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
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//
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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.Common;
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using Ryujinx.Audio.Renderer.Common;
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using System;
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using static Ryujinx.Audio.Renderer.Parameter.VoiceInParameter;
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namespace Ryujinx.Audio.Renderer.Dsp.Command
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{
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public class PcmInt16DataSourceCommandVersion1 : ICommand
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{
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public bool Enabled { get; set; }
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public int NodeId { get; }
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public CommandType CommandType => CommandType.PcmInt16DataSourceVersion1;
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public ulong EstimatedProcessingTime { get; set; }
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public ushort OutputBufferIndex { get; }
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public uint SampleRate { get; }
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public uint ChannelIndex { get; }
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public uint ChannelCount { get; }
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public float Pitch { get; }
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public WaveBuffer[] WaveBuffers { get; }
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public Memory<VoiceUpdateState> State { get; }
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public DecodingBehaviour DecodingBehaviour { get; }
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public PcmInt16DataSourceCommandVersion1(ref Server.Voice.VoiceState serverState, Memory<VoiceUpdateState> state, ushort outputBufferIndex, ushort channelIndex, int nodeId)
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{
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Enabled = true;
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NodeId = nodeId;
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OutputBufferIndex = (ushort)(channelIndex + outputBufferIndex);
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SampleRate = serverState.SampleRate;
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ChannelIndex = channelIndex;
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ChannelCount = serverState.ChannelsCount;
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Pitch = serverState.Pitch;
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WaveBuffers = new WaveBuffer[Constants.VoiceWaveBufferCount];
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for (int i = 0; i < WaveBuffers.Length; i++)
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{
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ref Server.Voice.WaveBuffer voiceWaveBuffer = ref serverState.WaveBuffers[i];
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WaveBuffers[i] = voiceWaveBuffer.ToCommon(1);
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}
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State = state;
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DecodingBehaviour = serverState.DecodingBehaviour;
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}
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public void Process(CommandList context)
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{
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Span<float> outputBuffer = context.GetBuffer(OutputBufferIndex);
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DataSourceHelper.WaveBufferInformation info = new DataSourceHelper.WaveBufferInformation()
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{
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State = State,
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SourceSampleRate = SampleRate,
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SampleFormat = SampleFormat.PcmInt16,
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Pitch = Pitch,
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DecodingBehaviour = DecodingBehaviour,
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WaveBuffers = WaveBuffers,
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ExtraParameter = 0,
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ExtraParameterSize = 0,
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ChannelIndex = (int)ChannelIndex,
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ChannelCount = (int)ChannelCount,
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};
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DataSourceHelper.ProcessWaveBuffers(context.MemoryManager, outputBuffer, info, context.SampleRate, (int)context.SampleCount);
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}
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}
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}
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