void startAudioCapture()
{
if (audioDeviceInfo.isNull())
{
audioInput = nullptr;
return;
}
audioInput = std::make_unique<QAudioInput>(audioDeviceInfo, audioFormat);
QIODevice * audioDevice = audioInput->start();
audioBuffer.clear();
QObject::connect(audioDevice, &QIODevice::readyRead, audioDevice, [=](){
audioBuffer += audioDevice->readAll();
int frameSize = 1024 * audioFormat.channelCount();
while (audioBuffer.size() > frameSize)
{
QByteArray data = audioBuffer.left(frameSize);
audioBuffer.remove(0, frameSize);
writeAudioFrame(audioStreamIndex, data);
}
});
}
void EncoderFfmpegCore::encodeBuffer(const CSAMPLE *samples, const int size) {
unsigned char *l_strBuffer = NULL;
int l_iBufferLen = 0;
//int l_iAudioCpyLen = m_iAudioInputFrameSize *
// av_get_bytes_per_sample(m_pEncoderAudioStream->codec->sample_fmt) *
// m_pEncoderAudioStream->codec->channels;
long l_iLeft = size;
long j = 0;
unsigned int l_iBufPos = 0;
unsigned int l_iPos = 0;
// TODO(XXX): Get rid of repeated malloc here!
float *l_fNormalizedSamples = (float *)malloc(size * sizeof(float));
// We use normalized floats in the engine [-1.0, 1.0] and FFMPEG expects
// samples in the range [-1.0, 1.0] so no conversion is required.
for (j = 0; j < size; j++) {
l_fNormalizedSamples[j] = samples[j];
}
// In MP3 this writes Header same In ogg
// They are written once front of the encoded stuff
if (m_bStreamInitialized == false) {
m_bStreamInitialized = true;
// Write a header.
avio_open_dyn_buf(&m_pEncodeFormatCtx->pb);
if (avformat_write_header(m_pEncodeFormatCtx, NULL) != 0) {
qDebug() << "EncoderFfmpegCore::encodeBuffer: failed to write a header.";
return;
}
l_iBufferLen = avio_close_dyn_buf(m_pEncodeFormatCtx->pb,
(uint8_t**)(&l_strBuffer));
m_pCallback->write(NULL, l_strBuffer, 0, l_iBufferLen);
av_free(l_strBuffer);
}
while (l_iLeft > (m_iFltAudioCpyLen / 4)) {
memset(m_pFltSamples, 0x00, m_iFltAudioCpyLen);
for (j = 0; j < m_iFltAudioCpyLen / 4; j++) {
if (m_lBufferSize > 0) {
m_pFltSamples[j] = m_SBuffer[ l_iBufPos++ ];
m_lBufferSize--;
m_lRecordedBytes++;
} else {
m_pFltSamples[j] = l_fNormalizedSamples[l_iPos++];
l_iLeft--;
m_lRecordedBytes++;
}
if (l_iLeft <= 0) {
qDebug() << "ffmpegencodercore: No samples left.. for encoding!";
break;
}
}
m_lBufferSize = 0;
// Open dynamic buffer for writing next bytes
if (avio_open_dyn_buf(&m_pEncodeFormatCtx->pb) < 0) {
qDebug() << "Can't alloc Dyn buffer!";
return;
}
// Write it to buffer (FILE) and then close buffer for waiting
// Next encoded buffe to come or we stop encode
if (! writeAudioFrame(m_pEncodeFormatCtx, m_pEncoderAudioStream)) {
l_iBufferLen = avio_close_dyn_buf(m_pEncodeFormatCtx->pb,
(uint8_t**)(&l_strBuffer));
m_pCallback->write(NULL, l_strBuffer, 0, l_iBufferLen);
av_free(l_strBuffer);
}
}
// Keep things clean
memset(m_SBuffer, 0x00, 65535);
for (j = 0; j < l_iLeft; j++) {
m_SBuffer[ j ] = l_fNormalizedSamples[ l_iPos++ ];
}
m_lBufferSize = l_iLeft;
free(l_fNormalizedSamples);
}
