Recherche avancée

Sur d’autres sites (8878)

• ffmpeg 3.3.2 and newer native aac encoder maximum number of channels [on hold]

  21 mars 2018, par AndySP

  I’m attempting to convert a MOGG (Multi-channel OGG) audio file with 16 channels into an AAC file using FFMPEG native AAC encoder (tried version 3.3.2 and also version 86344-gb5a0971). I’m using Audacity as the source and Export -> External Application to pipe the output to FFMPEG using the command :

  ffmpeg -i - -c:a aac -b:a 240k "%f" 

  Its returning "Unsupported number of channels : 16" and then failing. If I reduce the number of channels to 6 it works fine but obviously I’m not willing to remove 10 channels before exporting.

  Does anyone have an idea if its possible to get ffmpeg to accept more than 6 channels in AAC ?

• How to fill audio AVFrame (ffmpeg) with the data obtained from CMSampleBufferRef (AVFoundation) ?

  24 mai 2013, par Aleksei2414904

  I am writing program for streaming live audio and video from webcamera to rtmp-server. I work in MacOS X 10.8, so I use AVFoundation framework for obtaining audio and video frames from input devices. This frames come into delegate :

  -(void) captureOutput:(AVCaptureOutput*)captureOutput didOutputSampleBuffer:    (CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection*)connection ,

  where sampleBuffer contains audio or video data.

  When I recieve audio data in the sampleBuffer, I'm trying to convert this data into AVFrame and encode AVFramewith libavcodec :

      aframe = avcodec_alloc_frame();  //AVFrame *aframe;
  
      int got_packet, ret;
  
      CMItemCount numSamples = CMSampleBufferGetNumSamples(sampleBuffer); //CMSampleBufferRef
  
  
  
      NSUInteger channelIndex = 0;
  
  
  
      CMBlockBufferRef audioBlockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
  
  
  
      size_t audioBlockBufferOffset = (channelIndex * numSamples * sizeof(SInt16));
  
  
  
      size_t lengthAtOffset = 0;
  
  
  
      size_t totalLength = 0;
  
  
  
      SInt16 *samples = NULL;
  
  
  
      CMBlockBufferGetDataPointer(audioBlockBuffer, audioBlockBufferOffset, &lengthAtOffset, &totalLength, (char **)(&samples));
  
  
  
              const AudioStreamBasicDescription *audioDescription = CMAudioFormatDescriptionGetStreamBasicDescription(CMSampleBufferGetFormatDescription(sampleBuffer));
  
  
  
      aframe->nb_samples =(int) numSamples;
  
  
  
      aframe->channels=audioDescription->mChannelsPerFrame;
  
  
  
      aframe->sample_rate=(int)audioDescription->mSampleRate;
  
  
  
       //my webCamera configured to produce 16bit 16kHz LPCM mono, so sample format hardcoded here, and seems to be correct
  
      avcodec_fill_audio_frame(aframe, aframe->channels, AV_SAMPLE_FMT_S16,
  
  
  
                               (uint8_t *)samples,
  
  
  
                                aframe->nb_samples *
  
  
  
                               av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) *
  
  
  
                               aframe->channels, 0);  
  
      //encoding audio
  
      ret = avcodec_encode_audio2(c, &pkt, aframe, &got_packet);
  
      if (ret < 0) {
  
          fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
  
          exit(1);
  
      }

  The problem is that when I get so formed frames, I can hear the wanted sound, but it is slowing down and discontinuous (as if after each data frame comes the same frame of silence). It seems that something is wrong in the transformation from CMSampleBuffer to AVFrame , because the preview from the microphone created with AVFoundation from the same sample buffers played normally.

  I would be grateful for your help.

  UPD : Creating and initializing the AVCodceContext structure
   

      audio_codec= avcodec_find_encoder(AV_CODEC_ID_AAC);
  
      if (!(audio_codec)) {
  
          fprintf(stderr, "Could not find encoder for '%s'\n",
  
                  avcodec_get_name(AV_CODEC_ID_AAC));
  
          exit(1);
  
      }
  
      audio_st = avformat_new_stream(oc, audio_codec);  //AVFormatContext *oc;
  
      if (!audio_st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
  
  
        audio_st->id=1;
  
        audio_st->codec->sample_fmt= AV_SAMPLE_FMT_S16;
  
        audio_st->codec->bit_rate = 64000;
  
        audio_st->codec->sample_rate= 16000;
  
        audio_st->codec->channels=1;
  
        audio_st->codec->codec_type= AVMEDIA_TYPE_AUDIO;

• Why libffmpeg.so for opera for running on debian fail

  17 juin 2023, par Nonok cantik

  Why does libffmpeg.so for opera for running on debian fail

  


  `/opt/opera/opera: /lib/x86_64-linux-gnu/libm.so.6: version `GLIBC_2.29' not found (required by /opt/opera/lib_extra/libffmpeg.so)
/opt/opera/opera: /lib/x86_64-linux-gnu/libm.so.6: version `GLIBC_2.35' not found (required by /opt/opera/lib_extra/libffmpeg.so)
/opt/opera/opera: /lib/x86_64-linux-gnu/libc.so.6: version `GLIBC_2.34' not found (required by /opt/opera/lib_extra/libffmpeg.so)
`


  


  if it's used by arch, simply works
  
please help out solve such confusing