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• ffprobe different results video duration using pipe and reading a file from the file system

  5 février 2024, par alex

  I have a method to convert a video file, after processing the file I use pipe to pass bytes to a method to get meta information about the file using pipe. But in this case I get wrong duration of video file, 8.22, but if I save the file on file system and read it to get meta information I get result 15.85. Why is this happening ?

  


  Video Convert method :

  


  // ConvertVideoWithPath converts a video file specified by its path using FFmpeg.
// It returns the converted video data and any error that occurred during conversion.
func (f *FFmpeg) ConvertVideoWithPath(filePath string) (bytes []byte, err error) {
    if filePath == "" {
        return nil, ErrEmptyPath
    }

    // Create a CmdRunner instance for executing FFmpeg.
    commander := &CmdRunner{}
    commander.Command = "ffmpeg"
    args := []string{
        "-loglevel", "fatal",
        "-i", filePath,
        "-y",
        "-filter:v", "crop=trunc(iw/2)*2:trunc(ih/2)*2",
        "-c:v", f.videoCodec, // libx264
        "-c:a", f.audioCodec, // aac
        "-pix_fmt", "yuv420p",
        "-movflags", "frag_keyframe+faststart",
        "-map_metadata", "-1",
        "-crf", "5",
        "-vsync", "2",
        "-bufsize", "15000000",
        "-maxrate", "5000000",
        "-preset", "medium",
        "-f", "mp4",
        "pipe:1",
    }
    commander.Args = args

    // Initialize output pipe.
    reader := commander.InitStdOutPipe()

    // Use WaitGroup to synchronize goroutines.
    wg := &sync.WaitGroup{}
    wg.Add(1)

    // Goroutine for reading data from the output pipe.
    go func() {
        defer reader.Close()
        defer wg.Done()

        // Read data from the output pipe.
        data, _ := io.ReadAll(reader)
        // Safely update the 'bytes' variable.
        f.mutex.Lock()
        bytes = data
        f.mutex.Unlock()
    }()

    // Run the FFmpeg command with pipes and wait for completion.
    err = <-commander.RunWithPipe()
    wg.Wait()

    return
}


  


  // MetadataWithReader retrieves metadata from media data provided by an io.Reader using FFprobe.
// It returns the metadata and any error that occurred during metadata retrieval.
func (f *FFmpeg) MetadataWithReader(fileBytes io.Reader) (*Metadata, error) {
    if fileBytes == nil {
        return nil, ErrInvalidArgument
    }

    // Create a CmdRunner instance for executing FFprobe.
    commander := &CmdRunner{}
    commander.Command = "ffprobe"
    args := []string{
        "-loglevel", "fatal",
        "-i", "pipe:0",
        "-print_format", "json",
        "-show_format", "-show_streams",
        "-show_error",
    }
    commander.Args = args

    // Get output data from FFprobe with pipes.
    err := commander.GetOutputWithPipe(fileBytes)
    if err != nil {
        return nil, err
    }

    // Unmarshal JSON output into a Metadata struct.
    output := &Metadata{}
    err = json.Unmarshal(commander.GetOutput(), output)
    if err != nil {
        return nil, err
    }

    return output, err
}


  


  // MetadataWithPath extracts metadata of a file using FFprobe.
// It returns a Metadata struct or an error if the operation fails.
func (f *FFmpeg) MetadataWithPath(filePath string) (*Metadata, error) {
    if filePath == "" {
        return nil, ErrEmptyPath
    }

    // Create a CmdRunner instance for executing FFprobe.
    commander := &CmdRunner{}
    commander.Command = "ffprobe"
    args := []string{
        "-loglevel", "fatal",
        "-i", filePath,
        "-loglevel",
        "fatal",
        "-print_format", "json",
        "-show_format", "-show_streams", "-show_error",
    }
    commander.Args = args
    buffer := bytes.NewBuffer([]byte{})
    commander.StdOutWriter = buffer

    err := commander.Run()
    if err != nil {
        return nil, err
    }

    // Unmarshal JSON output into a Metadata struct.
    output := &Metadata{}
    err = json.Unmarshal(buffer.Bytes(), output)
    if err != nil {
        return nil, err
    }

    return output, nil
}


  


  The source code of the CmdRunner biblio library can be found here link , so as not to overload the question with a large piece of code.

  


  Unit test code

  


  t.Run("convert video", func(t *testing.T) {
        ffmpeg := NewFFmpeg("aac", "libx264", "24M", "12M")

        filePath := "../../test/testdata/input_video_ts.mp4"
        firstMeta, err := ffmpeg.MetadataWithPath(filePath)
        assert.NoError(t, err)
        fmt.Print("first meta duration: ", firstMeta.Format.DurationSeconds) // 15.75

        outFile := "../../test/testdata/output_mp4.mp4"
        newVideoOut, err := ffmpeg.ConvertVideoWithPath(filePath)
        assert.NoError(t, err)
        assert.NotEmpty(t, newVideoOut)

        meta, err := ffmpeg.MetadataWithReader(bytes.NewBuffer(newVideoOut))
        assert.NoError(t, err)
        assert.NotEmpty(t, meta)

        err = os.WriteFile(outFile, newVideoOut, 0644)
        assert.NoError(t, err)
        assert.FileExists(t, outFile)

        fmt.Print("meta duration: ", meta.Format.DurationSeconds) // 8.22

        secondMeta, err := ffmpeg.MetadataWithPath(outFile)
        assert.NoError(t, err)
        fmt.Print("second meta duration: ", secondMeta.Format.DurationSeconds) //15.85

        err = os.Remove(outFile)
        assert.NoError(t, err)
    })


  


• FFMPEG loudnorm filter does not work in combination with silenceremove filter

  12 mai 2021, par MareikeP

  I want to consistently normalize audio files for TTS model training. The output audio files should meet the following criteria :

  


  

  1. mono channel

  


  2. sample rate of 22050 Hz

  


  3. wav format

  


  4. no silence at beginning and end of audio clip

  


  5. volume of -24 dB

  


  


  I have already fulfilled the first 4 criteria. So far, it works properly.

  


  Normalizing the volume basically works as well with this ffmpeg command -af loudnorm=I=-24:LRA=11:TP=-1.5 , but not in combination with the silence removal : As soon as I remove silence with this ffmpeg command agate=threshold=0.045:attack=0.5:release=500:ratio=5000,silenceremove=start_periods=1:start_threshold=0.0075,areverse,silenceremove=start_periods=1:start_threshold=0.0075,areverse, the loudness normalization does not work any longer : the output volume now varies between -25dB and -32dB instead of the desired -24 dB.

  


  This is the complete ffmpeg command I used :

  


  ffmpeg -i filename.flac -ac 1 -af agate=threshold=0.045:attack=0.5:release=500:ratio=5000,silenceremove=start_periods=1:start_threshold=0.0075,areverse,silenceremove=start_periods=1:start_threshold=0.0075,areverse,loudnorm=I=-24:LRA=11:TP=-1.5,aresample=22050 -y -hide_banner filename.wav


  


  And this is the piece of code that I'm using to run it :

  


  import os

INPUT_DIR = '/home/username/all_data'
OUTPUT_DIR = '/home/username/normalized_data'
for filename in os.listdir(INPUT_DIR):
    wav_filename = filename[:-5] + '.wav'
    command = (f'ffmpeg -i {INPUT_DIR}/{filename} -ac 1 -af agate='
               f'threshold=0.045:attack=0.5:release=500:ratio=5000,'
               f'silenceremove=start_periods=1:start_threshold=0.0075,'
               f'areverse,silenceremove=start_periods=1:start_threshold='
               f'0.0075,areverse,loudnorm=I=-24:LRA=11:TP=-1.5,aresample'
               f'=22050 -y -hide_banner {OUTPUT_DIR}/{wav_filename}')
    os.system(command)


  


  EDIT :

  


  A complete log from the ffmpeg command can be seen here :

  


  username@pop-os:~$ ffmpeg -i /home/username/audios/filename.flac -ac 1 -af agate=threshold=0.045:attack=0.5:release=500:ratio=5000,silenceremove=start_periods=1:start_threshold=0.0075,areverse,silenceremove=start_periods=1:start_threshold=0.0075,areverse,loudnorm=I=-24:LRA=11:TP=-1.5,aresample=22050 /home/username/result.wav
ffmpeg version 4.2.4-1ubuntu0.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  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
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, flac, from '/home/mareike/tts_data/save/audios_flac/0a6c8520-7536-11eb-8338-b7015f354987.flac':
  Duration: 00:00:04.64, start: 0.000000, bitrate: 1090 kb/s
    Stream #0:0: Audio: flac, 44100 Hz, stereo, s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '/home/mareike/result_0a6c8520-7536-11eb-8338-b7015f354987.wav':
  Metadata:
    ISFT            : Lavf58.29.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
    Metadata:
      encoder         : Lavc58.54.100 pcm_s16le
size=     138kB time=00:00:03.19 bitrate= 353.0kbits/s speed=14.3x    
video:0kB audio:138kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.055375%


  


  Can anyone tell me what I'm doing wrong and how I can finally get the volume normalized to -24 dB (in combination with silence removal) ? Any help is appreciated, thank you very much !

  


• ffmpeg error on decode

  25 octobre 2013, par ademar111190

  I'm developing an android app with the libav and I'm trying decode a 3gp with code below :

  #define simbiLog(...) __android_log_print(ANDROID_LOG_DEBUG, "simbiose", __VA_ARGS__)

  ...

  AVCodec *codec;
  
  AVCodecContext *c = NULL;
  
  int len;
  
  FILE *infile, *outfile;
  
  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
  AVPacket avpkt;
  
  AVFrame *decoded_frame = NULL;
  
  
  
  simbiLog("inbuf size: %d", sizeof(inbuf) / sizeof(inbuf[0]));
  
  
  
  av_register_all();
  
  av_init_packet(&avpkt);
  
  
  
  codec = avcodec_find_decoder(AV_CODEC_ID_AMR_NB);
  
  if (!codec) {
  
      simbiLog("codec not found");
  
      return ERROR;
  
  }
  
  
  
  c = avcodec_alloc_context3(codec);
  
  if (!c) {
  
      simbiLog("Could not allocate audio codec context");
  
      return ERROR;
  
  }
  
  
  
  int open = avcodec_open2(c, codec, NULL);
  
  if (open < 0) {
  
      simbiLog("could not open codec %d", open);
  
      return ERROR;
  
  }
  
  
  
  infile = fopen(inputPath, "rb");
  
  if (!infile) {
  
      simbiLog("could not open %s", inputPath);
  
      return ERROR;
  
  }
  
  
  
  outfile = fopen(outputPath, "wb");
  
  if (!outfile) {
  
      simbiLog("could not open %s", outputPath);
  
      return ERROR;
  
  }
  
  
  
  avpkt.data = inbuf;
  
  avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, infile);
  
  int iterations = 0;
  
  
  
  while (avpkt.size > 0) {
  
      simbiLog("iteration %d", (++iterations));
  
      simbiLog("avpkt.size %d avpkt.data %X", avpkt.size, avpkt.data);
  
      int got_frame = 0;
  
  
  
      if (!decoded_frame) {
  
          if (!(decoded_frame = avcodec_alloc_frame())) {
  
              simbiLog("out of memory");
  
              return ERROR;
  
          }
  
      } else {
  
          avcodec_get_frame_defaults(decoded_frame);
  
      }
  
  
  
      //below the error, but it isn't occur on first time, only in 4th loop interation
  
      len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); 
  
      if (len < 0) {
  
          simbiLog("Error while decoding error %d frame %d duration %d", len, got_frame, avpkt.duration);
  
          return ERROR;
  
      } else {
  
          simbiLog("Decoding length %d frame %d duration %d", len, got_frame, avpkt.duration);
  
      }
  
  
  
      if (got_frame) {
  
          int data_size = av_samples_get_buffer_size(NULL, c->channels, decoded_frame->nb_samples, c->sample_fmt, 1);
  
          size_t* fwrite_size = fwrite(decoded_frame->data[0], 1, data_size, outfile);
  
          simbiLog("fwrite returned %d", fwrite_size);
  
      }
  
      avpkt.size -= len;
  
      avpkt.data += len;
  
      if (avpkt.size < AUDIO_REFILL_THRESH) {
  
          memmove(inbuf, avpkt.data, avpkt.size);
  
          avpkt.data = inbuf;
  
          len = fread(avpkt.data + avpkt.size, 1, AUDIO_INBUF_SIZE - avpkt.size, infile);
  
          if (len > 0)
  
              avpkt.size += len;
  
          simbiLog("fread returned %d", len);
  
      }
  
  }
  
  
  
  fclose(outfile);
  
  fclose(infile);
  
  
  
  avcodec_close(c);
  
  av_free(c);
  
  av_free(decoded_frame);

  but I'm getting the follow log and error :

  inbuf size: 20488
  
  iteration 1
  
  avpkt.size 3305 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 2
  
  avpkt.size 3292 avpkt.data BEEED40C
  
  Decoding length 13 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 3
  
  avpkt.size 3279 avpkt.data BEEED40C
  
  Decoding length 14 frame 1 duration 0
  
  fwrite returned 640
  
  fread returned 0
  
  iteration 4
  
  avpkt.size 3265 avpkt.data BEEED40C
  
  Error while decoding error -1052488119 frame 0 duration 0

  the audio file I'm trying decode :

  $ avprobe blue.3gp 
  
  avprobe version 0.8.6-6:0.8.6-1ubuntu2, Copyright (c) 2007-2013 the Libav developers
  
    built on Mar 30 2013 22:23:21 with gcc 4.7.2
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'blue.3gp':
  
    Metadata:
  
      major_brand     : 3gp4
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2013-09-19 18:53:38
  
    Duration: 00:00:01.52, start: 0.000000, bitrate: 17 kb/s
  
      Stream #0.0(eng): Audio: amrnb, 8000 Hz, 1 channels, flt, 12 kb/s
  
      Metadata:
  
        creation_time   : 2013-09-19 18:53:38

  thanks a lot !

  ---

  EDITED

  I read on ffmper documentation about the method avcodec_decode_audio4 the follow :

  @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE larger than the actual read bytes because some optimized bitstream readers read 32 or 64 bits at once and could read over the end.
  
  @note You might have to align the input buffer. The alignment requirements depend on the CPU and the decoder.

  and I see here a solution using posix_memalign, to android i founded a similar method called memalign, so i did the change :

  removed :

  uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];

  inserted :

  int inbufSize = sizeof(uint8_t) * (AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
  
  uint8_t *inbuf = memalign(FF_INPUT_BUFFER_PADDING_SIZE, inbufSize);
  
  simbiLog("inbuf size: %d", inbufSize);
  
  for (; inbufSize >= 0; inbufSize--)
  
      simbiLog("inbuf position: %d index: %p", inbufSize, &inbuf[inbufSize]);

  I'm getting the correct memory sequence position, but the error not changed.

  A piece of outpout :

  inbuf position: 37 index: 0x4e43d745
  
  inbuf position: 36 index: 0x4e43d744
  
  inbuf position: 35 index: 0x4e43d743
  
  inbuf position: 34 index: 0x4e43d742
  
  inbuf position: 33 index: 0x4e43d741
  
  inbuf position: 32 index: 0x4e43d740
  
  inbuf position: 31 index: 0x4e43d73f
  
  inbuf position: 30 index: 0x4e43d73e
  
  inbuf position: 29 index: 0x4e43d73d
  
  inbuf position: 28 index: 0x4e43d73c
  
  inbuf position: 27 index: 0x4e43d73b
  
  inbuf position: 26 index: 0x4e43d73a
  
  inbuf position: 25 index: 0x4e43d739
  
  inbuf position: 24 index: 0x4e43d738
  
  inbuf position: 23 index: 0x4e43d737
  
  inbuf position: 22 index: 0x4e43d736
  
  inbuf position: 21 index: 0x4e43d735
  
  inbuf position: 20 index: 0x4e43d734
  
  inbuf position: 19 index: 0x4e43d733