Choppy sound when using ffmpeg - c++

Question

I suffered some choppy audio when i try to capture audio from a live stream.
Another essential problem which could explain the problem is that the Wav file created is twice longer than the capture time.
The audio is perfect when i play the avs input file with ffplay, so the avs is ok, the problem is after whether in the capture or in the Wav writing.
To capture :
av_read_frame(pFormatCtx, &packet)
if(packet.stream_index == mAudioStream)
{
int buff_size = sizeof(mAudioBuffer);
std::cout << "Buff_size " << buff_size << std::endl;
len = avcodec_decode_audio3(pAudioCodecCtx,(int16_t*)mAudioBuffer, &buff_size,&packet);
if(len < 0){
qDebug("Extractor - Audio isEnd = -1;");
mAudioBufferSize = 0;
isEnd = ERROR_;
return isEnd;
}
// Set packet result type
mFrameType = AUDIO_PKT;
mAudioBufferSize = buff_size;
//store audio synchronization informations:
if(packet.pts != AV_NOPTS_VALUE) {
mAudioPts_ = av_q2d(pFormatCtx->streams[mAudioStream]->time_base);
mAudioPts_ *= packet.pts;
}
}
// store a copy of current audio frame in _frame
_frame.audioFrame = new decoded_frame_t::audio_frame_t();
_frame.audioFrame->sampleRate = mediaInfos.audioSampleRate;
_frame.audioFrame->sampleSize = mediaInfos.audioSampleSize;
_frame.audioFrame->nbChannels = mediaInfos.audioNbChannels;
_frame.audioFrame->nbSamples = mAudioBufferSize / ((mediaInfos.audioSampleSize/8) * mediaInfos.audioNbChannels);
_frame.audioFrame->buf.resize(mAudioBufferSize);
memcpy(&_frame.audioFrame->buf[0],mAudioBuffer,mAudioBufferSize);
Then i store in a Wav File using libsndfile :
SNDFILE* fd;
SF_INFO sfInf;
sfInf.frames = 0;
sfInf.channels = p_capt->ui_nbChannels;
sfInf.samplerate = p_capt->ui_sampleRate;
sfInf.format = SF_FORMAT_WAV | SF_FORMAT_PCM_U8;
sfInf.sections = 0;
sfInf.seekable = 0;
if (sf_format_check(&sfInf) == FALSE)
std::cout << "Format parameter are uncorrect ! Exit saving !" << std::endl;
else
{
fd = sf_open(fileName.toStdString().c_str(), SFM_WRITE, &sfInf);
if (fd == NULL)
{
std::cout << "Unable to open the file " << fileName.toStdString() << std::endl;
return GRAB_ST_NOK;
}
//little trick because v_buf is a uint8_t vector
sf_count_t l = sf_write_short(fd, (const short *)(&(p_capt->v_buf[0])), p_capt->v_buf.size()/2);
if (l != p_capt->v_buf.size()/2)
{
std::cout << "sf_write didn't write the right amoung of bits " << l << " != " << p_capt->v_buf.size()/2 << std::endl;
ret = GRAB_ST_NOK;
}
else
{
sf_write_sync(fd);
sf_close(fd);
ret = GRAB_ST_OK;
}
}
I hope it's understandable. Waiting for remarks.
Kurt

Answer 1

Ok problem solved.
There were two main problems :
resize DO add n element and is not just preparing the vector for further push etc...
the buff_size of avcodec_decode_audio3 return a length in bytes but is copying in a int16_t array so it can be disturbing.