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• FFMPEG Understanding AVFrame::linesize (Audio)

  4 décembre 2015, par user3584691

  As per the doucmentation of AVFrame, for audio, lineSize is size in bytes of each plane and only linesize[0] may be set. But however, am unsure whether lineszie[0] is holding per plane buffer size or is it the complete buffer size and we have to divide it by no of channels to get per plane buffer size.

  For Example, when I call
  int data_size = av_samples_get_buffer_size(NULL, iDesiredNoOfChannels, iAudioSamples, (AVSampleFormat)iDesiredFormat, 0) ; For iDesiredNoOfChannels = 2, iAudioSamples = 1024 & iDesiredFormat = AV_SAMPLE_FMT_FLTP data_size=8192. Pretty straightforward, as each sample is 4 bytes and since there are 2 channels total memory will be (1024 * 4 * 2) bytes. As such lineSize[0] should be 4096 for planar audio. data[0] & data[1] should be each of size 4096. However, pFrame->lineSize[0] is giving 8192. So to get the size per plane, I have to do pFrame->lineSize[0] / pFrame->channels. Isn’t this behaviour different from what the documentation suggests or is my understanding of the documentaion wrong.

• How do I convert flac to wav on iOS ?

  8 juin 2017, par popctrl

  I have a file which is encoded with FLAC and I want to convert it to WAV.

  I have added this FFMpeg lib to my project and imported it.

  I see some code from this answer, but I am unclear on how to use it :

  #import "avformat.h"
  
  
  
  // Some code goes here
  
  
  
  /*
  
   *   avformat_open_input(AVFormatContext **ps, const char *filename, AVInputFormat *fmt, AVDictionary **options)
  
   */
  
  int openInputValue = avformat_open_input(&pFormatCtx, utf8FilePath, inputFormat, nil);
  
  NSLog(@"%s - %d # openInputValue = %d", __PRETTY_FUNCTION__, __LINE__, openInputValue);

  I currently have this code in a function which takes NSData holding the FLAC file. If avformat_open_input is the correct call, how do I set the variable ? If it is not the correct call, what is ?

  This question seems like a duplicate but it doesn’t really have a good answer.

  Also note that I don’t want a player. This file contains MQA so I need to run it through my own custom decoder.

• How can I determine if a codec / container combination is compatible with FFmpeg ?

  26 juin 2023, par Dan

  I'm looking at re-muxing some containers holding audio and video such that I extract the best, first audio stream, and store it in a new container where e.g. only the audio stream is present.

  



  The output context for FFmpeg is created like so :

  



  AVFormatContext* output_context = NULL;
avformat_alloc_output_context2( &output_context, NULL, "mp4", NULL );


  



  I have a shortlist of acceptable outputs, e.g. MP4, M4A, etc … essentially those that are readable by Apple's Audio File Services :

  



  kAudioFileAIFFType              = 'AIFF',
kAudioFileAIFCType              = 'AIFC',
kAudioFileWAVEType              = 'WAVE',
kAudioFileSoundDesigner2Type    = 'Sd2f',
kAudioFileNextType              = 'NeXT',
kAudioFileMP3Type               = 'MPG3',   // mpeg layer 3
kAudioFileMP2Type               = 'MPG2',   // mpeg layer 2
kAudioFileMP1Type               = 'MPG1',   // mpeg layer 1
kAudioFileAC3Type               = 'ac-3',
kAudioFileAAC_ADTSType          = 'adts',
kAudioFileMPEG4Type             = 'mp4f',
kAudioFileM4AType               = 'm4af',
kAudioFileM4BType               = 'm4bf',
kAudioFileCAFType               = 'caff',
kAudioFile3GPType               = '3gpp',
kAudioFile3GP2Type              = '3gp2',
kAudioFileAMRType               = 'amrf'


  



  My question is this : is there an easy API in FFmpeg that can be leveraged to choose a compatible output container given the codec the audio stream is in ?