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• Ffmpeg frame extraction with Nvidia GPU acceleration throws "Output file #0 does not contain any stream"

  16 mai 2020, par nickthefreak

  I am trying to use nvidia gpu accelerated decoder api with ffmpeg, to extract all frames from a video file (.MTS) to a folder, but it looks like it's failing for some reason ; I could not find an answer or similar issues.

  



  Command used :

  



  ffmpeg -vsync 0 -hwaccel cuvid -c:v mpeg2_cuvid -i raw_video.MTS -q:v 2 -f image2 output_folder/image_%05d.jpg

  



  Traceback :

  



  ffmpeg version n4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.3.0 (Arch Linux 9.3.0-1)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mpegts @ 0x563fcc5616c0] start time for stream 0 is not set in estimate_timings_from_pts
[mpegts @ 0x563fcc5616c0] PES packet size mismatch
[mpegts @ 0x563fcc5616c0] Could not find codec parameters for stream 0 (Video: mpeg2video (HDMV / 0x564D4448), none(tv)): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpegts, from 'raw_video.MTS':
  Duration: 00:07:15.68, start: 1010.210356, bitrate: 41186 kb/s
  Program 1 
    Stream #0:0[0x1011]: Video: mpeg2video (HDMV / 0x564D4448), none(tv), 90k tbr, 90k tbn, 90k tbc
    Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s
    Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090), 1920x1080
Output #0, image2, to 'output_folder/image_%05d.jpg':
Output file #0 does not contain any stream


  



  I am pretty sure -hwaccel cuvid -c:v mpeg2_cuvid is correct as the file type seems to be MPEG-2 in the file properties, but similar issues happen with the other cuvid decoders as well :

  



  

  



  I have also tried to run without -c:v flag but then a cuda error is raised and it runs on the cpu :

  



  [h264 @ 0x55949e6d7e00] decoder->cvdl->cuvidCreateDecoder(&decoder->decoder, params) failed -> CUDA_ERROR_INVALID_VALUE: invalid argument
[h264 @ 0x55949e6d7e00] Failed setup for format cuda: hwaccel initialisation returned error.

  



  Any help will be much appreciated.

  



  Edit :

  



  

  • OS : Arch Linux

  


  • GPU : Nvidia 1050Ti

  


  • CUDA Version : 10.2

  


  • NVIDIA-SMI : 440.82

  


  



  Edit 2 :

  



  ffmpeg version n4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.3.0 (Arch Linux 9.3.0-1)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] SPS unavailable in decode_picture_timing
[h264 @ 0x557143911700] non-existing PPS 0 referenced
[h264 @ 0x557143911700] decode_slice_header error
[h264 @ 0x557143911700] no frame!
[mpegts @ 0x55714390c540] PES packet size mismatch
Input #0, mpegts, from 'raw_video.MTS':
  Duration: 00:18:30.97, start: 113.284733, bitrate: 16850 kb/s
  Program 1 
    Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p(top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s
    Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090), 1920x1080
At least one output file must be specified


  


• FFmpeg Opus choppy sound UPDATED DESCRIPTION

  2 juin 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 does 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, frameDecode->channels, AV_SAMPLE_FMT_S16);


  



  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 ?

  



  UPD 18.05.2020

  



  I tested my resampling logic, I did resampling of raw pcm sound without any encoding and decoding routines. First I tried to convert the sample rate of input sound from 8000 Hz to 48000 Hz than I took resampled samples from step above and convert it's sample rate from 48000 Hz to 8000 Hz and the result sound is perfect and clean, also I did the same steps but I converted not a sample rate but a sample format from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP and vice versa and again the result sound is perfect and clean, also I got the same result when I coverted both a sample rate and a sample format.
So I assume that the problem of distorted and choppy sound is in my encoding or decoding routine, I think most likely in decoding routine because after decoding I ALWAYS get AVFrame with 960 nb_samples despite what was the size of input sound.

  



  My decoding routine looks like this :

  



  std::vector decode(uint8_t *data, unsigned int dataLength)
{
    decodedData.clear();

    int dataSize = dataLength;

    while (dataSize > 0)
    {
        if (!frameDecode)
        {
            frameDecode = av_frame_alloc();
            if (!frameDecode)
            {
                LOGE(TAG, "[decode] Couldn't allocate the frame");
                return EMPTY_DATA;
            }
        }

        ret = av_parser_parse2(parser, contextDecoder, &packetDecode->data, &packetDecode->size, &data[0], dataSize, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
        if (ret < 0) {
            LOGE(TAG, "[decode] av_parser_parse2 error: %s", av_err2str(ret));
            return EMPTY_DATA;
        }

        data += ret;
        dataSize -= ret;

        doDecode();
    }
    return decodedData;
}

void doDecode()
{
    if (packetDecode->size) {
        /* send the packet with the compressed data to the decoder */
        int ret = avcodec_send_packet(contextDecoder, packetDecode);
        if (ret < 0) LOGE(TAG, "[decode] avcodec_send_packet error: %s", av_err2str(ret));

        /* read all the output frames (in general there may be any number of them */
        while (ret >= 0)
        {
            ret = avcodec_receive_frame(contextDecoder, frameDecode);
            if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) LOGE(TAG, "[decode] avcodec_receive_frame error: %s", av_err2str(ret));
            if (ret < 0) break;

            std::vector resampledData = resamplerDecoder->convert(frameDecode->data, frameDecode->nb_samples, frameDecode->channels, AV_SAMPLE_FMT_S16);
            if (!resampledData.size()) continue;
            std::copy(&resampledData.data()[0], &resampledData.data()[resampledData.size()], std::back_inserter(decodedData));
        }
    }
}


  



  UPD 30.05.2020

  



  I decided to refuse to use FFmpeg in my project and use libopus 1.3.1 instead, so I made a wrapper around it and it works fine.

  


• Taking care of silent priming frames converting mp4 to ts ?

  13 février 2019, par keepitterron

  videoOutputSettings = [
  
    AVVideoCodecKey: AVVideoCodecH264,
  
    AVVideoWidthKey: width,
  
    AVVideoHeightKey: height,
  
    AVVideoCompressionPropertiesKey: [
  
      AVVideoAverageBitRateKey: avgBitRate,
  
      AVVideoExpectedSourceFrameRateKey: fps,
  
      AVVideoProfileLevelKey: AVVideoProfileLevelH264BaselineAutoLevel
  
    ]
  
  ]
  
  audioOutputSettings = [
  
    AVFormatIDKey: kAudioFormatMPEG4AAC,
  
    AVSampleRateKey: 44100,
  
    AVNumberOfChannelsKey: 2
  
  ]

  I record a video + audio with AVCaptureSession (screen, camera and mic in OSX) and i encode it with AVAssetWriter (AVFileType.mp4), swapping the writers every 6 seconds.
  Every time a new part is written, a nodeJS app will be notified and using ffmpeg will convert it to ts with

  ffmpeg -y -i generatedFile.mp4 -c copy -copyts -muxdelay 0 -muxpreload 0 outputFile.ts

  I’ll manually write a m3u8 looking like :

  #EXTM3U
  
  #EXT-X-VERSION:3
  
  #EXT-X-TARGETDURATION:7
  
  #EXT-X-MEDIA-SEQUENCE:0
  
  #EXT-X-PLAYLIST-TYPE:VOD
  
  #EXTINF:6.140227,
  
  11214532343c4cae953d45e94a1660ea-0.ts
  
  #EXTINF:6.284218,
  
  11214532343c4cae953d45e94a1660ea-1.ts
  
  #EXTINF:6.099999999999998,
  
  11214532343c4cae953d45e94a1660ea-2.ts
  
  #EXTINF:6.133333,
  
  11214532343c4cae953d45e94a1660ea-3.ts
  
  #EXTINF:6.133333,
  
  11214532343c4cae953d45e94a1660ea-4.ts
  
  #EXTINF:6.100000000000001,
  
  11214532343c4cae953d45e94a1660ea-5.ts
  
  #EXTINF:6.133333,
  
  11214532343c4cae953d45e94a1660ea-6.ts
  
  #EXTINF:6.199999999999996,
  
  11214532343c4cae953d45e94a1660ea-7.ts
  
  #EXTINF:6.0666670000000025,
  
  11214532343c4cae953d45e94a1660ea-8.ts
  
  #EXT-X-ENDLIST

  The issue is : I have audio gaps between parts and I believe is AAC’s priming frames not being handled correctly.
  Am I correct in assuming so ?

  example file

  ❯ afinfo 11.mp4
  
  File:           11.mp4
  
  File type ID:   mp4f
  
  Num Tracks:     1
  
  ----
  
  Data format:     2 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
                  no channel layout.
  
  estimated duration: 6.128617 sec
  
  audio bytes: 87558
  
  audio packets: 266
  
  bit rate: 113407 bits per second
  
  packet size upper bound: 401
  
  maximum packet size: 401
  
  audio data file offset: 577163
  
  optimized
  
  audio 270272 valid frames + 2112 priming + 0 remainder = 272384
  
  format list:
  
  [ 0] format:      2 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
  
  Channel layout: Stereo (L R)
  
  ----

  example file

  ❯ ffprobe -v error -show_format -show_streams 1.mp4
  
  [STREAM]
  
  index=0
  
  codec_name=h264
  
  codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
  
  profile=Baseline
  
  codec_type=video
  
  codec_time_base=1/60
  
  codec_tag_string=avc1
  
  codec_tag=0x31637661
  
  width=1920
  
  height=1080
  
  coded_width=1920
  
  coded_height=1088
  
  has_b_frames=0
  
  sample_aspect_ratio=1:1
  
  display_aspect_ratio=16:9
  
  pix_fmt=yuv420p
  
  level=40
  
  color_range=tv
  
  color_space=bt709
  
  color_transfer=bt709
  
  color_primaries=bt709
  
  chroma_location=bottom
  
  field_order=unknown
  
  timecode=N/A
  
  refs=1
  
  is_avc=true
  
  nal_length_size=4
  
  id=N/A
  
  r_frame_rate=30/1
  
  avg_frame_rate=30/1
  
  time_base=1/600
  
  start_pts=3600
  
  start_time=6.000000
  
  duration_ts=7260
  
  duration=12.100000
  
  bit_rate=4227137
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=8
  
  nb_frames=183
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  TAG:creation_time=2019-02-13T13:21:51.000000Z
  
  TAG:language=und
  
  TAG:handler_name=Core Media Video
  
  [/STREAM]
  
  [STREAM]
  
  index=1
  
  codec_name=aac
  
  codec_long_name=AAC (Advanced Audio Coding)
  
  profile=LC
  
  codec_type=audio
  
  codec_time_base=1/44100
  
  codec_tag_string=mp4a
  
  codec_tag=0x6134706d
  
  sample_fmt=fltp
  
  sample_rate=44100
  
  channels=2
  
  channel_layout=stereo
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/44100
  
  start_pts=272991
  
  start_time=6.190272
  
  duration_ts=545352
  
  duration=12.366259
  
  bit_rate=115688
  
  max_bit_rate=128000
  
  bits_per_raw_sample=N/A
  
  nb_frames=269
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  TAG:creation_time=2019-02-13T13:21:51.000000Z
  
  TAG:language=und
  
  TAG:handler_name=Core Media Audio
  
  [/STREAM]
  
  [FORMAT]
  
  filename=1.mp4
  
  nb_streams=2
  
  nb_programs=0
  
  format_name=mov,mp4,m4a,3gp,3g2,mj2
  
  format_long_name=QuickTime / MOV
  
  start_time=6.000000
  
  duration=12.366259
  
  size=3317034
  
  bit_rate=2145860
  
  probe_score=100
  
  TAG:major_brand=mp42
  
  TAG:minor_version=1
  
  TAG:compatible_brands=mp41mp42isom
  
  TAG:creation_time=2019-02-13T13:21:51.000000Z
  
  [/FORMAT]

  Is there a way I can generate ts files with the encoder delay being taken care of ?