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• FFMpeg Muxer Internal Buffering

  7 mars 2016, par rkrishnan2012

  I am using libFFmpeg to mux h264 and aac packets into an flv container and streaming that via rtmp.

  I want to be able to detect when frames are being sent to the server or the size of the internal buffer, in order to detect the network lag and apply variable bit-rate. However, when I clock the time it takes for av_write_frame, it is much too fast for it to be syncronous (so we can’t use this to detect network speed).

  I already tried calling av_write_frame with a null parameter (to flush data) every frame, and also directly calling avio_flush but the bitrate is still too high. So, I am wondering if there is another buffer that I am missing (such as a socket-level buffer).

  I also tried to print the AVIOContext->buf_end, buffer_size, and buf_ptr values but none of those are indicative of my network speed getting throttled (when I limit the bandwidth on my router, I would expect the buffer size to increase).

  tl ;dr - I would like to know how to properly detect packets being sent to the server, or the network speed lagging compared to my av_write_frame calls.

  If it makes a difference, this is being run on an arm-v7a Android device.

• FFmpeg iOS -> output file is invalid

  15 juillet 2016, par Vive

  I’m using following library to convert mkv to mp4 : https://github.com/SterlingOnLoop/FFmpegWrapper.

  - (void)convertUsingFFmpegWrapper {
  
      NSString *mp4Extension = @"mp4";
  
      NSString *mkvExtension = @"mkv";
  
  
  
      NSString *videoName = @"file1";
  
  //    NSString *videoName = @"file2";
  
  
  
      NSString *mkvVideoFilePath = [[NSBundle mainBundle] pathForResource:videoName ofType:mkvExtension];
  
      NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
  
      NSString *documentsDirectory = paths[0];
  
      NSString *mp4VideoFilePath = [NSString stringWithFormat:@"%@/%@.%@", documentsDirectory, videoName, mp4Extension];
  
  
  
      FFmpegWrapper *ffmpegWrapper = [[FFmpegWrapper alloc] init];
  
      NSDictionary *options = @{kFFmpegInputFormatKey: mkvExtension, kFFmpegOutputFormatKey: mp4Extension};
  
      [ffmpegWrapper convertInputPath:mkvVideoFilePath outputPath:mp4VideoFilePath options:options progressBlock:nil completionBlock:^(BOOL success, NSError *error) {
  
          if (success && !error) {
  
              // delete mp4 file
  
          } else if (error) {
  
              NSLog(@"Error during .MKV -> .MP4 conversion occured: %@", error.localizedDescription);
  
          } else {
  
              NSLog(@"Unknown error during .MKV -> .MP4 conversion occured.");
  
          }
  
      }];
  
  }

  Here are the values from LLDB about the automatically detected codec types :

  (lldb) po inputStream.codecName
  
  aac
  
  
  
  (lldb) po outputStream.codecName
  
  aac

  I should mention here, that originally the file is generated on Linux with following codecs : vaapiencode_h264 for video and faac for sound.

  The issue is that the file simply does not work. I get huge amount of logs in the console, where most important is :

  [aac @ 0x7f7f65019200] Format aac detected only with low score of 1, misdetection possible!

  Inside the library, the following function is used :

  int streamInfoValue = avformat_find_stream_info(inputFormatContext, NULL);

  And exactly this line does the whole mess with the logs. Obviously, without this line I receive an error with invalid argument.

  When that line is turned on, the .mp4 file is generated. It lasts > 5 minutes, while input file is 11 seconds long. It cannot be played using VLC on my mac (seems to be broken). I get following errors (I’m pasting few of them, full track can be found here, it’s too long to quote it here) :

  [aac @ 0x7ff07c00b000] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  [aac @ 0x7ff07c00b000] channel element 0.0 is not allocated
  
  [aac @ 0x7ff07c00b000] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 0x7ff07c00b000] Inconsistent channel configuration.
  
  [aac @ 0x7ff07c00b000] get_buffer() failed
  
  [aac @ 0x7ff07c00b000] channel element 3.10 is not allocated
  
  [aac @ 0x7ff07c00b000] Reserved bit set.
  
  [aac @ 0x7ff07c00b000] invalid band type
  
  [aac @ 0x7ff07c00b000] Number of scalefactor bands in group (50) exceeds limit (41).
  
  [aac @ 0x7ff07c00b000] Reserved bit set.
  
  [aac @ 0x7ff07c00b000] Number of bands (7) exceeds limit (4).
  
  [aac @ 0x7ff07c00b000] Reserved bit set.
  
  [aac @ 0x7ff07c00b000] Dependent coupling is not supported together with LTP

  Any idea how to simply convert mkv to mp4 ? I don’t have any idea why the errors occurs. I’d claim that the file is not aac, but the linux uses this encoding, so it should be valid.

• AAC encoder : fix wrong gain sacalefactor being set

  26 novembre 2015, par Claudio Freire

  AAC encoder : fix wrong gain sacalefactor being set
  In some conditions, where the first band was being zeroed
  
  mainly, the wrong global gain scalefactor would be written
  
  to the stream since it’s always taken from the first band
  
  regardless of whether it’s been marked as zero or not.
  So, always make sure it contians something useful.
  

  • [DH] libavcodec/aaccoder_twoloop.h