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我正在编写将实时音频和视频从网络摄像头流式传输到 rtmp-server 的程序。我在 MacOS X 10.8 中工作，所以我使用 AVFoundation 框架从输入设备获取音频和视频帧。这个框架进入委托：

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

其中sampleBuffer包含音频或视频数据。

当我在 中接收音频数据时sampleBuffer，我正在尝试将此数据转换为并使用 libavcodecAVFrame进行编码：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);
    }

问题是，当我得到如此形成的帧时，我可以听到想要的声音，但它会变慢并且不连续（好像每个数据帧之后都会出现相同的静音帧）。CMSampleBuffer从到的转换似乎有问题 AVFrame，因为来自相同样本缓冲区的 AVFoundation 创建的麦克风的预览正常播放。

我会很感激你的帮助。

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;

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