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sampleBuffer中的音频数据时，我正在尝试将此数据转换为AVFrame并使用libavcodec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);
    }

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); }

问题在于，当我得到如此形成的帧时，我可以听到想要的声音，但它正在减速和不连续（好像在每个数据帧出现相同的静音帧之后）。从到CMSampleBuffer的转换似乎出现了问题，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：创建并初始化AVCodceContext结构 AVFrame

    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;