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

Question

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;