bool CDVDPlayerResampler::Retrieve(DVDAudioFrame &audioframe, double &pts)
{
//check if nr of channels changed so we can allocate new buffers if necessary
CheckResampleBuffers(audioframe.channel_count);
//value to divide samples by to get them into -1.0:1.0 range
float scale = (float)(1 << (audioframe.bits_per_sample - 1));
int nrframes = audioframe.size / audioframe.channel_count / (audioframe.bits_per_sample / 8);
//if we don't have enough in the samplebuffer, return false
if (nrframes > m_bufferfill)
{
return false;
}
//use the pts of the first fresh value in the samplebuffer
pts = m_ptsbuffer[0];
//add from samplebuffer to audioframe
float* inputptr = m_buffer;
int16_t* outputptr = (int16_t*)audioframe.data;
for (int i = 0; i < nrframes * m_nrchannels; i++)
*outputptr++ = MathUtils::round_int(Clamp(*inputptr++ * scale, scale * -1.0f, scale - 1.0f));
m_bufferfill -= nrframes;
//shift old data to the beginning of the buffer
memmove(m_buffer, m_buffer + (nrframes * m_nrchannels), m_bufferfill * m_nrchannels * sizeof(float));
memmove(m_ptsbuffer, m_ptsbuffer + nrframes, m_bufferfill * sizeof(double));
return true;
}
void CDVDPlayerResampler::Add(DVDAudioFrame &audioframe, double pts)
{
//check if nr of channels changed so we can allocate new buffers if necessary
CheckResampleBuffers(audioframe.channel_count);
//value to divide samples by to get them into -1.0:1.0 range
float scale = (float)(1 << (audioframe.bits_per_sample - 1));
int nrframes = audioframe.size / audioframe.channel_count / (audioframe.bits_per_sample / 8);
//resize sample buffer if necessary
//we want the buffer to be large enough to hold the current frames in it,
//the number of frames needed for libsamplerate's input
//and the maximum number of frames libsamplerate might generate, times 2 for safety
ResizeSampleBuffer(m_bufferfill + nrframes + nrframes * MathUtils::round_int(m_ratio + 0.5) * 2);
//assign samplebuffers
m_converterdata.input_frames = nrframes;
m_converterdata.output_frames = m_buffersize - m_bufferfill - m_converterdata.input_frames;
//output buffer starts at the place where the buffer doesn't hold samples
m_converterdata.data_out = m_buffer + m_bufferfill * m_nrchannels;
//intput buffer is a block of data at the end of the buffer
m_converterdata.data_in = m_converterdata.data_out + m_converterdata.output_frames * m_nrchannels;
//add samples to the resample input buffer
int16_t* inputptr = (int16_t*)audioframe.data;
float* outputptr = m_converterdata.data_in;
for (int i = 0; i < nrframes * m_nrchannels; i++)
*outputptr++ = (float)*inputptr++ / scale;
//resample
m_converterdata.src_ratio = m_ratio;
src_set_ratio(m_converter, m_ratio);
src_process(m_converter, &m_converterdata);
//calculate a pts for each sample
for (int i = 0; i < m_converterdata.output_frames_gen; i++)
{
m_ptsbuffer[m_bufferfill] = pts + i * (audioframe.duration / (double)m_converterdata.output_frames_gen);
m_bufferfill++;
}
}