diff options
| author | Mary <me@thog.eu> | 2021-02-26 01:11:56 +0100 |
|---|---|---|
| committer | GitHub <noreply@github.com> | 2021-02-26 01:11:56 +0100 |
| commit | f556c80d0230056335632b60c71f1567e177239e (patch) | |
| tree | 748aa6be62b93a8e941e25dbd83f39e1dbb37035 /Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs | |
| parent | 1c49089ff00fc87dc4872f135dc6a0d36169a970 (diff) | |
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
Diffstat (limited to 'Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs')
| -rw-r--r-- | Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs | 360 |
1 files changed, 0 insertions, 360 deletions
diff --git a/Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs b/Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs deleted file mode 100644 index 9a18a95a..00000000 --- a/Ryujinx.Audio.Renderer/Parameter/VoiceInParameter.cs +++ /dev/null @@ -1,360 +0,0 @@ -// -// 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.Renderer.Common; -using Ryujinx.Audio.Renderer.Dsp; -using Ryujinx.Common.Memory; -using System; -using System.Runtime.CompilerServices; -using System.Runtime.InteropServices; - -namespace Ryujinx.Audio.Renderer.Parameter -{ - /// <summary> - /// Input information for a voice. - /// </summary> - [StructLayout(LayoutKind.Sequential, Size = 0x170, Pack = 1)] - public struct VoiceInParameter - { - /// <summary> - /// Id of the voice. - /// </summary> - public int Id; - - /// <summary> - /// Node id of the voice. - /// </summary> - public int NodeId; - - /// <summary> - /// Set to true if the voice is new. - /// </summary> - [MarshalAs(UnmanagedType.I1)] - public bool IsNew; - - /// <summary> - /// Set to true if the voice is used. - /// </summary> - [MarshalAs(UnmanagedType.I1)] - public bool InUse; - - /// <summary> - /// The voice <see cref="PlayState"/> wanted by the user. - /// </summary> - public PlayState PlayState; - - /// <summary> - /// The <see cref="SampleFormat"/> of the voice. - /// </summary> - public SampleFormat SampleFormat; - - /// <summary> - /// The sample rate of the voice. - /// </summary> - public uint SampleRate; - - /// <summary> - /// The priority of the voice. - /// </summary> - public uint Priority; - - /// <summary> - /// Target sorting position of the voice. (Used to sort voices with the same <see cref="Priority"/>) - /// </summary> - public uint SortingOrder; - - /// <summary> - /// The total channel count used. - /// </summary> - public uint ChannelCount; - - /// <summary> - /// The pitch used on the voice. - /// </summary> - public float Pitch; - - /// <summary> - /// The output volume of the voice. - /// </summary> - public float Volume; - - /// <summary> - /// Biquad filters to apply to the output of the voice. - /// </summary> - public Array2<BiquadFilterParameter> BiquadFilters; - - /// <summary> - /// Total count of <see cref="WaveBufferInternal"/> of the voice. - /// </summary> - public uint WaveBuffersCount; - - /// <summary> - /// Current playing <see cref="WaveBufferInternal"/> of the voice. - /// </summary> - public uint WaveBuffersIndex; - - /// <summary> - /// Reserved/unused. - /// </summary> - private uint _reserved1; - - /// <summary> - /// User state address required by the data source. - /// </summary> - /// <remarks>Only used for <see cref="SampleFormat.Adpcm"/> as the address of the GC-ADPCM coefficients.</remarks> - public ulong DataSourceStateAddress; - - /// <summary> - /// User state size required by the data source. - /// </summary> - /// <remarks>Only used for <see cref="SampleFormat.Adpcm"/> as the size of the GC-ADPCM coefficients.</remarks> - public ulong DataSourceStateSize; - - /// <summary> - /// The target mix id of the voice. - /// </summary> - public int MixId; - - /// <summary> - /// The target splitter id of the voice. - /// </summary> - public uint SplitterId; - - /// <summary> - /// The wavebuffer parameters of this voice. - /// </summary> - public Array4<WaveBufferInternal> WaveBuffers; - - /// <summary> - /// The channel resource ids associated to the voice. - /// </summary> - public Array6<int> ChannelResourceIds; - - /// <summary> - /// Reset the voice drop flag during voice server update. - /// </summary> - [MarshalAs(UnmanagedType.I1)] - public bool ResetVoiceDropFlag; - - /// <summary> - /// Flush the amount of wavebuffer specified. This will result in the wavebuffer being skipped and marked played. - /// </summary> - /// <remarks>This was added on REV5.</remarks> - public byte FlushWaveBufferCount; - - /// <summary> - /// Reserved/unused. - /// </summary> - private ushort _reserved2; - - /// <summary> - /// Change the behaviour of the voice. - /// </summary> - /// <remarks>This was added on REV5.</remarks> - public DecodingBehaviour DecodingBehaviourFlags; - - /// <summary> - /// Change the Sample Rate Conversion (SRC) quality of the voice. - /// </summary> - /// <remarks>This was added on REV8.</remarks> - public SampleRateConversionQuality SrcQuality; - - /// <summary> - /// This was previously used for opus codec support on the Audio Renderer and was removed on REV3. - /// </summary> - public uint ExternalContext; - - /// <summary> - /// This was previously used for opus codec support on the Audio Renderer and was removed on REV3. - /// </summary> - public uint ExternalContextSize; - - /// <summary> - /// Reserved/unused. - /// </summary> - private unsafe fixed uint _reserved3[2]; - - /// <summary> - /// Input information for a voice wavebuffer. - /// </summary> - [StructLayout(LayoutKind.Sequential, Size = 0x38, Pack = 1)] - public struct WaveBufferInternal - { - /// <summary> - /// Address of the wavebuffer data. - /// </summary> - public ulong Address; - - /// <summary> - /// Size of the wavebuffer data. - /// </summary> - public ulong Size; - - /// <summary> - /// Offset of the first sample to play. - /// </summary> - public uint StartSampleOffset; - - /// <summary> - /// Offset of the last sample to play. - /// </summary> - public uint EndSampleOffset; - - /// <summary> - /// If set to true, the wavebuffer will loop when reaching <see cref="EndSampleOffset"/>. - /// </summary> - /// <remarks> - /// 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"/>) - /// </remarks> - [MarshalAs(UnmanagedType.I1)] - public bool ShouldLoop; - - /// <summary> - /// Indicates that this is the last wavebuffer to play of the voice. - /// </summary> - [MarshalAs(UnmanagedType.I1)] - public bool IsEndOfStream; - - /// <summary> - /// Indicates if the server should update its internal state. - /// </summary> - [MarshalAs(UnmanagedType.I1)] - public bool SentToServer; - - /// <summary> - /// Reserved/unused. - /// </summary> - private byte _reserved; - - /// <summary> - /// If set to anything other than 0, specifies how many times to loop the wavebuffer. - /// </summary> - /// <remarks>This was added in REV8.</remarks> - public int LoopCount; - - /// <summary> - /// Address of the context used by the sample decoder. - /// </summary> - /// <remarks>This is only currently used by <see cref="SampleFormat.Adpcm"/>.</remarks> - public ulong ContextAddress; - - /// <summary> - /// Size of the context used by the sample decoder. - /// </summary> - /// <remarks>This is only currently used by <see cref="SampleFormat.Adpcm"/>.</remarks> - public ulong ContextSize; - - /// <summary> - /// If set to anything other than 0, specifies the offset of the first sample to play when looping. - /// </summary> - /// <remarks>This was added in REV8.</remarks> - public uint LoopFirstSampleOffset; - - /// <summary> - /// If set to anything other than 0, specifies the offset of the last sample to play when looping. - /// </summary> - /// <remarks>This was added in REV8.</remarks> - public uint LoopLastSampleOffset; - - /// <summary> - /// Check if the sample offsets are in a valid range for generic PCM. - /// </summary> - /// <typeparam name="T">The PCM sample type</typeparam> - /// <returns>Returns true if the sample offset are in range of the size.</returns> - [MethodImpl(MethodImplOptions.AggressiveInlining)] - private bool IsSampleOffsetInRangeForPcm<T>() where T : unmanaged - { - uint dataTypeSize = (uint)Unsafe.SizeOf<T>(); - - return StartSampleOffset * dataTypeSize <= Size && - EndSampleOffset * dataTypeSize <= Size; - } - - /// <summary> - /// Check if the sample offsets are in a valid range for the given <see cref="SampleFormat"/>. - /// </summary> - /// <param name="format">The target <see cref="SampleFormat"/></param> - /// <returns>Returns true if the sample offset are in range of the size.</returns> - public bool IsSampleOffsetValid(SampleFormat format) - { - bool result; - - switch (format) - { - case SampleFormat.PcmInt16: - result = IsSampleOffsetInRangeForPcm<ushort>(); - break; - case SampleFormat.PcmFloat: - result = IsSampleOffsetInRangeForPcm<float>(); - break; - case SampleFormat.Adpcm: - result = AdpcmHelper.GetAdpcmDataSize((int)StartSampleOffset) <= Size && - AdpcmHelper.GetAdpcmDataSize((int)EndSampleOffset) <= Size; - break; - default: - throw new NotImplementedException($"{format} not implemented!"); - } - - return result; - } - } - - /// <summary> - /// Flag altering the behaviour of wavebuffer decoding. - /// </summary> - [Flags] - public enum DecodingBehaviour : ushort - { - /// <summary> - /// Default decoding behaviour. - /// </summary> - Default = 0, - - /// <summary> - /// Reset the played samples accumulator when looping. - /// </summary> - PlayedSampleCountResetWhenLooping = 1, - - /// <summary> - /// Skip pitch and Sample Rate Conversion (SRC). - /// </summary> - SkipPitchAndSampleRateConversion = 2 - } - - /// <summary> - /// Specify the quality to use during Sample Rate Conversion (SRC) and pitch handling. - /// </summary> - /// <remarks>This was added in REV8.</remarks> - public enum SampleRateConversionQuality : byte - { - /// <summary> - /// Resample interpolating 4 samples per output sample. - /// </summary> - Default, - - /// <summary> - /// Resample interpolating 8 samples per output sample. - /// </summary> - High, - - /// <summary> - /// Resample interpolating 1 samples per output sample. - /// </summary> - Low - } - } -} |
