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Audio Renderer improvements (#210)

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* 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
This commit is contained in:
gdkchan
2018-07-14 23:57:41 -03:00
committed by GitHub
parent be31f5b46d
commit 98c6ceede5
34 changed files with 1168 additions and 271 deletions
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91
Ryujinx.Audio/Adpcm/AdpcmDecoder.cs Normal file
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@ -0,0 +1,91 @@
namespace Ryujinx.Audio.Adpcm
{
public static class AdpcmDecoder
{
private const int SamplesPerFrame = 14;
private const int BytesPerFrame = 8;
public static int[] Decode(byte[] Buffer, AdpcmDecoderContext Context)
{
int Samples = GetSamplesCountFromSize(Buffer.Length);
int[] Pcm = new int[Samples * 2];
short History0 = Context.History0;
short History1 = Context.History1;
int InputOffset = 0;
int OutputOffset = 0;
while (InputOffset < Buffer.Length)
{
byte Header = Buffer[InputOffset++];
int Scale = 0x800 << (Header & 0xf);
int CoeffIndex = (Header >> 4) & 7;
short Coeff0 = Context.Coefficients[CoeffIndex * 2 + 0];
short Coeff1 = Context.Coefficients[CoeffIndex * 2 + 1];
int FrameSamples = SamplesPerFrame;
if (FrameSamples > Samples)
{
FrameSamples = Samples;
}
int Value = 0;
for (int SampleIndex = 0; SampleIndex < FrameSamples; SampleIndex++)
{
int Sample;
if ((SampleIndex & 1) == 0)
{
Value = Buffer[InputOffset++];
Sample = (Value << 24) >> 28;
}
else
{
Sample = (Value << 28) >> 28;
}
int Prediction = Coeff0 * History0 + Coeff1 * History1;
Sample = (Sample * Scale + Prediction + 0x400) >> 11;
short SaturatedSample = DspUtils.Saturate(Sample);
History1 = History0;
History0 = SaturatedSample;
Pcm[OutputOffset++] = SaturatedSample;
Pcm[OutputOffset++] = SaturatedSample;
}
Samples -= FrameSamples;
}
Context.History0 = History0;
Context.History1 = History1;
return Pcm;
}
public static long GetSizeFromSamplesCount(int SamplesCount)
{
int Frames = SamplesCount / SamplesPerFrame;
return Frames * BytesPerFrame;
}
public static int GetSamplesCountFromSize(long Size)
{
int Frames = (int)(Size / BytesPerFrame);
return Frames * SamplesPerFrame;
}
}
}

10
Ryujinx.Audio/Adpcm/AdpcmDecoderContext.cs Normal file
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@ -0,0 +1,10 @@
namespace Ryujinx.Audio.Adpcm
{
public class AdpcmDecoderContext
{
public short[] Coefficients;
public short History0;
public short History1;
}
}

13
Ryujinx.Audio/AudioFormat.cs
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@ -1,13 +0,0 @@
namespace Ryujinx.Audio
{
public enum AudioFormat
{
Invalid = 0,
PcmInt8 = 1,
PcmInt16 = 2,
PcmImt24 = 3,
PcmImt32 = 4,
PcmFloat = 5,
Adpcm = 6
}
}

16
Ryujinx.Audio/DspUtils.cs Normal file
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@ -0,0 +1,16 @@
namespace Ryujinx.Audio.Adpcm
{
public static class DspUtils
{
public static short Saturate(int Value)
{
if (Value > short.MaxValue)
Value = short.MaxValue;
if (Value < short.MinValue)
Value = short.MinValue;
return (short)Value;
}
}
}

8
Ryujinx.Audio/IAalOutput.cs
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@ -2,11 +2,7 @@ namespace Ryujinx.Audio
{
public interface IAalOutput
{
int OpenTrack(
int SampleRate,
int Channels,
ReleaseCallback Callback,
out AudioFormat Format);
int OpenTrack(int SampleRate, int Channels, ReleaseCallback Callback);
void CloseTrack(int Track);
@ -14,7 +10,7 @@ namespace Ryujinx.Audio
long[] GetReleasedBuffers(int Track, int MaxCount);
void AppendBuffer(int Track, long Tag, byte[] Buffer);
void AppendBuffer<T>(int Track, long Tag, T[] Buffer) where T : struct;
void Start(int Track);
void Stop(int Track);

53
Ryujinx.Audio/OpenAL/OpenALAudioOut.cs
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@ -3,6 +3,7 @@ using OpenTK.Audio.OpenAL;
using System;
using System.Collections.Concurrent;
using System.Collections.Generic;
using System.Runtime.InteropServices;
using System.Threading;
namespace Ryujinx.Audio.OpenAL
@ -226,15 +227,9 @@ namespace Ryujinx.Audio.OpenAL
while (KeepPolling);
}
public int OpenTrack(
int SampleRate,
int Channels,
ReleaseCallback Callback,
out AudioFormat Format)
public int OpenTrack(int SampleRate, int Channels, ReleaseCallback Callback)
{
Format = AudioFormat.PcmInt16;
Track Td = new Track(SampleRate, GetALFormat(Channels, Format), Callback);
Track Td = new Track(SampleRate, GetALFormat(Channels), Callback);
for (int Id = 0; Id < MaxTracks; Id++)
{
@ -247,38 +242,16 @@ namespace Ryujinx.Audio.OpenAL
return -1;
}
private ALFormat GetALFormat(int Channels, AudioFormat Format)
private ALFormat GetALFormat(int Channels)
{
if (Channels == 1)
switch (Channels)
{
switch (Format)
{
case AudioFormat.PcmInt8: return ALFormat.Mono8;
case AudioFormat.PcmInt16: return ALFormat.Mono16;
}
}
else if (Channels == 2)
{
switch (Format)
{
case AudioFormat.PcmInt8: return ALFormat.Stereo8;
case AudioFormat.PcmInt16: return ALFormat.Stereo16;
}
}
else if (Channels == 6)
{
switch (Format)
{
case AudioFormat.PcmInt8: return ALFormat.Multi51Chn8Ext;
case AudioFormat.PcmInt16: return ALFormat.Multi51Chn16Ext;
}
}
else
{
throw new ArgumentOutOfRangeException(nameof(Channels));
case 1: return ALFormat.Mono16;
case 2: return ALFormat.Stereo16;
case 6: return ALFormat.Multi51Chn16Ext;
}
throw new ArgumentException(nameof(Format));
throw new ArgumentOutOfRangeException(nameof(Channels));
}
public void CloseTrack(int Track)
@ -309,13 +282,15 @@ namespace Ryujinx.Audio.OpenAL
return null;
}
public void AppendBuffer(int Track, long Tag, byte[] Buffer)
public void AppendBuffer<T>(int Track, long Tag, T[] Buffer) where T : struct
{
if (Tracks.TryGetValue(Track, out Track Td))
{
int BufferId = Td.AppendBuffer(Tag);
AL.BufferData(BufferId, Td.Format, Buffer, Buffer.Length, Td.SampleRate);
int Size = Buffer.Length * Marshal.SizeOf<T>();
AL.BufferData<T>(BufferId, Td.Format, Buffer, Size, Td.SampleRate);
AL.SourceQueueBuffer(Td.SourceId, BufferId);
@ -366,7 +341,5 @@ namespace Ryujinx.Audio.OpenAL
return PlaybackState.Stopped;
}
}
}