April 2014 - FFmpeg-trac - lists.ffmpeg.org

#3572(undetermined:new): failure to remux mkv to mkv with pts < dts error
by FFmpeg 06 Sep '18

06 Sep '18

#3572: failure to remux mkv to mkv with pts < dts error -------------------------------------+------------------------------------- Reporter: bubbleguuum | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- This problematic mkv used in the command line belwos was recorded by TVheadend off Satellite on a UK HD (BBC) channel. The 5MB sample can be downloaded here: http://bubblesoftapps.com/tmp/ToyStory2.2014-04-13.20-00-sample.mkv The command line attempt to keep the h264 stream (copy) and convert the mp2 audio track to mp3. It fails with a pts < dts error, generating a very short 100Kb file. If the h264 is transcoded to h264 instead of just copied, it works. So there seems to be a problem keeping the original video track while transcoding the audio track. How to reproduce: {{{ % $ ffmpeg -y -i ToyStory2.2014-04-13.20-00-sample.mkv -map 0:0 -c:0 copy -map 0:1 -c:1 libmp3lame -ac:1 2 -b:1 320k test.mkv ffmpeg version N-62475-g256f530 Copyright (c) 2000-2014 the FFmpeg developers built on Apr 15 2014 22:01:57 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-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 76.100 / 52. 76.100 libavcodec 55. 58.103 / 55. 58.103 libavformat 55. 37.100 / 55. 37.100 libavdevice 55. 13.100 / 55. 13.100 libavfilter 4. 4.100 / 4. 4.100 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [h264 @ 02ca0360] mmco: unref short failure Last message repeated 4 times [h264 @ 02ca0360] number of reference frames (0+5) exceeds max (4; probably corrupt input), discarding one Input #0, matroska,webm, from 'ToyStory2.2014-04-13.20-00-sample.mkv': Metadata: title : Toy Story 2 encoder : Tvheadend Matroska muxer DATE_BROADCASTED: 2014-04-13 20:00:00 ORIGINAL_MEDIA_TYPE: TV CONTENT_TYPE : Movie / Drama TVCHANNEL : BBC Three HD SUMMARY : Animated comedy. When Woody is kidnapped, Buzz Lightyear and the other toys set out in hot pursuit. However, when they find him they discover him in a new life, with new friends. [1999] [AD,S] SUMMARY-eng : Animated comedy. When Woody is kidnapped, Buzz Lightyear and the other toys set out in hot pursuit. However, when they find him they discover him in a new life, with new friends. [1999] [AD,S] Duration: 01:42:05.14, start: 0.005000, bitrate: 6 kb/s Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 1k tbn, 50 tbc (default) Stream #0:1(nar): Audio: mp2, 48000 Hz, stereo, s16p, 256 kb/s (default) Stream #0:2(eng): Subtitle: dvb_subtitle (default) Stream #0:3(eng): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s (default) Stream #0:4(eng): Subtitle: subrip (default) Output #0, matroska, to 'test.mkv': Metadata: title : Toy Story 2 SUMMARY-eng : Animated comedy. When Woody is kidnapped, Buzz Lightyear and the other toys set out in hot pursuit. However, when they find him they discover him in a new life, with new friends. [1999] [AD,S] DATE_BROADCASTED: 2014-04-13 20:00:00 ORIGINAL_MEDIA_TYPE: TV CONTENT_TYPE : Movie / Drama TVCHANNEL : BBC Three HD SUMMARY : Animated comedy. When Woody is kidnapped, Buzz Lightyear and the other toys set out in hot pursuit. However, when they find him they discover him in a new life, with new friends. [1999] [AD,S] encoder : Lavf55.37.100 Stream #0:0(eng): Video: h264 (H264 / 0x34363248), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 1k tbn, 1k tbc (default) Stream #0:1(nar): Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 320 kb/s (default) Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (mp2 -> libmp3lame) Press [q] to stop, [?] for help [matroska @ 04d5a9e0] pts (155) < dts (255) in stream 0 av_interleaved_write_frame(): Invalid argument [libmp3lame @ 04d05f80] Trying to remove 1152 samples, but the queue is empty [matroska @ 04d5a9e0] Encoder did not produce proper pts, making some up. frame= 3 fps=0.0 q=-1.0 Lsize= 93kB time=00:00:00.23 bitrate=3231.7kbits/s video:109kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown }}} Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3572> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3574(undetermined:new): resolved bug "ffplay says cannot use next picture in error concealment while streaming from ffserver using feed file"
by FFmpeg 07 Jan '18

07 Jan '18

#3574: resolved bug "ffplay says cannot use next picture in error concealment while streaming from ffserver using feed file" -------------------------------------+------------------------------------- Reporter: Ashutosh | Type: Status: new | enhancement Component: | Priority: important undetermined | Version: 2.2.1 Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- The current scenario: I am using ffserver for live streaming.ffserver.conf file is as follows: Port 8090 BindAddress? 0.0.0.0 MaxHTTPConnections 2000 MaxClients? 1000 MaxBandwidth? 1000 CustomLog? - RTSPPort 7654 RTSPBindAddress 0.0.0.0 NoDaemon? <Feed exile1.ffm> #ffmpeg http://localhost:8090/exile1.ffm File exile%d.ffm FileMaxSize? 10M #Launch ffmpeg -i E:\ashu\ezio.mp4 ACL allow 127.0.0.1 ACL allow localhost </Feed> # if you want to use mpegts format instead of flv # then change "live.flv" to "live.ts" # and also change "Format flv" to "Format mpegts" <Stream live.h264> Format rtp NoAudio? Feed exile1.ffm #File "rtsp://127.0.0.1:8554/live.sdp?tcp" VideoCodec? libx264 VideoFrameRate? 10 VideoBitRate? 530 VideoSize? 640x480 VideoBufferSize? 40000 AVOptionVideo me_range 16 AVOptionVideo qcomp 0.6 AVOptionVideo qmin 10 VideoGopSize? 12 PreRoll? 15 #Strict -1 #NoLoop? #StartSendOnKey? # AVOptionVideo flags +global_header </Stream> <Stream status.html> # Server status URL Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> i transcode a video and push the output to feed file as follows: ffmpeg -i input.h264 http://localhost:8090/exile1.ffm and i play the video using ffplay using the command line as follows: ffplay -i rtsp://<IP>:7654/live.h264 <IP> - IP address. I ran first ffserver in one console window,ffmpeg in other console window and ffplay in other console window. The video plays normally by the way. When ffmpeg saves the streams to exile1.ffm it's file size increases.When filesize reaches max value i.e 10 M ,ffserver says [ffm 0x80048fe0 ] inavlid stream index 143 and ffplay stops playing and says. Cannot use next picture in error concealment I have resolved this bug and i need to discuss on this to the interested developer. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3574> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3474(ffserver:new): Smart output buffering (with no delay to start playback)
by FFmpeg 07 Jan '18

07 Jan '18

#3474: Smart output buffering (with no delay to start playback) ---------------------------------------+---------------------------------- Reporter: burek | Owner: Type: enhancement | Status: new Priority: wish | Component: ffserver Version: unspecified | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 ---------------------------------------+---------------------------------- If possible, it would be good to enhance the current output buffering of ffserver so that the output buffer size is calculated in such a way, which guarantees the buffer will always contain at least 1 key frame. Further more, after a client connects to the ffserver and requests a certain stream, it would be ideal for ffserver to skip sending any data in the buffer prior to that 1 key frame and make sure that key frame is the first thing that will be sent to the client, along with all the data that comes after that key frame in the buffer. This would make it possible to start the playback of the stream immediately, regardless of the chosen GOP size. Since the data (and the key frame) is usually in a compressed form, which makes it difficult to accomplish this in an ideal way (probably the key frame will be contained in a packet with some delta frames before and after, compressed altogether), it might (probably) be viable to make ffserver first mark all the packets in the buffer that contain a keyframe (for a faster lookup later on) and then, after a client connects and requests a stream, ffserver could do a fast look-up for a first packet with a keyframe, decode it, discard all the delta frames prior the keyframe and re-encode the data back to the packed form (thus creating a new packet). This would be done only once, when a new client connects to the ffserver, so it shouldn't be that much of the overhead (to decompress the packet, get rid of all the delta frames before the key frame and to re-compress the key frame and the rest of delta frames back to a new packet). I know this is probably a much more complicated task, but this would really be an awesome feature that a lot of people is looking for desperately, since the live TV re-encoding has become ultra popular and all that can be done to lower the playback start time is to tamper with GOP interval, which makes the entire stream encoding much more worse than it should be. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3474> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3418(ffserver:new): Cannot stream h264 to libvorbis with ffserver
by FFmpeg 07 Jan '18

07 Jan '18

#3418: Cannot stream h264 to libvorbis with ffserver ----------------------------------+---------------------------------- Reporter: pferland | Type: defect Status: new | Priority: normal Component: ffserver | Version: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+---------------------------------- When trying to stream a h264/libx264 to a ffserver feed that has an ogg stream fails with {{{Invalid data found when processing input}}} ffmpeg version: {{{ 2.1.4 }}} Built on: {{{ Feb 24 2014 14:55:01 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) }}} ffmpeg config options: {{{ --prefix=/home/pferland/ffmpeg_build --extra- cflags=-I/home/pferland/ffmpeg_build/include --extra- ldflags=-L/home/pferland/ffmpeg_build/lib --bindir=/home/pferland/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab }}} libx264 version: {{{ master at git://git.videolan.org/x264.git }}} Command line for ffmpeg: {{{% ffmpeg -i input.mkv http://172.16.0.88:8091/feed1.ffm}}} ffmpeg source file headers: {{{ Input #0, matroska,webm, from '/input.mkv': Metadata: creation_time : 2013-10-14 21:32:51 Duration: 00:43:24.06, start: 0.000000, bitrate: 3833 kb/s Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709), 1280x720, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc Stream #0:1: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s (default) }}} ffserver config file: {{{ Port 8091 BindAddress 172.16.0.88 MaxHTTPConnections 666 MaxClients 666 #MaxBandwidth 4000 CustomLog /var/log/ffserver/access.log NoDaemon <Feed feed1.ffm> File /tmp/feed1.ffm FileMaxSize 8G </Feed> <Stream stream1.ogg> Feed feed1.ffm Format ogg VideoCodec libtheora VideoFrameRate 15 VideoBitRate 500 VideoSize 640x480 # VideoQMin 1 # VideoQMax 31 VideoGopSize 12 Preroll 0 AVOptionVideo flags +global_header AudioCodec libvorbis AudioBitRate 64 AudioChannels 1 AudioSampleRate 44100 AVOptionAudio flags +global_header </Stream> }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3418> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3414(undetermined:new): ffmpeg stream to ffserver fails with "Not enough space"
by FFmpeg 07 Jan '18

07 Jan '18

#3414: ffmpeg stream to ffserver fails with "Not enough space" -------------------------------------+------------------------------------- Reporter: melsom | Type: defect Status: new | Priority: important Component: | Version: git- undetermined | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I'm having a problem streaming to ffserver with the latest Windows-build by Zeranoe. Works under Linux with the same version. Is this a bug or is there anything i'm missing? Same commandline works just fine under linux with the same git-version. There are no space issues. Both ffserver and machine running windows has plenty of space available. {{{ ffmpeg -i http://10.10.0.108:7522/upnp/channelstream/18.ts http://10.0.1.100:8090/feed1.ffm ffmpeg version N-60859-g313a6c6 Copyright (c) 2000-2014 the FFmpeg developers built on Feb 23 2014 22:01:50 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --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-libx265 --enable-libxavs --enable-libxvid --enable -zlib libavutil 52. 65.100 / 52. 65.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 1.103 / 4. 1.103 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [mpeg2video @ 0441f640] Invalid frame dimensions 0x0. Last message repeated 2 times Input #0, mpegts, from 'http://10.10.0.108:7522/upnp/channelstream/18.ts': Duration: N/A, start: 92735.966167, bitrate: 247 kb/s Program 163 Stream #0:0[0x200]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv), 720x576 [SAR 64:45 DAR 16:9], max. 3750 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x280]: Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 247 kb/s http://10.0.1.100:8090/feed1.ffm: Not enough space ffserver.conf: Port 8090 BindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 100000 CustomLog - #NoDaemon <Feed feed1.ffm> File /tmp/feed1.ffm FileMaxSize 20G ACL allow x.x.x.x # Removed </Feed> <Stream test.h264> Feed feed1.ffm Format h264 # Audio settings AudioCodec libfaac AudioBitRate 96 AudioSampleRate 44100 AudioChannels 2 # Video settings VideoCodec libx264 VideoSize 640x368 VideoFrameRate 50 AVOptionVideo flags +global_header AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey VideoBitRate 1024 </Stream> }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3414> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3479(undetermined:new): jpegls: support sample interleaved images
by FFmpeg 04 Mar '17

04 Mar '17

#3479: jpegls: support sample interleaved images -------------------------------------+------------------------------------- Reporter: ami_stuff | Owner: Type: defect | Status: new Priority: normal | Component: Version: | undetermined unspecified | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- file from charls archive, decodes correctly with hp plugin {{{ C:\>ffmpeg -i T8C2E0.JLS ffmpeg version N-61570-gaa86ccc Copyright (c) 2000-2014 the FFmpeg developers built on Mar 17 2014 22:01:34 with gcc 4.8.2 (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-libcaca --enable- libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable- libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable- libwavp ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable -zlib libavutil 52. 67.100 / 52. 67.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 34.101 / 55. 34.101 libavdevice 55. 11.100 / 55. 11.100 libavfilter 4. 3.100 / 4. 3.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [jpegls @ 02dd2660] Sample interleaved images is not implemented. Update your FF mpeg version to the newest one from Git. If the problem still occurs, it means t hat your file has a feature which has not been implemented. Input #0, image2, from 'T8C2E0.JLS': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0:0: Video: jpegls, rgb24, 256x256, 25 tbr, 25 tbn, 25 tbc At least one output file must be specified }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3479> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3529(undetermined:new): optimize cinepak encoder
by FFmpeg 12 Feb '17

12 Feb '17

#3529: optimize cinepak encoder -------------------------------------+------------------------------------- Reporter: ami_stuff | Owner: Type: defect | Status: new Priority: normal | Component: Version: | undetermined unspecified | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- it's too slow right now to be usable -- Ticket URL: <https://trac.ffmpeg.org/ticket/3529> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3480(ffmpeg:new): Playback video with planar audio to decklink card
by FFmpeg 31 Jan '17

31 Jan '17

#3480: Playback video with planar audio to decklink card -------------------------------------+------------------------------------- Reporter: abalan | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: git- Keywords: DeckLink | master Output | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- == Summary of the bug :== Playback video to the DeckLink 4K Extreme card. Consist crash in both Linux and Windows when using planar audio. Works with video if -an flag used. The video used is Windows Sample video. (Wildlife.wmv). == How to reproduce :== Input - Windows Sample video. (Wildlife.wmv) {{{ $ ffmpeg -vsync 1 -i Wildlife.wmv -c:v rawvideo -pix_fmt uyvy422 -c:a pcm_s16le -s 1920x1080 -r 30000/1001 -ar 48000 -f decklink 'DeckLink 4K Extreme' ffmpeg version git-2014-03-19-074db4b Copyright (c) 2000-2014 the FFmpeg developers built on Mar 19 2014 10:22:58 with gcc 4.8 (Ubuntu 4.8.1-2ubuntu1~12.04) configuration: --bindir=/usr/local/bin --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfdk-aac --enable-libmp3lame --enable-libtheora --enable-decklink --enable-libvorbis --enable-libvpx --enable-libx264 --extra-libs=-ldl --enable-gpl --enable-libass --enable- x11grab --enable-shared --prefix=/usr/local --extra- cflags=-I/home/dvrp/Downloads/Blackmagic_DeckLink_SDK_9.7.7/Linux/include --extra- ldflags=-L/home/dvrp/Downloads/Blackmagic_DeckLink_SDK_9.7.7/Linux/include libavutil 52. 67.100 / 52. 67.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 34.101 / 55. 34.101 libavdevice 55. 11.100 / 55. 11.100 libavfilter 4. 3.100 / 4. 3.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 }}} The sample video can be downloaded from http://www.datafilehost.com/d/53cc7cf4 -- Ticket URL: <https://trac.ffmpeg.org/ticket/3480> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3518(undetermined:new): ffmpeg lacks Raspberry Pi hardware encoding support
by FFmpeg 21 Jan '17

21 Jan '17

#3518: ffmpeg lacks Raspberry Pi hardware encoding support -------------------------------------+------------------------------------- Reporter: bmarwell | Type: Status: new | enhancement Component: | Priority: normal undetermined | Version: Keywords: | unspecified Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- Summary of the bug: While gstreamer has a omxh264enc sinc which does use the raspberry pi's built in h264 hardware encoder, ffmpeg will not use this one. How to reproduce: {{{ % ffmpeg -i input -c:v libx264 ... ffmpeg version <any> }}} Now that there's a floss driver, I hope such an enhancemant could be done. git of gst-omx: http://cgit.freedesktop.org/gstreamer/gst-omx/ Raspberry Pi driver announcement: http://www.raspberrypi.org/archives/6561 Beware, the raspberry pi page is barely useable today (April 1st). -- Ticket URL: <https://trac.ffmpeg.org/ticket/3518> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3576(ffmpeg:new): decoding MP3 (LAME) results in wrong length
by FFmpeg 24 Oct '16

24 Oct '16

#3576: decoding MP3 (LAME) results in wrong length -------------------------------------+------------------------------------- Reporter: tobwen | Type: defect Status: new | Priority: important Component: ffmpeg | Version: git- Keywords: MP3, LAME, | master audio, delay, padding | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- = What I was trying to accomplish = I was trying to transcode from MP3 to 44.1 kHz, 16-bit stereo WAV file. Since it uses libmp3lame, the output should be equal to the CLI version. = The problem I encountered = The output is too long by 1263 samples. I've used LAME v3.99.5 aka v3.99r (same es libmp3lame) to transcode a testfile with 441000 samples (10 seconds) to MP3 using this command line: {{{lame test_tobwen.wav -V 2 test_tobwen.mp3}}} When decoding it with the same lame version, it's exactly 441000 samples long. {{{lame --decode test_tobwen.mp3 output_lame.wav}}} LAME recognizes its own tag and skils encoder+decoder delay (1105 samples) and padding-decoder delay (263) samples. You can read more about this here: http://mp3decoders.mp3-tech.org/decoders_lame.html = excact command line I was using = {{{ffmpeg -i test_tobwen.mp3 -acodec pcm_s16le output_ffmpeg.wav}}} = full, uncut console output = {{{ffmpeg -v 9 -loglevel 99 -i test_tobwen.mp3 -acodec pcm_s16le output_ffmpeg.wav}}} {{{ ffmpeg version N-62569-gf57ac37 Copyright (c) 2000-2014 the FFmpeg developers built on Apr 19 2014 22:03:22 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-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 52. 77.100 / 52. 77.100 libavcodec 55. 58.105 / 55. 58.105 libavformat 55. 37.101 / 55. 37.101 libavdevice 55. 13.100 / 55. 13.100 libavfilter 4. 4.100 / 4. 4.100 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'. Reading option '-i' ... matched as input file with argument 'test_tobwen.mp3'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'pcm_s16le'. Reading option 'output_ffmpeg.wav' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument 9. Successfully parsed a group of options. Parsing a group of options: input file test_tobwen.mp3. Successfully parsed a group of options. Opening an input file: test_tobwen.mp3. [mp3 @ 04395000] Format mp3 probed with size=2048 and score=51 [mp3 @ 04395000] pad 576 792 [mp3 @ 04395000] Before avformat_find_stream_info() pos: 417 bytes read:65664 seeks:2 [mp3 @ 04395000] demuxer injecting skip 1105 [mp3 @ 04395820] skip 1105 samples due to side data [mp3 @ 04395820] skip 1105/1152 samples [mp3 @ 04395000] max_analyze_duration 5000000 reached at 5015510 microseconds [mp3 @ 04395000] After avformat_find_stream_info() pos: 57761 bytes read:98432 seeks:2 frames:194 Input #0, mp3, from 'test_tobwen.mp3': Duration: 00:00:10.03, start: 0.025057, bitrate: 104 kb/s Stream #0:0, 194, 1/14112000: Audio: mp3, 44100 Hz, stereo, s16p, 104 kb/s Metadata: encoder : LAME3.99r Successfully opened the file. Parsing a group of options: output file output_ffmpeg.wav. Applying option acodec (force audio codec ('copy' to copy stream)) with argument pcm_s16le. Successfully parsed a group of options. Opening an output file: output_ffmpeg.wav. Successfully opened the file. detected 4 logical cores [graph 0 input from stream 0:0 @ 029dd2c0] Setting 'time_base' to value '1/44100' [graph 0 input from stream 0:0 @ 029dd2c0] Setting 'sample_rate' to value '44100' [graph 0 input from stream 0:0 @ 029dd2c0] Setting 'sample_fmt' to value 's16p' [graph 0 input from stream 0:0 @ 029dd2c0] Setting 'channel_layout' to value '0x3' [graph 0 input from stream 0:0 @ 029dd2c0] tb:1/44100 samplefmt:s16p samplerate:44100 chlayout:0x3 [audio format for output stream 0:0 @ 029d7920] Setting 'sample_fmts' to value 's16' [audio format for output stream 0:0 @ 029d7920] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0' [AVFilterGraph @ 043bd100] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 029d7f00] ch:2 chl:stereo fmt:s16p r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz Output #0, wav, to 'output_ffmpeg.wav': Metadata: ISFT : Lavf55.37.101 Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Metadata: encoder : LAME3.99r Stream mapping: Stream #0:0 -> #0:0 (mp3 -> pcm_s16le) Press [q] to stop, [?] for help [mp3 @ 04395820] skip 1105 samples due to side data [mp3 @ 04395820] skip 1105/1152 samples [output stream 0:0 @ 029d78a0] EOF on sink link output stream 0:0:default. No more output streams to write to, finishing. size= 1724kB time=00:00:10.00 bitrate=1411.3kbits/s video:0kB audio:1724kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004532% Input file #0 (test_tobwen.mp3): Input stream #0:0 (audio): 384 packets read (130730 bytes); 384 frames decoded (441263 samples); Total: 384 packets (130730 bytes) demuxed Output file #0 (output_ffmpeg.wav): Output stream #0:0 (audio): 384 frames encoded (441263 samples); 384 packets muxed (1765052 bytes); Total: 384 packets (1765052 bytes) muxed 384 frames successfully decoded, 0 decoding errors [AVIOContext @ 043c4a60] Statistics: 4 seeks, 387 writeouts [AVIOContext @ 043949e0] Statistics: 164020 bytes read, 2 seeks }}} == Files attached to this ticket == I've uploaded test_tobwen.wav and test_tobwen.mp3 to the FTP. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3576> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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