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Ryujinx/Ryujinx.HLE/HOS/Services/Audio/AudioRendererManager/VoiceContext.cs
gdkchan f77694e4f7
Implement a new physical memory manager and replace DeviceMemory (#856)
* Implement a new physical memory manager and replace DeviceMemory

* Proper generic constraints

* Fix debug build

* Add memory tests

* New CPU memory manager and general code cleanup

* Remove host memory management from CPU project, use Ryujinx.Memory instead

* Fix tests

* Document exceptions on MemoryBlock

* Fix leak on unix memory allocation

* Proper disposal of some objects on tests

* Fix JitCache not being set as initialized

* GetRef without checks for 8-bits and 16-bits CAS

* Add MemoryBlock destructor

* Throw in separate method to improve codegen

* Address PR feedback

* QueryModified improvements

* Fix memory write tracking not marking all pages as modified in some cases

* Simplify MarkRegionAsModified

* Remove XML doc for ghost param

* Add back optimization to avoid useless buffer updates

* Add Ryujinx.Cpu project, move MemoryManager there and remove MemoryBlockWrapper

* Some nits

* Do not perform address translation when size is 0

* Address PR feedback and format NativeInterface class

* Remove ghost parameter description

* Update Ryujinx.Cpu to .NET Core 3.1

* Address PR feedback

* Fix build

* Return a well defined value for GetPhysicalAddress with invalid VA, and do not return unmapped ranges as modified

* Typo
2020-05-04 08:54:50 +10:00

201 lines
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5.4 KiB
C#

using Ryujinx.Audio.Adpcm;
using Ryujinx.Cpu;
using System;
namespace Ryujinx.HLE.HOS.Services.Audio.AudioRendererManager
{
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 * 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);
}
}
public void SetBufferIndex(int index)
{
_bufferIndex = index & 3;
_bufferReload = true;
}
}
}