private void UpdateBuffer(AMemory 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 (SampleFormat == SampleFormat.PcmInt16)
{
int SamplesCount = (int)(Wb.Size / (sizeof(short) * ChannelsCount));
Samples = new int[SamplesCount * AudioConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int Index = 0; Index < SamplesCount; Index++)
{
short Sample = Memory.ReadInt16(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.ReadInt16(Wb.Position + Index * 2);
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] Buffer = Memory.ReadBytes(Wb.Position, Wb.Size);
Samples = AdpcmDecoder.Decode(Buffer, AdpcmCtx);
}
else
{
throw new InvalidOperationException();
}
if (SampleRate != AudioConsts.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 / AudioConsts.HostChannelsCount;
SamplesCount = Math.Max(SamplesCount - 4, 0);
Samples = Resampler.Resample2Ch(
Samples,
SampleRate,
AudioConsts.HostSampleRate,
SamplesCount,
ref ResamplerFracPart);
}
}
public int[] GetBufferData(AMemory 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 * AudioConsts.HostChannelsCount;
if (Size > MaxSize)
{
Size = MaxSize;
}
int[] Output = new int[Size];
Array.Copy(Samples, Offset, Output, 0, Size);
SamplesCount = Size / AudioConsts.HostChannelsCount;
OutStatus.PlayedSamplesCount += SamplesCount;
Offset += Size;
if (Offset == Samples.Length)
{
Offset = 0;
if (Wb.Looping == 0)
{
SetBufferIndex((BufferIndex + 1) & 3);
}
OutStatus.PlayedWaveBuffersCount++;
if (Wb.LastBuffer != 0)
{
PlayState = PlayState.Paused;
}
}
return(Output);
}
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 * AudioConsts.HostChannelsCount;
if (size > maxSize)
{
size = maxSize;
}
int[] output = new int[size];
Array.Copy(_samples, _offset, output, 0, size);
samplesCount = size / AudioConsts.HostChannelsCount;
_outStatus.PlayedSamplesCount += samplesCount;
_offset += size;
if (_offset == _samples.Length)
{
_offset = 0;
if (wb.Looping == 0)
{
SetBufferIndex((_bufferIndex + 1) & 3);
}
_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 * AudioConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int index = 0; index < samplesCount; index++)
{
short sample = memory.ReadInt16(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.ReadInt16(wb.Position + index * 2);
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] buffer = memory.ReadBytes(wb.Position, wb.Size);
_samples = AdpcmDecoder.Decode(buffer, AdpcmCtx);
}
else
{
throw new InvalidOperationException();
}
if (SampleRate != AudioConsts.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 / AudioConsts.HostChannelsCount;
samplesCount = Math.Max(samplesCount - 4, 0);
_samples = Resampler.Resample2Ch(
_samples,
SampleRate,
AudioConsts.HostSampleRate,
samplesCount,
ref _resamplerFracPart);
}
}