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Revert "Adjust naming conventions and general refactoring in HLE Project (#490)" (#526)

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This reverts commit 85dbb9559a.
This commit is contained in:
gdkchan 2018-12-04 22:52:39 -02:00 • committed by GitHub
parent 85dbb9559a
commit 3615a70cae
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GPG key ID: 4AEE18F83AFDEB23
299 changed files with 12276 additions and 12268 deletions
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114
Ryujinx.HLE/HOS/Services/Aud/AudioRenderer/VoiceContext.cs
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@ -6,13 +6,13 @@ namespace Ryujinx.HLE.HOS.Services.Aud.AudioRenderer
{
class VoiceContext
{
private bool _acquired;
private bool _bufferReload;
private bool Acquired;
private bool BufferReload;
private int _resamplerFracPart;
private int ResamplerFracPart;
private int _bufferIndex;
private int _offset;
private int BufferIndex;
private int Offset;
public int SampleRate;
public int ChannelsCount;
@ -29,138 +29,138 @@ namespace Ryujinx.HLE.HOS.Services.Aud.AudioRenderer
public VoiceOut OutStatus;
private int[] _samples;
private int[] Samples;
public bool Playing => _acquired && PlayState == PlayState.Playing;
public bool Playing => Acquired && PlayState == PlayState.Playing;
public VoiceContext()
{
WaveBuffers = new WaveBuffer[4];
}
public void SetAcquireState(bool newState)
public void SetAcquireState(bool NewState)
{
if (_acquired && !newState)
if (Acquired && !NewState)
{
//Release.
Reset();
}
_acquired = newState;
Acquired = NewState;
}
private void Reset()
{
_bufferReload = true;
BufferReload = true;
_bufferIndex = 0;
_offset = 0;
BufferIndex = 0;
Offset = 0;
OutStatus.PlayedSamplesCount = 0;
OutStatus.PlayedWaveBuffersCount = 0;
OutStatus.VoiceDropsCount = 0;
}
public int[] GetBufferData(MemoryManager memory, int maxSamples, out int samplesCount)
public int[] GetBufferData(MemoryManager Memory, int MaxSamples, out int SamplesCount)
{
if (!Playing)
{
samplesCount = 0;
SamplesCount = 0;
return null;
}
if (_bufferReload)
if (BufferReload)
{
_bufferReload = false;
BufferReload = false;
UpdateBuffer(memory);
UpdateBuffer(Memory);
}
WaveBuffer wb = WaveBuffers[_bufferIndex];
WaveBuffer Wb = WaveBuffers[BufferIndex];
int maxSize = _samples.Length - _offset;
int MaxSize = Samples.Length - Offset;
int size = maxSamples * AudioConsts.HostChannelsCount;
int Size = MaxSamples * AudioConsts.HostChannelsCount;
if (size > maxSize)
if (Size > MaxSize)
{
size = maxSize;
Size = MaxSize;
}
int[] output = new int[size];
int[] Output = new int[Size];
Array.Copy(_samples, _offset, output, 0, size);
Array.Copy(Samples, Offset, Output, 0, Size);
samplesCount = size / AudioConsts.HostChannelsCount;
SamplesCount = Size / AudioConsts.HostChannelsCount;
OutStatus.PlayedSamplesCount += samplesCount;
OutStatus.PlayedSamplesCount += SamplesCount;
_offset += size;
Offset += Size;
if (_offset == _samples.Length)
if (Offset == Samples.Length)
{
_offset = 0;
Offset = 0;
if (wb.Looping == 0)
if (Wb.Looping == 0)
{
SetBufferIndex((_bufferIndex + 1) & 3);
SetBufferIndex((BufferIndex + 1) & 3);
}
OutStatus.PlayedWaveBuffersCount++;
if (wb.LastBuffer != 0)
if (Wb.LastBuffer != 0)
{
PlayState = PlayState.Paused;
}
}
return output;
return Output;
}
private void UpdateBuffer(MemoryManager memory)
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];
WaveBuffer Wb = WaveBuffers[BufferIndex];
if (wb.Position == 0)
if (Wb.Position == 0)
{
_samples = new int[0];
Samples = new int[0];
return;
}
if (SampleFormat == SampleFormat.PcmInt16)
{
int samplesCount = (int)(wb.Size / (sizeof(short) * ChannelsCount));
int SamplesCount = (int)(Wb.Size / (sizeof(short) * ChannelsCount));
_samples = new int[samplesCount * AudioConsts.HostChannelsCount];
Samples = new int[SamplesCount * AudioConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int index = 0; index < samplesCount; index++)
for (int Index = 0; Index < SamplesCount; Index++)
{
short sample = memory.ReadInt16(wb.Position + index * 2);
short Sample = Memory.ReadInt16(Wb.Position + Index * 2);
_samples[index * 2 + 0] = sample;
_samples[index * 2 + 1] = sample;
Samples[Index * 2 + 0] = Sample;
Samples[Index * 2 + 1] = Sample;
}
}
else
{
for (int index = 0; index < samplesCount * 2; index++)
for (int Index = 0; Index < SamplesCount * 2; Index++)
{
_samples[index] = memory.ReadInt16(wb.Position + index * 2);
Samples[Index] = Memory.ReadInt16(Wb.Position + Index * 2);
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] buffer = memory.ReadBytes(wb.Position, wb.Size);
byte[] Buffer = Memory.ReadBytes(Wb.Position, Wb.Size);
_samples = AdpcmDecoder.Decode(buffer, AdpcmCtx);
Samples = AdpcmDecoder.Decode(Buffer, AdpcmCtx);
}
else
{
@ -172,24 +172,24 @@ namespace Ryujinx.HLE.HOS.Services.Aud.AudioRenderer
//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;
int SamplesCount = Samples.Length / AudioConsts.HostChannelsCount;
samplesCount = Math.Max(samplesCount - 4, 0);
SamplesCount = Math.Max(SamplesCount - 4, 0);
_samples = Resampler.Resample2Ch(
_samples,
Samples = Resampler.Resample2Ch(
Samples,
SampleRate,
AudioConsts.HostSampleRate,
samplesCount,
ref _resamplerFracPart);
SamplesCount,
ref ResamplerFracPart);
}
}
public void SetBufferIndex(int index)
public void SetBufferIndex(int Index)
{
_bufferIndex = index & 3;
BufferIndex = Index & 3;
_bufferReload = true;
BufferReload = true;
}
}
}