitrate=2293.3kbits/frame= 364 fps= 17 q=24.0 size= 3481kB time=00:00:12.11 bitrate=2353.3kbits/frame= 405 fps= 19 q=24.0 size= 3993kB time=00:00:13.46 bitrate=2430.1kbits/frame= 450 fps= 20 q=24.0 size= 4526kB time=00:00:14.95 bitrate=2479.3kbits/frame= 495 fps= 22 q=24.0 size= 5041kB time=00:00:16.46 bitrate=2507.3kbits/frame= 537 fps= 23 q=24.0 size= 5522kB time=00:00:17.89 bitrate=2527.2kbits/frame= 578 fps= 24 q=24.0 size= 5984kB time=00:00:19.24 bitrate=2547.4kbits/frame= 624 fps= 26 q=24.0 size= 6517kB time=00:00:20.82 bitrate=2564.2kbits/frame= 667 fps= 27 q=24.0 size= 6972kB time=00:00:22.16 bitrate=2576.8kbits/frame= 705 fps= 28 q=24.0 size= 7425kB time=00:00:23.46 bitrate=2591.9kbits/frame= 750 fps= 29 q=24.0 size= 7915kB time=00:00:24.91 bitrate=2602.3kbits/frame= 793 fps= 30 q=24.0 size= 8364kB time=00:00:26.38 bitrate=2596.3kbits/frame= 837 fps= 31 q=24.0 size= 8873kB time=00:00:27.92 bitrate=2602.9kbits/frame= 878 fps= 32 q=24.0 size= 9363kB time=00:00:29.26 bitrate=2620.7kbits/frame= 915 fps= 33 q=24.0 size= 9806kB time=00:00:30.48 bitrate=2635.1kbits/frame= 957 fps= 34 q=24.0 size= 10281kB time=00:00:31.91 bitrate=2639.1kbits/frame= 997 fps= 35 q=24.0 size= 10737kB time=00:00:33.17 bitrate=2651.5kbits/frame= 1035 fps= 35 q=24.0 size= 11220kB time=00:00:34.47 bitrate=2666.1kbits/frame= 1077 fps= 36 q=24.0 size= 11771kB time=00:00:35.92 bitrate=2684.0kbits/frame= 1118 fps= 37 q=24.0 size= 12250kB time=00:00:37.29 bitrate=2691.0kbits/frame= 1160 fps= 38 q=24.0 size= 12737kB time=00:00:38.67 bitrate=2697.8kbits/frame= 1200 fps= 38 q=24.0 size= 13195kB time=00:00:39.97 bitrate=2703.8kbits/frame= 1239 fps= 39 q=24.0 size= 13638kB time=00:00:41.32 bitrate=2703.8kbits/frame= 1279 fps= 39 q=24.0 size= 14123kB time=00:00:42.60 bitrate=2715.6kbits/frame= 1324 fps= 40 q=24.0 size= 14680kB time=00:00:44.11 bitrate=2725.9kbits/frame= 1372 fps= 41 q=24.0 size= 15223kB time=00:00:45.65 bitrate=2731.7kbits/frame= 1414 fps= 42 q=24.0 size= 15648kB time=00:00:47.12 bitrate=2720.1kbits/frame= 1460 fps= 42 q=24.0 size= 16087kB time=00:00:48.66 bitrate=2708.2kbits/frame= 1500 fps= 43 q=24.0 size= 16534kB time=00:00:50.00 bitrate=2708.6kbits/frame= 1543 fps= 43 q=24.0 size= 17021kB time=00:00:51.43 bitrate=2710.9kbits/frame= 1582 fps= 44 q=24.0 size= 17451kB time=00:00:52.71 bitrate=2711.9kbits/frame= 1627 fps= 45 q=24.0 size= 17963kB time=00:00:54.18 bitrate=2715.6kbits/frame= 1665 fps= 45 q=24.0 size= 18407kB time=00:00:55.50 bitrate=2716.5kbits/frame= 1707 fps= 45 q=24.0 size= 18842kB time=00:00:56.93 bitrate=2710.9kbits/frame= 1748 fps= 46 q=24.0 size= 19312kB time=00:00:58.32 bitrate=2712.4kbits/frame= 1790 fps= 46 q=24.0 size= 19738kB time=00:00:59.66 bitrate=2709.9kbits/frame= 1832 fps= 47 q=24.0 size= 20145kB time=00:01:01.03 bitrate=2703.9kbits/frame= 1875 fps= 47 q=24.0 size= 20668kB time=00:01:02.48 bitrate=2709.7kbits/frame= 1908 fps= 48 q=24.0 size= 21041kB time=00:01:03.63 bitrate=2708.5kbits/frame= 1947 fps= 48 q=24.0 size= 21468kB time=00:01:04.93 bitrate=2708.2kbits/frame= 1987 fps= 48 q=24.0 size= 21893kB time=00:01:06.15 bitrate=2711.1kbits/frame= 2022 fps= 49 q=24.0 size= 22283kB time=00:01:07.43 bitrate=2707.0kbits/frame= 2065 fps= 49 q=24.0 size= 22744kB time=00:01:08.94 bitrate=2702.3kbits/frame= 2100 fps= 49 q=24.0 size= 23191kB time=00:01:10.03 bitrate=2712.5kbits/frame= 2140 fps= 50 q=24.0 size= 23640kB time=00:01:11.40 bitrate=2712.2kbits/frame= 2182 fps= 50 q=24.0 size= 24017kB time=00:01:12.72 bitrate=2705.4kbits/frame= 2221 fps= 50 q=24.0 size= 24378kB time=00:01:14.09 bitrate=2695.4kbits/frame= 2262 fps= 51 q=24.0 size= 24747kB time=00:01:15.43 bitrate=2687.5kbits/frame= 2303 fps= 51 q=24.0 size= 25207kB time=00:01:16.82 bitrate=2688.0kbits/frame= 2342 fps= 51 q=24.0 size= 25635kB time=00:01:18.10 bitrate=2688.9kbits/frame= 2382 fps= 51 q=24.0 size= 26089kB time=00:01:19.46 bitrate=2689.4kbits/frame= 2420 fps= 52 q=24.0 size= 26585kB time=00:01:20.66 bitrate=2700.0kbits/frame= 2458 fps= 52 q=24.0 size= 27008kB time=00:01:21.94 bitrate=2700.1kbits/frame= 2494 fps= 52 q=24.0 size= 27449kB time=00:01:23.17 bitrate=2703.4kbits/frame= 2535 fps= 52 q=24.0 size= 27939kB time=00:01:24.52 bitrate=2707.9kbits/frame= 2574 fps= 53 q=24.0 size= 28348kB time=00:01:25.86 bitrate=2704.5kbits/frame= 2613 fps= 53 q=24.0 size= 28774kB time=00:01:27.14 bitrate=2704.8kbits/frame= 2649 fps= 53 q=24.0 size= 29189kB time=00:01:28.36 bitrate=2706.1kbits/frame= 2686 fps= 53 q=24.0 size= 29635kB time=00:01:29.55 bitrate=2710.8kbits/frame= 2725 fps= 53 q=24.0 size= 30074kB time=00:01:30.92 bitrate=2709.6kbits/frame= 2760 fps= 54 q=24.0 size= 30499kB time=00:01:32.05 bitrate=2714.2kbits/frame= 2798 fps= 54 q=24.0 size= 30972kB time=00:01:33.33 bitrate=2718.4kbits/frame= 2835 fps= 54 q=24.0 size= 31395kB time=00:01:34.57 bitrate=2719.6kbits/frame= 2877 fps= 54 q=24.0 size= 31862kB time=00:01:36.02 bitrate=2718.3kbits/frame= 2918 fps= 54 q=24.0 size= 32312kB time=00:01:37.30 bitrate=2720.5kbits/frame= 2955 fps= 55 q=24.0 size= 32748kB time=00:01:38.53 bitrate=2722.5kbits/frame= 2994 fps= 55 q=24.0 size= 33179kB time=00:01:39.86 bitrate=2721.8kbits/frame= 3027 fps= 55 q=24.0 size= 33582kB time=00:01:40.97 bitrate=2724.6kbits/frame= 3067 fps= 55 q=24.0 size= 34076kB time=00:01:42.22 bitrate=2730.6kbits/frame= 3105 fps= 55 q=24.0 size= 34526kB time=00:01:43.57 bitrate=2730.8kbits/frame= 3144 fps= 55 q=24.0 size= 34958kB time=00:01:44.93 bitrate=2729.0kbits/frame= 3188 fps= 56 q=21.0 size= 35367kB time=00:01:46.38 bitrate=2723.3kbits/frame= 3226 fps= 56 q=24.0 size= 35822kB time=00:01:47.56 bitrate=2728.2kbits/frame= 3262 fps= 56 q=24.0 size= 36225kB time=00:01:48.77 bitrate=2728.1kbits/frame= 3282 fps= 55 q=-1.0 Lsize= 37302kB time=00:01:49.50 bitrate=2790.5kbits/s dup=657 drop=0 video:30647kB audio:6536kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.319933% [libx264 @ 0x7feffc022400] frame I:26 Avg QP:16.36 size: 63960 [libx264 @ 0x7feffc022400] frame P:987 Avg QP:19.48 size: 19693 [libx264 @ 0x7feffc022400] frame B:2269 Avg QP:22.33 size: 4532 [libx264 @ 0x7feffc022400] consecutive B-frames: 5.2% 1.9% 17.6% 75.3% [libx264 @ 0x7feffc022400] mb I I16..4: 1.5% 93.0% 5.5% [libx264 @ 0x7feffc022400] mb P I16..4: 0.0% 6.3% 0.2% P16..4: 36.0% 27.0% 17.1% 0.0% 0.0% skip:13.5% [libx264 @ 0x7feffc022400] mb B I16..4: 0.0% 0.4% 0.0% B16..8: 38.1% 7.0% 1.4% direct: 6.6% skip:46.4% L0:37.9% L1:39.5% BI:22.7% [libx264 @ 0x7feffc022400] 8x8 transform intra:95.5% inter:76.0% [libx264 @ 0x7feffc022400] direct mvs spatial:99.7% temporal:0.3% [libx264 @ 0x7feffc022400] coded y,uvDC,uvAC intra: 98.3% 97.7% 90.1% inter: 33.9% 43.8% 11.2% [libx264 @ 0x7feffc022400] i16 v,h,dc,p: 49% 11% 11% 29% [libx264 @ 0x7feffc022400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 11% 12% 8% 10% 12% 10% 11% 11% [libx264 @ 0x7feffc022400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 10% 8% 2% 9% 17% 17% 14% 13% 10% [libx264 @ 0x7feffc022400] i8c dc,h,v,p: 52% 15% 16% 17% [libx264 @ 0x7feffc022400] Weighted P-Frames: Y:3.9% UV:0.9% [libx264 @ 0x7feffc022400] ref P L0: 38.7% 14.0% 27.6% 9.0% 8.3% 2.3% 0.1% [libx264 @ 0x7feffc022400] ref B L0: 59.3% 30.2% 7.2% 3.2% [libx264 @ 0x7feffc022400] ref B L1: 90.6% 9.4% [libx264 @ 0x7feffc022400] kb/s:2292.56 To playback full audio of test.mp4 with ffplay, I need to use pan filter. $ ffplay -i test.mp4 -af "pan=stereo|FL Message-ID: Kimio Miyamura apost.plala.or.jp> writes: > If I use "-ac 2" with ffplay like the following, > seems to right side channel sound have lost. I cannot reproduce this with a random sample here... Do you see anything suspicious if you press "w" while playing to see a visualization? > $ ffplay -i test.mp4 -ac 2 Please test the following: $ ffplay -i test.mp4 -af aresample=ocl=stereo Carl Eugen From xanadu at apost.plala.or.jp Wed Nov 19 14:38:41 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Wed, 19 Nov 2014 22:38:41 +0900 Subject: [FFmpeg-user] ffplay does not seems to downmix 5.1ch audio to Stereo with "-ac 2" In-Reply-To: References: Message-ID: <578EC50D-18E3-4EEA-ACAF-5D14392CFCED@apost.plala.or.jp> 2014/11/19 21:46, Carl Eugen Hoyos ag.or.at> wrote: > Kimio Miyamura apost.plala.or.jp> writes: > >> If I use "-ac 2" with ffplay like the following, >> seems to right side channel sound have lost. > > Do you see anything suspicious if you press "w" while > playing to see a visualization? Yes, the video would be lost. pressing "w" one time, the video goes to black with white and purple oblique line. pressing "w" again, the video goes to complete black, pressing "w" one more time, the video comes back. Am I correctly describe the video state? >> $ ffplay -i test.mp4 -ac 2 > > Please test the following: > $ ffplay -i test.mp4 -af aresample=ocl=stereo Nice!! With above command, I can playback all audio correctly. I'm happy aren't I? So if I can playback audio with above ffplay command, don't I need to downmix audio at the timing of encoding? // Miya From oinos at web.de Wed Nov 19 18:31:08 2014 From: oinos at web.de (=?UTF-8?B?UGFibG8gUm9kcsOtZ3Vleg==?=) Date: Wed, 19 Nov 2014 18:31:08 +0100 Subject: [FFmpeg-user] error compressing Flash video Message-ID: <546CD3DC.3040906@web.de> Hi there, I recorded an screencast in WinXP using the following command: ffmpeg -f gdigrab -framerate 5 -i desktop -f dshow -i audio="Realtek HD Audio Input" -vcodec libx264 -crf 0 -acodec pcm_s16le output.flv It worked fine and I stopped it pressing Ctrl+C in the command-line window. I tried to convert it, but I got an error: >ffmpeg -i output.flv -acodec libvo_aacenc -vcodec h264 -pix_fmt yuv420p output.flv.flv ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 Input #0, flv, from 'output.flv': Metadata: encoder : Lavf56.4.101 Duration: 00:03:03.40, start: 0.000000, bitrate: 1605 kb/s Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p, 1280x1024, 5 fps, 5 tbr, 1k tbn, 10 tbc Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s [libx264 @ 02edf020] using cpu capabilities: MMX2 SSE2 SSE3 Cache64 [libx264 @ 02edf020] profile High, level 3.2 [libx264 @ 02edf020] 264 - core 142 r2479 dd79a61 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=5 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, flv, to 'output.flv.flv': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1280x1024, q=-1--1, 5 fps, 1k tbn, 5 tbc Metadata: encoder : Lavc56.1.100 libx264 Stream #0:1: Audio: aac (libvo_aacenc) ([10][0][0][0] / 0x000A), 44100 Hz, s tereo, s16, 128 kb/s Metadata: encoder : Lavc56.1.100 libvo_aacenc Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (libvo_aacenc)) Press [q] to stop, [?] for help frame= 8 fps=0.0 q=0.0 size= 0kB time=00:00:02.48 bitrate= 1.3kbits/s frame= 17 fps= 16 q=0.0 size= 0kB time=00:00:04.48 bitrate= 0.7kbits/s frame= 26 fps= 16 q=0.0 size= 0kB time=00:00:06.49 bitrate= 0.5kbits/s frame= 36 fps= 17 q=0.0 size= 0kB time=00:00:08.51 bitrate= 0.4kbits/s frame= 45 fps= 17 q=0.0 size= 1kB time=00:00:09.99 bitrate= 0.6kbits/s frame= 46 fps= 12 q=0.0 size= 9kB time=00:00:10.49 bitrate= 6.8kbits/s [flv @ 03ca2020] Packets are not in the proper order with respect to DTS av_interleaved_write_frame(): Invalid argument frame= 48 fps=6.1 q=-1.0 Lsize= 173kB time=00:00:10.54 bitrate= 134.2kbits/s video:133kB audio:165kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [libx264 @ 02edf020] frame I:1 Avg QP: 4.93 size:135384 [libx264 @ 02edf020] frame P:12 Avg QP:12.46 size: 394 [libx264 @ 02edf020] frame B:35 Avg QP: 7.82 size: 69 [libx264 @ 02edf020] consecutive B-frames: 2.1% 0.0% 6.3% 91.7% [libx264 @ 02edf020] mb I I16..4: 76.3% 0.8% 22.9% [libx264 @ 02edf020] mb P I16..4: 0.0% 0.0% 0.0% P16..4: 0.4% 0.0% 0.0% 0.0% 0.0% skip:99.6% [libx264 @ 02edf020] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.2% 0.0% 0.0% direct: 0.0% skip:99.8% L0:16.2% L1:83.8% BI: 0.0% [libx264 @ 02edf020] 8x8 transform intra:0.8% inter:5.6% [libx264 @ 02edf020] coded y,uvDC,uvAC intra: 15.4% 27.2% 26.9% inter: 0.0% 0.0% 0.0% [libx264 @ 02edf020] i16 v,h,dc,p: 93% 5% 2% 0% [libx264 @ 02edf020] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 16% 30% 34% 3% 2% 1% 4% 6% 2% [libx264 @ 02edf020] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 30% 26% 2% 3% 3% 4% 2% 4% [libx264 @ 02edf020] i8c dc,h,v,p: 80% 13% 6% 1% [libx264 @ 02edf020] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 02edf020] ref P L0: 92.8% 0.6% 4.0% 2.6% [libx264 @ 02edf020] ref B L0: 68.2% 31.8% [libx264 @ 02edf020] kb/s:111.77 Conversion failed! Sorry, but I?m afraid that I don?t know why this goes wrong. Many thanks for your help, Pablo -- http://www.ousia.tk From paynito at outlook.com Wed Nov 19 11:53:34 2014 From: paynito at outlook.com (paynito) Date: Wed, 19 Nov 2014 02:53:34 -0800 (PST) Subject: [FFmpeg-user] brew install ffmpeg on 10.10 Yosemite for wma to mp3 batch conv. Message-ID: <1416394414529-4668194.post@n4.nabble.com> bash // brew install ffmpeg // seems to work cd folder with wmas for f in *.wma; do ffmpeg -y -i "$f" -c:a libfdk_aac -b:a 192k "${f%.wma}.m4a"; done; // try a script -> error Unknown encoder 'libfdk_aac' any ideas? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/brew-install-ffmpeg-on-10-10-Yosemite-for-wma-to-mp3-batch-conv-tp4668194.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From thanos_n7 at yahoo.com Wed Nov 19 09:17:19 2014 From: thanos_n7 at yahoo.com (Thanos Natsiopoulos) Date: Wed, 19 Nov 2014 08:17:19 +0000 (UTC) Subject: [FFmpeg-user] how to install and use FFmpeg Message-ID: <460929487.860078.1416385039895.JavaMail.yahoo@jws10734.mail.gq1.yahoo.com> Hello, ?? Reading about FFmpeg I saw that I can create a multiplexed transport stream with it. I am completely new to this environment and I tried to follow the guides that I found online but without success.?? May I please have a simple guide for a begginer user on how to install and create the multiplexed transport stream? I can have access to both windows and ubuntu. ?? Thank you in advance. Kind regards,Thanos From lou at lrcd.com Wed Nov 19 20:26:51 2014 From: lou at lrcd.com (Lou) Date: Wed, 19 Nov 2014 10:26:51 -0900 Subject: [FFmpeg-user] brew install ffmpeg on 10.10 Yosemite for wma to mp3 batch conv. In-Reply-To: <1416394414529-4668194.post@n4.nabble.com> References: <1416394414529-4668194.post@n4.nabble.com> Message-ID: <20141119102651.5732cc7d@lrcd.com> On Wed, 19 Nov 2014 02:53:34 -0800 (PST) paynito wrote: > bash // > brew install ffmpeg // seems to work > cd folder with wmas > for f in *.wma; do ffmpeg -y -i "$f" -c:a libfdk_aac -b:a 192k > "${f%.wma}.m4a"; done; > // try a script > -> error Unknown encoder 'libfdk_aac' > > > any ideas? I do not believe brew includes libfdk_aac support by default. As far as I know you have to use: brew install ffmpeg --with-fdk-aac Also see: From lou at lrcd.com Wed Nov 19 20:38:35 2014 From: lou at lrcd.com (Lou) Date: Wed, 19 Nov 2014 10:38:35 -0900 Subject: [FFmpeg-user] how to install and use FFmpeg In-Reply-To: <460929487.860078.1416385039895.JavaMail.yahoo@jws10734.mail.gq1.yahoo.com> References: <460929487.860078.1416385039895.JavaMail.yahoo@jws10734.mail.gq1.yahoo.com> Message-ID: <20141119103835.74a41a87@lrcd.com> On Wed, 19 Nov 2014 08:17:19 +0000 (UTC) Thanos Natsiopoulos wrote: > May I please have a simple guide for a begginer user on how to install Simplest method is to download, extract, and execute a static build: Linux: http://johnvansickle.com/ffmpeg/ Windows: http://ffmpeg.zeranoe.com/builds/ From radpopl at gmail.com Wed Nov 19 20:58:24 2014 From: radpopl at gmail.com (radpopl) Date: Wed, 19 Nov 2014 11:58:24 -0800 (PST) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: Hello, there is a sample. https://www.sendspace.com/file/leir3w This happens only with this mts format (1440x1080, 50i) with my camera. 1920x1080 50p converts well (of course without the option interlace). Regards, Radek On Tue, Nov 18, 2014 at 10:59 PM, Carl Eugen Hoyos [via FFmpeg-users] < ml-node+s933282n4668188h98 at n4.nabble.com> wrote: > radpopl gmail.com> writes: > > > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > > "%%~dF%%~pF%%~nF.avi" > > > > but ffmpeg crashed. > > Please provide the input sample. > (I cannot reproduce with a random sample.) > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > [hidden email] > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668188.html > To unsubscribe from H264 (mts) interlaced to XVID interlaced, click here > > . > NAML > > -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668202.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From radpopl at gmail.com Wed Nov 19 21:16:22 2014 From: radpopl at gmail.com (=?UTF-8?B?UmFkb3PFgmF3IFBvcMWCYXdza2k=?=) Date: Wed, 19 Nov 2014 21:16:22 +0100 Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: I tried to use VirtualDub - getting the message out there, such as in the attached image. On Wed, Nov 19, 2014 at 8:58 PM, radpopl wrote: > Hello, > > there is a sample. > https://www.sendspace.com/file/leir3w > > This happens only with this mts format (1440x1080, 50i) with my camera. > 1920x1080 50p converts well (of course without the option interlace). > > Regards, > Radek > > > On Tue, Nov 18, 2014 at 10:59 PM, Carl Eugen Hoyos [via FFmpeg-users] < > ml-node+s933282n4668188h98 at n4.nabble.com> wrote: > > > radpopl gmail.com> writes: > > > > > ffmpeg.exe -n -i "%%F" -c:v mpeg4 -qscale:v 4 -vtag xvid > > > -flags +qpel *+ildct+ilme* -acodec libmp3lame -b:a 320k > > > "%%~dF%%~pF%%~nF.avi" > > > > > > but ffmpeg crashed. > > > > Please provide the input sample. > > (I cannot reproduce with a random sample.) > > > > Carl Eugen > > > > _______________________________________________ > > ffmpeg-user mailing list > > [hidden email] > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > ------------------------------ > > If you reply to this email, your message will be added to the discussion > > below: > > > > > http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668188.html > > To unsubscribe from H264 (mts) interlaced to XVID interlaced, click here > > < > http://ffmpeg-users.933282.n4.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=4668119&code=cmFkcG9wbEBnbWFpbC5jb218NDY2ODExOXwxMTMwNzIwMDI3 > > > > . > > NAML > > < > http://ffmpeg-users.933282.n4.nabble.com/template/NamlServlet.jtp?macro=macro_viewer&id=instant_html%21nabble%3Aemail.naml&base=nabble.naml.namespaces.BasicNamespace-nabble.view.web.template.NabbleNamespace-nabble.view.web.template.NodeNamespace&breadcrumbs=notify_subscribers%21nabble%3Aemail.naml-instant_emails%21nabble%3Aemail.naml-send_instant_email%21nabble%3Aemail.naml > > > > > > > > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668202.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -------------- next part -------------- A non-text attachment was scrubbed... Name: virtualdub-ffmpeg.png Type: image/png Size: 39183 bytes Desc: not available URL: From cehoyos at ag.or.at Thu Nov 20 01:22:14 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 00:22:14 +0000 (UTC) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: radpopl gmail.com> writes: > there is a sample. > https://www.sendspace.com/file/leir3w I opened ticket #4121, thank you for the sample! https://trac.ffmpeg.org/ticket/4121 Please do not top-post here, Carl Eugen From dave at dericed.com Thu Nov 20 02:22:50 2014 From: dave at dericed.com (Dave Rice) Date: Wed, 19 Nov 2014 20:22:50 -0500 Subject: [FFmpeg-user] make error with --enable-decklink Message-ID: <1238FA57-D623-4A13-B22D-64CB11A0FDC1@dericed.com> Hi all, I?m trying to build ffmpeg with --enable-decklink but get an error during make. For a c++ compiler I have g++. Any advice that would help get this built would be appreciated. u813s:ffmpeg rice$ make CC libavdevice/alldevices.o CC libavdevice/avdevice.o CC libavdevice/avfoundation.o CXX libavdevice/decklink_common.o error: invalid argument '-std=c99' not allowed with 'C++/ObjC++' make: *** [libavdevice/decklink_common.o] Error 1 full version: u813s:ffmpeg rice$ ./configure --enable-gpl --enable-decklink --extra-cflags="-I/usr/local/include" --extra-ldflags="-L/usr/local/lib" Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 install prefix /usr/local source path . C compiler gcc C library ARCH x86 (generic) big-endian no runtime cpu detection yes yasm yes MMX enabled yes MMXEXT enabled yes 3DNow! enabled yes 3DNow! extended enabled yes SSE enabled yes SSSE3 enabled yes AVX enabled yes XOP enabled yes FMA3 enabled yes FMA4 enabled yes i686 features enabled yes CMOV is fast yes EBX available yes EBP available yes debug symbols yes strip symbols yes optimize for size no optimizations yes static yes shared no postprocessing support yes new filter support yes network support yes threading support pthreads safe bitstream reader yes SDL support yes opencl enabled no texi2html enabled yes perl enabled yes pod2man enabled yes makeinfo enabled yes makeinfo supports HTML no External libraries: bzlib iconv zlib decklink sdl Enabled decoders: aac bink gsm_ms aac_latm binkaudio_dct h261 aasc binkaudio_rdft h263 ac3 bintext h263i ac3_fixed bmp h263p adpcm_4xm bmv_audio h264 adpcm_adx bmv_video h264_vda adpcm_afc brender_pix hevc adpcm_ct c93 hnm4_video adpcm_dtk cavs huffyuv adpcm_ea cdgraphics iac adpcm_ea_maxis_xa cdxl idcin adpcm_ea_r1 cinepak idf adpcm_ea_r2 cljr iff_byterun1 adpcm_ea_r3 cllc iff_ilbm adpcm_ea_xas comfortnoise imc adpcm_g722 cook indeo2 adpcm_g726 cpia indeo3 adpcm_g726le cscd indeo4 adpcm_ima_amv cyuv indeo5 adpcm_ima_apc dca interplay_dpcm adpcm_ima_dk3 dfa interplay_video adpcm_ima_dk4 dirac jacosub adpcm_ima_ea_eacs dnxhd jpeg2000 adpcm_ima_ea_sead dpx jpegls adpcm_ima_iss dsd_lsbf jv adpcm_ima_oki dsd_lsbf_planar kgv1 adpcm_ima_qt dsd_msbf kmvc adpcm_ima_rad dsd_msbf_planar lagarith adpcm_ima_smjpeg dsicinaudio loco adpcm_ima_wav dsicinvideo mace3 adpcm_ima_ws dvbsub mace6 adpcm_ms dvdsub mdec adpcm_sbpro_2 dvvideo metasound adpcm_sbpro_3 dxa microdvd adpcm_sbpro_4 dxtory mimic adpcm_swf eac3 mjpeg adpcm_thp eacmv mjpegb adpcm_vima eamad mlp adpcm_xa eatgq mmvideo adpcm_yamaha eatgv motionpixels aic eatqi movtext alac eightbps mp1 alias_pix eightsvx_exp mp1float als eightsvx_fib mp2 amrnb escape124 mp2float amrwb escape130 mp3 amv evrc mp3adu anm exr mp3adufloat ansi ffv1 mp3float ape ffvhuff mp3on4 ass ffwavesynth mp3on4float asv1 fic mpc7 asv2 flac mpc8 atrac1 flashsv mpeg1video atrac3 flashsv2 mpeg2video atrac3p flic mpeg4 aura flv mpegvideo aura2 fourxm mpl2 avrn fraps msa1 avrp frwu msmpeg4v1 avs g2m msmpeg4v2 avui g723_1 msmpeg4v3 ayuv g729 msrle bethsoftvid gif mss1 bfi gsm mss2 msvideo1 qpeg txd mszh qtrle ulti mts2 r10k utvideo mvc1 r210 v210 mvc2 ra_144 v210x mxpeg ra_288 v308 nellymoser ralf v408 nuv rawvideo v410 on2avc realtext vb opus rl2 vble paf_audio roq vc1 paf_video roq_dpcm vc1image pam rpza vcr1 pbm rv10 vima pcm_alaw rv20 vmdaudio pcm_bluray rv30 vmdvideo pcm_dvd rv40 vmnc pcm_f32be s302m vorbis pcm_f32le sami vp3 pcm_f64be sanm vp5 pcm_f64le sgi vp6 pcm_lxf sgirle vp6a pcm_mulaw shorten vp6f pcm_s16be sipr vp7 pcm_s16be_planar smackaud vp8 pcm_s16le smacker vp9 pcm_s16le_planar smc vplayer pcm_s24be smvjpeg vqa pcm_s24daud snow wavpack pcm_s24le sol_dpcm webp pcm_s24le_planar sonic webvtt pcm_s32be sp5x wmalossless pcm_s32le srt wmapro pcm_s32le_planar ssa wmav1 pcm_s8 stl wmav2 pcm_s8_planar subrip wmavoice pcm_u16be subviewer wmv1 pcm_u16le subviewer1 wmv2 pcm_u24be sunrast wmv3 pcm_u24le svq1 wmv3image pcm_u32be svq3 wnv1 pcm_u32le tak ws_snd1 pcm_u8 targa xan_dpcm pcm_zork targa_y216 xan_wc3 pcx text xan_wc4 pgm theora xbin pgmyuv thp xbm pgssub tiertexseqvideo xface pictor tiff xl pjs tmv xsub png truehd xwd ppm truemotion1 y41p prores truemotion2 yop prores_lgpl truespeech yuv4 ptx tscc zero12v qcelp tscc2 zerocodec qdm2 tta zlib qdraw twinvq zmbv Enabled encoders: a64multi jpegls prores a64multi5 ljpeg prores_aw aac mjpeg prores_ks ac3 movtext qtrle ac3_fixed mp2 r10k adpcm_adx mp2fixed r210 adpcm_g722 mpeg1video ra_144 adpcm_g726 mpeg2video rawvideo adpcm_ima_qt mpeg4 roq adpcm_ima_wav msmpeg4v2 roq_dpcm adpcm_ms msmpeg4v3 rv10 adpcm_swf msvideo1 rv20 adpcm_yamaha nellymoser s302m alac pam sgi alias_pix pbm snow amv pcm_alaw sonic ass pcm_f32be sonic_ls asv1 pcm_f32le srt asv2 pcm_f64be ssa avrp pcm_f64le subrip avui pcm_mulaw sunrast ayuv pcm_s16be svq1 bmp pcm_s16be_planar targa cinepak pcm_s16le tiff cljr pcm_s16le_planar tta comfortnoise pcm_s24be utvideo dca pcm_s24daud v210 dnxhd pcm_s24le v308 dpx pcm_s24le_planar v408 dvbsub pcm_s32be v410 dvdsub pcm_s32le vorbis dvvideo pcm_s32le_planar wavpack eac3 pcm_s8 webvtt ffv1 pcm_s8_planar wmav1 ffvhuff pcm_u16be wmav2 flac pcm_u16le wmv1 flashsv pcm_u24be wmv2 flashsv2 pcm_u24le xbm flv pcm_u32be xface g723_1 pcm_u32le xsub gif pcm_u8 xwd h261 pcx y41p h263 pgm yuv4 h263p pgmyuv zlib huffyuv png zmbv jpeg2000 ppm Enabled hwaccels: h264_vda h264_vda_old Enabled parsers: aac dvd_nav mpegvideo aac_latm dvdsub opus ac3 flac png adx gsm pnm bmp h261 rv30 cavsvideo h263 rv40 cook h264 tak dca hevc vc1 dirac mjpeg vorbis dnxhd mlp vp3 dpx mpeg4video vp8 dvbsub mpegaudio vp9 Enabled demuxers: aac h263 nc ac3 h264 nistsphere act hevc nsv adf hls nut adp hnm nuv adx ico ogg aea idcin oma afc idf paf aiff iff pcm_alaw amr ilbc pcm_f32be anm image2 pcm_f32le apc image2_alias_pix pcm_f64be ape image2_brender_pix pcm_f64le aqtitle image2pipe pcm_mulaw asf image_bmp_pipe pcm_s16be ass image_dpx_pipe pcm_s16le ast image_exr_pipe pcm_s24be au image_j2k_pipe pcm_s24le avi image_jpeg_pipe pcm_s32be avr image_jpegls_pipe pcm_s32le avs image_pictor_pipe pcm_s8 bethsoftvid image_png_pipe pcm_u16be bfi image_sgi_pipe pcm_u16le bink image_sunrast_pipe pcm_u24be bintext image_tiff_pipe pcm_u24le bit image_webp_pipe pcm_u32be bmv ingenient pcm_u32le boa ipmovie pcm_u8 brstm ircam pjs c93 iss pmp caf iv8 pva cavsvideo ivf pvf cdg jacosub qcp cdxl jv r3d cine latm rawvideo concat live_flv realtext data lmlm4 redspark daud loas rl2 dfa lrc rm dirac lvf roq dnxhd lxf rpl dsf m4v rsd dsicin matroska rso dts mgsts rtp dtshd microdvd rtsp dv mjpeg sami dxa mlp sap ea mlv sbg ea_cdata mm sdp eac3 mmf sdr2 epaf mov segafilm ffm mp3 shorten ffmetadata mpc siff filmstrip mpc8 sln flac mpegps smacker flic mpegts smjpeg flv mpegtsraw smush fourxm mpegvideo sol frm mpl2 sox g722 mpsub spdif g723_1 msnwc_tcp srt g729 mtv stl gif mv str gsm mvi subviewer gxf mxf subviewer1 h261 mxg sup swf vc1t webvtt tak vivo wsaud tedcaptions vmd wsvqa thp vobsub wtv tiertexseq voc wv tmv vplayer xa truehd vqf xbin tta w64 xmv tty wav xwma txd wc3 yop vc1 webm_dash_manifest yuv4mpegpipe Enabled muxers: a64 image2pipe pcm_s24be ac3 ipod pcm_s24le adts ircam pcm_s32be adx ismv pcm_s32le aiff ivf pcm_s8 amr jacosub pcm_u16be asf latm pcm_u16le asf_stream lrc pcm_u24be ass m4v pcm_u24le ast matroska pcm_u32be au matroska_audio pcm_u32le avi md5 pcm_u8 avm2 microdvd psp bit mjpeg rawvideo caf mkvtimestamp_v2 rm cavsvideo mlp roq crc mmf rso dash mov rtp data mp2 rtsp daud mp3 sap dirac mp4 segment dnxhd mpeg1system smjpeg dts mpeg1vcd smoothstreaming dv mpeg1video sox eac3 mpeg2dvd spdif f4v mpeg2svcd spx ffm mpeg2video srt ffmetadata mpeg2vob stream_segment filmstrip mpegts swf flac mpjpeg tee flv mxf tg2 framecrc mxf_d10 tgp framemd5 null truehd g722 nut uncodedframecrc g723_1 oga vc1 gif ogg vc1t gxf oma voc h261 opus w64 h263 pcm_alaw wav h264 pcm_f32be webm hds pcm_f32le webm_dash_manifest hevc pcm_f64be webp hls pcm_f64le webvtt ico pcm_mulaw wtv ilbc pcm_s16be wv image2 pcm_s16le yuv4mpegpipe Enabled protocols: cache http rtp concat httpproxy srtp crypto icecast subfile data md5 tcp ffrtmphttp mmsh udp file mmst udplite ftp pipe unix gopher rtmp hls rtmpt Enabled filters: adelay dejudder owdenoise aecho delogo pad aeval deshake pan aevalsrc drawbox perms afade drawgrid perspective aformat earwax phase ainterleave ebur128 pixdesctest allpass edgedetect pp alphaextract elbg psnr alphamerge equalizer pullup amerge extractplanes removelogo amix fade replaygain amovie field rgbtestsrc anull fieldmatch rotate anullsink fieldorder sab anullsrc flanger scale apad format select aperms fps sendcmd aphaser framepack separatefields aresample framestep setdar aselect geq setfield asendcmd gradfun setpts asetnsamples haldclut setsar asetpts haldclutsrc settb asetrate hflip showcqt asettb highpass showinfo ashowinfo histeq showspectrum asplit histogram showwaves astats hqdn3d shuffleplanes astreamsync hqx signalstats atempo hue silencedetect atrim idet silenceremove avectorscope il sine bandpass interlace smartblur bandreject interleave smptebars bass join smptehdbars bbox kerndeint split biquad lenscorrection spp blackdetect life stereo3d blackframe lowpass super2xsai blend lut swapuv boxblur lut3d telecine cellauto lutrgb testsrc channelmap lutyuv thumbnail channelsplit mandelbrot tile codecview mcdeint tinterlace color mergeplanes transpose colorbalance movie treble colorchannelmixer mp trim colormatrix mpdecimate unsharp compand mptestsrc vflip concat negate vignette copy noformat volume crop noise volumedetect cropdetect null w3fdif curves nullsink xbr dctdnoiz nullsrc yadif decimate overlay zoompan Enabled bsfs: aac_adtstoasc imx_dump_header mp3_header_decompress chomp mjpeg2jpeg noise dump_extradata mjpega_dump_header remove_extradata h264_mp4toannexb mov2textsub text2movsub Enabled indevs: avfoundation lavfi decklink qtkit Enabled outdevs: decklink sdl License: GPL version 2 or later Creating config.mak, config.h, and doc/config.texi... config.asm is unchanged libavutil/avconfig.h is unchanged u813s:ffmpeg rice$ make CC libavdevice/alldevices.o CC libavdevice/avdevice.o CC libavdevice/avfoundation.o CXX libavdevice/decklink_common.o error: invalid argument '-std=c99' not allowed with 'C++/ObjC++' make: *** [libavdevice/decklink_common.o] Error 1 Thanks, Dave Rice From paynito at outlook.com Thu Nov 20 05:27:54 2014 From: paynito at outlook.com (Brandon Payne) Date: Thu, 20 Nov 2014 11:27:54 +0700 Subject: [FFmpeg-user] brew install ffmpeg on 10.10 Yosemite for wma to mp3 batch conv. Message-ID: Thanks of course I copied that off a discussion of bringing wma into iTunes. Now I see the .m4a so it should work for mp3 through homebrew without an installation flag. Cool. The end use is actually to listen to these tracks on a wp8 phone ??. But from a mac music gets to the phone from the iTunes library. ________________________________ From: Lou Sent: ?20/?11/?2014 02:27 To: ffmpeg-user at ffmpeg.org Cc: paynito Subject: Re: [FFmpeg-user] brew install ffmpeg on 10.10 Yosemite for wma to mp3 batch conv. On Wed, 19 Nov 2014 02:53:34 -0800 (PST) paynito wrote: > bash // > brew install ffmpeg // seems to work > cd folder with wmas > for f in *.wma; do ffmpeg -y -i "$f" -c:a libfdk_aac -b:a 192k > "${f%.wma}.m4a"; done; > // try a script > -> error Unknown encoder 'libfdk_aac' > > > any ideas? I do not believe brew includes libfdk_aac support by default. As far as I know you have to use: brew install ffmpeg --with-fdk-aac Also see: _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bobm-ffmpeg at burner.com Thu Nov 20 07:58:36 2014 From: bobm-ffmpeg at burner.com (Bob Maple) Date: Wed, 19 Nov 2014 23:58:36 -0700 Subject: [FFmpeg-user] Codec "same" as opposed to "copy"? Message-ID: <546D911C.3010403@burner.com> I'm trying to do some audio channel processing, stripping off a bunch of audio channels from an audio stream with 16 channels, taking the first 2 channels and turning them into 2 mono streams, dumping the remaining 14 channels: ffmpeg -i manyaudio.mov -map 0:v -map 0:a -map 0:a \ -codec:v copy -codec:a copy \ -map_channel 0.1.0:0.1.0 -map_channel 0.1.1:0.2.0 \ dualmono.mov However when I use '-codec:a copy' it ignores my channel mapping and I wind up with 2 x 16-channel streams. My source's audio codec will typically be either pcm_s16le or pcm_s24le, but ideally I don't want to care and want to say "use the same codec as the source." Is there a way to accomplish this that I'm missing, so I don't have to probe the source file first to determine the codec in order to specifically pass it along? From radpopl at gmail.com Thu Nov 20 09:49:32 2014 From: radpopl at gmail.com (radpopl) Date: Thu, 20 Nov 2014 00:49:32 -0800 (PST) Subject: [FFmpeg-user] H264 (mts) interlaced to XVID interlaced In-Reply-To: References: <1416055135474-4668119.post@n4.nabble.com> <006901d0019d$03e83ff0$0bb8bfd0$@gmail.com> Message-ID: <546DAC79.2000107@gmail.com> W dniu 2014-11-20 01:16, Carl Eugen Hoyos [via FFmpeg-users] pisze: > radpopl gmail.com> writes: > > > there is a sample. > > https://www.sendspace.com/file/leir3w > > I opened ticket #4121, thank you for the sample! > https://trac.ffmpeg.org/ticket/4121 OK, thank you for answer. > Please do not top-post here, Carl Eugen Sorry, it was not supposed to be sent to the list. :). Regards, rp -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/H264-mts-interlaced-to-XVID-interlaced-tp4668119p4668208.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Thu Nov 20 10:44:36 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 09:44:36 +0000 (UTC) Subject: [FFmpeg-user] make error with --enable-decklink References: <1238FA57-D623-4A13-B22D-64CB11A0FDC1@dericed.com> Message-ID: Dave Rice dericed.com> writes: > CXX libavdevice/decklink_common.o > error: invalid argument '-std=c99' not allowed with 'C++/ObjC++' > make: *** [libavdevice/decklink_common.o] Error 1 I only get a warning here with gcc 4.7 and gcc 4.9.1. What does "g++ -v" show for you? Carl Eugen From cehoyos at ag.or.at Thu Nov 20 10:47:56 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 09:47:56 +0000 (UTC) Subject: [FFmpeg-user] Codec "same" as opposed to "copy"? References: <546D911C.3010403@burner.com> Message-ID: Bob Maple burner.com> writes: > ffmpeg -i manyaudio.mov -map 0:v -map 0:a -map 0:a \ > -codec:v copy -codec:a copy \ > -map_channel 0.1.0:0.1.0 -map_channel 0.1.1:0.2.0 \ > dualmono.mov > > However when I use '-codec:a copy' it ignores my channel > mapping and I wind up with 2 x 16-channel streams. Yes, the channel mapping cannot work with -codec copy > My source's audio codec will typically be either > pcm_s16le or pcm_s24le, but ideally I don't want to care > and want to say "use the same codec as the source." This is not implemented, patch (or enhancement request) welcome. Carl Eugen From cehoyos at ag.or.at Thu Nov 20 10:52:16 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 09:52:16 +0000 (UTC) Subject: [FFmpeg-user] error compressing Flash video References: <546CD3DC.3040906@web.de> Message-ID: Pablo Rodr?guez web.de> writes: > I tried to convert it, but I got an error: > > $ ffmpeg -i output.flv -acodec libvo_aacenc -vcodec h264 > -pix_fmt yuv420p output.flv.flv Please provide output.flv. Carl Eugen From dave at dericed.com Thu Nov 20 13:20:33 2014 From: dave at dericed.com (Dave Rice) Date: Thu, 20 Nov 2014 07:20:33 -0500 Subject: [FFmpeg-user] make error with --enable-decklink In-Reply-To: References: <1238FA57-D623-4A13-B22D-64CB11A0FDC1@dericed.com> Message-ID: <3422C27C-EA90-477D-B4E1-FE71D94FFFBF@dericed.com> > On Nov 20, 2014, at 4:44 AM, Carl Eugen Hoyos wrote: > > Dave Rice dericed.com> writes: > >> CXX libavdevice/decklink_common.o >> error: invalid argument '-std=c99' not allowed with 'C++/ObjC++' >> make: *** [libavdevice/decklink_common.o] Error 1 > > I only get a warning here with gcc 4.7 and gcc 4.9.1. > What does "g++ -v" show for you? g++ -v Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.10.sdk/usr/include/c++/4.2.1 Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) Target: x86_64-apple-darwin14.0.0 Thread model: posix Dave From cehoyos at ag.or.at Thu Nov 20 13:41:56 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 12:41:56 +0000 (UTC) Subject: [FFmpeg-user] make error with --enable-decklink References: <1238FA57-D623-4A13-B22D-64CB11A0FDC1@dericed.com> <3422C27C-EA90-477D-B4E1-FE71D94FFFBF@dericed.com> Message-ID: Dave Rice dericed.com> writes: > >> error: invalid argument '-std=c99' not allowed with 'C++/ObjC++' > Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) I opened ticket #4124. Work-around is to use "make V=1", copy the commands, remove -std=c99 and continue with make. Carl Eugen From jazzman at misalpina.net Thu Nov 20 13:46:22 2014 From: jazzman at misalpina.net (Claudiu Rad) Date: Thu, 20 Nov 2014 14:46:22 +0200 Subject: [FFmpeg-user] Codec "same" as opposed to "copy"? In-Reply-To: References: <546D911C.3010403@burner.com> Message-ID: <546DE29E.40807@misalpina.net> On 11/20/2014 11:47 AM, Carl Eugen Hoyos wrote: >> My source's audio codec will typically be either >> pcm_s16le or pcm_s24le, but ideally I don't want to care >> and want to say "use the same codec as the source." > This is not implemented, patch (or enhancement request) > welcome. without trying to hijack the thread, sorry if i force it a bit but it is quite related: is there a possibility (or is anything foreseen) in ffmpeg to *conditionally* copy the stream? an example would be that in a batch process, i would only want to convert audio to aac if it is not already aac. are there any workarounds on this other than first ffmpeg -i , parse the output, try to detect what you have there and if it is not what you want convert, otherwise copy? this workaround is reasonable for some sources, but for others where seeking is problematic or a large portion of the file must be read for detection it raises many problems. -- Claudiu From cehoyos at ag.or.at Thu Nov 20 13:55:55 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 12:55:55 +0000 (UTC) Subject: [FFmpeg-user] Codec "same" as opposed to "copy"? References: <546D911C.3010403@burner.com> <546DE29E.40807@misalpina.net> Message-ID: Claudiu Rad misalpina.net> writes: > is there a possibility (or is anything foreseen) > in ffmpeg to *conditionally* copy the stream? Risking to repeat myself: As soon as somebody sends a clean patch, why not? Please don't forget that time is the only limiting factor for FFmpeg development. (And I believe we have more important open bugs.) Carl Eugen From licinio.alexandre at gmail.com Thu Nov 20 14:48:37 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 05:48:37 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts Message-ID: <1416491317633-4668216.post@n4.nabble.com> Hi all, I'm using ffmpeg for a lot of tricks (mainly for encoding) BUT i can't use professional decoders (Ateme, Ericsson, Tandberg, etc...) to decoding my stream (udp or rtp over mpeg2ts). Do you know why ? The stream is really good decoding by any avconv/ffmpeg player or vlc or raspberry or whatever BUT NOT by professional decoders. They "see" the stream (they probe it) but they can't decode it and show the service, pid program, video and audio. I did this : avconv -i udp://@:5006 -map 0 -codec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://192.168.1.54:5004?pkt_size=1316?ttl=8 Thanks, -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Thu Nov 20 15:05:15 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 14:05:15 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts References: <1416491317633-4668216.post@n4.nabble.com> Message-ID: AlexandreL gmail.com> writes: > I did this : > > avconv This looks like an intentionally broken version of FFmpeg that contains several hundred known bugs not present in FFmpeg, some of the security relevant. Please understand that we cannot support it here. Carl Eugen From anshul.ffmpeg at gmail.com Thu Nov 20 15:07:48 2014 From: anshul.ffmpeg at gmail.com (Anshul) Date: Thu, 20 Nov 2014 19:37:48 +0530 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416491317633-4668216.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> Message-ID: <546DF5B4.5000105@gmail.com> On 11/20/2014 07:18 PM, AlexandreL wrote: > Hi all, > > I'm using ffmpeg for a lot of tricks (mainly for encoding) BUT i can't use > professional decoders (Ateme, Ericsson, Tandberg, etc...) to decoding my > stream (udp or rtp over mpeg2ts). > > Do you know why ? > The stream is really good decoding by any avconv/ffmpeg player or vlc or > raspberry or whatever BUT NOT by professional decoders. > They "see" the stream (they probe it) but they can't decode it and show the > service, pid program, video and audio. > I did this : > > avconv -i udp://@:5006 -map 0 -codec copy -streamid 0:512 -streamid 1:4112 > -metadata:s:a:0 language=eng -metadata service_name=test -metadata > service_provider=test2 -mpegts_service_id 0x1 -f mpegts > udp://192.168.1.54:5004?pkt_size=1316?ttl=8 On ffmpeg mailing list people would like to know, what command you used by ffmpeg not avconv. Complete uncut ffmpeg console output is what any one want to provide any info. from avconv command it looks like you have some encoder installed on your PC, which is providing stream, check whether bare output of encoder of each stream is decodable by its refrence decoder, FFmpeg is very stable code, to give user maximum out of video. It supress some warning, may be debug log from FFmpeg would be helpful to you. -Anshul From licinio.alexandre at gmail.com Thu Nov 20 15:05:44 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 06:05:44 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: References: <1416491317633-4668216.post@n4.nabble.com> Message-ID: <1416492344772-4668219.post@n4.nabble.com> Sorry for avconv but also with ffmpeg it's not working ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test1 -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://192.168.1.54:5004?pkt_size=1316?ttl=8 Thanks, -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668219.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Thu Nov 20 15:17:50 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 14:17:50 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts References: <1416491317633-4668216.post@n4.nabble.com> <1416492344772-4668219.post@n4.nabble.com> Message-ID: AlexandreL gmail.com> writes: > Sorry for avconv but also with ffmpeg it's not working Complete, uncut console output missing. Carl Eugen From licinio.alexandre at gmail.com Thu Nov 20 15:11:42 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 06:11:42 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <546DF5B4.5000105@gmail.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> Message-ID: <1416492702833-4668220.post@n4.nabble.com> Sorry, i'm a new user i post my ffmpeg message during transcoding. *c-100-2 at c1002-ProLiant-DL360-G6*:~$ ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://192.168.1.10:5004?pkt_size=1316?ttl=8 ffmpeg version 0.8.16-4:0.8.16-0ubuntu0.12.04.1, Copyright (c) 2000-2014 the Libav developers built on Sep 16 2014 18:33:49 with gcc 4.6.3 The ffmpeg program is only provided for script compatibility and will be removed in a future release. It has been deprecated in the Libav project to allow for incompatible command line syntax improvements in its replacement called avconv (see Changelog for details). Please use avconv instead. [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! 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[h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! 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[h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! 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[h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [h264 @ 0x1cc6ac0] non-existing PPS referenced [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced [h264 @ 0x1cc6ac0] decode_slice_header error [h264 @ 0x1cc6ac0] no frame! [mpegts @ 0x1cb09c0] max_analyze_duration reached [mpegts @ 0x1cb09c0] Estimating duration from bitrate, this may be inaccurate Input #0, mpegts, from 'udp://@:5006': Duration: N/A, start: 14.840000, bitrate: 224 kb/s Program 1 Metadata: service_name : test-servicename service_provider: test-provider Stream #0.0[0x200]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 30.45 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x1010](eng): Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s [mpegts @ 0x1cc7200] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'udp://192.168.1.10:5004?pkt_size=1316?ttl=8': Metadata: service_name : test service_provider: test2 encoder : Lavf53.21.1 Stream #0.0: Video: libx264, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 90k tbn, 25 tbc Stream #0.1(eng): Audio: mp2, 48000 Hz, stereo, 224 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press ctrl-c to stop encoding ^Cframe= 431 fps= 33 q=-1.0 Lsize= 6833kB time=18.20 bitrate=3075.7kbits/s video:5779kB audio:499kB global headers:0kB muxing overhead 8.831937% Received signal 2: terminating -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668220.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From anshul.ffmpeg at gmail.com Thu Nov 20 15:22:30 2014 From: anshul.ffmpeg at gmail.com (Anshul) Date: Thu, 20 Nov 2014 19:52:30 +0530 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416492702833-4668220.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> Message-ID: <546DF926.5020608@gmail.com> On 11/20/2014 07:41 PM, AlexandreL wrote: > Sorry, i'm a new user > i post my ffmpeg message during transcoding. > > > *c-100-2 at c1002-ProLiant-DL360-G6*:~$ ffmpeg -i udp://@:5006 -vcodec copy > -acodec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng > -metadata service_name=test -metadata service_provider=test2 > -mpegts_service_id 0x1 -f mpegts > udp://192.168.1.10:5004?pkt_size=1316?ttl=8 > ffmpeg version 0.8.16-4:0.8.16-0ubuntu0.12.04.1, Copyright (c) 2000-2014 the Its Historical version :) > Libav developers > built on Sep 16 2014 18:33:49 with gcc 4.6.3 > The ffmpeg program is only provided for script compatibility and will be > removed > in a future release. It has been deprecated in the Libav project to allow > for > incompatible command line syntax improvements in its replacement called > avconv > (see Changelog for details). Please use avconv instead. > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [h264 @ 0x1cc6ac0] non-existing PPS referenced > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] non-existing PPS 0 referenced > [h264 @ 0x1cc6ac0] decode_slice_header error > [h264 @ 0x1cc6ac0] no frame! > [mpegts @ 0x1cb09c0] max_analyze_duration reached > [mpegts @ 0x1cb09c0] Estimating duration from bitrate, this may be > inaccurate > Input #0, mpegts, from 'udp://@:5006': > Duration: N/A, start: 14.840000, bitrate: 224 kb/s > Program 1 > Metadata: > service_name : test-servicename > service_provider: test-provider > Stream #0.0[0x200]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR > 16:9], 30.45 fps, 25 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1010](eng): Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s > [mpegts @ 0x1cc7200] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt > every 40 pkts > Output #0, mpegts, to 'udp://192.168.1.10:5004?pkt_size=1316?ttl=8': > Metadata: > service_name : test > service_provider: test2 > encoder : Lavf53.21.1 > Stream #0.0: Video: libx264, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], > q=2-31, 90k tbn, 25 tbc > Stream #0.1(eng): Audio: mp2, 48000 Hz, stereo, 224 kb/s > Stream mapping: > Stream #0.0 -> #0.0 > Stream #0.1 -> #0.1 > Press ctrl-c to stop encoding > ^Cframe= 431 fps= 33 q=-1.0 Lsize= 6833kB time=18.20 > bitrate=3075.7kbits/s > video:5779kB audio:499kB global headers:0kB muxing overhead 8.831937% > Received signal 2: terminating > > > > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668220.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From licinio.alexandre at gmail.com Thu Nov 20 15:18:56 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 06:18:56 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <546DF926.5020608@gmail.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> Message-ID: <1416493136227-4668223.post@n4.nabble.com> Sorry all, Where can i find the "Complete, uncut console output" ? Thanks, -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668223.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From barsnick at gmx.net Thu Nov 20 15:26:15 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Thu, 20 Nov 2014 15:26:15 +0100 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416492344772-4668219.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <1416492344772-4668219.post@n4.nabble.com> Message-ID: <20141120142615.GB27506@sunshine.barsnick.net> On Thu, Nov 20, 2014 at 06:05:44 -0800, AlexandreL wrote: > ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 -streamid > 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test1 -metadata > service_provider=test2 -mpegts_service_id 0x1 -f mpegts > udp://192.168.1.54:5004?pkt_size=1316?ttl=8 As Anshul said, please also provide the full, uncut output of the command in addition to the commandline. Moritz From cehoyos at ag.or.at Thu Nov 20 15:26:13 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 14:26:13 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> Message-ID: AlexandreL gmail.com> writes: > ffmpeg version 0.8.16-4:0.8.16-0ubuntu0.12.04.1 This looks like an intentionally broken version of FFmpeg that contains several hundred known bugs not present in FFmpeg, some of them security relevant, please understand that we cannot support it here. Please see http://ffmpeg.org/download.html for supported versions, please test *current git head* before reporting problems here. See http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html for more information about the fraud that hit you. Carl Eugen From cehoyos at ag.or.at Thu Nov 20 15:28:19 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 14:28:19 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> Message-ID: Anshul gmail.com> writes: > > ffmpeg version 0.8.16-4:0.8.16-0ubuntu0.12.04.1 > > Its Historical version :) No, this is *not* an old FFmpeg version, such a version has never existed. Carl Eugen From cehoyos at ag.or.at Thu Nov 20 15:47:16 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 14:47:16 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg play continuous loop! References: Message-ID: Suri Shelvapille baymicrosystems.com> writes: > I need to play all the videos in a loop. Then please try with the movie filter as input, there is no loop option for files passed via "-i". Carl Eugen From anshul.ffmpeg at gmail.com Thu Nov 20 15:47:50 2014 From: anshul.ffmpeg at gmail.com (Anshul) Date: Thu, 20 Nov 2014 20:17:50 +0530 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416493136227-4668223.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416493136227-4668223.post@n4.nabble.com> Message-ID: <546DFF16.7090105@gmail.com> On 11/20/2014 07:48 PM, AlexandreL wrote: > Sorry all, > Where can i find the "Complete, uncut console output" ? > > Thanks, > Though I am not sure, from long time I was not here, Most of the times things printed on console is known as Console Output of ffmpeg. Use git version and your console output would look like as console output. you can download snapshot from here https://www.ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 As u are vey new user here, I would suggest scroll through following links. https://www.ffmpeg.org/download.html https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu. If your problem still persist, then come again. -Anshul From licinio.alexandre at gmail.com Thu Nov 20 15:43:12 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 06:43:12 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> Message-ID: <1416494592507-4668229.post@n4.nabble.com> so sorry i am i just updated to the last version it's not working again here the command c-100-2 at c1002-ProLiant-DL360-G6:~/Downloads/ffmpeg-git-20141120-64bit-static$ ./ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://193.252.151.106:5004?pkt_size=1316?ttl=128 ffmpeg version N-42326-gc661601- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2014 the FFmpeg developers built on Nov 20 2014 01:23:11 with gcc 4.8 (Debian 4.8.3-13) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable-libass --enable-gnutls --cc=gcc-4.8 libavutil 54. 14.100 / 54. 14.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 14.100 / 56. 14.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! [h264 @ 0x2d10c00] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] non-existing PPS 0 referenced [h264 @ 0x2d10c00] decode_slice_header error [h264 @ 0x2d10c00] no frame! Input #0, mpegts, from 'udp://@:5006': Duration: N/A, start: 1934.888000, bitrate: 224 kb/s Program 1 Metadata: service_name : test-servicename service_provider: test-provider Stream #0:0[0x200]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x1010](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 224 kb/s Output #0, mpegts, to 'udp://193.252.151.106:5004?pkt_size=1316?ttl=128': Metadata: service_name : test service_provider: test2 encoder : Lavf56.14.100 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 90k tbn, 25 tbc Stream #0:1(eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, 224 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 130 fps=0.0 q=-1.0 size= 2646kB time=00:00:05.61 bitrate=3859.4kbits/frame= 143 fps=142 q=-1.0 size= 2907kB time=00:00:06.09 bitrate=3906.6kbits/frame= 156 fps=103 q=-1.0 size= 3173kB time=00:00:06.60 bitrate=3938.9kbits/frame= 168 fps= 84 q=-1.0 size= 3414kB time=00:00:07.10 bitrate=3936.8kbits/frame= 181 fps= 72 q=-1.0 size= 3676kB time=00:00:07.60 bitrate=3957.9kbits/frame= 193 fps= 64 q=-1.0 size= 3919kB time=00:00:08.13 bitrate=3945.8kbits/frame= 206 fps= 58 q=-1.0 size= 4180kB time=00:00:08.61 bitrate=3974.5kbits/frame= 219 fps= 54 q=-1.0 size= 4443kB time=00:00:09.12 bitrate=3990.7kbits/frame= 231 fps= 51 q=-1.0 size= 4684kB time=00:00:09.62 bitrate=3987.4kbits/frame= 244 fps= 48 q=-1.0 size= 4937kB time=00:00:10.12 bitrate=3993.0kbits/frame= 256 fps= 46 q=-1.0 size= 5182kB time=00:00:10.65 bitrate=3983.7kbits/frame= 269 fps= 44 q=-1.0 size= 5447kB time=00:00:11.13 bitrate=4006.9kbits/frame= 282 fps= 43 q=-1.0 size= 5711kB time=00:00:11.64 bitrate=4019.5kbits/frame= 294 fps= 42 q=-1.0 size= 5952kB time=00:00:12.14 bitrate=4015.4kbits/frame= 307 fps= 41 q=-1.0 size= 6218kB time=00:00:12.64 bitrate=4027.1kbits/frame= 319 fps= 40 q=-1.0 size= 6459kB time=00:00:13.17 bitrate=4015.6kbits/frame= 332 fps= 39 q=-1.0 size= 6721kB time=00:00:13.65 bitrate=4031.8kbits/frame= 345 fps= 38 q=-1.0 size= 6980kB time=00:00:14.16 bitrate=4037.9kbits/frame= 357 fps= 37 q=-1.0 size= 7221kB time=00:00:14.66 bitrate=4034.2kbits/frame= 370 fps= 37 q=-1.0 size= 7485kB time=00:00:15.16 bitrate=4042.6kbits/frame= 382 fps= 36 q=-1.0 size= 7727kB time=00:00:15.69 bitrate=4032.6kbits/frame= 395 fps= 36 q=-1.0 size= 7988kB time=00:00:16.17 bitrate=4045.3kbits/frame= 408 fps= 35 q=-1.0 size= 8240kB time=00:00:16.68 bitrate=4047.0kbits/frame= 420 fps= 35 q=-1.0 size= 8478kB time=00:00:17.18 bitrate=4041.8kbits/frame= 433 fps= 34 q=-1.0 size= 8741kB time=00:00:17.68 bitrate=4048.4kbits/frame= 445 fps= 34 q=-1.0 size= 8983kB time=00:00:18.21 bitrate=4039.8kbits/frame= 458 fps= 34 q=-1.0 size= 9250kB time=00:00:18.69 bitrate=4052.9kbits/frame= 471 fps= 33 q=-1.0 size= 9512kB time=00:00:19.20 bitrate=4058.4kbits/frame= 483 fps= 33 q=-1.0 size= 9755kB time=00:00:19.70 bitrate=4055.9kbits/frame= 496 fps= 33 q=-1.0 size= 10011kB time=00:00:20.20 bitrate=4058.4kbits/frame= 508 fps= 33 q=-1.0 size= 10251kB time=00:00:20.73 bitrate=4049.8kbits/frame= 521 fps= 32 q=-1.0 size= 10512kB time=00:00:21.21 bitrate=4058.8kbits/frame= 526 fps= 32 q=-1.0 Lsize= 10616kB time=00:00:21.43 bitrate=4057.6kbits/s video:9187kB audio:586kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.621732% Received signal 2: terminating. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668229.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From anshul.ffmpeg at gmail.com Thu Nov 20 15:51:08 2014 From: anshul.ffmpeg at gmail.com (Anshul) Date: Thu, 20 Nov 2014 20:21:08 +0530 Subject: [FFmpeg-user] ffmpeg play continuous loop! In-Reply-To: References: Message-ID: <546DFFDC.7000108@gmail.com> On 11/20/2014 08:17 PM, Carl Eugen Hoyos wrote: > Suri Shelvapille baymicrosystems.com> writes: > >> I need to play all the videos in a loop. > Then please try with the movie filter as input, > there is no loop option for files passed via "-i". > > Carl Eugen > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user For playing on display you can also use ffplay -loop 0 blah blah blah From anshul.ffmpeg at gmail.com Thu Nov 20 15:59:04 2014 From: anshul.ffmpeg at gmail.com (Anshul) Date: Thu, 20 Nov 2014 20:29:04 +0530 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416494592507-4668229.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> Message-ID: <546E01B8.8080000@gmail.com> On 11/20/2014 08:13 PM, AlexandreL wrote: > so sorry i am > i just updated to the last version > it's not working again > > here the command > > c-100-2 at c1002-ProLiant-DL360-G6:~/Downloads/ffmpeg-git-20141120-64bit-static$ > ./ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 > -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test > -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts > udp://193.252.151.106:5004?pkt_size=1316?ttl=128 > ffmpeg version N-42326-gc661601- http://johnvansickle.com/ffmpeg/ > Copyright (c) 2000-2014 the FFmpeg developers > built on Nov 20 2014 01:23:11 with gcc 4.8 (Debian 4.8.3-13) > configuration: --enable-gpl --enable-version3 --disable-shared > --disable-debug --enable-runtime-cpudetect --enable-libmp3lame > --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex > --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig > --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver > --enable-libass --enable-gnutls --cc=gcc-4.8 > libavutil 54. 14.100 / 54. 14.100 > libavcodec 56. 12.101 / 56. 12.101 > libavformat 56. 14.100 / 56. 14.100 > libavdevice 56. 3.100 / 56. 3.100 > libavfilter 5. 2.103 / 5. 2.103 > libswscale 3. 1.101 / 3. 1.101 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 3.100 / 53. 3.100 > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] non-existing PPS 0 referenced > [h264 @ 0x2d10c00] decode_slice_header error > [h264 @ 0x2d10c00] no frame! > Input #0, mpegts, from 'udp://@:5006': > Duration: N/A, start: 1934.888000, bitrate: 224 kb/s > Program 1 > Metadata: > service_name : test-servicename > service_provider: test-provider > Stream #0:0[0x200]: Video: h264 (High) ([27][0][0][0] / 0x001B), > yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, > 50 tbc > Stream #0:1[0x1010](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, > stereo, s16p, 224 kb/s > Output #0, mpegts, to 'udp://193.252.151.106:5004?pkt_size=1316?ttl=128': > Metadata: > service_name : test > service_provider: test2 > encoder : Lavf56.14.100 > Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 > [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 90k tbn, 25 tbc > Stream #0:1(eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, > 224 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Stream #0:1 -> #0:1 (copy) > Press [q] to stop, [?] for help > frame= 130 fps=0.0 q=-1.0 size= 2646kB time=00:00:05.61 > bitrate=3859.4kbits/frame= 143 fps=142 q=-1.0 size= 2907kB > time=00:00:06.09 bitrate=3906.6kbits/frame= 156 fps=103 q=-1.0 size= > 3173kB time=00:00:06.60 bitrate=3938.9kbits/frame= 168 fps= 84 q=-1.0 size= > 3414kB time=00:00:07.10 bitrate=3936.8kbits/frame= 181 fps= 72 q=-1.0 size= > 3676kB time=00:00:07.60 bitrate=3957.9kbits/frame= 193 fps= 64 q=-1.0 size= > 3919kB time=00:00:08.13 bitrate=3945.8kbits/frame= 206 fps= 58 q=-1.0 size= > 4180kB time=00:00:08.61 bitrate=3974.5kbits/frame= 219 fps= 54 q=-1.0 size= > 4443kB time=00:00:09.12 bitrate=3990.7kbits/frame= 231 fps= 51 q=-1.0 size= > 4684kB time=00:00:09.62 bitrate=3987.4kbits/frame= 244 fps= 48 q=-1.0 size= > 4937kB time=00:00:10.12 bitrate=3993.0kbits/frame= 256 fps= 46 q=-1.0 size= > 5182kB time=00:00:10.65 bitrate=3983.7kbits/frame= 269 fps= 44 q=-1.0 size= > 5447kB time=00:00:11.13 bitrate=4006.9kbits/frame= 282 fps= 43 q=-1.0 size= > 5711kB time=00:00:11.64 bitrate=4019.5kbits/frame= 294 fps= 42 q=-1.0 size= > 5952kB time=00:00:12.14 bitrate=4015.4kbits/frame= 307 fps= 41 q=-1.0 size= > 6218kB time=00:00:12.64 bitrate=4027.1kbits/frame= 319 fps= 40 q=-1.0 size= > 6459kB time=00:00:13.17 bitrate=4015.6kbits/frame= 332 fps= 39 q=-1.0 size= > 6721kB time=00:00:13.65 bitrate=4031.8kbits/frame= 345 fps= 38 q=-1.0 size= > 6980kB time=00:00:14.16 bitrate=4037.9kbits/frame= 357 fps= 37 q=-1.0 size= > 7221kB time=00:00:14.66 bitrate=4034.2kbits/frame= 370 fps= 37 q=-1.0 size= > 7485kB time=00:00:15.16 bitrate=4042.6kbits/frame= 382 fps= 36 q=-1.0 size= > 7727kB time=00:00:15.69 bitrate=4032.6kbits/frame= 395 fps= 36 q=-1.0 size= > 7988kB time=00:00:16.17 bitrate=4045.3kbits/frame= 408 fps= 35 q=-1.0 size= > 8240kB time=00:00:16.68 bitrate=4047.0kbits/frame= 420 fps= 35 q=-1.0 size= > 8478kB time=00:00:17.18 bitrate=4041.8kbits/frame= 433 fps= 34 q=-1.0 size= > 8741kB time=00:00:17.68 bitrate=4048.4kbits/frame= 445 fps= 34 q=-1.0 size= > 8983kB time=00:00:18.21 bitrate=4039.8kbits/frame= 458 fps= 34 q=-1.0 size= > 9250kB time=00:00:18.69 bitrate=4052.9kbits/frame= 471 fps= 33 q=-1.0 size= > 9512kB time=00:00:19.20 bitrate=4058.4kbits/frame= 483 fps= 33 q=-1.0 size= > 9755kB time=00:00:19.70 bitrate=4055.9kbits/frame= 496 fps= 33 q=-1.0 size= > 10011kB time=00:00:20.20 bitrate=4058.4kbits/frame= 508 fps= 33 q=-1.0 > size= 10251kB time=00:00:20.73 bitrate=4049.8kbits/frame= 521 fps= 32 > q=-1.0 size= 10512kB time=00:00:21.21 bitrate=4058.8kbits/frame= 526 fps= > 32 q=-1.0 Lsize= 10616kB time=00:00:21.43 bitrate=4057.6kbits/s > video:9187kB audio:586kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 8.621732% > Received signal 2: terminating. > > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668229.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user I dont see any problem in ffmpeg, you r not using ffmpeg codecs to encode either video or audio. so if the generated video does not work on any professional decoder then there is problem in Encoder of Input provider. Thanks Anshul From licinio.alexandre at gmail.com Thu Nov 20 16:03:51 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 07:03:51 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <546E01B8.8080000@gmail.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> <546E01B8.8080000@gmail.com> Message-ID: <1416495831758-4668232.post@n4.nabble.com> i'm agree but the encoder used is a professional one (same brand as the decoder), Ateme. When i stream directly from the Ateme encoder to the Ateme decoder, it works. When i stream from the Ateme encoder through ffmpeg and ffmpeg streaming to the decoder, it's not working. What i don't understand is why it's not working ? because professional equipment are only working with professional equipment ? i'm sure we can do the trick, i'm sure of that maybe it's just something about the PCR pid or program pid or whatever i'm also compiling ffmpeg with the tutorial https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu and i will try in few minutes -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668232.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From licinio.alexandre at gmail.com Thu Nov 20 16:15:23 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 07:15:23 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416495831758-4668232.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> <546E01B8.8080000@gmail.com> <1416495831758-4668232.post@n4.nabble.com> Message-ID: <1416496523657-4668233.post@n4.nabble.com> no it's the same with the compiled version c-100-2 at c1002-ProLiant-DL360-G6:~/bin$ cd ~/bin && ./ffmpeg -i udp://@:5006 -vcodec copy -acodec copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://193.252.151.106:5004?pkt_size=1316?ttl=128 ffmpeg version 2.4.git Copyright (c) 2000-2014 the FFmpeg developers built on Nov 20 2014 16:12:29 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --prefix=/home/c-100-2/ffmpeg_build --extra-cflags=-I/home/c-100-2/ffmpeg_build/include --extra-ldflags=-L/home/c-100-2/ffmpeg_build/lib --bindir=/home/c-100-2/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 54. 14.100 / 54. 14.100 libavcodec 56. 12.101 / 56. 12.101 libavformat 56. 14.100 / 56. 14.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! [h264 @ 0x312efe0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] non-existing PPS 0 referenced [h264 @ 0x312efe0] decode_slice_header error [h264 @ 0x312efe0] no frame! Input #0, mpegts, from 'udp://@:5006': Duration: N/A, start: 3996.440000, bitrate: 224 kb/s Program 1 Metadata: service_name : test-servicename service_provider: test-provider Stream #0:0[0x200]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x1010](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 224 kb/s Output #0, mpegts, to 'udp://193.252.151.106:5004?pkt_size=1316?ttl=128': Metadata: service_name : test service_provider: test2 encoder : Lavf56.14.100 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 90k tbn, 25 tbc Stream #0:1(eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, 224 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 178 fps= 59 q=-1.0 Lsize= 3609kB time=00:00:08.08 bitrate=3655.5kbits/s video:3099kB audio:221kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.698487% Received signal 2: terminating. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668233.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From george at nsup.org Thu Nov 20 19:18:23 2014 From: george at nsup.org (Nicolas George) Date: Thu, 20 Nov 2014 19:18:23 +0100 Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> Message-ID: <20141120181823.GA11268@phare.normalesup.org> Le decadi 30 brumaire, an CCXXIII, Carl Eugen Hoyos a ?crit?: > This looks like an intentionally broken version of > FFmpeg that contains several hundred known bugs not > present in FFmpeg, some of them security relevant, > please understand that we cannot support it here. > Please see http://ffmpeg.org/download.html for > supported versions, please test *current git head* > before reporting problems here. > > See http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html > for more information about the fraud that hit you. ^^^^^ Please avoid derogatory terms for old issues, dwelling on the past will not bring anything. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 819 bytes Desc: Digital signature URL: From cehoyos at ag.or.at Thu Nov 20 19:21:38 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 18:21:38 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> Message-ID: AlexandreL gmail.com> writes: > i just updated to the last version > it's not working again Does it work if you don't use -codec copy but if you encode to mpeg2video or libx264? Did you already try audio-only or video-only? Carl Eugen From ganeshprasad2 at gmail.com Thu Nov 20 20:04:23 2014 From: ganeshprasad2 at gmail.com (ganesh Prasad) Date: Thu, 20 Nov 2014 14:04:23 -0500 Subject: [FFmpeg-user] Extraction of frames in H.264 encoded data Message-ID: Hello , I am receiving H264 encoded video over a tcp socket , the decoder that I have works only on a complete frame . can I use ffmpeg to extract a complete frame from the received H264 data ? From licinio.alexandre at gmail.com Thu Nov 20 20:22:02 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 11:22:02 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> Message-ID: <1416511322744-4668237.post@n4.nabble.com> i tried with video only and audio only > no success i tried this command c-100-2 at c1002-ProLiant-DL360-G6:~/bin$ cd ~/bin && ./ffmpeg -i udp://@:5006 -c:v libx264 -profile:v high -preset ultrafast -threads 12 -b:v 4000k -maxrate 4000k -bufsize 8000k -c:a copy -streamid 0:512 -streamid 1:4112 -metadata:s:a:0 language=eng -metadata service_name=test -metadata service_provider=test2 -mpegts_service_id 0x1 -f mpegts udp://193.252.151.106:5004?pkt_size=1316?ttl=128 great in vlc but not working with the professional decoder WHY WHY WHY WHY ??? most of the professional devices are working through a ffmpeg core ! i will try in mpeg2 -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668237.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From licinio.alexandre at gmail.com Thu Nov 20 20:58:50 2014 From: licinio.alexandre at gmail.com (AlexandreL) Date: Thu, 20 Nov 2014 11:58:50 -0800 (PST) Subject: [FFmpeg-user] ffmpeg profesional decoder compatibility mpegts In-Reply-To: <1416511322744-4668237.post@n4.nabble.com> References: <1416491317633-4668216.post@n4.nabble.com> <546DF5B4.5000105@gmail.com> <1416492702833-4668220.post@n4.nabble.com> <546DF926.5020608@gmail.com> <1416494592507-4668229.post@n4.nabble.com> <1416511322744-4668237.post@n4.nabble.com> Message-ID: <1416513530250-4668238.post@n4.nabble.com> no success too with mpeg2 i'm sure it's possible to do it i need more coffee !! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-profesional-decoder-compatibility-mpegts-tp4668216p4668238.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From cehoyos at ag.or.at Thu Nov 20 21:13:46 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 20 Nov 2014 20:13:46 +0000 (UTC) Subject: [FFmpeg-user] Extraction of frames in H.264 encoded data References: Message-ID: ganesh Prasad gmail.com> writes: > I am receiving H264 encoded video over a tcp socket, > the decoder that I have works only on a complete frame. > can I use ffmpeg to extract a complete frame from the > received H264 data ? This is the default, you can only extract complete frames with FFmpeg. Carl Eugen From jonathan.viney at gmail.com Thu Nov 20 21:20:09 2014 From: jonathan.viney at gmail.com (Jonathan Viney) Date: Fri, 21 Nov 2014 09:20:09 +1300 Subject: [FFmpeg-user] Problems saving rtsp stream directly as mpegts In-Reply-To: References: Message-ID: I had another look at this today and managed to find a solution. Adding "-bsf:v dump_extra" makes the mpeg2ts output file readable by ffmpeg. On Mon, Nov 3, 2014 at 3:38 PM, Jonathan Viney wrote: > > On Mon, Nov 3, 2014 at 2:45 PM, Carl Eugen Hoyos wrote: >> >> Jonathan Viney gmail.com> writes: >> >> > ffmpeg -i rtsp://10.9.9.3:554/axis-media/media.amp -c copy -f mpegts >> > out.ts >> > >> > This seems to work fine, but the output file is >> > not recognised by ffmpeg/ffprobe. >> >> Complete, uncut console output missing. >> (I would have expected above command to return an >> error message and I would like to know why there >> is no such message.) > > > Here are the command outputs. > > ffmpeg -rtsp_transport tcp -i rtsp://10.9.9.3:554/axis-media/media.amp -c > copy -y -t 5 out.ts > > ffmpeg version N-67343-gd457478 Copyright (c) 2000-2014 the FFmpeg > developers > built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) > (based on LLVM 3.5svn) > configuration: --enable-shared --enable-pthreads --enable-gpl > --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx > --prefix=/tmp/ffmpeg-test > libavutil 54. 11.100 / 54. 11.100 > libavcodec 56. 10.101 / 56. 10.101 > libavformat 56. 12.100 / 56. 12.100 > libavdevice 56. 2.100 / 56. 2.100 > libavfilter 5. 2.101 / 5. 2.101 > libswscale 3. 1.101 / 3. 1.101 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 3.100 / 53. 3.100 > Input #0, rtsp, from 'rtsp://10.9.9.3:554/axis-media/media.amp': > Metadata: > title : Media Presentation > Duration: N/A, start: 0.040011, bitrate: N/A > Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 > DAR 16:9], 25 fps, 25 tbr, 90k tbn, 180k tbc > Output #0, mpegts, to 'out.ts': > Metadata: > title : Media Presentation > encoder : Lavf56.12.100 > Stream #0:0: Video: h264, yuvj420p, 1920x1080 [SAR 1:1 DAR 16:9], > q=2-31, 25 fps, 90k tbn, 90k tbc > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Press [q] to stop, [?] for help > [mpegts @ 0x7f9f22826a00] Non-monotonous DTS in output stream 0:0; previous: > 0, current: 0; changing to 1. This may result in incorrect timestamps in the > output file. > frame= 127 fps= 32 q=-1.0 Lsize= 5774kB time=00:00:04.99 > bitrate=9460.9kbits/s > video:5341kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 8.106906% > > > ffprobe out.ts > > ffprobe out.ts > ffprobe version N-67343-gd457478 Copyright (c) 2007-2014 the FFmpeg > developers > built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) > (based on LLVM 3.5svn) > configuration: --enable-shared --enable-pthreads --enable-gpl > --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx > --prefix=/tmp/ffmpeg-test > libavutil 54. 11.100 / 54. 11.100 > libavcodec 56. 10.101 / 56. 10.101 > libavformat 56. 12.100 / 56. 12.100 > libavdevice 56. 2.100 / 56. 2.100 > libavfilter 5. 2.101 / 5. 2.101 > libswscale 3. 1.101 / 3. 1.101 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 3.100 / 53. 3.100 > [h264 @ 0x7fe783808000] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x7fe783808000] decode_slice_header error > [h264 @ 0x7fe783808000] no frame! > [h264 @ 0x7fe783808000] non-existing PPS 0 referenced > Last message repeated 1 times > [h264 @ 0x7fe783808000] decode_slice_header error > [h264 @ 0x7fe783808000] no frame! > [h264 @ 0x7fe783808000] non-existing PPS 0 referenced > (duplicate messages removed) > [mpegts @ 0x7fe78300da00] decoding for stream 0 failed > [mpegts @ 0x7fe78300da00] Could not find codec parameters for stream 0 > (Video: h264 ([27][0][0][0] / 0x001B), none): unspecified size > Consider increasing the value for the 'analyzeduration' and 'probesize' > options > out.ts: Operation not permitted > >> >> >> > If I do it in two steps, first saving the stream >> > as a .mp4, it converts to mpegts fine with: >> > >> > ffmpeg -i out.mp4 -c copy -bsf h264_mp4toannexb out.ts >> > >> > Is it possible to do this in one step without the >> > intermediary file? >> >> What happens if you add the bitstreamfilter to the >> first command? > > > ffmpeg -rtsp_transport tcp -i rtsp://10.9.9.3:554/axis-media/media.amp -c > copy -y -bsf h264_mp4toannexb -t 5 out.ts > > ffmpeg version N-67343-gd457478 Copyright (c) 2000-2014 the FFmpeg > developers > built on Nov 3 2014 15:19:39 with Apple LLVM version 6.0 (clang-600.0.54) > (based on LLVM 3.5svn) > configuration: --enable-shared --enable-pthreads --enable-gpl > --enable-version3 --enable-nonfree --enable-libx264 --enable-libvpx > --prefix=/tmp/ffmpeg-test > libavutil 54. 11.100 / 54. 11.100 > libavcodec 56. 10.101 / 56. 10.101 > libavformat 56. 12.100 / 56. 12.100 > libavdevice 56. 2.100 / 56. 2.100 > libavfilter 5. 2.101 / 5. 2.101 > libswscale 3. 1.101 / 3. 1.101 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 3.100 / 53. 3.100 > Input #0, rtsp, from 'rtsp://10.9.9.3:554/axis-media/media.amp': > Metadata: > title : Media Presentation > Duration: N/A, start: 0.040000, bitrate: N/A > Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 > DAR 16:9], 25 fps, 25 tbr, 90k tbn, 180k tbc > Output #0, mpegts, to 'out.ts': > Metadata: > title : Media Presentation > encoder : Lavf56.12.100 > Stream #0:0: Video: h264, yuvj420p, 1920x1080 [SAR 1:1 DAR 16:9], > q=2-31, 25 fps, 90k tbn, 90k tbc > Stream mapping: > Stream #0:0 -> #0:0 (copy) > Press [q] to stop, [?] for help > [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, > corrupted stream or invalid MP4/AVCC bitstream > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec > copy: Invalid argument > [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, > corrupted stream or invalid MP4/AVCC bitstream > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec > copy: Invalid argument > [mpegts @ 0x7fa97a809600] Non-monotonous DTS in output stream 0:0; previous: > 0, current: 0; changing to 1. This may result in incorrect timestamps in the > output file. > [NULL @ 0x7fa97a820600] Packet header is not contained in global extradata, > corrupted stream or invalid MP4/AVCC bitstream > Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec > copy: Invalid argument > (duplicate messages removed) > frame= 127 fps= 31 q=-1.0 Lsize= 5280kB time=00:00:04.99 > bitrate=8651.6kbits/s > video:4883kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 8.132199% > > The resulting output file gives the same errors as above when passed to > ffprobe. > > Thanks for the help. > > Regards, > -Jonathan. > > >> Carl Eugen >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From dvtuan1610 at gmail.com Fri Nov 21 01:38:45 2014 From: dvtuan1610 at gmail.com (Tuan Dinh) Date: Fri, 21 Nov 2014 11:38:45 +1100 Subject: [FFmpeg-user] ERROR: ffmpeg[F30B4700]: [ac3] unsupported frame type : skipping frame Message-ID: Hi, Please help with the following issue: I have audio breaks when playing LiveTV stream that seems to use AC3-Dolby digital audio format. My ffmpeg library is the 2.4.3. The application is XBMC (under OpenElec 3.2). Behaviour: There are constant breaks in audio when playing the stream. The synchronization between audio/video is fine. When checking the log of the application, I got this: 09:09:35 T:139832223315712 INFO: CDVDAudioCodecFFmpeg::GetChannelMap - FFmpeg reported 3 channels, but the layout contains 0 ignoring 09:09:35 T:139833305450240 DEBUG: FactoryCodec - Audio: FFmpeg - Opening 09:09:35 T:139833305450240 DEBUG: FactoryCodec - Audio: FFmpeg - Opened 09:26:08 T:139832223315712 DEBUG: ffmpeg[39FFB700]: [ac3] Additional substreams not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. 09:26:08 T:139832223315712 DEBUG: ffmpeg[39FFB700]: [ac3] 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. [b]09:26:08 T:139832223315712 ERROR: ffmpeg[39FFB700]: [ac3] unsupported frame type : skipping frame[/b] 09:35:25 T:139833305450240 DEBUG: FactoryCodec - Audio: FFmpeg - Opening 09:35:25 T:139833305450240 DEBUG: FactoryCodec - Audio: FFmpeg - Opened 09:35:25 T:139832223315712 ERROR: ffmpeg[39FFB700]: [ac3] frame CRC mismatch Already check: ffmpeg library 2.4.3 is the latest up to 21/11/14. Best regards, -- Dinh, Tuan From cehoyos at ag.or.at Fri Nov 21 09:47:40 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 21 Nov 2014 08:47:40 +0000 (UTC) Subject: [FFmpeg-user] ERROR: ffmpeg[F30B4700]: [ac3] unsupported frame type : skipping frame References: Message-ID: Tuan Dinh gmail.com> writes: > 09:26:08 T:139832223315712 DEBUG: ffmpeg[39FFB700]: > [ac3] If you want to help, upload a sample of this > file to ftp://upload.ffmpeg.org/MPlayer/incoming/ Did you do this? As an alternative, please provide a download link here on the mailing list. > Already check: ffmpeg library 2.4.3 is the latest up to 21/11/14. No, the latest is f0ae0354, if you think our download page does not explain this well, please tell us! Carl Eugen From xanadu at apost.plala.or.jp Fri Nov 21 14:28:03 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Fri, 21 Nov 2014 22:28:03 +0900 Subject: [FFmpeg-user] Question about ffplay build on Mac OS X Message-ID: Hello list members, I know that SDL can be build without X11 on Mac OS X. Without X11, ffplay works fine on Mac OS X. My question is that ffplay should be build with X11 or should be build without X11 on Mac OS X. Is there any recommendation? Thanks for help!! // Miya From suri at baymicrosystems.com Fri Nov 21 21:05:15 2014 From: suri at baymicrosystems.com (Suri Shelvapille) Date: Fri, 21 Nov 2014 20:05:15 +0000 Subject: [FFmpeg-user] ffmpeg play continuous loop! In-Reply-To: References: Message-ID: Carl: As per your suggestion tried the below, unfortunately did not work....any other suggestions? Should the base video be a movie filter instead of nullsrc? -------------------------------------------- ffmpeg -i a1.avi -i a2.avi -filter_complex "nullsrc=size=320x90 [base];movie=a1.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos0];movie=a2.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos1];[base][pos0] overlay=shortest=0 [tmp1];[tmp1][pos1] overlay=shortest=0:x=160" -c:v libx264 -f avi - | ffplay - -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: Thursday, November 20, 2014 9:47 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] ffmpeg play continuous loop! Suri Shelvapille baymicrosystems.com> writes: > I need to play all the videos in a loop. Then please try with the movie filter as input, there is no loop option for files passed via "-i". Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From suri at baymicrosystems.com Fri Nov 21 22:38:09 2014 From: suri at baymicrosystems.com (Suri Shelvapille) Date: Fri, 21 Nov 2014 21:38:09 +0000 Subject: [FFmpeg-user] ffmpeg play continuous loop! References: Message-ID: Carl: I tried this as well, did not help: ----------------------------------------------------------------- ffmpeg -filter_complex "nullsrc=size=320x90 [base];movie=a1.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos0];movie=a2.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos1];[base][pos0] overlay=shortest=0 [tmp1];[tmp1][pos1] overlay=shortest=0:x=160" -c:v libx264 -f avi - | ffplay - ------------------------------------------- Thanks, Suri -----Original Message----- From: Suri Shelvapille Sent: Friday, November 21, 2014 3:05 PM To: ffmpeg-user at ffmpeg.org; 'Carl Eugen Hoyos' Subject: RE: [FFmpeg-user] ffmpeg play continuous loop! Carl: As per your suggestion tried the below, unfortunately did not work....any other suggestions? Should the base video be a movie filter instead of nullsrc? -------------------------------------------- ffmpeg -i a1.avi -i a2.avi -filter_complex "nullsrc=size=320x90 [base];movie=a1.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos0];movie=a2.avi:loop=100, setpts=PTS-STARTPTS, scale=160X90 [pos1];[base][pos0] overlay=shortest=0 [tmp1];[tmp1][pos1] overlay=shortest=0:x=160" -c:v libx264 -f avi - | ffplay - -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: Thursday, November 20, 2014 9:47 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] ffmpeg play continuous loop! Suri Shelvapille baymicrosystems.com> writes: > I need to play all the videos in a loop. Then please try with the movie filter as input, there is no loop option for files passed via "-i". Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From murat.maman at gmail.com Sat Nov 22 12:24:55 2014 From: murat.maman at gmail.com (Murat Maman) Date: Sat, 22 Nov 2014 13:24:55 +0200 Subject: [FFmpeg-user] Audio and Video Out of Sync with UPD stream input and flv RTMP output Message-ID: Hello everyone, I have troubles with A/V sync for a live RTMP output stream generated by ffmpeg. After playing around with the parameters for a few days now, I still wasn?t able to get rid of the sync issues. I hope that you could spot an issue in my command or advise me in how to avoid sync troubles. I would extremely appreciate any help here. Here?s the my problem: In order to publish a live RTMP stream, I use one UDP input, this is coming from an application that is only able to output raw formats. Ffmpeg is used for encoding resizing and encoding video to H.264. Finally, streams are muxed into flv and published via RTMP. In general, this works very well, and the results look great. However, unfortunately A/V seems to drift after running for a couple of days or 5-6 hours. Here?s the my command: sudo ffmpeg -y -f rawvideo mpegts \ -fflags nobuffer \ -i 'udp://@UPD LINK?fifo_size=1000000&overrun_nonfatal=1' \ -maxrate 4000k -bufsize 4000k \ -af volume=1.0 \ -strict experimental \ -c:a libfaac -ar 44100 -ac 2 -b:a 128k \ -vsync 1 -c:v libx264 -r 25 -vf yadif,scale=720:576 -aspect:v 16:9 -b:v 4000k -pix_fmt yuv420p \ -preset superfast -tune zerolatency -threads 0 -crf 25 \ -fflags nobuffer \ -f flv 'rtmp://[URL]' I have tried several vsync settings and ?map? variations. I believe the above command should try to sync the video stream to the audio stream, which seems most sensible for our scenario. I have tried it the other around as well, with no different results, i.e. A/V is still drifting. As I am able to confirm that the source application feeds very well. Also i have changed output rtmp to hls but result was same still drifting. In case of problem i checking source by dumping video and saw that there is no problem. I am break current connection and reconnect to same udp link then problem was solving but i can do this everytime. Here is my questions what is the source of problem and how can i understand video and audio out of sync. I am thinking if i can undestand drift between audio and video then i can reset the connection and then reconnect the source. This is console output, ffmpeg version git-2013-12-30-61d43a2 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 30 2013 23:01:15 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-gpl --enable-pthreads --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libass --enable-libfdk-aac --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-mmx --enable-nonfree --enable-version3 --enable-libx264 libavutil 52. 59.100 / 52. 59.100 libavcodec 55. 47.100 / 55. 47.100 libavformat 55. 22.101 / 55. 22.101 libavdevice 55. 5.102 / 55. 5.102 libavfilter 4. 0.103 / 4. 0.103 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [mpeg2video @ 0x2e4ace0] Invalid frame dimensions 0x0. Last message repeated 16 times Input #0, mpegts, from 'udp://@UDP LINK?fifo_size=1000000&overrun_nonfatal=1': Duration: N/A, start: 75687.034456, bitrate: 256 kb/s Program 1101 Stream #0:0[0x65]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv), 720x576 [SAR 64:45 DAR 16:9], max. 15000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0xc9](tur): Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 256 kb/s [libx264 @ 0x2eb0640] using SAR=64/45 [libx264 @ 0x2eb0640] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x2eb0640] profile High, level 3.0 [libx264 @ 0x2eb0640] 264 - core 140 r2 1ca7bb9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x3 me=dia subme=1 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=9 lookahead_threads=9 sliced_threads=1 slices=9 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=25.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=4000 vbv_bufsize=4000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00 Output #0, rawvideo, to 'mpegts': Metadata: encoder : Lavf55.22.101 Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 25 tbc Output #1, flv, to 'rtmp://[URL]': Metadata: encoder : Lavf55.22.101 Stream #1:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=-1--1, 4000 kb/s, 1k tbn, 25 tbc Stream #1:1(tur): Audio: aac (libfaac) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (mpeg2video -> rawvideo) Stream #0:0 -> #1:0 (mpeg2video -> libx264) Stream #0:1 -> #1:1 (mp2 -> libfaac) Press [q] to stop, [?] for help [flv @ 0x2eafe40] Failed to update header with correct duration.8 bitrate=124416.0kbits/s dup=78 drop=0 [flv @ 0x2eafe40] Failed to update header with correct filesize. frame= 98 fps= 43 q=0.0 Lq=27.0 size= 59535kB time=00:00:03.92 bitrate=124416.0kbits/s dup=78 drop=0 video:59989kB audio:45kB subtitle:0 global headers:0kB muxing overhead -0.832512% [libx264 @ 0x2eb0640] frame I:2 Avg QP:19.43 size: 47187 [libx264 @ 0x2eb0640] frame P:95 Avg QP:20.19 size: 3896 [libx264 @ 0x2eb0640] mb I I16..4: 8.9% 17.0% 74.1% [libx264 @ 0x2eb0640] mb P I16..4: 0.4% 0.6% 0.4% P16..4: 40.7% 0.0% 0.0% 0.0% 0.0% skip:58.0% [libx264 @ 0x2eb0640] 8x8 transform intra:26.4% inter:44.9% [libx264 @ 0x2eb0640] coded y,uvDC,uvAC intra: 75.2% 81.0% 55.3% inter: 11.6% 12.6% 1.1% [libx264 @ 0x2eb0640] i16 v,h,dc,p: 42% 41% 14% 3% [libx264 @ 0x2eb0640] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 16% 17% 6% 7% 9% 6% 9% 10% [libx264 @ 0x2eb0640] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 13% 13% 9% 7% 9% 6% 10% 8% [libx264 @ 0x2eb0640] i8c dc,h,v,p: 47% 20% 22% 11% [libx264 @ 0x2eb0640] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x2eb0640] kb/s:957.69 From rogerdpack2 at gmail.com Sat Nov 22 20:17:28 2014 From: rogerdpack2 at gmail.com (Roger Pack) Date: Sat, 22 Nov 2014 12:17:28 -0700 Subject: [FFmpeg-user] screencast in WinXP In-Reply-To: <546B830A.2020004@web.de> References: <5467C9A7.6030401@web.de> <5467CA98.8070705@gmail.com> <5467CD02.5010608@web.de> <5467D27C.2000501@gmail.com> <54685CFD.7090207@web.de> <54689C0A.7040307@web.de> <4B3C4839-46A3-4014-ACFE-6D10EA8AACB7@gmail.com> <54690303.6030401@web.de> <546A3CD6.3030609@web.de> <546B76E4.2020605@web.de> <546B830A.2020004@web.de> Message-ID: On Tue, Nov 18, 2014 at 10:34 AM, Pablo Rodr?guez wrote: > On 11/18/2014 06:28 PM, Roger Pack wrote: > > what's your output to > > > > ffmpeg -list_devices true -f dshow -i dummy > > Many thanks for your reply, Roger. > > The weird thing is that here it is detected: > > C:>ffmpeg -list_devices true -f dshow -i dummy > ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg > developers > built on Sep 15 2014 22:02:10 with gcc 4.8.3 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r > --enable-gnutls --enable-iconv --enable-libass --enable-libbluray > --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme > --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libopenjpeg --enable-libopus --enable-librtmp > --enable-libschroedinger --enable-libsoxr --enable-libspeex > --enable-libtheora --enable-libtwolame --enable-libvidstab > --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis > --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 > --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink > --enable-zlib > libavutil 54. 7.100 / 54. 7.100 > libavcodec 56. 1.100 / 56. 1.100 > libavformat 56. 4.101 / 56. 4.101 > libavdevice 56. 0.100 / 56. 0.100 > libavfilter 5. 1.100 / 5. 1.100 > libswscale 3. 0.100 / 3. 0.100 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 0.100 / 53. 0.100 > [dshow @ 02dde020] DirectShow video devices > [dshow @ 02dde020] Could not enumerate video devices. > [dshow @ 02dde020] DirectShow audio devices > [dshow @ 02dde020] "Realtek HD Audio Input" > [dshow @ 02dde020] "Realtek HD Digital input" > If you're trying to use that with -i audio="Microphone" You may want to try -i audio="Realtek HD Audio Input" instead. From rogerdpack2 at gmail.com Sat Nov 22 20:22:45 2014 From: rogerdpack2 at gmail.com (Roger Pack) Date: Sat, 22 Nov 2014 12:22:45 -0700 Subject: [FFmpeg-user] Problem with GDIGRAB device on Windows 7 In-Reply-To: <54320BF2.4030302@xopnetworks.com> References: <54320BF2.4030302@xopnetworks.com> Message-ID: On Sun, Oct 5, 2014 at 9:26 PM, Yan Brenman wrote: > Hello ffmpeg/gdigrab gurus! > > We are using "gdigrab" device on Windows to share/stream video for a > specific application window (identified by a title as "gdigrab" requires). > Everything works great on Window 8.x - actual application window (and not > the region on the desktop occupied by the application window) > is getting shared and successfully played by the player on the receiving > side. Which means that even if shared window is overlapped by the > other window on the source - still the application window is getting > played. > Unfortunately, as we discovered, "gdigrab" doesn't work correctly on > Window 7. And instead window sharing it actually does the region sharing. > Which pretty much means that if shared window is being overlapped by the > other window on the desktop - that's what getting played on the > receiving side. > Just for reference - we tried to do window sharing with > screen-capture-recorder and unfortunately got exactly the same results. > > I would greatly appreciate any help/advise anybody can provide on the > subject. And please let me know if there is any other information I can > provide. > Unfortunately there's no easy "fix" unless someone were to implement the "PrintWindow" API. Patches welcome, for sure. -roger- _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From chris at grimtech.net Sat Nov 22 11:08:55 2014 From: chris at grimtech.net (Chris Grimmett) Date: Sat, 22 Nov 2014 02:08:55 -0800 Subject: [FFmpeg-user] ffmpeg exits before streaming Message-ID: I'm trying to combine a music file, /persist/01.m4a with video, /persist/bars2.mp4 and stream it to ffserver. the following command works: ffmpeg -loglevel debug -i /persist/01.m4a -i /persist/bars2.mp4 -pix_fmt yuv420p -c:v libx264 -c:a libfdk_aac -s 320x240 -f flv /persist/test.flv the following command exits seemingly before anything is streamed: ffmpeg -loglevel debug -i /persist/01.m4a -i /persist/bars2.mp4 -pix_fmt yuv420p -c:v libx264 -c:a libfdk_aac -s 320x240 -f flv http://192.168.1.117:8090/feed1.ffm2 ffmpeg version git-2014-06-16-389d453 Copyright (c) 2000-2014 the FFmpeg developers built on Jun 16 2014 19:03:15 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree libavutil 52. 89.100 / 52. 89.100 libavcodec 55. 67.100 / 55. 67.100 libavformat 55. 43.100 / 55. 43.100 libavdevice 55. 13.101 / 55. 13.101 libavfilter 4. 8.100 / 4. 8.100 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 19.100 / 0. 19.100 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-i' ... matched as input file with argument '/persist/01.m4a'. Reading option '-i' ... matched as input file with argument '/persist/bars2.mp4'. Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument 'yuv420p'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'libx264'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'libfdk_aac'. Reading option '-s' ... matched as option 's' (set frame size (WxH or abbreviation)) with argument '320x240'. Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'. Reading option 'htp://192.168.1.117:8090/feed1.ffm2' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file /persist/01.m4a. Successfully parsed a group of options. Opening an input file: /persist/01.m4a. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] ISO: File Type Major Brand: M4A [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] stream 0, timescale not set [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] Before avformat_find_stream_info() pos: 592860 bytes read:163840 seeks:1 [mjpeg @ 0x1e11a00] marker=d8 avail_size_in_buf=35296 [mjpeg @ 0x1e11a00] marker parser used 0 bytes (0 bits) [mjpeg @ 0x1e11a00] marker=e0 avail_size_in_buf=35294 [mjpeg @ 0x1e11a00] marker parser used 16 bytes (128 bits) [mjpeg @ 0x1e11a00] marker=db avail_size_in_buf=35276 [mjpeg @ 0x1e11a00] index=0 [mjpeg @ 0x1e11a00] qscale[0]: 1 [mjpeg @ 0x1e11a00] marker parser used 67 bytes (536 bits) [mjpeg @ 0x1e11a00] marker=db avail_size_in_buf=35207 [mjpeg @ 0x1e11a00] index=1 [mjpeg @ 0x1e11a00] qscale[1]: 2 [mjpeg @ 0x1e11a00] marker parser used 67 bytes (536 bits) [mjpeg @ 0x1e11a00] marker=c0 avail_size_in_buf=35138 [mjpeg @ 0x1e11a00] sof0: picture: 600x600 [mjpeg @ 0x1e11a00] component 0 1:1 id: 0 quant:0 [mjpeg @ 0x1e11a00] component 1 1:1 id: 1 quant:1 [mjpeg @ 0x1e11a00] component 2 1:1 id: 2 quant:1 [mjpeg @ 0x1e11a00] pix fmt id 11111100 [mjpeg @ 0x1e11a00] marker parser used 17 bytes (136 bits) [mjpeg @ 0x1e11a00] marker=c4 avail_size_in_buf=35119 [mjpeg @ 0x1e11a00] class=0 index=0 nb_codes=11 [mjpeg @ 0x1e11a00] marker parser used 30 bytes (240 bits) [mjpeg @ 0x1e11a00] marker=c4 avail_size_in_buf=35087 [mjpeg @ 0x1e11a00] class=1 index=0 nb_codes=243 [mjpeg @ 0x1e11a00] marker parser used 110 bytes (880 bits) [mjpeg @ 0x1e11a00] marker=c4 avail_size_in_buf=34975 [mjpeg @ 0x1e11a00] class=0 index=1 nb_codes=8 [mjpeg @ 0x1e11a00] marker parser used 27 bytes (216 bits) [mjpeg @ 0x1e11a00] marker=c4 avail_size_in_buf=34946 [mjpeg @ 0x1e11a00] class=1 index=1 nb_codes=178 [mjpeg @ 0x1e11a00] marker parser used 49 bytes (392 bits) [mjpeg @ 0x1e11a00] escaping removed 106 bytes [mjpeg @ 0x1e11a00] marker=da avail_size_in_buf=34895 [mjpeg @ 0x1e11a00] component: 0 [mjpeg @ 0x1e11a00] component: 1 [mjpeg @ 0x1e11a00] component: 2 [mjpeg @ 0x1e11a00] marker parser used 34788 bytes (278302 bits) [mjpeg @ 0x1e11a00] marker=d9 avail_size_in_buf=0 [mjpeg @ 0x1e11a00] decode frame unused 0 bytes [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] demuxer injecting skip 2112 [aac @ 0x1e10720] skip 2112 samples due to side data [aac @ 0x1e10720] skip whole frame, skip left: 1088 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] max_analyze_duration 5000000 reached at 5015510 microseconds [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e0fd20] After avformat_find_stream_info() pos: 775167 bytes read:327680 seeks:1 frames:219 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/persist/01.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2033-04-24 17:18:04 iTunNORM : 0000138E 00001755 00007734 00008265 0000D0FB 0000C55F 00007C24 00007C24 000184EF 00001A1F title : It Goes On artist : Louis Cole and Genevieve Artadi & Pomplamoose album_artist : Louis Cole and Genevieve Artadi & Pomplamoose album : It Goes On - Single genre : Pop track : 1/1 disc : 1/1 compilation : 0 date : 2011-02-09T08:00:00Z media_type : 1 Duration: 00:03:13.05, start: 0.000000, bitrate: 314 kb/s Stream #0:0(eng), 218, 1/44100: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 289 kb/s (default) Metadata: creation_time : 2033-04-24 17:18:04 Stream #0:1, 1, 1/90000: Video: mjpeg, yuvj444p(pc), 600x600 [SAR 72:72 DAR 1:1], 1/90000, 90k tbr, 90k tbn, 90k tbc Successfully opened the file. Parsing a group of options: input file /persist/bars2.mp4. Successfully parsed a group of options. Opening an input file: /persist/bars2.mp4. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e2fa60] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e2fa60] ISO: File Type Major Brand: isom [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e2fa60] Before avformat_find_stream_info() pos: 8018567 bytes read:426995 seeks:1 [h264 @ 0x1e2e820] no picture [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e2fa60] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1e2fa60] After avformat_find_stream_info() pos: 4890 bytes read:459763 seeks:2 frames:4 Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '/persist/bars2.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.43.100 Duration: 00:16:40.00, start: 0.000000, bitrate: 64 kb/s Stream #1:0(und), 3, 1/12800: Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 320x240 [SAR 1:1 DAR 4:3], 1/50, 36 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: handler_name : VideoHandler Stream #1:1(eng), 1, 1/44100: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default) Metadata: handler_name : SoundHandler Successfully opened the file. Parsing a group of options: output file http://192.168.1.117:8090/feed1.ffm2 . Applying option pix_fmt (set pixel format) with argument yuv420p. Applying option c:v (codec name) with argument libx264. Applying option c:a (codec name) with argument libfdk_aac. Applying option s (set frame size (WxH or abbreviation)) with argument 320x240. Applying option f (force format) with argument flv. Successfully parsed a group of options. Opening an output file: http://192.168.1.117:8090/feed1.ffm2. [http @ 0x1e2e700] request: POST /feed1.ffm2 HTTP/1.1 Transfer-Encoding: chunked User-Agent: Lavf/55.43.100 Accept: */* Connection: close Host: 192.168.1.117:8090 Successfully opened the file. detected 4 logical cores [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'video_size' to value '320x240' [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'pix_fmt' to value '5' [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'time_base' to value '1/12800' [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'pixel_aspect' to value '1/1' [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'sws_param' to value 'flags=2' [graph 0 input from stream 1:0 @ 0x1df6880] Setting 'frame_rate' to value '25/1' [graph 0 input from stream 1:0 @ 0x1df6880] w:320 h:240 pixfmt:yuv444p tb:1/12800 fr:25/1 sar:1/1 sws_param:flags=2 [scaler for output stream 0:0 @ 0x1df6ec0] Setting 'w' to value '320' [scaler for output stream 0:0 @ 0x1df6ec0] Setting 'h' to value '240' [scaler for output stream 0:0 @ 0x1df6ec0] Setting 'flags' to value '0x4' [scaler for output stream 0:0 @ 0x1df6ec0] w:320 h:240 flags:'0x4' interl:0 [format @ 0x1e135c0] compat: called with args=[yuv420p] [format @ 0x1e135c0] Setting 'pix_fmts' to value 'yuv420p' [AVFilterGraph @ 0x1df6e20] query_formats: 5 queried, 4 merged, 0 already done, 0 delayed [scaler for output stream 0:0 @ 0x1df6ec0] w:320 h:240 fmt:yuv444p sar:1/1 -> w:320 h:240 fmt:yuv420p sar:1/1 flags:0x4 [graph 1 input from stream 0:0 @ 0x1e0c680] Setting 'time_base' to value '1/44100' [graph 1 input from stream 0:0 @ 0x1e0c680] Setting 'sample_rate' to value '44100' [graph 1 input from stream 0:0 @ 0x1e0c680] Setting 'sample_fmt' to value 'fltp' [graph 1 input from stream 0:0 @ 0x1e0c680] Setting 'channel_layout' to value '0x3' [graph 1 input from stream 0:0 @ 0x1e0c680] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:0x3 [audio format for output stream 0:1 @ 0x1e0cfe0] Setting 'sample_fmts' to value 's16' [audio format for output stream 0:1 @ 0x1e0cfe0] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000' [audio format for output stream 0:1 @ 0x1e0cfe0] Setting 'channel_layouts' to value '0x4|0x3|0x7|0x107|0x37|0x3f|0xff|0x63f' [audio format for output stream 0:1 @ 0x1e0cfe0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1' [AVFilterGraph @ 0x1e07460] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x1dffb80] ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz [libx264 @ 0x1e2b6a0] using mv_range_thread = 24 [libx264 @ 0x1e2b6a0] using SAR=1/1 [libx264 @ 0x1e2b6a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x1e2b6a0] profile High, level 1.3 [libx264 @ 0x1e2b6a0] 264 - core 142 r2 a5831aa - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'http://192.168.1.117:8090/feed1.ffm2': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom media_type : 1 iTunNORM : 0000138E 00001755 00007734 00008265 0000D0FB 0000C55F 00007C24 00007C24 000184EF 00001A1F title : It Goes On artist : Louis Cole and Genevieve Artadi & Pomplamoose album_artist : Louis Cole and Genevieve Artadi & Pomplamoose album : It Goes On - Single genre : Pop track : 1/1 disc : 1/1 compilation : 0 date : 2011-02-09T08:00:00Z encoder : Lavf55.43.100 Stream #0:0(und), 0, 1/1000: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 1/25, q=-1--1, 25 fps, 1k tbn, 25 tbc (default) Metadata: handler_name : VideoHandler encoder : Lavc55.67.100 libx264 Stream #0:1(eng), 0, 1/1000: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 128 kb/s (default) Metadata: creation_time : 2033-04-24 17:18:04 encoder : Lavc55.67.100 libfdk_aac Stream mapping: Stream #1:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:0 -> #0:1 (aac (native) -> aac (libfdk_aac)) Press [q] to stop, [?] for help [h264 @ 0x2a69860] no picture [h264 @ 0x2aaea00] no picture [libx264 @ 0x1e2b6a0] frame= 0 QP=20.56 NAL=3 Slice:I Poc:0 I:300 P:0 SKIP:0 size=3511 bytes [libx264 @ 0x1e2b6a0] frame= 1 QP=21.65 NAL=2 Slice:P Poc:8 I:9 P:62 SKIP:229 size=605 bytes [libx264 @ 0x1e2b6a0] frame= 2 QP=14.78 NAL=2 Slice:B Poc:4 I:1 P:27 SKIP:272 size=88 bytes [libx264 @ 0x1e2b6a0] frame= 3 QP=15.60 NAL=0 Slice:B Poc:2 I:1 P:20 SKIP:279 size=52 bytes [aac @ 0x1e2f120] skip 2112 samples due to side data [aac @ 0x1e2f120] skip whole frame, skip left: 1088 [aac @ 0x1e2f120] skip whole frame, skip left: 64 [aac @ 0x1e2f120] skip 64/1024 samples root at 1ba4795b8c0c:/tmp# Using the command that saves to disk, the process takes several minutes. When I attempt to stream, ffmpeg seems like it doesn't do anything, it just skips doing what I asked it to do and exits. Lots of "skip"s everywhere. I'm assuming the "side data" is the problem. Also there's that h264 saying, "no picture" which might be the problem. My questions... What does "skip 2112 samples due to side data" mean? What is side data? Is there a way I can fix the skipping? From hugh.welles at gmail.com Fri Nov 21 17:57:59 2014 From: hugh.welles at gmail.com (Hugh Welles) Date: Fri, 21 Nov 2014 11:57:59 -0500 Subject: [FFmpeg-user] config error Message-ID: Hello, I've tried to install the latest ffmpeg on ubuntu 14.04. The configuration didn't go so well. In summary, I tried to configure and yasm was a problem, so I tried the crippled build(no idea what that is). Then when I try to make, I get an error. I've copied the terminal out put here: hugh at Dat-Pad:~/Documents/ffmpeg-2.4.3$ PKG_CONFIG_PATH=/tmp/ffmpeg/lib/pkgconfig ./configure yasm/nasm not found or too old. Use --disable-yasm for a crippled build. 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 solve the problem. hugh at Dat-Pad:~/Documents/ffmpeg-2.4.3$ ^C hugh at Dat-Pad:~/Documents/ffmpeg-2.4.3$ PKG_CONFIG_PATH=/tmp/ffmpeg/lib/pkgconfig ./configure --disable-yasm install prefix /usr/local source path . C compiler gcc C library glibc ARCH x86 (generic) big-endian no runtime cpu detection yes yasm no MMX enabled yes MMXEXT enabled yes 3DNow! enabled yes 3DNow! extended enabled yes SSE enabled yes SSSE3 enabled yes AVX enabled yes XOP enabled yes FMA3 enabled yes FMA4 enabled yes i686 features enabled yes CMOV is fast yes EBX available yes EBP available yes debug symbols yes strip symbols yes optimize for size no optimizations yes static yes shared no postprocessing support no new filter support yes network support yes threading support pthreads safe bitstream reader yes SDL support no opencl enabled no texi2html enabled no perl enabled yes pod2man enabled yes makeinfo enabled no makeinfo supports HTML no External libraries: iconv xlib zlib Enabled decoders: aac bink gsm_ms aac_latm binkaudio_dct h261 aasc binkaudio_rdft h263 ac3 bintext h263i ac3_fixed bmp h263p adpcm_4xm bmv_audio h264 adpcm_adx bmv_video hevc adpcm_afc brender_pix hnm4_video adpcm_ct c93 huffyuv adpcm_dtk cavs iac adpcm_ea cdgraphics idcin adpcm_ea_maxis_xa cdxl idf adpcm_ea_r1 cinepak iff_byterun1 adpcm_ea_r2 cljr iff_ilbm adpcm_ea_r3 cllc imc adpcm_ea_xas comfortnoise indeo2 adpcm_g722 cook indeo3 adpcm_g726 cpia indeo4 adpcm_g726le cscd indeo5 adpcm_ima_amv cyuv interplay_dpcm adpcm_ima_apc dca interplay_video adpcm_ima_dk3 dfa jacosub adpcm_ima_dk4 dirac jpeg2000 adpcm_ima_ea_eacs dnxhd jpegls adpcm_ima_ea_sead dpx jv adpcm_ima_iss dsd_lsbf kgv1 adpcm_ima_oki dsd_lsbf_planar kmvc adpcm_ima_qt dsd_msbf lagarith adpcm_ima_rad dsd_msbf_planar loco adpcm_ima_smjpeg dsicinaudio mace3 adpcm_ima_wav dsicinvideo mace6 adpcm_ima_ws dvbsub mdec adpcm_ms dvdsub metasound adpcm_sbpro_2 dvvideo microdvd adpcm_sbpro_3 dxa mimic adpcm_sbpro_4 dxtory mjpeg adpcm_swf eac3 mjpegb adpcm_thp eacmv mlp adpcm_vima eamad mmvideo adpcm_xa eatgq motionpixels adpcm_yamaha eatgv movtext aic eatqi mp1 alac eightbps mp1float alias_pix eightsvx_exp mp2 als eightsvx_fib mp2float amrnb escape124 mp3 amrwb escape130 mp3adu amv evrc mp3adufloat anm exr mp3float ansi ffv1 mp3on4 ape ffvhuff mp3on4float ass ffwavesynth mpc7 asv1 fic mpc8 asv2 flac mpeg1video atrac1 flashsv mpeg2video atrac3 flashsv2 mpeg4 atrac3p flic mpegvideo aura flv mpl2 aura2 fourxm msa1 avrn fraps msmpeg4v1 avrp frwu msmpeg4v2 avs g2m msmpeg4v3 avui g723_1 msrle ayuv g729 mss1 bethsoftvid gif mss2 bfi gsm msvideo1 mszh qtrle ulti mts2 r10k utvideo mvc1 r210 v210 mvc2 ra_144 v210x mxpeg ra_288 v308 nellymoser ralf v408 nuv rawvideo v410 on2avc realtext vb opus rl2 vble paf_audio roq vc1 paf_video roq_dpcm vc1image pam rpza vcr1 pbm rv10 vima pcm_alaw rv20 vmdaudio pcm_bluray rv30 vmdvideo pcm_dvd rv40 vmnc pcm_f32be s302m vorbis pcm_f32le sami vp3 pcm_f64be sanm vp5 pcm_f64le sgi vp6 pcm_lxf sgirle vp6a pcm_mulaw shorten vp6f pcm_s16be sipr vp7 pcm_s16be_planar smackaud vp8 pcm_s16le smacker vp9 pcm_s16le_planar smc vplayer pcm_s24be smvjpeg vqa pcm_s24daud snow wavpack pcm_s24le sol_dpcm webp pcm_s24le_planar sonic webvtt pcm_s32be sp5x wmalossless pcm_s32le srt wmapro pcm_s32le_planar ssa wmav1 pcm_s8 subrip wmav2 pcm_s8_planar subviewer wmavoice pcm_u16be subviewer1 wmv1 pcm_u16le sunrast wmv2 pcm_u24be svq1 wmv3 pcm_u24le svq3 wmv3image pcm_u32be tak wnv1 pcm_u32le targa ws_snd1 pcm_u8 targa_y216 xan_dpcm pcm_zork text xan_wc3 pcx theora xan_wc4 pgm thp xbin pgmyuv tiertexseqvideo xbm pgssub tiff xface pictor tmv xl pjs truehd xsub png truemotion1 xwd ppm truemotion2 y41p prores truespeech yop prores_lgpl tscc yuv4 ptx tscc2 zero12v qcelp tta zerocodec qdm2 twinvq zlib qdraw txd zmbv qpeg Enabled encoders: a64multi jpegls prores a64multi5 ljpeg prores_aw aac mjpeg prores_ks ac3 movtext qtrle ac3_fixed mp2 r10k adpcm_adx mp2fixed r210 adpcm_g722 mpeg1video ra_144 adpcm_g726 mpeg2video rawvideo adpcm_ima_qt mpeg4 roq adpcm_ima_wav msmpeg4v2 roq_dpcm adpcm_ms msmpeg4v3 rv10 adpcm_swf msvideo1 rv20 adpcm_yamaha nellymoser s302m alac pam sgi alias_pix pbm snow amv pcm_alaw sonic ass pcm_f32be sonic_ls asv1 pcm_f32le srt asv2 pcm_f64be ssa avrp pcm_f64le subrip avui pcm_mulaw sunrast ayuv pcm_s16be svq1 bmp pcm_s16be_planar targa cinepak pcm_s16le tiff cljr pcm_s16le_planar tta comfortnoise pcm_s24be utvideo dca pcm_s24daud v210 dnxhd pcm_s24le v308 dpx pcm_s24le_planar v408 dvbsub pcm_s32be v410 dvdsub pcm_s32le vorbis dvvideo pcm_s32le_planar wavpack eac3 pcm_s8 webvtt ffv1 pcm_s8_planar wmav1 ffvhuff pcm_u16be wmav2 flac pcm_u16le wmv1 flashsv pcm_u24be wmv2 flashsv2 pcm_u24le xbm flv pcm_u32be xface g723_1 pcm_u32le xsub gif pcm_u8 xwd h261 pcx y41p h263 pgm yuv4 h263p pgmyuv zlib huffyuv png zmbv jpeg2000 ppm Enabled hwaccels: Enabled parsers: aac dvd_nav mpegvideo aac_latm dvdsub opus ac3 flac png adx gsm pnm bmp h261 rv30 cavsvideo h263 rv40 cook h264 tak dca hevc vc1 dirac mjpeg vorbis dnxhd mlp vp3 dpx mpeg4video vp8 dvbsub mpegaudio vp9 Enabled demuxers: aac h263 nistsphere ac3 h264 nsv act hevc nut adf hls nuv adp hnm ogg adx ico oma aea idcin paf afc idf pcm_alaw aiff iff pcm_f32be amr ilbc pcm_f32le anm image2 pcm_f64be apc image2_alias_pix pcm_f64le ape image2_brender_pix pcm_mulaw aqtitle image2pipe pcm_s16be asf image_bmp_pipe pcm_s16le ass image_dpx_pipe pcm_s24be ast image_exr_pipe pcm_s24le au image_j2k_pipe pcm_s32be avi image_jpegls_pipe pcm_s32le avr image_pictor_pipe pcm_s8 avs image_png_pipe pcm_u16be bethsoftvid image_sgi_pipe pcm_u16le bfi image_sunrast_pipe pcm_u24be bink image_tiff_pipe pcm_u24le bintext image_webp_pipe pcm_u32be bit ingenient pcm_u32le bmv ipmovie pcm_u8 boa ircam pjs brstm iss pmp c93 iv8 pva caf ivf pvf cavsvideo jacosub qcp cdg jv r3d cdxl latm rawvideo cine live_flv realtext concat lmlm4 redspark data loas rl2 daud lrc rm dfa lvf roq dirac lxf rpl dnxhd m4v rsd dsf matroska rso dsicin mgsts rtp dts microdvd rtsp dtshd mjpeg sami dv mlp sap dxa mlv sbg ea mm sdp ea_cdata mmf sdr2 eac3 mov segafilm epaf mp3 shorten ffm mpc siff ffmetadata mpc8 sln filmstrip mpegps smacker flac mpegts smjpeg flic mpegtsraw smush flv mpegvideo sol fourxm mpl2 sox frm mpsub spdif g722 msnwc_tcp srt g723_1 mtv str g729 mv subviewer gif mvi subviewer1 gsm mxf swf gxf mxg tak h261 nc tedcaptions thp vmd wsaud tiertexseq vobsub wsvqa tmv voc wtv truehd vplayer wv tta vqf xa tty w64 xbin txd wav xmv vc1 wc3 xwma vc1t webm_dash_manifest yop vivo webvtt yuv4mpegpipe Enabled muxers: a64 ipod pcm_s24be ac3 ircam pcm_s24le adts ismv pcm_s32be adx ivf pcm_s32le aiff jacosub pcm_s8 amr latm pcm_u16be asf lrc pcm_u16le asf_stream m4v pcm_u24be ass matroska pcm_u24le ast matroska_audio pcm_u32be au md5 pcm_u32le avi microdvd pcm_u8 avm2 mjpeg psp bit mkvtimestamp_v2 rawvideo caf mlp rm cavsvideo mmf roq crc mov rso data mp2 rtp daud mp3 rtsp dirac mp4 sap dnxhd mpeg1system segment dts mpeg1vcd smjpeg dv mpeg1video smoothstreaming eac3 mpeg2dvd sox f4v mpeg2svcd spdif ffm mpeg2video spx ffmetadata mpeg2vob srt filmstrip mpegts stream_segment flac mpjpeg swf flv mxf tee framecrc mxf_d10 tg2 framemd5 null tgp g722 nut truehd g723_1 oga uncodedframecrc gif ogg vc1 gxf oma vc1t h261 opus voc h263 pcm_alaw w64 h264 pcm_f32be wav hds pcm_f32le webm hevc pcm_f64be webm_dash_manifest hls pcm_f64le webvtt ico pcm_mulaw wtv ilbc pcm_s16be wv image2 pcm_s16le yuv4mpegpipe image2pipe Enabled protocols: cache hls rtmp concat http rtmpt crypto httpproxy rtp data icecast srtp ffrtmphttp md5 subfile file mmsh tcp ftp mmst udp gopher pipe unix Enabled filters: aconvert copy nullsink adelay crop nullsrc aecho curves overlay aeval dctdnoiz pad aevalsrc decimate pan afade dejudder perms aformat deshake pixdesctest ainterleave drawbox psnr allpass drawgrid removelogo alphaextract earwax replaygain alphamerge edgedetect rgbtestsrc amerge elbg rotate amix equalizer scale amovie extractplanes select anull fade sendcmd anullsink field separatefields anullsrc fieldmatch setdar apad fieldorder setfield aperms flanger setpts aphaser format setsar aresample fps settb aselect framepack showcqt asendcmd framestep showinfo asetnsamples gradfun showspectrum asetpts haldclut showwaves asetrate haldclutsrc shuffleplanes asettb hflip signalstats ashowinfo highpass silencedetect asplit histogram silenceremove astats hqx sine astreamsync hue smptebars atempo idet smptehdbars atrim il split avectorscope interleave swapuv bandpass join telecine bandreject lenscorrection testsrc bass life thumbnail bbox lowpass tile biquad lut transpose blackdetect lut3d treble blend lutrgb trim cellauto lutyuv unsharp channelmap mandelbrot vflip channelsplit mergeplanes vignette codecview movie volume color negate volumedetect colorbalance noformat w3fdif colorchannelmixer noise yadif compand null zoompan concat Enabled bsfs: aac_adtstoasc imx_dump_header mp3_header_decompress chomp mjpeg2jpeg noise dump_extradata mjpega_dump_header remove_extradata h264_mp4toannexb mov2textsub text2movsub Enabled indevs: dv1394 lavfi v4l2 fbdev oss Enabled outdevs: fbdev oss v4l2 License: LGPL version 2.1 or later Creating config.mak, config.h, and doc/config.texi... libavutil/avconfig.h is unchanged hugh at Dat-Pad:~/Documents/ffmpeg-2.4.3$ make touch: cannot touch ?.version?: Permission denied CC libavdevice/alldevices.o libavdevice/alldevices.c:77:1: fatal error: opening dependency file libavdevice/alldevices.d: Permission denied } ^ compilation terminated. make: *** [libavdevice/alldevices.o] Error 1 Thanks for any help! Hugh -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: text/x-log Size: 147929 bytes Desc: not available URL: From jshupert at theppsgroup.com Sat Nov 22 23:16:11 2014 From: jshupert at theppsgroup.com (Jim Shupert, Jr.) Date: Sat, 22 Nov 2014 17:16:11 -0500 Subject: [FFmpeg-user] brew install ffmpeg on 10.10 Yosemite for wma to mp3 batch conv. In-Reply-To: <1416394414529-4668194.post@n4.nabble.com> References: <1416394414529-4668194.post@n4.nabble.com> Message-ID: <4ee19d400d6fd7ce6c5e5576bd926b51.squirrel@webmail.theppsgroup.com> > bash // > brew install ffmpeg // seems to work > cd folder with wmas > for f in *.wma; do ffmpeg -y -i "$f" -c:a libfdk_aac -b:a 192k > "${f%.wma}.m4a"; done; > // try a script > -> error Unknown encoder 'libfdk_aac' > > > any ideas? > > the happiest solution would be that you simply do not have libfdk_aac but DO have an alternative aac codec try a ffmpeg -codecs | grep aac and see what aac codecs you have From mohanraj.k at stellentsoft.com Sat Nov 22 07:03:19 2014 From: mohanraj.k at stellentsoft.com (mohanraj kandregula) Date: Sat, 22 Nov 2014 11:33:19 +0530 Subject: [FFmpeg-user] ffmpeg is not building in my mac os Message-ID: FFmpeg library is not building in my mac os giving the error gcc is unable to execute the executable files and the c compiler test failed.see the attached config.log file attached to the mail. -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 141968 bytes Desc: not available URL: From skwaap at gmail.com Sat Nov 22 19:31:01 2014 From: skwaap at gmail.com (Steven D) Date: Sat, 22 Nov 2014 13:31:01 -0500 Subject: [FFmpeg-user] Accurate Seeking without Re-encoding Message-ID: Hello, I'm trying to accurately (not just nearest keyframe) cut out a subtrack of an mp4 without having to re-encode the *entire* subtrack. It seems like overkill to re-encode everything just to get a few additional seconds of accuracy at the beginning. Is there a good way to avoid this? I'm perfectly happy to re-encode a bit of the subtrack, I just don't want to have to re-encode everything. My current thought was to re-encode from an accurate start point to the next keyframe, then copy from the keyframe to the end of the subtrack, and then concatenate the pieces using the concat demuxer. Is this a reasonable approach, or is there a better way to go about things? I can get the pieces of the mp4 to start and stop at the right times, but so far I haven't been able to get the concat demuxer to work on one re-encoded mp4 and one stream copied mp4. I'm not exactly sure what has to match between the re-encoded piece and the copied piece. I'd like the final output to be an mp4. Thanks in advance! From cvadim at me.com Sun Nov 23 16:41:08 2014 From: cvadim at me.com (Vadim Chernykh) Date: Sun, 23 Nov 2014 18:41:08 +0300 Subject: [FFmpeg-user] yasm/nasm not found or too old. Use --disable-yasm for a crippled build. Message-ID: Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 yasm/nasm not found or too old. Use --disable-yasm for a crippled build. 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 solve the problem. From h.reindl at thelounge.net Sun Nov 23 20:19:25 2014 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 23 Nov 2014 20:19:25 +0100 Subject: [FFmpeg-user] yasm/nasm not found or too old. Use --disable-yasm for a crippled build. In-Reply-To: References: Message-ID: <5472333D.1020309@thelounge.net> Am 23.11.2014 um 16:41 schrieb Vadim Chernykh: > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr --with-gxx-include-dir=/usr/include/c++/4.2.1 > yasm/nasm not found or too old. Use --disable-yasm for a crippled build. > > 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 > solve the problem and *what* is your question? just follow the instructions you posted -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 181 bytes Desc: OpenPGP digital signature URL: From bza.salman at gmail.com Sun Nov 23 20:19:56 2014 From: bza.salman at gmail.com (Walid Salman) Date: Sun, 23 Nov 2014 22:49:56 +0330 Subject: [FFmpeg-user] yasm/nasm not found or too old. Use --disable-yasm for a crippled build. In-Reply-To: References: Message-ID: Hello, Kindly some one told me wear I can find the configuration file of the ffmpeg on Centos6.5 server ? Thanks, On Sun, Nov 23, 2014 at 7:11 PM, Vadim Chernykh wrote: > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr > --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr > --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr > --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr > --with-gxx-include-dir=/usr/include/c++/4.2.1 > Configured with: --prefix=/Applications/Xcode.app/Contents/Developer/usr > --with-gxx-include-dir=/usr/include/c++/4.2.1 > yasm/nasm not found or too old. Use --disable-yasm for a crippled build. > > 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 > solve the problem. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From yiminhe at gmail.com Mon Nov 24 12:12:21 2014 From: yiminhe at gmail.com (yiminhe at gmail.com) Date: Mon, 24 Nov 2014 19:12:21 +0800 Subject: [FFmpeg-user] duration problem of concatenated audio in powerpoint Message-ID: <201411241912178814793@gmail.com> Hi all, I'm using ffmpeg to concatenate some mp3 audios. There is no problem when playing the concatenated audio with VLC and WMP. But if I inserted them into PPT files with Microsoft PowerPoint, the duration displayed in powerpoint is smaller than actual duration. And the player stopped very early (at position ~70%) if I playing them in fullscreen mode in powerpoint. Here is the minimal example code: [root at localhost audio-concatenate]# ll total 24 -rw-r--r-- 1 root root 19688 Nov 24 11:19 a.mp3 drwxr-xr-x 3 root root 4096 Nov 18 11:37 mp3 [root at localhost audio-concatenate]# ffmpeg -i a.mp3 -i a.mp3 -filter_complex "concat=n=2:v=0:a=1" two_a.mp3 ffmpeg version 2.4.3- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2014 the FFmpeg developers built on Nov 4 2014 13:14:24 with gcc 4.8 (Debian 4.8.3-13) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable-libass --enable-gnutls --cc=gcc-4.8 libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [mp3 @ 0x27d45c0] Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'a.mp3': Metadata: encoder : Lavf56.4.101 Duration: 00:00:04.91, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 32 kb/s [mp3 @ 0x27e4d40] Estimating duration from bitrate, this may be inaccurate Input #1, mp3, from 'a.mp3': Metadata: encoder : Lavf56.4.101 Duration: 00:00:04.91, start: 0.000000, bitrate: 32 kb/s Stream #1:0: Audio: mp3, 22050 Hz, mono, s16p, 32 kb/s Output #0, mp3, to 'two_a.mp3': Metadata: TSSE : Lavf56.4.101 Stream #0:0: Audio: mp3 (libmp3lame), 22050 Hz, mono, s16p (default) Metadata: encoder : Lavc56.1.100 libmp3lame Stream mapping: Stream #0:0 (mp3) -> concat:in0:a0 Stream #1:0 (mp3) -> concat:in1:a0 concat -> Stream #0:0 (libmp3lame) Press [q] to stop, [?] for help size= 39kB time=00:00:09.82 bitrate= 32.3kbits/s video:0kB audio:39kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.506368% How can I fix this? Thanks! yiminhe at gmail.com From n32 at email.cz Mon Nov 24 15:11:56 2014 From: n32 at email.cz (n32 at email.cz) Date: Mon, 24 Nov 2014 15:11:56 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder Message-ID: Hi all, I used these commands to encode my videos: for pass in 1 2; do ? mencoder -oac mp3lame -lameopts cbr:br=128 -ovc x264 -x264encopts pass=$ pass:preset=veryslow:fast_pskip=0:tune=film:frameref=15:bitrate=1000 -o out. avi in.avi done I wanted to change to ffmpeg because of troubles with A/V sync, so I encoded by these commands: for pass in 1 2; do ? ffmpeg -y -i in.avi -c:a libmp3lame -b:a 128k -c:v libx264 -b:v 1000k - minrate:v 1000k -maxrate:v 1000k -bufsize 1835k -pass $pass -preset veryslow -tune film -x264-params fast_pskip=0:frameref=15 out.avi done But I didn't get the same quality, I had to add 15 % to bitrate to get the same visual quality. Is there any parameter which is set in mencoder but not in ffmpeg? I tried to set a dozen of parameters in ffmpeg to the defaults of mencoder but haven't found the responsible option. Thank you in advance for any advice, Jan Sever From james.darnley at gmail.com Mon Nov 24 15:30:01 2014 From: james.darnley at gmail.com (James Darnley) Date: Mon, 24 Nov 2014 15:30:01 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: References: Message-ID: <547340E9.3080704@gmail.com> On 2014-11-24 15:11, n32 at email.cz wrote: > Hi all, > > I used these commands to encode my videos: > > for pass in 1 2; do > mencoder -oac mp3lame -lameopts cbr:br=128 -ovc x264 -x264encopts pass=$ > pass:preset=veryslow:fast_pskip=0:tune=film:frameref=15:bitrate=1000 -o out. > avi in.avi > done > > I wanted to change to ffmpeg because of troubles with A/V sync, so I encoded > by these commands: > > for pass in 1 2; do > ffmpeg -y -i in.avi -c:a libmp3lame -b:a 128k -c:v libx264 -b:v 1000k - > minrate:v 1000k -maxrate:v 1000k -bufsize 1835k -pass $pass -preset veryslow > -tune film -x264-params fast_pskip=0:frameref=15 out.avi > done You are setting VBV options for ffmpeg and not in mencoder (and forcing CBR at that) do I'm not surprised you get different output. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From n32 at email.cz Mon Nov 24 15:40:30 2014 From: n32 at email.cz (n32 at email.cz) Date: Mon, 24 Nov 2014 15:40:30 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder References: <547340E9.3080704@gmail.com> Message-ID: If you mean "-b:v 1000k", ffmpeg doesn't allow me to run 2-pass without it. Is there another way to get CBR with ffmpeg? P. S. Thanks for your very quick answer. On 2014-11-24 15:30, James Darnley wrote: "You are setting VBV options for ffmpeg and not in mencoder (and forcing CBR at that) do I'm not surprised you get different output." From james.darnley at gmail.com Mon Nov 24 15:44:29 2014 From: james.darnley at gmail.com (James Darnley) Date: Mon, 24 Nov 2014 15:44:29 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: References: <547340E9.3080704@gmail.com> Message-ID: <5473444D.4050002@gmail.com> On 2014-11-24 15:40, n32 at email.cz wrote: > If you mean "-b:v 1000k", ffmpeg doesn't allow me to run 2-pass without it. > Is there another way to get CBR with ffmpeg? Why do you want CBR? I don't think mencoder is giving you CBR. I did not mean that bitrate but the maxrate and bufsize you have given. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 603 bytes Desc: OpenPGP digital signature URL: From h.reindl at thelounge.net Mon Nov 24 15:45:00 2014 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 24 Nov 2014 15:45:00 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: References: <547340E9.3080704@gmail.com> Message-ID: <5473446C.2050608@thelounge.net> Am 24.11.2014 um 15:40 schrieb n32 at email.cz: > If you mean "-b:v 1000k", ffmpeg doesn't allow me to run 2-pass without it. > Is there another way to get CBR with ffmpeg? that's not the problem *but* you set the 1000k on more places and so force CBR instead VBR which is bad, especially in context of 2-pass because that can *heavily* benefit of VBR in both directions for a optimized balance quality/size > On 2014-11-24 15:30, James Darnley wrote: > > "You are setting VBV options for ffmpeg and not in mencoder (and forcing > CBR at that) do I'm not surprised you get different output." -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 181 bytes Desc: OpenPGP digital signature URL: From n32 at email.cz Mon Nov 24 15:55:23 2014 From: n32 at email.cz (n32 at email.cz) Date: Mon, 24 Nov 2014 15:55:23 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder References: <547340E9.3080704@gmail.com> <5473444D.4050002@gmail.com> Message-ID: On 2014-11-24 15:44, James Darnley wrote: "Why do you want CBR? I don't think mencoder is giving you CBR." Mainly because of network throughput. Mencoder can make it pretty well. "I did not mean that bitrate but the maxrate and bufsize you have given." I tried to encode it without maxrate and the result was unfortunately worse. "" From n32 at email.cz Mon Nov 24 15:57:16 2014 From: n32 at email.cz (n32 at email.cz) Date: Mon, 24 Nov 2014 15:57:16 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> Message-ID: On 2014-11-24 15:45, Reindl Harald wrote: "that's not the problem *but* you set the 1000k on more places and so force CBR instead VBR which is bad, especially in context of 2-pass because that can *heavily* benefit of VBR in both directions for a optimized balance quality/size" Omitting maxrate gave me worse results. "" From cehoyos at ag.or.at Mon Nov 24 15:57:11 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 24 Nov 2014 14:57:11 +0000 (UTC) Subject: [FFmpeg-user] duration problem of concatenated audio in powerpoint References: <201411241912178814793@gmail.com> Message-ID: yiminhe gmail.com gmail.com> writes: > But if I inserted them into PPT files with Microsoft > PowerPoint, the duration displayed in powerpoint is > smaller than actual duration. [...] > How can I fix this? 1. Send an email to Microsoft. 2. Try -write_xing 0 Carl Eugen From cehoyos at ag.or.at Mon Nov 24 16:03:14 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 24 Nov 2014 15:03:14 +0000 (UTC) Subject: [FFmpeg-user] config error References: Message-ID: Hugh Welles gmail.com> writes: > $ PKG_CONFIG_PATH=/tmp/ffmpeg/lib/pkgconfig ./configure --disable-yasm Do not use --disable-yasm unless you want a slow, unsupported output. yasm is a small, selfcontained binary, compile it yourself and put it somewhere in your patch (no need to "make install" if you don't want to). > touch: cannot touch ?.version?: Permission denied Remove your current working directory and check out again. (Do not use sudo or su when building FFmpeg.) Carl Eugen From cehoyos at ag.or.at Mon Nov 24 16:00:29 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Mon, 24 Nov 2014 15:00:29 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg is not building in my mac os References: Message-ID: mohanraj kandregula stellentsoft.com> writes: > /bin/ld: error: cannot open crtbegin_dynamic.o: No such file or directory Does it work if you try to compile hello world for Android? Carl Eugen From alfredo.dinapoli at gmail.com Mon Nov 24 17:04:24 2014 From: alfredo.dinapoli at gmail.com (Alfredo Di Napoli) Date: Mon, 24 Nov 2014 17:04:24 +0100 Subject: [FFmpeg-user] Mixing two audio files to left/right channels, but using a (stream-copied) video from a third input Message-ID: Hello everyone, and sorry if the title is convoluted, I couldn't find a better one. What I'm trying to accomplish is simple, in practice: Assume I have three files: - Left_Audio.mp3 - Right_Audio.mp3 - Video.mov (Extensions are not important). What I would like to achieve is to come up with a CLI command which would give me a resulting video such that: - The left audio channel of the output will be mixed from Left_Audio.mp3 - The right audio channel of the output will be mixed from Right_Audio.mp3 - The video will be a stream copy of Video.mov To be even clearer, I guess what I need is something along the lines of what's described here (covers exactly my audio needs) but with the extra step of the video copy: https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a2stereostereo (last picture of the section, where we send FL and FR of input 0 to FL of output 0 and so on and so forth. If what I'm looking for is not possible with a single pass, I would also accept a simplification: - Generate the audio mix, produce an intermediate file and then create a new output obtained with the resulting audio mix plus the aforementioned stream copy of the video. Is that possible? Thanks in advance! Alfredo Di Napoli From lou at lrcd.com Mon Nov 24 20:20:58 2014 From: lou at lrcd.com (Lou) Date: Mon, 24 Nov 2014 10:20:58 -0900 Subject: [FFmpeg-user] Mixing two audio files to left/right channels, but using a (stream-copied) video from a third input In-Reply-To: References: Message-ID: <20141124102058.61f09e32@lrcd.com> On Mon, 24 Nov 2014 17:04:24 +0100 Alfredo Di Napoli wrote: > Hello everyone, > > and sorry if the title is convoluted, I couldn't find a better one. What > I'm trying to accomplish is simple, in practice: Assume I have three files: > - Left_Audio.mp3 > - Right_Audio.mp3 > - Video.mov > > (Extensions are not important). What I would like to achieve is to come up > with a CLI command which would give me a resulting video such that: > > - The left audio channel of the output will be mixed from Left_Audio.mp3 > - The right audio channel of the output will be mixed from Right_Audio.mp3 > - The video will be a stream copy of Video.mov Since there is no info (ffmpeg console output) about your inputs I'll assume your audio inputs are both stereo, and that you only want the video stream from Video.mov while ignoring any other stream types: ffmpeg -i Left_Audio.mp3 -i Right_Audio.mp3 -i Video.mov \ -filter_complex "[0:a][1:a]amerge,pan=stereo|c0 <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> Message-ID: <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> On 2014-11-24 15:30, James Darnley wrote: > You are setting VBV options for ffmpeg and not in mencoder (and forcing > CBR at that) do I'm not surprised you get different output. I think this is not the source of the problem: I saw the same bitrate in the time of frame which I visually compared; so there should be another different option. Do you please know what? From Eric.Lovelace at msdmail.net Mon Nov 24 21:28:22 2014 From: Eric.Lovelace at msdmail.net (Eric Lovelace) Date: Mon, 24 Nov 2014 20:28:22 +0000 Subject: [FFmpeg-user] RTP Multicast doesn't play Message-ID: <5A651CA75E25F940A89A85F2A41F388D8C2C6C@EXCHANGE-DB2.msdtech.net> Hello all, I am attempting to take an rtmp stream and push it out over multicast. It appears like it is working on the server end and I can see the packets pushed out; however the video does not play in any player. If I run wireshark on the receiving machine I can see the rtp packets arriving; however the video player (ffplay, vlc, or quicktime) just sits as though it were waiting for data. No players give an error, they just go into what appears to be a ready state. Any thoughts on what is causing this or what I could do to further troubleshoot? ffserver.conf HTTPPort 8090 RTSPPort 554 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 1000 MaxClients 200 CustomLog - File /tmp/1.ffm FileMaxSize 5M Launch /usr/bin/ffmpeg -i "rtmp://172.16.10.86/pos/test" -flags:a +global_header -pixel_format yuv420p -acodec libfdk_aac -vcodec libx264 -s 640x368 Format rtp MulticastAddress 224.1.1.17 MulticastPort 5000 MulticastTTL 63 NoLoop VideoCodec libx264 VideoSize 640x388 AVOptionAudio flags +global_header AudioCodec libfdk_aac Feed 1.ffm Format status ACL allow localhost Thanks, Eric From alfredo.dinapoli at gmail.com Mon Nov 24 21:01:04 2014 From: alfredo.dinapoli at gmail.com (Alfredo Di Napoli) Date: Mon, 24 Nov 2014 21:01:04 +0100 Subject: [FFmpeg-user] Mixing two audio files to left/right channels, but using a (stream-copied) video from a third input In-Reply-To: <20141124102058.61f09e32@lrcd.com> References: <20141124102058.61f09e32@lrcd.com> Message-ID: You are exactly right Lou, that were my assumptions. Thank you very much for the help and the ready-to-use video filter! Many thanks, Alfredo Sent from my iPad > On 24/nov/2014, at 20:20, Lou wrote: > > On Mon, 24 Nov 2014 17:04:24 +0100 > Alfredo Di Napoli wrote: > >> Hello everyone, >> >> and sorry if the title is convoluted, I couldn't find a better one. What >> I'm trying to accomplish is simple, in practice: Assume I have three files: >> - Left_Audio.mp3 >> - Right_Audio.mp3 >> - Video.mov >> >> (Extensions are not important). What I would like to achieve is to come up >> with a CLI command which would give me a resulting video such that: >> >> - The left audio channel of the output will be mixed from Left_Audio.mp3 >> - The right audio channel of the output will be mixed from Right_Audio.mp3 >> - The video will be a stream copy of Video.mov > > Since there is no info (ffmpeg console output) about your inputs I'll > assume your audio inputs are both stereo, and that you only want the > video stream from Video.mov while ignoring any other stream types: > > ffmpeg -i Left_Audio.mp3 -i Right_Audio.mp3 -i Video.mov \ > -filter_complex "[0:a][1:a]amerge,pan=stereo|c0 -map 2:v -map "[aout]" -c:v copy -shortest output From yiminhe at gmail.com Tue Nov 25 07:26:03 2014 From: yiminhe at gmail.com (yiminhe at gmail.com) Date: Tue, 25 Nov 2014 14:26:03 +0800 Subject: [FFmpeg-user] duration problem of concatenated audio in powerpoint References: <201411241912178814793@gmail.com>, Message-ID: <201411251425576816863@gmail.com> -write_xing 0 could fix the problem. Thank you very much!! yiminhe at gmail.com From: Carl Eugen Hoyos Date: 2014-11-24 22:57 To: ffmpeg-user Subject: Re: [FFmpeg-user] duration problem of concatenated audio in powerpoint yiminhe gmail.com gmail.com> writes: > But if I inserted them into PPT files with Microsoft > PowerPoint, the duration displayed in powerpoint is > smaller than actual duration. [...] > How can I fix this? 1. Send an email to Microsoft. 2. Try -write_xing 0 Carl Eugen _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From belcampo at zonnet.nl Tue Nov 25 17:41:21 2014 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Tue, 25 Nov 2014 17:41:21 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> Message-ID: <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> On 24 Nov 2014, at 21:01, n32 at email.cz wrote: > On 2014-11-24 15:30, James Darnley wrote: >> You are setting VBV options for ffmpeg and not in mencoder (and forcing >> CBR at that) do I'm not surprised you get different output. > > I think this is not the source of the problem: I saw the same bitrate in the > time of frame which I visually compared; so there should be another different > option. Do you please know what? You could check if the ffmpeg and mencoder file sizes match to see if your assumption is true. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From belcampo at zonnet.nl Tue Nov 25 17:59:51 2014 From: belcampo at zonnet.nl (Henk D. Schoneveld) Date: Tue, 25 Nov 2014 17:59:51 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: References: <547340E9.3080704@gmail.com> Message-ID: On 24 Nov 2014, at 15:40, wrote: > If you mean "-b:v 1000k", ffmpeg doesn't allow me to run 2-pass without it. > Is there another way to get CBR with ffmpeg? I think you should remove the min and maxrate:v options, you don?t use them in mencoder either > > P. S. Thanks for your very quick answer. > > > On 2014-11-24 15:30, James Darnley wrote: > > "You are setting VBV options for ffmpeg and not in mencoder (and forcing > CBR at that) do I'm not surprised you get different output." > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From n32 at email.cz Tue Nov 25 18:24:32 2014 From: n32 at email.cz (n32 at email.cz) Date: Tue, 25 Nov 2014 18:24:32 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder References: <547340E9.3080704@gmail.com> Message-ID: <44S.2lWK9.6W6GFzEKESM.1KTBjG@seznam.cz> On 2014-11-25 17:41, Henk D. Schoneveld wrote: > You could check if the ffmpeg and mencoder file sizes match to see if your assumption is true. Yes, the size is almost the same (+/- 0.2 %). > I think you should remove the min and maxrate:v options, you don?t use them in mencoder either OK, I removed it, but I still cannot get the same quality. Do you know about any parameter which could cause it? As I've said the problem disappears when I raise the bitrate by 10-15 %. From ffmpeg at itvc.pl Tue Nov 25 18:25:57 2014 From: ffmpeg at itvc.pl (ffmpeg) Date: Tue, 25 Nov 2014 18:25:57 +0100 Subject: [FFmpeg-user] FFmpeg + FFserver problem - please help! Message-ID: <43b8088cf0f7adeb4ee282d7dff605b2@itvc.pl> Hello guys, When i`m trying to send stream from ffmpeg to ffserver over HTTP i have Connection reset by peer error. When i`m sending via udp to server where ffserver is (external IP), i don`t have any error (Maybe anyone can share own ffserver feed for while for tests, i will try to send my stream?) Full debug bellow, i wll be v. glad for help. Maybe anyone can share for some while own ffserver for test. ############# Client cmd: ffmpeg -i udp://@231.1.2.199:1234 -b:v 1500k -pix_fmt yuv420p -vcodec libx264 -tune zerolatency -preset ultrafast -f mpegts "http : / / xx.xx.xx.xx:8090/dupa.ffm" File /home/ffmpeg/dupa.ffm FileMaxSize 2G # This is roughly 24h of media Feed dupa.ffm Format mpegts ############# Server debug: ffserver version N-67244-g1a25c33 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 28 2014 20:12:25 with gcc 4.8.2 (GCC) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --docdir=/usr/doc/ffmpeg-2.4/html --mandir=/usr/man --disable-debug --enable-shared --disable-static --enable-pthreads --enable-libtheora --enable-libvorbis --enable-gpl --enable-version3 --enable-libx264 --enable-postproc --enable-swscale --disable-x11grab --enable-avfilter --enable-gnutls --enable-libcdio --enable-libssh --arch=x86_64 --enable-libmp3lame --enable-libx264 libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.100 / 56. 10.100 libavformat 56. 11.100 / 56. 11.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.100 / 5. 2.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Tue Oct 28 22:46:23 2014 [ffm @ 0xd53ae0]Format ffm probed with size=2048 and score=101 Tue Oct 28 22:46:23 2014 [AVIOContext @ 0xd4f940]Statistics: 4096 bytes read, 0 seeks Tue Oct 28 22:46:23 2014 FFserver started. Tue Oct 28 22:46:40 2014 xx.xx.xx.xx - - [POST] "/dupa.ffm HTTP/1.1" 200 4096[/code] Client debug: [code]ffmpeg version N-67244-g1a25c33 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 28 2014 20:12:25 with gcc 4.8.2 (GCC) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --docdir=/usr/doc/ffmpeg-2.4/html --mandir=/usr/man --disable-debug --enable-shared libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 10.100 / 56. 10.100 libavformat 56. 11.100 / 56. 11.100 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.100 / 5. 2.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Splitting the commandline. Reading option '-i' ... matched as input file with argument 'udp://@231.1.2.199:1234'. Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '1500k'. Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument 'yuv420p'. Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'. Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'. Reading option '-preset' ... matched as AVOption 'preset' with argument 'ultrafast'. Reading option '-f' ... matched as option 'f' (force format) with argument 'mpegts'. Reading option 'htt p : / / xx.xx.xx.xx:8090/dupa.ffm' ... matched as output file. Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file udp://@231.1.2.199:1234. Successfully parsed a group of options. Opening an input file: udp://@231.1.2.199:1234. [udp @ 0x10ebea0] end receive buffer size reported is 131072 [mpegts @ 0x10ed240] Format mpegts probed with size=2048 and score=100 [mpegts @ 0x10ed240] stream=0 stream_type=1b pid=7d2 prog_reg_desc= [mpegts @ 0x10ed240] stream=1 stream_type=4 pid=bba prog_reg_desc= [mpegts @ 0x10ed240] Before avformat_find_stream_info() pos: 0 bytes read:25004 seeks:0 [mpegts @ 0x10ed240] All programs have pmt, headers found [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] non-existing SPS 0 referenced in buffering period [h264 @ 0x1100480] non-existing PPS 0 referenced [h264 @ 0x1100480] decode_slice_header error [h264 @ 0x1100480] no frame! [h264 @ 0x1100480] Current profile doesn't provide more RBSP data in PPS, skipping [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] Increasing reorder buffer to 1 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 1 times [h264 @ 0x1100480] Increasing reorder buffer to 2 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] Increasing reorder buffer to 3 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture ooo [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 [h264 @ 0x1100480] no picture. [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 5 times [h264 @ 0x1100480] Current profile doesn't provide more RBSP data in PPS, skipping [h264 @ 0x1100480] unknown SEI type 128 Last message repeated 3 times [mpegts @ 0x10ed240] All info found [mpegts @ 0x10ed240] After avformat_find_stream_info() pos: 285572 bytes read:285572 seeks:0 frames:51 Input #0, mpegts, from 'udp://@231.1.2.199:1234': Duration: N/A, start: 86384.517867, bitrate: 247 kb/s Program 5090. Metadata: service_name : TV Disco service_provider:. Stream #0:0[0x7d2], 21, 1/90000: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt470bg, left), 720x576 [SAR 16:11 DAR 20:11], 1/50, 25 fps, 25 Stream #0:1[0xbba], 30, 1/90000: Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 247 kb/s Successfully opened the file. Parsing a group of options: output file htt p : / / xx.xx.xx.xx:8090/dupa.ffm. Applying option b:v (video bitrate (please use -b:v)) with argument 1500k. Applying option pix_fmt (set pixel format) with argument yuv420p. Applying option vcodec (force video codec ('copy' to copy stream)) with argument libx264. Applying option f (force format) with argument mpegts. Successfully parsed a group of options. Opening an output file: htt p :/ / xx.xx.xx.xx:8090/dupa.ffm. [http @ 0x1101c80] request: POST /dupa.ffm HTTP/1.1^M Transfer-Encoding: chunked^M User-Agent: Lavf/56.11.100^M Accept: */*^M Connection: close^M Host: xx.xx.xx.xx:8090^M Icy-MetaData: 1^M ^M Successfully opened the file. detected 8 logical cores [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'video_size' to value '720x576' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'pix_fmt' to value '0' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'time_base' to value '1/90000' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'pixel_aspect' to value '16/11' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'sws_param' to value 'flags=2' [graph 0 input from stream 0:0 @ 0x13778e0] Setting 'frame_rate' to value '25/1' [graph 0 input from stream 0:0 @ 0x13778e0] w:720 h:576 pixfmt:yuv420p tb:1/90000 fr:25/1 sar:16/11 sws_param:flags=2 [format @ 0x10ec040] compat: called with args=[yuv420p] [format @ 0x10ec040] Setting 'pix_fmts' to value 'yuv420p' [AVFilterGraph @ 0x1139700] query_formats: 4 queried, 3 merged, 0 already done, 0 delayed [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'time_base' to value '1/48000' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'sample_rate' to value '48000' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'sample_fmt' to value 's16p' [graph 1 input from stream 0:1 @ 0x1377dc0] Setting 'channel_layout' to value '0x3' [graph 1 input from stream 0:1 @ 0x1377dc0] tb:1/48000 samplefmt:s16p samplerate:48000 chlayout:0x3 [audio format for output stream 0:1 @ 0x113a520] Setting 'sample_fmts' to value 's16' [audio format for output stream 0:1 @ 0x113a520] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000' [audio format for output stream 0:1 @ 0x113a520] Setting 'channel_layouts' to value '0x4|0x3' [audio format for output stream 0:1 @ 0x113a520] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio [AVFilterGraph @ 0x10ecae0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x119e820] ch:2 chl:stereo fmt:s16p r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz [libx264 @ 0x10e4760] using SAR=16/11 [libx264 @ 0x10e4760] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 [libx264 @ 0x10e4760] profile Constrained Baseline, level 3.0 [mpegts @ 0x112b240] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'htt p :/ /xx.xx.xx.xx:8090/dupa.ffm': Metadata: encoder : Lavf56.11.100 Stream #0:0, 0, 1/90000: Video: h264 (libx264), yuv420p(left), 720x576 [SAR 16:11 DAR 20:11], 1/25, q=-1--1, 1500 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc56.10.100 libx264 Stream #0:1, 0, 1/90000: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata: encoder : Lavc56.10.100 mp2 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (mp2 (native) -> mp2 (native)) Press [q] to stop, [?] for help [h264 @ 0x1264d60] Frame num gap 10 5 [h264 @ 0x1264d60] Frame num gap 10 6 [h264 @ 0x1264d60] Frame num gap 10 7 [h264 @ 0x1264d60] Frame num gap 10 8 [h264 @ 0x1264d60] no picture. [h264 @ 0x1265de0] mmco: unref short failure [h264 @ 0x1265de0] number of reference frames (0+5) exceeds max (4; probably corrupt input), discarding one [h264 @ 0x1265de0] no picture. [h264 @ 0x1266660] no picture. *** 18 dup! [libx264 @ 0x10e4760] frame= 0 QP=25.00 NAL=3 Slice:I Poc:0 I:1620 P:0 SKIP:0 size=18259 bytes [libx264 @ 0x10e4760] frame= 1 QP=26.00 NAL=2 Slice:P Poc:2 I:0 P:0 SKIP:1620 size=86 bytes [libx264 @ 0x10e4760] frame= 2 QP=22.00 NAL=2 Slice:P Poc:4 I:5 P:649 SKIP:966 size=2547 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 3 QP=18.00 NAL=2 Slice:P Poc:6 I:27 P:1321 SKIP:272 size=11388 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 4 QP=16.00 NAL=2 Slice:P Poc:8 I:46 P:1416 SKIP:158 size=12070 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 5 QP=15.00 NAL=2 Slice:P Poc:10 I:3 P:883 SKIP:734 size=5210 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 6 QP=14.00 NAL=2 Slice:P Poc:12 I:12 P:1200 SKIP:408 size=7699 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 7 QP=13.00 NAL=2 Slice:P Poc:14 I:8 P:1360 SKIP:252 size=11766 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 8 QP=13.00 NAL=2 Slice:P Poc:16 I:0 P:18 SKIP:1602 size=169 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 9 QP=12.00 NAL=2 Slice:P Poc:18 I:19 P:1342 SKIP:259 size=9715 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 10 QP=12.00 NAL=2 Slice:P Poc:20 I:5 P:617 SKIP:998 size=2314 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 11 QP=12.00 NAL=2 Slice:P Poc:22 I:3 P:580 SKIP:1037 size=2117 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 12 QP=11.00 NAL=2 Slice:P Poc:24 I:3 P:1260 SKIP:357 size=7489 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 13 QP=11.00 NAL=2 Slice:P Poc:26 I:3 P:832 SKIP:785 size=3103 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 14 QP=10.00 NAL=2 Slice:P Poc:28 I:5 P:1398 SKIP:217 size=12018 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 15 QP=11.00 NAL=2 Slice:P Poc:30 I:2 P:386 SKIP:1232 size=1176 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 16 QP=10.00 NAL=2 Slice:P Poc:32 I:2 P:343 SKIP:1275 size=1064 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 17 QP=10.00 NAL=2 Slice:P Poc:34 I:2 P:116 SKIP:1502 size=463 bytes av_interleaved_write_frame(): Connection reset by peer [libx264 @ 0x10e4760] frame= 18 QP=9.00 NAL=2 Slice:P Poc:36 I:2 P:779 SKIP:839 size=4851 bytes av_interleaved_write_frame(): Connection reset by peer No more output streams to write to, finishing. frame= 19 fps=0.0 q=9.0 Lsize= 92kB time=00:00:00.76 bitrate= 987.5kbits/s dup=18 drop=0 ^M video:111kB audio:22kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Input file #0 (udp://@231.1.2.199:1234): Input stream #0:0 (video): 15 packets read (159198 bytes); 2 frames decoded;. Input stream #0:1 (audio): 20 packets read (15360 bytes); 20 frames decoded (23040 samples);. Total: 35 packets (174558 bytes) demuxed Output file #0 (htt p :/ / xx.xx.xx.xx:8090/dupa.ffm): Output stream #0:0 (video): 19 frames encoded; 19 packets muxed (113504 bytes);. Output stream #0:1 (audio): 20 frames encoded (23040 samples); 20 packets muxed (23040 bytes);. Total: 39 packets (136544 bytes) muxed 36 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x127e240] Statistics: 0 seeks, 16 writeouts [libx264 @ 0x10e4760] frame I:1 Avg QP:25.00 size: 18259 [libx264 @ 0x10e4760] frame P:18 Avg QP:13.61 size: 5291 [libx264 @ 0x10e4760] mb I I16..4: 100.0% 0.0% 0.0% [libx264 @ 0x10e4760] mb P I16..4: 0.5% 0.0% 0.0% P16..4: 49.7% 0.0% 0.0% 0.0% 0.0% skip:49.8% [libx264 @ 0x10e4760] final ratefactor: 20.23 [libx264 @ 0x10e4760] coded y,uvDC,uvAC intra: 45.6% 56.9% 16.4% inter: 19.6% 21.4% 5.9% [libx264 @ 0x10e4760] i16 v,h,dc,p: 29% 34% 14% 23% [libx264 @ 0x10e4760] i8c dc,h,v,p: 47% 21% 24% 8% [libx264 @ 0x10e4760] kb/s:1194.78 [AVIOContext @ 0x10ed980] Statistics: 285572 bytes read, 0 seeks Conversion failed! It`s not firewall problem. Maybe linux distribution? Its Slackware.. Thanks for help! Lucas From barsnick at gmx.net Tue Nov 25 18:31:19 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Tue, 25 Nov 2014 18:31:19 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> Message-ID: <20141125173119.GC2833@sunshine.barsnick.net> On Tue, Nov 25, 2014 at 17:41:21 +0100, Henk D. Schoneveld wrote: > On 24 Nov 2014, at 21:01, n32 at email.cz wrote: > > I think this is not the source of the problem: I saw the same bitrate in the > > time of frame which I visually compared; so there should be another different > > option. Do you please know what? > You could check if the ffmpeg and mencoder file sizes match to see if your assumption is true. >From what I can tell, libx264 embeds its encoder settings into the resulting movie, whether employed from mencoder or from ffmpeg. As discussed here: http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2014-October/023982.html those settings aren't exposed by any programs other than mediainfo, but you can also see that embedded info doing something like this: $ strings < out.avi | grep -E "x264.*core" You should at least compare the settings from mencoder and ffmpeg (and share the output here), as you/we might get an indicator as to which setting is leading to bad results. I also haven't seen the full uncut console output of either encoding command yet, please do provide. Just in case. :-) Moritz From n32 at email.cz Tue Nov 25 19:14:00 2014 From: n32 at email.cz (n32 at email.cz) Date: Tue, 25 Nov 2014 19:14:00 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> <20141125173119.GC2833@sunshine.barsnick.net> Message-ID: <4Bl.2lWKJ.6ECj653Lyyb.1KTCRe@seznam.cz> On 2014-11-25 18:31, Moritz Barsnick wrote: "From what I can tell, libx264 embeds its encoder settings into the resulting movie, whether employed from mencoder or from ffmpeg. As discussed here: http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2014-October/023982.html those settings aren't exposed by any programs other than mediainfo, but you can also see that embedded info doing something like this: $ strings < out.avi | grep -E "x264.*core"" You read my thoughts: I just wanted to ask how I could get the parameters from mencoder (ffmpeg has it on its output). I should have known that it's as easy as plain text in the files. "" ? "You should at least compare the settings from mencoder and ffmpeg (and share the output here), as you/we might get an indicator as to which setting is leading to bad results." The output is surprisingly the same: x264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www. videolan.org/x264.html - options: cabac=1 ref=15 deblock=1:-1:-1 analyse=0x 3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_ offset=-3 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_ adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_ min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=2pass mbtree=1 bitrate =1000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur= 0.5 ip_ratio=1.40 aq=1:1.00 "I also haven't seen the full uncut console output of either encoding command yet, please do provide. Just in case. :-) " It's very long, especially from mencoder, so I dropped some insignificant lines (marked as {more times}), which are similar to previous, so hopefully nobody feels flooded; and changed \r to \n to read it easier. "" -------------- next part -------------- A non-text attachment was scrubbed... Name: ffmpeg.txt.xz Type: application/x-xz Size: 2732 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: mencoder.txt.xz Type: application/x-xz Size: 2108 bytes Desc: not available URL: From manuel_songokuh at yahoo.it Tue Nov 25 22:26:25 2014 From: manuel_songokuh at yahoo.it (manuel_songokuh at yahoo.it) Date: Tue, 25 Nov 2014 21:26:25 +0000 (UTC) Subject: [FFmpeg-user] ffmpeg SRT FONT SIZE Message-ID: <669011512.1129128.1416950785972.JavaMail.yahoo@jws11128.mail.ir2.yahoo.com> hello i need your help for ffmpeg: i want to add srt in video but i want chose the font dejevu.tiff and size so how i add parameter ffmpeg? "ffmpeg -i video.avi -vf subtitles=subtitle.srt out.avi" From maziar.mehrabi at gmail.com Wed Nov 26 10:06:29 2014 From: maziar.mehrabi at gmail.com (Maziar Mehrabi) Date: Wed, 26 Nov 2014 11:06:29 +0200 Subject: [FFmpeg-user] Problem with instances of ffplay Message-ID: Hello everyone, I'm not sure whether this problems relates to ffplay, ffmpeg or my system's OS. I would appreciate any kind of comment, suggestion or solution. here is the scenario: I use ffmpeg to stream video files to UDP and then I use ffplay to play these streams (all on the same machine using ffmpeg -re -i input1 -f mpegts udp://localhost:port1 and ffplay udp://localhost:port1 and similar commands for other streams). I first run the ffplay commands and they seem to be waiting (or listening) to udp port for incoming streams, then I start streaming with ffmpeg. This works fine in most of the cases and if I have for example two streams of video then two instances of ffplay will start playing these videos in separate windows right after the streaming starts. The problem arises in some cases when I start ffmpeg streams shortly one after another then the system does not open two separate instances of ffplay in separate windows, instead both video streams get merged and played in one ffplay instance and in one window. Although I dedicate two different ports for each stream, but one of the players seems to be still on the waiting mode showing the following in the console while the other one plays both streams. nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 This problem appears more frequently when instantiations are somehow automated (e.g. using bash). But also appears when I switch between console tabs rapidly and run ffmpeg instances. What do you think about this? Is it related to streaming with ffmpeg? or ffplay binding to udp ports? or OS's inabilities to dedicate resources for each instance of ffplay? How can I track it down and narrow this scope? My ffmpeg version: ffmpeg version 2.3.git. Does it need upgrade? I was simply waiting for 2.5 to be released. OS: Ubuntu 14.04 LTS Thanks a lot, Maziar From xanadu at apost.plala.or.jp Wed Nov 26 10:12:02 2014 From: xanadu at apost.plala.or.jp (Kimio Miyamura) Date: Wed, 26 Nov 2014 18:12:02 +0900 Subject: [FFmpeg-user] ffmpeg SRT FONT SIZE In-Reply-To: <669011512.1129128.1416950785972.JavaMail.yahoo@jws11128.mail.ir2.yahoo.com> References: <669011512.1129128.1416950785972.JavaMail.yahoo@jws11128.mail.ir2.yahoo.com> Message-ID: 2014/11/26 6:26, ffmpeg.org> ffmpeg.org> > i need your help for ffmpeg: i want to add srt in video but i want chose the font dejevu.tiff and size so how i add parameter ffmpeg? > "ffmpeg -i video.avi -vf subtitles=subtitle.srt out.avi" I don't think ffmpeg have parameter for srt or ass styling. (If my thought is incorrect, please let me know, experts) In my thought, srt does not have style information. If you want to use srt with styling, you have to provide a srt.style file. But I don't know ffmpeg can handle srt.style file with srt file. Another thought, ass subtitle can handle styling information in it's file. You can convert srt subtitle into ass subtitle with the following command. $ ffmpeg -i subtitle.srt subtitle.ass After conversion, you can add any styling information into ass file using your favorite text editor. see this URL for your reference: http://stackoverflow.com/questions/21363334/how-to-add-font-size-in-subtitles-in-ffmpeg-video-filter // Miya From barsnick at gmx.net Wed Nov 26 12:29:49 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Wed, 26 Nov 2014 12:29:49 +0100 Subject: [FFmpeg-user] Worse quality than mencoder In-Reply-To: <4Bl.2lWKJ.6ECj653Lyyb.1KTCRe@seznam.cz> References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> <20141125173119.GC2833@sunshine.barsnick.net> <4Bl.2lWKJ.6ECj653Lyyb.1KTCRe@seznam.cz> Message-ID: <20141126112949.GA15873@sunshine.barsnick.net> Hi n32, On Tue, Nov 25, 2014 at 19:14:00 +0100, n32 at email.cz wrote: > The output is surprisingly the same: Are you absolutely sure? > It's very long, especially from mencoder, so I dropped some insignificant > lines (marked as {more times}), which are similar to previous, so hopefully > nobody feels flooded; and changed \r to \n to read it easier. No problem, that's fine. You're doing two-pass, right? If that is why these logs each contain two outputs from the encoders, then you're doing something significantly wrong: mencoder.txt:x264 [info]: profile Main, level 3.0 mencoder.txt:x264 [info]: profile High, level 4.0 ffmpeg.txt:[libx264 @ 0x63c9c0] profile Main, level 4.0 ffmpeg.txt:[libx264 @ 0x63c9c0] profile High, level 4.0 The libx264 settings from the first pass and second pass differ significantly from each other - and in a different manner in ffmpeg and mencoder. Only the second pass settings seem identical, which may be why you see the same embedded in the file. AFAIU you need to use identical codec options in both passes, and it seems you're not doing that. That _may_ be the cause for your observations. Moritz From n32 at email.cz Wed Nov 26 14:36:52 2014 From: n32 at email.cz (Jan Sever) Date: Wed, 26 Nov 2014 14:36:52 +0100 (CET) Subject: [FFmpeg-user] Worse quality than mencoder [SOLVED] References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> <20141125173119.GC2833@sunshine.barsnick.net> <4Bl.2lWKJ.6ECj653Lyyb.1KTCRe@seznam.cz> <20141126112949.GA15873@sunshine.barsnick.net> Message-ID: <6E7.2lWJX.2RGpZ0ZoN5N.1KTTTq@seznam.cz> On 2014-11-26 12:30, Moritz Barsnick wrote: > Are you absolutely sure? I'm sorry, they're the same only for the second pass: that was the problem. > You're doing two-pass, right? If that is why these logs each contain > two outputs from the encoders, then you're doing something > significantly wrong: > > mencoder.txt:x264 [info]: profile Main, level 3.0 > mencoder.txt:x264 [info]: profile High, level 4.0 > ffmpeg.txt:[libx264 @ 0x63c9c0] profile Main, level 4.0 > ffmpeg.txt:[libx264 @ 0x63c9c0] profile High, level 4.0 Level was not the problem. ? > The libx264 settings from the first pass and second pass differ > significantly from each other - and in a different manner in ffmpeg and > mencoder. Only the second pass settings seem identical, which may be > why you see the same embedded in the file.? Yes, you're right, that was the problem. The encoders set some parameters for quicker first pass (I knew that), but what I didn't know that ffmpeg doesn't set fast_pskip=1 and ref=1 while mencoder does. > AFAIU you need to use identical codec options in both passes, and it > seems you're not doing that. That _may_ be the cause for your > observations. In fact, it is. I compared the achieved quality after changing the two mentioned parameters and it looks almost the same now. So marking as SOLVED. Thanks everybody for your precious help. Jan Sever From dpeterson478 at gmail.com Wed Nov 26 14:46:48 2014 From: dpeterson478 at gmail.com (David Peterson) Date: Wed, 26 Nov 2014 08:46:48 -0500 Subject: [FFmpeg-user] Problem with GDIGRAB device on Windows 7 In-Reply-To: References: <54320BF2.4030302@xopnetworks.com> Message-ID: Is there a way to use ffmpeg to "grab" an IOS screen" Dave On Sat, Nov 22, 2014 at 2:22 PM, Roger Pack wrote: > On Sun, Oct 5, 2014 at 9:26 PM, Yan Brenman > wrote: > > > Hello ffmpeg/gdigrab gurus! > > > > We are using "gdigrab" device on Windows to share/stream video for a > > specific application window (identified by a title as "gdigrab" > requires). > > Everything works great on Window 8.x - actual application window (and not > > the region on the desktop occupied by the application window) > > is getting shared and successfully played by the player on the receiving > > side. Which means that even if shared window is overlapped by the > > other window on the source - still the application window is getting > > played. > > Unfortunately, as we discovered, "gdigrab" doesn't work correctly on > > Window 7. And instead window sharing it actually does the region sharing. > > Which pretty much means that if shared window is being overlapped by the > > other window on the desktop - that's what getting played on the > > receiving side. > > Just for reference - we tried to do window sharing with > > screen-capture-recorder and unfortunately got exactly the same results. > > > > I would greatly appreciate any help/advise anybody can provide on the > > subject. And please let me know if there is any other information I can > > provide. > > > > Unfortunately there's no easy "fix" unless someone were to implement the > "PrintWindow" API. > Patches welcome, for sure. > -roger- > _______________________________________________ > 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 barsnick at gmx.net Wed Nov 26 15:37:00 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Wed, 26 Nov 2014 15:37:00 +0100 Subject: [FFmpeg-user] Worse quality than mencoder [SOLVED] In-Reply-To: <6E7.2lWJX.2RGpZ0ZoN5N.1KTTTq@seznam.cz> References: <547340E9.3080704@gmail.com> <5473446C.2050608@thelounge.net> <1kb.2lWKv.1PCRibMDuXh.1KSuwg@seznam.cz> <040538D1-30B4-47F5-A804-8C84B7525BC5@zonnet.nl> <20141125173119.GC2833@sunshine.barsnick.net> <4Bl.2lWKJ.6ECj653Lyyb.1KTCRe@seznam.cz> <20141126112949.GA15873@sunshine.barsnick.net> <6E7.2lWJX.2RGpZ0ZoN5N.1KTTTq@seznam.cz> Message-ID: <20141126143700.GB15873@sunshine.barsnick.net> On Wed, Nov 26, 2014 at 14:36:52 +0100, Jan Sever wrote: > The encoders set some parameters for quicker first pass (I knew > that), I wasn't aware of that, but didn't actually find any source indicating to having to use the same parameters either (which is why I added "AFAIU"). > but what I didn't know that ffmpeg doesn't set fast_pskip=1 and ref=1 > while mencoder does. Interesting. > In fact, it is. I compared the achieved quality after changing the two mentioned > parameters and it looks almost the same now. So marking as SOLVED. Thanks > everybody for your precious help. Excellent, thanks for letting us know. My guesswork actually provided some value. :-) Cheers, Moritz From minhkhoi15 at yahoo.com Wed Nov 26 04:47:40 2014 From: minhkhoi15 at yahoo.com (likelook) Date: Tue, 25 Nov 2014 19:47:40 -0800 (PST) Subject: [FFmpeg-user] Use build_libstagefright in ffmpeg to build file .so for android In-Reply-To: References: Message-ID: <1416973660005-4668282.post@n4.nabble.com> i'm a new user of ffmpeg, please help me build a new libstagefright.so for a x86 android 4.2.2 base, i use android intel architecture r1-ia3 4.2.2 on acer w700, almost thing is ok but the video hardware accelerator. Thank u very much! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Use-build-libstagefright-in-ffmpeg-to-build-file-so-for-android-tp4654772p4668282.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rens at onlinemedia.nl Thu Nov 27 00:41:14 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Thu, 27 Nov 2014 00:41:14 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <3508177.241417045170176.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <24107656.261417045274462.JavaMail.root@webmail.onlinemedia.nl> Hi, with the latest versions of ffmpeg converting a SONY AVC100CBG_1920_1080_H422IP at L41 to Quicktime Movie the resulting movie plays but display remains green in the quicktime player, both mplayer and ffplay will play the resulting mov. The same command tested with ffmbc-7.1 works fine. macbook-2:~ macbook$ ffmpeg -i XAVC_1920 at 1080.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 TEST.MOV ffmpeg version 2.4.3 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 17 2014 11:26:31 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 --enable-libx265 --enable-libopenjpeg --enable-gpl --enable-nonfree --enable-pthreads --enable-avfilter --enable-libfreetype --enable-libass --enable-shared libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [mxf @ 0x7fd25301a000] Stream #0: not enough frames to estimate rate; consider increasing probesize Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono Guessed Channel Layout for Input Stream #0.3 : mono Guessed Channel Layout for Input Stream #0.4 : mono Guessed Channel Layout for Input Stream #0.5 : mono Guessed Channel Layout for Input Stream #0.6 : mono Guessed Channel Layout for Input Stream #0.7 : mono Guessed Channel Layout for Input Stream #0.8 : mono Input #0, mxf, from 'XAVC_1920 at 1080.MXF': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 timecode : 17:30:04:08 Duration: 00:00:33.88, start: 0.000000, bitrate: 124027 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:9: Data: none Metadata: data_type : vbi_vanc_smpte_436M File 'TEST.MOV' already exists. Overwrite ? [y/N] y Output #0, mov, to 'TEST.MOV': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 timecode : 17:30:04:08 encoder : Lavf56.4.101 Stream #0:0: Video: h264 (ai12 / 0x32316961), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 12800 tbn, 25 tbc Stream #0:1: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 847 fps= 42 q=-1.0 Lsize= 480050kB time=00:00:33.88 bitrate=116073.4kbits/s video:470508kB audio:9529kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.002571% macbook-2:~ macbook$ Any ideas how to fix this Regards Rens Dijkshoorn From cehoyos at ag.or.at Thu Nov 27 00:46:13 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Wed, 26 Nov 2014 23:46:13 +0000 (UTC) Subject: [FFmpeg-user] XAVC to Quicktime MOV References: <3508177.241417045170176.JavaMail.root@webmail.onlinemedia.nl> <24107656.261417045274462.JavaMail.root@webmail.onlinemedia.nl> Message-ID: Rens Dijkshoorn onlinemedia.nl> writes: > with the latest versions of ffmpeg converting a SONY > AVC100CBG_1920_1080_H422IP L41 to Quicktime Movie > the resulting movie plays but display remains green in > the quicktime player, both mplayer and ffplay will > play the resulting mov. > > The same command tested with ffmbc-7.1 works fine. Please provide (three) sample files. > ffmpeg version 2.4.3 Copyright (c) 2000-2014 the FFmpeg developers This is not the latest version;-( Carl Eugen From rogerdpack2 at gmail.com Thu Nov 27 02:51:50 2014 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 26 Nov 2014 18:51:50 -0700 Subject: [FFmpeg-user] Problem with GDIGRAB device on Windows 7 In-Reply-To: References: <54320BF2.4030302@xopnetworks.com> Message-ID: On Wed, Nov 26, 2014 at 6:46 AM, David Peterson wrote: > Is there a way to use ffmpeg to "grab" an IOS screen" > I doubt it. Maybe OS X but I'm not even sure there. > > Dave > > On Sat, Nov 22, 2014 at 2:22 PM, Roger Pack wrote: > > > On Sun, Oct 5, 2014 at 9:26 PM, Yan Brenman > > wrote: > > > > > Hello ffmpeg/gdigrab gurus! > > > > > > We are using "gdigrab" device on Windows to share/stream video for a > > > specific application window (identified by a title as "gdigrab" > > requires). > > > Everything works great on Window 8.x - actual application window (and > not > > > the region on the desktop occupied by the application window) > > > is getting shared and successfully played by the player on the > receiving > > > side. Which means that even if shared window is overlapped by the > > > other window on the source - still the application window is getting > > > played. > > > Unfortunately, as we discovered, "gdigrab" doesn't work correctly on > > > Window 7. And instead window sharing it actually does the region > sharing. > > > Which pretty much means that if shared window is being overlapped by > the > > > other window on the desktop - that's what getting played on the > > > receiving side. > > > Just for reference - we tried to do window sharing with > > > screen-capture-recorder and unfortunately got exactly the same results. > > > > > > I would greatly appreciate any help/advise anybody can provide on the > > > subject. And please let me know if there is any other information I can > > > provide. > > > > > > > Unfortunately there's no easy "fix" unless someone were to implement the > > "PrintWindow" API. > > Patches welcome, for sure. > > -roger- > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From hanqz at mail.ustc.edu.cn Thu Nov 27 06:22:56 2014 From: hanqz at mail.ustc.edu.cn (hanqz at mail.ustc.edu.cn) Date: Thu, 27 Nov 2014 13:22:56 +0800 (GMT+08:00) Subject: [FFmpeg-user] About compile fail on ffmpeg Message-ID: <4b968865.5b08.149efb3c4aa.Coremail.hanqz@mail.ustc.edu.cn> Hello,I'm using the Cygwin and NDK r5b to compile the ffmpeg , but it always fail,and the tips are : /toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc is unable to create an executable file. the config.log is added. waitting for your reply . thank you -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: config.log URL: From battistel at gmail.com Thu Nov 27 10:21:23 2014 From: battistel at gmail.com (Massimo Battistel) Date: Thu, 27 Nov 2014 10:21:23 +0100 Subject: [FFmpeg-user] Relevant image degradation after 6-8 hour of encoding Message-ID: Hello, I'm trying to transcode a live stream and I faced the issue in object. Picture looks much more "pixellated" especially on I-frames, giving the video a "pulsing" effect. I've attached a screenshot of image quality after ~1h and after ~8h. My test stream was a single video in loop. This is the command line I used: ffmpeg -probesize 20M -channel_layout stereo -i -vcodec mpeg2video -pix_fmt yuv420p -aspect 16:9 -b:v 1200k -minrate 1200k -maxrate 1200k -trellis 2 -g 24 -bf 2 -acodec mp2 -ac 2 -ar 48000 -b:a 128k -streamid 0:70 -streamid 1:71 -f mpegts -flush_packets 0 udp:// 127.0.0.1:1500?pkt_size=1316 ffmpeg version N-66232-g5e3da25 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 10 2014 22:06:30 with gcc 4.8.3 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 [h264 @ 02660fc0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] no frame! [h264 @ 02660fc0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] non-existing PPS 0 referenced [h264 @ 02660fc0] decode_slice_header error [h264 @ 02660fc0] no frame! Input #0, mpegts, from 'udp://127.0.0.1:1400': Duration: N/A, start: 152.956000, bitrate: 122 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 122 kb/s [mpeg2video @ 049f0740] Automatically choosing VBV buffer size of 224 kbyte [mpeg2video @ 049f0740] Warning vbv_delay will be set to 0xFFFF (=VBR) as the specified vbv buffer is too large for the given bitrate! Output #0, mpegts, to 'udp://127.0.0.1:1500?pkt_size=1316': Metadata: encoder : Lavf56.4.101 Stream #0:0: Video: mpeg2video, yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 1200 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc56.1.100 mpeg2video Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s Metadata: encoder : Lavc56.1.100 mp2 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg2video (native)) Stream #0:1 -> #0:1 (aac (native) -> mp2 (native)) Press [q] to stop, [?] for help I would like to know which encoding parameters change over time and how to control them (if possible.) thanks, MB -------------- next part -------------- A non-text attachment was scrubbed... Name: degradation_8h.jpg Type: image/jpeg Size: 8881 bytes Desc: not available URL: From barsnick at gmx.net Thu Nov 27 11:47:47 2014 From: barsnick at gmx.net (Moritz Barsnick) Date: Thu, 27 Nov 2014 11:47:47 +0100 Subject: [FFmpeg-user] About compile fail on ffmpeg In-Reply-To: <4b968865.5b08.149efb3c4aa.Coremail.hanqz@mail.ustc.edu.cn> References: <4b968865.5b08.149efb3c4aa.Coremail.hanqz@mail.ustc.edu.cn> Message-ID: <20141127104747.GC17494@sunshine.barsnick.net> On Thu, Nov 27, 2014 at 13:22:56 +0800, hanqz at mail.ustc.edu.cn wrote: > Hello,I'm using the Cygwin and NDK r5b to compile the ffmpeg , but it always fail,and the tips are : > /toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc is unable to create an executable file. Of course it's unable, because: > /toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi-gcc: > No such file or directory So this must be wrong: > --cross-prefix=/toolchains/arm-linux-androideabi-4.4.3/prebuilt/windows/bin/arm-linux-androideabi- Moritz From rens at onlinemedia.nl Thu Nov 27 20:11:23 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Thu, 27 Nov 2014 20:11:23 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <28632321.1051417114861288.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <31492935.1071417115483551.JavaMail.root@webmail.onlinemedia.nl> Rens Dijkshoorn onlinemedia.nl> writes: > with the latest versions of ffmpeg converting a SONY > AVC100CBG_1920_1080_H422IP L41 to Quicktime Movie > the resulting movie plays but display remains green in > the quicktime player, both mplayer and ffplay will > play the resulting mov. > > The same command tested with ffmbc-7.1 works fine. Please provide (three) sample files. > ffmpeg version 2.4.3 Copyright (c) 2000-2014 the FFmpeg developers This is not the latest version;-( Just to be sure I checked today with a fresh build(s) from git and did some tests to see if i could come up with some more relevant information. ffmpeg -i AVC100CBG_1920_1080_H422IP at L41.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 FFMPEG.MOV ffmpeg version N-68057-g92fa1d9 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 27 2014 19:29:47 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --arch=x86_64 libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.100 / 56. 15.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 The resulting file is now playing fine. When building with my standard config configuration: --arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 --enable-libx265 --enable-libopenjpeg --enable-gpl --enable-nonfree --enable-pthreads --enable-avfilter --enable-libfreetype --enable-libass --enable-shared The greenscreen when playingback in the quicktimeplayer has returned so is it safe to say that its related to one of the external library's ? Tested with a static build found on the internet ffmpeg version 2.4.1-tessus Copyright (c) 2000-2014 the FFmpeg developers configuration: --cc=/usr/bin/clang --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm --extra-version=tessus --disable-shared --enable-static --disable-ffplay --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libx265 --enable-libxvid --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libxavs --enable-libsoxr --enable-libwavpack --enable-version3 --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig --enable-libfreetype --enable-libass --enable-libbluray --enable-filters --disable-indev=qtkit --enable-runtime-cpudetect libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat 56. 4.101 / 56. 4.101 libavdevice 56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 0.100 / 53. 0.100 Converted to mov with the same command ffmpeg -i AVC100CBG_1920_1080_H422IP at L41.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 FFMPEG.MOV This config give's the green screen when playing in the Quicktimeplayer, mplayer is fine. For final test I build with ffmpeg version N-68057-g92fa1d9 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 27 2014 19:54:47 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --arch=x86_64 --enable-libx264 --enable-gpl --enable-nonfree libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.100 / 56. 15.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Now the resulting mov file plays fine. When converting there is a difference in printout for the working version Input #0, mxf, from '/Volumes/Media/XAVC/F55/B001C001_141122B0.MXF': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 material_package_uid: 71673eb8-8369-05d3-0800-4602026b87d2 timecode : 17:30:04:08 Duration: 00:00:33.88, start: 0.000000, bitrate: 124027 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:9: Data: none Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 data_type : vbi_vanc_smpte_436M File 'FFMPEG.MOV' already exists. Overwrite ? [y/N] y Output #0, mov, to 'FFMPEG.MOV': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 material_package_uid: 71673eb8-8369-05d3-0800-4602026b87d2 timecode : 17:30:04:08 encoder : Lavf56.15.100 Stream #0:0: Video: h264 (ai12 / 0x32316961), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 12800 tbn, 25 tbc Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:1: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream #0:2: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Metadata: file_package_uid: 72673eb8-8369-05d3-0800-4602026b87d2 Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 847 fps=155 q=-1.0 Lsize= 480050kB time=00:00:33.88 bitrate=116073.4kbits/s video:470508kB audio:9529kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.002553% for the file that playback fine to Input #0, mxf, from '/Volumes/Media/XAVC/F55/B001C001_141122B0.MXF': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 timecode : 17:30:04:08 Duration: 00:00:33.88, start: 0.000000, bitrate: 124027 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:9: Data: none Metadata: data_type : vbi_vanc_smpte_436M File 'FFMPEG.MOV' already exists. Overwrite ? [y/N] y Output #0, mov, to 'FFMPEG.MOV': Metadata: uid : 482f3ada-726b-11e4-a6b2-0800466b87d2 generation_uid : 482f3ae4-726b-11e4-a881-0800466b87d2 company_name : Sony product_name : Mem product_version : 2.00 product_uid : cede1104-8280-11de-8a39-08004678031c modification_date: 2014-11-22 17:16:22 timecode : 17:30:04:08 encoder : Lavf56.4.101 Stream #0:0: Video: h264 (ai12 / 0x32316961), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 12800 tbn, 25 tbc Stream #0:1: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24 bit), 1152 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Stream #0:2 -> #0:2 (copy) Press [q] to stop, [?] for help frame= 847 fps=186 q=-1.0 Lsize= 480050kB time=00:00:33.88 bitrate=116073.4kbits/s video:470508kB audio:9529kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.002571% this one for the quicktime that won't playpack properly, no idea if this could be relevant. Regards Rens From mboufleur at gmail.com Thu Nov 27 20:33:00 2014 From: mboufleur at gmail.com (Marcelo Boufleur) Date: Thu, 27 Nov 2014 17:33:00 -0200 Subject: [FFmpeg-user] H.264 Quicktime encoded file stutters Message-ID: Hello everyone, I am still quite new to ffmpeg, but I'm already using it to convert some content - mostly to Quicktime format. During one of these attempts, I wanted to convert a JPEG2000 MXF file from a DCP into Quicktime file with H.264 codec, that would mimic the same experience as if the file would be encoded with Quicktime player itself (or maybe Compressor). At the end, this Quicktime file would be played using a windows PC with the Quicktime Player, so my goal was to create the exact same experience. I created a small sample from Quicktime player and then compared it to the same file created using ffmpeg. At first, I was able to adjust all settings in ffmpeg (CAVLC, ref frames, b-frames, resolution, bitrate, etc) to mimic all configurations used by the H.264 file created by Quicktime player. The info was provided by Mediainfo, and although not all parameters for H.264 were there, the ffmpeg generated file seemed to equalize all settings. Unfortunately, when played in Quicktime player, the final ffmpeg H.264 file would stutter. The original movie frame rate had 24fps, and Quicktime would play only 16fps on average. This is the command line I am currently using: ffmpeg -r 24 -i Input.mxf -vf "scale=1920:1038,pad=1920:1080:0:21" -pix_fmt yuv420p -c:v libx264 -r 24 -profile:v main -level 4.1 -preset faster -coder 0 -crf 21 -x264opts keyint=60:bframes=1:ref=2:qpmin=4:b-pyramid=0 Output.mov And this is the console output: ffmpeg version N-67501-g064a237 Copyright (c) 2000-2014 the FFmpeg developers built on Nov 9 2014 22:52:31 with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 11.100 / 54. 11.100 libavcodec 56. 12.100 / 56. 12.100 libavformat 56. 12.103 / 56. 12.103 libavdevice 56. 2.100 / 56. 2.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [mxf @ 0000000004e833a0] "OPAtom" with 2 ECs - assuming OP1a Input #0, mxf, from 'MORTDECAI_TLR_S_QBP-XX_ENC-reel-1-jp2k.mxf': Metadata: uid : 5831cb6f-9270-4545-a789-5b79221e0074 generation_uid : 9583e162-55e1-46ac-b111-4ee032c5f74e company_name : QubeCinema, Inc. product_name : QubeMaster Pro product_version : 2.5 product_uid : a6d3ea56-8155-4dfc-86f6-664b12671427 modification_date: 1969-08-14 04:58:00 application_platform: win32 timecode : 00:00:00:00 Duration: 00:01:38.42, start: 0.000000, bitrate: 91072 kb/s Stream #0:0: Video: jpeg2000 (JPEG 2000 digital cinema 2K), xyz12le, 2048x858, 24 tbr, 24 tbn, 24 tbc [libx264 @ 0000000002def280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 [libx264 @ 0000000002def280] profile Main, level 4.1 [libx264 @ 0000000002def280] 264 - core 142 r2479 dd79a61 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=0 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=4 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=36 lookahead_threads=8 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=1 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=60 keyint_min=6 scenecut=40 intra_refresh=0 rc_lookahead=20 rc=crf mbtree=1 crf=21.0 qcomp=0.60 qpmin=4 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mov, to 'Test.mov': Metadata: uid : 5831cb6f-9270-4545-a789-5b79221e0074 generation_uid : 9583e162-55e1-46ac-b111-4ee032c5f74e company_name : QubeCinema, Inc. product_name : QubeMaster Pro product_version : 2.5 product_uid : a6d3ea56-8155-4dfc-86f6-664b12671427 modification_date: 1969-08-14 04:58:00 application_platform: win32 timecode : 00:00:00:00 encoder : Lavf56.12.103 Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 1920x1080, q=-1--1, 24 fps, 12288 tbn, 24 tbc Metadata: encoder : Lavc56.12.100 libx264 Stream mapping: Stream #0:0 -> #0:0 (jpeg2000 (native) -> h264 (libx264)) Press [q] to stop, [?] for help frame= 2362 fps=9.4 q=-1.0 Lsize= 52668kB time=00:01:38.37 bitrate=4385.8kbits/s video:52642kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.049134% [libx264 @ 0000000002def280] frame I:73 Avg QP:17.39 size:139477 [libx264 @ 0000000002def280] frame P:1431 Avg QP:19.90 size: 25280 [libx264 @ 0000000002def280] frame B:858 Avg QP:20.86 size: 8797 [libx264 @ 0000000002def280] consecutive B-frames: 27.3% 72.7% [libx264 @ 0000000002def280] mb I I16..4: 51.5% 0.0% 48.5% [libx264 @ 0000000002def280] mb P I16..4: 15.0% 0.0% 2.6% P16..4: 29.8% 6.5% 1.2% 0.0% 0.0% skip:44.9% [libx264 @ 0000000002def280] mb B I16..4: 1.6% 0.0% 0.3% B16..8: 13.1% 3.0% 0.1% direct:11.6% skip:70.4% L0:38.5% L1:54.3% BI: 7.1% [libx264 @ 0000000002def280] coded y,uvDC,uvAC intra: 34.6% 64.0% 28.7% inter: 8.6% 22.5% 1.1% [libx264 @ 0000000002def280] i16 v,h,dc,p: 49% 21% 17% 13% [libx264 @ 0000000002def280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 18% 15% 7% 8% 8% 7% 7% 5% [libx264 @ 0000000002def280] i8c dc,h,v,p: 55% 18% 21% 6% [libx264 @ 0000000002def280] Weighted P-Frames: Y:10.3% UV:5.0% [libx264 @ 0000000002def280] ref P L0: 74.0% 26.0% [libx264 @ 0000000002def280] kb/s:4381.77 Any ideas what may be wrong? From cehoyos at ag.or.at Thu Nov 27 20:36:59 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Thu, 27 Nov 2014 19:36:59 +0000 (UTC) Subject: [FFmpeg-user] XAVC to Quicktime MOV References: <28632321.1051417114861288.JavaMail.root@webmail.onlinemedia.nl> <31492935.1071417115483551.JavaMail.root@webmail.onlinemedia.nl> Message-ID: Rens Dijkshoorn onlinemedia.nl> writes: > The resulting file is now playing fine. Is --enable-libx264 responsible? > configuration: The following options have no effect, please remove them, they often cause confusion: > --arch=x86_64 > --enable-pthreads > --enable-avfilter Carl Eugen From mboufleur at gmail.com Thu Nov 27 21:24:38 2014 From: mboufleur at gmail.com (Marcelo Boufleur) Date: Thu, 27 Nov 2014 18:24:38 -0200 Subject: [FFmpeg-user] H.264 Quicktime encoded file stutters In-Reply-To: References: Message-ID: After some more testing, it seems to me that the problem does not lie in H.264 encoding, but in Quicktime container itself. I made another test, and encoded the MXF into Quicktime ProRes, which played fine in Quicktime Player. No stutter, no issues, it was ok. Then, I grabbed this ffmpeg-ProRes file and fed it to Compressor in order to create a final H.264 file. The resulting file ended up having a stutter just like the original H.264 ffmpeg file. When played side by side, the H.264 file from ffmpeg and the H.264 file from ffmpeg>compressor seemed very similar. The main difference was that the original H.264 ffmpeg file had all frames (i.e. when moving frame by frame in Quicktime player, it was possible to see that all frames were indeed in the file), and the H.264 file converted by Compressor had duplicated frames that indeed caused the stutter effect when played. So there may be something or some timebase in the final Quicktime container that my get messed up in Apple software and/or workflow (the "tbn" perhaps?). 2014-11-27 17:33 GMT-02:00 Marcelo Boufleur : > Hello everyone, > > I am still quite new to ffmpeg, but I'm already using it to convert some > content - mostly to Quicktime format. > > During one of these attempts, I wanted to convert a JPEG2000 MXF file from > a DCP into Quicktime file with H.264 codec, that would mimic the same > experience as if the file would be encoded with Quicktime player itself (or > maybe Compressor). > > At the end, this Quicktime file would be played using a windows PC with > the Quicktime Player, so my goal was to create the exact same experience. > > I created a small sample from Quicktime player and then compared it to the > same file created using ffmpeg. At first, I was able to adjust all settings > in ffmpeg (CAVLC, ref frames, b-frames, resolution, bitrate, etc) to mimic > all configurations used by the H.264 file created by Quicktime player. > > The info was provided by Mediainfo, and although not all parameters for > H.264 were there, the ffmpeg generated file seemed to equalize all settings. > > Unfortunately, when played in Quicktime player, the final ffmpeg H.264 > file would stutter. The original movie frame rate had 24fps, and Quicktime > would play only 16fps on average. > > This is the command line I am currently using: > > ffmpeg -r 24 -i Input.mxf -vf "scale=1920:1038,pad=1920:1080:0:21" > -pix_fmt yuv420p -c:v libx264 -r 24 -profile:v main -level 4.1 -preset > faster -coder 0 -crf 21 -x264opts > keyint=60:bframes=1:ref=2:qpmin=4:b-pyramid=0 Output.mov > > And this is the console output: > > ffmpeg version N-67501-g064a237 Copyright (c) 2000-2014 the FFmpeg > developers > built on Nov 9 2014 22:52:31 with gcc 4.9.2 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads > --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r > --enable-gnutls --enable-iconv --enable-libass --enable-libbluray > --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme > --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg > --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr > --enable-libspeex --enable-libtheora --enable-libtwolame > --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc > --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp > --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid > --enable-zlib > libavutil 54. 11.100 / 54. 11.100 > libavcodec 56. 12.100 / 56. 12.100 > libavformat 56. 12.103 / 56. 12.103 > libavdevice 56. 2.100 / 56. 2.100 > libavfilter 5. 2.103 / 5. 2.103 > libswscale 3. 1.101 / 3. 1.101 > libswresample 1. 1.100 / 1. 1.100 > libpostproc 53. 3.100 / 53. 3.100 > [mxf @ 0000000004e833a0] "OPAtom" with 2 ECs - assuming OP1a > Input #0, mxf, from 'MORTDECAI_TLR_S_QBP-XX_ENC-reel-1-jp2k.mxf': > Metadata: > uid : 5831cb6f-9270-4545-a789-5b79221e0074 > generation_uid : 9583e162-55e1-46ac-b111-4ee032c5f74e > company_name : QubeCinema, Inc. > product_name : QubeMaster Pro > product_version : 2.5 > product_uid : a6d3ea56-8155-4dfc-86f6-664b12671427 > modification_date: 1969-08-14 04:58:00 > application_platform: win32 > timecode : 00:00:00:00 > Duration: 00:01:38.42, start: 0.000000, bitrate: 91072 kb/s > Stream #0:0: Video: jpeg2000 (JPEG 2000 digital cinema 2K), xyz12le, > 2048x858, 24 tbr, 24 tbn, 24 tbc > [libx264 @ 0000000002def280] using cpu capabilities: MMX2 SSE2Fast SSSE3 > SSE4.2 > [libx264 @ 0000000002def280] profile Main, level 4.1 > [libx264 @ 0000000002def280] 264 - core 142 r2479 dd79a61 - H.264/MPEG-4 > AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - > options: cabac=0 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=4 psy=1 > psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 > cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=36 > lookahead_threads=8 sliced_threads=0 nr=0 decimate=1 interlaced=0 > bluray_compat=0 constrained_intra=0 bframes=1 b_pyramid=0 b_adapt=1 > b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=60 keyint_min=6 > scenecut=40 intra_refresh=0 rc_lookahead=20 rc=crf mbtree=1 crf=21.0 > qcomp=0.60 qpmin=4 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 > Output #0, mov, to 'Test.mov': > Metadata: > uid : 5831cb6f-9270-4545-a789-5b79221e0074 > generation_uid : 9583e162-55e1-46ac-b111-4ee032c5f74e > company_name : QubeCinema, Inc. > product_name : QubeMaster Pro > product_version : 2.5 > product_uid : a6d3ea56-8155-4dfc-86f6-664b12671427 > modification_date: 1969-08-14 04:58:00 > application_platform: win32 > timecode : 00:00:00:00 > encoder : Lavf56.12.103 > Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, > 1920x1080, q=-1--1, 24 fps, 12288 tbn, 24 tbc > Metadata: > encoder : Lavc56.12.100 libx264 > Stream mapping: > Stream #0:0 -> #0:0 (jpeg2000 (native) -> h264 (libx264)) > Press [q] to stop, [?] for help > frame= 2362 fps=9.4 q=-1.0 Lsize= 52668kB time=00:01:38.37 > bitrate=4385.8kbits/s > video:52642kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 0.049134% > [libx264 @ 0000000002def280] frame I:73 Avg QP:17.39 size:139477 > [libx264 @ 0000000002def280] frame P:1431 Avg QP:19.90 size: 25280 > [libx264 @ 0000000002def280] frame B:858 Avg QP:20.86 size: 8797 > [libx264 @ 0000000002def280] consecutive B-frames: 27.3% 72.7% > [libx264 @ 0000000002def280] mb I I16..4: 51.5% 0.0% 48.5% > [libx264 @ 0000000002def280] mb P I16..4: 15.0% 0.0% 2.6% P16..4: > 29.8% 6.5% 1.2% 0.0% 0.0% skip:44.9% > [libx264 @ 0000000002def280] mb B I16..4: 1.6% 0.0% 0.3% B16..8: > 13.1% 3.0% 0.1% direct:11.6% skip:70.4% L0:38.5% L1:54.3% BI: 7.1% > [libx264 @ 0000000002def280] coded y,uvDC,uvAC intra: 34.6% 64.0% 28.7% > inter: 8.6% 22.5% 1.1% > [libx264 @ 0000000002def280] i16 v,h,dc,p: 49% 21% 17% 13% > [libx264 @ 0000000002def280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 18% 15% > 7% 8% 8% 7% 7% 5% > [libx264 @ 0000000002def280] i8c dc,h,v,p: 55% 18% 21% 6% > [libx264 @ 0000000002def280] Weighted P-Frames: Y:10.3% UV:5.0% > [libx264 @ 0000000002def280] ref P L0: 74.0% 26.0% > [libx264 @ 0000000002def280] kb/s:4381.77 > > > Any ideas what may be wrong? > From rens at onlinemedia.nl Thu Nov 27 23:19:22 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Thu, 27 Nov 2014 23:19:22 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <4319619.1101417126605100.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <16025984.1121417126762506.JavaMail.root@webmail.onlinemedia.nl> Rens Dijkshoorn onlinemedia.nl> writes: > The resulting file is now playing fine. > Is --enable-libx264 responsible? No, this configuration: --arch=x86_64 --enable-libx264 --enable-gpl --enable-nonfree works > configuration: The following options have no effect, please remove them, they often cause confusion: > --arch=x86_64 > --enable-pthreads > --enable-avfilter I will try to figure out which option is responsible rens From nichot20 at yahoo.com Fri Nov 28 10:08:59 2014 From: nichot20 at yahoo.com (tim nicholson) Date: Fri, 28 Nov 2014 09:08:59 +0000 Subject: [FFmpeg-user] compiling with libx265 Message-ID: <54783BAB.2040609@yahoo.com> Thought I'd have a go at this, but suffered an odd epic fail. x265 builds and installs fine, and the standalone at least responds to x265 -h. there are the expected:- /usr/local/lib64/x265.a /usr/local/lib64/pkgconfig/x265.pc /usr/local/include/x265.h /usr/local/include/x265_config.h however ./configure reports:- "ERROR: x265 not found" This is a lie as the end of config log shows it is found, but throws up lots of "undefined reference to " errors:- check_pkg_config x265 x265.h x265_encoder_encode pkg-config --exists --print-errors x265 check_func_headers x265.h x265_encoder_encode -I/usr/local/include -L/usr/local/lib64 -lx265 check_ld cc -I/usr/local/include -L/usr/local/lib64 -lx265 check_cc -I/usr/local/include -L/usr/local/lib64 BEGIN /tmp/ffconf.8IJVvbxA.c 1 #include 2 long check_x265_encoder_encode(void) { return (long) x265_encoder_encode; } 3 int main(void) { return 0; } END /tmp/ffconf.8IJVvbxA.c gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I/usr/local/include -static -std=c99 -fomit-frame-pointer -pthread -I/mnt/msds-store-0/tims/ffmpeg-tux/usr/local/include/freetype2 -I/mnt/msds-store-0/tims/ffmpeg-tux/usr/local/include -I/usr/local/include -I/usr/local/include -L/usr/local/lib64 -c -o /tmp/ffconf.1n1Ue3xn.o /tmp/ffconf.8IJVvbxA.c gcc -L/usr/local/lib64 -static -ldl -Wl,--as-needed -Wl,-z,noexecstack -I/usr/local/include -L/usr/local/lib64 -o /tmp/ffconf.QTquSvyv /tmp/ffconf.1n1Ue3xn.o -lx265 -L/usr/local/lib64 -lx264 -lpthread -lm -L/mnt/msds-store-0/tims/ffmpeg-tux/usr/local/lib64 -lfreetype -lfdk-aac -lfaac -lm -lz -pthread /usr/local/lib64/libx265.a(bitcost.cpp.o): In function `x265::BitCost::CalculateLogs()': bitcost.cpp:(.text+0x26): undefined reference to `operator new[](unsigned long)' [..... lots more undefined reference to ...] /usr/local/lib64/libx265.a(wavefront.cpp.o): In function `x265::WaveFront::~WaveFront()': wavefront.cpp:(.text+0x4e): undefined reference to `operator delete(void*)' /usr/local/lib64/libx265.a(wavefront.cpp.o):(.data.rel.ro._ZTIN4x2659WaveFrontE[_ZTIN4x2659WaveFrontE]+0x0): undefined reference to `vtable for __cxxabiv1::__si_class_type_info' /usr/local/lib64/libx265.a(wavefront.cpp.o):(.data.rel.ro._ZTVN4x2659WaveFrontE[_ZTVN4x2659WaveFrontE]+0x28): undefined reference to `__cxa_pure_virtual' /usr/local/lib64/libx265.a(deblock.cpp.o): In function `x265::Deblock::getBoundaryStrength(x265::CUData const*, int, unsigned int, unsigned char const*)': deblock.cpp:(.text+0x488): undefined reference to `__cxa_guard_acquire' deblock.cpp:(.text+0x502): undefined reference to `__cxa_guard_release' collect2: error: ld returned 1 exit status ERROR: x265 not found all with a fresh clone of x265 and ffmpeg ea38e5a... on gcc 4.8.3 -- Tim. Key Fingerprint 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83 From cehoyos at ag.or.at Fri Nov 28 10:55:50 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 28 Nov 2014 09:55:50 +0000 (UTC) Subject: [FFmpeg-user] compiling with libx265 References: <54783BAB.2040609@yahoo.com> Message-ID: tim nicholson ffmpeg.org> writes: > Thought I'd have a go at this, but suffered an odd epic fail. Use this patch as a work-around: http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/178167 Carl Eugen From nichot20 at yahoo.com Fri Nov 28 11:31:21 2014 From: nichot20 at yahoo.com (tim nicholson) Date: Fri, 28 Nov 2014 10:31:21 +0000 Subject: [FFmpeg-user] compiling with libx265 In-Reply-To: References: <54783BAB.2040609@yahoo.com> Message-ID: <54784EF9.9010604@yahoo.com> On 28/11/14 09:55, Carl Eugen Hoyos wrote: > tim nicholson ffmpeg.org> writes: > >> Thought I'd have a go at this, but suffered an odd epic fail. > > Use this patch as a work-around: > http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/178167 > thanks Carl, not sure if it will work for me, as I am trying to make a static build and the above thread has made me realise I am missing the glibcpp static libraries, only having the glibc ones... but at least I now understand why it was such an epic fail.... > Carl Eugen > -- Tim. Key Fingerprint 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83 From guido.holz at googlemail.com Fri Nov 28 15:50:33 2014 From: guido.holz at googlemail.com (Guido Holz) Date: Fri, 28 Nov 2014 15:50:33 +0100 Subject: [FFmpeg-user] (no subject) Message-ID: my problem is after exporting from Adobe Premiere and postwork with ffmpeg I get more frames of each mp4-footage. I minimalized it to the following example: I exported 2:00.0 min of black-screen from Adobe Premiere and after encoding it through ffmpeg like :\>ffmpeg.exe -y -i before.mp4 after.mp4 it has 4 frames more (in Adobe Premiere only 1 see screenshot). When I make :\>ffmpeg.exe -y -i before.mp4 -c copy after.mp4 everything is fine. Somethng happens with the encoder? What I saw - but don't know wht it means is: before.mp4 : yuv420p(tv) after.mp4 : yuv420p Ouptut for before.mp4 (Adobe Premiere export) --------------------------------------- :\> ffmpeg.exe -i before.mp4 ffmpeg version N-60215-g2a9c507 Copyright (c) 2000-2014 the FFmpeg developers built on Jan 27 2014 22:06:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 63.100 / 52. 63.100 libavcodec 55. 49.100 / 55. 49.100 libavformat 55. 28.100 / 55. 28.100 libavdevice 55. 7.100 / 55. 7.100 libavfilter 4. 1.101 / 4. 1.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'before.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 creation_time : 2014-11-28 14:29:09 Duration: 00:02:00.00, start: 0.040000, bitrate: 70 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], 68 kb/s, 25 fps, 25 tbr, 25k tbn, 50 tbc (default) Metadata: creation_time : 2014-11-28 14:29:09 handler_name : ?Mainconcept Video Media Handler At least one output file must be specified Output for after.mp4 -------------------------------------------------------------------------- ffmpeg.exe -i after.mp4 ffmpeg version N-60215-g2a9c507 Copyright (c) 2000-2014 the FFmpeg developers built on Jan 27 2014 22:06:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 63.100 / 52. 63.100 libavcodec 55. 49.100 / 55. 49.100 libavformat 55. 28.100 / 55. 28.100 libavdevice 55. 7.100 / 55. 7.100 libavfilter 4. 1.101 / 4. 1.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'after.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.28.100 Duration: 00:02:00.04, start: 0.000000, bitrate: 17 kb/s Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 14 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: handler_name : VideoHandler At least one output file must be specified I don't understand where the differences are. thanks for helping -------------- next part -------------- A non-text attachment was scrubbed... Name: ffmpeg.jpg Type: image/jpeg Size: 13684 bytes Desc: not available URL: From rens at onlinemedia.nl Fri Nov 28 16:06:57 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 28 Nov 2014 16:06:57 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <14067675.1541417187000897.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <3777684.1561417187217045.JavaMail.root@webmail.onlinemedia.nl> The ffmpeg-2.4.3 version is broken for this command ffmpeg -i AVC100CBG_1920_1080_H422IP.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 FFMPEG.MOV The latest version from git this afternoon works fine with this configuration ffmpeg version N-68088-gf001a2b Copyright (c) 2000-2014 the FFmpeg developers built on Nov 28 2014 14:25:15 with Apple LLVM version 6.0 (clang-600.0.54) (based on LLVM 3.5svn) configuration: --enable-libvpx --enable-libx264 --enable-libx265 --enable-libopenjpeg --enable-libmp3lame --enable-libfaac --enable-libfdk-aac --enable-libfreetype --enable-libass --enable-gpl --enable-nonfree --enable-shared libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.101 / 56. 15.101 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... Regards Rens From krueger at lesspain.de Fri Nov 28 16:36:41 2014 From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=) Date: Fri, 28 Nov 2014 16:36:41 +0100 Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <3777684.1561417187217045.JavaMail.root@webmail.onlinemedia.nl> References: <14067675.1541417187000897.JavaMail.root@webmail.onlinemedia.nl> <3777684.1561417187217045.JavaMail.root@webmail.onlinemedia.nl> Message-ID: On Fri, Nov 28, 2014 at 4:06 PM, Rens Dijkshoorn wrote: > > The ffmpeg-2.4.3 version is broken for this command > > ffmpeg -i AVC100CBG_1920_1080_H422IP.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 FFMPEG.MOV > Just curious but why ai12 when it's XAVC? Isn't aivx the correct fourCC in this case? http://ffmpeg.org/pipermail/ffmpeg-user/2013-October/018055.html From rens at onlinemedia.nl Fri Nov 28 17:49:24 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 28 Nov 2014 17:49:24 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <8888861.1751417193148034.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <23108839.1771417193364064.JavaMail.root@webmail.onlinemedia.nl> On Fri, Nov 28, 2014 at 4:06 PM, Rens Dijkshoorn wrote: > > The ffmpeg-2.4.3 version is broken for this command > > ffmpeg -i AVC100CBG_1920_1080_H422IP.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag ai12 FFMPEG.MOV > Just curious but why ai12 when it's XAVC? Isn't aivx the correct fourCC in this case? -- When i rewrap using -vtag aivx ffmpeg -i AVC100CBG_1920_1080_H422IP.MXF -map 0:0 -map 0:1 -map 0:2 -c copy -vtag aivx FFMPEG.MOV Then Quicktime ( only tested with 7 ) can't play the file, furter more FCPX-10.1.3 uses the same vtag ai12 when rewrapping this to an original AVC-Intra format in mov. The logic I found checking what FCPX uses to rewrap to mov but not completly tested all possible combinations - ai12 is used for HD 25, 29,97 - ai13 is used for HD 23.98 - aivx is used for 4K formats If i use -vtag aivx in ffmpeg then playing in Quicktime won't work but cmd-I shows the correct info but the player window remains green. Mplayer will playback video. FFmbc refuses to use -vtag aivx but when using -vtag ai12 produces a 4K file that plays in quicktime the cmd-I info is not consistend with the video as would be expected ( wrong frame rate). Trying same thing in FFMPEG but this won't work for an AVC_4096_2160_H422IP at L52 50FPS file. A AVC_4096_2160_H422IP at L51 25FPS -vtag ai12 plays when rewrapped bij FFMBC and FFMPEG in Quicktime-7, Quicktime PlayerX prompts an error " operation could not be completed ". There are a lot more options to investigate but the initial conclusion is that everything above HD frame size has problems in just rewrapping to quickitme mov. FFmbc-7.1 seems to produce a more tolerant version but is not prefect either. Om my test system i have FCPX installed and the Sony XAVC plugin. Any Suggestion to get this fixed are more then welcome. Rens From cehoyos at ag.or.at Fri Nov 28 19:07:26 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 28 Nov 2014 18:07:26 +0000 (UTC) Subject: [FFmpeg-user] XAVC to Quicktime MOV References: <14067675.1541417187000897.JavaMail.root@webmail.onlinemedia.nl> <3777684.1561417187217045.JavaMail.root@webmail.onlinemedia.nl> Message-ID: Rens Dijkshoorn onlinemedia.nl> writes: > The latest version from git this afternoon works fine So there is no issue that you want to report? Please try to fix your quoting, Carl Eugen From rens at onlinemedia.nl Fri Nov 28 19:33:46 2014 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 28 Nov 2014 19:33:46 +0100 (CET) Subject: [FFmpeg-user] XAVC to Quicktime MOV In-Reply-To: <28821208.1801417199336699.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <10189496.1821417199626018.JavaMail.root@webmail.onlinemedia.nl> > Rens Dijkshoorn onlinemedia.nl> writes: > > > The latest version from git this afternoon works fine > > So there is no issue that you want to report? > > Please try to fix your quoting, Carl Eugen > Maybe I should refrase this to " The latest version from git this afternoon works fine for HD " If you read my previous posting there are still a number of issues with rewrapping 4K XAVC MXF content to quicktime MOV that would be nice if they could be resolved in the future. If these are bugs or not implemented at this time is for me as user difficult to determin. Thanks for quick response. Regards Rens From john at dimis.fim.uni-passau.de Fri Nov 28 15:29:16 2014 From: john at dimis.fim.uni-passau.de (S. John) Date: Fri, 28 Nov 2014 15:29:16 +0100 Subject: [FFmpeg-user] Error with parameterization using h264 with profiles Message-ID: <547886BC.9060506@dimis.fim.uni-passau.de> Hi everybody! I'm struggling with a simple task. Converting an .avi containing a single cvid video stream with rgb24 color model to a .mp4 with h264 encoding. For some reason, ffmpeg won't let me use any other h264 profile other than high444. With every other profile i use, the encoder still seems to use yuv444p pixel format (which is only allowed in high444). Doing so / ffmpeg -i input.avi -vcodec libx264 -vprofile main -pix_fmt yuv420p out.mp4/ results in an error: Output #0, mp4, to 'out.mp4': Stream #0:0: Video: h264, yuv444p, 320x240, q=-1--1, 90k tbn, 15 tbc Stream mapping: Stream #0:0 -> #0:0 (cinepak -> libx264) /Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width o//r height/ I can circumvent the problem by manually specifying the correct pixel format: /ffmpeg -i input.avi -vcodec libx264 -vprofile main -pix_fmt yuv420p out.mp4/ Is this an intended behavior? For what I read in the documentation, I expected ffmpeg to automatically choose parameters in accordance to the selected profile. Or is the profile selection more like an assert, just telling me whenever I'm not in accordance with the constraints of a selected profile? Thank you for any information on this topic! Stefan From m.nalis at verbavoice.de Fri Nov 28 16:27:08 2014 From: m.nalis at verbavoice.de (Marko Nalis) Date: Fri, 28 Nov 2014 15:27:08 +0000 Subject: [FFmpeg-user] ffserver http header Message-ID: <969FA4F0799A04429645DA802D8202C3DF92A5@Exchange.verbavoice.local> Hello, I need to put custom http headers into my ffserver output stream. In ffmpeg you would do it with the -headers command. Is it possible to achieve this in ffserver? I particularly need to set "Transfer-Encoding: chunked", "Content-Length:0" and "AcceptRanges:none" Help would be much appreciated Thanks in advance From m.nalis at verbavoice.de Fri Nov 28 16:33:59 2014 From: m.nalis at verbavoice.de (Marko Nalis) Date: Fri, 28 Nov 2014 15:33:59 +0000 Subject: [FFmpeg-user] ffserver set min_udp port and max_port Message-ID: <969FA4F0799A04429645DA802D8202C3DF92B4@Exchange.verbavoice.local> Hello, Is it possible to set min_port and max_port for rtsp udp streaming in the ffserver config file? Because the default range of 5000-65000 won't work for me because of firewall restrictions. But I need to do udp streaming and not tcp Thanks for any help I can get From cehoyos at ag.or.at Fri Nov 28 21:44:17 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Fri, 28 Nov 2014 20:44:17 +0000 (UTC) Subject: [FFmpeg-user] Error with parameterization using h264 with profiles References: <547886BC.9060506@dimis.fim.uni-passau.de> Message-ID: S. John dimis.fim.uni-passau.de> writes: > Doing so > ffmpeg -i input.avi -vcodec libx264 -vprofile main -pix_fmt yuv420p out.mp4 > results in an error: Complete, uncut console output missing. Carl Eugen From kranthi9s at gmail.com Sun Nov 30 16:31:23 2014 From: kranthi9s at gmail.com (kranthi kumar) Date: Sun, 30 Nov 2014 21:01:23 +0530 Subject: [FFmpeg-user] How To Play YouTube Videos using FFPLAy Message-ID: Hi, This is very Important to me, for playing or receiving live streaming videos from YouTube .So,please tell me How to play YouTube videos using FFplay. I would also like to know how to re-stream Video to local servers from YouTube using UDP protocol . I Tried " [b]ffplay -i http://youtube.com/xxxxxx[/b] " for just playing YouTube live streaming video. but I'm getting TSL and Input/Output error. So,please give a solution for the above. Thank You. Kranthi -- Bavuluru Kranthi Kumar From h.reindl at thelounge.net Sun Nov 30 20:08:15 2014 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 30 Nov 2014 20:08:15 +0100 Subject: [FFmpeg-user] How To Play YouTube Videos using FFPLAy In-Reply-To: References: Message-ID: <547B6B1F.90705@thelounge.net> Am 30.11.2014 um 16:31 schrieb kranthi kumar: > This is very Important to me, for playing or receiving live streaming > videos from YouTube .So,please tell me How to play YouTube videos using > FFplay. > I would also like to know how to re-stream Video to local servers from > YouTube using UDP protocol . > I Tried " [b]ffplay -i http://youtube.com/xxxxxx[/b] " for just playing > YouTube live streaming video. but I'm getting TSL and Input/Output error. > > So,please give a solution for the above no mediaplayer on this planet will play a http://youtube.com/xxxxxx URL which is just a *website* and not a videofile you need a way to parse that HTML page, follow all sort of links and embeds and finally get the URL of the video itself -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 181 bytes Desc: OpenPGP digital signature URL: From cehoyos at ag.or.at Sun Nov 30 20:12:34 2014 From: cehoyos at ag.or.at (Carl Eugen Hoyos) Date: Sun, 30 Nov 2014 19:12:34 +0000 (UTC) Subject: [FFmpeg-user] How To Play YouTube Videos using FFPLAy References: <547B6B1F.90705@thelounge.net> Message-ID: Reindl Harald thelounge.net> writes: > no mediaplayer on this planet will play a > http://youtube.com/xxxxxx URL I believe it should work with FFplay if compiled with --enable-libquvi. Carl Eugen From lou at lrcd.com Sun Nov 30 20:13:37 2014 From: lou at lrcd.com (Lou) Date: Sun, 30 Nov 2014 10:13:37 -0900 Subject: [FFmpeg-user] How To Play YouTube Videos using FFPLAy In-Reply-To: References: Message-ID: <1417374817.2292863.197005457.313732C0@webmail.messagingengine.com> On Sun, Nov 30, 2014, at 06:31 AM, kranthi kumar wrote: > Hi, > This is very Important to me, for playing or receiving live streaming > videos from YouTube .So,please tell me How to play YouTube videos using > FFplay. > I would also like to know how to re-stream Video to local servers from > YouTube using UDP protocol . > I Tried " [b]ffplay -i http://youtube.com/xxxxxx[/b] " for just playing > YouTube live streaming video. but I'm getting TSL and Input/Output error. > > So,please give a solution for the above. If your build supports libquvi then you may be able to just do: ffplay http://www.youtube.com/watch?v=xxxxxx However I haven't tried it myself. From u at pkh.me Sun Nov 30 20:20:59 2014 From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=) Date: Sun, 30 Nov 2014 20:20:59 +0100 Subject: [FFmpeg-user] How To Play YouTube Videos using FFPLAy In-Reply-To: References: <547B6B1F.90705@thelounge.net> Message-ID: <20141130192059.GB16842@leki> On Sun, Nov 30, 2014 at 07:12:34PM +0000, Carl Eugen Hoyos wrote: > Reindl Harald thelounge.net> writes: > > > no mediaplayer on this planet will play a > > http://youtube.com/xxxxxx URL > > I believe it should work with FFplay if compiled > with --enable-libquvi. > Note: I'm curious about the outcome of this nowadays since libquvi seems discontinued. We might want to check if youtube-dl provides a maintained alternative. -- Cl?ment B. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 473 bytes Desc: not available URL: From maillist35650 at gmail.com Sun Nov 30 22:18:32 2014 From: maillist35650 at gmail.com (mail list) Date: Sun, 30 Nov 2014 22:18:32 +0100 Subject: [FFmpeg-user] bug in zoompan with overlay filters ? Message-ID: Hello, It seems there is a bug in zoompan. If we use an overlay filter on the output of zoompan, the zoom progress stops at 2 seconds whatever the 'd' parameter of the zoompan. If we remove the overlay fliter, it works. Here is the command; ffmpeg -loglevel info -y -f lavfi -r 25 -i color=black -loop 1 -r 25 -i pic.png -filter_complex " \ [0:v]scale=640x480[background]; [\ 1:v]scale=640x480,zoompan=z='zoom+0.001':x='ow/2':y='oh/2':s=640x480:d=250,setpts=PTS-STARTPTS+1/TB [vshot2]; \ [background][vshot2] overlay[vfinal]" \ -strict -2 -c:v h264 -b:a 96k -aspect 640/480 -pix_fmt yuv420p -t 10 -map [vfinal] final.mp4 Here are some logs: ffmpeg version N-41370-gb4d8724- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2014 the FFmpeg developers built on Nov 24 2014 00:39:29 with gcc 4.8 (Debian 4.8.3-13) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable-libass --enable-gnutls --cc=gcc-4.8 libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.100 / 56. 15.100 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 ... [swscaler @ 0x9fe99e0] Warning: data is not aligned! This can lead to a speedloss [Parsed_overlay_4 @ 0xa044e40] [framesync @ 0xa044ee4] Buffer queue overflow, dropping. Last message repeated 184 times [Parsed_overlay_4 @ 0xa044e40] [framesync @ 0xa044ee4] Buffer queue overflow, dropping. Last message repeated 185 times rgds Jean