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• avcodec_find_decoder() can't find AV_CODEC_ID_WMAV2 even through CLI can parse WMAs on macOS ?

  3 octobre 2023, par grendell

  I am following the decode_audio.c example from FFmpeg, but I am unable to initialize a parser for AV_CODEC_ID_WMAV2.

  


  Test code :

  


  #include &#xA;#include <libavcodec></libavcodec>avcodec.h>&#xA;&#xA;int main() {&#xA;    // codec is found successfully&#xA;    const AVCodec * codec = avcodec_find_decoder(AV_CODEC_ID_WMAV2);&#xA;    if (!codec) {&#xA;        fprintf(stderr, "codec not found\n");&#xA;        return 1;&#xA;    }&#xA;&#xA;    // parser is always NULL&#xA;    AVCodecParserContext * parser = av_parser_init(codec->id);&#xA;    if (!parser) {&#xA;        fprintf(stderr, "parser not found\n");&#xA;        return 1;&#xA;    }&#xA;&#xA;    av_parser_close(parser);&#xA;    return 0;&#xA;}&#xA;

  


  Build commands :

  


  clang -c -I/opt/homebrew/Cellar/ffmpeg/6.0_1/include wma2mp3.c -o obj/wma2mp3.o
clang -L/opt/homebrew/Cellar/ffmpeg/6.0_1/lib -lavcodec obj/wma2mp3.o -o wma2mp3


  


  I'm surprised by the fact that the FFmpeg CLI can perform this operation on the same machine :

  


  % ffmpeg -i test.wma test.mp3
ffmpeg version 6.0 Copyright (c) 2000-2023 the FFmpeg developers
  built with Apple clang version 14.0.3 (clang-1403.0.22.14.1)
  configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/6.0_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon
  libavutil      58.  2.100 / 58.  2.100
  libavcodec     60.  3.100 / 60.  3.100
  libavformat    60.  3.100 / 60.  3.100
  libavdevice    60.  1.100 / 60.  1.100
  libavfilter     9.  3.100 /  9.  3.100
  libswscale      7.  1.100 /  7.  1.100
  libswresample   4. 10.100 /  4. 10.100
  libpostproc    57.  1.100 / 57.  1.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, asf, from 'test.wma':
  Metadata:
    ToolName        : Windows Media Encoding Utility
    ToolVersion     : 8.00.00.0343
  Duration: 00:00:00.74, start: 0.000000, bitrate: 80 kb/s
  Stream #0:0: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 1 channels, fltp, 48 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (wmav2 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'test.mp3':
  Metadata:
    ToolName        : Windows Media Encoding Utility
    ToolVersion     : 8.00.00.0343
    TSSE            : Lavf60.3.100
  Stream #0:0: Audio: mp3, 44100 Hz, mono, fltp
    Metadata:
      encoder         : Lavc60.3.100 libmp3lame
[libmp3lame @ 0x130706320] Queue input is backward in timeed=N/A    
[mp3 @ 0x1307056e0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 15668 >= 14764
size=       8kB time=00:00:00.97 bitrate=  65.8kbits/s speed= 103x    
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 4.048112%


  


  I am using an Apple M1 machine running MacOS 13.5.2 (22G91).

  


  Is the CLI using a different mechanism than av_parser_parse2 to perform this conversion, and is there a better way to accomplish this via the C API ?

  


• FFmpeg - Overlay multiple layers of transparent webm files [closed]

  27 octobre 2023, par Jens

  I am trying to overlay multiple layers ( up to 8 layers ) of transparent webm files to an mp4 file.

  


  If I do this for 2 layers, it works but ignores the alpha channel.

  


  ffmpeg  -i 1.webm -i 2.webm  -c:a copy -filter_complex "[0:v][1:v] overlay=0:0:enable='between(t,0,20)'" output.mp4


  


  I specify libvpx-vp9 to get the alpha channel as well

  


  ffmpeg -c:v libvpx-vp9 -i 1.webm -c:v libvpx-vp9 -i 2.webm -c:a copy -filter_complex "[0:v][1:v] overlay=0:0:enable='between(t,0,20)'" output.mp4


  


  Then I get an error.

  


  [libvpx-vp9 @ 0x7f9bd8d04840] Failed to decode frame: Unspecified internal error
    Last message repeated 3 times
[libvpx-vp9 @ 0x7f9bd8d04840] Failed to decode frame: Bitstream not supported by this decoder
    Last message repeated 8 times
[matroska,webm @ 0x7f9bd8f04880] Could not find codec parameters for stream 0 (Video: vp9 (libvpx-vp9) (Profile 3), none, 1080x1080): unspecified pixel format
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, matroska,webm, from '1.webm':
  Metadata:
    encoder         : WS Matroska Muxer
    creation_time   : 2023-09-11T14:55:34.000000Z
  Duration: 00:00:20.00, start: 0.000000, bitrate: 1775 kb/s
  Stream #0:0: Video: vp9 (Profile 3), none, 1080x1080, SAR 1:1 DAR 1:1, 30 fps, 30 tbr, 1k tbn (default)
    Metadata:
      alpha_mode      : 1
  Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)

and

Failed to decode frame: Unspecified internal error
Error while decoding stream #0:0: Invalid data found when processing input
Cannot determine format of input stream 0:0 after EOF


  


  I have tried to increasing the value for the 'analyzeduration' and 'probesize', but it makes no difference.

  


  ffprobe of 1.webm

  


  ffprobe version 6.0 Copyright (c) 2007-2023 the FFmpeg developers
  built with Apple clang version 14.0.3 (clang-1403.0.22.14.1)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/6.0-with-options_4 --enable-shared --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libaom --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-demuxer=dash --enable-opencl --enable-audiotoolbox --enable-videotoolbox --disable-htmlpages --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfdk-aac --enable-libgme --enable-libgsm --enable-libmodplug --enable-libopenh264 --enable-libopenjpeg --enable-libopenmpt --enable-librav1e --enable-libsvtav1 --enable-librist --enable-librsvg --enable-librtmp --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtesseract --enable-libtwolame --enable-libvidstab --enable-libvmaf --enable-libwebp --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-openssl --enable-nonfree --enable-libjack --enable-indev=jack --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb
  libavutil      58.  2.100 / 58.  2.100
  libavcodec     60.  3.100 / 60.  3.100
  libavformat    60.  3.100 / 60.  3.100
  libavdevice    60.  1.100 / 60.  1.100
  libavfilter     9.  3.100 /  9.  3.100
  libswscale      7.  1.100 /  7.  1.100
  libswresample   4. 10.100 /  4. 10.100
  libpostproc    57.  1.100 / 57.  1.100
Input #0, matroska,webm, from '1.webm':
  Metadata:
    encoder         : WS Matroska Muxer
    creation_time   : 2023-09-11T14:55:34.000000Z
  Duration: 00:00:20.00, start: 0.000000, bitrate: 1775 kb/s
  Stream #0:0: Video: vp8, yuv420p(progressive), 1080x1080, SAR 1:1 DAR 1:1, 30 fps, 30 tbr, 1k tbn (default)
    Metadata:
      alpha_mode      : 1
  Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)


  


  Any ffmpeg wizards with an idea ?

  


• ffmpeg mp3 chunk to wav chunk adds gap in the start of the audio

  13 décembre 2023, par 1Mayur

  I have an mp3 streaming from a URL, I save the chunks in 1024 byes buffer size.
After I get all the chunks, I'm using ffmpeg to convert the incoming mp3 chunk (22050 mono) to a wav chunk.

  


  When I open/play the wav chunk I see that there is an empty gap at the start of every chunk.

  


  here is the code I'm running in Python subprocess in a loop for all the saved chunks

  


  subprocess.run(["ffmpeg", "-i",
    f"{Path.cwd()}/input/{path}",
    f"{Path.cwd()}/temp_output/{path.replace('.mp3', '')}.wav"
])


  


  here is the output in the terminal

  


  processing: test-016.mp3
ffmpeg version 6.0 Copyright (c) 2000-2023 the FFmpeg developers
  built with Apple clang version 15.0.0 (clang-1500.0.40.1)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/6.0_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox
  libavutil      58.  2.100 / 58.  2.100
  libavcodec     60.  3.100 / 60.  3.100
  libavformat    60.  3.100 / 60.  3.100
  libavdevice    60.  1.100 / 60.  1.100
  libavfilter     9.  3.100 /  9.  3.100
  libswscale      7.  1.100 /  7.  1.100
  libswresample   4. 10.100 /  4. 10.100
  libpostproc    57.  1.100 / 57.  1.100
[mp3 @ 0x7fd48e104480] Format mp3 detected only with low score of 25, misdetection possible!
[mp3 @ 0x7fd48e104480] Skipping 463 bytes of junk at 0.
[mp3 @ 0x7fd48e104480] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '/Users/mayur/Projects/input/test-016.mp3':
  Duration: 00:00:00.39, start: 0.000000, bitrate: 169 kb/s
  Stream #0:0: Audio: mp3, 22050 Hz, mono, fltp, 160 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '/Users/mayur/Projects/temp_output/test-016.wav':
  Metadata:
    ISFT            : Lavf60.3.100
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
    Metadata:
      encoder         : Lavc60.3.100 pcm_s16le
size=      17kB time=00:00:00.36 bitrate= 379.7kbits/s speed= 253x    
video:0kB audio:17kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.451389%


  


  I tried the pydub as well and faced similar issue.