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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)

• Can not play local rtsp URL using FFmpeg in iOS

  12 juin 2015, par Priyanka

  I am using FFmpeg for stream rtsp URL in iOS.
  I am trying to stream a local url but my app is failed to open url
  avformat_open_input method always return -5

  I have played the same url rtsp://172.16.1.226:5544/1 on VLC media player on my iPhone and macbook it works on both.

  After few research i have found there is some problem with rtsp_transport

  I was using av_dict_set(&serverOpt, "rtsp_transport", "tcp", 0); for the server configuration while opening url and the result is can not open feed.

  When I changed it to av_dict_set(&serverOpt, "rtsp_transport", "udp", 1);
  I am able to open url successfully but I continuously getting error rtsp 1 missing packet and so on.

  Can anybody help what should be the right configuration while opening a local rtsp url using ffmpeg.
  Should i need to update av_dict_set(&serverOpt, "rtsp_transport", "udp", 1)

  Thanks in advance

• C - FFmpeg streaming from a C program ?

  16 août 2017, par golmschenk

  I’m looking to replicate an FFmpeg command-line command in my C code. Specifically I would like to be able to run :

  ffmpeg -re -i video.mp4 -f mpegts udp://localhost:7777

  One thing I’ve noticed when looking at people’s code who have used the libraries of FFmpeg in their own code is that they often have a few hundred lines of code for a single command similar to an FFmpeg command-line command. I’m guessing this is just because they are doing something very specific, because if I can run that short command on my command line and get what I want it should probably only take about ten lines of code to do the same thing in my C code. This should only take about that much work right ? Why would it take much more ?

  I’m having a bit of difficulty finding explanations on how to use the streaming capabilities of the FFmpeg libraries that aren’t overly complex because they’re for a very specific purpose. Can anyone explain how I might go about writing the code for the above command ? Or at the very least point me to some documentation explaining how to write such a script/program ? Thank you much !

  EDIT : I do hope to run this from an iPhone app eventually so I won’t just be able to straight up call FFmpeg from my program. I’ll need to use the libraries used by FFmpeg.