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• ffmpeg to convert mov to flv

  27 octobre 2014, par jeet

  I’m trying to convert a mov video to flv, but keep getting these errors below.
  There are 2 commands I used, both are below.

  ffmpeg -y -i video.mov -deinterlace -acodec copy -r 25 -qmin 3 -qmax 6 video.flv
  
  
  
  
  
  FFmpeg version SVN-r16573, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  
  configuration: --extra-cflags=-fno-common --enable-memalign-hack --enable-pthreads --enable-libmp3lame --enable-libxvid --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libfaac --enable-libgsm --enable-libx264 --enable-libschroedinger --enable-avisynth --enable-swscale --enable-gpl
  
  libavutil     49.12. 0 / 49.12. 0
  
  libavcodec    52.10. 0 / 52.10. 0
  
  libavformat   52.23. 1 / 52.23. 1
  
  libavdevice   52. 1. 0 / 52. 1. 0
  
  libswscale     0. 6. 1 /  0. 6. 1
  
  built on Jan 13 2009 02:57:09, gcc: 4.2.4
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'vid\video.mov':
  
  Duration: 00:03:16.00, start: 0.000000, bitrate: 398 kb/s
  
  Stream #0.0(eng): Video: mpeg4, yuv420p, 800x600 [PAR 1:1 DAR 4:3], 30.00 tb(r)
  
  Stream #0.1(eng): Audio: pcm_u8, 8000 Hz, mono, s16, 64 kb/s
  
  Output #0, flv, to 'vid\video.flv':
  
  Stream #0.0(eng): Video: flv, yuv420p, 800x600 [PAR 1:1 DAR 4:3], q=3-6, 200 kb/s, 25.00 tb(c)
  
  Stream #0.1(eng): Audio: pcm_u8, 8000 Hz, mono, s16, 64 kb/s
  
  Stream mapping:
  
  Stream #0.0 -> #0.0
  
  Stream #0.1 -> #0.1
  
  [NULL @ 0x1714390]codec not compatible with flv
  
  Could not write header for output file #0 (incorrect codec parameters ?)

  Second command :

  ffmpeg -y -i video.mov -deinterlace -ar 44100 -r 25 -qmin 3 -qmax 6 video.flv

  Audio resampler only works with 16 bits per sample, patch welcome.

  With a newer version of ffmpeg :

  ffmpeg version N-49610-gc2dd5a1 Copyright (c) 2000-2013 the FFmpeg developers
  
  built on Feb  5 2013 13:20:59 with gcc 4.7.2 (GCC)
  
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      52. 17.101 / 52. 17.101
  
    libavcodec     54. 91.100 / 54. 91.100
  
    libavformat    54. 61.104 / 54. 61.104
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 35.101 /  3. 35.101
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  Guessed Channel Layout for  Input Stream #0.1 : mono
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'vid\video.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 512
  
      compatible_brands: qt  
  
      creation_time   : 1970-01-01 00:00:00
  
    Duration: 00:02:50.39, start: 0.000000, bitrate: 370 kb/s
  
      Stream #0:0(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1366x768 [SAR 1:1 DAR 683:384], 308 kb/s, 11.50 fps, 11.50 tbr, 23 tbn, 23 tbc
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : DataHandler
  
      Stream #0:1(eng): Audio: pcm_u8 (raw  / 0x20776172), 8000 Hz, mono, u8, 64 kb/s
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : DataHandler
  
  [flv @ 026347a0] FLV does not support sample rate 8000, choose from (44100, 22050, 11025)
  
  Output #0, flv, to 'vid\video.flv':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 512
  
  compatible_brands: qt  
  
      encoder         : Lavf54.61.104
  
      Stream #0:0(eng): Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 1366x768 [SAR 1:1 DAR 683:384], q=2-31, 200 kb/s, 1k tbn, 11.50 tbc
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : DataHandler
  
      Stream #0:1(eng): Audio: mp3 ([2][0][0][0] / 0x0002), 8000 Hz, mono, s16p
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : DataHandler
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mpeg4 -> flv)
  
    Stream #0:1 -> #0:1 (pcm_u8 -> libmp3lame)
  
  Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input

  1 more thing please :
  If I use this newer version of ffmpeg to create a video with the below command, I get a video with a very hazy display.
  It’s like a few black dots on a blank screen :

  ffmpeg -i img%d.png -i audio.wav -acodec copy output.mov

  what could be the reason for this display ?

• Tracking granular duration from an ffmpeg stream piped into Nodejs

  20 mai 2020, par slifty

  The Situation

  



  I have a video file that I want to process in nodejs, but I want to run it through ffmpeg first in order to normalize the encoding. As I get the data / for a given data event I would like to be able to know how "far" into the video the piped stream has progressed, at as close to frame-level granularity as possible.

  



  0.1 second granularity would be fine.

  



  The Code

  



  I am using Nodejs to invoke ffmpeg to take the video file path and then output the data to stdout :

  



  const ffmpegSettings = [
    '-i', './path/to/video.mp4',       // Ingest a file
    '-f', 'mpegts',                    // Specify the wrapper
    '-'                                // Output to stdout
]
const ffmpegProcess = spawn('ffmpeg', ffmpegSettings)
const readableStream = ffmpegProcess.stdout
readableStream.on('data', async (data) => {
    readableStream.pause() // Pause the stream to make sure we process in order.

    // Process the data.
    // This is where I want to know the "timestamp" for the data.

    readableStream.resume() // Restart the stream.
})


  



  The Question

  



  How can I efficiently and accurately keep track of how far into the video a given 'data' event represents ? Keeping in mind that a given on event in this context might not even have enough data to represent even a full frame.

  



  ---

  



  Temporary Edit

  



  (I will delete this section once a conclusive answer is identified)

  



  Since posting this question I've done some exploration using ffprobe.

  



      start = () => {
        this.ffmpegProcess = spawn('ffmpeg', this.getFfmpegSettings())
        this.ffprobeProcess = spawn('ffprobe', this.getFfprobeSettings())
        this.inputStream = this.getInputStream()

        this.inputStream.on('data', (rawData) => {
            this.inputStream.pause()
            if(rawData) {
                this.ffmpegProcess.stdin.write(rawData)
            }
            this.inputStream.resume()
        })

        this.ffmpegProcess.stdout.on('data', (mpegtsData) => {
            this.ffmpegProcess.stdout.pause()
            this.ffprobeProcess.stdin.write(mpegtsData)
            console.log(this.currentPts)
            this.ffmpegProcess.stdout.resume()
        })

        this.ffprobeProcess.stdout.on('data', (probeData) => {
            this.ffprobeProcess.stdout.pause()
            const lastLine = probeData.toString().trim().split('\n').slice(-1).pop()
            const lastPacket = lastLine.split(',')
            const pts = lastPacket[4]
            console.log(`FFPROBE: ${pts}`)
            this.currentPts = pts
            this.ffprobeProcess.stdout.resume()
        })

        logger.info(`Starting ingestion from ${this.constructor.name}...`)
    }

    /**
     * Returns an ffmpeg settings array for this ingestion engine.
     *
     * @return {String[]} A list of ffmpeg command line parameters.
     */
    getFfmpegSettings = () => [
        '-loglevel', 'info',
        '-i', '-',
        '-f', 'mpegts',
        // '-vf', 'fps=fps=30,signalstats,metadata=print:key=lavfi.signalstats.YDIF:file=\'pipe\\:3\'',
        '-',
    ]

    /**
     * Returns an ffprobe settings array for this ingestion engine.
     *
     * @return {String[]} A list of ffprobe command line parameters.
     */
    getFfprobeSettings = () => [
        '-f', 'mpegts',
        '-i', '-',
        '-print_format', 'csv',
        '-show_packets',
    ]


  



  This pipes the ffmpeg output into ffprobe and uses that to "estimate" where in the stream the processing has gotten. It starts off wildly inaccurate, but after about 5 seconds of processed-video ffprobe and ffmpeg are producing data at a similar pace.

  



  This is a hack, but a step towards the granularity I want. It may be that I need an mpegts parser / chunker that can run on the ffmpeg output directly in NodeJS.

  



  The output of the above is something along the lines of :

  



  undefined (repeated around 100x as I assume ffprobe needs more data to start)
FFPROBE: 1.422456 // Note that this represents around 60 ffprobe messages sent at once.
FFPROBE: 1.933867
1.933867 // These lines represent mpegts data sent from ffmpeg, and the latest pts reported by ffprobe
FFPROBE: 2.388989
2.388989
FFPROBE: 2.728578
FFPROBE: 2.989811
FFPROBE: 3.146544
3.146544
FFPROBE: 3.433889
FFPROBE: 3.668989
FFPROBE: 3.802400
FFPROBE: 3.956333
FFPROBE: 4.069333
4.069333
FFPROBE: 4.426544
FFPROBE: 4.609400
FFPROBE: 4.870622
FFPROBE: 5.184089
FFPROBE: 5.337267
5.337267
FFPROBE: 5.915522
FFPROBE: 6.104700
FFPROBE: 6.333478
6.333478
FFPROBE: 6.571833
FFPROBE: 6.705300
6.705300
FFPROBE: 6.738667
6.738667
FFPROBE: 6.777567
FFPROBE: 6.772033
6.772033
FFPROBE: 6.805400
6.805400
FFPROBE: 6.829811
FFPROBE: 6.838767
6.838767
FFPROBE: 6.872133
6.872133
FFPROBE: 6.882056
FFPROBE: 6.905500
6.905500


  


• Write wave audio samples to vorbis by FFMPEG

  3 avril 2014, par Ankush

  I am recording screen, mic & speaker to webm video. Video part is working fine but when I add audio samples recorded from mic to video, I can't hear sound in generated video. I am adding 3 screenshots per sec. & After each screenshot I add audio buffer gathered by NAudio. Any clues what I am doing wrong ?

  c# part :

  static void wmic_DataAvailable(object sender, NAudio.Wave.WaveInEventArgs e)
  
          {
  
              if(e.BytesRecorded>0)
  
                  vencoder.AddAudioSampleBuffer(e.Buffer, e.BytesRecorded);
  
          }

  c++ part :

   AVStream * VideoEncoder::AddAudioStream(AVFormatContext *pContext, AVCodecID codec_id)
  
  {
  
    AVCodecContext *pCodecCxt = NULL;
  
    AVStream *pStream = NULL;
  
  
  
    // Try create stream.
  
    pStream = av_new_stream(pContext, 1);
  
    if (!pStream) 
  
    {
  
      printf("Cannot add new audio stream\n");
  
      return NULL;
  
    }
  
  
  
    // Codec.
  
    pCodecCxt = pStream->codec;
  
    pCodecCxt->codec_id = codec_id;
  
    pCodecCxt->codec_type = AVMEDIA_TYPE_AUDIO;
  
    // Set format
  
    pCodecCxt->bit_rate    = 128000;
  
    pCodecCxt->sample_rate = 44100;
  
    pCodecCxt->channels    = 1;
  
    pCodecCxt->sample_fmt  = AV_SAMPLE_FMT_S16P;//AV_SAMPLE_FMT_S16; //libvorbis supports only AV_SAMPLE_FMT_FLTP
  
  
  
    nSizeAudioEncodeBuffer = 4 * MAX_AUDIO_PACKET_SIZE;
  
    if (pAudioEncodeBuffer == NULL)
  
    {      
  
      pAudioEncodeBuffer = (uint8_t * )av_malloc(nSizeAudioEncodeBuffer);
  
    }
  
  
  
    // Some formats want stream headers to be separate.
  
    if(pContext->oformat->flags & AVFMT_GLOBALHEADER)
  
    {
  
      pCodecCxt->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
    }
  
  
  
    return pStream;
  
  }
  
  
  
  bool VideoEncoder::OpenAudio(AVFormatContext *pContext, AVStream *pStream)
  
  {
  
    AVCodecContext *pCodecCxt = NULL;
  
    AVCodec *pCodec = NULL;
  
    pCodecCxt = pStream->codec;
  
  
  
    // Find the audio encoder.
  
    pCodec = avcodec_find_encoder(pCodecCxt->codec_id);
  
    if (!pCodec) 
  
    {
  
      printf("Cannot open audio codec\n");
  
      return false;
  
    }
  
  
  
    // Open it.
  
    if (avcodec_open2(pCodecCxt, pCodec,NULL) < 0) 
  
    {
  
      printf("Cannot open audio codec\n");
  
      return false;
  
    }
  
  
  
    if (pCodecCxt->frame_size <= 1) 
  
    {
  
      // Ugly hack for PCM codecs (will be removed ASAP with new PCM
  
      // support to compute the input frame size in samples. 
  
      audioInputSampleSize = nSizeAudioEncodeBuffer / pCodecCxt->channels;
  
      switch (pStream->codec->codec_id) 
  
      {
  
        case CODEC_ID_PCM_S16LE:
  
        case CODEC_ID_PCM_S16BE:
  
        case CODEC_ID_PCM_U16LE:
  
        case CODEC_ID_PCM_U16BE:
  
          audioInputSampleSize >>= 1;
  
          break;
  
        default:
  
          break;
  
      }
  
      pCodecCxt->frame_size = audioInputSampleSize;
  
    } 
  
    else 
  
    {
  
      audioInputSampleSize = pCodecCxt->frame_size;
  
    }
  
  
  
    return true;
  
  }
  
  
  
  bool VideoEncoder::AddAudioSample(AVFormatContext *pFormatContext, AVStream *pStream, 
  
                                          const char* soundBuffer, int soundBufferSize)
  
  {
  
    AVCodecContext *pCodecCxt;    
  
    bool res = true;  
  
  
  
    pCodecCxt       = pStream->codec;
  
    memcpy(audioBuffer + nAudioBufferSizeCurrent, soundBuffer, soundBufferSize);
  
    nAudioBufferSizeCurrent += soundBufferSize;
  
  
  
    BYTE * pSoundBuffer = (BYTE *)audioBuffer;
  
    int nCurrentSize    = nAudioBufferSizeCurrent;
  
  
  
    // Size of packet on bytes.
  
    // FORMAT s16
  
    DWORD packSizeInSize = 2 * audioInputSampleSize;
  
  
  
    while(nCurrentSize >= packSizeInSize)
  
    {
  
      AVPacket pkt;
  
      av_init_packet(&pkt);
  
  
  
      pkt.size = avcodec_encode_audio(pCodecCxt, pAudioEncodeBuffer, 
  
        nSizeAudioEncodeBuffer, (const short *)pSoundBuffer);
  
  
  
      if (pCodecCxt->coded_frame && pCodecCxt->coded_frame->pts != AV_NOPTS_VALUE)
  
      {
  
        pkt.pts = av_rescale_q(pCodecCxt->coded_frame->pts, pCodecCxt->time_base, pStream->time_base);
  
      }
  
  
  
      pkt.flags |= AV_PKT_FLAG_KEY;
  
      pkt.stream_index = pStream->index;
  
      pkt.data = pAudioEncodeBuffer;
  
  
  
      // Write the compressed frame in the media file.
  
      if (av_interleaved_write_frame(pFormatContext, &pkt) != 0) 
  
      {
  
        res = false;
  
        break;
  
      }
  
  
  
      nCurrentSize -= packSizeInSize;  
  
      pSoundBuffer += packSizeInSize;      
  
    }
  
  
  
    // save excess
  
    memcpy(audioBuffer, audioBuffer + nAudioBufferSizeCurrent - nCurrentSize, nCurrentSize);
  
    nAudioBufferSizeCurrent = nCurrentSize; 
  
  
  
    return res;
  
  }
  
  
  
  char* ByteArray_to_charptr(array^ byteArray)
  
  {
  
      pin_ptr p = &byteArray[0];
  
      unsigned char* pby = p;
  
      char* pch = reinterpret_cast(pby);
  
      return pch;
  
      // use it...
  
  }
  
  
  
  void VideoEncoder::AddAudioSampleBuffer(array^ byteArray,int^ buffersize)
  
  {
  
      char * buffer=ByteArray_to_charptr(byteArray);
  
  
  
      if (buffer && (int)buffersize > 0)
  
      {
  
          AddAudioSample(pFormatContext,pVideoStream,buffer,(int)buffersize);
  
      }
  
  }

  Q2. How can I mix mic & speaker samples realtime so that I hear both in one AudioStream ?