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• Live streaming .webm file with ffmpeg and nginx

  9 décembre 2017, par seekahead

  Hi I am new to ffmpeg and I need some help with live streaming of .webm file which is still being written into. I have a .webm file containing only video in vp9 codecwhich needs to be live streamed (preferably without transcoding and streamed using DASH`. I have compiled nginx along with nginx-ts-module from (https://github.com/arut/nginx-ts-module#features) with the following configuration

  nginx version: nginx/1.12.1
  
  built by gcc 4.8.4 (Ubuntu 4.8.4-2ubuntu1~14.04.3) 
  
  built with OpenSSL 1.0.1f 6 Jan 2014
  
  TLS SNI support enabled
  
  configure arguments: --with-cc-opt='-g -O2 -fPIE -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -fPIC -D_FORTIFY_SOURCE=2' --with-ld-opt='-Wl,-Bsymbolic-functions -fPIE -pie -Wl,-z,relro -Wl,-z,now -fPIC' --prefix=/usr/share/nginx --conf-path=/etc/nginx/nginx.conf --http-log-path=/var/log/nginx/access.log --error-log-path=/var/log/nginx/error.log --lock-path=/var/lock/nginx.lock --pid-path=/run/nginx.pid --modules-path=/usr/lib/nginx/modules --http-client-body-temp-path=/var/lib/nginx/body --http-fastcgi-temp-path=/var/lib/nginx/fastcgi --http-proxy-temp-path=/var/lib/nginx/proxy --http-scgi-temp-path=/var/lib/nginx/scgi --http-uwsgi-temp-path=/var/lib/nginx/uwsgi --with-debug --with-pcre-jit --with-http_ssl_module --with-http_stub_status_module --with-http_realip_module --with-http_auth_request_module --with-http_v2_module --with-http_dav_module --with-http_slice_module --with-threads --with-http_addition_module --with-http_gunzip_module --with-http_gzip_static_module --with-http_sub_module --with-stream=dynamic --with-stream_ssl_module --with-stream_ssl_preread_module --with-mail=dynamic --with-mail_ssl_module --add-dynamic-module=/home/ubuntu/NGINX/nginx-ts-module-master

  The mediafile has the following properties(obtained with mediainfo) :

  General
  
  Complete name                            : video_4_1_Admin_0.webm
  
  Format                                   : WebM
  
  Format version                           : Version 2
  
  File size                                : 19.7 MiB
  
  Duration                                 : 1ms
  
  Overall bit rate                         : 165 Gbps
  
  Writing library                          : libwebm-0.2.1.0
  
  Video
  
  ID                                       : 1
  
  Format                                   : VP9
  
  Codec ID                                 : V_VP9
  
  Width                                    : 320 pixels
  
  Height                                   : 180 pixels
  
  Display aspect ratio                     : 16:9
  
  Frame rate mode                          : Variable
  
  Language                                 : English
  
  Default                                  : Yes
  
  Forced                                   : No

  I am trying to stream the file with the following

  ffmpeg -re -i video_4_1_Admin_0.webm -bsf:v vp9_superframe -c copy -f webm_chunk  http://127.0.0.1:8000/publish/sintel

  And I am getting the following error

  Input #0, matroska,webm, from 'video_4_1_Admin_0.webm':
  
    Metadata:
  
      encoder         : libwebm-0.2.1.0
  
    Duration: 00:00:00.00, start: 0.000000, bitrate: 174444152 kb/s
  
      Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv), 320x180, SAR 1:1 DAR 16:9, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Output #0, webm_chunk, to 'http://127.0.0.1:8000/publish/sintel':
  
  Output file #0 does not contain any stream

  I am not sure what are the exact options to be passed to ffmpeg so that it creates the .mpd and needed chunks which will allow me to LiveStream the media, I tried replacing the options given in the example (https://github.com/arut/nginx-ts-module#features) with what seems like vp9 equivalent., but, I am not sure. Can someone help me out here ? Thanks in advance

• 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

• Merge commit '883ce264d9ffc5bdaf477e09ee155b03339c46a6'

  23 octobre 2017, par James Almer

  Merge commit '883ce264d9ffc5bdaf477e09ee155b03339c46a6'
  * commit '883ce264d9ffc5bdaf477e09ee155b03339c46a6' :
  
    vf_showinfo : Display spherical properties
  Merged-by : James Almer <jamrial@gmail.com>
  

  • [DH] libavfilter/vf_showinfo.c