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* 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
193 lines
6.6 KiB
C#
193 lines
6.6 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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namespace Ryujinx.Audio
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{
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/// <summary>
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/// Define constants used by the audio system.
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/// </summary>
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public static class Constants
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{
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/// <summary>
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/// The default device output name.
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/// </summary>
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public const string DefaultDeviceOutputName = "DeviceOut";
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/// <summary>
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/// The default device input name.
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/// </summary>
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public const string DefaultDeviceInputName = "BuiltInHeadset";
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/// <summary>
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/// The maximum number of channels supported. (6 channels for 5.1 surround)
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/// </summary>
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public const int ChannelCountMax = 6;
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/// <summary>
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/// The maximum number of channels supported per voice.
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/// </summary>
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public const int VoiceChannelCountMax = ChannelCountMax;
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/// <summary>
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/// The maximum count of mix buffer supported per operations (volumes, mix effect, ...)
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/// </summary>
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public const int MixBufferCountMax = 24;
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/// <summary>
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/// The maximum count of wavebuffer per voice.
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/// </summary>
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public const int VoiceWaveBufferCount = 4;
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/// <summary>
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/// The maximum count of biquad filter per voice.
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/// </summary>
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public const int VoiceBiquadFilterCount = 2;
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/// <summary>
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/// The lowest priority that a voice can have.
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/// </summary>
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public const int VoiceLowestPriority = 0xFF;
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/// <summary>
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/// The highest priority that a voice can have.
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/// </summary>
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/// <remarks>Voices with the highest priority will not be dropped if a voice drop needs to occur.</remarks>
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public const int VoiceHighestPriority = 0;
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/// <summary>
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/// Maximum <see cref="Common.BehaviourParameter.ErrorInfo"/> that can be returned by <see cref="Parameter.BehaviourErrorInfoOutStatus"/>.
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/// </summary>
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public const int MaxErrorInfos = 10;
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/// <summary>
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/// Default alignment for buffers.
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/// </summary>
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public const int BufferAlignment = 0x40;
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/// <summary>
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/// Alignment required for the work buffer.
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/// </summary>
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public const int WorkBufferAlignment = 0x1000;
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/// <summary>
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/// Alignment required for every performance metrics frame.
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/// </summary>
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public const int PerformanceMetricsPerFramesSizeAlignment = 0x100;
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/// <summary>
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/// The id of the final mix.
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/// </summary>
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public const int FinalMixId = 0;
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/// <summary>
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/// The id defining an unused mix id.
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/// </summary>
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public const int UnusedMixId = int.MaxValue;
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/// <summary>
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/// The id defining an unused splitter id as a signed integer.
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/// </summary>
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public const int UnusedSplitterIdInt = -1;
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/// <summary>
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/// The id defining an unused splitter id.
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/// </summary>
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public const uint UnusedSplitterId = uint.MaxValue;
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/// <summary>
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/// The id of invalid/unused node id.
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/// </summary>
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public const int InvalidNodeId = -268435456;
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/// <summary>
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/// The indice considered invalid for processing order.
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/// </summary>
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public const int InvalidProcessingOrder = -1;
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/// <summary>
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/// The maximum number of audio renderer sessions allowed to be created system wide.
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/// </summary>
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public const int AudioRendererSessionCountMax = 2;
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/// <summary>
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/// The maximum number of audio output sessions allowed to be created system wide.
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/// </summary>
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public const int AudioOutSessionCountMax = 12;
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/// <summary>
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/// The maximum number of audio input sessions allowed to be created system wide.
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/// </summary>
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public const int AudioInSessionCountMax = 4;
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/// <summary>
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/// Maximum buffers supported by one audio device session.
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/// </summary>
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public const int AudioDeviceBufferCountMax = 32;
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/// <summary>
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/// The target sample rate of the audio renderer. (48kHz)
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/// </summary>
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public const uint TargetSampleRate = 48000;
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/// <summary>
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/// The target sample size of the audio renderer. (PCM16)
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/// </summary>
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public const int TargetSampleSize = sizeof(ushort);
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/// <summary>
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/// The target sample count per audio renderer update.
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/// </summary>
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public const int TargetSampleCount = 240;
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/// <summary>
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/// The size of an upsampler entry to process upsampling to <see cref="TargetSampleRate"/>.
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/// </summary>
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public const int UpSampleEntrySize = TargetSampleCount * VoiceChannelCountMax;
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/// <summary>
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/// The target audio latency computed from <see cref="TargetSampleRate"/> and <see cref="TargetSampleCount"/>.
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/// </summary>
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public const int AudioProcessorMaxUpdateTimeTarget = 1000000000 / ((int)TargetSampleRate / TargetSampleCount); // 5.00 ms
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/// <summary>
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/// The maximum update time of the DSP on original hardware.
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/// </summary>
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public const int AudioProcessorMaxUpdateTime = 5760000; // 5.76 ms
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/// <summary>
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/// The maximum update time per audio renderer session.
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/// </summary>
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public const int AudioProcessorMaxUpdateTimePerSessions = AudioProcessorMaxUpdateTime / AudioRendererSessionCountMax;
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/// <summary>
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/// Guest timer frequency used for system ticks.
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/// </summary>
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public const int TargetTimerFrequency = 19200000;
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/// <summary>
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/// The default coefficients used for standard 5.1 surround to stereo downmixing.
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/// </summary>
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public static float[] DefaultSurroundToStereoCoefficients = new float[4]
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{
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1.0f,
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0.707f,
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0.251f,
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0.707f,
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};
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}
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}
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