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• Why do i get avc1.000000 ? (FFMPEG, H.264 Video Encoding, C++)

  17 février 2020, par elastoss

  I have a bunch of bitmaps, and need to make them encoded in h.264 in fragmented .mp4.
  I’m using C++.
  What could cause that my AVC Profile is set to 0, SPS[], PPS[] and codec string to avc1.000000 ?

  Output from the mp4info :

  File :

  minor version : 200
  compatible brand : iso6
  compatible brand : mp41
  fast start : yes

  Movie :

  duration : 0 ms
  time scale : 1000
  fragments : yes

  Found 1 Tracks

  Track 1 :
  flags : 3 ENABLED IN-MOVIE
  id : 1
  type : Video
  duration : 0 ms
  language : und

  media :

  sample count : 0
  timescale : 90000
  duration : 0 (media timescale units)
  duration : 0 (ms)
  bitrate (computed) : 412.672 Kbps
  sample count with fragments : 35
  duration with fragments : 540000
  duration with fragments : 6000 (ms)

  display width : 1280.000000
  display height : 720.000000

  Sample Description 0

  Coding : avc1 (H.264)
  Width:n 1280
  Height : 720
  Depth : 24
  AVC Profile : 0
  AVC Profile Compat : 0
  AVC Level : 0
  AVC NALU Length Size : 0
  AVC SPS : []
  AVC PPS : []
  Codecs String : avc1.000000

  I’m using things like

  if (stream->codecpar->codec_id == AVCodecID.AV_CODEC_ID_H264)
  
  {
  
      err = ffmpeg.av_opt_set(cctx->priv_data, "preset", "ultrafast", 0);
  
      err = ffmpeg.av_opt_set(cctx->priv_data, "tune", "zerolatency", 0);
  
      err = ffmpeg.av_opt_set(cctx->priv_data, "profile", "high", 0);
  
  } 
  
  
  
  ...
  
  
  
  AVDictionary* opts = null;
  
  ffmpeg.av_dict_set(&opts, "movflags", "default_base_moof+frag_keyframe+empty_moov", 0); 
  
  
  
  ...
  
  
  
  AVPacket* pPacket = ffmpeg.av_packet_alloc();
  
  try
  
  {
  
      int error;
  
      do
  
      {
  
          ffmpeg.avcodec_send_frame(cctx, &convertedFrame).ThrowExceptionIfError();
  
  
  
          error = ffmpeg.avcodec_receive_packet(cctx, pPacket);
  
      } while (error == ffmpeg.AVERROR(ffmpeg.EAGAIN));
  
  
  
      error.ThrowExceptionIfError();
  
  
  
  }
  
  finally
  
  {
  
      ffmpeg.av_packet_rescale_ts(pPacket, cctx->time_base, stream->time_base);
  
      pPacket->stream_index = stream->index;
  
  
  
      ffmpeg.av_interleaved_write_frame(ofctx, pPacket);
  
      ffmpeg.av_packet_unref(pPacket);
  
  }

  What am I missing ? I’m using examples from internet. Thought that if AVFrame is encoded (send_frame) with H264 with profiles and presets and received as AVPacket. It should be done automatically.

  This is my first post, please be nice. Thanks in advance for helping.

• Receiving RTP stream - AudioStream, AudioGroup

  21 février 2020, par totten

  I would like to listen an RTP audio stream, however the voice has little gaps in it - not continues. What may be the solution ? Am I missing something on Receiver(android) side or Streamer(ffmpeg) side ?

  I’m using ffmpeg to stream RTP audio,

  ffmpeg -f lavfi -i aevalsrc="sin(400*2*PI*t)" -ar 8000 -vcodec pcm_u8 -f rtp rtp://192.168.0.15:41954 (port changes.)

  And here is my related android code :

  AudioStream audioStream;
  
  AudioGroup audioGroup;
  
  @Override
  
  public void onStart() {
  
      super.onStart();
  
      StrictMode.ThreadPolicy policy = new StrictMode.ThreadPolicy.Builder().permitNetwork().build();
  
      StrictMode.setThreadPolicy(policy);
  
      AudioManager audio = (AudioManager)getSystemService(AUDIO_SERVICE);
  
      audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
  
      audioGroup = new AudioGroup();
  
      audioGroup.setMode(AudioGroup.MODE_ECHO_SUPPRESSION);
  
      InetAddress inetAddress;
  
      try {
  
          inetAddress = InetAddress.getByName("192.168.0.15");
  
          audioStream = new AudioStream(inetAddress);
  
          audioStream.setCodec(AudioCodec.PCMU);
  
          audioStream.setMode(RtpStream.MODE_NORMAL);
  
          InetAddress inetAddressRemote = InetAddress.getByName("192.168.0.14");
  
          audioStream.associate(inetAddressRemote, 6000);
  
          ((TextView)findViewById(R.id.tv_port)).setText("Port : " + String.valueOf(audioStream.getLocalPort()));
  
          audioStream.join(audioGroup);
  
      }
  
      catch ( UnknownHostException e ) {
  
          e.printStackTrace();
  
      }
  
      catch ( SocketException e ) {
  
          e.printStackTrace();
  
      }
  
  }

• ffmpeg processing the live screen streaming very slowly

  27 février 2020, par Ant's

  I’m trying to do live screen sharing using ffmpeg using the following command :

  ffmpeg -f avfoundation -i "1:1" -c:v libx264 -threads 4 -preset ultrafast -c:a aac -ar 44100 -f flv rtmp://localhost/live/test

  Now I have a rtmp server, which receives the data and using flv.js, I’m showing the live stream video on the browser. The integration works perfectly fine, but the problem was the stream is getting delayed very much. There is a delay for say atleast 10s ; I’m not sure, whether we can make it less delay (more like instant screen share).

  Note : I’m using the Node RTMP server using https://github.com/illuspas/Node-Media-Server. The code for that is over here :

  const NodeMediaServer = require('node-media-server');
  
  
  
  const config = {
  
    rtmp: {
  
      port: 1935,
  
      chunk_size: 6000,
  
      gop_cache: true,
  
      ping: 30,
  
      ping_timeout: 60
  
    },
  
    http: {
  
      port: 8000,
  
      allow_origin: '*'
  
    }
  
  };
  
  
  
  var nms = new NodeMediaServer(config)
  
  nms.run();

  Any suggestions ? I’m on MacOS