FFmpeg audio encoder new encode function - c++

Question

I would like to update an AV Audio encoder using function avcodec_encode_audio (deprecated) to avcodec_encode_audio2, without modifying the structure of existing encoder:
outBytes = avcodec_encode_audio(m_handle, dst, sizeBytes, (const short int*)m_samBuf);
where:
1) m_handle AVCodecContext
2) dst, uint8_t * destination buffer
3) sizeBytes, uint32_t size of the destination buffer
4) m_samBuf void * to the input chunk of data to encode (this is casted to: const short int*)
is there a simply way to do it?
Im tryng with:
int gotPack = 1;
memset (&m_Packet, 0, sizeof (m_Packet));
m_Frame = av_frame_alloc();
av_init_packet(&m_Packet);
m_Packet.data = dst;
m_Packet.size = sizeBytes;
uint8_t* buffer = (uint8_t*)m_samBuf;
m_Frame->nb_samples = m_handle->frame_size;
avcodec_fill_audio_frame(m_Frame,m_handle->channels,m_handle->sample_fmt,buffer,m_FrameSize,1);
outBytes = avcodec_encode_audio2(m_handle, &m_Packet, m_Frame, &gotPack);
char error[256];
av_strerror(outBytes,error,256);
if (outBytes<0){
m_server->log(1,1,"Input data: %d, encode function call error: %s \n",gotPack, error);
return AUDIOWRAPPER_ERROR;
}
av_frame_free(&m_Frame);
it compiles but it does not encode anything, i dont here audio at the output if I pipe the output stream on mplayer, wich was warking prior to the upgrade.
What am I doing wrong?
The encoder accept only two sample formats:
AV_SAMPLE_FMT_S16, ///< signed 16 bits
AV_SAMPLE_FMT_FLT, ///< float
here is how the buffer is allocated:
free(m_samBuf);
int bps = 2;
if(m_handle->codec->sample_fmts[0] == AV_SAMPLE_FMT_FLT) {
bps = 4;
}
m_FrameSize = bps * m_handle->frame_size * m_handle->channels;
m_samBuf = malloc(m_FrameSize);
m_numSam = 0;

Answer 1

avcodec_fill_audio_frame should get you there
memset (&m_Packet, 0, sizeof (m_Packet));
memset (&m_Frame, 0, sizeof (m_Frame));
av_init_packet(&m_Packet);
m_Packet.data = dst;
m_Packet.size = sizeBytes;
m_Frame->nb_samples = //you need to get this value from somewhere, it is the number of samples (per channel) this frame represents
avcodec_fill_audio_frame(m_Frame, m_handle->channels, m_handle->sample_fmt,
buffer,
sizeBytes, 1);
int gotPack = 1;
avcodec_encode_audio2(m_handle, &m_Packet, &m_Frame, &gotPack);