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https://github.com/Ryujinx/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
361 lines
12 KiB
C#
361 lines
12 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.Common;
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using Ryujinx.Audio.Renderer.Common;
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using Ryujinx.Audio.Renderer.Dsp;
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using Ryujinx.Common.Memory;
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using System;
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using System.Runtime.CompilerServices;
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using System.Runtime.InteropServices;
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namespace Ryujinx.Audio.Renderer.Parameter
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{
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/// <summary>
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/// Input information for a voice.
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/// </summary>
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[StructLayout(LayoutKind.Sequential, Size = 0x170, Pack = 1)]
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public struct VoiceInParameter
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{
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/// <summary>
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/// Id of the voice.
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/// </summary>
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public int Id;
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/// <summary>
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/// Node id of the voice.
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/// </summary>
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public int NodeId;
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/// <summary>
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/// Set to true if the voice is new.
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/// </summary>
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[MarshalAs(UnmanagedType.I1)]
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public bool IsNew;
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/// <summary>
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/// Set to true if the voice is used.
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/// </summary>
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[MarshalAs(UnmanagedType.I1)]
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public bool InUse;
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/// <summary>
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/// The voice <see cref="PlayState"/> wanted by the user.
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/// </summary>
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public PlayState PlayState;
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/// <summary>
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/// The <see cref="SampleFormat"/> of the voice.
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/// </summary>
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public SampleFormat SampleFormat;
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/// <summary>
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/// The sample rate of the voice.
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/// </summary>
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public uint SampleRate;
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/// <summary>
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/// The priority of the voice.
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/// </summary>
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public uint Priority;
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/// <summary>
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/// Target sorting position of the voice. (Used to sort voices with the same <see cref="Priority"/>)
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/// </summary>
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public uint SortingOrder;
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/// <summary>
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/// The total channel count used.
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/// </summary>
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public uint ChannelCount;
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/// <summary>
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/// The pitch used on the voice.
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/// </summary>
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public float Pitch;
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/// <summary>
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/// The output volume of the voice.
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/// </summary>
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public float Volume;
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/// <summary>
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/// Biquad filters to apply to the output of the voice.
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/// </summary>
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public Array2<BiquadFilterParameter> BiquadFilters;
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/// <summary>
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/// Total count of <see cref="WaveBufferInternal"/> of the voice.
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/// </summary>
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public uint WaveBuffersCount;
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/// <summary>
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/// Current playing <see cref="WaveBufferInternal"/> of the voice.
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/// </summary>
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public uint WaveBuffersIndex;
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/// <summary>
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/// Reserved/unused.
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/// </summary>
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private uint _reserved1;
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/// <summary>
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/// User state address required by the data source.
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/// </summary>
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/// <remarks>Only used for <see cref="SampleFormat.Adpcm"/> as the address of the GC-ADPCM coefficients.</remarks>
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public ulong DataSourceStateAddress;
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/// <summary>
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/// User state size required by the data source.
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/// </summary>
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/// <remarks>Only used for <see cref="SampleFormat.Adpcm"/> as the size of the GC-ADPCM coefficients.</remarks>
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public ulong DataSourceStateSize;
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/// <summary>
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/// The target mix id of the voice.
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/// </summary>
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public int MixId;
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/// <summary>
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/// The target splitter id of the voice.
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/// </summary>
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public uint SplitterId;
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/// <summary>
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/// The wavebuffer parameters of this voice.
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/// </summary>
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public Array4<WaveBufferInternal> WaveBuffers;
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/// <summary>
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/// The channel resource ids associated to the voice.
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/// </summary>
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public Array6<int> ChannelResourceIds;
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/// <summary>
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/// Reset the voice drop flag during voice server update.
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/// </summary>
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[MarshalAs(UnmanagedType.I1)]
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public bool ResetVoiceDropFlag;
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/// <summary>
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/// Flush the amount of wavebuffer specified. This will result in the wavebuffer being skipped and marked played.
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/// </summary>
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/// <remarks>This was added on REV5.</remarks>
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public byte FlushWaveBufferCount;
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/// <summary>
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/// Reserved/unused.
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/// </summary>
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private ushort _reserved2;
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/// <summary>
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/// Change the behaviour of the voice.
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/// </summary>
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/// <remarks>This was added on REV5.</remarks>
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public DecodingBehaviour DecodingBehaviourFlags;
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/// <summary>
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/// Change the Sample Rate Conversion (SRC) quality of the voice.
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/// </summary>
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/// <remarks>This was added on REV8.</remarks>
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public SampleRateConversionQuality SrcQuality;
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/// <summary>
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/// This was previously used for opus codec support on the Audio Renderer and was removed on REV3.
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/// </summary>
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public uint ExternalContext;
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/// <summary>
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/// This was previously used for opus codec support on the Audio Renderer and was removed on REV3.
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/// </summary>
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public uint ExternalContextSize;
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/// <summary>
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/// Reserved/unused.
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/// </summary>
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private unsafe fixed uint _reserved3[2];
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/// <summary>
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/// Input information for a voice wavebuffer.
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/// </summary>
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[StructLayout(LayoutKind.Sequential, Size = 0x38, Pack = 1)]
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public struct WaveBufferInternal
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{
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/// <summary>
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/// Address of the wavebuffer data.
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/// </summary>
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public ulong Address;
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/// <summary>
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/// Size of the wavebuffer data.
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/// </summary>
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public ulong Size;
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/// <summary>
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/// Offset of the first sample to play.
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/// </summary>
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public uint StartSampleOffset;
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/// <summary>
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/// Offset of the last sample to play.
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/// </summary>
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public uint EndSampleOffset;
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/// <summary>
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/// If set to true, the wavebuffer will loop when reaching <see cref="EndSampleOffset"/>.
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/// </summary>
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/// <remarks>
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/// Starting with REV8, you can specify how many times to loop the wavebuffer (<see cref="LoopCount"/>) and where it should start and end when looping (<see cref="LoopFirstSampleOffset"/> and <see cref="LoopLastSampleOffset"/>)
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/// </remarks>
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[MarshalAs(UnmanagedType.I1)]
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public bool ShouldLoop;
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/// <summary>
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/// Indicates that this is the last wavebuffer to play of the voice.
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/// </summary>
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[MarshalAs(UnmanagedType.I1)]
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public bool IsEndOfStream;
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/// <summary>
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/// Indicates if the server should update its internal state.
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/// </summary>
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[MarshalAs(UnmanagedType.I1)]
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public bool SentToServer;
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/// <summary>
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/// Reserved/unused.
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/// </summary>
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private byte _reserved;
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/// <summary>
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/// If set to anything other than 0, specifies how many times to loop the wavebuffer.
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/// </summary>
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/// <remarks>This was added in REV8.</remarks>
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public int LoopCount;
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/// <summary>
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/// Address of the context used by the sample decoder.
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/// </summary>
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/// <remarks>This is only currently used by <see cref="SampleFormat.Adpcm"/>.</remarks>
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public ulong ContextAddress;
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/// <summary>
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/// Size of the context used by the sample decoder.
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/// </summary>
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/// <remarks>This is only currently used by <see cref="SampleFormat.Adpcm"/>.</remarks>
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public ulong ContextSize;
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/// <summary>
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/// If set to anything other than 0, specifies the offset of the first sample to play when looping.
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/// </summary>
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/// <remarks>This was added in REV8.</remarks>
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public uint LoopFirstSampleOffset;
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/// <summary>
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/// If set to anything other than 0, specifies the offset of the last sample to play when looping.
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/// </summary>
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/// <remarks>This was added in REV8.</remarks>
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public uint LoopLastSampleOffset;
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/// <summary>
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/// Check if the sample offsets are in a valid range for generic PCM.
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/// </summary>
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/// <typeparam name="T">The PCM sample type</typeparam>
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/// <returns>Returns true if the sample offset are in range of the size.</returns>
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[MethodImpl(MethodImplOptions.AggressiveInlining)]
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private bool IsSampleOffsetInRangeForPcm<T>() where T : unmanaged
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{
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uint dataTypeSize = (uint)Unsafe.SizeOf<T>();
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return StartSampleOffset * dataTypeSize <= Size &&
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EndSampleOffset * dataTypeSize <= Size;
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}
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/// <summary>
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/// Check if the sample offsets are in a valid range for the given <see cref="SampleFormat"/>.
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/// </summary>
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/// <param name="format">The target <see cref="SampleFormat"/></param>
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/// <returns>Returns true if the sample offset are in range of the size.</returns>
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public bool IsSampleOffsetValid(SampleFormat format)
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{
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bool result;
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switch (format)
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{
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case SampleFormat.PcmInt16:
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result = IsSampleOffsetInRangeForPcm<ushort>();
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break;
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case SampleFormat.PcmFloat:
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result = IsSampleOffsetInRangeForPcm<float>();
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break;
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case SampleFormat.Adpcm:
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result = AdpcmHelper.GetAdpcmDataSize((int)StartSampleOffset) <= Size &&
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AdpcmHelper.GetAdpcmDataSize((int)EndSampleOffset) <= Size;
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break;
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default:
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throw new NotImplementedException($"{format} not implemented!");
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}
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return result;
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}
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}
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/// <summary>
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/// Flag altering the behaviour of wavebuffer decoding.
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/// </summary>
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[Flags]
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public enum DecodingBehaviour : ushort
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{
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/// <summary>
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/// Default decoding behaviour.
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/// </summary>
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Default = 0,
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/// <summary>
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/// Reset the played samples accumulator when looping.
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/// </summary>
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PlayedSampleCountResetWhenLooping = 1,
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/// <summary>
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/// Skip pitch and Sample Rate Conversion (SRC).
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/// </summary>
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SkipPitchAndSampleRateConversion = 2
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}
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/// <summary>
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/// Specify the quality to use during Sample Rate Conversion (SRC) and pitch handling.
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/// </summary>
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/// <remarks>This was added in REV8.</remarks>
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public enum SampleRateConversionQuality : byte
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{
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/// <summary>
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/// Resample interpolating 4 samples per output sample.
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/// </summary>
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Default,
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/// <summary>
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/// Resample interpolating 8 samples per output sample.
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/// </summary>
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High,
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/// <summary>
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/// Resample interpolating 1 samples per output sample.
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/// </summary>
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Low
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}
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}
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}
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