Ryujinx/Ryujinx.HLE/OsHle/Services/Aud/AudioRenderer/VoiceContext.cs
gdkchan 98c6ceede5
Audio Renderer improvements (#210)
* Partial voice implementation on audio renderer

* Implemented audren resampler (based on original impl)

* Fix BiquadFilter struct

* Pause audio playback on last stream buffer

* Split audren/audout files into separate folders, some minor cleanup

* Use AudioRendererParameter on GetWorkBufferSize aswell

* Bump audren version to REV4, name a few things, increase sample buffer size

* Remove useless new lines
2018-07-14 23:57:41 -03:00

188 lines
4.9 KiB
C#

using ChocolArm64.Memory;
using Ryujinx.Audio.Adpcm;
using System;
namespace Ryujinx.HLE.OsHle.Services.Aud.AudioRenderer
{
class VoiceContext
{
private bool Acquired;
private bool BufferReload;
private int ResamplerFracPart;
private int BufferIndex;
private int Offset;
public int SampleRate;
public int ChannelsCount;
public float Volume;
public PlayState PlayState;
public SampleFormat SampleFormat;
public AdpcmDecoderContext AdpcmCtx;
public WaveBuffer[] WaveBuffers;
public VoiceOut 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(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;
}
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 void SetBufferIndex(int Index)
{
BufferIndex = Index & 3;
BufferReload = true;
}
}
}