Haydn: Part 1 (#2007)

* 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

Ryujinx.Audio/Renderer/Dsp/Command/DeviceSinkCommand.cs

@@ -0,0 +1,108 @@
+//
+// Copyright (c) 2019-2021 Ryujinx
+//
+// This program is free software: you can redistribute it and/or modify
+// it under the terms of the GNU Lesser General Public License as published by
+// the Free Software Foundation, either version 3 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public License
+// along with this program.  If not, see <https://www.gnu.org/licenses/>.
+//
+
+using Ryujinx.Audio.Integration;
+using Ryujinx.Audio.Renderer.Server.Sink;
+using System;
+using System.Runtime.CompilerServices;
+using System.Text;
+
+namespace Ryujinx.Audio.Renderer.Dsp.Command
+{
+    public class DeviceSinkCommand : ICommand
+    {
+        public bool Enabled { get; set; }
+
+        public int NodeId { get; }
+
+        public CommandType CommandType => CommandType.DeviceSink;
+
+        public ulong EstimatedProcessingTime { get; set; }
+
+        public string DeviceName { get; }
+
+        public int SessionId { get; }
+
+        public uint InputCount { get; }
+        public ushort[] InputBufferIndices { get; }
+
+        public Memory<float> Buffers { get; }
+
+        public DeviceSinkCommand(uint bufferOffset, DeviceSink sink, int sessionId, Memory<float> buffers, int nodeId)
+        {
+            Enabled = true;
+            NodeId = nodeId;
+
+            DeviceName = Encoding.ASCII.GetString(sink.Parameter.DeviceName).TrimEnd('\0');
+            SessionId = sessionId;
+            InputCount = sink.Parameter.InputCount;
+            InputBufferIndices = new ushort[InputCount];
+
+            for (int i = 0; i < InputCount; i++)
+            {
+                InputBufferIndices[i] = (ushort)(bufferOffset + sink.Parameter.Input[i]);
+            }
+
+            if (sink.UpsamplerState != null)
+            {
+                Buffers = sink.UpsamplerState.OutputBuffer;
+            }
+            else
+            {
+                Buffers = buffers;
+            }
+        }
+
+        [MethodImpl(MethodImplOptions.AggressiveInlining)]
+        private Span<float> GetBuffer(int index, int sampleCount)
+        {
+            return Buffers.Span.Slice(index * sampleCount, sampleCount);
+        }
+
+        public void Process(CommandList context)
+        {
+            IHardwareDevice device = context.OutputDevice;
+
+            if (device.GetSampleRate() == Constants.TargetSampleRate)
+            {
+                int channelCount = (int)device.GetChannelCount();
+                uint bufferCount = Math.Min(device.GetChannelCount(), InputCount);
+
+                const int sampleCount = Constants.TargetSampleCount;
+
+                short[] outputBuffer = new short[bufferCount * sampleCount];
+
+                for (int i = 0; i < bufferCount; i++)
+                {
+                    ReadOnlySpan<float> inputBuffer = GetBuffer(InputBufferIndices[i], sampleCount);
+
+                    for (int j = 0; j < sampleCount; j++)
+                    {
+                        outputBuffer[i + j * channelCount] = PcmHelper.Saturate(inputBuffer[j]);
+                    }
+                }
+
+                device.AppendBuffer(outputBuffer, InputCount);
+            }
+            else
+            {
+                // TODO: support resampling for device only supporting something different
+                throw new NotImplementedException();
+            }
+        }
+    }
+}