mirror of
https://github.com/GreemDev/Ryujinx.git
synced 2024-12-04 22:37:57 +01:00
f77694e4f7
* 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
201 lines
5.4 KiB
C#
201 lines
5.4 KiB
C#
using Ryujinx.Audio.Adpcm;
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using Ryujinx.Cpu;
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using System;
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namespace Ryujinx.HLE.HOS.Services.Audio.AudioRendererManager
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{
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class VoiceContext
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{
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private bool _acquired;
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private bool _bufferReload;
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private int _resamplerFracPart;
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private int _bufferIndex;
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private int _offset;
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public int SampleRate { get; set; }
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public int ChannelsCount { get; set; }
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public float Volume { get; set; }
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public PlayState PlayState { get; set; }
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public SampleFormat SampleFormat { get; set; }
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public AdpcmDecoderContext AdpcmCtx { get; set; }
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public WaveBuffer[] WaveBuffers { get; }
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public WaveBuffer CurrentWaveBuffer => WaveBuffers[_bufferIndex];
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private VoiceOut _outStatus;
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public VoiceOut OutStatus => _outStatus;
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private int[] _samples;
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public bool Playing => _acquired && PlayState == PlayState.Playing;
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public VoiceContext()
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{
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WaveBuffers = new WaveBuffer[4];
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}
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public void SetAcquireState(bool newState)
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{
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if (_acquired && !newState)
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{
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// Release.
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Reset();
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}
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_acquired = newState;
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}
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private void Reset()
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{
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_bufferReload = true;
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_bufferIndex = 0;
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_offset = 0;
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_outStatus.PlayedSamplesCount = 0;
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_outStatus.PlayedWaveBuffersCount = 0;
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_outStatus.VoiceDropsCount = 0;
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}
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public int[] GetBufferData(MemoryManager memory, int maxSamples, out int samplesCount)
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{
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if (!Playing)
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{
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samplesCount = 0;
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return null;
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}
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if (_bufferReload)
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{
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_bufferReload = false;
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UpdateBuffer(memory);
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}
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WaveBuffer wb = WaveBuffers[_bufferIndex];
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int maxSize = _samples.Length - _offset;
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int size = maxSamples * AudioRendererConsts.HostChannelsCount;
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if (size > maxSize)
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{
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size = maxSize;
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}
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int[] output = new int[size];
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Array.Copy(_samples, _offset, output, 0, size);
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samplesCount = size / AudioRendererConsts.HostChannelsCount;
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_outStatus.PlayedSamplesCount += samplesCount;
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_offset += size;
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if (_offset == _samples.Length)
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{
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_offset = 0;
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if (wb.Looping == 0)
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{
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SetBufferIndex(_bufferIndex + 1);
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}
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_outStatus.PlayedWaveBuffersCount++;
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if (wb.LastBuffer != 0)
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{
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PlayState = PlayState.Paused;
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}
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}
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return output;
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}
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private void UpdateBuffer(MemoryManager memory)
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{
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// TODO: Implement conversion for formats other
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// than interleaved stereo (2 channels).
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// As of now, it assumes that HostChannelsCount == 2.
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WaveBuffer wb = WaveBuffers[_bufferIndex];
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if (wb.Position == 0)
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{
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_samples = new int[0];
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return;
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}
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if (SampleFormat == SampleFormat.PcmInt16)
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{
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int samplesCount = (int)(wb.Size / (sizeof(short) * ChannelsCount));
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_samples = new int[samplesCount * AudioRendererConsts.HostChannelsCount];
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if (ChannelsCount == 1)
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{
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for (int index = 0; index < samplesCount; index++)
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{
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short sample = memory.Read<short>((ulong)(wb.Position + index * 2));
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_samples[index * 2 + 0] = sample;
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_samples[index * 2 + 1] = sample;
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}
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}
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else
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{
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for (int index = 0; index < samplesCount * 2; index++)
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{
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_samples[index] = memory.Read<short>((ulong)(wb.Position + index * 2));
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}
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}
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}
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else if (SampleFormat == SampleFormat.Adpcm)
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{
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byte[] buffer = new byte[wb.Size];
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memory.Read((ulong)wb.Position, buffer);
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_samples = AdpcmDecoder.Decode(buffer, AdpcmCtx);
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}
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else
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{
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throw new InvalidOperationException();
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}
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if (SampleRate != AudioRendererConsts.HostSampleRate)
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{
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// TODO: We should keep the frames being discarded (see the 4 below)
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// on a buffer and include it on the next samples buffer, to allow
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// the resampler to do seamless interpolation between wave buffers.
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int samplesCount = _samples.Length / AudioRendererConsts.HostChannelsCount;
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samplesCount = Math.Max(samplesCount - 4, 0);
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_samples = Resampler.Resample2Ch(
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_samples,
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SampleRate,
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AudioRendererConsts.HostSampleRate,
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samplesCount,
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ref _resamplerFracPart);
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}
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}
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public void SetBufferIndex(int index)
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{
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_bufferIndex = index & 3;
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_bufferReload = true;
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}
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}
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} |