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• Add sound tracks to a video file with FFmpeg [on hold]

  26 juillet 2014, par user3877422

  I look this page : https://zoid.cc/ffmpeg-audio-video/ but i don’t understand how to add multiple audio to a video, i have a problem.

  My video length : 04:30
  My audio1 length : 01:30
  My audio2 length : 02:00

  I tried :

  ffmpeg -i vid.avi -i audio1.wav -i audio2.wav -map 0:0 -map 0:1 -map 1:0 -map 2:0 -c:v copy -c:a copy output.avi

  FFMpeg set is video length for audio1.wav length (01:30)

  I want add audio but not change video length (04:30)
  How to fix this problem ?

  Thanks for help.

• Video concatenation puts sound out of sync

  9 août 2019, par mmorin

  (Cross-posted from Video Production, where the question received no answers and may be more technical than usual video production.)

  I have several MOV files from a DSLR camera. I concatenate them with directions from this thread :

  ffmpeg -safe 0 -f concat -i files_to_combine -vcodec copy -acodec copy temp.MOV

  where files_to_combine is :

  file ./DSC_0013.MOV
  
  ...
  
  file ./DSC_0019.MOV

  The result has image and sound in sync for the first clip and is out of sync by fractions of a second in the second clip, and out of sync by around a second for the last clip. It is probably related to this error from the log :

  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] st: 0 edit list: 1 Missing key frame while searching for timestamp: 1000
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] st: 0 edit list 1 Cannot find an index entry before timestamp: 1000.
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] Auto-inserting h264_mp4toannexb bitstream filter

  How can I trim the frames to the available sound stream, then concatenate the two videos ?

  The full log from the ffmpeg command is :

  ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.3_1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dc00e000] Auto-inserting h264_mp4toannexb bitstream filter
  
  Input #0, concat, from 'files_to_combine':
  
    Duration: N/A, start: -0.592000, bitrate: 36888 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, smpte170m/bt709/bt470m), 1920x1080, 35352 kb/s, 50 fps, 50 tbr, 50k tbn, 100 tbc
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Output #0, mov, to 'temp.MOV':
  
    Metadata:
  
      encoder         : Lavf58.20.100
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, smpte170m/bt709/bt470m), 1920x1080, q=2-31, 35352 kb/s, 50 fps, 50 tbr, 50k tbn, 50k tbc
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] st: 0 edit list: 1 Missing key frame while searching for timestamp: 1000
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] st: 0 edit list 1 Cannot find an index entry before timestamp: 1000.
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f82dd802200] Auto-inserting h264_mp4toannexb bitstream filter
  
  frame=41886 fps=547 q=-1.0 Lsize= 3789826kB time=00:13:58.75 bitrate=37014.8kbits/s speed=10.9x    
  
  video:3631879kB audio:157123kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.021759%

  Update (1 July 2019)

  I thought that the files had a problem at the beginning or at the end, so I
  trimmed one second from each end, but it still had the sound out of sync :

  FILES=files_to_combine
  
  OUTPUT=show2.MOV
  
  rm $FILES
  
  for i in 3 4 5 6 7 8 9; do
  
      rm ${i}.MOV
  
      duration=$(ffprobe -v 0 -show_entries format=duration -of compact=p=0:nk=1  DSC_001${i}.MOV)
  
      trimmed=$(echo $duration - 1 | bc)
  
      ffmpeg -ss 1 -t $trimmed -i DSC_001${i}.MOV -vcodec copy -acodec copy ${i}.MOV
  
      echo file ./${i}.MOV >> $FILES
  
  done
  
  
  
  rm $OUTPUT
  
  ffmpeg -safe 0 -f concat -i $FILES -vcodec copy -acodec copy $OUTPUT

  When I trim a single file near the end, the sound and video do not seem out of sync :

  ffmpeg -ss 00:09:20 -t 20 -i DSC_0014.MOV -vcodec copy -acodec copy end.MOV

  When I concatenate only 30 seconds from each video, the result seems OK :

  FILES=files_to_combine
  
  OUTPUT=show2.MOV
  
  rm $FILES
  
  for i in 3 4 5 6 7 8 9; do
  
      rm ${i}.MOV
  
      duration=$(ffprobe -v 0 -show_entries format=duration -of compact=p=0:nk=1  DSC_001${i}.MOV)
  
      start=$(echo $duration - 30 | bc)
  
      end=$(echo $duration - 1 | bc)
  
      ffmpeg -ss $start -t $end -i DSC_001${i}.MOV -vcodec copy -acodec copy ${i}.MOV
  
      echo file ./${i}.MOV >> $FILES
  
  done
  
  
  
  rm $OUTPUT
  
  ffmpeg -safe 0 -f concat -i $FILES -vcodec copy -acodec copy $OUTPUT

  This last concatenation gives this error multiple times :

  [mov @ 0x7fc3c7837400] Non-monotonous DTS in output stream 0:0; previous: 9080205, current: 9080200; changing to 9080206. This may result in incorrect timestamps in the output file.

  So I am guessing that the problem is small differences in timestamps that
  accumulate and become more noticeable with longer durations and the
  concatenation of multiple files.

  For reference, the DSLR that shot these clips is a Nikon D3300 and the result
  of ffprobe on one of the files is :

  $ ffprobe DSC_0017.MOV -hide_banner
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fab70003800] st: 0 edit list: 1 Missing key frame while searching for timestamp: 1000
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fab70003800] st: 0 edit list 1 Cannot find an index entry before timestamp: 1000.
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'DSC_0017.MOV':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537331968
  
      compatible_brands: qt  niko
  
      creation_time   : 2019-06-12T23:52:37.000000Z
  
    Duration: 00:09:53.58, start: 0.000000, bitrate: 36843 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, smpte170m/bt709/bt470m), 1920x1080, 35300 kb/s, 50 fps, 50 tbr, 50k tbn, 100 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-06-12T23:52:37.000000Z
  
      Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 2 channels, s16, 1536 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-06-12T23:52:37.000000Z

  Update (9 August 2019)

  I concatenated the files in iMovie and the sound and image are not as out of sync as with FFMPEG. Maybe iMovie aligns the timestamps at the end of each clip instead of concatenating the audio and image streams separately.

  I ran the concatenation again with the latest ffmpeg 4.1.4_1 on these files and others from the same camera. The audio and image are in sync in one case (the results lasts 46 minutes) out of sync in another (the result lasts 48 minutes).

• FFmpeg Opus choppy sound

  15 mai 2020, par easy_breezy

  I'm using FFmpeg and try to encode and decode a raw PCM sound to Opus using a built-in FFmpeg "opus" codec. My input samples are raw PCM 8000 Hz 16 bit mono, in AV_SAMPLE_FMT_S16 format. Since Opus requires sample format AV_SAMPLE_FMT_FLTP and sample rate 48000 Hz only, so I resample my samples before encode them.

  



  I have two instances of ResamplerAudio class that does the work of resampling audio samples and has a member of SwrContext, I use the first instance of ResamplerAudio for resampling a raw PCM input audio before encoding and the second for resampling decoded audio to get it's format and sample rate the same as source values of input raw audio.

  



  ResamplerAudio class has a function that init it's SwrContext member like this :

  



  void ResamplerAudio::init(AVCodecContext *codecContext, int inSampleRate, int outSampleRate, AVSampleFormat inSampleFmt, AVSampleFormat outSampleFmt)
{
    swrContext = swr_alloc();
    if (!swrContext)
    {
        LOGE(TAG, "[init] Couldn't allocate swr context");
        return;
    }

    av_opt_set_int(swrContext, "in_channel_layout", (int64_t) codecContext->channel_layout, 0);
    av_opt_set_int(swrContext, "out_channel_layout", (int64_t) codecContext->channel_layout,  0);

    av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
    av_opt_set_int(swrContext, "out_channel_count", codecContext->channels, 0);

    av_opt_set_int(swrContext, "in_sample_rate", inSampleRate, 0);
    av_opt_set_int(swrContext, "out_sample_rate", outSampleRate, 0);

    av_opt_set_sample_fmt(swrContext, "in_sample_fmt", inSampleFmt, 0);
    av_opt_set_sample_fmt(swrContext, "out_sample_fmt", outSampleFmt,  0);

    int ret = swr_init(swrContext);
    if (ret < 0)
    {
        LOGE(TAG, "[init] swr_init error: %s", av_err2str(ret));
        return;
    }

    LOGD(TAG, "[init] success codecContext->channel_layout: %d; inSampleRate: %d; outSampleRate: %d; inSampleFmt: %d; outSampleFmt: %d", (int) codecContext->channel_layout, inSampleRate, outSampleRate, inSampleFmt, outSampleFmt);
}


  



  And I call ResamplerAudio::init function for the first instance of ResamplerAudio (this instance do resamping a raw PCM input audio before encoding and I called it resamplerEncoder) with the following args :

  



  resamplerEncoder->init(contextEncoder, 8000, 48000, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP);


  



  The second instance of ResamplerAudio (this instance do resamping after decoding audio from Opus and I called it resamplerDecoder) I init with the following args :

  



  resamplerDecoder->init(contextDecoder, 48000, 8000, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16);


  



  The function of ResamplerAudio that do resampling looks like this :

  



  std::vector ResamplerAudio::convert(uint8_t **inData, int inSamplesCount, int outChannels, int outFormat)
{
    std::vector result;
    uint8_t *dstData = NULL;
    const int dstNbSamples = swr_get_out_samples(swrContext, inSamplesCount);
    av_samples_alloc(&dstData, NULL, outChannels, dstNbSamples, AVSampleFormat(outFormat), 1);
    int resampledSize = swr_convert(swrContext, &dstData, dstNbSamples, (const uint8_t **)inData, inSamplesCount);
    int dstBufSize = av_samples_get_buffer_size(NULL, outChannels, resampledSize, AVSampleFormat(outFormat), 1);

    if (dstBufSize <= 0) return result;

    std::copy(&dstData[0], &dstData[dstBufSize], std::back_inserter(result));

    return result;
}


  



  And I call ResamplerAudio::convert function before encoding with the following args :

  



  // data - an array of raw pcm audio
// dataLength - the length of data array
// getSamplesCount() - function that calculates samples count
// frameEncode - AVFrame that using for encode audio
std::vector resampledData = resamplerEncoder->convert(&data, getSamplesCount(dataLength, frameEncode->channels, AV_SAMPLE_FMT_S16), frameEncode->channels, frameEncode->format);


  



  getSamplesCount() function looks like this :

  



  getSamplesCount(int bytesCount, int channels, AVSampleFormat format)
{
    return bytesCount / av_get_bytes_per_sample(format) / channels;
}


  



  After that I fill my frameEncode with resampled samples :

  



  memcpy(&frame->data[0][0], &resampledData[0], sizeof(uint8_t) * resampledDataLength);


  



  And pass frameEncode to encoding like this encodeFrame(resampledDataLength) :

  



  void encodeFrame(int dataLength)
{
    /* send the frame for encoding */
    int ret = avcodec_send_frame(contextEncoder, frameEncode);
    if (ret < 0)
    {
        LOGE(TAG, "[encodeFrame] avcodec_send_frame error: %s", av_err2str(ret));
        return;
    }

    /* read all the available output packets (in general there may be any number of them */
    while (ret >= 0)
    {
        ret = avcodec_receive_packet(contextEncoder, packetEncode);
        if (ret < 0 && ret != AVERROR(EAGAIN)) LOGE(TAG, "[encodeFrame] error in avcodec_receive_packet: %s", av_err2str(ret));
        if (ret < 0) break;

        // encodedData - std::vector that stores encoded data
        std::copy(&packetEncode->data[0], &packetEncode->data[dataLength], std::back_inserter(encodedData));
        av_packet_unref(packetEncode);
    }
}


  



  Then I decode my encoded samples and do resampling to get back them in source sample format and sample rate so I call ResamplerAudio::convert function for resamplerDecoder with the following args :

  



  // frameDecode - AVFrame that holds decoded audio
std::vector resampledData = resamplerDecoder->convert(frameDecode->data, frameDecode->nb_samples, AV_SAMPLE_FMT_S16, frameDecode->channels);


  



  And result sound is choppy and I also noticed that the decoded array size is bigger than the source array size with raw pcm audio.

  



  Please any ideas what I'm doing wrong ?