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• IframeExtractor don't output sound with rtsp

  9 janvier 2013, par Kamax

  I use IframeExtractor from the git mooncatventure, it play nice the .mov file.
  But when i try to read a rtsp stream, i hear no sound.

  This is the FFMEG dump from the rtsp stream :

  Metadata:
  
  title           : unknown
  
  comment         : unknown
  
  Duration: N/A, start: 49435.000589, bitrate: 258 kb/s
  
  Program 3223 
  
  No Program
  
  Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
  Stream #0:1(fra): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 142 kb/s
  
  Stream #0:2(fra): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006)
  
  Stream #0:3(qad): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, mono, fltp, 47 kb/s
  
  Stream #0:4(qaa): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 68 kb/s

  And this is the dump from the local .mov file that work :

  Metadata:
  
  major_brand     : qt  
  
  minor_version   : 0
  
  compatible_brands: qt  
  
  creation_time   : 2010-01-17 21:52:33
  
  model           : iPhone 3GS
  
  model-eng       : iPhone 3GS
  
  date            : 2010-01-17T16:52:33-0500
  
  date-eng        : 2010-01-17T16:52:33-0500
  
  encoder         : 3.1.2
  
  encoder-eng     : 3.1.2
  
  make            : Apple
  
  make-eng        : Apple
  
  Duration: 00:00:03.25, start: 0.000000, bitrate: 3836 kb/s
  
  Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 640x480, 3695 kb/s, 30.02 fps, 30 tbr, 600 tbn, 1200 tbc
  
  Metadata:
  
    rotate          : 90
  
    creation_time   : 2010-01-17 21:52:33
  
    handler_name    : Core Media Data Handler
  
  Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 63 kb/s
  
  Metadata:
  
    creation_time   : 2010-01-17 21:52:33
  
    handler_name    : Core Media Data Handler

  The audio class that manage sounds contain a codec detector which say that the codec CODEC_ID_AAC is found for the two input :

  audioStreamBasicDesc_.mFormatFlags = 0;
  
  switch (_audioCodecContext->codec_id) {
  
      case CODEC_ID_MP3:
  
           audioStreamBasicDesc_.mFormatID = kAudioFormatMPEGLayer3;
  
          break;
  
      case CODEC_ID_AAC:
  
           audioStreamBasicDesc_.mFormatID = kAudioFormatMPEG4AAC;
  
           audioStreamBasicDesc_.mFormatFlags = kMPEG4Object_AAC_Main;
  
          NSLog(@"audio format aac %s (%d) is  supported",  _audioCodecContext->codec_name, _audioCodecContext->codec_id);
  
          break;
  
  }

  I see data going into the buffer but i hear nothing. It's maybe audioStreamBasicDesc_ which has wrong settings but i can't find what.

  Is it possible that it's not the same AAC codec ?

  Has someone experienced the same issue ?

  Any help are welcome, i'm on this problem since some days now.

  Edit :
  I have found a error that i had not before, i don't know how to resolve it. If i change audioStreamBasicDesc.mFramesPerPacket to 0 or divided by 2, the error message dissapear.

  AudioConverterNew returned 'fmt?'
  
  Prime failed ('fmt?'); will stop (72000/0 frames)

• ffmpeg settings or alternatives to ffmpeg on raspberry pi for video streaming

  14 octobre 2016, par andrei

  I have a Raspberry Pi (model B) running raspbian wheezy on a 16gb SD card. I also have a 32gb flash storage attached on the usb. I’m trying to stream a video (h264 encoded mp4 file 1280x720) over the ethernet from that flash storage.
  I’m using ffmpeg+ffserver. Here is ffserver.conf (relevant parts) :

  ...
  
  MaxBandwidth 10000
  
  <feed>
  
  ...
  
  FileMaxSize 100M
  
  ACL allow 127.0.0.1
  
  </feed>
  
  ...
  
  <stream>
  
  Feed feed1.ffm
  
  Format flv
  
  VideoSize 288x176 #made small just for testing
  
  NoAudio
  
  </stream>
  
  ....

  I start the ffserver, then call ffmpeg with this command :

  ffmpeg -re -an -i /mnt/u32/main.mp4 -r 25 -bit_rate 300k http://localhost:8090/feed1.ffm

  And I’m getting fps 3-5 at most. Naturally when I try to view the stream on another computer it’s very choppy and virtually unusable.

  Am I missing some settings ? Or perhaps there is another streaming solution that leverages the GPU instead of just the CPU as ffmpeg does ? I’m even open to suggestions about other boards (e.g. a pandaboard ? or clustering several RPi’s ?) Also, I’m flexible about the output format.

• Is there a Heroku-supported method for getting the length of an audio file in seconds ?

  12 juillet 2013, par Ben West

  I'm trying to implement

  http://blog.firmhouse.com/validate-length-of-an-audio-file-when-using-paperclip-and-s3

  But I'm unsure what gem provided FFMpegWrapper, and I'd like to use Heroku to deploy my app for testing. Does heroku support something that will work for this ?

  full Rails Application source :

  https://github.com/mrgenixus/sound-byte

  Relevant Model :

  https://github.com/mrgenixus/sound-byte/blob/master/app/models/episode.rb