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using ChocolArm64.Memory;
using Ryujinx.Audio.Adpcm;
using System;
namespace Ryujinx.HLE.HOS.Services.Aud.AudioRenderer
{
class VoiceContext
{
private bool _acquired;
private bool _bufferReload;
private int _resamplerFracPart;
private int _bufferIndex;
private int _offset;
public int SampleRate { get; set; }
public int ChannelsCount { get; set; }
public float Volume { get; set; }
public PlayState PlayState { get; set; }
public SampleFormat SampleFormat { get; set; }
public AdpcmDecoderContext AdpcmCtx { get; set; }
public WaveBuffer[] WaveBuffers { get; }
public WaveBuffer CurrentWaveBuffer => WaveBuffers[_bufferIndex];
private VoiceOut _outStatus;
public VoiceOut OutStatus => _outStatus;
private int[] _samples;
public bool Playing => _acquired && PlayState == PlayState.Playing;
public VoiceContext()
{
WaveBuffers = new WaveBuffer[4];
}
public void SetAcquireState(bool newState)
{
if (_acquired && !newState)
{
//Release.
Reset();
}
_acquired = newState;
}
private void Reset()
{
_bufferReload = true;
_bufferIndex = 0;
_offset = 0;
_outStatus.PlayedSamplesCount = 0;
_outStatus.PlayedWaveBuffersCount = 0;
_outStatus.VoiceDropsCount = 0;
}
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);
}
}
public void SetBufferIndex(int index)
{
_bufferIndex = index & 3;
_bufferReload = true;
}
}
}
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