PCDJ DEX 3.8.0.1   

PCDJ DEX to potężne narzędzie do miksowania muzyki, przeznaczone jednocześnie dla początkujących jak i zaawansowanych użytkowników. Aplikacja obsługuje najpopularniejsze formaty plików audio: *.MP3, *.OGG, *.WMA, *.FLAC, *.WAV. Program umożliwia obsługę wtyczki VST. Ponadto narzędzie pozwala na zapis w czasie rzeczywistym granego setu do formatów *.MP3, *.WAV, *.OGG.

PCDJ DEX to aplikacja idealnie nadająca się do występów na żywo. Dużą zaletą jest możliwość współpracy z urządzeniami zewnętrznymi takimi jak: PCDJ DAC-3, DENON DN-HC4500, Behringer BCD-2000, Hercules DJ Console, Vestax VCI-100, EKS XP10, FinalScratch, VirtualDJ, SSL, MsPinky, oraz ze wszelkimi kontrolerami MIDI.
Uwaga:

Dostępna jest równiez darmowa wersja programu na naszej stronie.

 


          Musicator 1.0.0.2   
Программа для проигрывания .MPJ, воспроизведения .WAV, .MID-файлов.
          Comment on Opeth – Deliverance [Bruce Soord Vinyl Remix] – Review by Yalcin Inel   
While I don't own a vinyl setup to experience this, I can say that the remix on CD is leaps and bounds better than the original. The LP-rips of the original scattered around the net are still better than the remix cd tho'. Anyway, everyone who owns the dvd's of the remix can produce a little miracle in downmixing the surround mix to stereo with sox.exe commands. You need to rip the dts stream from the dvd with a program such as dvd-decrypter, convert it to at least 24 bit wav and with a little experimentation on sox.exe you can get the most dynamic, lush and detailed sound of the album which will create eargasm after eargasm. I guess this is right now the best way to experience what Lord AMG described here, if you don't own a vinyl player. If anyone is keen on trying this, my version was created using the command: Sox.exe -V -G deliverance.wav c:outputdeliverance.wav remix 1v0.39,3v0.06,4v0.52,5v0.34 2v0.39,3v0.06,4v0.52,6v0.34 norm -1
          #2325: MP4 AAC Audio is delayed by 2ms when converted to PCM   

Summary of the bug:

When using ffmpeg to convert an aac audio stream from an mp4 to pcm, the result is out of sync by about 2ms. By adding -ss 00:00:00.02 after the input, then output is correctly aligned.

How to reproduce:

% ffmpeg -i test100.mp4 -c:a pcm_s16le test100_audio.wav
ffmpeg version 1.1.git
git revision: faa0068
built on Mar  4 2013 11:40:27

          #5866: mxf regression with xiph sample file   

Lutz Vieweg reported an mxf regression since bf2290a80c124c5168f78be64ee87d9efdd4040d for the following file: https://media.xiph.org/video/derf/meridian/MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV/MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf

$ ffmpeg -i MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf out.wav
ffmpeg version N-81774-g64545dd Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55. 30.100 / 55. 30.100
  libavcodec     57. 58.100 / 57. 58.100
  libavformat    57. 50.100 / 57. 50.100
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 63.100 /  6. 63.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[mxf @ 0x293c3a0] OPAtom misinterpreted as OP1a? KLV for edit unit 0 extending into next edit unit is 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.
[mxf @ 0x293c3a0] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, mxf, from 'MERIDIAN_SHR_C_EN-XX_US-NR_51_LTRT_UHD_20160909_OV_01_EN_20_B.mxf':
  Metadata:
    application_platform: Microsoft Windows 7 Professional Service Pack 1 (Build 7601)
    uid             : cdb57bb1-4f06-4c77-64e4-c0293f2c7163
    generation_uid  : 2d654592-04ae-4cf4-35ab-69707dc0de3a
    company_name    : Colorfront
    product_name    : Transkoder
    product_uid     : 3a4fe380-0d01-11e4-869f-3cd92b5c1dfc
    product_version : 2.7.3.20150121
    modification_date: 2016-09-09T11:29:38.000000Z
    material_package_umid: 0x060A2B340101010501010F20130000002C3E49A0BD1548ACD65D70EA10FAEB6B
    timecode        : 00:00:17;12
  Duration: 00:11:58.92, start: 0.000000, bitrate: 2304 kb/s
    Stream #0:0: Audio: pcm_s24le, 48000 Hz, stereo, s32 (24 bit), 2304 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010F2013000000AE2179BC93E24600C47112668D9E21FB
Output #0, wav, to 'out.wav':
  Metadata:
    application_platform: Microsoft Windows 7 Professional Service Pack 1 (Build 7601)
    uid             : cdb57bb1-4f06-4c77-64e4-c0293f2c7163
    generation_uid  : 2d654592-04ae-4cf4-35ab-69707dc0de3a
    company_name    : Colorfront
    product_name    : Transkoder
    product_uid     : 3a4fe380-0d01-11e4-869f-3cd92b5c1dfc
    product_version : 2.7.3.20150121
    modification_date: 2016-09-09T11:29:38.000000Z
    material_package_umid: 0x060A2B340101010501010F20130000002C3E49A0BD1548ACD65D70EA10FAEB6B
    ISMP            : 00:00:17;12
    ISFT            : Lavf57.50.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010F2013000000AE2179BC93E24600C47112668D9E21FB
      encoder         : Lavc57.58.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)

          #1670: -filter_complex doesn't handle well finite video + infinite audio   

Given a 60 seconds video stream:

./ffmpeg -v 0 -f lavfi -i testsrc=d=60 -y video.mp4

and a 5 seconds audio stream:

./ffmpeg -v 0 -f lavfi -i 'aevalsrc=sin(440*2*PI*t)::d=5' -y audio.wav

We can make a video presentation of 60 seconds with the 5 seconds audio padded with silence with the following command line:

./ffmpeg -f lavfi -i 'movie=video.mp4 [out0]; amovie=audio.wav [audio]; aevalsrc=0 [silence]; [audio][silence] concat=v=0:a=1 [out1]' -y out.avi

Unfortunately, this doesn't work with the -filter_complex option:

% ./ffmpeg -i video.mp4 -i audio.wav -filter_complex 'aevalsrc=0 [silence]; [1:0][silence] concat=v=0:a=1 [padded_audio]' -map 0:0 -map '[padded_audio]' -y out.avi
ffmpeg version N-43791-g2c7a622 Copyright (c) 2000-2012 the FFmpeg developers
  built on Aug 21 2012 21:01:23 with gcc 4.7.1 (GCC) 20120721 (prerelease)
  configuration: --enable-gpl --enable-fontconfig --enable-libfreetype --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-x11grab --enable-libopenjpeg --enable-libass --enable-libmodplug --enable-libv4l2 --cc='ccache colorgcc' --samples=/home/ubitux/fate-samples --prefix=/tmp/ffinstall
  libavutil      51. 70.100 / 51. 70.100
  libavcodec     54. 54.100 / 54. 54.100
  libavformat    54. 25.104 / 54. 25.104
  libavdevice    54.  2.100 / 54.  2.100
  libavfilter     3. 13.100 /  3. 13.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.25.104
  Duration: 00:01:00.00, start: 0.000000, bitrate: 37 kb/s
    Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 320x240 [SAR 1:1 DAR 4:3], 34 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
    Metadata:
      handler_name    : VideoHandler
Guessed Channel Layout for  Input Stream #1.0 : mono
Input #1, wav, from 'audio.wav':
  Duration: 00:00:05.01, bitrate: 705 kb/s
    Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Output #0, avi, to 'out.avi':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    ISFT            : Lavf54.25.104
    Stream #0:0(und): Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 25 tbn, 25 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, mono, s16
Stream mapping:
  Stream #1:0 (pcm_s16le) -> concat:in0:a0 (graph 0)
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  concat (graph 0) -> Stream #0:1 (libmp3lame)
Press [q] to stop, [?] for help
frame= 1500 fps=177 q=9.6 Lsize=   13088kB time=00:01:00.00 bitrate=1786.9kbits/s

This creates some kind of infinite loop, with fps slightly decreasing over time (+memory leak)


          #2120: mmf files generated by ffmpeg does not play with yamaha software player   

Currently FFmpeg creates mmf files which are uplayable by yamaha software.

"ma2_mono_8khz.mmf" is a file created with yamaham software. What I can notice comparing this file and output from FFmpeg is that yamaha file have additional AspI chunk, so maybe the problem is that FFmpeg doesn not write it.

Mobile Contents Player (MCP-MA7) Ver.1.6.2

http://smaf-yamaha.com/tools/downloads.html

C:\>ffmpeg -i test.wav -ac 1 -ar 8000 ffmpeg_ma2_mono_8khz.mmf
ffmpeg version N-48610-gb23aff6 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jan  8 2013 02:37:29 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      52. 13.100 / 52. 13.100
  libavcodec     54. 86.100 / 54. 86.100
  libavformat    54. 59.106 / 54. 59.106
  libavdevice    54.  3.102 / 54.  3.102
  libavfilter     3. 32.100 /  3. 32.100
  libswscale      2.  1.103 /  2.  1.103
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[wav @ 0x1efea60] max_analyze_duration 5000000 reached at 5015510
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'test.wav':
  Duration: 00:00:30.01, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Output #0, mmf, to 'ffmpeg_ma2_mono_8khz.mmf':
  Metadata:
    encoder         : Lavf54.59.106
    Stream #0:0: Audio: adpcm_yamaha, 8000 Hz, mono, s16, 32 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> adpcm_yamaha)
Press [q] to stop, [?] for help
size=     118kB time=00:00:30.20 bitrate=  32.0kbits/s
video:0kB audio:118kB subtitle:0 global headers:0kB muxing overhead 0.076139%

          #2283: Slow counter / incorrect duration for aac-in-flv   

(videolan ticket 8195) I will upload an audio-only flv sample that can be decoded but the shown duration is wrong, both the duration shown before decoding and the duration shown after decoding. The time counter is wrong for both ffplay and MPlayer, seeking works fine with MPlayer, fails with ffplay.

$ ffmpeg -i sky_the80s_aacplus.flv out.wav
ffmpeg version N-50086-gd41efc1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[flv @ 0x32bdfc0] max_analyze_duration 5000000 reached at 5014000 microseconds
Input #0, flv, from 'sky_the80s_aacplus.flv':
  Duration: 00:01:08.77, start: 0.000000, bitrate: 74 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf54.63.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[flv @ 0x32bdfc0] Stream discovered after head already parsed
[flv @ 0x32bdfc0] New data stream 0:1 at pos:254270 and DTS:27.274s
size=   21168kB time=00:01:08.81 bitrate=2519.9kbits/s
video:0kB audio:21168kB subtitle:0 global headers:0kB muxing overhead 0.000369%

Duration and time claim that the sample is ~68 seconds, but it actually is over two minutes long:

$ ffmpeg -i out.wav -f null -
ffmpeg version N-50086-gd41efc1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[wav @ 0x1c49fc0] max_analyze_duration 5000000 reached at 5015510 microseconds
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'out.wav':
  Metadata:
    encoder         : Lavf54.63.100
  Duration: 00:02:02.88, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf54.63.100
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=N/A time=00:02:02.88 bitrate=N/A
video:0kB audio:21168kB subtitle:0 global headers:0kB muxing overhead -100.000101%

The same can be shown with ffplay:

$ time ffplay sky_the80s_aacplus.flv
ffplay version N-50086-gd41efc1 Copyright (c) 2003-2013 the FFmpeg developers
  built on Feb 19 2013 09:36:55 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 17.102 / 52. 17.102
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.100 / 54. 63.100
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 38.103 /  3. 38.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[flv @ 0x7f7d6c0008c0] max_analyze_duration 5000000 reached at 5014000 microseconds
Input #0, flv, from 'sky_the80s_aacplus.flv':
  Duration: 00:01:08.77, start: 0.000000, bitrate: 74 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp
  23.20 A-V:  0.000 fd=   0 aq=    1KB vq=    0KB sq=    0B f=0/0
real    0m43.481s
user    0m0.937s
sys     0m0.474s

          #2504: Audio glitches and distortion when recording alsa   

Summary of the bug: I try to record from integrated audio card input line/micro and get distortions. How to reproduce:

$ ffmpeg -debug 1 -vn -f alsa -ac 2 -channel_layout stereo -i hw:0,0 -c:a pcm_s16le /Store3/Test/pcm_`date +%m%d_%H%M`.wav
ffmpeg version N-52061-g8137ea3 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 13 2013 21:10:51 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
  configuration: --prefix=/opt/ffmpeg-opt/usr --enable-shared --bindir=/opt/ffmpeg-opt/usr/bin --datadir=/opt/ffmpeg-opt/usr/share/ffmpeg-opt --incdir=/opt/ffmpeg-opt/usr/include/ffmpeg-opt --libdir=/opt/ffmpeg-opt/usr/lib64 --mandir=/opt/ffmpeg-opt/usr/share/man --shlibdir=/opt/ffmpeg-opt/usr/lib64 --arch=x86_64 --optflags='-O2 -g' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-libass --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --disable-libv4l2 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --build-suffix=1 --disable-doc --enable-runtime-cpudetect
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 53.101 /  3. 53.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
 matched as AVOption 'debug' with argument '1'.
Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-channel_layout' ... matched as option 'channel_layout' (set channel layout) with argument 'stereo'.
Reading option '-i' ... matched as input file with argument 'hw:0,0'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_s16le'.
Reading option '/Store3/Test/pcm_0426_1018.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Successfully parsed a group of options.
Parsing a group of options: input file hw:0,0.
Applying option vn (disable video) with argument 1.
Applying option f (force format) with argument alsa.
Applying option ac (set number of audio channels) with argument 2.
Applying option channel_layout (set channel layout) with argument stereo.
Successfully parsed a group of options.
Opening an input file: hw:0,0.
[alsa @ 0x20f7720] All info found
Input #0, alsa, from 'hw:0,0':
  Duration: N/A, start: 1366967936.157276, bitrate: 1536 kb/s
    Stream #0:0, 1, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file /Store3/Test/pcm_0426_1018.wav.
Applying option c:a (codec name) with argument pcm_s16le.
Successfully parsed a group of options.
Opening an output file: /Store3/Test/pcm_0426_1018.wav.
Successfully opened the file.
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0x20e8280] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0x20e8280] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x20effc0] Setting 'sample_fmts' to value 's16'
[AVFilterGraph @ 0x20f7320] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, wav, to '/Store3/Test/pcm_0426_1018.wav':
  Metadata:
    ISFT            : Lavf55.2.100
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=    1076kB time=00:00:05.73 bitrate=1536.1kbits/s
video:0kB audio:1076kB subtitle:0 global headers:0kB muxing overhead 0.007263%
31021 frames successfully decoded, 0 decoding errors
Statistics: 4 seeks, 31024 writeouts

Result: very bad audio, distortion Same result with

$ ffmpeg -debug 1 -vn -f alsa -ac 2 -channel_layout stereo -i hw:0,0 -ar 44100 -ab 320k -c:a mp3 /Store3/Test/mp3_`date +%m%d_%H%M`.mp3

Result: very bad audio, distortion

$ ffplay -debug 1 -f alsa -ac 2 -i hw:0,0

Result: statics after almost every bip.

More tests with other applications: Audacity With Alsa, input: default then pulse then ALC892 (hw:0,0) Result: with these 3 different parameters, audio is perfect.

recordmydesktop input: DEFAULT result: audio is perfect.

gnome-sound-recorder 2.29.91 Record from input: Capture format flac, ogg, pm3 Result: with these 3 different formats, audio is perfect.

arecord arecord -f cd -t wav out.wav Result: audio is perfect.

VLC input hw:0,0 Result: static after every bip.


          #3433: using "shortest" results in mismatched audio and video lengths   

Summary of the bug:

"-shortest" does not add trim/atrim filters, so the length of each output stream will be a multiple of sample size. In the given example, the video is exactly 1s, but the resulting output is 1.02s. If "-t 1.000" is passed instead of "-shortest", the resulting output is exactly 1s.

The difference is that using "-t" adds trim/atrim filters, so the last frame->nb_samples encoded will be truncated to fit the given duration.

How to reproduce:

% ffmpeg -y -probesize 500000 -f image2 -r 25 -i colorbars.%04d.png -i 100Hz_44100Hz_16bit_05sec.wav -shortest -c:v png -c:a pcm_s16le -q:v 1 -2 -f mov colorbars.mov
Note that input streams have durations of 5s and 1s.
% ffprobe colorbars.mov
Note that the output stream has a duration of 1.02s.
ffmpeg version N-60960-gf3eef02
built on Feb 28 2014 10:31:20

          #3455: aresample=matrix_encoding=dplii doesn't downmix backleft and center correctly   

Summary of the bug: DPLII files have too-loud right channel. Seems to be some problem with the interaction of back left and center channel How to reproduce:

ffmpeg -i surround_blbrc.wav -af "aresample=matrix_encoding=dplii" -ac 2 dplii_blbrc.wav
ffmpeg version git-2014-03-09-8dfd0e6
built on Mar  9 2014 22:53:26 with gcc 4.8.2 (Gentoo 4.8.2 p1.3r1, pie-0.5.8r1)

You can see the problem easiest by crafting a 5.1 surround wave file with empty channels except the center channel with noise and the two back channels, first half of the back left with noise and the second half of the back right with noise. However, all surround files that use the bl and center channels at the same time will be too loud in the resulting file's right channel.

Can provide example files if you like.


          #3853: silence in front of generated audio mpegts   

Let a.ts be an mpegts with aac encoded audio only. Then

/ffmpeg/ffmpeg -i a.ts -acodec aac -strict -2 -loglevel debug audio.ts

Input #0, aac, from 'a.ts': Duration: 00:00:05.06, bitrate: 126 kb/s Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 126 kb/s

if I convert a.ts and audio.ts into .wav I will get two wav files, one of them has silence in front of it. I used audacity to determine that about 0.031s audio.ts will have some silence. I can see that a.ts has 215 audio frames and audio.ts 216.

Any explanations why is silence added to ts? As far as I can see it could be a bug. Summary: the command at top introduces one extra frame to the start of aac stream, making aac longer as it should.


          #3887: ffmpeg cant multiplex IMX video essence (m2v) with correct vbv into mov or mxf   

Summary of the bug: How to reproduce:

% ffmpeg -i <file>  -vcodec mpeg2video -pix_fmt yuv422p -b:v 30000k -maxrate 30000k -minrate 30000k -g 1 -flags +ildct+low_delay -dc 10 -intra_vlc 1 -non_linear_quant 1 -qmin 1 -qmax 8 -q:v 1 -ps 1 -top 1 -aspect 16:9 -bufsize 9437184 -rc_init_occupancy 9437184 -rc_buf_aggressivity 0.25 -an <intermediate.m2v>
ffmpeg -i <intermediate.m2v> -i stereo.wav -vcodec copy -r 25 -acodec copy -vtag mx3p <final.mov>
ffmpeg -i <intermediate.m2v> -i stereo.wav -vcodec copy -r 25 -acodec copy -d10_channelcount 4 -f mxf_d10 <final.mxf>
mov file shows complete green field (like video error). mxf file shows "[mxf_d10 @ 04714260] cannot fill d-10 video packet" errors during mpxing, and final video has many video errors.
vbv = 9437184 is correct value for MPEG2 422P@ML, and e.g. Telestream encoder will correctly generate mxf at least with this vbv.
Apple's own IMX.mov encoding shows vbv 1835008, which is actually MPEG2 MP@ML vbv, meant for max bitrate of 15Mbit/sec
Maximum vbv before error appears to be 1200000 for IMX30 and 2000000 for IMX50, assuming -rc_init_occupancy is set to same value.
ffmpeg version git 1aa153d
built on … Aug 23rd 2014

Thanks you very much for looking at this.


          #4160: problem encoding opus file with unusual channel mask   

Summary of the bug: trying to encode opus audio file with unusual channel mask (for example FL+FR+BL) generates an error. How to reproduce: ffmpeg -i 3flfrbl.wav 3flfrbl.opus

% ffmpeg -i 3flfrbl.wav 3flfrbl.opus
ffmpeg version 2.5
built on December 3rd 2014

          #4692: opus + webm - inconsistent meta?   
ffmpeg -i src.wav -ar 48000 -af atrim=start_sample=24000*0:end_sample=240000*1 -c:a opus 0.webm

I'm a bit confused by ffmpeg while encoding opus into webm. I have a wav file with exactly 240000 samples @ 48kHz. The resulting file gives me the following information:

Duration: 5.01, Start 0.007 discard 648/900 samples 240312 samples decoded.

If I have several of these files how would I play them seamlessly without gaps?

i.e. in a browser I've tried:

sourceBuffer.timestampOffset = 5 * n - 0.007;
sourceBuffer.appendWindowStart = 5 * n;
sourceBuffer.appendWindowEnd = 5 * (n+1);
sourceBuffer.appendBuffer(new Uint8Array(buffer[n]));

However, there are audible gaps.

How many samples am I actually supposed to discard? 0.007 * 48000, 648, or 240312 - 240000?

ffprobe -i 0.webm
ffprobe version 2.6.3 Copyright (c) 2007-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.53) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libass --enable-ffplay --enable-libfdk-aac --enable-openssl --enable-libopus --enable-libwebp --enable-nonfree --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, matroska,webm, from '0.webm':
  Metadata:
    encoder         : Lavf56.25.101
  Duration: 00:00:05.01, start: 0.007000, bitrate: 94 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp (default)
h59ec2765:sintel ronagy$ ffmpeg -loglevel debug -i 1.webm asd.wav
ffmpeg version 2.6.3 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.53) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libass --enable-ffplay --enable-libfdk-aac --enable-openssl --enable-libopus --enable-libwebp --enable-nonfree --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  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 '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-i' ... matched as input file with argument '1.webm'.
Reading option 'asd.wav' ... 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 1.webm.
Successfully parsed a group of options.
Opening an input file: 1.webm.
[matroska,webm @ 0x7fbfa300da00] Format matroska,webm probed with size=2048 and score=100
st:0 removing common factor 1000000 from timebase
[matroska,webm @ 0x7fbfa300da00] Before avformat_find_stream_info() pos: 473 bytes read:32768 seeks:0
[matroska,webm @ 0x7fbfa300da00] All info found
[matroska,webm @ 0x7fbfa300da00] After avformat_find_stream_info() pos: 942 bytes read:32768 seeks:0 frames:1
Input #0, matroska,webm, from '1.webm':
  Metadata:
    encoder         : Lavf56.25.101
  Duration: 00:00:05.01, start: 0.007000, bitrate: 94 kb/s
    Stream #0:0, 1, 1/1000: Audio: opus, 48000 Hz, stereo, fltp (default)
Successfully opened the file.
Parsing a group of options: output file asd.wav.
Successfully parsed a group of options.
Opening an output file: asd.wav.
File 'asd.wav' already exists. Overwrite ? [y/N] y
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0x7fbfa42003a0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x7fbfa42008a0] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x7fbfa42008a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x7fbfa4000160] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x7fbfa4201060] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
Output #0, wav, to 'asd.wav':
  Metadata:
    ISFT            : Lavf56.25.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s (default)
    Metadata:
      encoder         : Lavc56.26.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[opus @ 0x7fbfa3011400] skip 0 samples due to side data
[opus @ 0x7fbfa3011400] discard 648/960 samples
[output stream 0:0 @ 0x7fbfa4200620] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=     939kB time=00:00:05.00 bitrate=1536.0kbits/s
video:0kB audio:939kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.008114%
Input file #0 (1.webm):
  Input stream #0:0 (audio): 251 packets read (57217 bytes); 251 frames decoded (240312 samples);
  Total: 251 packets (57217 bytes) demuxed
Output file #0 (asd.wav):
  Output stream #0:0 (audio): 251 frames encoded (240312 samples); 251 packets muxed (961248 bytes);
  Total: 251 packets (961248 bytes) muxed
251 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x7fbfa2f00000] Statistics: 4 seeks, 254 writeouts
[AVIOContext @ 0x7fbfa2c357c0] Statistics: 59467 bytes read, 0 seeks

          #5039: Incorrect TrueHD decoding   

http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/203994 A user uploaded a TrueHD sample that is supposed to only contain 0xFFFFFF but the two first channels of the output file always contain 0x01 or 0x02.

$ ffmpeg -i warnings_cut.thd -acodec pcm_s24le out.wav
ffmpeg version N-76844-g3885ef0 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55.  9.100 / 55.  9.100
  libavcodec     57. 16.100 / 57. 16.100
  libavformat    57. 19.100 / 57. 19.100
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 15.100 /  6. 15.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, truehd, from 'warnings_cut.thd':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32 (24 bit)
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.19.100
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32 (24 bit), 9216 kb/s
    Metadata:
      encoder         : Lavc57.16.100 pcm_s24le
Stream mapping:
  Stream #0:0 -> #0:0 (truehd (native) -> pcm_s24le (native))
Press [q] to stop, [?] for help
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 05.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0b.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 08.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
...
...
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0c.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 09.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0e.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0f.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 04.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 03.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0a.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 01.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 02.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 0a.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 09.
[truehd @ 0x2d32460] Lossless check failed - expected 00, calculated 05.
size=   56934kB time=00:00:50.60 bitrate=9216.0kbits/s
video:0kB audio:56933kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000175%

          #5053: Cannot select sample rate for the link between filters audio format   

Summary of the bug:

I upgraded from 2.8 to 2.8.3 and my project no longer worked.

Query

ffmpeg -y \
-ss 0 -i testdata/video.webm \
-ss 0 -i backingtrack.mp3 \
-ss 0 -i testdata/audio.wav \
-i overlay.png \
-i suffix.mp4 \
-filter_complex \
\
        "[1:a][2:a] amix=inputs=2:duration=shortest [mergedAudio]; \
        [mergedAudio] asplit=3 [fullAudio][a1][a2]; \
        [a1] atrim=0:2 [vineAudio]; \
        [a2] atrim=0:4 [instagramAudio]; \
        [0]setpts=PTS-STARTPTS[video]; \
        [video][3]overlay[overlayed]; \
        [4]setpts=PTS-STARTPTS[suffix]; \
        [suffix] split=3 [suffix1][suffix2][suffix3]; \
        [overlayed] split=3 [fullIn][instagramIn][vineIn]; \
        [vineIn] trim=0:2 [vineOut]; \
        [instagramIn] trim=0:4 [instagramOut]; \
        [fullIn][suffix1] concat=n=2:v=1:a=0 [full]; \
        [vineOut][suffix2] concat=n=2:v=1:a=0 [vine]; \
        [instagramOut][suffix3] concat=n=2:v=1:a=0 [instagram]" \
\
-acodec libvorbis -b:a 128k -ac 2 -r 25 -vcodec libvpx -b:v 1024k \
-map [full] -map [fullAudio] testdata/output/full.webm \
-map [instagram] -map [instagramAudio] testdata/output/instagram.webm \
-map [vine] -map [vineAudio] testdata/output/vine.webm

Output:

ffmpeg version 2.8.3-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-26) 20151125
  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 --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Input #0, matroska,webm, from 'testdata/video.webm':
  Metadata:
    encoder         : whammy
  Duration: 00:00:08.08, start: 0.000000, bitrate: 2097 kb/s
    Stream #0:0: Video: vp8, yuv420p, 480x320, SAR 1:1 DAR 3:2, 1k tbr, 1k tbn, 1k tbc (default)
[mp3 @ 0xbc38740] Skipping 0 bytes of junk at 4096.
[mp3 @ 0xbc38740] Estimating duration from bitrate, this may be inaccurate
Input #1, mp3, from 'backingtrack.mp3':
  Metadata:
    TSS             : Logic 10.2.0
  Duration: 00:00:08.07, start: 0.000000, bitrate: 164 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 160 kb/s
Guessed Channel Layout for  Input Stream #2.0 : stereo
Input #2, wav, from 'testdata/audio.wav':
  Duration: 00:00:08.92, bitrate: 1411 kb/s
    Stream #2:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Input #3, png_pipe, from 'overlay.png':
  Duration: N/A, bitrate: N/A
    Stream #3:0: Video: png, rgba(pc), 480x320 [SAR 2835:2835 DAR 3:2], 25 tbr, 25 tbn, 25 tbc
Input #4, mov,mp4,m4a,3gp,3g2,mj2, from 'suffix.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.33.100
  Duration: 00:00:01.57, start: 0.000000, bitrate: 117 kb/s
    Stream #4:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 480x320 [SAR 1:1 DAR 3:2], 111 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
[audio format for output stream 0:1 @ 0xbc74960] Cannot select sample rate for the link between filters audio format for output stream 0:1 and output stream 0:1.
Error configuring complex filters.
Invalid argument

Also tested with git build g6b978da built on 20151130, same result.

It works if I add a "aformat=sample_rates=44100" step in the audio filter chain, but this was not needed before.


          #5075: Duration of M4A file is wrong?   

I think duration is an issue after all. I attached files to demonstrate it.

8.wav and 9.wav are original ones. There should not be any gap in between (those are small pieces of tracks 8 and 9 from Depeche Mode Violator).

When I concatenate files 8.wav and 9.wav and then open resulting 10w.wav in Audacity I see that there's no gap in between, so this is OK. It sound fine as well.

ffmpeg -f concat -i ccw.txt  10w.wav

8.m4a and 9.m4a are ones created with FFmpeg. When I concatenate them like this and then open resulting 10.wav in Audacity I see that there is a gap in between. That can be heard as well.

ffmpeg -f concat -i cc.txt  10.wav

          #5091: Icecast SSL support (icecast:// is http-only)   

Summary of the bug: icecast:// does not work if server is running with https How to reproduce: Output icecast:// to an icecast server running https

% ffmpeg -i song.wav -acodec libfdk_aac -vbr 4 -content_type "audio/aac" -f adts icecast://source:hackme@exampleserver.com:8000/stream.aac
ffmpeg version 2.8.3
built on ... Linux, today

Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker.


          #5179: Native AAC encoder does not respect channelmap assignments   

Tried to generate a 4-channel AAC stream with quad(side) layout, but the output MP4 reports stream layout as 4.0. WAV is generated with correct layout. Using Zeranoe static binary.


Generation of dummy stereo source.

ffmpeg -f lavfi -i "aevalsrc=0|0:d=5" null0.mp4 -report 2> null0.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (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-libdcadec --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-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, lavfi, from 'aevalsrc=0|0:d=5':
  Duration: N/A, start: 0.000000, bitrate: 5644 kb/s
    Stream #0:0: Audio: pcm_f64le, 44100 Hz, stereo, dbl, 5644 kb/s
Output #0, mp4, to 'null0.mp4':
  Metadata:
    encoder         : Lavf57.21.101
    Stream #0:0: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.21.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_f64le (native) -> aac (native))
Press [q] to stop, [?] for help
size=       3kB time=00:00:05.01 bitrate=   4.7kbits/s speed=79.6x
video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 122.289612%
[aac @ 00000000003ad8c0] Qavg: 65536.000

Generation of 4 channel AAC in MP4

ffmpeg -i null0.mp4 -lavfi "[0]asplit[a][b];[a][b]amerge[o];[o]channelmap=0|1|2|3:quad(side)[out]" -map "[out]" null1.mp4 2> null1.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (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-libdcadec --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-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null0.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, start: 0.023220, bitrate: 4 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
[Parsed_amerge_1 @ 00000000029a3cc0] No channel layout for input 1
[Parsed_amerge_1 @ 00000000029a3cc0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, mp4, to 'null1.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
    Stream #0:0: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, quad(side), fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc57.21.100 aac
Stream mapping:
  Stream #0:0 (aac) -> asplit
  channelmap -> Stream #0:0 (aac)
Press [q] to stop, [?] for help
size=       4kB time=00:00:05.03 bitrate=   7.1kbits/s speed=46.7x
video:0kB audio:3kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 56.716942%
[aac @ 00000000029a22c0] Qavg: 65536.000
ffprobe null1.mp4
=================
ffprobe version N-77715-gfc703f5 Copyright (c) 2007-2016 the FFmpeg developers
  built with gcc 5.2.0 (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-libdcadec --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-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null1.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.06, start: 0.023220, bitrate: 7 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, 4.0, fltp, 4 kb/s (default)
    Metadata:
      handler_name    : SoundHandler

Generation of 4-channel WAV

ffmpeg -i null0.mp4 -lavfi "[0]asplit[a][b];[a][b]amerge[o];[o]channelmap=0|1|2|3:quad(side)[out]" -map [out] null1.wav 2> null1wav.log
ffmpeg version N-77715-gfc703f5 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.2.0 (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-libdcadec --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-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'null0.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, start: 0.023220, bitrate: 4 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
[Parsed_amerge_1 @ 0000000002613c20] No channel layout for input 1
[Parsed_amerge_1 @ 0000000002613c20] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, wav, to 'null1.wav':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    ISFT            : Lavf57.21.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, quad(side), s16, 2822 kb/s (default)
    Metadata:
      encoder         : Lavc57.21.100 pcm_s16le
Stream mapping:
  Stream #0:0 (aac) -> asplit
  channelmap -> Stream #0:0 (pcm_s16le)
Press [q] to stop, [?] for help
size=    1736kB time=00:00:05.03 bitrate=2822.6kbits/s speed= 157x
video:0kB audio:1736kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.005738%
ffprobe null1.wav
=================
ffprobe version N-77715-gfc703f5 Copyright (c) 2007-2016 the FFmpeg developers
  built with gcc 5.2.0 (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-libdcadec --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-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 12.100 / 55. 12.100
  libavcodec     57. 21.100 / 57. 21.100
  libavformat    57. 21.101 / 57. 21.101
  libavdevice    57.  0.100 / 57.  0.100
  libavfilter     6. 23.100 /  6. 23.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, wav, from 'null1.wav':
  Metadata:
    encoder         : Lavf57.21.101
  Duration: 00:00:05.04, bitrate: 2822 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, quad(side), s16, 2822 kb/s

          #5273: adpcm_swf: 11khz mode decodes as 5khz   

I don't know if this is a bug in ffmpeg or in encoding software.

C:\>ffmpeg -i 11khz_mono_44kbps.flv out.wav
ffmpeg version N-78410-g588e2e3 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 17.100 / 55. 17.100
  libavcodec     57. 24.102 / 57. 24.102
  libavformat    57. 25.100 / 57. 25.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 31.100 /  6. 31.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, flv, from '11khz_mono_44kbps.flv':
  Metadata:
    creationdate    : Thu Feb 25 04:13:06 2016
    Encoded_With    : Sorenson Squeeze
    Encoded_By      :
  Duration: 00:00:02.43, start: 0.000000, bitrate: 183 kb/s
    Stream #0:0: Audio: adpcm_swf, 5512 Hz, mono, s16, 22 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    creationdate    : Thu Feb 25 04:13:06 2016
    Encoded_With    : Sorenson Squeeze
    ITCH            :
    ISFT            : Lavf57.25.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 5512 Hz, mono, s16, 8
8 kb/s
    Metadata:
      encoder         : Lavc57.24.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_swf (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=     208kB time=00:00:09.74 bitrate= 175.0kbits/s speed=N/A
video:0kB audio:208kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.036584%

          #5457: mp3 delay issue   

This issue is reproduced with

ffmpeg version N-79546-g13406b6 Copyright (c) 2000-2016 the FFmpeg developers

Description: Run below commands to generate a.wav and b.wav

ffmpeg.exe -i big50.mp4 -acodec libmp3lame a.mp3
ffmpeg.exe -i a.mp3 a.wav
ffmpeg.exe -i big50.mp4 -vcodec libx264 -acodec libmp3lame b.avi
ffmpeg.exe -i b.avi b.wav

Open a.wav and b.wav in wav file editor you can see there's 50 ms delay in b.wav. (sample rate is 22050, so that 50 ms should be delay introduced by LAME that are don't properly handled by ffmpeg?)

Source mp4 is here: https://dl.dropboxusercontent.com/u/89678527/big50.mp4 and generated wav files: https://dl.dropboxusercontent.com/u/89678527/a.wav https://dl.dropboxusercontent.com/u/89678527/b.wav

here's comparison of a.wav and b.wav https://dl.dropboxusercontent.com/u/89678527/mp3.PNG


          #5481: g723 in au: fix decoding   

decodes ok with sox

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g723_44khz_3bit.au

C:\>ffmpegx -i C:\g723_44khz_3bit.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, au, from 'C:\g723_44khz_3bit.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 132 kb/s
    Stream #0:0: Audio: adpcm_g726le ([25][0][0][0] / 0x0019), 44100 Hz, mono, s
16, 132 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[wav @ 0x2c3eb60] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.103
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.38.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_g726le (g726le) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[g726le @ 0x2c3e6a0] Frame invalidly split, missing parser?
    Last message repeated 484 times
size=    2585kB time=00:00:30.00 bitrate= 705.6kbits/s speed=66.2x
video:0kB audio:2585kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.002947%

          #5725: ffmpeg: error during decoding mp3 with APE tag   

FFmpeg tool as well as certain applications have problems decoding MP3 files with APE tags. The tag is reported as an error while decoding stream and some applications using ffmpeg libraries abort decoding in such case.

Issue is also reproducible in ffmpeg 2.7.7 and probably in all releases up to date.

ffmpeg -loglevel trace -i Ice\ Wharf.mp3 Ice\ Wharf.wav
ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
  libavutil      55. 28.100 / 55. 28.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 41.100 / 57. 41.100
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 47.100 /  6. 47.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'trace'.
Reading option '-i' ... matched as input file with argument 'Ice Wharf.mp3'.
Reading option 'Ice Wharf.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument trace.
Successfully parsed a group of options.
Parsing a group of options: input file Ice Wharf.mp3.
Successfully parsed a group of options.
Opening an input file: Ice Wharf.mp3.
[file @ 0xa9af6e0] Setting default whitelist 'file,crypto'
Probing aac score:1 size:1875
Probing mp3 score:25 size:1875
Probing aac score:1 size:3923
Probing mp3 score:51 size:3923
[mp3 @ 0xa9aef60] Format mp3 probed with size=4096 and score=51
[mp3 @ 0xa9aef60] id3v2 ver:3 flags:00 len:163
[mp3 @ 0xa9aef60] pad 576 1681
[mp3 @ 0xa9aef60] Skipping 0 bytes of junk at 590.
[mp3 @ 0xa9aef60] Before avformat_find_stream_info() pos: 590 bytes read:32768 seeks:0 nb_streams:1
[mp3 @ 0xa9aef60] demuxer injecting skip 1105 / discard 0
[mp3 @ 0xa9b01e0] skip 1105 / discard 0 samples due to side data
[mp3 @ 0xa9b01e0] skip 1105/1152 samples
[mp3 @ 0xa9aef60] All info found
[mp3 @ 0xa9aef60] stream 0: start_time: 0.025 duration: 238.341
[mp3 @ 0xa9aef60] format: start_time: 0.025 duration: 238.341 bitrate=128 kb/s
[mp3 @ 0xa9aef60] After avformat_find_stream_info() pos: 22094 bytes read:32768 seeks:0 frames:50
Input #0, mp3, from 'Ice Wharf.mp3':
  Metadata:
    title           : Ice Wharf
    artist          : Work Drugs
    album           : Aurora Lies
    track           : 6
  Duration: 00:03:58.34, start: 0.025057, bitrate: 128 kb/s
    Stream #0:0, 50, 1/14112000: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
    Metadata:
      encoder         : LAME3.96r
    Side data:
      replaygain: track gain - -6.200000, track peak - unknown, album gain - unknown, album peak - unknown,
Successfully opened the file.
Parsing a group of options: output file Ice Wharf.wav.
Successfully parsed a group of options.
Opening an output file: Ice Wharf.wav.
[file @ 0xa9bfd40] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'sample_fmt' to value 's16p'
[graph 0 input from stream 0:0 @ 0xa9b2880] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0xa9b2880] tb:1/44100 samplefmt:s16p samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0xa9c0820] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0xa9c0820] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0xa9b2800] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0xa9c1e80] [SWR @ 0xa9db240] Using s16p internally between filters
[auto-inserted resampler 0 @ 0xa9c1e80] ch:2 chl:stereo fmt:s16p r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz
[wav @ 0xa9b16c0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'Ice Wharf.wav':
  Metadata:
    INAM            : Ice Wharf
    IART            : Work Drugs
    IPRD            : Aurora Lies
    IPRT            : 6
    ISFT            : Lavf57.41.100
    Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.48.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp3 @ 0xa9b0720] skip 1105 / discard 0 samples due to side data
[mp3 @ 0xa9b0720] skip 1105/1152 samples
[mp3 @ 0xa9aef60] demuxer injecting skip 0 / discard 1152eed= 242x
[mp3 @ 0xa9b0720] skip 0 / discard 1152 samples due to side data
[mp3 @ 0xa9b0720] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
[output stream 0:0 @ 0xa9c0b80] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=   41049kB time=00:03:58.29 bitrate=1411.2kbits/s speed= 245x
video:0kB audio:41049kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000347%
Input file #0 (Ice Wharf.mp3):
  Input stream #0:0 (audio): 9125 packets read (3813711 bytes); 9123 frames decoded (10508591 samples);
  Total: 9125 packets (3813711 bytes) demuxed
Output file #0 (Ice Wharf.wav):
  Output stream #0:0 (audio): 9123 frames encoded (10508591 samples); 9123 packets muxed (42034364 bytes);
  Total: 9123 packets (42034364 bytes) muxed
9123 frames successfully decoded, 1 decoding errors
[AVIOContext @ 0xa9b26c0] Statistics: 4 seeks, 9126 writeouts
[AVIOContext @ 0xa9b7880] Statistics: 3814255 bytes read, 0 seeks

          #5820: WMA Lossless 24-bit Distortions via RTSP   

Summary of the bug: When trans-coding an RTSP stream with WMA Lossless 24-bit audio to a WAV file, ffmpeg produces many audible artifacts. Stream is publicly available for testing: rtsp://djxmmx.net/rap-rnb. Please see next post for links to the output files produced by ffmpeg. How to reproduce:

ffmpeg -i rtsp://djxmmx.net/rap-rnb C:\Temp\Out-RTSP.wav
ffmpeg version N-81516-gbe07c25 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --ena
ble-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --ena
ble-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --e
nable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-lib
openh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-lib
soxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvo
rbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-li
bxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 54.101 / 57. 54.101
  libavformat    57. 48.101 / 57. 48.101
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 58.100 /  6. 58.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[wmalossless @ 00000000003fe4e0] Decoding audio at 24 bit-depth
Input #0, rtsp, from 'rtsp://djxmmx.net/rap-rnb':
  Metadata:
    MediaFoundationVersion: 2.112
    DeviceConformanceTemplate: N2
    WM/WMADRCPeakReference: 32767
    WM/WMADRCPeakTarget: 32767
    WM/WMADRCAverageReference: 10194
    WM/WMADRCAverageTarget: 10194
    genre           : R&B
    WMFSDKVersion   : 11.0.5721.5145
    WMFSDKNeeded    : 0.0.0.0000
    album           : Thr33 Ringz
    track           : 11
    WM/Track        : 10
    WM/Year         : 2008
    PeakValue       : 32673
    AverageLevel    : 7035
    IsVBR           : 1
    title           : Long Lap Dance
    artist          : T-Pain
  Duration: 00:04:36.47, start: 199.296000, bitrate: N/A
    Stream #0:0: Audio: wmalossless (c[1][0][0] / 0x0163), 96000 Hz, stereo, s32p (24 bit), 1152 kb/s
[wmalossless @ 0000000002d04120] Decoding audio at 24 bit-depth
[wav @ 0000000002d045c0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar in
stead.
Output #0, wav, to 'C:\Temp\Out-RTSP.wav':
  Metadata:
    MediaFoundationVersion: 2.112
    DeviceConformanceTemplate: N2
    WM/WMADRCPeakReference: 32767
    WM/WMADRCPeakTarget: 32767
    WM/WMADRCAverageReference: 10194
    WM/WMADRCAverageTarget: 10194
    IGNR            : R&B
    WMFSDKVersion   : 11.0.5721.5145
    WMFSDKNeeded    : 0.0.0.0000
    IPRD            : Thr33 Ringz
    IPRT            : 11
    WM/Track        : 10
    WM/Year         : 2008
    PeakValue       : 32673
    AverageLevel    : 7035
    IsVBR           : 1
    INAM            : Long Lap Dance
    IART            : T-Pain
    ISFT            : Lavf57.48.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, stereo, s16, 3072 kb/s
    Metadata:
      encoder         : Lavc57.54.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (wmalossless (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 141 packets
[wmalossless @ 0000000002d04120] broken frame: channel len(6656) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(7680) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(5632) > samples_per_frame(4096)
[wmalossless @ 0000000002d04120] broken frame: channel len(4864) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][1] 608 > max (256), not supported
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 15 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 3 vs 6
[wmalossless @ 0000000002d04120] broken frame: channel len(5120) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Arithmetic coding is 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.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(4352) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][0] 776 > max (256), not supported
Error while decoding stream #0:0: Invalid data found when processing input
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 6 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 6 vs 8
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] broken frame: channel len(6912) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 24 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 2 vs 6
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 6 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq c vs e
[rtsp @ 000000000030a920] max delay reached. need to consume packet
[rtsp @ 000000000030a920] RTP: missed 24 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 4 vs 8
[rtsp @ 000000000030a920] RTP: missed 9 packets
[wmalossless @ 0000000002d04120] Packet loss detected! seq 8 vs a
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
[wmalossless @ 0000000002d04120] broken frame: channel len(6144) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Expect wrong output since inverse LPC filter is not implemented. Update your FFmpeg ver
sion to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been i
mplemented.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
[wmalossless @ 0000000002d04120] Expect wrong output since inverse LPC filter is not implemented. Update your FFmpeg ver
sion to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been i
mplemented.
[wmalossless @ 0000000002d04120] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/incoming/
and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
[wmalossless @ 0000000002d04120] broken frame: channel len(4608) > samples_per_frame(4096)
Error while decoding stream #0:0: Invalid data found when processing input
[wmalossless @ 0000000002d04120] Order[0][0] 968 > max (256), not supported
Error while decoding stream #0:0: Invalid data found when processing input
size=   24720kB time=00:01:08.18 bitrate=2970.1kbits/s speed=1.27x
video:0kB audio:24720kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000735%
Exiting normally, received signal 2.

          #5829: ADPCM SWF wave decode error, but directly decode *.flv right   

Summary of the bug:

ADPCM SWF wave decode error, but directly decode *.flv right

% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav % ./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav Got mono_11k_pcm.wav error.

% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn mono_11k_pcm_right.wav Got mono_11k_pcm_right.wav right.

*.flv ------> right.wav *.flv --> swf.wav --> error.wav

Also, use SWF ADPCM encode to a wav, then decode it to PCM, meet error. PCM --> swf.wav --> PCM(error)

Root cause in my opinion: SWF adpcm take block_align as once decoded frame_size. When decode *.flv, we can get the right frame_size. When decode swf.wav get the wrong frame_size.

How to reproduce:

CMD:
% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav
% ./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav
% ./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn mono_11k_pcm_right.wav
Version:
From https://git.ffmpeg.org/ffmpeg
   369ed11..ba7be8c  master     -> origin/master
   8887daf..c32ce24  release/2.8 -> origin/release/2.8
   b62191f..3512ed3  release/3.0 -> origin/release/3.0
   4275b27..677ea4a  release/3.1 -> origin/release/3.1
 * [new tag]         n3.0.3     -> n3.0.3
From https://git.ffmpeg.org/ffmpeg
 * [new tag]         n3.1.3     -> n3.1.3
Updating 369ed11..ba7be8c
Log:
ligd@bjdevel03 ffmpeg $./ffmpeg -i ~/program/swf_adpcm/mono_11k.flv -vn -acodec copy mono_11k_bs.wav
ffmpeg version N-81571-gba7be8c Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.4.5 (Debian 4.4.5-8)
  configuration: --prefix=/home/ligd/git/
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 55.101 / 57. 55.101
  libavformat    57. 48.102 / 57. 48.102
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 60.100 /  6. 60.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
[flv @ 0xa48b260] video stream discovered after head already parsed
[flv @ 0xa48b260] audio stream discovered after head already parsed
Input #0, flv, from '/home/ligd/program/swf_adpcm/mono_11k.flv':
  Duration: 00:01:00.46, start: 0.000000, bitrate: 810 kb/s
    Stream #0:0: Video: flv1, yuv420p, 640x352, 12 fps, 12 tbr, 1k tbn
    Stream #0:1: Audio: adpcm_swf, 11025 Hz, mono, s16, 44 kb/s
[wav @ 0xa4e4b00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[wav @ 0xa4e4b00] requested bits_per_coded_sample (16) and actually stored (4) differ
Output #0, wav, to 'mono_11k_bs.wav':
  Metadata:
    ISFT            : Lavf57.48.102
    Stream #0:0: Audio: adpcm_swf (FS[0][0] / 0x5346), 11025 Hz, mono, 44 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=     327kB time=00:01:00.46 bitrate=  44.3kbits/s speed=1.73e+03x
video:0kB audio:327kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.027772%
ligd@bjdevel03 ffmpeg $
ligd@bjdevel03 ffmpeg $
ligd@bjdevel03 ffmpeg $./ffmpeg -i mono_11k_bs.wav mono_11k_pcm.wav
ffmpeg version N-81571-gba7be8c Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.4.5 (Debian 4.4.5-8)
  configuration: --prefix=/home/ligd/git/
  libavutil      55. 29.100 / 55. 29.100
  libavcodec     57. 55.101 / 57. 55.101
  libavformat    57. 48.102 / 57. 48.102
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 60.100 /  6. 60.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from 'mono_11k_bs.wav':
  Metadata:
    encoder         : Lavf57.48.102
  Duration: 00:01:00.46, bitrate: 44 kb/s
    Stream #0:0: Audio: adpcm_swf (FS[0][0] / 0x5346), 11025 Hz, 1 channels, s16, 44 kb/s
[wav @ 0xb44c400] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'mono_11k_pcm.wav':
  Metadata:
    ISFT            : Lavf57.48.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, mono, s16, 176 kb/s
    Metadata:
      encoder         : Lavc57.55.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_swf (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1904kB time=00:01:28.42 bitrate= 176.4kbits/s speed=2.61e+03x
video:0kB audio:1904kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004001%

Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker.


          #6071: loudnorm filter always returns LRA of 0.0 LU with short files (less than ~3 seconds)   

I was trying to execute a 2 pass loudnorm filter on a file less than 1 second and it wouldn't work properly - the output of the second pass was not changed by much and the target output values were missed by a long way. On the first pass, the measured input_lra value was always 0.0 which seems wrong and the output_i is nowhere near the target.

I verified my build was working ok by running a test on the loudnorm_before.mp3 found at http://k.ylo.ph/2016/04/04/loudnorm.html

ffmpeg version N-83034-gf48b6b8 Copyright (c) 2000-2017 the FFmpeg developers

built with gcc 4.8.3 (GCC) 20140911 (Red Hat 4.8.3-9) configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags='-L/root/ffmpeg_build/lib -ldl' --bindir=/root/bin --pkg-config-flags=--static --enable-gpl --enable-nonfree --enable-libfdk_aac libavutil 55. 43.100 / 55. 43.100 libavcodec 57. 71.100 / 57. 71.100 libavformat 57. 62.100 / 57. 62.100 libavdevice 57. 2.100 / 57. 2.100 libavfilter 6. 68.100 / 6. 68.100 libswscale 4. 3.101 / 4. 3.101 libswresample 2. 4.100 / 2. 4.100 libpostproc 54. 2.100 / 54. 2.100

I setup a new test file and concatenated it together several times. The file bonjour.wav is 1.17 seconds long and the LRA could not be measured. The file bonjour_twice.wav is 2.33 seconds and the LRA could not be measured. The file bonjour_three.wav is 3.49 seconds and the LRA CAN be measured.

I have successfully used those measurements to properly normalise all 3 input files. ie. the normalisation pass works properly on small files.

ffmpeg -i bonjour_twice.wav -af loudnorm=I=-16:tp=-1.5:LRA=11:print_format=json -f null -

size=N/A time=00:00:02.33 bitrate=N/A speed=74.1x

{

"input_i" : "-22.85", "input_tp" : "-10.19", "input_lra" : "0.00", "input_thresh" : "-32.85", "output_i" : "-22.33", "output_tp" : "-9.66", "output_lra" : "0.00", "output_thresh" : "-32.33", "normalization_type" : "linear", "target_offset" : "6.33"

}

ffmpeg -i bonjour_three.wav -af loudnorm=I=-16:tp=-1.5:LRA=11:print_format=json -f null -

size=N/A time=00:00:03.49 bitrate=N/A speed=46.2x

{

"input_i" : "-22.84", "input_tp" : "-10.19", "input_lra" : "0.30", "input_thresh" : "-32.84", "output_i" : "-15.94", "output_tp" : "-3.29", "output_lra" : "0.00", "output_thresh" : "-25.94", "normalization_type" : "dynamic", "target_offset" : "-0.06"

}


          #6098: qcelp: "IFQ: Badly received packets in frame."   
C:\>ffmpeg -i qcelp_13k.mov out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl --disable-ffprobe
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 72.100 / 57. 72.100
  libavformat    57. 62.100 / 57. 62.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 69.100 /  6. 69.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[qcelp @ 0667d680] Warning: not compiled with thread support, using thread emula
tion
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'qcelp_13k.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2016-12-29T13:36:49.000000Z
  Duration: 00:01:00.00, start: 0.000000, bitrate: 14 kb/s
    Stream #0:0(eng): Audio: qcelp (Qclp / 0x706C6351), 8000 Hz, mono, flt (defa
ult)
    Metadata:
      creation_time   : 2016-12-29T13:36:49.000000Z
      handler_name    : Procedura obs│ugi skrˇtˇw danych Apple
[qcelp @ 06a221c0] Warning: not compiled with thread support, using thread emula
tion
[pcm_s16le @ 06a36fc0] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    ISFT            : Lavf57.62.100
    Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s
16, 128 kb/s (default)
    Metadata:
      creation_time   : 2016-12-29T13:36:49.000000Z
      handler_name    : Procedura obs│ugi skrˇtˇw danych Apple
      encoder         : Lavc57.72.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (qcelp (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[qcelp @ 06a221c0] Frame #105, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #577, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1152, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1154, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1155, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1156, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1157, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1158, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1159, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1160, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1161, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1162, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1163, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1164, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1165, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1166, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1167, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1168, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1169, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1170, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1171, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1172, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1173, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1825, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1826, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1828, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1831, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1832, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1833, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1834, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1835, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1836, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1837, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1838, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1839, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1840, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1841, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1842, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1843, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1844, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1845, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1846, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1848, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1849, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1850, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1851, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1852, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1853, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1854, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1855, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1856, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1857, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1858, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1859, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1860, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1861, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1862, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1863, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1864, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1865, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1866, IFQ: Badly received packets in frame.
[qcelp @ 06a221c0] Frame #1867, IFQ: Badly received packets in frame.
size=     938kB time=00:01:00.00 bitrate= 128.0kbits/s speed=240x
video:0kB audio:938kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.008125%

          #6156: Issue with FFMPEG ALSA CPU usage   

I am on a Raspberry Pi Zero with a Pi Camera and an I2S MEMS mic, running the latest Raspbian Jessie. I am attempting to livestream audio and video. I've discovered that when I record ALSA audio with FFMPEG, that process takes 95% of my CPU and produces stuttering, unusable audio in the stream.

I isolated the issue to the following:

  1. ffmpeg -f alsa -acodec pcm_s32le -i mic_sv temp.wav

This single command takes 95% of my CPU.

  1. arecord -Dmic_sv -c2 -r48000 -fS32_LE -twav temp.wav

This single command takes <5% of my CPU.

I have added a custom hook to ffmpeg so we can see the PCM settings. I have ensured that settings (buffer size, period size, etc.) are identical between the two commands.

Here is the full console output of FFMPEG:

$ ALSA_CONFIG_PATH=/usr/share/alsa/alsa.conf:./dumpasoundrc  /home/pi/special/ffmpeg/ffmpeg -v 9 -loglevel 99  -f alsa -acodec pcm_s32le -i michooks -f  null -
ffmpeg version git-2017-02-05-e57fd92 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.9.2 (Raspbian 4.9.2-10)
  configuration: --extra-cflags=-I/opt/vc/include/IL --enable-nonfree --enable-omx-rpi --logfile=CONFIG.TXT
  libavutil      55. 46.100 / 55. 46.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 66.101 / 57. 66.101
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 73.100 /  6. 73.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
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 '-f' ... matched as option 'f' (force format) with argument 'alsa'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'pcm_s32le'.
Reading option '-i' ... matched as input url with argument 'michooks'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'null'.
Reading option '-' ... matched as output url.
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 url michooks.
Applying option f (force format) with argument alsa.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument pcm_s32le.
Successfully parsed a group of options.
Opening an input file: michooks.
[ALSA Debug Hook] Installing hooks.
Hooks PCM
Slave: Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Slave: Mmap emulation PCM
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
[ALSA Debug Hook] hw_params:
Hooks PCM
Slave: Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Mmap emulation PCM
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1073741824
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
  appl_ptr     : 0
  hw_ptr       : 0
[alsa @ 0x31ea450] All info found
[alsa @ 0x31ea450] stream 0: start_time: 1487382765.290 duration: -9223372036854.775
[alsa @ 0x31ea450] format: start_time: 1487382765.290 duration: -9223372036854.775 bitrate=3072 kb/s
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'michooks':
  Duration: N/A, start: 1487382765.289814, bitrate: 3072 kb/s
    Stream #0:0, 1, 1/1000000: Audio: pcm_s32le, 48000 Hz, stereo, s32, 3072 kb/s
Successfully opened the file.
Parsing a group of options: output url -.
Applying option f (force format) with argument null.
Successfully parsed a group of options.
Opening an output file: -.
Successfully opened the file.
detected 1 logical cores
[graph_0_in_0_0 @ 0x31f9310] Setting 'time_base' to value '1/48000'
[graph_0_in_0_0 @ 0x31f9310] Setting 'sample_rate' to value '48000'
[graph_0_in_0_0 @ 0x31f9310] Setting 'sample_fmt' to value 's32'
[graph_0_in_0_0 @ 0x31f9310] Setting 'channel_layout' to value '0x3'
[graph_0_in_0_0 @ 0x31f9310] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x3
[format_out_0_0 @ 0x31f9f10] Setting 'sample_fmts' to value 's16'
[format_out_0_0 @ 0x31f9f10] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0x31f9980] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0x31faf10] [SWR @ 0x321e130] Using fltp internally between filters
[auto_resampler_0 @ 0x31faf10] ch:2 chl:stereo fmt:s32 r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    Metadata:
      encoder         : Lavc57.75.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s32le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
size=N/A time=00:00:17.84 bitrate=N/A speed=   1x
video:0kB audio:3345kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (michooks):
  Input stream #0:0 (audio): 26372 packets read (6851424 bytes); 26372 frames decoded (856428 samples);
  Total: 26372 packets (6851424 bytes) demuxed
Output file #0 (pipe:):
  Output stream #0:0 (audio): 26372 frames encoded (856428 samples); 26372 packets muxed (3425712 bytes);
  Total: 26372 packets (3425712 bytes) muxed
26372 frames successfully decoded, 0 decoding errors
Exiting normally, received signal 2.

Here is the console output of Arecord:

$ arecord -v -r48000 -c2 -fS32_LE --period-size=16384 --buffer-size=65536 -Dmic_sv -twav /dev/null
Recording WAVE '/dev/null' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo
Soft volume PCM
Control: Boost Capture Volume
min_dB: -3
max_dB: 50
resolution: 256
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Route conversion PCM
  Transformation table:
    0 <- 0
    1 <- 0
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Mmap emulation PCM
Its setup is:
  stream       : CAPTURE
  access       : MMAP_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
Slave: Hardware PCM card 1 'mems-mic' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S32_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 65536
  period_size  : 16384
  period_time  : 341333
  tstamp_mode  : NONE
  period_step  : 1
  avail_min    : 16384
  period_event : 0
  start_threshold  : 1073741824
  stop_threshold   : 65536
  silence_threshold: 0
  silence_size : 0
  boundary     : 1073741824
  appl_ptr     : 0
  hw_ptr       : 0
^CAborted by signal Interrupt...

After a long chat in the FFMPEG IRC channel, we believe there may be a bug in the way FFMPEG waits for ALSA packets. Arecord uses snd_pcm_wait() to wait for packets, and this command is not used in FFMPEG sources. Inserting -re before -i into my FFMPEG command reduces CPU usage to <5%, but is not the right approach, and causes other problems (bitrate is almost 0kbps, audio file generated is <5 kilobytes).


          #6216: TrueHD encoder creates non bitexact files   

Summary of the bug: How to reproduce:

./ffmpeg -i truehd.wav -f md5 -
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'truehd.wav':
  Metadata:
    encoder         : Lavf57.66.103
  Duration: 00:00:02.02, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.82.100 pcm_s16le
MD5=d886c9feaf7dddcdff3a5adcfa4c57bc
size=       0kB time=00:00:02.02 bitrate=   0.1kbits/s speed= 952x
video:0kB audio:348kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
% ./ffmpeg -i truehd.wav -c:a truehd -strict -2 truehd.thd
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'truehd.wav':
  Metadata:
    encoder         : Lavf57.66.103
  Duration: 00:00:02.02, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> truehd (native))
Press [q] to stop, [?] for help
Output #0, truehd, to 'truehd.thd':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: truehd, 44100 Hz, stereo, s16, 128 kb/s
    Metadata:
      encoder         : Lavc57.82.100 truehd
size=      57kB time=00:00:02.02 bitrate= 231.8kbits/s speed= 103x
video:0kB audio:57kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
./ffmpeg -i truehd.thd -f md5 -
ffmpeg version N-83737-g68ee800a9d Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-libx265 --enable-libfdk_aac --enable-libvpx --enable-libmp3lame --enable-libvorbis --enable-libopus --target-os=mingw32 --arch=x86_64 --cpu=haswell --extra-cflags='-D_WIN32_WINNT=0x0602' --cc='ccache gcc' --samples=../samples --prefix=/mingw64
  libavutil      55. 47.101 / 55. 47.101
  libavcodec     57. 82.100 / 57. 82.100
  libavformat    57. 66.103 / 57. 66.103
  libavdevice    57.  3.100 / 57.  3.100
  libavfilter     6. 74.100 /  6. 74.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
Input #0, truehd, from 'truehd.thd':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: truehd, 44100 Hz, stereo, s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (truehd (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf57.66.103
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc57.82.100 pcm_s16le
[truehd @ 0000000002544f40] Lossless check failed - expected 07, calculated 04.
[truehd @ 0000000002544f40] Lossless check failed - expected 06, calculated 04.
[truehd @ 0000000002544f40] Lossless check failed - expected 00, calculated 02.
[truehd @ 0000000002544f40] Lossless check failed - expected 01, calculated 03.
MD5=cba280ea373fa6287016c9bca7900bac
size=       0kB time=00:00:01.99 bitrate=   0.1kbits/s speed= 104x
video:0kB audio:346kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown

The resulting stream is also for some reason reported as 24bit (s32) instead of 16bit.


          #6330: Strange behavior with native FFmpeg AAC encoder, when converting audio to mono   

Summary of the bug:

When converting stereo to mono, even it seems to really convert to mono, it's been reported as stereo and with a strange bitrate (not CBR??) by MediaInfo? and VLC.

How to reproduce:

% ffmpeg -i input -c:v libx264 -b:v 750k -profile:v baseline -level 3.0 -pix_fmt yuv420p -c:a aac -b:a 64k -ac 1 output
ffmpeg version FFmpeg version: 20170330-ad7aff0

MediaInfo? and VLC report stereo audio.

MediaInfo? also reports a strange 64.5 kbps bitrate (when I wrote 64k), with 2 channels, 1 original channel and Front C position:

Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 1min.
Duration_LastFrame                       : -11ms
Bit rate mode                            : Constante
Bit rate                                 : 64,5Kbps
Channel(s)                               : 2canales
Channel(s)_Original                      : 1canal
Channel positions                        : Front: C
Sampling rate                            : 48,0KHz
Stream size                              : 472Kib (8%)
Language                                 : Español

Testing other bitrates: 48k -> 48Kbps (Constant) 64k -> 64,5Kbps (Constant) 96k -> 96,0Kbps (Constant) 112k -> 114Kbps (Constant) 128k -> 129Kbps (Constant) 192k -> 159Kbps and Variable (with a max bitrate of 192Kbps)

So, it makes me think maybe it's not really CBR?

And althought "-b:a" is supossed to make CBR, when compressing only audio, MediaInfo? always reports Variable Bitrate, so no bitrate is visible with it.

ffmpeg -i "1min stereo.wav" -c:a aac -b:a 64k -ac 1 "1min 64kb mono.aac"
Audio
Formato                                  : AAC
Formato/Info                             : Advanced Audio Codec
Formato de la versión                    : Version 4
Formato del perfil                       : LC
Tipo de tasa de bits                     : Variable
Canal(es)                                : 1canal
Posiciones del canal                     : Front: C
Velocidad de muestreo                    : 48,0KHz
Tamaño de pista                          : 491Kib (100%)

          #6351: encodec lossy audio LFE how to do not to process   

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" "F:\8.wav"

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" -c aac "F:\8.m4a"

ffmpeg -i "F:\8.flac" -af "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7" -c dca -strict -2 "F:\8.dts"

flac & wav,LFE is no process,but aac & dca is do EQ and down volume. aac & dca,the eq is no same,dca is lower.

??? which set can do not to process LFE ???


          #1520: allow mathematical expressions for -t and -ss   

Summary of the bug:

It seems that -t and -ss support integer and decimals up to the six places after the decimal (-t 1.033366 works but -t 1.0333666 doesn't). Sometimes greater precision is needed. Is it possible to support mathematical expression with -t and -ss as it is with some of the timing in the filters.

For instance if I try to create a wav file that contains one NTSC frame's worth of silence, it will produce an error. The closest I can seem to get is with '-t 1.033367'.

How to reproduce:

ffmpeg -t '1/(30000/1001)' -f lavfi -i aevalsrc=0 -c:a pcm_s16le out.wav
ffmpeg version 0.11.1.git Copyright (c) 2000-2012 the FFmpeg developers
  built on Jul  5 2012 11:22:03 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-libfreetype --cc=/usr/bin/gcc-4.2 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libass --enable-libvo-aacenc
  libavutil      51. 64.100 / 51. 64.100
  libavcodec     54. 33.100 / 54. 33.100
  libavformat    54. 15.100 / 54. 15.100
  libavdevice    54.  1.100 / 54.  1.100
  libavfilter     3.  0.101 /  3.  0.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
Invalid duration specification for t: 1/(30000/1001)

          #2493: support filter options within ffprobe   

Summary of the bug:

This ticket is inspired by comments by Nicolas George at this post: http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2013-March/141071.html.

I'd like to make use of metadata from filters within ffprobe without having to use amovie|movie to provide the input.

For instance:

ffprobe 2>/dev/null -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format

provides

ffprobe -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format
ffprobe version 1.1.git Copyright (c) 2007-2013 the FFmpeg developers
  built on Apr 13 2013 10:04:18 with Apple clang version 3.1 (tags/Apple/clang-318.0.61) (based on LLVM 3.1svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libfreetype --enable-ffplay --enable-libcaca
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 53.101 /  3. 53.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Failed to set value 'ebur128=metadata=1' for option 'af': Option not found

The workaround seems to be to use lavfi as the input:

ffprobe 2>/dev/null -f lavfi amovie=038Robin.wav,ebur128=metadata=1 -show_frames -show_format

However although the latter commands provides the frame metadata that I want, the data provided by -show_format refers to lavfi and not the original file input.

Is it feasible to support -vf and -af in ffprobe to allow something like:

ffprobe -i 038Robin.wav -af ebur128=metadata=1 -show_frames -show_format

          #4175: support phase shift for Dolby Pro Logic II / Dolby matrix downmix   

Summary of the bug:

Hello guys. I am trying to downmix a 5.1 audio to 2.0 (stereo) using the audio resampler filter "matrix_encoding=dplii". The resulting stereo audio presents some audio level bias towards the right speaker when there is more than one active channel on the surround stream. That is, it seems that the audio volume on the right speaker is higher than it should be. This is, of course, using a test signal where all surround channels produce the same sound at the same volume.

How to reproduce:

% ffmpeg -i ChID-BLITS-EBU-Narration441-16b.wav -ac 2 -filter:a "aresample=matrix_encoding=dplii" dplii.wav
ffmpeg version 2.4.4 Copyright (c) 2000-2014 the FFmpeg developers
  built on Dec  4 2014 14:13:11 with Apple LLVM version 6.0 (clang-600.0.56) (based on LLVM 3.5svn)

You may freely obtain the surround test file I used on the command above from Fraunhoffer's website. Direct link to the surround wave file: https://www2.iis.fraunhofer.de/AAC/ChID-BLITS-EBU-Narration441-16b.wav Fraunhoffer's page where that file's URL is located: https://www2.iis.fraunhofer.de/AAC/multichannel.html

If you analyze both the surround input file and the resulting stereo wave file on Audacity, for example, you will notice that by 19 seconds into the audio stream all channels are active with a test signal. The same point in time on the resulting DPL-II downmixed shows a "higher volume" on the right speaker. This effect is even more noticeable by 22 seconds when they change the test signal. It is interesting to note that in the initial seconds of the file, when a male's voice is announcing each channel individually, I could not notice this right speaker bias. The channel downmixing in that segment sounds correct. This effect is very noticeable on sound tracks with very active surround channels, they all seem to be stronger on the right speaker.

Thank you.


          #4998: SanDisk Clip Sport/Jam is buggy, support generating "simpler" mp4 files that can be played by it.   

Hi m4a files created by FFmpeg aren't supported by SanDisk? Clip Sport/Jam?. I've attached 3 files.

1 nero.m4a This plays OK. Created with neroAacEnc version 1.5.4.0 neroAacEnc -if foo.wav -of nero.m4a

2 FFmpeg.m4a This doesn't play. Created with FFmpeg ffmpeg version git-2015-10-28-cceed83 FFmpeg -i foo.wav -c:a aac -strict -2 FFmpeg.m4a

3 nero2.m4a This doesn't play either (nero file re-muxed by FFmpeg). ffmpeg -i nero.m4a -c copy nero2.m4a

So I think that it is not FFmpeg's aac encoder that is causing the problem, but maybe the muxing into m4a by FFmpeg.

My operating system Ubuntu-12.04. The mp3 player is Clip Jam 8GB model number SDMX26-008G-G46K


          #5239: support options for MPEGTS muxer when using RTP_MPEGTS   

I'm trying to produce a MPEG TS over RTP with a constant muxrate. I am able produce a MPEG TS with constant muxrate over UDP using the "mpegts" output format (as shown in the first pasted console output below), but after changing the command to use "rtp_mpegts" and a RTP:// address, the output bit rate simply follows the A/V rate. I.E the -muxrate option is ignored and the null packets are not inserted. Having compared the outputs using a TS analyzer, I've also noticed that the -streamid and -metadata options are also ignored because the PIDs and service names are different. Perhaps all options are being ignored in that case?

COMMAND 1: MPEGTS over UDP

c:\ffmpeg-2-8-3\bin\ffmpeg -loop 1 -re -i HOLPGKO01/bars/NFLbars%03d.jpg -f lavfi -i "amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME_RATE*TB)" -c:v libx264 -profile:v high -pix_fmt yuvj420p -g 5 -b:v 1M -minrate 1M -maxrate 1M -bufsize 512k -b:a 384k -map 0:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -f mpegts -streamid 0:31 -streamid 1:32 -streamid 2:33 -streamid 3:34 -streamid 4:35 -muxrate 8M -mpegts_pmt_start_pid 0x001E -metadata service_provider="Provider" -metadata service_name="Service 1" udp://239.118.0.20:11820?pkt_size=1316

ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers

built with gcc 5.2.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc

--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl

e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib

libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100

[mjpeg @ 0000000000328100] Changeing bps to 8 Input #0, image2, from 'HOLPGKO01/bars/NFLbars%03d.jpg':

Duration: 00:00:12.60, start: 0.000000, bitrate: N/A

Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1920x1080

[SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc [Parsed_amovie_0 @ 00000000056cf0a0] Channel layout is not set in output stream 0, guessed channel layout is 'stereo' Input #1, lavfi, from 'amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME _RATE*TB)':

Duration: N/A, bitrate: 1536 kb/s

Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s

[libx264 @ 0000000000335ba0] using SAR=1/1 [libx264 @ 0000000000335ba0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0000000000335ba0] profile High, level 4.0 Output #0, mpegts, to 'udp://239.118.0.20:11820?pkt_size=1316':

Metadata:

service_provider: Provider service_name : Service 1 encoder : Lavf56.40.101 Stream #0:0: Video: h264 (libx264), yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:

9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc

Metadata:

encoder : Lavc56.60.100 libx264

Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:2: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:3: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream #0:4: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 mp2

Stream mapping:

Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264)) Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:2 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:3 (pcm_s16le (native) -> mp2 (native)) Stream #1:0 -> #0:4 (pcm_s16le (native) -> mp2 (native))

Press [q] to stop, ? for help [image2 @ 00000000003251e0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8) frame= 37 fps= 36 q=33.0 size= 509kB time=00:00:00.52 bitrate=8011.7kbits/ frame= 49 fps= 32 q=33.0 size= 1017kB time=00:00:01.00 bitrate=8330.7kbits/ frame= 62 fps= 30 q=33.0 size= 1485kB time=00:00:01.52 bitrate=8004.8kbits/ frame= 74 fps= 29 q=33.0 size= 1994kB time=00:00:02.00 bitrate=8142.4kbits/ frame= 87 fps= 29 q=33.0 size= 2501kB time=00:00:02.52 bitrate=8128.8kbits/ frame= 100 fps= 28 q=32.0 size= 2973kB time=00:00:03.04 bitrate=8012.3kbits/ frame= 112 fps= 28 q=32.0 size= 3478kB time=00:00:03.52 bitrate=8093.4kbits/ frame= 125 fps= 27 q=33.0 size= 3950kB time=00:00:04.04 bitrate=8008.8kbits/ frame= 137 fps= 27 q=33.0 size= 4454kB time=00:00:04.52 bitrate=8061.4kbits/ frame= 150 fps= 27 q=32.0 size= 4962kB time=00:00:05.04 bitrate=8064.6kbits/ frame= 163 fps= 27 q=30.0 size= 5430kB time=00:00:05.56 bitrate=8001.0kbits/ frame= 175 fps= 27 q=31.0 size= 5938kB time=00:00:06.04 bitrate=8053.9kbits/ frame= 188 fps= 27 q=30.0 size= 6407kB time=00:00:06.56 bitrate=8001.0kbits/ frame= 200 fps= 26 q=31.0 size= 6915kB time=00:00:07.04 bitrate=8039.4kbits/ frame= 213 fps= 26 q=33.0 size= 7439kB time=00:00:07.56 bitrate=8061.3kbits/ frame= 226 fps= 26 q=32.0 size= 7894kB time=00:00:08.08 bitrate=8003.2kbits/ frame= 238 fps= 26 q=32.0 size= 8415kB time=00:00:08.56 bitrate=8053.4kbits/ frame= 251 fps= 26 q=32.0 size= 8870kB time=00:00:09.08 bitrate=8002.8kbits/ frame= 263 fps= 26 q=30.0 size= 9390kB time=00:00:09.56 bitrate=8041.7kbits/ frame= 276 fps= 26 q=33.0 size= 9884kB time=00:00:10.08 bitrate=8032.4kbits/ frame= 288 fps= 26 q=31.0 size= 10368kB time=00:00:10.56 bitrate=8042.7kbits/ frame= 301 fps= 26 q=33.0 size= 10860kB time=00:00:11.08 bitrate=8029.3kbits/ frame= 314 fps= 26 q=32.0 size= 11345kB time=00:00:11.60 bitrate=8012.0kbits/ frame= 326 fps= 26 q=32.0 size= 11836kB time=00:00:12.08 bitrate=8022.9kbits/ frame= 339 fps= 26 q=31.0 size= 12322kB time=00:00:12.60 bitrate=8011.2kbits/ frame= 351 fps= 26 q=31.0 size= 12813kB time=00:00:13.08 bitrate=8024.9kbits/ frame= 364 fps= 26 q=31.0 size= 13300kB time=00:00:13.60 bitrate=8011.5kbits/ frame= 376 fps= 26 q=32.0 size= 13790kB time=00:00:14.08 bitrate=8023.2kbits/ frame= 389 fps= 26 q=32.0 size= 14299kB time=00:00:14.60 bitrate=8019.7kbits/ frame= 402 fps= 26 q=32.0 size= 14769kB time=00:00:15.12 bitrate=8001.6kbits/ frame= 408 fps= 25 q=-1.0 Lsize= 15900kB time=00:00:16.24 bitrate=8020.5kbits /s video:1574kB audio:2880kB subtitle:0kB other streams:0kB global headers:0kB muxi ng overhead: 256.951935% [libx264 @ 0000000000335ba0] frame I:82 Avg QP:20.83 size: 16417 [libx264 @ 0000000000335ba0] frame P:102 Avg QP:25.24 size: 1420 [libx264 @ 0000000000335ba0] frame B:224 Avg QP:38.17 size: 541 [libx264 @ 0000000000335ba0] consecutive B-frames: 25.2% 2.5% 6.6% 65.7% [libx264 @ 0000000000335ba0] mb I I16..4: 62.6% 29.7% 7.7% [libx264 @ 0000000000335ba0] mb P I16..4: 1.5% 0.7% 0.0% P16..4: 3.6% 0.6 % 0.2% 0.0% 0.0% skip:93.4% [libx264 @ 0000000000335ba0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 3.9% 0.3 % 0.0% direct: 0.0% skip:95.6% L0:39.1% L1:60.7% BI: 0.2% [libx264 @ 0000000000335ba0] 8x8 transform intra:29.7% inter:53.3% [libx264 @ 0000000000335ba0] coded y,uvDC,uvAC intra: 6.4% 2.1% 1.1% inter: 0.2%

0.1% 0.1%

[libx264 @ 0000000000335ba0] i16 v,h,dc,p: 82% 15% 2% 0% [libx264 @ 0000000000335ba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 19% 42% 1% 0%

0% 0% 0% 1%

[libx264 @ 0000000000335ba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 26% 11% 8% 3%

3% 3% 4% 3%

[libx264 @ 0000000000335ba0] i8c dc,h,v,p: 83% 11% 6% 0% [libx264 @ 0000000000335ba0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000000335ba0] ref P L0: 94.9% 2.7% 1.7% 0.6% [libx264 @ 0000000000335ba0] ref B L0: 89.7% 10.3% [libx264 @ 0000000000335ba0] ref B L1: 91.2% 8.8% [libx264 @ 0000000000335ba0] kb/s:790.27 Exiting normally, received signal 2.

COMMAND 2: MPEGTS over RTP

c:\ffmpeg-2-8-3\bin\ffmpeg -loop 1 -re -i HOLPGKO01/bars/NFLbars%03d.jpg -f lavfi -i "amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME_RATE*TB)" -c:v libx264 -profile:v high -pix_fmt yuvj420p -g 5 -b:v 1M -minrate 1M -maxrate 1M -bufsize 512k -b:a 384k -map 0:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -f rtp_mpegts -streamid 0:31 -streamid 1:32 -streamid 2:33 -streamid 3:34 -streamid 4:35 -muxrate 8M -mpegts_pmt_start_pid 0x001E -metadata service_provider="Provider" -metadata service_name="Service 1" rtp://239.118.0.20:11820?pkt_size=1316

ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers

built with gcc 5.2.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc

--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl

e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib

libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100

[mjpeg @ 0000000000328100] Changeing bps to 8 Input #0, image2, from 'HOLPGKO01/bars/NFLbars%03d.jpg':

Duration: 00:00:12.60, start: 0.000000, bitrate: N/A

Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1920x1080

[SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc [Parsed_amovie_0 @ 00000000057bf0a0] Channel layout is not set in output stream 0, guessed channel layout is 'stereo' Input #1, lavfi, from 'amovie=holpgko01/tone/tone.wav:loop=0, asetpts=N/(FRAME _RATE*TB)':

Duration: N/A, bitrate: 1536 kb/s

Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s

[libx264 @ 0000000000335ba0] using SAR=1/1 [libx264 @ 0000000000335ba0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0000000000335ba0] profile High, level 4.0 Output #0, rtp_mpegts, to 'rtp://239.118.0.20:11820?pkt_size=1316':

Metadata:

service_provider: Provider service_name : Service 1 encoder : Lavf56.40.101 Stream #0:0: Video: h264 (libx264), yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:

9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc

Metadata:

encoder : Lavc56.60.100 libx264

Stream #0:1: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:2: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:3: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream #0:4: Audio: aac (libvo_aacenc), 48000 Hz, stereo, s16, 384 kb/s Metadata:

encoder : Lavc56.60.100 libvo_aacenc

Stream mapping:

Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264)) Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:2 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:3 (pcm_s16le (native) -> aac (libvo_aacenc)) Stream #1:0 -> #0:4 (pcm_s16le (native) -> aac (libvo_aacenc))

Press [q] to stop, ? for help [image2 @ 00000000003251e0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8) frame= 36 fps= 35 q=33.0 size= 75kB time=00:00:00.48 bitrate=1273.5kbits/ frame= 48 fps= 31 q=31.0 size= 169kB time=00:00:00.96 bitrate=1444.2kbits/ frame= 61 fps= 30 q=33.0 size= 252kB time=00:00:01.48 bitrate=1392.8kbits/ frame= 74 fps= 29 q=33.0 size= 351kB time=00:00:02.00 bitrate=1436.5kbits/ frame= 86 fps= 28 q=33.0 size= 435kB time=00:00:02.48 bitrate=1438.0kbits/ frame= 99 fps= 28 q=33.0 size= 532kB time=00:00:03.00 bitrate=1453.2kbits/ frame= 111 fps= 27 q=32.0 size= 615kB time=00:00:03.48 bitrate=1446.7kbits/ frame= 124 fps= 27 q=33.0 size= 716kB time=00:00:04.00 bitrate=1466.1kbits/ frame= 137 fps= 27 q=33.0 size= 797kB time=00:00:04.52 bitrate=1444.7kbits/ frame= 149 fps= 27 q=32.0 size= 896kB time=00:00:05.00 bitrate=1468.0kbits/ frame= 162 fps= 27 q=32.0 size= 974kB time=00:00:05.52 bitrate=1445.7kbits/ frame= 174 fps= 26 q=31.0 size= 1070kB time=00:00:06.00 bitrate=1460.8kbits/ frame= 187 fps= 26 q=31.0 size= 1151kB time=00:00:06.52 bitrate=1446.4kbits/ frame= 200 fps= 26 q=31.0 size= 1242kB time=00:00:07.04 bitrate=1445.8kbits/ frame= 212 fps= 26 q=32.0 size= 1334kB time=00:00:07.52 bitrate=1453.0kbits/ frame= 225 fps= 26 q=32.0 size= 1422kB time=00:00:08.04 bitrate=1448.6kbits/ frame= 237 fps= 26 q=32.0 size= 1503kB time=00:00:08.52 bitrate=1444.9kbits/ frame= 250 fps= 26 q=32.0 size= 1603kB time=00:00:09.04 bitrate=1452.8kbits/ frame= 263 fps= 26 q=31.0 size= 1689kB time=00:00:09.56 bitrate=1447.2kbits/ frame= 275 fps= 26 q=33.0 size= 1787kB time=00:00:10.04 bitrate=1458.0kbits/ frame= 288 fps= 26 q=31.0 size= 1868kB time=00:00:10.56 bitrate=1449.2kbits/ frame= 300 fps= 26 q=33.0 size= 1967kB time=00:00:11.04 bitrate=1459.8kbits/ frame= 313 fps= 26 q=30.0 size= 2047kB time=00:00:11.56 bitrate=1450.9kbits/ frame= 325 fps= 26 q=32.0 size= 2142kB time=00:00:12.04 bitrate=1457.3kbits/ frame= 338 fps= 26 q=30.0 size= 2240kB time=00:00:12.56 bitrate=1461.0kbits/ frame= 350 fps= 26 q=31.0 size= 2321kB time=00:00:13.04 bitrate=1458.3kbits/ frame= 363 fps= 26 q=30.0 size= 2420kB time=00:00:13.56 bitrate=1462.2kbits/ frame= 376 fps= 26 q=32.0 size= 2503kB time=00:00:14.08 bitrate=1456.1kbits/ frame= 388 fps= 26 q=30.0 size= 2584kB time=00:00:14.56 bitrate=1453.9kbits/ frame= 401 fps= 26 q=32.0 size= 2686kB time=00:00:15.08 bitrate=1459.4kbits/ frame= 413 fps= 26 q=31.0 size= 2784kB time=00:00:15.56 bitrate=1465.8kbits/ frame= 415 fps= 25 q=-1.0 Lsize= 2889kB time=00:00:16.52 bitrate=1432.6kbits /s video:1595kB audio:982kB subtitle:0kB other streams:0kB global headers:0kB muxin g overhead: 12.116439% [libx264 @ 0000000000335ba0] frame I:83 Avg QP:20.79 size: 16442 [libx264 @ 0000000000335ba0] frame P:105 Avg QP:25.40 size: 1400 [libx264 @ 0000000000335ba0] frame B:227 Avg QP:38.45 size: 537 [libx264 @ 0000000000335ba0] consecutive B-frames: 25.3% 2.9% 7.2% 64.6% [libx264 @ 0000000000335ba0] mb I I16..4: 63.3% 29.0% 7.7% [libx264 @ 0000000000335ba0] mb P I16..4: 1.5% 0.6% 0.0% P16..4: 3.6% 0.5 % 0.2% 0.0% 0.0% skip:93.5% [libx264 @ 0000000000335ba0] mb B I16..4: 0.1% 0.0% 0.0% B16..8: 3.9% 0.2 % 0.0% direct: 0.0% skip:95.7% L0:39.0% L1:60.8% BI: 0.2% [libx264 @ 0000000000335ba0] 8x8 transform intra:29.1% inter:51.0% [libx264 @ 0000000000335ba0] coded y,uvDC,uvAC intra: 6.4% 2.0% 1.1% inter: 0.2%

0.1% 0.1%

[libx264 @ 0000000000335ba0] i16 v,h,dc,p: 82% 15% 2% 0% [libx264 @ 0000000000335ba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 35% 18% 44% 1% 0%

0% 0% 0% 1%

[libx264 @ 0000000000335ba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 26% 11% 8% 3%

3% 3% 4% 3%

[libx264 @ 0000000000335ba0] i8c dc,h,v,p: 83% 11% 5% 0% [libx264 @ 0000000000335ba0] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0000000000335ba0] ref P L0: 95.1% 2.6% 1.7% 0.6% [libx264 @ 0000000000335ba0] ref B L0: 89.4% 10.6% [libx264 @ 0000000000335ba0] ref B L1: 91.4% 8.6% [libx264 @ 0000000000335ba0] kb/s:787.29 Exiting normally, received signal 2.


          #3552: Initial warnings shown when playing aac   

http://thread.gmane.org/gmane.comp.video.ffmpeg.user/51341 I will attach (the first 25 seconds of) a sample that shows warnings before the first second is decoded.

$ ffmpeg -i iv_cut.aacp -t 24 out.wav
ffmpeg version N-62354-g671f908 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 11 2014 19:43:39 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      52. 75.100 / 52. 75.100
  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
[aac @ 0x1ef0480] channel element 2.8 is not allocated
[aac @ 0x1eef900] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'iv_cut.aacp':
  Duration: 00:00:25.03, bitrate: 32 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 32 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.37.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x1ef0480] channel element 2.8 is not allocated
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] illegal icc
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
[aac @ 0x1ef0480] Reserved SBR extensions is 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.
[aac @ 0x1ef0480] 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.
size=    4128kB time=00:00:24.00 bitrate=1408.5kbits/s
video:0kB audio:4128kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.001893%

          #4685: Misleading warning shown for mov file with multiple edit list entries   

FFmpeg shows a warning about missing mp3 frame size for a mov file with multiple edit list entries. The stream can be played and decoded.

$ ffmpeg -i leopard-intro.mov -map 0:3 out.wav
ffmpeg version N-73240-g55397b0 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 27.100 / 54. 27.100
  libavcodec     56. 45.101 / 56. 45.101
  libavformat    56. 39.100 / 56. 39.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 19.100 /  5. 19.100
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
  libpostproc    53.  3.100 / 53.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] multiple edit list entries, a/v desync might occur, patch welcome
    Last message repeated 2 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] Could not find codec parameters for stream 1 (Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x26a92a0] Could not find codec parameters for stream 3 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for  Input Stream #0.3 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'leopard-intro.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2007-09-23 05:43:34
    timecode        : 00:00:00:00
  Duration: 00:00:46.25, start: 0.000000, bitrate: 7847 kb/s
    Stream #0:0(eng): Video: mjpeg (jpeg / 0x6765706A), yuvj420p(pc, bt470bg/unknown/unknown), 1008x630 [SAR 72:72 DAR 8:5], 9602 kb/s, 22.66 fps, 22.66 tbr, 2266 tbn, 2266 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:1(eng): Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s, 24 fps, 24 tbr, 2400 tbn, 2400 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:3(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, stereo, s16p, 159 kb/s (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    ISMP            : 00:00:00:00
    ISFT            : Lavf56.39.100
    Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Lavc56.45.101 pcm_s16le
Stream mapping:
  Stream #0:3 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    2097kB time=00:00:46.31 bitrate= 370.9kbits/s
video:0kB audio:2097kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004564%
$ ffplay -vn -ast 3 leopard-intro.mov -autoexit
ffplay version N-73240-g55397b0 Copyright (c) 2003-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 27.100 / 54. 27.100
  libavcodec     56. 45.101 / 56. 45.101
  libavformat    56. 39.100 / 56. 39.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 19.100 /  5. 19.100
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
  libpostproc    53.  3.100 / 53.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] multiple edit list entries, a/v desync might occur, patch welcome
    Last message repeated 2 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] Could not find codec parameters for stream 1 (Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82a0000920] Could not find codec parameters for stream 3 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'leopard-intro.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2007-09-23 05:43:34
    timecode        : 00:00:00:00
  Duration: 00:00:46.25, start: 0.000000, bitrate: 7847 kb/s
    Stream #0:0(eng): Video: mjpeg (jpeg / 0x6765706A), yuvj420p(pc, bt470bg/unknown/unknown), 1008x630 [SAR 72:72 DAR 8:5], 9602 kb/s, 22.66 fps, 22.66 tbr, 2266 tbn, 2266 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:1(eng): Video: mjpeg (jpeg / 0x6765706A), none(bt470bg/unknown/unknown), 1008x630, 2055 kb/s, 24 fps, 24 tbr, 2400 tbn, 2400 tbc (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      encoder         : Photo - JPEG
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:3(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 173 kb/s
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, stereo, s16p, 159 kb/s (default)
    Metadata:
      creation_time   : 2007-09-23 05:43:34
      handler_name    : Apple Alias Data Handler
  46.37    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

          #5573: Incorrect duration when decoding raw G.729   
$ ffmpeg -i fate-suite/act/REC03.act -acodec copy -map 0:0 -f rawvideo out.g729
$ ffmpeg -i out.g729 out.wav
ffmpeg version N-80026-g936751b Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      55. 24.100 / 55. 24.100
  libavcodec     57. 42.100 / 57. 42.100
  libavformat    57. 37.100 / 57. 37.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 45.100 /  6. 45.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[g729 @ 0x2dc93a0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, g729, from 'out.g729':
  Duration: 00:01:05.79, start: 0.000000, bitrate: 8 kb/s
    Stream #0:0: Audio: g729, 8000 Hz, 1 channels, s16
[wav @ 0x2dcbf40] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.37.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, mono, s16, 128 kb/s
    Metadata:
      encoder         : Lavc57.42.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (g729 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1028kB time=00:00:00.83 bitrate=10119.2kbits/s speed=6.01x
video:0kB audio:1028kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.007410%

Both input and output file have a duration of 65 seconds.


          #4085: Support PLII upmixing   

http://thread.gmane.org/gmane.comp.video.ffmpeg.user/54545

FFmpeg does support PLII down-mixing (see tickets #709 and #3455) but does not support upmixing afaict. I will attach a sample from the MythTV collection that plays surround sound on my receiver, upmixing to six channels does not produce a similar result.

$ ffmpeg -i channelsPLII.mpg -ac 6 out.wav
ffmpeg version N-67396-g90cdec5 Copyright (c) 2000-2014 the FFmpeg developers
  built on Nov  6 2014 00:16:59 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 11.100 / 54. 11.100
  libavcodec     56. 10.102 / 56. 10.102
  libavformat    56. 12.101 / 56. 12.101
  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, mpeg, from 'channelsPLII.mpg':
  Duration: 00:00:31.82, start: 0.740000, bitrate: 462 kb/s
    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 480x576 [SAR 8:5 DAR 4:3], max. 2420 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 224 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf56.12.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s
    Metadata:
      encoder         : Lavc56.10.102 pcm_s16le
Stream mapping:
  Stream #0:1 -> #0:0 (mp2 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=   17915kB time=00:00:31.84 bitrate=4608.0kbits/s
video:0kB audio:17914kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000556%

MythTV and VLC contain PLII decoder that support upmixing under GPL.


          #4553: Support G.729 stereo files   

http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/193145 A user uploaded a stereo G.729 file that is currently unsupported.

$ ffmpeg -i g729stereo.wav
ffmpeg version N-72081-gec38a1b Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 23.101 / 54. 23.101
  libavcodec     56. 38.101 / 56. 38.101
  libavformat    56. 33.101 / 56. 33.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
[g729 @ 0x2897e80] Only mono sound is supported (requested channels: 2).
[wav @ 0x2897460] Failed to open codec in av_find_stream_info
[g729 @ 0x2897e80] Only mono sound is supported (requested channels: 2).
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'g729stereo.wav':
  Duration: 00:00:14.14, bitrate: 8 kb/s
    Stream #0:0: Audio: g729 ([131][0][0][0] / 0x0083), 8000 Hz, 2 channels, 16 kb/s
At least one output file must be specified

          #5700: Read id3v2 in wav   

When i try to obtain the embedded CoverArt? from a WAV files, output is not generated.

Command line: ffmpeg -v 9 -loglevel 99 -i semesta.wav coverwav.jpg -report

ffmpeg version N-80694-g074fdf4 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --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-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  libavutil      55. 26.100 / 55. 26.100
  libavcodec     57. 47.100 / 57. 47.100
  libavformat    57. 40.101 / 57. 40.101
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 46.102 /  6. 46.102
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
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 'semesta.wav'.
Reading option 'coverwav.jpg' ... matched as output file.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option loglevel (set logging level) with argument 99.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file semesta.wav.
Successfully parsed a group of options.
Opening an input file: semesta.wav.
[file @ 033ca2c0] Setting default whitelist 'file,crypto'
[wav @ 033c9940] Format wav probed with size=2048 and score=99
[wav @ 033c9940] Before avformat_find_stream_info() pos: 44 bytes read:131072 seeks:3 nb_streams:1
[wav @ 033c9940] probing stream 0 pp:32
[wav @ 033c9940] probing stream 0 pp:31
[wav @ 033c9940] probing stream 0 pp:30
[wav @ 033c9940] probing stream 0 pp:29
[wav @ 033c9940] probing stream 0 pp:28
[wav @ 033c9940] probing stream 0 pp:27
[wav @ 033c9940] probing stream 0 pp:26
[wav @ 033c9940] probing stream 0 pp:25
[wav @ 033c9940] probing stream 0 pp:24
[wav @ 033c9940] probing stream 0 pp:23
[wav @ 033c9940] probing stream 0 pp:22
[wav @ 033c9940] probing stream 0 pp:21
[wav @ 033c9940] probing stream 0 pp:20
[wav @ 033c9940] probing stream 0 pp:19
[wav @ 033c9940] probing stream 0 pp:18
[wav @ 033c9940] probing stream 0 pp:17
[wav @ 033c9940] probing stream 0 pp:16
[wav @ 033c9940] probing stream 0 pp:15
[wav @ 033c9940] probing stream 0 pp:14
[wav @ 033c9940] probing stream 0 pp:13
[wav @ 033c9940] probing stream 0 pp:12
[wav @ 033c9940] probing stream 0 pp:11
[wav @ 033c9940] probing stream 0 pp:10
[wav @ 033c9940] probing stream 0 pp:9
[wav @ 033c9940] probing stream 0 pp:8
[wav @ 033c9940] probing stream 0 pp:7
[wav @ 033c9940] probing stream 0 pp:6
[wav @ 033c9940] probing stream 0 pp:5
[wav @ 033c9940] probing stream 0 pp:4
[wav @ 033c9940] probing stream 0 pp:3
[wav @ 033c9940] probing stream 0 pp:2
[wav @ 033c9940] probing stream 0 pp:1
[wav @ 033c9940] probed stream 0
[wav @ 033c9940] parser not found for codec pcm_s16le, packets or times may be invalid.
[wav @ 033c9940] All info found
[wav @ 033c9940] After avformat_find_stream_info() pos: 204844 bytes read:360448 seeks:3 frames:50
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'semesta.wav':
  Duration: 00:04:47.47, bitrate: 1416 kb/s
    Stream #0:0, 50, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output file coverwav.jpg.
Successfully parsed a group of options.
Opening an output file: coverwav.jpg.
Output #0, image2, to 'coverwav.jpg':
Output file #0 does not contain any stream
[AVIOContext @ 033ca400] Statistics: 360448 bytes read, 3 seeks

          #6140: Add support for AMR-WB+ codec   

FFmpeg currently does not have a decoder for AMR-WB+ codec.

A free encoder + decoder for AMR-WB+ can be downloaded from https://portal.3gpp.org/desktopmodules/Specifications/SpecificationDetails.aspx?specificationId=1451. The ZIP file has both binaries (encoder.exe, decoder.exe) and source code.

Here are some usage instructions,

$ wget http://www.music.helsinki.fi/tmt/opetus/uusmedia/esim/a2002011001-e02-16kHz.wav
$ wine encoder.exe -rate 16 -if a2002011001-e02-16kHz.wav -of out.3gp
3gp File Format
Encoding @  16.00kbps
 --- Running ---
$ mediainfo out.3gp
General
Complete name                            : out.3gp
Format                                   : MPEG-4
Format profile                           : 3GPP Media Release 6 Basic
Codec ID                                 : 3gp6 (3gp6/isom)
File size                                : 110 KiB
Duration                                 : 54 s 160 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 16.7 kb/s
Encoded date                             : UTC 2017-02-10 06:10:58
Tagged date                              : UTC 2017-02-10 06:10:58
Audio
ID                                       : 1
Format                                   : sawp
Codec ID                                 : sawp
Duration                                 : 54 s 160 ms
Bit rate mode                            : Constant
Bit rate                                 : 16.2 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 6 464 Hz
Bit depth                                : 16 bits
Stream size                              : 107 KiB (97%)
Encoded date                             : UTC 2017-02-10 06:10:58
Tagged date                              : UTC 2017-02-10 06:10:58
$ wine decoder.exe  -if out.3gp  -of out.wav
...
File type brand '3gp6' minor version 256
Compatible brands: 'isom' (yes) '3gp4' (no) '3gp5' (no)
Track 1 'sawp'  Duration 54.160s  Time scale 72000
Decoding @  16.00kbps
$ mediainfo out.wav
General
Complete name                            : out.wav
Format                                   : Wave
File size                                : 9.92 MiB
Duration                                 : 54 s 160 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 1 536 kb/s
Audio
Format                                   : PCM
Format settings, Endianness              : Little
Format settings, Sign                    : Signed
Codec ID                                 : 1
Duration                                 : 54 s 160 ms
Bit rate mode                            : Constant
Bit rate                                 : 1 536 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 48.0 kHz
Bit depth                                : 16 bits
Stream size                              : 9.92 MiB (100%)

It would be great to get support for AMR-WB+ codec in FFmpeg.


          #3815: libcdio only works with -ss 00   

Summary of the bug: The (undocumented) libcdio device does not work directly. How to reproduce:

ffmpeg -f libcdio -i /dev/sr0 test.wav

always end instantly and did not produce any valid file.

A possible workaround is:

ffmpeg -ss 00 -f libcdio -i /dev/sr0 test.wav

I use git version.

(I tried on another computer with ffmpeg stable release and it didn't work, too.)

Here is the output: $ ffmpeg -f libcdio -i /dev/sr0 test.wav ffmpeg version N-65021-gc2ef844 Copyright (c) 2000-2014 the FFmpeg developers

built on Jul 27 2014 14:04:31 with gcc 4.8.2 (Gentoo 4.8.2 p1.3r1, pie-0.5.8r1) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame-pointer' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable-nonfree --disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=oss --disable-outdev=oss --enable-bzlib --disable-runtime-cpudetect --disable-debug --disable-doc --enable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --disable-libsmbclient --enable-ffplay --enable-vaapi --enable-vdpau --enable-xlib --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame --enable-libaacplus --disable-libfaac --enable-libtheora --disable-libtwolame --enable-libwavpack --disable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libcdio --disable-libiec61883 --disable-libdc1394 --disable-libcaca --disable-openal --enable-opengl --disable-libv4l2 --disable-libpulse --enable-x11grab --disable-libbs2b --disable-libflite --disable-frei0r --disable-libfribidi --disable-fontconfig --enable-ladspa --disable-libass --enable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore-amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --enable-libbluray --enable-libcelt --disable-libgme --disable-libgsm --disable-libmodplug --enable-libopus --enable-libquvi --disable-librtmp --disable-libssh --disable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libvpx --disable-libzvbi --disable-armv5te --disable-armv6 --disable-armv6t2 --disable-neon --disable-vfp --disable-mips32r2 --disable-mipsdspr1 --disable-mipsdspr2 --disable-mipsfpu --disable-altivec --disable-amd3dnow --disable-amd3dnowext --disable-avx --disable-avx2 --disable-fma3 --disable-fma4 --disable-sse3 --disable-sse4 --disable-sse42 --disable-xop --cpu=host 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

Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, libcdio, from '/dev/sr0':

Duration: 00:39:21.61, bitrate: N/A

Chapter #0.0: start 0.000000, end 89.360000 Metadata:

title : track 01

Chapter #0.1: start 89.360000, end 100.453333 Metadata:

title : track 02

Chapter #0.2: start 100.453333, end 346.680000 Metadata:

title : track 03

Chapter #0.3: start 346.680000, end 414.746667 Metadata:

title : track 04

Chapter #0.4: start 414.746667, end 576.333333 Metadata:

title : track 05

Chapter #0.5: start 576.333333, end 631.093333 Metadata:

title : track 06

Chapter #0.6: start 631.093333, end 723.906667 Metadata:

title : track 07

Chapter #0.7: start 723.906667, end 771.813333 Metadata:

title : track 08

Chapter #0.8: start 771.813333, end 862.746667 Metadata:

title : track 09

Chapter #0.9: start 862.746667, end 940.346667 Metadata:

title : track 10

Chapter #0.10: start 940.346667, end 1079.186667 Metadata:

title : track 11

Chapter #0.11: start 1079.186667, end 1190.653333 Metadata:

title : track 12

Chapter #0.12: start 1190.653333, end 1238.746667 Metadata:

title : track 13

Chapter #0.13: start 1238.746667, end 1393.866667 Metadata:

title : track 14

Chapter #0.14: start 1393.866667, end 1467.586667 Metadata:

title : track 15

Chapter #0.15: start 1467.586667, end 1548.733333 Metadata:

title : track 16

Chapter #0.16: start 1548.733333, end 1595.493333 Metadata:

title : track 17

Chapter #0.17: start 1595.493333, end 1720.413333 Metadata:

title : track 18

Chapter #0.18: start 1720.413333, end 1801.480000 Metadata:

title : track 19

Chapter #0.19: start 1801.480000, end 1932.146667 Metadata:

title : track 20

Chapter #0.20: start 1932.146667, end 1969.053333 Metadata:

title : track 21

Chapter #0.21: start 1969.053333, end 2135.746667 Metadata:

title : track 22

Chapter #0.22: start 2135.746667, end 2176.933333 Metadata:

title : track 23

Chapter #0.23: start 2176.933333, end 2361.613333 Metadata:

title : track 24

Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s

Output #0, wav, to 'test.wav':

Metadata:

ISFT : Lavf55.49.100 Chapter #0.0: start 0.000000, end 89.360000 Metadata:

title : track 01

Chapter #0.1: start 89.360000, end 100.453333 Metadata:

title : track 02

Chapter #0.2: start 100.453333, end 346.680000 Metadata:

title : track 03

Chapter #0.3: start 346.680000, end 414.746667 Metadata:

title : track 04

Chapter #0.4: start 414.746667, end 576.333333 Metadata:

title : track 05

Chapter #0.5: start 576.333333, end 631.093333 Metadata:

title : track 06

Chapter #0.6: start 631.093333, end 723.906667 Metadata:

title : track 07

Chapter #0.7: start 723.906667, end 771.813333 Metadata:

title : track 08

Chapter #0.8: start 771.813333, end 862.746667 Metadata:

title : track 09

Chapter #0.9: start 862.746667, end 940.346667 Metadata:

title : track 10

Chapter #0.10: start 940.346667, end 1079.186667 Metadata:

title : track 11

Chapter #0.11: start 1079.186667, end 1190.653333 Metadata:

title : track 12

Chapter #0.12: start 1190.653333, end 1238.746667 Metadata:

title : track 13

Chapter #0.13: start 1238.746667, end 1393.866667 Metadata:

title : track 14

Chapter #0.14: start 1393.866667, end 1467.586667 Metadata:

title : track 15

Chapter #0.15: start 1467.586667, end 1548.733333 Metadata:

title : track 16

Chapter #0.16: start 1548.733333, end 1595.493333 Metadata:

title : track 17

Chapter #0.17: start 1595.493333, end 1720.413333 Metadata:

title : track 18

Chapter #0.18: start 1720.413333, end 1801.480000 Metadata:

title : track 19

Chapter #0.19: start 1801.480000, end 1932.146667 Metadata:

title : track 20

Chapter #0.20: start 1932.146667, end 1969.053333 Metadata:

title : track 21

Chapter #0.21: start 1969.053333, end 2135.746667 Metadata:

title : track 22

Chapter #0.22: start 2135.746667, end 2176.933333 Metadata:

title : track 23

Chapter #0.23: start 2176.933333, end 2361.613333 Metadata:

title : track 24

Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Metadata:

encoder : Lavc55.69.100 pcm_s16le

Stream mapping:

Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))

Press [q] to stop, ? for help size= 0kB time=00:00:00.00 bitrate=N/A video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)


          #5297: mpeg-4 als: "predictor order too large"   

http://www.datafilehost.com/d/e51ea572

C:\>mp4alsRM23.exe -o1023 -MP4 pred.wav
C:\>ffmpeg -i pred.mp4 out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pred.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-03-06 18:17:21
  Duration: 00:00:30.01, start: 0.000000, bitrate: 916 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 44100 Hz, 2 channels, s
16, 916 kb/s (default)
    Metadata:
      creation_time   : 2016-03-06 18:17:21
Output #0, wav, to 'out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo
, s16, 1411 kb/s (default)
    Metadata:
      creation_time   : 2016-03-06 18:17:21
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[output stream 0:0 @ 038f2740] 100 buffers queued in output stream 0:0, somethin
g may be wrong.
[als @ 038f7020] Predictor order too large.
[als @ 038f7020] Reading frame data failed. Skipping RA unit.
size=    5171kB time=00:00:30.01 bitrate=1411.2kbits/s speed=9.02x
video:0kB audio:5170kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%

          #5671: FFmpeg unable to decode PCM in MXF   

Summary of the bug: FFmpeg is unable to decode PCM which is wrapped in an MXF file. The MXF file is Avid compatible, but was apparently not created with Avid. Sample can be found here: http://dl.flavoursys.com/samples/ffmpeg/TVOKids_6.mxf

How to reproduce:

% $ ffmpeg -i TVOKids_6.mxf -loglevel debug -y out.wav                                                                                                                                                                                                 [17:02:14]
ffmpeg version N-80780-gd693392 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.1 (Ubuntu 5.3.1-14ubuntu2.1) 20160413
  configuration: --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-version3 --enable-libfdk-aac --enable-libx264 --enable-libfreetype
  libavutil      55. 27.100 / 55. 27.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 40.101 / 57. 40.101
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 46.102 /  6. 46.102
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Splitting the commandline.
Reading option '-i' ... matched as input file with argument 'TVOKids_6.mxf'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'out.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file TVOKids_6.mxf.
Successfully parsed a group of options.
Opening an input file: TVOKids_6.mxf.
[file @ 0x362acc0] Setting default whitelist 'file,crypto'
[mxf @ 0x362a380] Format mxf probed with size=2048 and score=100
[mxf @ 0x362a380] Dark key 06.0e.2b.34.02.53.01.01.0d.01.01.01.01.01.23.00
[mxf @ 0x362a380] Dark key 06.0e.2b.34.01.01.01.02.03.01.02.10.01.00.00.00
    Last message repeated 2 times
[mxf @ 0x362a380] Dark key 06.0e.2b.34.02.05.01.01.0d.01.02.01.01.11.01.00
[mxf @ 0x362a380] none: Universal Label: 00000000.0000.0000.00000000.00000000
[mxf @ 0x362a380] Before avformat_find_stream_info() pos: 32744 bytes read:66172 seeks:5 nb_streams:1
[mxf @ 0x362a380] IndexSID 1 segment at 0 missing EditUnitByteCount and IndexEntryArray
[mxf @ 0x362a380] After avformat_find_stream_info() pos: 32744 bytes read:66172 seeks:5 frames:0
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, mxf, from 'TVOKids_6.mxf':
  Metadata:
    uid             : 27023931-e9a3-4549-946d-a5aea01cec3e
    generation_uid  : bd18e86b-ee7a-4f8f-8ceb-7ab5ac3f175f
    company_name    : Matrox Electronic
    product_name    : DSX
    product_version : 4.0
    product_uid     : 060e2b34-0401-0109-0e0e-010101000001
    modification_date: 2016-05-31 17:30:17
    material_package_umid: 0x060A2B340101010501010D20130000004F5296D2A16741ED9D11A972EA7C9C30
    material_package_name: 0004PE
    timecode        : 11:22:33;00
  Duration: 00:00:18.95, start: 0.000000, bitrate: 1166 kb/s
    Stream #0:0, 0, 1/48000: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010D20130000000B45AF7186674D708C09A0FCF3D5B364
Successfully opened the file.
Parsing a group of options: output file out.wav.
Successfully parsed a group of options.
Opening an output file: out.wav.
[file @ 0x3631540] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 16 logical cores
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'sample_fmt' to value 's32'
[graph 0 input from stream 0:0 @ 0x3643980] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0x3643980] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0x3644560] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x3644560] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x3631b80] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x3645d40] [SWR @ 0x3646120] Using fltp internally between filters
[auto-inserted resampler 0 @ 0x3645d40] ch:1 chl:mono fmt:s32 r:48000Hz -> ch:1 chl:mono fmt:s16 r:48000Hz
[wav @ 0x362fa00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    uid             : 27023931-e9a3-4549-946d-a5aea01cec3e
    generation_uid  : bd18e86b-ee7a-4f8f-8ceb-7ab5ac3f175f
    company_name    : Matrox Electronic
    product_name    : DSX
    product_version : 4.0
    product_uid     : 060e2b34-0401-0109-0e0e-010101000001
    modification_date: 2016-05-31 17:30:17
    material_package_umid: 0x060A2B340101010501010D20130000004F5296D2A16741ED9D11A972EA7C9C30
    material_package_name: 0004PE
    ISMP            : 11:22:33;00
    ISFT            : Lavf57.40.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16 (24 bit), 768 kb/s
    Metadata:
      file_package_umid: 0x060A2B340101010501010D20130000000B45AF7186674D708C09A0FCF3D5B364
      encoder         : Lavc57.48.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mxf @ 0x362a380] IndexSID 1 segment at 0 missing EditUnitByteCount and IndexEntryArray
TVOKids_6.mxf: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    Last message repeated 1 times
[output stream 0:0 @ 0x36444c0] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (TVOKids_6.mxf):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (out.wav):
  Output stream #0:0 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes);
  Total: 0 packets (0 bytes) muxed
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x3631640] Statistics: 4 seeks, 3 writeouts
[AVIOContext @ 0x3633020] Statistics: 66172 bytes read, 5 seeks

          #6106: Some pcm wavs detected as eac3   

Summary of the bug:

I use ffmpeg extensively to identify file formats.

There are some pcm wavs that are mis-identified as varieties of ac3 - recompiling with --disable-demuxer=ac3 fixes it, but that's not nice if you also want to be able to deal with ac3 audio.

This was not a case with a very old FFmpeg I have on a different machine (N-34304-gc0dbab9, built Oct 31 2011), but bisecting 5 years seems... unproductive.

How to reproduce:

Compile ffmpeg (git master) with default settings, and run ffmpeg -i with the file I'm supplying.

$ ./ffmpeg -i pcm_wav_misidentified_as_eac3.wav
ffmpeg version N-83248-gb4a13d442a Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration:
  libavutil      55. 45.100 / 55. 45.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 65.100 / 57. 65.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 70.100 /  6. 70.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
Input #0, wav, from 'pcm_wav_misidentified_as_eac3.wav':
  Duration: 00:00:05.00, bitrate: 1411 kb/s
    Stream #0:0: Audio: eac3 ([1][0][0][0] / 0x0001), 48000 Hz, 5.0(side), fltp, 64 kb/s (visual impaired)
At least one output file must be specified

Built Jan 25 13:03 CET.

Additional comment:

Based on what I read on http://soundfile.sapp.org/doc/WaveFormat/ the stream shouldn't even have to be interpreted by all different codecs as long as the AudioFormat? field (at bytes 20-21) is 0x0001 ? Just a thought...

Full output:

$ ./ffmpeg -v 9 -loglevel 99 -report -i pcm_wav_misidentified_as_eac3.wav
ffmpeg started on 2017-01-27 at 11:50:13
Report written to "ffmpeg-20170127-115013.log"
ffmpeg version N-83248-gb4a13d442a Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration:
  libavutil      55. 45.100 / 55. 45.100
  libavcodec     57. 75.100 / 57. 75.100
  libavformat    57. 65.100 / 57. 65.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 70.100 /  6. 70.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
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 '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-i' ... matched as input url with argument 'pcm_wav_misidentified_as_eac3.wav'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url pcm_wav_misidentified_as_eac3.wav.
Successfully parsed a group of options.
Opening an input file: pcm_wav_misidentified_as_eac3.wav.
[file @ 0x7fb764d02680] Setting default whitelist 'file,crypto'
Probing mp3 score:1 size:2048
Probing wav score:99 size:2048
[wav @ 0x7fb765800000] Format wav probed with size=2048 and score=99
[wav @ 0x7fb765800000] Before avformat_find_stream_info() pos: 44 bytes read:65580 seeks:1 nb_streams:1
[wav @ 0x7fb765800000] probing stream 0 pp:32
Probing mp3 score:1 size:4096
[wav @ 0x7fb765800000] Probe with size=4096, packets=2469 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:31
Probing mp3 score:1 size:8192
[wav @ 0x7fb765800000] Probe with size=8192, packets=2470 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:30
[wav @ 0x7fb765800000] probing stream 0 pp:29
Probing mp3 score:1 size:16384
[wav @ 0x7fb765800000] Probe with size=16384, packets=2472 detected mp3 with score=1
[wav @ 0x7fb765800000] probing stream 0 pp:28
[wav @ 0x7fb765800000] probing stream 0 pp:27
[wav @ 0x7fb765800000] probing stream 0 pp:26
[wav @ 0x7fb765800000] probing stream 0 pp:25
[wav @ 0x7fb765800000] probing stream 0 pp:24
[wav @ 0x7fb765800000] probing stream 0 pp:23
[wav @ 0x7fb765800000] probing stream 0 pp:22
[wav @ 0x7fb765800000] probing stream 0 pp:21
[wav @ 0x7fb765800000] probing stream 0 pp:20
[wav @ 0x7fb765800000] probing stream 0 pp:19
[wav @ 0x7fb765800000] probing stream 0 pp:18
[wav @ 0x7fb765800000] probing stream 0 pp:17
Probing ac3 score:25 size:65536
[wav @ 0x7fb765800000] Probe with size=65536, packets=2484 detected ac3 with score=25
[wav @ 0x7fb765800000] probing stream 0 pp:16
[wav @ 0x7fb765800000] probing stream 0 pp:15
[wav @ 0x7fb765800000] probing stream 0 pp:14
[wav @ 0x7fb765800000] probing stream 0 pp:13
[wav @ 0x7fb765800000] probing stream 0 pp:12
[wav @ 0x7fb765800000] probing stream 0 pp:11
[wav @ 0x7fb765800000] probing stream 0 pp:10
[wav @ 0x7fb765800000] probing stream 0 pp:9
[wav @ 0x7fb765800000] probing stream 0 pp:8
[wav @ 0x7fb765800000] probing stream 0 pp:7
[wav @ 0x7fb765800000] probing stream 0 pp:6
[wav @ 0x7fb765800000] probing stream 0 pp:5
[wav @ 0x7fb765800000] probing stream 0 pp:4
[wav @ 0x7fb765800000] probing stream 0 pp:3
[wav @ 0x7fb765800000] probing stream 0 pp:2
[wav @ 0x7fb765800000] probing stream 0 pp:1
Probing ac3 score:50 size:131072
[wav @ 0x7fb765800000] Probe with size=131072, packets=2500 detected ac3 with score=50
[wav @ 0x7fb765800000] probed stream 0
[wav @ 0x7fb765800000] stream 0: start_time: -209146758205323.719 duration: 5.000
[wav @ 0x7fb765800000] format: start_time: -9223372036854.775 duration: 5.000 bitrate=1411 kb/s
[wav @ 0x7fb765800000] After avformat_find_stream_info() pos: 882044 bytes read:947580 seeks:1 frames:16
Input #0, wav, from 'pcm_wav_misidentified_as_eac3.wav':
  Duration: 00:00:05.00, bitrate: 1411 kb/s
    Stream #0:0, 16, 1/44100: Audio: eac3 ([1][0][0][0] / 0x0001), 48000 Hz, 5.0(side), fltp, 64 kb/s (visual impaired)
Successfully opened the file.
At least one output file must be specified
[AVIOContext @ 0x7fb764d028c0] Statistics: 947580 bytes read, 1 seeks

          #6112: ffmpeg not responding, stuck at converting flac to wav file   

I am trying to convert a flac to a wav file. Unfortunately the ffmpeg is getting stuck in the conversion process. I have tried with multiple ffmpeg versions (including the last one) and the same issue occurs.

I have attached the source flac file (AFN8429768236582_01_11_FLAC.flac)

How to reproduce:

% ffmpeg.exe -i AFN8429768236582_01_11_FLAC.flac -y -ar 44100 -ac 2 -sample_fmt s16 AFN8429768236582_01_11_wav.wav
ffmpeg version N-67063-g282c935 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct 20 2014 22:01:37 with gcc 4.9.1 (GCC)

Below if the output with the debug log level:

ffmpeg.exe -v 9 -loglevel 99 -i AFN8429768236582_01_11_FLAC.flac -y -ar 44100 -ac 2 -sample_fmt s16 AFN8429768236582_01_11_wav.wav
ffmpeg version N-67063-g282c935 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct 20 2014 22:01:37 with gcc 4.9.1 (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 --e
nable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libw
ebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      54. 10.100 / 54. 10.100
  libavcodec     56.  8.102 / 56.  8.102
  libavformat    56.  9.101 / 56.  9.101
  libavdevice    56.  1.100 / 56.  1.100
  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 '-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 'AFN8429768236582_01_11_FLAC.flac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-sample_fmt' ... matched as option 'sample_fmt' (set sample format) with argument 's16'.
Reading option 'AFN8429768236582_01_11_wav.wav' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file AFN8429768236582_01_11_FLAC.flac.
Successfully parsed a group of options.
Opening an input file: AFN8429768236582_01_11_FLAC.flac.
[flac @ 04485720] Format flac probed with size=2048 and score=50
[flac @ 04485720] Before avformat_find_stream_info() pos: 8288 bytes read:32768 seeks:0
[flac @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 14 times
[flac @ 04485720] All info found
[flac @ 04485720] After avformat_find_stream_info() pos: 73728 bytes read:98304 seeks:0 frames:1
Input #0, flac, from 'AFN8429768236582_01_11_FLAC.flac':
  Metadata:
    ENCODER         : Lavf55.12.100
  Duration: 00:01:54.81, start: 0.000000, bitrate: 830 kb/s
    Stream #0:0, 1, 1/44100: Audio: flac, 44100 Hz, stereo, s16
Successfully opened the file.
Parsing a group of options: output file AFN8429768236582_01_11_wav.wav.
Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
Applying option ac (set number of audio channels) with argument 2.
Applying option sample_fmt (set sample format) with argument s16.
Successfully parsed a group of options.
Opening an output file: AFN8429768236582_01_11_wav.wav.
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0446e820] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0446e820] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0446e820] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0446ed80] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0446ed80] Setting 'sample_rates' to value '44100'
[audio format for output stream 0:0 @ 0446ed80] Setting 'channel_layouts' to value '0x3'
[AVFilterGraph @ 02b3fe80] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Output #0, wav, to 'AFN8429768236582_01_11_wav.wav':
  Metadata:
    ISFT            : Lavf56.9.101
    Stream #0:0, 0, 1/44100: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc56.8.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 187 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 171 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 157 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 157 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 169 times
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 7 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 70650 (frame 436) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 61719 (frame 435) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 52626 (frame 434) to 74071 (frame 243)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 62483 (frame 436) to 83256 (frame 244)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 53552 (frame 435) to 83256 (frame 244)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
[NULL @ 0446fc60] crc check failed from offset 53909 (frame 436) to 91005 (frame 245)
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample rate change detected in adjacent frames
[NULL @ 0446fc60] bits per sample change detected in adjacent frames
[NULL @ 0446fc60] sample/frame number mismatch in adjacent frames
    Last message repeated 1 times

Thanks! Ovidiu


          #6179: ffmpeg cannot recognize an wave file PCM detected as mp1   

Summary of the bug:when i use ffmpeg convert an wave file,the ffmpeg cannot recognize my file and probe result is mp1 How to reproduce:

% ./ffmpeg -i wav_format_recongnize_error.wav out.wav
ffmpeg version is 3.2.4
built with gcc 4.1.2 (GCC) 20070115 (prerelease) (SUSE Linux)

The full report is below:

ffmpeg started on 2017-02-22 at 22:00:00
Report written to "ffmpeg-20170222-220000.log"
Command line:
./ffmpeg -report -v 9 -loglevel 99 -i wav_format_recongnize_error.wav out.wav
ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 4.1.2 (GCC) 20070115 (prerelease) (SUSE Linux)
  configuration: --prefix=/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/ffmpeg_3.2.4_video --disable-yasm --enable-static --disable-shared --enable-libx264 --enable-gpl --extra-cflags=-I/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/x264_static/include --extra-ldflags=-L/data/AutoBuild/src/isd_musicadmin_proj/trunk/biz_pack/lesterkong/x264_static/lib --extra-libs='-lx264 -ldl' --enable-cross-compile --arch=i486 --target-os=linux
  libavutil      55. 34.101 / 55. 34.101
  libavcodec     57. 64.101 / 57. 64.101
  libavformat    57. 56.101 / 57. 56.101
  libavdevice    57.  1.100 / 57.  1.100
  libavfilter     6. 65.100 /  6. 65.100
  libswscale      4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc    54.  1.100 / 54.  1.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
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 url with argument 'wav_format_recongnize_error.wav'.
Reading option 'out.wav' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option v (set logging level) with argument 9.
Applying option loglevel (set logging level) with argument 99.
Successfully parsed a group of options.
Parsing a group of options: input url wav_format_recongnize_error.wav.
Successfully parsed a group of options.
Opening an input file: wav_format_recongnize_error.wav.
[file @ 0x97cfb20] Setting default whitelist 'file,crypto'
[wav @ 0x97cf400] Format wav probed with size=2048 and score=99
[wav @ 0x97cf400] Before avformat_find_stream_info() pos: 44 bytes read:65580 seeks:1 nb_streams:1
[wav @ 0x97cf400] probing stream 0 pp:32
[wav @ 0x97cf400] probing stream 0 pp:31
[wav @ 0x97cf400] probing stream 0 pp:30
[wav @ 0x97cf400] probing stream 0 pp:29
[wav @ 0x97cf400] Probe with size=16384, packets=2472 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:28
[wav @ 0x97cf400] probing stream 0 pp:27
[wav @ 0x97cf400] probing stream 0 pp:26
[wav @ 0x97cf400] probing stream 0 pp:25
[wav @ 0x97cf400] Probe with size=32768, packets=2476 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:24
[wav @ 0x97cf400] probing stream 0 pp:23
[wav @ 0x97cf400] probing stream 0 pp:22
[wav @ 0x97cf400] probing stream 0 pp:21
[wav @ 0x97cf400] probing stream 0 pp:20
[wav @ 0x97cf400] probing stream 0 pp:19
[wav @ 0x97cf400] probing stream 0 pp:18
[wav @ 0x97cf400] probing stream 0 pp:17
[wav @ 0x97cf400] Probe with size=65536, packets=2484 detected mp3 with score=25
[wav @ 0x97cf400] probing stream 0 pp:16
[wav @ 0x97cf400] probing stream 0 pp:15
[wav @ 0x97cf400] probing stream 0 pp:14
[wav @ 0x97cf400] probing stream 0 pp:13
[wav @ 0x97cf400] probing stream 0 pp:12
[wav @ 0x97cf400] probing stream 0 pp:11
[wav @ 0x97cf400] probing stream 0 pp:10
[wav @ 0x97cf400] probing stream 0 pp:9
[wav @ 0x97cf400] probing stream 0 pp:8
[wav @ 0x97cf400] probing stream 0 pp:7
[wav @ 0x97cf400] probing stream 0 pp:6
[wav @ 0x97cf400] probing stream 0 pp:5
[wav @ 0x97cf400] probing stream 0 pp:4
[wav @ 0x97cf400] probing stream 0 pp:3
[wav @ 0x97cf400] probing stream 0 pp:2
[wav @ 0x97cf400] probing stream 0 pp:1
[wav @ 0x97cf400] Probe with size=131072, packets=2500 detected mp3 with score=50
[wav @ 0x97cf400] probed stream 0
[mp1 @ 0x97d0500] Header missing
[mp1 @ 0x97d0500] Header missing
[wav @ 0x97cf400] All info found
[wav @ 0x97cf400] Estimating duration from bitrate, this may be inaccurate
[wav @ 0x97cf400] After avformat_find_stream_info() pos: 131116 bytes read:196652 seeks:1 frames:50
Input #0, wav, from 'wav_format_recongnize_error.wav':
  Duration: 00:00:58.05, bitrate: 1411 kb/s
    Stream #0:0, 50, 1/44100: Audio: mp1 ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16p, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output url out.wav.
Successfully parsed a group of options.
Opening an output file: out.wav.
[file @ 0x97d3760] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 6 logical cores
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'sample_fmt' to value 's16p'
[graph 0 input from stream 0:0 @ 0x980dec0] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0x980dec0] tb:1/48000 samplefmt:s16p samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0x97e5700] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0x97e5700] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 0x98036e0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x97d0b60] [SWR @ 0x9817580] Using s16p internally between filters
[auto-inserted resampler 0 @ 0x97d0b60] ch:1 chl:mono fmt:s16p r:48000Hz -> ch:1 chl:mono fmt:s16 r:48000Hz
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.56.101
    Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      encoder         : Lavc57.64.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp1 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[mp1 @ 0x97d31c0] Header missing
.....
Error while decoding stream #0:0: Invalid data found when processing input
[mp1 @ 0x97d31c0] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
[output stream 0:0 @ 0x97d2340] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
size=     620kB time=00:00:06.61 bitrate= 767.0kbits/s speed=7.94x
video:0kB audio:620kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.012296%
Input file #0 (wav_format_recongnize_error.wav):
  Input stream #0:0 (audio): 4837 packets read (10239956 bytes); 827 frames decoded (317568 samples);
  Total: 4837 packets (10239956 bytes) demuxed
Output file #0 (out.wav):
  Output stream #0:0 (audio): 826 frames encoded (317184 samples); 826 packets muxed (634368 bytes);
  Total: 826 packets (634368 bytes) muxed
827 frames successfully decoded, 4010 decoding errors
[AVIOContext @ 0x97e0a60] Statistics: 4 seeks, 829 writeouts
[AVIOContext @ 0x97d7cc0] Statistics: 10305536 bytes read, 1 seeks
Conversion failed!}}}

          #6349: libmp3lame encode segfaults with audio filter   

Summary of the bug:

FFmpeg segfaults with this file when encoding with libmp3lame with an audio filter. The same segfault happens when using one of several other audio filters. FFmpeg was just built from the current git head (dfc4ce5f5ddb7ae0934bb3ca40b99932cf3e1cb0).

ffprobe file info:

Input #0, wav, from 'clip.wav':
  Metadata:
    date            : 2017-01-0
    encoder         : Lavf57.72.101
    encoded_by      : Pro Tools
  Duration: 00:00:15.02, bitrate: 1152 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s

How to reproduce:

ffmpeg -i clip.wav -af alimiter -c:a mp3 -b:a 128k -ar 48k -f null -
ffmpeg version N-85642-gdfc4ce5 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 7.0.2 (clang-700.1.81)
  configuration: --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda --enable-libvpx --enable-libopus
  libavutil      55. 61.100 / 55. 61.100
  libavcodec     57. 93.100 / 57. 93.100
  libavformat    57. 72.101 / 57. 72.101
  libavdevice    57.  7.100 / 57.  7.100
  libavfilter     6. 87.100 /  6. 87.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale      4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc    54.  6.100 / 54.  6.100
Input #0, wav, from 'clip.wav':
  Metadata:
    date            : 2017-01-0
    encoder         : Lavf57.72.101
    encoded_by      : Pro Tools
  Duration: 00:00:15.02, bitrate: 1152 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':
  Metadata:
    date            : 2017-01-0
    encoded_by      : Pro Tools
    encoder         : Lavf57.72.101
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, mono, fltp (24 bit), 128 kb/s
    Metadata:
      encoder         : Lavc57.93.100 libmp3lame
zsh: segmentation fault  ffmpeg -i clip.wav -af alimiter -c:a mp3 -b:a 128k -ar 48k -f null -

gdb backtrace:

Program received signal SIGSEGV, Segmentation fault.
0x0000000101ba55cd in av_assert0_fpu () from /usr/local/lib/libavutil.55.dylib
(gdb) backtrace
#0  0x0000000101ba55cd in av_assert0_fpu () from /usr/local/lib/libavutil.55.dylib
#1  0x00000001007bdb10 in av_jni_get_java_vm () from /usr/local/lib/libavcodec.57.dylib
#2  0x00007fff5fbfdc0c in ?? ()
#3  0x0000000102601c80 in ?? ()
#4  0x00000000000001f0 in ?? ()
#5  0x0000000000000006 in ?? ()
#6  0x000000000000000c in ?? ()
#7  0x0000000101ef1000 in ?? ()
#8  0x00007fff5fbfdbb0 in ?? ()
#9  0x00000001026027e0 in ?? ()
#10 0x00007fff5fbfdc0c in ?? ()
#11 0x0000000102601c80 in ?? ()
#12 0x0000000000000000 in ?? ()

MacOS crash report:

Process:               ffmpeg [28218]
Path:                  /usr/local/bin/ffmpeg
Identifier:            ffmpeg
Version:               0
Code Type:             X86-64 (Native)
Parent Process:        zsh [15966]
Responsible:           Terminal [90035]
User ID:               1921560808
Date/Time:             2017-04-24 14:15:16.635 -0500
OS Version:            Mac OS X 10.10.5 (14F2109)
Report Version:        11
Anonymous UUID:        9A022765-A1D6-4133-507D-AF6B8EDF895E
Sleep/Wake UUID:       EE41D361-72E6-423A-86A8-FC2740611B59
Time Awake Since Boot: 530000 seconds
Time Since Wake:       4300 seconds
Crashed Thread:        0  Dispatch queue: com.apple.main-thread
Exception Type:        EXC_BAD_ACCESS (SIGSEGV)
Exception Codes:       EXC_I386_GPFLT
Thread 0 Crashed:: Dispatch queue: com.apple.main-thread
0   libavutil.55.dylib            	0x00000001098f25cd av_assert0_fpu + 1485
Thread 1:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 2:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 3:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 4:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 5:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 6:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 7:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 8:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 9:
0   libsystem_kernel.dylib        	0x00007fff96431136 __psynch_cvwait + 10
Thread 0 crashed with X86 Thread State (64-bit):
  rax: 0x00000000000002d0  rbx: 0x00007fa2b042a780  rcx: 0x00007fa2b0423100  rdx: 0x0000000000000b30
  rdi: 0x00007fa2b080f600  rsi: 0x00007fa2b089a014  rbp: 0x0000000000000000  rsp: 0x00007fff57ebfb68
   r8: 0x00000000000f4240   r9: 0x0000000000000000  r10: 0x000000000007a120  r11: 0x00007fa2b0400000
  r12: 0x00007fa2b0820c00  r13: 0x00007fa2b0429c00  r14: 0x00007fa2b0423b00  r15: 0x00007fa2b0820c00
  rip: 0x00000001098f25cd  rfl: 0x0000000000010202  cr2: 0x0000000109c0b5b0
Logical CPU:     0
Error Code:      0x00000000
Trap Number:     13
Binary Images:
       0x107d3e000 -        0x107d6bfff +ffmpeg (0) <EEABF0E9-23E0-3924-85E0-DC4724D5123E> /usr/local/bin/ffmpeg
       0x107d7e000 -        0x107d86ff7 +libavdevice.57.dylib (0) <3950BA15-C048-3CDB-9EEA-8CAD234DCECE> /usr/local/lib/libavdevice.57.dylib
       0x107d8e000 -        0x107f27fff +libavfilter.6.dylib (0) <EAF7411E-8A52-3FFA-AF89-95BD2B0A9EF5> /usr/local/lib/libavfilter.6.dylib
       0x107f91000 -        0x10812dff7 +libavformat.57.dylib (0) <77435D7B-A942-3BD8-BB97-1D4AAD5254B8> /usr/local/lib/libavformat.57.dylib
       0x108173000 -        0x1090c2f4f +libavcodec.57.dylib (0) <ACA9F35A-557F-311F-B6FD-D24D99EBD1EC> /usr/local/lib/libavcodec.57.dylib
       0x1097ca000 -        0x1097e5ff7 +libavresample.3.dylib (0) <0A355673-F3F3-3989-9CD1-53909B589911> /usr/local/lib/libavresample.3.dylib
       0x1097ec000 -        0x109809ff7 +libpostproc.54.dylib (0) <D4A1CAE3-E011-3C9B-A32B-0FD1E9958590> /usr/local/lib/libpostproc.54.dylib
       0x10980e000 -        0x109826fff +libswresample.2.dylib (0) <80518108-514D-337C-ADBA-27197EB2003D> /usr/local/lib/libswresample.2.dylib
       0x10982c000 -        0x1098afff7 +libswscale.4.dylib (0) <09A2EAE9-8F62-3883-99AF-08B67217C66F> /usr/local/lib/libswscale.4.dylib
       0x1098bd000 -        0x109906ff7 +libavutil.55.dylib (0) <B7957198-75E9-36BF-96CB-21ED3DA10F2E> /usr/local/lib/libavutil.55.dylib
       0x109929000 -        0x1099d2ff7 +libSDL2-2.0.0.dylib (0) <13519299-19D0-3E7B-B282-30793CD07E04> /usr/local/opt/sdl2/lib/libSDL2-2.0.0.dylib
       0x109a0e000 -        0x109a0ffff  com.apple.VideoDecodeAcceleration (1.1 - 8) <B624EA7A-EECA-3A30-9FE9-25C8339535BE> /System/Library/Frameworks/VideoDecodeAcceleration.framework/Versions/A/VideoDecodeAcceleration
       0x109a1a000 -        0x109af9fff +libx264.148.dylib (0) <3F494DDD-A841-3C1A-9B36-2A18A89A7102> /usr/local/opt/x264/lib/libx264.148.dylib
       0x109b8d000 -        0x109bceff7 +libopus.0.dylib (0) <0BFFA36E-D9E6-3ECB-A106-1380FF0E04FE> /usr/local/opt/opus/lib/libopus.0.dylib
       0x109bd6000 -        0x109c0bfff +libmp3lame.0.dylib (0) <2D449F39-3AE5-3C4A-B56D-AF82F6A481F3> /usr/local/opt/lame/lib/libmp3lame.0.dylib
       0x109c48000 -        0x109c4afff  com.apple.ForceFeedback (1.0.6 - 1.0.6) <A411AA46-5DB6-3C6C-B4B4-613EC1EF33C3> /System/Library/Frameworks/ForceFeedback.framework/Versions/A/ForceFeedback
    0x7fff67cb2000 -     0x7fff67ce892f  dyld (353.2.4) <708E385E-50C6-32A3-983C-42E97BEE4346> /usr/lib/dyld
    0x7fff8a0f9000 -     0x7fff8a115ff7  libsystem_malloc.dylib (53.30.1) <DDA8928B-CC0D-3255-BD8A-3FEA0982B890> /usr/lib/system/libsystem_malloc.dylib
    0x7fff8a285000 -     0x7fff8a5a0fcf  com.apple.vImage (8.0 - 8.0) <1183FE6A-FDB6-3B3B-928D-50C7909F2308> /System/Library/Frameworks/Accelerate.framework/Versions/A/Frameworks/vImage.framework/Versions/A/vImage
    0x7fff8a5a1000 -     0x7fff8a5f2ff7  com.apple.AppleVAFramework (5.0.31 - 5.0.31) <17C35538-7C2B-3735-A1B3-E8487EED0668> /System/Library/PrivateFrameworks/AppleVA.framework/Versions/A/AppleVA
    0x7fff8a5f3000 -     0x7fff8a5fcfff  libsystem_pthread.dylib (105.40.1) <ACE90967-ECD0-3251-AEEB-461E3C6414F7> /usr/lib/system/libsystem_pthread.dylib
    0x7fff8a5fd000 -     0x7fff8b17eff7  com.apple.AppKit (6.9 - 1348.17) <E485D56D-3E72-34B7-99BB-BFDEE2D07BF5> /System/Library/Frameworks/AppKit.framework/Versions/C/AppKit
    0x7fff8bace000 -     0x7fff8bb3fffb  com.apple.ApplicationServices.ATS (360 - 375.4) <888FE0B1-835C-30B1-80AD-5066CADB0364> /System/Library/Frameworks/ApplicationServices.framework/Versions/A/Frameworks/ATS.framework/Versions/A/ATS
    0x7fff8bb62000 -     0x7fff8bb6bff7  libsystem_notify.dylib (133.1.1) <61147800-F320-3DAA-850C-BADF33855F29> /usr/lib/system/libsystem_notify.dylib
    0x7fff8bb6c000 -     0x7fff8bb88fff  com.apple.GenerationalStorage (2.0 - 209.11) <9FF8DD11-25FB-3047-A5BF-9415339B3EEC> /System/Library/PrivateFrameworks/GenerationalStorage.framework/Versions/A/GenerationalStorage
    0x7fff8be5c000 -     0x7fff8be76ff3  com.apple.Ubiquity (1.3 - 313) <DF56A657-CC6E-3BE2-86A0-71F07127724C> /System/Library/PrivateFrameworks/Ubiquity.framework/Versions/A/Ubiquity
    0x7fff8be90000 -     0x7fff8bf2fe27  com.apple.AppleJPEG (1.0 - 1) <6627DDD9-A8FE-3968-B23A-B6A29AA3919A> /System/Library/PrivateFrameworks/AppleJPEG.framework/Versions/A/AppleJPEG
    0x7fff8c1cd000 -     0x7fff8c2f5ff7  com.apple.coreui (2.1 - 308.6) <9E0E9C6A-68F5-34C1-A17C-96226D401D4D> /System/Library/PrivateFrameworks/CoreUI.framework/Versions/A/CoreUI
    0x7fff8c2f6000 -     0x7fff8c2f6fff  com.apple.CoreServices (62 - 62) <C69DA8A7-B536-34BF-A93F-1C170E2C6D58> /System/Library/Frameworks/CoreServices.framework/Versions/A/CoreServices
    0x7fff8c2f7000 -     0x7fff8c30eff7  libLinearAlgebra.dylib (1128) <E78CCBAA-A999-3B65-8EC9-06DB15E67C37> /System/Library/Frameworks/Accelerate.framework/Versions/A/Frameworks/vecLib.framework/Versions/A/libLinearAlgebra.dylib
    0x7fff8c30f000 -     0x7fff8c315ff7  libsystem_networkextension.dylib (167.40.3) <BA58B30B-8377-3B0A-8AE3-4F84021D9D4E> /usr/lib/system/libsystem_networkextension.dylib
    0x7fff8c358000 -     0x7fff8c3d9ff7  com.apple.CoreUtils (1.1 - 110.1) <3808CBC5-1DF1-3821-8AAE-57C30D4F514B> /System/Library/PrivateFrameworks/CoreUtils.framework/Versions/A/CoreUtils
    0x7fff8c415000 -     0x7fff8c506fff  libJP2.dylib (1247) <D74A8708-C27E-3D7C-89D0-BE999596E2B1> /System/Library/Frameworks/ImageIO.framework/Versions/A/Resources/libJP2.dylib
    0x7fff8c87f000 -     0x7fff8c8f3ffb  com.apple.securityfoundation (6.0 - 55126) <344401F5-4016-3AEB-BFDC-CE8E5546D39F> /System/Library/Frameworks/SecurityFoundation.framework/Versions/A/SecurityFoundation
    0x7fff8cb53000 -     0x7fff8cbbffff  com.apple.framework.CoreWLAN (5.0 - 500.35.2) <03697149-1CDD-32FF-B564-1C1EF5E9E5C3> /System/Library/Frameworks/CoreWLAN.framework/Versions/A/CoreWLAN
    0x7fff8cbc3000 -     0x7fff8cd73ff3  com.apple.QuartzCore (1.10 - 361.19) <E59000AA-DC30-3F73-A103-0240576C9556> /System/Library/Frameworks/QuartzCore.framework/Versions/A/QuartzCore
    0x7fff8cd74000 -     0x7fff8cda1fff  com.apple.CoreVideo (1.8 - 145.1) <18DB07E0-B927-3260-A234-636F298D1917> /System/Library/Frameworks/CoreVideo.framework/Versions/A/CoreVideo
    0x7fff8cdff000 -     0x7fff8cf2ffff  com.apple.UIFoundation (1.0 - 1) <466BDFA8-0B9F-3AB0-989D-F9779422926A> /System/Library/PrivateFrameworks/UIFoundation.framework/Versions/A/UIFoundation
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    0x7fff97251000 -     0x7fff97251fff  libOpenScriptingUtil.dylib (162.2) <D6A2216D-ADB2-3F24-AD30-F6D00829F545> /usr/lib/libOpenScriptingUtil.dylib
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    0x7fff97285000 -     0x7fff9728aff7  libmacho.dylib (862) <126CA2ED-DE91-308F-8881-B9DAEC3C63B6> /usr/lib/system/libmacho.dylib
    0x7fff9728b000 -     0x7fff9728bfff  com.apple.ApplicationServices (48 - 48) <5BF7910B-C328-3BF8-BA4F-CE52B574CE01> /System/Library/Frameworks/ApplicationServices.framework/Versions/A/ApplicationServices
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    0x7fff98c06000 -     0x7fff98c10ff7  com.apple.NetAuth (5.2 - 5.2) <2BBD749A-8E18-35B8-8E48-A90347C1CCA7> /System/Library/PrivateFrameworks/NetAuth.framework/Versions/A/NetAuth
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External Modification Summary:
  Calls made by other processes targeting this process:
    task_for_pid: 0
    thread_create: 0
    thread_set_state: 0
  Calls made by this process:
    task_for_pid: 0
    thread_create: 0
    thread_set_state: 0
  Calls made by all processes on this machine:
    task_for_pid: 3299541
    thread_create: 0
    thread_set_state: 57
VM Region Summary:
ReadOnly portion of Libraries: Total=213.5M resident=146.4M(69%) swapped_out_or_unallocated=67.1M(31%)
Writable regions: Total=49.3M written=2932K(6%) resident=3192K(6%) swapped_out=0K(0%) unallocated=46.2M(94%)
REGION TYPE                      VIRTUAL
===========                      =======
Kernel Alloc Once                     4K
MALLOC                             28.5M
MALLOC (admin)                       32K
MALLOC_LARGE (reserved)             132K        reserved VM address space (unallocated)
STACK GUARD                        56.0M
Stack                              12.6M
VM_ALLOCATE                           8K
__DATA                             23.7M
__IMAGE                             528K
__LINKEDIT                         71.5M
__TEXT                            142.1M
__UNICODE                           552K
shared memory                         4K
===========                      =======
TOTAL                             335.5M
TOTAL, minus reserved VM space    335.4M

          #1659: g723_1: wrong current duration after seeking   

ffplay displays wrong current duration after seeking in g723_1 (in wav)

C:\>ffmpeg -i g723_1.wav
ffmpeg version N-43678-gceb8773 Copyright (c) 2000-2012 the FFmpeg developers
  built on Aug 18 2012 02:22:03 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 70.100 / 51. 70.100
  libavcodec     54. 53.100 / 54. 53.100
  libavformat    54. 25.100 / 54. 25.100
  libavdevice    54.  2.100 / 54.  2.100
  libavfilter     3. 10.100 /  3. 10.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[wav @ 0x1eace20] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from 'g723_1.wav':
  Duration: 00:00:30.07, bitrate: 6 kb/s
    Stream #0:0: Audio: g723_1 (B[0][0][0] / 0x0042), 8000 Hz, mono, s16, 6 kb/s
At least one output file must be specified

          #1712: Audio precision in lossless and long time to encode.   

Summary of the bug: Hi, first of all, i'm sorry for my english. I've tried the last ffmpeg build and i get a problem with Audio precision in lossless audio file, the audio is delayed about 2 seconds. The audio is passed to another program throught a pipe, but i know that the other program isn't the problem, because with a very old ffmpeg version (SVN 16573, Jan 13 2009) works fine, and i've tried with wav output without pipe and get the same problem.

If you want, here's a test wav files: http://www.mediafire.com/?1ewceliyjhspedk

The command line that i've used is : ffmpeg_old -i "input.ape" -ss 00:59:18.71 -t 00:00:15.55 out_oldffmpeg.wav ffmpeg -i "input.ape" -ss 00:59:18.71 -t 00:00:15.55 out_newffmpeg.wav

How to reproduce:

% ffmpeg -i input -ss 00:45:25 -t 00:05:25 -ac 2 -f wav -
The output is a pipe to another program.
'''ffmpeg version''' N-44080-gdfb4757
'''built on''' Aug 31 2012 16:58:51 with gcc 4.7.1 (GCC)
configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt
ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass -
-enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-l
ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenj
peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo
ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena
ble-zlib

The other thing that i wan to answer is why the program take a long time to convert a portion of a file, for example: ffmpeg -i input.ape -ss 00:45:15.42 -t 00:00:15.25 output.wav

the program stay a long time without do anything and then start to convert. I take about 2 minutes to convert 15s of audio.


          #1841: raac: fail to seek   

ffmpeg fails to seek above 26s

http://www.datafilehost.com/download-733c41c3.html

C:\>ffmpeg -ss 26 -i raac.rm out.wav
ffmpeg version N-45867-g3318d6b Copyright (c) 2000-2012 the FFmpeg developers
  built on Oct 23 2012 02:39:33 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 77.100 / 51. 77.100
  libavcodec     54. 68.100 / 54. 68.100
  libavformat    54. 33.100 / 54. 33.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 20.104 /  3. 20.104
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[rm @ 0x1edbe20] Invalid stream index 1 for index at pos 1233529
Input #0, rm, from 'raac.rm':
  Metadata:
    title           :
    author          :
    copyright       :
    comment         :
    Creation Date   : 4/21/2012 13:08:18
    Generated By    : Helix Producer SDK 13.1 for Windows, Build 13.1.1.3436
    Modification Date: 4/21/2012 13:08:18
  Duration: 00:00:30.07, start: 0.000000, bitrate: 328 kb/s
    Stream #0:0: Audio: aac (raac / 0x63616172), 44100 Hz, stereo, s16, 320 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    INAM            :
    author          :
    ICOP            :
    ICMT            :
    Creation Date   : 4/21/2012 13:08:18
    Generated By    : Helix Producer SDK 13.1 for Windows, Build 13.1.1.3436
    Modification Date: 4/21/2012 13:08:18
    ISFT            : Lavf54.33.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
[aac @ 0x1edc920] Reserved bit set.
Error while decoding stream #0:0: Invalid data found when processing input
Truncating packet of size 11122 to 7128
[aac @ 0x1edc920] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
raac.rm: Input/output error
size=       0kB time=00:00:00.00 bitrate=   0.0kbits/s
video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters i
f used)

          #1958: mp3 24khz (in wav): overreads   

This happens only with 24khz mode.

I can hear some distortions.

C:\>ffmpeg -i 24khz_2ch_160kbps.wav -t 2 -f null -
ffmpeg version N-46755-ge70144c Copyright (c) 2000-2012 the FFmpeg developers
  built on Nov 15 2012 02:38:20 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      52.  6.100 / 52.  6.100
  libavcodec     54. 71.100 / 54. 71.100
  libavformat    54. 36.100 / 54. 36.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 23.100 /  3. 23.100
  libswscale      2.  1.102 /  2.  1.102
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[wav @ 0x1edc3c0] max_analyze_duration 5000000 reached at 5016000
[wav @ 0x1edc3c0] Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from '24khz_2ch_160kbps.wav':
  Duration: 00:00:33.12, bitrate: 127 kb/s
    Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 24000 Hz, stereo, s16, 128 kb
/s
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf54.36.100
    Stream #0:0: Audio: pcm_s16le, 24000 Hz, stereo, s16, 768 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 -> pcm_s16le)
Press [q] to stop, [?] for help
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -6 -6
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -5 -5
[mp3 @ 0x1edcb60] overread, skip -9 enddists: -1 -1
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -3 -3
    Last message repeated 1 times
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -2 -2
[mp3 @ 0x1edcb60] overread, skip -6 enddists: -4 -4
[mp3 @ 0x1edcb60] overread, skip -7 enddists: -4 -4
size=       0kB time=00:00:02.01 bitrate=   0.0kbits/s
video:0kB audio:189kB subtitle:0 global headers:0kB muxing overhead -100.000000%

          #2458: raw aac fails with FFmpeg, works fine with faad   

(videolan ticket 8309) I will upload an AAC sample (provided by a vlc user) that decodes fine with faad, FFmpeg fails both auto-detection and decoding.

$ ffmpeg -probesize 2G -analyzeduration 2G -i trac_8309_raw.aac
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
trac_8309_raw.aac: Invalid data found when processing input
$ faad trac_8309_raw.aac
 *********** Ahead Software MPEG-4 AAC Decoder V2.7 ******************
 Build: Apr  3 2012
 Copyright 2002-2004: Ahead Software AG
 http://www.audiocoding.com
 Floating point version
 This program is free software; you can redistribute it and/or modify
 it under the terms of the GNU General Public License.
 **************************************************************************
trac_8309_raw.aac file info:
RAW
  ---------------------
 | Config:  2 Ch       |
  ---------------------
 | Ch |    Position    |
  ---------------------
 | 00 | Left front     |
 | 01 | Right front    |
  ---------------------
Decoding trac_8309_raw.aac took:  0.76 sec.  0.00x real-time.
$ ffmpeg -i trac_8309_raw.wav
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[wav @ 0x1d4ca20] max_analyze_duration 5000000 reached at 5015510 microseconds
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'trac_8309_raw.wav':
  Duration: 00:05:41.22, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
At least one output file must be specified
$ ffmpeg -f aac -i trac_8309_raw.aac
ffmpeg version N-51927-geabc0c7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:23:56 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 52.101 /  3. 52.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] More than one AAC RDB per ADTS frame is 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.
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 3.1 is not allocated
[aac @ 0x1ab25a0] Number of bands (7) exceeds limit (4).
[aac @ 0x1ab25a0] ms_present = 3 is reserved.
[aac @ 0x1ab25a0] Number of bands (15) exceeds limit (12).
[aac @ 0x1ab25a0] channel element 1.13 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 2.5 is not allocated
[aac @ 0x1ab25a0] channel element 3.2 is not allocated
[aac @ 0x1ab25a0] channel element 0.1 is not allocated
[aac @ 0x1ab25a0] Number of bands (6) exceeds limit (2).
[aac @ 0x1ab25a0] channel element 3.8 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] channel element 1.11 is not allocated
[aac @ 0x1ab25a0] channel element 1.0 is not allocated
[aac @ 0x1ab25a0] Number of bands (10) exceeds limit (9).
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] Number of bands (17) exceeds limit (12).
[aac @ 0x1ab25a0] Number of bands (20) exceeds limit (4).
[aac @ 0x1ab25a0] channel element 2.7 is not allocated
[aac @ 0x1ab25a0] channel element 1.10 is not allocated
[aac @ 0x1ab25a0] channel element 0.13 is not allocated
[aac @ 0x1ab25a0] channel element 2.1 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
...
[aac @ 0x1ab25a0] channel element 2.14 is not allocated
[aac @ 0x1ab25a0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Number of bands (40) exceeds limit (32).
[aac @ 0x1ab25a0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x1ab25a0] Inconsistent channel configuration.
[aac @ 0x1ab25a0] get_buffer() failed
[aac @ 0x1ab25a0] Number of bands (30) exceeds limit (13).
[aac @ 0x1ab25a0] channel element 2.7 is not allocated
[aac @ 0x1ab25a0] Reserved bit set.
[aac @ 0x1ab25a0] Dependent coupling is not supported together with LTP
    Last message repeated 31 times
[aac @ 0x1ab25a0] invalid band type
[aac @ 0x1ab25a0] Number of bands (6) exceeds limit (4).
[aac @ 0x1ab25a0] Dependent coupling is not supported together with LTP
    Last message repeated 31 times
[aac @ 0x1aa9a60] max_analyze_duration 5000000 reached at 5120000 microseconds
[aac @ 0x1aa9a60] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'trac_8309_raw.aac':
  Duration: 00:07:48.76, bitrate: 91 kb/s
    Stream #0:0: Audio: aac, 8000 Hz, 7.1(wide), fltp, 91 kb/s
At least one output file must be specified

MPlayer stops after 15 seconds, fails with -ac faad:

$ mplayer trac_8309_raw.aac -demuxer rawaudio -rawaudio format=0xFF -ac ffaac
MPlayer SVN-r36089-4.7 (C) 2000-2013 MPlayer Team
Playing trac_8309_raw.aac.
rawaudio file format detected.
Load subtitles in ./
==========================================================================
Forced audio codec: ffaac
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
libavcodec version 55.2.100 (internal)
AUDIO: 44100 Hz, 2 ch, floatle, 1411.2 kbit/50.00% (ratio: 176400->352800)
Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
AO: [alsa] 44100Hz 2ch floatle (4 bytes per sample)
Video: no video
Starting playback...
A:  15.7 (15.6) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]decode_band_types: Input buffer exhausted before END element found
A:  15.7 (15.7) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.7 is not allocated
A:   0.5 (00.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   1.5 (01.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:   2.5 (02.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   3.5 (03.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Number of bands (4) exceeds limit (3).
A:   4.5 (04.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Number of bands (16) exceeds limit (4).
A:   5.5 (05.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:   6.5 (06.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.7 is not allocated
A:   7.5 (07.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:   8.5 (08.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:   9.5 (09.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]invalid band type
A:  10.5 (10.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 2.3 is not allocated
A:  11.5 (11.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  12.5 (12.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xf75c1540]Inconsistent channel configuration.
[aac @ 0xf75c1540]get_buffer() failed
A:  13.5 (13.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  14.5 (14.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]channel element 3.4 is not allocated
A:  15.5 (15.5) of 30.2 (30.2)  0.2%
[aac @ 0xf75c1540]Reserved bit set.
A:  16.5 (16.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]invalid band type
A:  17.5 (17.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Reserved bit set.
A:  18.5 (18.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Reserved bit set.
A:  19.5 (19.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.6 is not allocated
A:  20.5 (20.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]SBR was found before the first channel element.
[aac @ 0xf75c1540]Reserved bit set.
A:  21.5 (21.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.2 is not allocated
A:  22.5 (22.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.2 is not allocated
A:  23.5 (23.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.0 is not allocated
A:  24.5 (24.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]Number of bands (11) exceeds limit (2).
A:  25.5 (25.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:  26.5 (26.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 3.7 is not allocated
A:  27.5 (27.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.15 is not allocated
A:  28.5 (28.5) of 30.2 (30.2)  0.3%
[aac @ 0xf75c1540]channel element 2.9 is not allocated
A:  29.5 (29.5) of 30.2 (30.2)  0.3%
Exiting... (End of file)

          #3789: shortest command doesnt work when using a filter_complex with audio and video filters   

Summary of the bug: shortest command doesn't work when using a filter_complex with audio and video filters

I uploaded output.mp4 at 8:21 ET on 7/20/14 to upload.ffmpeg.org

The background audio is 4 mins long, but it should be truncated to the video length.

Shortest works if I take the drawtext off the filter_complex

How to reproduce:

ffmpeg -framerate 1/5 -i foo%03d.png -i audio.wav -i speak.wav -filter_complex '[1:a]volume=volume=0.2[aquiet];[aquiet][2:a]amix=inputs=2[a];[0:v]drawtext=fontsize=40:fontcolor=white@0.9:fontfile=FreeSerif.ttf:shadowy=2:shadowx=2:text="I am getting the hang of ffmpeg\!":x=(w-text_w)/2:y=(h-text_h-line_h)/2[v]' -map '[a]' -map '[v]' -shortest -r 32 -c:v libx264 -b:a 192k -pix_fmt yuv420p output.mp4
ffmpeg version 2.2.4-   http://johnvansickle.com/ffmpeg/    Copyright (c) 2000-2014 the FFmpeg developers
  built on Jul  1 2014 22:10:48 with gcc 4.8 (Debian 4.8.3-4)
  configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --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      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, image2, from 'foo%03d.png':
  Duration: 00:00:30.00, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: png, rgb24, 640x360 [SAR 2835:2835 DAR 16:9], 0.20 fps, 1 tbr, 0.20 tbn, 0.20 tbc
Input #1, mp3, from 'audio.wav':
  Metadata:
    major_brand     : dash
    minor_version   : 0
    compatible_brands: iso6mp41
    encoder         : Lavf54.29.104
  Duration: 00:04:29.09, start: 0.025057, bitrate: 238 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 238 kb/s
Guessed Channel Layout for  Input Stream #2.0 : mono
Input #2, wav, from 'speak.wav':
  Duration: 00:00:03.33, bitrate: 256 kb/s
    Stream #2:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
File 'output.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x2fdf7a0] using SAR=1/1
[libx264 @ 0x2fdf7a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x2fdf7a0] profile High, level 3.0
[libx264 @ 0x2fdf7a0] 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=18 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=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 'output.mp4':
  Metadata:
    encoder         : Lavf55.33.100
    Stream #0:0: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 192 kb/s (default)
    Stream #0:1: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 640x360 [SAR 1:1 DAR 16:9], q=-1--1, 16384 tbn, 32 tbc (default)
Stream mapping:
  Stream #0:0 (png) -> drawtext
  Stream #1:0 (mp3) -> volume
  Stream #2:0 (pcm_s16le) -> amix:input1
  amix -> Stream #0:0 (libvo_aacenc)
  drawtext -> Stream #0:1 (libx264)
Press [q] to stop, [?] for help
frame=  800 fps=116 q=-1.0 Lsize=    6606kB time=00:04:29.05 bitrate= 201.1kbits/s dup=794 drop=0
video:238kB audio:6307kB subtitle:0 data:0 global headers:0kB muxing overhead 0.933773%
[libvo_aacenc @ 0x2fde340] 1 frames left in the queue on closing
[libx264 @ 0x2fdf7a0] frame I:5     Avg QP:17.56  size: 37668
[libx264 @ 0x2fdf7a0] frame P:201   Avg QP:17.22  size:   219
[libx264 @ 0x2fdf7a0] frame B:594   Avg QP:25.96  size:    18
[libx264 @ 0x2fdf7a0] consecutive B-frames:  0.8%  0.2%  1.5% 97.5%
[libx264 @ 0x2fdf7a0] mb I  I16..4:  4.7% 64.5% 30.8%
[libx264 @ 0x2fdf7a0] mb P  I16..4:  0.1%  0.2%  0.3%  P16..4:  0.5%  0.0%  0.0%  0.0%  0.0%    skip:99.0%
[libx264 @ 0x2fdf7a0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.1%  0.0%  0.0%  direct: 0.0%  skip:99.9%  L0:16.4% L1:83.6% BI: 0.0%
[libx264 @ 0x2fdf7a0] 8x8 transform intra:58.6% inter:30.7%
[libx264 @ 0x2fdf7a0] coded y,uvDC,uvAC intra: 83.7% 87.7% 75.6% inter: 0.0% 0.1% 0.0%
[libx264 @ 0x2fdf7a0] i16 v,h,dc,p: 50% 11%  7% 32%
[libx264 @ 0x2fdf7a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 15% 19% 21%  6%  6%  5%  8%  6% 13%
[libx264 @ 0x2fdf7a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 22% 11%  6%  7%  6%  7%  5% 12%
[libx264 @ 0x2fdf7a0] i8c dc,h,v,p: 50% 22% 20%  8%
[libx264 @ 0x2fdf7a0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x2fdf7a0] ref P L0: 94.2%  0.5%  3.9%  1.4%
[libx264 @ 0x2fdf7a0] ref B L0: 53.1% 45.3%  1.6%
[libx264 @ 0x2fdf7a0] ref B L1: 95.1%  4.9%
[libx264 @ 0x2fdf7a0] kb/s:77.80

          #4590: concat filter does not work to loop audio   

As reported by irc user chama The following works as expected:

$ ffmpeg -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -filter_complex [0:a][1:a][2:a]concat=n=3:v=0:a=1 out.wav

The output file has a correct length of 2:10

The following claims to encode 2:10 but the output file is much shorter:

$ ffmpeg -i fate-suite/svq3/Vertical400kbit.sorenson3.mov -filter_complex [0:a][0:a][0:a]concat=n=3:v=0:a=1 out.wav
ffmpeg version N-72624-g196b885 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 26.101 / 54. 26.101
  libavcodec     56. 41.101 / 56. 41.101
  libavformat    56. 34.100 / 56. 34.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.1 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'fate-suite/svq3/Vertical400kbit.sorenson3.mov':
  Metadata:
    creation_time   : 2001-03-20 16:17:18
    title           : Vertical Online SV3 Demo
    title-eng       : Vertical Online SV3 Demo
    artist          : Logan Kelsey
    artist-eng      : Logan Kelsey
    copyright       : (c) Vertical Online 2001
    copyright-eng   : (c) Vertical Online 2001
    encoder         : Sorenson Video 3
    encoder-eng     : Sorenson Video 3
  Duration: 00:00:43.58, start: 0.000000, bitrate: 580 kb/s
    Stream #0:0(eng): Video: svq3 (SVQ3 / 0x33515653), yuvj420p(pc), 320x240, 391 kb/s, 30.02 fps, 30 tbr, 600 tbn, 600 tbc (default)
    Metadata:
      creation_time   : 2001-03-20 16:17:18
      handler_name    : Apple Alias Data Handler
      encoder         : Sorenson Video 3
    Stream #0:1(eng): Audio: adpcm_ima_qt (ima4 / 0x34616D69), 44100 Hz, 1 channels, s16p, 176 kb/s (default)
    Metadata:
      creation_time   : 2001-03-20 16:17:18
      handler_name    : Apple Alias Data Handler
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf56.34.100
    INAM            : Vertical Online SV3 Demo
    title-eng       : Vertical Online SV3 Demo
    IART            : Logan Kelsey
    artist-eng      : Logan Kelsey
    ICOP            : (c) Vertical Online 2001
    copyright-eng   : (c) Vertical Online 2001
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s (default)
    Metadata:
      encoder         : Lavc56.41.101 pcm_s16le
Stream mapping:
  Stream #0:1 (adpcm_ima_qt) -> concat:in0:a0
  Stream #0:1 (adpcm_ima_qt) -> concat:in1:a0
  Stream #0:1 (adpcm_ima_qt) -> concat:in2:a0
  concat -> Stream #0:0 (pcm_s16le)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 1
[Parsed_concat_0 @ 0x26bed40] Buffer queue overflow, dropping.
    Last message repeated 59539 times
[output stream 0:0 @ 0x26bf600] 100 buffers queued in output stream 0:0, something may be wrong.
size=    3817kB time=00:02:10.72 bitrate= 239.2kbits/s
video:0kB audio:3817kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004247%
$ ffmpeg -i out.wav
ffmpeg version N-72624-g196b885 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl
  libavutil      54. 26.101 / 54. 26.101
  libavcodec     56. 41.101 / 56. 41.101
  libavformat    56. 34.100 / 56. 34.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, wav, from 'out.wav':
  Metadata:
    artist          : Logan Kelsey
    copyright       : (c) Vertical Online 2001
    title           : Vertical Online SV3 Demo
    encoder         : Lavf56.34.100
  Duration: 00:00:44.32, bitrate: 705 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s
At least one output file must be specified

          #5033: Incorrect duration when converting WAV/MP3 files to AAC with -shortest   

Summary of the bug: Incorrect duration of encoded audio files when converting WAV/MP3 (MP3 created with the same WAV file) to AAC. I assume that this is affecting the duration of MP4 videos created with an still image and audio (which is the reason I'm reporting this bug).

How to reproduce: (espeak is a TTS engine. -w in espeak will save the output speech to a wav file)

$ espeak -s 80 -w /tmp/in.wav 'This product is meant for educational purposes only. Any resemblance to real persons, living or dead is purely coincidental. Void where prohibited. Some assembly required. List each check separately by bank number. Batteries not included. Contents may settle during shipment. Use only as directed. No other warranty expressed or implied. Do not use while operating a motor vehicle or heavy equipment. Postage will be paid by addressee. Subject to CARB approval.'
$ ffmpeg -y -i /tmp/in.wav -vn -c:a aac -strict -2 -ab 24k -ar 16000 /tmp/out.aac
ffmpeg version 2.8.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-23) 20151028
  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 --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, wav, from '/tmp/in.wav':
  Duration: 00:01:00.33, bitrate: 352 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
Output #0, adts, to '/tmp/out.aac':
  Metadata:
    encoder         : Lavf56.40.101
    Stream #0:0: Audio: aac, 16000 Hz, mono, fltp, 24 kb/s
    Metadata:
      encoder         : Lavc56.60.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
size=     195kB time=00:01:00.35 bitrate=  26.5kbits/s
video:0kB audio:189kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.422557%
$ for i in /tmp/in.wav /tmp/out.aac; do ffprobe -i $i 2>&1 | grep Duration; done
  Duration: 00:01:00.33, bitrate: 352 kb/s
  Duration: 00:03:19.68, bitrate: 8 kb/s

If you encode the WAV files to MP3 and then to AAC, once again, you get different, longer audio streams:

$ ffmpeg -y -i /tmp/in.wav -vn -strict -2 -ab 24k -ar 16000 -f mp3 /tmp/out.mp3
...
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
...
size=     177kB time=00:01:00.33 bitrate=  24.1kbits/s
$ ffmpeg -y -i /tmp/out.mp3 -vn -c:a aac -strict -2 -ab 24k -ar 16000 /tmp/out.aac
$ for i in /tmp/out.mp3 /tmp/out.aac; do ffprobe -i $i 2>&1 | grep Duration; done
  Duration: 00:01:00.41, start: 0.069063, bitrate: 24 kb/s
  Duration: 00:01:06.31, bitrate: 23 kb/s

Interestingly, I ran espeak with a different input ('This is some sample text to test audio encoding. ' repeated 10 times) and it gave me a shorter difference between the duration of the encoded output files.

I tried this with OGG/libvorbis files, the duration of output AAC file didn't seem to be affected.

For MP4s:

$ ffprobe -i /tmp/in.wav 2>&1 | grep Duration
  Duration: 00:01:00.33, bitrate: 352 kb/s
$ ffmpeg -y -framerate 1 -r 1 -loop 1 -i /tmp/in.jpg -i /tmp/in.wav -c:v libx264 -preset veryfast -tune stillimage -c:a aac -ab 32k -ar 16000 -strict experimental -shortest -pix_fmt yuv420p -movflags faststart -f mp4 /tmp/out.mp4
ffmpeg version 2.8.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.1 (Debian 5.2.1-23) 20151028
  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 --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --enable-frei0r --enable-libfribidi --disable-indev=sndio --disable-outdev=sndio --cc=gcc
  libavutil      54. 31.100 / 54. 31.100
  libavcodec     56. 60.100 / 56. 60.100
  libavformat    56. 40.101 / 56. 40.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 40.101 /  5. 40.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc    53.  3.100 / 53.  3.100
[mjpeg @ 0x4b1daa0] Changeing bps to 8
Input #0, image2, from '/tmp/in.jpg':
  Duration: 00:00:01.00, start: 0.000000, bitrate: 272 kb/s
    Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 300x300 [SAR 1:1 DAR 1:1], 1 fps, 1 tbr, 1 tbn, 1 tbc
Guessed Channel Layout for  Input Stream #1.0 : mono
Input #1, wav, from '/tmp/in.wav':
  Duration: 00:01:00.33, bitrate: 352 kb/s
    Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 1 channels, s16, 352 kb/s
[swscaler @ 0x4b59e20] deprecated pixel format used, make sure you did set range correctly
[libx264 @ 0x4b7eba0] using SAR=1/1
[libx264 @ 0x4b7eba0] using cpu capabilities: MMX2 SSE2Fast LZCNT
[libx264 @ 0x4b7eba0] profile High, level 1.2
[libx264 @ 0x4b7eba0] 264 - core 148 r209 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:-3:-3 analyse=0x3:0x113 me=hex subme=2 psy=1 psy_rd=2.00:0.70 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=6 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.20
Output #0, mp4, to '/tmp/out.mp4':
  Metadata:
    encoder         : Lavf56.40.101
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 300x300 [SAR 1:1 DAR 1:1], q=-1--1, 1 fps, 16384 tbn, 1 tbc
    Metadata:
      encoder         : Lavc56.60.100 libx264
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 16000 Hz, mono, fltp, 32 kb/s
    Metadata:
      encoder         : Lavc56.60.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[mp4 @ 0x4b38d40] Starting second pass: moving the moov atom to the beginning of the file
frame=   84 fps=0.0 q=-1.0 Lsize=     298kB time=00:01:22.00 bitrate=  29.8kbits/s
video:48kB audio:243kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.420870%
[libx264 @ 0x4b7eba0] frame I:1     Avg QP:13.91  size: 46896
[libx264 @ 0x4b7eba0] frame P:21    Avg QP:11.16  size:    31
[libx264 @ 0x4b7eba0] frame B:62    Avg QP:16.35  size:    15
[libx264 @ 0x4b7eba0] consecutive B-frames:  1.2%  0.0%  3.6% 95.2%
[libx264 @ 0x4b7eba0] mb I  I16..4:  0.6%  5.5% 93.9%
[libx264 @ 0x4b7eba0] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.7%  0.0%  0.0%  0.0%  0.0%    skip:99.3%
[libx264 @ 0x4b7eba0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.0%  0.0%  0.0%  direct: 0.0%  skip:100.0%
[libx264 @ 0x4b7eba0] 8x8 transform intra:5.5% inter:6.7%
[libx264 @ 0x4b7eba0] coded y,uvDC,uvAC intra: 97.9% 97.5% 94.7% inter: 0.0% 0.1% 0.0%
[libx264 @ 0x4b7eba0] i16 v,h,dc,p:  0%  0% 50% 50%
[libx264 @ 0x4b7eba0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu:  8% 51% 14%  4%  2%  1%  9%  5%  6%
[libx264 @ 0x4b7eba0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 14% 22% 16%  6%  8%  5%  8%  7% 13%
[libx264 @ 0x4b7eba0] i8c dc,h,v,p: 41% 27% 18% 14%
[libx264 @ 0x4b7eba0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x4b7eba0] kb/s:4.62
$ ffprobe -i /tmp/out.mp4 2>&1 | grep Duration
  Duration: 00:01:24.00, start: 0.064000, bitrate: 29 kb/s

I talked in #ffmpeg @ freenode with another user who has the same problem with MP4s:

<user> llogan: http://sprunge.us/JYOS
<user> when i run that with a 4m45s audio stream, the first one has a duration of 4:50, the second one 4:56
<user> s/4:56/4:46/
<user> it's also not specific to aac or mp4
<user> and also according to mediainfo it's the video stream which is too long, not the audio stream
<user> and finally, increasing -r reduces the offset
<user> i assume it's an issue with the image2 muxer, but i don't know for sure
<user> and by for sure i of course mean at all

          #5258: ffmpeg fails to decode opus test vectors correctly   

taken from https://trac.ffmpeg.org/ticket/5238#comment:3

ffmpeg fails to decode opus test vectors correctly, in especially: http://people.xiph.org/~greg/opus_testvectors/correctness_trimming_nobeeps.opus

even using ffmpeg git:

$ ./ffmpeg -i correctness_trimming_nobeeps.opus correctness_trimming_nobeeps.wav
ffmpeg version N-78622-g2b5b001 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.1 (Debian 5.3.1-9) 20160220
  configuration: --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv --enable-version3 --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libvo_amrwbenc
  libavutil      55. 18.100 / 55. 18.100
  libavcodec     57. 24.105 / 57. 24.105
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 34.100 /  6. 34.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, ogg, from 'correctness_trimming_nobeeps.opus':
  Duration: 00:00:12.72, start: 0.000000, bitrate: 40 kb/s
    Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
    Metadata:
      ENCODER         : Xiph.Org Opus testvectormaker
      TESTDESCRIPTION : Beeps at the beginning and end of this stream MUST be hidden by preskip and end-trimming.
                      : If the beeps are heard the decoder is broken.
Output #0, wav, to 'correctness_trimming_nobeeps.wav':
  Metadata:
    ISFT            : Lavf57.26.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      TESTDESCRIPTION : Beeps at the beginning and end of this stream MUST be hidden by preskip and end-trimming.
                      : If the beeps are heard the decoder is broken.
      encoder         : Lavc57.24.105 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1193kB time=00:00:11.35 bitrate= 860.4kbits/s speed= 211x
video:0kB audio:1193kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.006388%

the resulting WAV files still contains the beep


          #5468: on2avc (0x0501): improve decoding of stereo modes   

https://samples.mplayerhq.hu/V-codecs/VP6/wagner-700.vp6

https://samples.mplayerhq.hu/V-codecs/VP5/BeautifulMind_640vp5-750.vp5

https://samples.mplayerhq.hu/V-codecs/VP6/Xmen-avc-500.vp6

three more samples are here:

https://www.datafilehost.com/d/05f10ede

C:\>ffmpeg -i wagner-700.vp6 out.wav
ffmpeg version N-79546-g13406b6 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.100 / 55. 22.100
  libavcodec     57. 35.100 / 57. 35.100
  libavformat    57. 34.102 / 57. 34.102
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[on2avc @ 0x2c47d80] Stereo mode support is not good, patch is welcome
Input #0, avi, from 'wagner-700.vp6':
  Duration: 00:02:49.23, start: 0.000000, bitrate: 816 kb/s
    Stream #0:0: Video: vp6 (VP62 / 0x32365056), yuv420p, 624x352, 700 kb/s, 29.
97 fps, 29.97 tbr, 29.97 tbn
    Stream #0:1: Audio: avc ([1][5][0][0] / 0x0501), 44100 Hz, stereo, fltp, 96
kb/s
[on2avc @ 0x2899020] Stereo mode support is not good, patch is welcome
[wav @ 0x289c020] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
    Metadata:
      encoder         : Lavc57.35.100 pcm_s16le
Stream mapping:
  Stream #0:1 -> #0:0 (avc (on2avc) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=   29152kB time=00:02:49.22 bitrate=1411.2kbits/s speed= 285x
video:0kB audio:29152kB subtitle:0kB other streams:0kB global headers:0kB muxing
 overhead: 0.000261%

          #5482: g721 (g726): fix audiable artefacts   

decodes ok with sox

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g721_1ch_44khz.au

also reproducible with g726 sample:

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g726_44khz_2bit.au

C:\>ffmpeg -i C:\g721_1ch_44khz.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Input #0, au, from 'C:\g721_1ch_44khz.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 176 kb/s
    Stream #0:0: Audio: adpcm_g726le ([23][0][0][0] / 0x0017), 44100 Hz, mono, s
16, 176 kb/s
[wav @ 0x2c3eb60] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.103
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.38.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_g726le (g726le) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    2585kB time=00:00:30.01 bitrate= 705.6kbits/s speed= 213x
video:0kB audio:2585kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.002946%

          #5942: mpeg-4 als: "quant_cof 237 is out of range" with multichannel 16bit   

no such problem with mono/stereo files

encoded with:

C:\>mp4alsRM23.exe -7 -MP4 D:\16multi.wav
C:\>ffmpeg -i D:\16multi.mp4  D:\out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : 5.1
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\16multi.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-11-11 17:28:53
  Duration: 00:00:05.00, start: 0.000000, bitrate: 1220 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 48000 Hz, 6 channels, s
16, 1219 kb/s (default)
    Metadata:
      creation_time   : 2016-11-11 17:28:53
Output #0, wav, to 'D:\out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s
16, 4608 kb/s (default)
    Metadata:
      creation_time   : 2016-11-11 17:28:53
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[als @ 038f0d60] quant_cof 237 is out of range.
[als @ 038f0d60] Reading frame data failed. Skipping RA unit.
size=     720kB time=00:00:01.28 bitrate=4608.6kbits/s speed=N/A
video:0kB audio:720kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.013835%

          #5943: mpeg-4 als: "Overread 8" with 8bit file   

encoded with:

C:\>mp4alsRM23.exe -MP4 -7 -i d:\8mono_7_i.wav
C:\>ffmpeg -i D:\8mono_7_i.mp4 D:\out.wav
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\8mono_7_i.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-11-12 14:06:09
  Duration: 00:00:05.00, start: 0.000000, bitrate: 45 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 48000 Hz, 1 channels, s
16 (8 bit), 44 kb/s (default)
    Metadata:
      creation_time   : 2016-11-12 14:06:09
Output #0, wav, to 'D:\out.wav':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    ISFT            : Lavf57.26.100
    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono,
s16 (8 bit), 768 kb/s (default)
    Metadata:
      creation_time   : 2016-11-12 14:06:09
      encoder         : Lavc57.27.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (mp4als (als) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[als @ 03906020] Overread 8
[als @ 03906020] Reading frame data failed. Skipping RA unit.
size=     509kB time=00:00:05.42 bitrate= 768.1kbits/s speed= 174x
video:0kB audio:509kB subtitle:0kB other streams:0kB global headers:0kB muxing o
verhead: 0.014972%

          #6035: wmavoice: files decodes incorrectly   

4kbps 8khz and 5kbps 8khz modes decodes ok, the rest don't

https://trac.ffmpeg.org/raw-attachment/ticket/5426/wmavoice.7z

C:\>ffmpeg.exe -i C:\2_16kbps_16khz.wma out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[wmavoice @ 0224d5c0] Warning: not compiled with thread support, using thread em
ulation
Input #0, asf, from 'C:\2_16kbps_16khz.wma':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: S1
    WMFSDKVersion   : 11.0.5721.5275
    IsVBR           : 0
  Duration: 00:00:37.98, start: 0.000000, bitrate: 17 kb/s
    Stream #0:0(pol): Audio: wmavoice ([10][0][0][0] / 0x000A), 16000 Hz, mono,
flt, 16 kb/s
[wmavoice @ 025ffa20] Warning: not compiled with thread support, using thread em
ulation
[pcm_s16le @ 02247e00] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: S1
    WMFSDKVersion   : 11.0.5721.5275
    IsVBR           : 0
    ISFT            : Lavf57.61.100
    Stream #0:0(pol): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono,
s16, 256 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (wmavoice (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Error while decoding stream #0:0: Function not implementeded=13.5x
    Last message repeated 3 times
size=    1188kB time=00:00:37.97 bitrate= 256.2kbits/s speed=13.4x
video:0kB audio:1187kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.006415%

          #6036: sipr: improve decoding of 16khz 16kbps mode   

this bug deserves it's own ticket

there are metalic artifacts between 21s-32s

C:\>ffmpeg -i sipr_16k_could_be_better.rm out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[rm @ 02243d60] Invalid stream index 1 for index at pos 81173
[sipr @ 0224d380] Warning: not compiled with thread support, using thread emulat
ion
Input #0, rm, from 'sipr_16k_could_be_better.
rm':
  Metadata:
    Generated By    : RealProducer 6.1.0.153 Windows
    Content Rating  : (PICS-1.1 "http://www.classify.org/safesurf" labels commen
t "RealProducer 6.1.0.153 Windows" ratings (SS~~000 1))
    Target Audiences: 28K Modem (20 Kbps);
    Audio Format    : Voice with Background Music
    Video Quality   : No Video
    Creation Date   : 12/19/2016 3:26:44
    Modification Date: 12/19/2016 3:26:44
    File ID         : 5001d8cc-b0c8-5861-36e0-07a19838af8b
    Keywords        :
    Abstract        :
    title           : Inp
    author          :
    copyright       : ę2016
    comment         :
  Duration: 00:00:38.31, start: 0.000000, bitrate: 16 kb/s
    Stream #0:0: Audio: sipr (sipr / 0x72706973), 16000 Hz, mono, flt, 16 kb/s
[sipr @ 025d9c00] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 02247140] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    Generated By    : RealProducer 6.1.0.153 Windows
    Content Rating  : (PICS-1.1 "http://www.classify.org/safesurf" labels commen
t "RealProducer 6.1.0.153 Windows" ratings (SS~~000 1))
    Target Audiences: 28K Modem (20 Kbps);
    Audio Format    : Voice with Background Music
    Video Quality   : No Video
    Creation Date   : 12/19/2016 3:26:44
    Modification Date: 12/19/2016 3:26:44
    File ID         : 5001d8cc-b0c8-5861-36e0-07a19838af8b
    Keywords        :
    Abstract        :
    INAM            : Inp
    author          :
    ICOP            : ę2016
    ICMT            :
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16,
256 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (sipr (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=    1200kB time=00:00:38.31 bitrate= 256.6kbits/s speed= 189x
video:0kB audio:1200kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.008464%

          #6046: mpc8: "maxband 19 too large" / "overread 8"   
C:\>ffmpeg -i q1.mpc out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[mpc8 @ 0224d2e0] Warning: not compiled with thread support, using thread emulat
ion
Input #0, mpc8, from 'q1.mpc':
  Duration: 00:00:58.51, start: 0.000000, bitrate: 17 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, mono, s16p
[mpc8 @ 025db260] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 025dc9c0] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (musepack8 (mpc8) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[mpc8 @ 025db260] Multiple frames in a packet.
[mpc8 @ 025db260] maxband 19 too large
Error while decoding stream #0:0: Invalid data found when processing input
q1.mpc: I/O error
size=    5171kB time=00:01:00.02 bitrate= 705.6kbits/s speed= 384x
video:0kB audio:5170kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%
C:\>ffmpeg -i q10.mpc out.wav
ffmpeg version 3.2.git Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.6.2 (GCC)
  configuration: --disable-pthreads --enable-gpl
  libavutil      55. 43.100 / 55. 43.100
  libavcodec     57. 68.100 / 57. 68.100
  libavformat    57. 61.100 / 57. 61.100
  libavdevice    57.  2.100 / 57.  2.100
  libavfilter     6. 68.100 /  6. 68.100
  libswscale      4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc    54.  2.100 / 54.  2.100
[mpc8 @ 0224d300] Warning: not compiled with thread support, using thread emulat
ion
Input #0, mpc8, from 'q10.mpc':
  Duration: 00:00:58.51, start: 0.000000, bitrate: 123 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, mono, s16p
[mpc8 @ 0224c040] Warning: not compiled with thread support, using thread emulat
ion
[pcm_s16le @ 025fd600] Warning: not compiled with thread support, using thread e
mulation
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.61.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16,
705 kb/s
    Metadata:
      encoder         : Lavc57.68.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (musepack8 (mpc8) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[mpc8 @ 0224c040] Multiple frames in a packet.
[mpc8 @ 0224c040] Overread 8
q10.mpc: I/O error
size=    5173kB time=00:01:00.05 bitrate= 705.6kbits/s speed= 384x
video:0kB audio:5173kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001473%

          #667: Error when decoding TrueHD 1.0 track that also contains AC-3   

I am trying to use ffmpeg to decode a TrueHD 1.0 track to WAV for encoding to FLAC. Originally I was using eac3to to accomplish this, and received an error. I was told this had to do with the outdated ffmpeg used by eac3to, and that I should run the following command line with the latest ffmpeg build:

ffmpeg -i japanese.thd -ac 1 -acodec pcm_s24le -f wav japanese.wav

This command produces the following error message:

[truehd @ 0000000001C80290] mlpparse: Parity check failed.

Last message repeated 228 times

[truehd @ 000000000029D570] max_analyze_duration 5000000 reached at 5000000 [truehd @ 000000000029D570] Estimating duration from bitrate, this may be inaccurate Input #0, truehd, from 'japanese.thd':

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0:0: Audio: truehd, 48000 Hz, mono, s32

Output #0, wav, to 'japanese.wav':

Metadata:

encoder : Lavf53.20.0 Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s32, 1152 kb/s

Stream mapping:

Stream #0:0 -> #0:0 (truehd -> pcm_s24le)

Press [q] to stop, ? for help Input stream #0:0 frame changed from rate:48000 fmt:s32 ch:1 to rate:48000 fmt:s32 ch:2 [SWR @ 0000000001CA18D0] Input channel layout has a different number of channels than the number of used channels, ignoring layout [truehd @ 0000000001C80290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C80290] Lossless check failed - expected 05, calculated 00. [truehd @ 0000000001C80290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C80290] Lossless check failed - expected 09, calculated 06. [truehd @ 0000000001C80290] Lossless check failed - expected 0c, calculated 12. [truehd @ 0000000001C80290] Lossless check failed - expected 00, calculated 0a. [truehd @ 0000000001C80290] Invalid nonrestart_substr. Error while decoding stream #0:0

This error continues until the program halts and stops processing the file. According to the error above, this track is a mix of 1.0 and 2.0, so I tried running the same command with -ac 2 instead and received the same error message.

Here is the output produced by ffmpeg -v 9 -loglevel 99 -i japanese.thd -f wav japanese.wav:

ffmpeg version N-34906-g4e7b3ef, Copyright (c) 2000-2011 the FFmpeg developers

built on Nov 16 2011 12:39:54 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 25. 0 / 51. 25. 0 libavcodec 53. 34. 0 / 53. 34. 0 libavformat 53. 20. 0 / 53. 20. 0 libavdevice 53. 4. 0 / 53. 4. 0 libavfilter 2. 48. 1 / 2. 48. 1 libswscale 2. 1. 0 / 2. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0

[truehd @ 00000000003AD540] Format truehd probed with size=2048 and score=50 [truehd @ 0000000001C10290] err{or,}_recognition separate: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] Unsupported bit depth: 0 [truehd @ 0000000001C10290] mlpparse: Parity check failed.

Last message repeated 228 times

[truehd @ 00000000003AD540] max_analyze_duration 5000000 reached at 5000000 [truehd @ 00000000003AD540] Estimating duration from bitrate, this may be inaccurate Input #0, truehd, from 'japanese.thd':

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0:0, 6002, 1/90000: Audio: truehd, 48000 Hz, mono, s32

Incompatible sample format 's32' for codec 'pcm_s16le', auto-selecting format 's16' [pcm_s16le @ 0000000003B05820] err{or,}_recognition separate: 1; 1 [pcm_s16le @ 0000000003B05820] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition separate: 1; 1 [truehd @ 0000000001C10290] err{or,}_recognition combined: 1; 1 [truehd @ 0000000001C10290] Unsupported bit depth: 24 Output #0, wav, to 'japanese.wav':

Metadata:

encoder : Lavf53.20.0 Stream #0:0, 0, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s

Stream mapping:

Stream #0:0 -> #0:0 (truehd -> pcm_s16le)

Press [q] to stop, ? for help [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. Input stream #0:0 frame changed from rate:48000 fmt:s32 ch:1 to rate:48000 fmt:s32 ch:2 [SWR @ 0000000001C318D0] Input channel layout has a different number of channels than the number of used channels, ignoring layout 0.500000 0.500000 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 2 times

[truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 00. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 1 times

[truehd @ 0000000001C10290] Lossless check failed - expected 09, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 12. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 00, calculated 0a. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated 1d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 96, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 18, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 8e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 69, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2d, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected db, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated c9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated 1e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8d, calculated ee. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a9, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f3, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 21. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 93, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped.

Last message repeated 1 times

[truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated b2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 65, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated d8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 00. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4b, calculated 3c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated 99. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5a, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected af, calculated 6a. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d2, calculated c5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 9a, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated e2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 39, calculated 06. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 21, calculated ee. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 81, calculated 6a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 12. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated d2. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 36, calculated 69. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 59, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 48, calculated dd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 24, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated ed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f3, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 17, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 9a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 87. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 56. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a3, calculated 8d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected cc, calculated 59. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 50, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 95, calculated d7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 78, calculated 71. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated 72. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated c6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated aa. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated a6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated d1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 9c, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 24. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 28, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 41. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated b7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b7, calculated 88. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 69, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5a, calculated 44. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated e8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 93, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected af, calculated d2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 11, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 03, calculated ed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 03, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e1, calculated bb. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 41, calculated 50. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 22, calculated 0a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 8b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated 41. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 18, calculated e2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated ff. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 0f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected aa, calculated 0a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated 8d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 41, calculated 90. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected bb, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated 09. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c0, calculated 7d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 69. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e7, calculated 7e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated 6f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated f0. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ff, calculated e4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated 4e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 14. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e8, calculated 50. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b2, calculated 5c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 55, calculated b4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated 8b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 81. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated f3. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated cf. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated 39. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 33. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected db, calculated 47. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c5, calculated a9. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 88, calculated 2d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated 9f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 59, calculated 6c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated 27. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2b, calculated 66. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 56. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 48, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected be, calculated e7. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 28. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated af. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2d, calculated 59. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 84, calculated ac. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 56, calculated d4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b1, calculated 4b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 33, calculated 6c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fa, calculated 2b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated 59. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated 5c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated 66. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated e1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e2, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 96, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1e, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1d, calculated 7d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 22, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated 71. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4b, calculated 95. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated e1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e1, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b4, calculated aa. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b8, calculated 6a. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 17, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 2b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected be, calculated 4d. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1b, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 6f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 30, calculated eb. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a3, calculated a5. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5f, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated 05. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a5, calculated b7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a9, calculated 87. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated a0. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a6, calculated de. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated cf. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated 9c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected a5, calculated e7. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0f, calculated c3. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ed, calculated a6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 42. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 82, calculated 72. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 8b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2e, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated fc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 1b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 39, calculated 47. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7b, calculated 30. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Stream parameters not seen; skipping frame. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0c, calculated 93. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 66, calculated 1e. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated de. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated bd. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 17. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected fc, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3f, calculated 11. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b4, calculated 0c. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 05, calculated b2. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 56, calculated 30. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7e, calculated b7. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d1, calculated 7b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated db. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 81, calculated ff. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 5f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 47, calculated b4. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 4b. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated 9f. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected de, calculated 03. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 1b, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated b8. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated b1. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated f6. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c3, calculated cc. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 93. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 12, calculated 5f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6a, calculated f6. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 00, calculated 0f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated cc. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated 18. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated 81. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated be. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 4d, calculated 5c. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated bb. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated a0. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 30, calculated 66. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 5c, calculated a0. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c0, calculated e2. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f5, calculated 3f. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 35, calculated 0a. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 78. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated b7. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ac, calculated 56. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 0a, calculated ff. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 14, calculated d1. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3a, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated 56. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 53. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 2e, calculated aa. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 77, calculated 03. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ee, calculated 12. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c9, calculated 1b. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected f9, calculated 87. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected b2, calculated 55. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated 17. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 14, calculated 18. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 06, calculated aa. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 53, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 63, calculated 1d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 71, calculated d8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ff, calculated 66. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected dd, calculated d2. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 87, calculated 4b. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected d8, calculated 8e. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected eb, calculated 95. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 55. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 65, calculated f9. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 90, calculated ee. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 3c, calculated 33. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected c6, calculated e8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 7d, calculated 44. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 6c, calculated cc. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 09, calculated 24. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 66, calculated ed. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected eb, calculated 3a. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 99, calculated 4d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected ca, calculated e8. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 8e, calculated 30. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 84, calculated 1d. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected e4, calculated 48. [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Lossless check failed - expected 95, calculated 78. [truehd @ 0000000001C10290] Invalid nonrestart_substr. Error while decoding stream #0:0 [truehd @ 0000000001C10290] mlpparse: Parity check failed. [truehd @ 0000000001C10290] Extracting 2 channel downmix from substream 0. Further substreams will be skipped. [truehd @ 0000000001C10290] Loss

          #2459: mpc8: wrong duration   

ffmpeg reports 28s, mpcdec 30s

1323648 / 44100 gives the correct duration

http://ffmpeg.org/trac/ffmpeg/raw-attachment/ticket/1254/musepack8.mpc

$ mpcdec -i musepack8.mpc
mpcdec - Musepack (MPC) decoder v1.0.0 (C) 2006-2009 MDT
Built Feb 26 2009 20:20:48
file: musepack8.mpc
stream version 8
encoder: --Stable-- 1.30.0
profile: 'Standard' (q=5.00)
PNS: off
mid/side stereo: on
gapless: on
average bitrate:  176.5 kbps
samplerate: 44100 Hz
channels: 2
length: 0:30 (1323648 samples)
file size: 662373 Bytes
track peak: 0.00 dB
track gain: 0.00 dB / 0.00 dB
album peak: 0.00 dB
album gain: 0.00 dB / 0.00 dB
C:\>ffmpeg -i musepack8.mpc out.wav
ffmpeg version N-51683-g9dc88ac Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr  8 2013 21:19:21 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrn
b --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libr
tmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheo
ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena
ble-zlib
  libavutil      52. 25.100 / 52. 25.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  1.100 / 55.  1.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 49.101 /  3. 49.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[mpc8 @ 026e4020] max_analyze_duration 5000000 reached at 5015510 microseconds
Input #0, mpc8, from 'musepack8.mpc':
  Duration: 00:00:28.42, start: 0.000000, bitrate: 186 kb/s
    Stream #0:0: Audio: musepack8, 44100 Hz, stereo, s16p
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.1.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16
, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mpc8 -> pcm_s16le)
Press [q] to stop, [?] for help
Multiple frames in a packet from stream 0
musepack8.mpc: Input/output error
size=    5175kB time=00:00:30.04 bitrate=1411.2kbits/s
video:0kB audio:5175kB subtitle:0 global headers:0kB muxing overhead 0.001510%

          #5448: aac: wrong calculation of duration   

it used to be correct long time ago:

C:\>ffmpeg -i 1.aac
ffmpeg version 2.1.3 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jan 21 2014 18:55:26 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-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[aac @ 02a7a980] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '1.aac':
  Duration: 00:00:29.58, bitrate: 73 kb/s
    Stream #0:0: Audio: aac, 88200 Hz, mono, fltp, 73 kb/s
At least one output file must be specified
C:\>ffmpeg -i 1.aac out.wav
ffmpeg version N-79468-ga676c29 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.100 / 55. 22.100
  libavcodec     57. 34.102 / 57. 34.102
  libavformat    57. 34.101 / 57. 34.101
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 42.100 /  6. 42.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[aac @ 0x2894160] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '1.aac':
  Duration: 00:04:47.72, bitrate: 7 kb/s
    Stream #0:0: Audio: aac (LC), 88200 Hz, mono, fltp, 7 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[wav @ 0x2897420] Using AVStream.codec to pass codec parameters to muxers is dep
recated, use AVStream.codecpar instead.
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf57.34.101
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 88200 Hz, mono, s16,
1411 kb/s
    Metadata:
      encoder         : Lavc57.34.102 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 0; changing to 88057. This may result in incorrect timestamps in the output
 file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 1024; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 2048; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 3072; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 4096; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 5120; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 6144; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 7168; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 8192; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 9216; changing to 88057. This may result in incorrect timestamps in the out
put file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 10240; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 11264; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 12288; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 13312; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 14336; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 15360; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 16384; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 17408; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 18432; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 19456; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 20480; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 21504; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 22528; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 23552; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 24576; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 25600; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 26624; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 27648; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 28672; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 29696; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 30720; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 31744; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 32768; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 33792; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 34816; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 35840; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 36864; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 37888; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 38912; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 39936; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 40960; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 41984; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 43008; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 44032; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 45056; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 46080; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 47104; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 48128; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 49152; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 50176; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 51200; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 52224; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 53248; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 54272; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 55296; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 56320; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 57344; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 58368; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 59392; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 60416; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 61440; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 62464; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 63488; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 64512; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 65536; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 66560; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 67584; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 68608; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 69632; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 70656; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 71680; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 72704; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 73728; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 74752; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 75776; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 76800; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 77824; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 78848; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 79872; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 80896; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 81920; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 82944; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 83968; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 84992; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 86016; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
[wav @ 0x2897420] Non-monotonous DTS in output stream 0:0; previous: 88057, curr
ent: 87040; changing to 88057. This may result in incorrect timestamps in the ou
tput file.
size=    5188kB time=00:00:29.10 bitrate=1460.2kbits/s speed= 133x
video:0kB audio:5188kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.001920%

          #5501: EAC3 decoding   

Problem decoding EAC3 from TS, if track is repackaged with "-copy_unkown", then the stream is a FFmpeg decodable AC3 384k...

Any way to decode original file without the temp file?

  1. FAILED TO DECODE

ffmpeg -i part.ts -map 0:1 -t 10 -f null -

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

[mpegts @ 00000000007a2700] probed stream 1 failed [mpegts @ 00000000007a2700] Could not find codec parameters for stream 1 (Unknown: none (EAC3 / 0x33434145)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from 'part.ts':

Duration: N/A, start: 1.083400, bitrate: N/A Program 2

Stream #0:0[0x1e1]: Video: h264 (High 4:2:2) ([27][0][0][0] / 0x001B), yuv422p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9],

Closed Captions, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc

Stream #0:1[0x1e2]: Unknown: none (EAC3 / 0x33434145)

Cannot map stream #0:1 - unsupported type. If you want unsupported types ignored instead of failing, please use the -ignore_unknown option If you want them copied, please use -copy_unknown

  1. COPIED TRACK TO TS with "-copy_unknown"

ffmpeg -i 108GB_9.ts -map 0:1 -t 10 -copy_unknown -c copy audio_test.ts

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

[mpegts @ 0000000000662820] probed stream 1 failed [mpegts @ 0000000000662820] Could not find codec parameters for stream 1 (Unknown: none (EAC3 / 0x33434145)): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '108GB_9.ts':

Duration: N/A, start: 1.083400, bitrate: N/A Program 2

Stream #0:0[0x1e1]: Video: h264 (High 4:2:2) ([27][0][0][0] / 0x001B), yuv422p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9],

Closed Captions, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc

Stream #0:1[0x1e2]: Unknown: none (EAC3 / 0x33434145)

Output #0, mpegts, to 'audio_test.ts':

Metadata:

encoder : Lavf57.28.100 Stream #0:0: Unknown: none (EAC3 / 0x33434145)

Stream mapping:

Stream #0:1 -> #0:0 (copy)

Press [q] to stop, ? for help size= 540kB time=00:00:09.98 bitrate= 443.5kbits/s speed= 35x video:0kB audio:0kB subtitle:0kB other streams:473kB global headers:0kB muxing overhead: 14.242310%

  1. CONVERT TEST AUDIO FILE to WAV - OK!

ffmpeg -i audio_test.ts 1.wav

ffmpeg version N-79000-g66edd86 Copyright (c) 2000-2016 the FFmpeg developers

built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 28.100 / 57. 28.100 libavformat 57. 28.100 / 57. 28.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 39.102 / 6. 39.102 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100

Input #0, mpegts, from 'audio_test.ts':

Duration: 00:00:09.98, start: 1.400000, bitrate: 443 kb/s Program 1

Metadata:

service_name : Service01 service_provider: FFmpeg

Stream #0:0[0x100]: Audio: eac3 ([6][0][0][0] / 0x0006), 48000 Hz, 5.1(side), fltp, 384 kb/s

Output #0, wav, to '1.wav':

Metadata:

ISFT : Lavf57.28.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s16, 4608 kb/s Metadata:

encoder : Lavc57.28.100 pcm_s16le

Stream mapping:

Stream #0:0 -> #0:0 (eac3 (native) -> pcm_s16le (native))

Press [q] to stop, ? for help [eac3 @ 000000000078ef20] incomplete frame size= 5652kB time=00:00:10.04 bitrate=4608.1kbits/s speed= 141x video:0kB audio:5652kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.001762%


          #5828: Duration mismatch on opus to pcm conversion   

Summary of the bug:

There is a random non negligible duration mismatch when converting from opus to PCM

How to reproduce:

% ffmpeg -i test.opus out.wav
ffmpeg version N-81528-g39f7564-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.1 (Debian 5.4.1-1) 20160803
libopus version 1.1.3
% ffprobe test.opus
Input #0, ogg, from 'test.opus':
  Duration: 00:00:22.81, start: 1.346000, bitrate: 27 kb/s
    Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp, delay 48, padding 0
    Metadata:
      LANGUAGE        : eng
      ENCODER         : Lavf57.48.101
% ffprobe output.wav
Input #0, wav, from 'output.wav':
  Metadata:
    encoder         : Lavf57.48.101
  Duration: 00:00:21.48, bitrate: 768 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s16, 768 kb/s

          #5931: showwavespic, showspectrumpic innacurate length of resulting image   

Summary of the bug:showwavespic, showspectrumpic innacurate length of resulting image How to reproduce:

ffmpeg -i voice.wav -lavfi  showwavespic=scale=log:s=1000x180 waveform.png

In this example, the waveform is rendered correctly (end of waveform on image1.png). But if we increase the with to 10000, the end of waveform have an extra gap (end of waveform on image2.png).

ffmpeg -i voice.wav -lavfi  showwavespic=scale=log:s=10000x180 waveform.png

          #6500: Fails to playback a certain .WAV audio file   

Summary of the bug: ffplay fails to playback a certain .WAV audio file. VLC and other non-ffmpeg-based players can play it correctly; I'd assume the audio file is malformed in some way because it seems to be detected as a MP3 file. How to reproduce:

% ffplay -i FF_massfowl.wav
FF_massfowl.wav: Invalid data found when processing input0B f=0/0

          #357: Trying to transcode a vbk file to wav   

Hi Team, I was trying to transcode a vbk file to a wav file. But I encoutered an error log pasted below. Also I verified that the file is not corrupt by playing the same in CallPilot? Player and it plays fine.

C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest>ffmpeg -y -i C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\v060102.vbk -acodec pcm_mulaw C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\output.wav ffmpeg version N-31548-g78accb8, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 17 2011 22:41:27 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable- runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib

libavutil 51. 11. 0 / 51. 11. 0 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 0 / 2. 27. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0

C:\Users\himanshu.bhardwaj\Desktop\FFMpegTest\v060102.vbk: Invalid data found when processing input


          #1158: support iXML riff chunk   

This sample file contains metadata in a bext chunk and iXML chunk.

http://www.gallery.co.uk/ixml/examples/usesEntireiXMLSpec.WAV

With the flac utility there is an option called --keep-foreign-metadata that allows these chunks to be copied to the flac output. When I use ffmpeg to transcode the wav to a flac file there is no means to keep the bext chunk or ixml chunk of source file.

ffmpeg -i usesEntireiXMLSpec.WAV usesEntireiXMLSpec.flac
ffmpeg version N-35765-gf9143d2 Copyright (c) 2000-2012 the FFmpeg developers
  built on Mar 31 2012 12:44:33 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2335.15.00)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-libfreetype --cc=/usr/bin/llvm-gcc --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libass --disable-ffplay
  libavutil      51. 44.100 / 51. 44.100
  libavcodec     54. 12.100 / 54. 12.100
  libavformat    54.  3.100 / 54.  3.100
  libavdevice    53.  4.100 / 53.  4.100
  libavfilter     2. 66.101 /  2. 66.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 10.100 /  0. 10.100
  libpostproc    52.  0.100 / 52.  0.100
Input #0, wav, from 'usesEntireiXMLSpec.WAV':
  Metadata:
    comment         : gSCENE=66a
                    : gTAKE=002
                    : gTAPE=007
                    : gNOTE=Circle
                    : gUBITS=00000000
                    :
    encoded_by      : Gallery Metacorder
    date            : 2005:08:04
    creation_time   : 16:55:54
    time_reference  : 4090608000
    coding_history  :
  Duration: 00:00:02.18, bitrate: 781 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels, s16, 768 kb/s
Output #0, flac, to 'usesEntireiXMLSpec.flac':
  Metadata:
    comment         : gSCENE=66a
                    : gTAKE=002
                    : gTAPE=007
                    : gNOTE=Circle
                    : gUBITS=00000000
                    :
    encoded_by      : Gallery Metacorder
    date            : 2005:08:04
    creation_time   : 16:55:54
    time_reference  : 4090608000
    coding_history  :
    encoder         : Lavf54.3.100
    Stream #0:0: Audio: flac, 48000 Hz, 1 channels, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> flac)
Press [q] to stop, [?] for help
size=      42kB time=00:00:02.20 bitrate= 155.8kbits/s
video:0kB audio:34kB global headers:0kB muxing overhead 24.469009%

          #1325: Different results win7 vs. linux on pcm_dvd --> pcm_s24le   

Version : ffmpeg version N-40824-g31dfe20 Job : pcm_dvd --> pcm_s24le Args[0] : -i E:\VIDEO_TS\VTS_01_1.VOB test.wav Args[1] : -acodec pcm_s24le -i E:\VIDEO_TS\VTS_01_1.VOB test.wav Result[Win7] : pcm_s16le Result[Ubu] : pcm_s32le

Desc : Sourceformat (24Bit 96kHz) detected, but ffmpeg@windows switch to pcm_s16le for output. FFmpeg@ubuntu create the wav-output with 24Bit. Set the codes pcm_s24le explicit don't help:

Microsoft Windows [Version 6.1.7601]
Copyright (c) 2009 Microsoft Corporation. Alle Rechte vorbehalten.
Z:\Music\Ripp\DVD-A>ffmpeg -acodec pcm_s24le -i E:\VIDEO_TS\VTS_01_1.VOB test.wav
ffmpeg version N-40824-g31dfe20 Copyright (c) 2000-2012 the FFmpeg developers
  built on May 19 2012 00:49:25 with gcc 4.6.3
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enab
le-bzlib --enable-frei0r --enable-libass --enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable
-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-librtmp --enable-libschro
edinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-
libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      51. 53.100 / 51. 53.100
  libavcodec     54. 21.101 / 54. 21.101
  libavformat    54.  5.100 / 54.  5.100
  libavdevice    53.  4.100 / 53.  4.100
  libavfilter     2. 74.101 /  2. 74.101
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 12.100 /  0. 12.100
  libpostproc    52.  0.100 / 52.  0.100
[mpeg @ 0000000001e6eee0] max_analyze_duration 5000000 reached at 5001044
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, mpeg, from 'E:\VIDEO_TS\VTS_01_1.VOB':
  Duration: 00:10:59.06, start: 0.262200, bitrate: 13030 kb/s
    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 7500 kb/s, 29.97 tbr, 90k tbn, 59.
94 tbc
    Stream #0:1[0xa0]: Audio: '''pcm_s24le, 96000 Hz, stereo, s32''', 4608 kb/s
[abuffer @ 000000000a0fbf20] format:s32 layout:stereo rate:96000
[aformat @ 000000000ec50d00] auto-inserting filter 'auto-inserted resampler 0' between the filter 'src' and the filter '
aformat'
[aresample @ 000000000dd40da0] r:96000Hz -> r:96000Hz
Output #0, wav, to 'test.wav':
  Metadata:
    encoder         : Lavf54.5.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, stereo, s16, 3072 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 ('''pcm_s24le -> pcm_s16le''')
Press [q] to stop, [?] for help
size=   85217kB time=00:03:47.24 bitrate=3072.0kbits/s
video:0kB audio:85217kB global headers:0kB muxing overhead 0.000078%
Received signal 2: terminating.

          #1864: support 0x350 TwoCC (Micronas speech ADPCM codec)   

it's "Micronas InterMetall? Speech MI-SC4 ADPCM", decodes with noise for example with "-acodec adpcm_ct"

http://wiki.multimedia.cx/index.php?title=Micronas_SC4

C:\>ffmpeg -i 32khz_stereo.wav
ffmpeg version N-45867-g3318d6b Copyright (c) 2000-2012 the FFmpeg developers
  built on Oct 23 2012 02:39:33 with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.
0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-w32threads --enable-memalign-hack --enable-runtime-cpudetect
--enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -
lwinmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snap
shots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/s
napshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3
--enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-
libvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libmp3lame --enable-libvpx --disable-decoder=libvpx
  libavutil      51. 77.100 / 51. 77.100
  libavcodec     54. 68.100 / 54. 68.100
  libavformat    54. 33.100 / 54. 33.100
  libavdevice    54.  3.100 / 54.  3.100
  libavfilter     3. 20.104 /  3. 20.104
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 16.100 /  0. 16.100
  libpostproc    52.  1.100 / 52.  1.100
[wav @ 0x1edbe00] Could not find codec parameters for stream 0 (Audio: none (P[3
][0][0] / 0x0350), 32000 Hz, 2 channels, 264 kb/s): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[wav @ 0x1edbe00] Estimating duration from bitrate, this may be inaccurate
32khz_stereo.wav: could not find codec parameters

          #2837: Unsupported tak codec   

(bug 546) A user uploaded a tak sample (cut with dd) that cannot be decoded with current git head.

$ ffmpeg -i 01.tak out.wav
ffmpeg version N-55249-ga7bb12a Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  6 2013 08:47:17 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --disable-indev=jack
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 20.100 / 55. 20.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, tak, from '01.tak':
  Duration: 00:51:26.69, start: 0.000000, bitrate: 2 kb/s
    Stream #0:0: Audio: tak, 44100 Hz, stereo, s16p
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.13.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (tak -> pcm_s16le)
Press [q] to stop, [?] for help
[tak @ 0x2357cc0] unsupported codec: 0
[tak @ 0x2358140] unsupported codec: 0
[tak @ 0x23694e0] unsupported codec: 0
[tak @ 0x2369d40] unsupported codec: 0
[tak @ 0x2341540] unsupported codec: 0
[tak @ 0x2343200] unsupported codec: 0
[tak @ 0x2344ee0] unsupported codec: 0
[tak @ 0x2346bc0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x23488a0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
...
[tak @ 0x2346bc0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x23488a0] unsupported codec: 0
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[tak @ 0x2357cc0] unsupported codec: 0
size=       0kB time=00:00:00.00 bitrate=N/A
video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)

          #3718: ffmpeg does not correctly read input text file.   

Summary of the bug: How to reproduce:

> ffmpeg -f concat -i t
mp.txt -c copy output.wav
ffmpeg version N-60592-gfd982f2 Copyright (c) 2000-2014 the FFmpeg developers
  built on Feb 13 2014 22:01:02 with gcc 4.8.2 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopenco
re-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li
bvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 63.101 / 52. 63.101
  libavcodec     55. 52.101 / 55. 52.101
  libavformat    55. 32.101 / 55. 32.101
  libavdevice    55.  9.100 / 55.  9.100
  libavfilter     4.  1.102 /  4.  1.102
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[concat @ 003b36e0] Line 1: unknown keyword 'file'
tmp.txt: Invalid data found when processing input

This is the Windows Zeranoe Build downloaded from here: http://ffmpeg.zeranoe.com/builds/win32/static/ffmpeg-20140612-git-3a1c895-win32-static.7z

The file is written from a self-made Visual Basic program using the method described here: http://msdn.microsoft.com/en-us/library/ms128035(v=vs.110).aspx. As you can see under the Remarks section, it uses the UTF-8 encoding.

Turns out, that method also writes 3 extra chars to the file, ef bb bf. It seems like this throws off FFMPEG and it gives the error above.


          #4140: wav: Demux cover art   

Summary of the bug:

ffplay does not show cover picture in it's gui window. Attached file contains picture described by id3v2.2 tags. File is wav. From libavformat/utils.c I found that id3 parsing happens only with mp3, aac, tta formats. If add wav condition parsing done well. Is it correct to add wav id3 parsing functionality?

How to reproduce: % ffplay test.wav

ffplay version N-68018-g8b8cb30 Copyright (c) 2003-2014 the FFmpeg developers

built on Nov 26 2014 16:52:25 with gcc 4.6 (Ubuntu/Linaro? 4.6.3-1ubuntu5) configuration: 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

Input #0, wav, from 'test.wav': 0KB vq= 0KB sq= 0B f=0/0

Metadata:

title : The Wailing Wall artist : Inner Sanctum album : CBS, Tue 11-06-45 date : 1945

Duration: 00:54:50.70, bitrate: 12 kb/s

Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 22050 Hz, mono, s16p, 24 kb/s

3.29 M-A: 0.000 fd= 0 aq= 1KB vq= 0KB sq= 0B f=0/0

ffmpeg version: master (8b8cb30d11885b406ce0dd896b42992386d98594)

built on: Ubuntu 12.04 LTS 64bit


          #4799: audio codec conversion codec 0x0135 sipro lab kelvin   

I'm trying to convert codec 0x0135 sipro lab kelvin to mp3. I think ffmpeg currently doesn't support this conversion.

commandline - ffmpeg -i sipro_codec_wav.wav test.mp3 http://www.datafilehost.com/d/cef9a4d9 http://www.datafilehost.com/d/d87133da http://www.datafilehost.com/d/05a4dc9e - sample wav


          #5313: mpeg-4 als: add support for adaptive RLS-LMS prediction   
C:\>mp4alsRM23.exe -[z1/z2/z3] -MP4 [z1/z2/z3].wav

http://www.datafilehost.com/d/220f9dee

C:\>ffmpeg -i z1.mp4
ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (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-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
ble-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --
enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib
x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d
ecklink --enable-zlib
  libavutil      55. 19.100 / 55. 19.100
  libavcodec     57. 27.100 / 57. 27.100
  libavformat    57. 26.100 / 57. 26.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 37.100 /  6. 37.100
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[als @ 038ef940] Adaptive RLS-LMS prediction is not implemented. Update your FFm
peg version to the newest one from Git. If the problem still occurs, it means th
at your file has a feature which has not been implemented.
[mov,mp4,m4a,3gp,3g2,mj2 @ 038de520] Failed to open codec in av_find_stream_info
[als @ 038ef940] Adaptive RLS-LMS prediction is not implemented. Update your FFm
peg version to the newest one from Git. If the problem still occurs, it means th
at your file has a feature which has not been implemented.
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'z1.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42isom
    creation_time   : 2016-03-08 19:47:15
  Duration: 00:00:30.01, start: 0.000000, bitrate: 811 kb/s
    Stream #0:0(und): Audio: mp4als (mp4a / 0x6134706D), 44100 Hz, 2 channels, 8
10 kb/s (default)
    Metadata:
      creation_time   : 2016-03-08 19:47:15
At least one output file must be specified

          #5407: WavPack correction file (hybrid mode)   

As far as I can tell ffmpeg is currently not able to use WavePack? correction file when decoding a WavePack? track.

It would be great if this could be implemented.

Relevant excerpt from WavePack? user documentation:

If the -c option is specified (in addition to the -b option), then WavPack? will generate an additional file that will contain the information that was discarded to generate the lossy .wv file. This file will have the same name as the .wv file but will have the extension .wvc (the 'c' is for "correction"). When WvUnpack? is used to restore the .wav file, it will attempt to find the .wvc file in the same directory as the .wv file. If it is found then the decompression will be lossless and will be reported as such, otherwise lossy decompression will be reported (assuming that any information was actually discarded during the original compression).


          #5483: g721: support stereo modes   

decodes ok with winamp

decodes not so good with sox (probably buggy support)

https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket1955/g721_2ch_44khz.au

C:\>ffmpeg -i C:\g721_2ch_44khz.au out.wav
ffmpeg version N-79632-g3ce1988 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.5.0 (GCC) 20100414 (Fedora MinGW 4.5.0-1.fc14)
  configuration: --prefix=/var/www/users/research/ffmpeg/snapshots/build --arch=
x86 --target-os=mingw32 --cross-prefix=i686-pc-mingw32- --cc='ccache i686-pc-min
gw32-gcc' --enable-pthreads --enable-memalign-hack --enable-runtime-cpudetect --
enable-cross-compile --enable-static --disable-shared --extra-libs='-lws2_32 -lw
inmm -lpthread' --extra-cflags='--static -I/var/www/users/research/ffmpeg/snapsh
ots/build/include' --extra-ldflags='-static -L/var/www/users/research/ffmpeg/sna
pshots/build/lib' --enable-bzlib --enable-zlib --enable-gpl --enable-version3 --
enable-nonfree --enable-libx264 --enable-libspeex --enable-libtheora --enable-li
bvorbis --enable-libfaac --enable-libxvid --enable-libopencore-amrnb --enable-li
bopencore-amrwb --enable-libmp3lame --enable-libfreetype --enable-libvpx --disab
le-decoder=libvpx
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[g726le @ 0x28ad4a0] Decoding more than one channel is not implemented. Update y
our FFmpeg version to the newest one from Git. If the problem still occurs, it m
eans that your file has a feature which has not been implemented.
[g726le @ 0x28ad4a0] If you want to help, upload a sample of this file to ftp://
upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-d
evel@ffmpeg.org)
[au @ 0x28a4020] Failed to open codec in av_find_stream_info
[g726le @ 0x28ad4a0] Decoding more than one channel is not implemented. Update y
our FFmpeg version to the newest one from Git. If the problem still occurs, it m
eans that your file has a feature which has not been implemented.
[g726le @ 0x28ad4a0] If you want to help, upload a sample of this file to ftp://
upload.ffmpeg.org/incoming/ and contact the ffmpeg-devel mailing list. (ffmpeg-d
evel@ffmpeg.org)
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, au, from 'C:\g721_2ch_44khz.au':
  Duration: 00:00:30.01, start: 0.000000, bitrate: 352 kb/s
    Stream #0:0: Audio: adpcm_g726le ([23][0][0][0] / 0x0017), 44100 Hz, 2 chann
els, 352 kb/s
File 'out.wav' already exists. Overwrite ? [y/N] y
[abuffer @ 0x2c49700] Unable to parse option value "(null)" as sample format
    Last message repeated 1 times
[abuffer @ 0x2c49700] Error setting option sample_fmt to value (null).
[graph 0 input from stream 0:0 @ 0x2c49640] Error applying options to the filter
.
Error opening filters!

          #2786: last packet of TTA stream in Matroska isn't decoded correctly   

Using inside.tta from the FATE test suit as an example:

Testing MD5 of the decoded stream:

$ ./ffmpeg -i ../samples/lossless-audio/inside.tta -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.101 /  3. 81.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Input #0, tta, from '../samples/lossless-audio/inside.tta':
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=d0beb768d860b4776358077dd9fcb1e9
size=       0kB time=00:00:11.88 bitrate=   0.0kbits/s
video:0kB audio:2048kB subtitle:0 global headers:0kB muxing overhead -99.998236%

Remuxing into Matroska:

$ ./ffmpeg -i ../samples/lossless-audio/inside.tta -c:a copy inside.mka
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, tta, from '../samples/lossless-audio/inside.tta':
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16
Output #0, matroska, to 'inside.mka':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: tta, 44100 Hz, stereo
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    1182kB time=00:00:12.53 bitrate= 772.4kbits/s
video:0kB audio:1182kB subtitle:0 global headers:0kB muxing overhead 0.053638%

Testing MD5 of the decoded stream, this time from the Matroska file:

$ ./ffmpeg -i inside.mka -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, matroska,webm, from 'inside.mka':
  Metadata:
    ENCODER         : Lavf55.12.102
  Duration: 00:00:11.89, start: 0.000000, bitrate: 814 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16 (default)
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=eb96f9b80e0ce6bc5ddd841a8843620a
size=       0kB time=00:00:11.49 bitrate=   0.0kbits/s
video:0kB audio:1980kB subtitle:0 global headers:0kB muxing overhead -99.998175%

Remuxing inside.tta using mkvtoolnix, then testing MD5 of the decoded stream:

$ ./ffmpeg -i inside_mkvtoolnix.mka -f md5 -
ffmpeg version N-54738-g04b9836 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 14 2013 17:57:33 with gcc 4.7.3 (GCC)
  configuration: --enable-nonfree --enable-gpl --enable-libfdk-aac --enable-libm
p3lame --enable-libvorbis --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32
 --arch=x86_64 --cpu=amdfam10 --prefix=/mingw64
[...]
Input #0, matroska,webm, from 'inside_mkvtoolnix.mka':
  Metadata:
    creation_time   : 2013-07-15 07:41:51
  Duration: 00:00:11.89, start: 0.000000, bitrate: 818 kb/s
    Stream #0:0: Audio: tta, 44100 Hz, stereo, s16 (default)
Output #0, md5, to 'pipe:':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
Stream mapping:
  Stream #0:0 -> #0:0 (tta -> pcm_s16le)
Press [q] to stop, [?] for help
MD5=eb96f9b80e0ce6bc5ddd841a8843620a
size=       0kB time=00:00:11.49 bitrate=   0.0kbits/s
video:0kB audio:1980kB subtitle:0 global headers:0kB muxing overhead -99.998175%

To make sure I tried the bit-compare plugin for foobar2000, and unlike with avformat, both inside.tta and any of the Matroska files are the same:

All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside.mka"
No differences in decoded data found.
All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_mkvtoolnix.mka"
No differences in decoded data found.

After converting inside.tta into WAV pcm_s16le with "./ffmpeg -i ../samples/lossless-audio/inside.tta inside_tta.wav":

All tracks decoded fine, no differences found.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_tta.wav"
No differences in decoded data found.

After converting inside.mka into WAV pcm_s16le with "./ffmpeg -i inside.mka inside_mka.wav":

Differences found in 1 out of 1 track pairs.
Comparing:
"D:\MinGW\msys\1.0\ffmpeg\samples\lossless-audio\inside.tta"
"D:\MinGW\msys\1.0\ffmpeg\build\inside_mka.wav"
Length mismatch : 0:11.888367 vs 0:11.493878, 524277 vs 506880 samples

Looks like the Matroska demuxer is discarding data at the end of the stream or something, without giving an error or warning.


          #5238: gapless playback doesn't work (with at least Opus)   

Hi.

It seems that ffmpeg has quite some issues with respect to gapless playback at several formats. I'll show it here for Opus but it probably doesn't work for at least AAC either.

The following is basically from https://github.com/mpv-player/mpv/issues/2823, which AFAIU uses ffmpeg for opus playback:


The approach is basically: take and two tracks that play gapless (or take a single track and split it at a position with audio into two WAVs), encode them with opusenc, play it back with mpv... and one will hear a really very tiny gap (which sounds therefore more like a click) between the two.

Encoding settings I've used were: opusenc --bitrate 128 --vbr in.wav out.opus. Further, I decoded the opus files again with opusdec, opened the 6 files in audactiy (which AFAICS also uses ffmpeg to import Opus, just as mpv) and concatenated the corresponding files:

1st track: original WAV 2nd track: ffmpeg imported Opus 3rd track: WAV, gained by decoding the Opus files with opusdec

The beginning looks like: screenshot from 2016-02-14 23-26-40 The approach is basically: take and two tracks that play gapless (or take a single track and split it at a position with audio into two WAVs), encode them with opusenc, play it back with mpv... and one will hear a really very tiny gap (which sounds therefore more like a click) between the two.

Encoding settings I've used were: opusenc --bitrate 128 --vbr in.wav out.opus. Further, I decoded the opus files again with opusdec, opened the 6 files in audactiy (which AFAICS also uses ffmpeg to import Opus, just as mpv) and concatenated the corresponding files:

1st track: original WAV 2nd track: ffmpeg imported Opus 3rd track: WAV, gained by decoding the Opus files with opusdec

The beginning looks like beginning.png.

The join section looks like intersection.png

And the end like end.png

One can quite clearly see that 1st and 3rd track are pretty fine... but the directly read Opus is not only shifted, but also badly broken at the intersection.


I have basically the same problem with AAC, more detailed described here: https://github.com/mpv-player/mpv/issues/2284 and https://github.com/mstorsjo/fdk-aac/issues/35 https://github.com/nu774/fdkaac/issues/19

Interestingly, mpv does perfectly playback MP3s encoded gaplessly with lame (which confusingly does mean NOT to use the --gapless* options, which are broken for a long time).

mplayer OTOH, fails to badly with playing back anything gapless (including WAVs)

Cheers, Chris.


          #4916: AAC decoder generates noise if compiled on OSX 10.10.3   

We've a custom audio player built upon libav. We discovered that if we compile libav on osx 10.10.3 without changing a single line of code, then any AAC file decoding (see below) generates noise (you can't hear the audio at all, just noise). However, if we compile ffmpeg on osx 10.9.5 and we move the .dylib to 10.10.3, then AAC decoding works perfectly.

FFmpeg version: 2.8 (we also tested 2.3.6 and they issue is there too)

How we compile the libraries:

./configure \

--enable-shared \ --disable-static \ --enable-gpl \ --enable-version3 \ --enable-nonfree \ --disable-runtime-cpudetect \ --disable-all \ --disable-doc \ --enable-avcodec \ --enable-avformat \ --enable-avutil \ --enable-swresample \ --disable-w32threads \ --disable-os2threads \ --disable-network \ --disable-dxva2 \ --disable-vaapi \ --disable-vda \ --disable-vdpau \ --enable-protocol="file" \ --enable-decoder="aac" \ --enable-decoder="cook" \ --enable-decoder="flac" \ --enable-decoder="mp3" \ --enable-decoder="mp3adu" \ --enable-decoder="mp3adufloat" \ --enable-decoder="mp3float" \ --enable-decoder="mp3on4" \ --enable-decoder="mp3on4float" \ --enable-decoder="pcm_alaw" \ --enable-decoder="pcm_bluray" \ --enable-decoder="pcm_dvd" \ --enable-decoder="pcm_f32be" \ --enable-decoder="pcm_f32le" \ --enable-decoder="pcm_f64be" \ --enable-decoder="pcm_f64le" \ --enable-decoder="pcm_lxf" \ --enable-decoder="pcm_mulaw" \ --enable-decoder="pcm_s16be" \ --enable-decoder="pcm_s16be_planar" \ --enable-decoder="pcm_s16le" \ --enable-decoder="pcm_s16le_planar" \ --enable-decoder="pcm_s24be" \ --enable-decoder="pcm_s24daud" \ --enable-decoder="pcm_s24le" \ --enable-decoder="pcm_s24le_planar" \ --enable-decoder="pcm_s32be" \ --enable-decoder="pcm_s32le" \ --enable-decoder="pcm_s32le_planar" \ --enable-decoder="pcm_s8" \ --enable-decoder="pcm_s8_planar" \ --enable-decoder="pcm_u16be" \ --enable-decoder="pcm_u16le" \ --enable-decoder="pcm_u24be" \ --enable-decoder="pcm_u24le" \ --enable-decoder="pcm_u32be" \ --enable-decoder="pcm_u32le" \ --enable-decoder="pcm_u8" \ --enable-decoder="pcm_zork" \ --enable-decoder="ra_144" \ --enable-decoder="ra_288" \ --enable-decoder="ralf" \ --enable-decoder="vorbis" \ --enable-decoder="wmav1" \ --enable-decoder="wmav2" \ --enable-decoder="wmavoice" \ --enable-parser="aac" \ --enable-parser="flac" \ --enable-parser="mpegaudio" \ --enable-parser="vorbis" \ --enable-demuxer="aac" \ --enable-demuxer="asf" \ --enable-demuxer="flac" \ --enable-demuxer="ogg" \ --enable-demuxer="mp3" \ --enable-demuxer="mov" \ --enable-demuxer="pcm_alaw" \ --enable-demuxer="pcm_f32be" \ --enable-demuxer="pcm_f32le" \ --enable-demuxer="pcm_f64be" \ --enable-demuxer="pcm_f64le" \ --enable-demuxer="pcm_mulaw" \ --enable-demuxer="pcm_s16be" \ --enable-demuxer="pcm_s16le" \ --enable-demuxer="pcm_s24be" \ --enable-demuxer="pcm_s24le" \ --enable-demuxer="pcm_s32be" \ --enable-demuxer="pcm_s32le" \ --enable-demuxer="pcm_s8" \ --enable-demuxer="pcm_u16be" \ --enable-demuxer="pcm_u16le" \ --enable-demuxer="pcm_u24be" \ --enable-demuxer="pcm_u24le" \ --enable-demuxer="pcm_u32be" \ --enable-demuxer="pcm_u32le" \ --enable-demuxer="rm" \ --enable-demuxer="wav" \ --disable-debug \ --enable-stripping

make -j4 && make install

AAC test files: All the files listed here generate noise: http://download.wavetlan.com/SVV/Media/HTTP/http-aac.htm

Feel free to ask me any further information required to debug it. Thanks!


          #3803: Calculate actual bitrate of first frame when reading cbr dts   

Summary of the bug:

When adding a WAV header to a dts stream, FFmpeg writes the value read from the dts "target bitrate" field, which almost-always is not the actual bitrate of the stream. DTS in a WAV container is accepted by VirtualDub?'s AVI muxer, and so far I have to hex-edit the WAVved DTS files created through FFmpeg.

How to reproduce:

1) Create a dts file from an Audio CD track;

dcaenc -i CD_Track_01.wav -o track_01.dts -b 441

2) Mux with FFmpeg;

ffmpeg -i track_01.dts -acodec copy track_01.WAV

The RIFF header will report "448kbps", instead of *441kbps*, which is the correct value.


          Clé USB 3.0 Leef iBridge Lightning 16 Go Noire   
49,99€

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Profiter de vos films ou de toute votre bibliothèque de musiques sans devoir occuper un seul Mo d'espace sur votre appareil iOS

Designed for your on-the-go lifestyle

Le design en forme de J de la clé Leef iBridge s'adapte à vos appareils et étuis iOS

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REMARQUE : Apple ne permet pas que des fichiers DRM audio & video d'iTunes soient joués sur des applications externes

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Shhh. Don't tell anyone. Today is Doc's birthday.
          【转载】每天健康养生十分钟   

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客
 健康养生十分钟 - 逍遥客 - 逍遥客的网易博客
 健康养生十分钟 - 逍遥客 - 逍遥客的网易博客
 


1两手对搓一分钟——治疗肩痛



健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

2手指摩头一分钟——头发乌黑

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

3轻揉耳轮一分钟——防治耳鸣目眩

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

4转动眼睛一分钟——提神醒目

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

5拇指揉鼻一分钟——防治心脏病

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

6叩齿卷舌一分钟——牙齿保健

健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

7轻按肚脐一分钟——促进消化


健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

8收腹提肛一分钟——促进血液循环


健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

9伸屈四肢一分钟——防治心脑血管疾病


健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 

10蹬摩脚心一分钟——清肝明目


健康养生十分钟 - 逍遥客 - 逍遥客的网易博客

 


逍遥客的网易博客 - 逍遥客 - 逍遥客
 
 
 

http://zgzmzr.blog.163.com/


逍遥客祝福朋友幸福、快乐!


          AMBER Alert issued for three Anderson, Indiana children - WAVE 3   

WLKY Louisville

AMBER Alert issued for three Anderson, Indiana children
WAVE 3
Jayden Rindahl, an eight-month-old white male Anderson Police believe has been abducted. (Source: Anderson Police). (Source: Anderson Police). (Source: Anderson Police). ANDERSON, IN (WAVE) - Anderson Police have issued AMBER Alerts for three ...
Amber alert: 3 children abducted in Anderson, police sayThe Herald-Times (subscription)

all 15 news articles »

          LGBT Religion News Updates for August 24, 2011   
Sangamon County Circuit Judge John Schmidt ruled last week that the state of Illinois has no legal obligation to renew its annual foster care contract with Catholic Charities, due to Catholic Charities’ refusal to place children in families with gay and lesbian parents. After being investigated earlier this month simply for providing meeting space for PFLAG meetings, Journey Fellowship, a Baptist congregation in Owensboro, Kentucky, has adopted a position of “radical acceptance” of LGBT people and, as a result, has been removed from the Daviess-McLean Baptist Association. Progressive Christian magazine Sojourners, which had previously received criticism after rejecting an advertisement by the Believe Out Loud campaign, printed an ad in this month’s issue highlighting statistics on LGBT youth homelessness and the Ali Forney Center, an LGBT youth homeless shelter in New York City. Meeting in Orlando for their 2011 Churchwide Assembly, the Evangelical Lutheran Church in America (ELCA), the nation’s largest Lutheran denomination, passed a resolution last week to seek out and eliminate bullying in churches and schools. The Rev. Dr. Janet Edwards wrote in The Huffington Post this week about Mt. Auburn Presbyterian Church in Cincinnati, Ohio, which recently celebrated its 20th year as an inclusive and affirming congregation, noting that “while many faithful Christians today are beginning to move in the direction of full inclusion, Mt. Auburn has been showing us the way to the future of the church, since the early 1990s” and has served as a guide for the progression of many denominations and congregations towards affirmation and acceptance of all people of faith. Finally, clergy in Australia are speaking out in favor of marriage equality, while a poll found that the majority of Australian Christians support marriage for all couples.
August 24, 2011
Issues: 

          引用 常用降压药物的用法、用量及注意事项   


 

一、血管紧张素转化酶抑制剂


常用降压药物的用法、用量及注意事项 - 逍遥客 - 逍遥客的网易博客
 
注意事项

1、ACEI类降压药与利尿药合用:应从小剂量开始,以确定其对血压的起始效应,然后根据病人的需要对剂量加以调整。

2、ACEI最常见的不良反应是“干咳”,其特征为:无痰性、持续性咳嗽,停药后可消失。

3、ACEI最严重的不良反应是血管神经性水肿,可发生面部、四肢、唇、舌、声门和/或喉的血管神经性水肿者,可在治疗期的任何时间发生。

4、血管神经性水肿伴有喉部水肿可能导致死亡,当水肿发生在舌、声门或喉部时,可能引起气道阻塞,应立即给予适当治疗,包括诸如皮下注射1:1000肾上腺素溶液(0.3-0.5mL)和/或立即采取保持呼吸道通畅的措施。




二、血管紧张素受体拮抗剂 

常用降压药物的用法、用量及注意事项 - 逍遥客 - 逍遥客的网易博客

注意事项

1、ARB与ACEI均可引起高血钾症,也可引起血管神经性水肿。

2、坎地沙坦偶尔可引起横纹肌溶解和间质性肺炎。如出现上述情况,应停止服药,并进行适当处理。

 
 


 三、β受体阻滞剂

常用降压药物的用法、用量及注意事项 - 逍遥客 - 逍遥客的网易博客

注意事项

1、降压作用强度与血药浓度并不平行,而心率的减少则与血药浓度呈直线关系,剂量调整应以临床效应为准。

2、不能突然停药,否则可引起心绞痛。整个撤药过程约需要用2周时间,剂量逐渐减低。

 


 四、钙拮抗剂


常用降压药物的用法、用量及注意事项 - 逍遥客 - 逍遥客的网易博客



注意事项

1、地尔硫卓有明显的负性肌力作用,不能擅自增加或减少用药剂量。

2、地尔硫卓、尼群地平对可使地高辛的体内清除率降低,血药浓度增加;氨氯地平对地高辛的血药浓度无影响。

3、氨氯地平:可显著增加辛伐他汀的血药浓度,服用氨氯地平的患者应将辛伐他汀的剂量限制在20mg/日以下;但氨氯地平对阿托伐他汀的血药浓度无影响。



 五、噻嗪类利尿剂

常用降压药物的用法、用量及注意事项 - 逍遥客 - 逍遥客的网易博客


注意事项

1、交叉过敏:噻嗪类利尿药与磺胺类药物、呋塞米、布美他尼、碳酸酐酶抑制剂有交叉过敏。

2、当利尿药每日用药一次时,应在早晨用药,以免夜间排尿次数增多。

3、利尿药常见不良反应:低血钾、高血糖、高尿酸血症。