public int[] GetBufferData(MemoryManager memory, int maxSamples, out int samplesCount)
{
if (!Playing)
{
samplesCount = 0;
return(null);
}
if (_bufferReload)
{
_bufferReload = false;
UpdateBuffer(memory);
}
WaveBuffer wb = WaveBuffers[_bufferIndex];
int maxSize = _samples.Length - _offset;
int size = maxSamples * AudioRendererConsts.HostChannelsCount;
if (size > maxSize)
{
size = maxSize;
}
int[] output = new int[size];
Array.Copy(_samples, _offset, output, 0, size);
samplesCount = size / AudioRendererConsts.HostChannelsCount;
_outStatus.PlayedSamplesCount += samplesCount;
_offset += size;
if (_offset == _samples.Length)
{
_offset = 0;
if (wb.Looping == 0)
{
SetBufferIndex(_bufferIndex + 1);
}
_outStatus.PlayedWaveBuffersCount++;
if (wb.LastBuffer != 0)
{
PlayState = PlayState.Paused;
}
}
return(output);
}
private void UpdateBuffer(MemoryManager memory)
{
// TODO: Implement conversion for formats other
// than interleaved stereo (2 channels).
// As of now, it assumes that HostChannelsCount == 2.
WaveBuffer wb = WaveBuffers[_bufferIndex];
if (wb.Position == 0)
{
_samples = new int[0];
return;
}
if (SampleFormat == SampleFormat.PcmInt16)
{
int samplesCount = (int)(wb.Size / (sizeof(short) * ChannelsCount));
_samples = new int[samplesCount * AudioRendererConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int index = 0; index < samplesCount; index++)
{
short sample = memory.Read <short>((ulong)(wb.Position + index * 2));
_samples[index * 2 + 0] = sample;
_samples[index * 2 + 1] = sample;
}
}
else
{
for (int index = 0; index < samplesCount * 2; index++)
{
_samples[index] = memory.Read <short>((ulong)(wb.Position + index * 2));
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] buffer = new byte[wb.Size];
memory.Read((ulong)wb.Position, buffer);
_samples = AdpcmDecoder.Decode(buffer, AdpcmCtx);
}
else
{
throw new InvalidOperationException();
}
if (SampleRate != AudioRendererConsts.HostSampleRate)
{
// TODO: We should keep the frames being discarded (see the 4 below)
// on a buffer and include it on the next samples buffer, to allow
// the resampler to do seamless interpolation between wave buffers.
int samplesCount = _samples.Length / AudioRendererConsts.HostChannelsCount;
samplesCount = Math.Max(samplesCount - 4, 0);
_samples = Resampler.Resample2Ch(
_samples,
SampleRate,
AudioRendererConsts.HostSampleRate,
samplesCount,
ref _resamplerFracPart);
}
}