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• How do I change audio language using ffmpeg in video files contaning two or more audio tracks

  19 mai 2020, par warezsoftwarez

  I am transcoding video files, that contain 2 audio tracks - Polish and English. Unfortunately my transcoder messes somehow with metadata, and at the output, when I'm checking video file using MediaInfo, i see that both audio tracks are described as English. I'd like to change description of 1st audio track to Polish, so accordingly to ffmpeg documentation :

  



  To set the language of the first audio stream:
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT


  



  Unfortunately, running command :

  



  ffmpeg -i INPUTPATH -metadata:s:a:0 language=pol OUTPUTPATH


  



  to overwrite input file resulted in saving file thats size is 63KB but it is described as Polish audio (no second audio track - English somehow was deleted)

  



  When I used this command, to save output file as another file, to avoid overwriting if something misses (which is current situation), but using parameter -metadata:s:a:4 because I thought that maybe I'm reading metadata from ffmpeg incorrectly, it resulted in saving output file as 800Mb with only English (no Polish track).

  



  I don't know what should I change in this command to run as I want it to - just changing metadata of first audio track language to Polish.

  



  Here is full ffmpeg output :

  



  C:\Users\user>ffmpeg -i F:\Path\33244813_6000_1_trailer.mp4 -metadata:s:a:4 language=pol F:\Path\testtest.mp4
ffmpeg version N-91931-gb69ea742ab Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 8.2.1 (GCC) 20180813
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enabl
e-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr
wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enab
le-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --
enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc -
-enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e
nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  libavutil      56. 19.101 / 56. 19.101
  libavcodec     58. 30.100 / 58. 30.100
  libavformat    58. 18.100 / 58. 18.100
  libavdevice    58.  4.103 / 58.  4.103
  libavfilter     7. 29.100 /  7. 29.100
  libswscale      5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc    55.  2.100 / 55.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'F:\Path\33244813_6000_1_trailer.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 1
    compatible_brands: isom
    creation_time   : 2020-04-22T16:18:04.000000Z
  Duration: 00:40:39.12, start: 0.000000, bitrate: 8848 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], Closed C
aptions, 5068 kb/s, 25 fps, 25 tbr, 120k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Video
    Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], Closed C
aptions, 1857 kb/s, 25 fps, 25 tbr, 120k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Video
    Stream #0:2(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], Closed Ca
ptions, 902 kb/s, 25 fps, 25 tbr, 120k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Video
    Stream #0:3(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 640x480 [SAR 4:3 DAR 16:9], Closed Cap
tions, 628 kb/s, 25 fps, 25 tbr, 120k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Video
    Stream #0:4(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Sound
    Stream #0:5(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 253 kb/s (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Sound
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:5 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 00000012bef91f80] using SAR=1/1
[libx264 @ 00000012bef91f80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 00000012bef91f80] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 00000012bef91f80] 264 - core 157 r2932 303c484 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.vid
eolan.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=34 looka
head_threads=5 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_lookah
ead=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 'F:\Path\testtest.mp4'
:
  Metadata:
    major_brand     : isom
    minor_version   : 1
    compatible_brands: isom
    encoder         : Lavf58.18.100
    Stream #0:0(und): Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps,
 12800 tbn, 25 tbc (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Video
      encoder         : Lavc58.30.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 341 kb/s (default)
    Metadata:
      creation_time   : 2020-04-22T16:18:04.000000Z
      handler_name    : Sound
      encoder         : Lavc58.30.100 aac
frame=60981 fps= 84 q=-1.0 Lsize=  816313kB time=00:40:39.12 bitrate=2741.7kbits/s dup=3 drop=0 speed=3.36x
video:713292kB audio:101322kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.208512%
[libx264 @ 00000012bef91f80] frame I:769   Avg QP:16.36  size: 46713
[libx264 @ 00000012bef91f80] frame P:37125 Avg QP:19.81  size: 15059
[libx264 @ 00000012bef91f80] frame B:23087 Avg QP:21.11  size:  5866
[libx264 @ 00000012bef91f80] consecutive B-frames: 40.4% 24.0% 10.2% 25.5%
[libx264 @ 00000012bef91f80] mb I  I16..4: 40.3% 58.0%  1.7%
[libx264 @ 00000012bef91f80] mb P  I16..4:  6.2% 12.5%  0.1%  P16..4: 29.8%  3.7%  2.7%  0.0%  0.0%    skip:45.0%
[libx264 @ 00000012bef91f80] mb B  I16..4:  1.0%  1.7%  0.0%  B16..8: 27.9%  1.1%  0.1%  direct: 2.0%  skip:66.2%  L0:49
.4% L1:49.3% BI: 1.3%
[libx264 @ 00000012bef91f80] 8x8 transform intra:65.3% inter:95.1%
[libx264 @ 00000012bef91f80] coded y,uvDC,uvAC intra: 24.7% 39.7% 3.2% inter: 7.0% 14.6% 0.0%
[libx264 @ 00000012bef91f80] i16 v,h,dc,p: 34% 26% 13% 28%
[libx264 @ 00000012bef91f80] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 15% 41%  3%  3%  4%  3%  3%  2%
[libx264 @ 00000012bef91f80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 25% 20%  3%  6%  5%  4%  3%  2%
[libx264 @ 00000012bef91f80] i8c dc,h,v,p: 58% 19% 20%  3%
[libx264 @ 00000012bef91f80] Weighted P-Frames: Y:0.1% UV:0.0%
[libx264 @ 00000012bef91f80] ref P L0: 72.2%  6.3% 15.8%  5.7%  0.0%
[libx264 @ 00000012bef91f80] ref B L0: 86.6% 11.5%  1.9%
[libx264 @ 00000012bef91f80] ref B L1: 97.6%  2.4%
[libx264 @ 00000012bef91f80] kb/s:2395.53
[aac @ 00000012bef93b00] Qavg: 471.415


  


• 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.

  


• FFmpeg loglevel with level prefix

  13 mai 2020, par Matan Marciano

  I want to add level prefix to ffmpeg logs.
Im using ffmpeg -loglevel level+info -i input
And it is working as well in my Mac (ios) but is not working in my Ubuntu 18 with ffmpeg version 3.4.6-0ubuntu0.18.04.1

  



  When I am running it in my Ubuntu I got :

  



  Invalid loglevel "level+info". Possible levels are numbers or:
"quite"
"panic"
...
...
...
"trace"


  



  Please advise