如何使用从CMSampleBufferRef(AVFoundation)获得的数据填充音频AVFrame(ffmpeg)？

Question

我正在编写用于将实时音频和视频从webcamera流式传输到rtmp-server的程序.我在MacOS X 10.8中工作,因此我使用AVFoundation框架从输入设备获取音频和视频帧.这个框架进入代表:

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

其中sampleBuffer包含音频或视频数据.

当我收到音频数据时sampleBuffer,我试图将这些数据转换为libavcodec AVFrame并用其编码AVFrame:

    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);
    }

CMSampleBuffer出现问题AVFrame,因为使用AVFoundation从同一样本缓冲区创建的麦克风预览正常播放.

我很感激你的帮助.

UPD:创建和初始化AVCodceContext结构

    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;

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