mirror of
https://github.com/GreemDev/Ryujinx.git
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f556c80d02
* Haydn: Part 1 Based on my reverse of audio 11.0.0. As always, core implementation under LGPLv3 for the same reasons as for Amadeus. This place the bases of a more flexible audio system while making audout & audin accurate. This have the following improvements: - Complete reimplementation of audout and audin. - Audin currently only have a dummy backend. - Dramatically reduce CPU usage by up to 50% in common cases (SoundIO and OpenAL). - Audio Renderer now can output to 5.1 devices when supported. - Audio Renderer init its backend on demand instead of keeping two up all the time. - All backends implementation are now in their own project. - Ryujinx.Audio.Renderer was renamed Ryujinx.Audio and was refactored because of this. As a note, games having issues with OpenAL haven't improved and will not because of OpenAL design (stopping when buffers finish playing causing possible audio "pops" when buffers are very small). * Update for latest hexkyz's edits on Switchbrew * audren: Rollback channel configuration changes * Address gdkchan's comments * Fix typo in OpenAL backend driver * Address last comments * Fix a nit * Address gdkchan's comments
255 lines
8.1 KiB
C#
255 lines
8.1 KiB
C#
//
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// Copyright (c) 2019-2021 Ryujinx
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//
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// This program is free software: you can redistribute it and/or modify
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// it under the terms of the GNU Lesser General Public License as published by
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// the Free Software Foundation, either version 3 of the License, or
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// (at your option) any later version.
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//
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// This program is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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// GNU Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public License
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// along with this program. If not, see <https://www.gnu.org/licenses/>.
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//
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using Ryujinx.Audio.Integration;
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using Ryujinx.Audio.Renderer.Dsp.Command;
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using Ryujinx.Audio.Renderer.Utils;
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using Ryujinx.Common;
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using Ryujinx.Common.Logging;
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using System;
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using System.Threading;
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namespace Ryujinx.Audio.Renderer.Dsp
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{
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public class AudioProcessor : IDisposable
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{
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private const int MaxBufferedFrames = 5;
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private const int TargetBufferedFrames = 3;
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private enum MailboxMessage : uint
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{
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Start,
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Stop,
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RenderStart,
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RenderEnd
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}
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private class RendererSession
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{
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public CommandList CommandList;
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public int RenderingLimit;
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public ulong AppletResourceId;
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}
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private Mailbox<MailboxMessage> _mailbox;
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private RendererSession[] _sessionCommandList;
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private Thread _workerThread;
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public IHardwareDevice[] OutputDevices { get; private set; }
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private long _lastTime;
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private long _playbackEnds;
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private ManualResetEvent _event;
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public AudioProcessor()
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{
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_event = new ManualResetEvent(false);
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}
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private static uint GetHardwareChannelCount(IHardwareDeviceDriver deviceDriver)
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{
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// Get the real device driver (In case the compat layer is on top of it).
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deviceDriver = deviceDriver.GetRealDeviceDriver();
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if (deviceDriver.SupportsChannelCount(6))
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{
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return 6;
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}
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else
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{
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// NOTE: We default to stereo as this will get downmixed to mono by the compat layer if it's not compatible.
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return 2;
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}
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}
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public void Start(IHardwareDeviceDriver deviceDriver)
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{
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OutputDevices = new IHardwareDevice[Constants.AudioRendererSessionCountMax];
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// TODO: Before enabling this, we need up-mixing from stereo to 5.1.
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// uint channelCount = GetHardwareChannelCount(deviceDriver);
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uint channelCount = 2;
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for (int i = 0; i < OutputDevices.Length; i++)
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{
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// TODO: Don't hardcode sample rate.
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OutputDevices[i] = new HardwareDeviceImpl(deviceDriver, channelCount, Constants.TargetSampleRate);
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}
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_mailbox = new Mailbox<MailboxMessage>();
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_sessionCommandList = new RendererSession[Constants.AudioRendererSessionCountMax];
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_event.Reset();
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_lastTime = PerformanceCounter.ElapsedNanoseconds;
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StartThread();
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_mailbox.SendMessage(MailboxMessage.Start);
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if (_mailbox.ReceiveResponse() != MailboxMessage.Start)
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{
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throw new InvalidOperationException("Audio Processor Start response was invalid!");
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}
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}
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public void Stop()
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{
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_mailbox.SendMessage(MailboxMessage.Stop);
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if (_mailbox.ReceiveResponse() != MailboxMessage.Stop)
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{
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throw new InvalidOperationException("Audio Processor Stop response was invalid!");
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}
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foreach (IHardwareDevice device in OutputDevices)
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{
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device.Dispose();
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}
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}
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public void Send(int sessionId, CommandList commands, int renderingLimit, ulong appletResourceId)
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{
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_sessionCommandList[sessionId] = new RendererSession
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{
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CommandList = commands,
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RenderingLimit = renderingLimit,
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AppletResourceId = appletResourceId
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};
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}
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public void Signal()
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{
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_mailbox.SendMessage(MailboxMessage.RenderStart);
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}
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public void Wait()
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{
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if (_mailbox.ReceiveResponse() != MailboxMessage.RenderEnd)
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{
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throw new InvalidOperationException("Audio Processor Wait response was invalid!");
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}
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long increment = Constants.AudioProcessorMaxUpdateTimeTarget;
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long timeNow = PerformanceCounter.ElapsedNanoseconds;
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if (timeNow > _playbackEnds)
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{
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// Playback has restarted.
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_playbackEnds = timeNow;
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}
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_playbackEnds += increment;
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// The number of frames we are behind where the timer says we should be.
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long framesBehind = (timeNow - _lastTime) / increment;
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// The number of frames yet to play on the backend.
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long bufferedFrames = (_playbackEnds - timeNow) / increment + framesBehind;
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// If we've entered a situation where a lot of buffers will be queued on the backend,
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// Skip some audio frames so that playback can catch up.
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if (bufferedFrames > MaxBufferedFrames)
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{
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// Skip a few frames so that we're not too far behind. (the target number of frames)
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_lastTime += increment * (bufferedFrames - TargetBufferedFrames);
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}
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while (timeNow < _lastTime + increment)
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{
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_event.WaitOne(1);
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timeNow = PerformanceCounter.ElapsedNanoseconds;
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}
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_lastTime += increment;
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}
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private void StartThread()
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{
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_workerThread = new Thread(Work)
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{
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Name = "AudioProcessor.Worker"
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};
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_workerThread.Start();
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}
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private void Work()
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{
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if (_mailbox.ReceiveMessage() != MailboxMessage.Start)
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{
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throw new InvalidOperationException("Audio Processor Start message was invalid!");
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}
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_mailbox.SendResponse(MailboxMessage.Start);
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_mailbox.SendResponse(MailboxMessage.RenderEnd);
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Logger.Info?.Print(LogClass.AudioRenderer, "Starting audio processor");
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while (true)
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{
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MailboxMessage message = _mailbox.ReceiveMessage();
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if (message == MailboxMessage.Stop)
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{
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break;
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}
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if (message == MailboxMessage.RenderStart)
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{
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long startTicks = PerformanceCounter.ElapsedNanoseconds;
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for (int i = 0; i < _sessionCommandList.Length; i++)
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{
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if (_sessionCommandList[i] != null)
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{
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_sessionCommandList[i].CommandList.Process(OutputDevices[i]);
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_sessionCommandList[i] = null;
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}
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}
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long endTicks = PerformanceCounter.ElapsedNanoseconds;
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long elapsedTime = endTicks - startTicks;
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if (Constants.AudioProcessorMaxUpdateTime < elapsedTime)
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{
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Logger.Debug?.Print(LogClass.AudioRenderer, $"DSP too slow (exceeded by {elapsedTime - Constants.AudioProcessorMaxUpdateTime}ns)");
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}
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_mailbox.SendResponse(MailboxMessage.RenderEnd);
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}
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}
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Logger.Info?.Print(LogClass.AudioRenderer, "Stopping audio processor");
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_mailbox.SendResponse(MailboxMessage.Stop);
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}
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public void Dispose()
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{
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Dispose(true);
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}
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protected virtual void Dispose(bool disposing)
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{
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if (disposing)
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{
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_event.Dispose();
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}
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}
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}
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}
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