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• ffmpeg error opening filters at mobile browser

  1er novembre 2017, par lzmhhh123

  I use the ffmpeg at mobile browser. But it responds Error opening filter both at safari and chrome. I need help. The log are as following.

  The IOS 11.0.2 Safari logs :

  [Log] ffmpeg version n3.1.2 Copyright (c) 2000-2016 the FFmpeg developers (bundle.js, line 43469)
  
    built with emcc (Emscripten gcc/clang-like replacement) 1.36.7 ()
  
    configuration: --cc=emcc --enable-cross-compile --target-os=none --arch=x86 --disable-runtime-cpudetect --disable-asm --disable-fast-unaligned --disable-pthreads --disable-w32threads --disable-os2threads --disable-debug --disable-stripping --disable-all --enable-ffmpeg --enable-avcodec --enable-avformat --enable-avutil --enable-swresample --enable-swscale --enable-avfilter --disable-network --disable-d3d11va --disable-dxva2 --disable-vaapi --disable-vda --disable-vdpau --enable-decoder=vp8 --enable-decoder=vp9 --enable-decoder=theora --enable-decoder=mpeg2video --enable-decoder=mpeg4 --enable-decoder=h264 --enable-decoder=hevc --enable-decoder=png --enable-decoder=mjpeg --enable-decoder=vorbis --enable-decoder=opus --enable-decoder=mp3 --enable-decoder=ac3 --enable-decoder=aac --enable-decoder=ass --enable-decoder=ssa --enable-decoder=srt --enable-decoder=webvtt --enable-demuxer=matroska --enable-demuxer=ogg --enable-demuxer=avi --enable-demuxer=mov --enable-demuxer=flv --enable-demuxer=mpegps --enable-demuxer=image2 --enable-demuxer=mp3 --enable-demuxer=concat --enable-protocol=file --enable-filter=aresample --enable-filter=scale --enable-filter=crop --enable-filter=overlay --disable-bzlib --disable-iconv --disable-libxcb --disable-lzma --disable-sdl --disable-securetransport --disable-xlib --disable-zlib --enable-encoder=libx264 --enable-encoder=libmp3lame --enable-encoder=aac --enable-muxer=mp4 --enable-muxer=mp3 --enable-muxer=null --enable-gpl --enable-libmp3lame --enable-libx264 --extra-cflags=-I../lame/dist/include --extra-ldflags=-L../lame/dist/lib
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 48.101 / 57. 48.101
  
    libavformat    57. 41.100 / 57. 41.100
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
  [h264 @ 0x809df0] Warning: not compiled with thread support, using thread emulation
  
  [aac @ 0x80e380] Warning: not compiled with thread support, using thread emulation
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '53119917068__7A6AFE7C-86EC-42F4-8995-A9B09B16C2DB.MOV':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2017-11-01 03:19:31
  
      com.apple.quicktime.make: Apple
  
      com.apple.quicktime.model: iPhone 6
  
      com.apple.quicktime.software: 11.0.2
  
      com.apple.quicktime.creationdate: 2017-11-01T11:19:31+0800
  
    Duration: 00:00:00.20, start: 0.000000, bitrate: 1467 kb/s
  
      Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(tv, smpte170m/bt709/bt709), 480x360, 1131 kb/s, 30 fps, 30 tbr, 600 tbn, 1200 tbc (default)
  
      Metadata:
  
        rotate          : 90
  
        creation_time   : 2017-11-01 03:19:31
  
        handler_name    : Core Media Data Handler
  
        encoder         : H.264
  
      Side data:
  
        displaymatrix: rotation of -90.00 degrees
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 80 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-01 03:19:31
  
        handler_name    : Core Media Data Handler
  
      Stream #0:2(und): Data: none (mebx / 0x7862656D), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-01 03:19:31
  
        handler_name    : Core Media Data Handler
  
      Stream #0:3(und): Data: none (mebx / 0x7862656D), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-01 03:19:31
  
        handler_name    : Core Media Data Handler
  
  Error opening filters!

  The Android 7.0 Chrome logs :

  ffmpeg version n3.1.2 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with emcc (Emscripten gcc/clang-like replacement) 1.36.7 ()
  
    configuration: --cc=emcc --enable-cross-compile --target-os=none --arch=x86 --disable-runtime-cpudetect --disable-asm --disable-fast-unaligned --disable-pthreads --disable-w32threads --disable-os2threads --disable-debug --disable-stripping --disable-all --enable-ffmpeg --enable-avcodec --enable-avformat --enable-avutil --enable-swresample --enable-swscale --enable-avfilter --disable-network --disable-d3d11va --disable-dxva2 --disable-vaapi --disable-vda --disable-vdpau --enable-decoder=vp8 --enable-decoder=vp9 --enable-decoder=theora --enable-decoder=mpeg2video --enable-decoder=mpeg4 --enable-decoder=h264 --enable-decoder=hevc --enable-decoder=png --enable-decoder=mjpeg --enable-decoder=vorbis --enable-decoder=opus --enable-decoder=mp3 --enable-decoder=ac3 --enable-decoder=aac --enable-decoder=ass --enable-decoder=ssa --enable-decoder=srt --enable-decoder=webvtt --enable-demuxer=matroska --enable-demuxer=ogg --enable-demuxer=avi --enable-demuxer=mov --enable-demuxer=flv --enable-demuxer=mpegps --enable-demuxer=image2 --enable-demuxer=mp3 --enable-demuxer=concat --enable-protocol=file --enable-filter=aresample --enable-filter=scale --enable-filter=crop --enable-filter=overlay --disable-bzlib --disable-iconv --disable-libxcb --disable-lzma --disable-sdl --disable-securetransport --disable-xlib --disable-zlib --enable-encoder=libx264 --enable-encoder=libmp3lame --enable-encoder=aac --enable-muxer=mp4 --enable-muxer=mp3 --enable-muxer=null --enable-gpl --enable-libmp3lame --enable-libx264 --extra-cflags=-I../lame/dist/include --extra-ldflags=-L../lame/dist/lib
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 48.101 / 57. 48.101
  
    libavformat    57. 41.100 / 57. 41.100
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
  [hevc @ 0x809db0] Warning: not compiled with thread support, using thread emulation
  
  [aac @ 0x80ea90] Warning: not compiled with thread support, using thread emulation
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'VID_20171101_113436.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: isommp42
  
      creation_time   : 2017-11-01 03:34:40
  
      com.android.version: 7.0
  
    Duration: 00:00:02.69, start: 0.000000, bitrate: 32551 kb/s
  
      Stream #0:0(eng): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv), 3840x2160, 29756 kb/s, SAR 1:1 DAR 16:9, 29.60 fps, 29.83 tbr, 90k tbn, 90k tbc (default)
  
      Metadata:
  
        rotate          : 90
  
        creation_time   : 2017-11-01 03:34:40
  
        handler_name    : VideoHandle
  
      Side data:
  
        displaymatrix: rotation of -90.00 degrees
  
      Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-11-01 03:34:40
  
        handler_name    : SoundHandle
  
  Error opening filters!

• FFmpeg HLS Live Stream Stuck with iOS 11

  1er novembre 2017, par Lamorun

  I’ve exp with FFmpeg but i recent problem with iOS 11 (Recent release for Apple)

  I used FFmpeg for stream HLS Live Stream for only iPhone and iPad in closed app for online courses. My problem is the next :

  I used this code for HLS Live Stream :

  ffmpeg -i "Input" -preset fast -c:v libx264 -c:a aac -ac 1 -strict -2 -crf 18 -profile:v baseline -maxrate 1000k -bufsize 1835k -pix_fmt yuv420p -b:a 64k -flags -global_header -hls_time 20 -hls_list_size 6 -hls_wrap 10 -start_number 1 /m3u8/test.m3u8

  This work perfect in iOS 8, 9 and 10 devices. But with new version iOS 11, video is stuck after about 30 to 40 seconds, as seen in the image :

  ios11stuck

  I have searched for hours and hours. I have done many tests, but I can not get it out, since it works perfectly with iOS 8,9 and 10 (test in somes devices, even working perfectly on a device with iOS 10, update to iOS 11, and stop working immediately, downgrade to iOS 10 and work again)

  I hope someone can help me out, thank you and greetings to all

• Windows(WASAPI):Converting IEEE float PCM to PCM 16bit

  10 octobre 2017, par Michael IV

  I do a hook into some of WASAPI methods to read out audio buffer that contains system sound.

  The source is WAV with audio format = WAVE_FORMAT_IEEE_FLOAT and

  subformat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT

  I am trying to convert it to PCM 16bit on the fly using FFMPEG libswresample. The resulting soundtrack has alot of forground noise with the original track’s data playing in the background.

  Here is how I do it (I skip WASAPI init part) :

  First,the source WAVEFORMATEX struct has the following properties :

  Format tag:65534
  
  Sample rate:44100
  
  Num channels:2
  
  CB size:22
  
  Average bytes per sec:352800
  
  Block align:8
  
  Bits per sample:32

  Where 65534 stands for WAVE_FORMAT_IEEE_FLOAT.

   WAVEFORMATEX *w;//filled with the props above
  
   int samples = 0;  
  
   int bufferSize = 0;
  
   int buffer_frames = 0; 
  
   int bWritten = 0;
  
   char* bufferIn = NULL;
  
   char* bufferOut = NULL;
  
   BYTE* wavBuffer = NULL;
  
   void InitSWRContext()
  
   {
  
      swrctx = swr_alloc_set_opts(NULL,
  
      AV_CH_LAYOUT_STEREO, //stereo layout
  
      AV_SAMPLE_FMT_S16, // sample out format
  
      44100, // out sample rate
  
      2, //number of channels 
  
      AV_SAMPLE_FMT_FLT, //sample in format
  
      w->nSamplesPerSec ,//in sample rate
  
      0,
  
     NULL);
  
  
  
      swr_init(swrctx);
  
  
  
   //also tried like this:
  
    /*
  
     swrctx = swr_alloc();
  
     av_opt_set_int(swrctx, "in_channel_layout", CA2SWR_chlayout(w->nChannels), 0);
  
     av_opt_set_int(swrctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
     av_opt_set_int(swrctx, "in_sample_rate", 44100, 0);
  
     av_opt_set_int(swrctx, "out_sample_rate", 44100, 0);
  
     av_opt_set_sample_fmt(swrctx, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
     av_opt_set_sample_fmt(swrctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
  
  
  
     */
  
  
  
      samples = (int)av_rescale_rnd(CA_MAX_SAMPLES, 44100, w->nSamplesPerSec, 
  
      AV_ROUND_UP);
  
      bufferSize = av_samples_get_buffer_size(NULL,2, samples * 2,AV_SAMPLE_FMT_S16, 1/*no-alignment*/);
  
      bufferOut = ( char *)malloc(bufferSize ))
  
  
  
   }

  These two methods are invoked by the system and I hook them to access audio buffer :

    DEXPORT HRESULT __stdcall
  
  _GetBuffer(
  
  IAudioRenderClient *thiz,
  
  UINT32 NumFramesRequested,
  
  BYTE **ppData)
  
  {
  
      HRESULT hr;
  
      hr = orig_GetBuffer(thiz, NumFramesRequested, ppData);
  
      bufferIn = (char*)*ppData;
  
      buffer_frames = NumFramesRequested;
  
      return S_OK;
  
  }

  Then on buffer release I perform the conversion and readout.

   DEXPORT HRESULT __stdcall
  
  _ReleaseBuffer(IAudioRenderClient *thiz,
  
  UINT32 NumFramesWritten, DWORD dwFlags)
  
  {
  
      const unsigned char *srcplanes[2];
  
      unsigned char *dstplanes[2];
  
      int samples;
  
      srcplanes[0] = (unsigned char*)bufferIn;
  
      srcplanes[1] = NULL;
  
      dstplanes[0] = (unsigned char*)bufferOut; 
  
      dstplanes[1] = NULL;
  
      samples = (int)av_rescale_rnd(NumFramesWritten, 44100, 44100, AV_ROUND_UP);
  
      int samplesConverted = swr_convert(swrctx, dstplanes, samples , srcplanes, NumFramesWritten);
  
      int framesize = (/*PCM16*/16 / 8) * pwfx->nChannels * NumFramesWritten;
  
      if (!wavBuffer)
  
      {
  
        wavBuffer = (BYTE*)malloc(4096 * 1000); //write out a few seconds
  
      }
  
      //copy converted buffer into wavBuffer
  
      memcpy(wavBuffer + bWritten, bufferOut, framesize);
  
      bWritten += framesize;
  
      if (bWritten > 4096 * 1000)
  
      {
  
          pwfx->wFormatTag = WAVE_FORMAT_PCM;
  
          pwfx->wBitsPerSample = 16;
  
          pwfx->nBlockAlign = (pwfx->nChannels * pwfx->wBitsPerSample) / 8;
  
          pwfx->cbSize = 0;
  
          SaveWaveData(wavBuffer, bWritten, pwfx, L"test.wav");
  
  
  
          exit(0);
  
       }
  
  
  
    }

  SaveWaveData borrowed from here.

  Resulting sound file(WAV)

  Original file : a2002011001-e02.wav

  I tried to play and analyze the output in VLC,Audacity, and FMOD Studio.
  The strange thing is that VLC show (codec info) that it is PCM16, while FMOD studio interprets the data as PCM32. I also tried to store without conversion the original buffer which also produces a sound with noise,though not as significant as when converting to PCM16