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

* Rename enum fields

* Naming conventions

* Remove unneeded ".this"

* Remove unneeded semicolons

* Remove unused Usings

* Don't use var

* Remove unneeded enum underlying types

* Explicitly label class visibility

* Remove unneeded @ prefixes

* Remove unneeded commas

* Remove unneeded if expressions

* Method doesn't use unsafe code

* Remove unneeded casts

* Initialized objects don't need an empty constructor

* Remove settings from DotSettings

* Revert "Explicitly label class visibility"

This reverts commit ad5eb5787c.

* Small changes

* Revert external enum renaming

* Changes from feedback

* Remove unneeded property setters
This commit is contained in:
Alex Barney 2018-12-04 14:23:37 -06:00 committed by gdkchan
parent c86aacde76
commit 85dbb9559a
299 changed files with 12268 additions and 12276 deletions

View file

@ -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;
}
}
}