forked from Mirror/Ryujinx
a389dd59bd
* Amadeus: Final Act This is my requiem, I present to you Amadeus, a complete reimplementation of the Audio Renderer! This reimplementation is based on my reversing of every version of the audio system module that I carried for the past 10 months. This supports every revision (at the time of writing REV1 to REV8 included) and all features proposed by the Audio Renderer on real hardware. Because this component could be used outside an emulation context, and to avoid possible "inspirations" not crediting the project, I decided to license the Ryujinx.Audio.Renderer project under LGPLv3. - FE3H voices in videos and chapter intro are not present. - Games that use two audio renderer **at the same time** are probably going to have issues right now **until we rewrite the audio output interface** (Crash Team Racing is the only known game to use two renderer at the same time). - Persona 5 Scrambler now goes ingame but audio is garbage. This is caused by the fact that the game engine is syncing audio and video in a really aggressive way. This will disappears the day this game run at full speed. * Make timing more precise when sleeping on Windows Improve precision to a 1ms resolution on Windows NT based OS. This is used to avoid having totally erratic timings and unify all Windows users to the same resolution. NOTE: This is only active when emulation is running.
360 lines
12 KiB
C#
360 lines
12 KiB
C#
//
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// Copyright (c) 2019-2020 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.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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