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• Revision d2059b0d2d : Skip mode check when mv has been tested This commit allows the non-RD mode deci

  7 mars 2014, par Jingning Han

  Changed Paths :
   Modify /vp9/encoder/vp9_pickmode.c

  
  
  Skip mode check when mv has been tested

  This commit allows the non-RD mode decision to skip mode RD modelling
  check, if the motion vector associated with the current mode is
  same as that of NEARESTMV mode. This makes speed -7 about 2% faster.
  Previous change that converts cost metric from SAD to model based RD
  value makes the codec 6% slower at speed -7.

  Change-Id : I30cfec5452f606a671b8432a2f7f0c94fbb49fc8

• FFMPEG - windows - MJPEG stream - convert to MP4 and re-stream - C# - newbie [on hold]

  27 décembre 2014, par ganesh

  Am a newbie to work with videos and stuff.

  Doing a project with a panasonic IP cam (WV-ns-202A) which will be the source of an MJPEG stream.

  I need to Display this content on a ASP.net,C# web app under IIS, as well as store the same in filesystem.

  sofar have tried out some ffmpeg/vlc stuff not helping much.

  (tried #transcoding to flv and save in local filesystem - success
  #transcoding and restream over http - failed
  #transcoding with vlc & streaming (sout) - failed
  )

  looks like am lost.

  Do i must need a meadia server to stream this or can i manage with ffmpeg itself ?.
  one camera to one user, no users more then that at any point of time.

  Any help appreciated - guidence, sample code, links, c#,

  Thanks for your time.

• 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)