400 lines
19 KiB
C#
400 lines
19 KiB
C#
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using System;
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using System.Linq;
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using System.Runtime.InteropServices;
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using UnityEngine;
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namespace E7.Native
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{
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/// <summary>
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/// Several properties about the device asked from the native side that might help you.
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/// Returned from <see cref="NativeAudio.GetDeviceAudioInformation"/>
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///
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/// The content of this `struct` changes completely depending on active build platform.
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/// You will want to use a preprocessor directive wrapping it.
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/// </summary>
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[StructLayout(LayoutKind.Sequential)]
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public struct DeviceAudioInformation
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{
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#if UNITY_IOS
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/// <summary>
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/// It is from [AVAudioSessionPortDescription](https://developer.apple.com/documentation/avfoundation/avaudiosessionportdescription).
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/// </summary>
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public enum IosAudioPortType
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{
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//---Output---
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/// <summary>
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/// Line-level output to the dock connector.
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/// </summary>
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LineOut = 0,
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/// <summary>
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/// Output to a wired headset.
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/// </summary>
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Headphones = 1,
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/// <summary>
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/// Output to a speaker intended to be held near the ear.
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/// </summary>
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BuiltInReceiver = 2,
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/// <summary>
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/// Output to the device’s built-in speaker.
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/// </summary>
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BuiltInSpeaker = 3,
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/// <summary>
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/// Output to a device via the High-Definition Multimedia Interface (HDMI) specification.
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/// </summary>
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HDMI = 4,
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/// <summary>
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/// Output to a remote device over AirPlay.
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/// </summary>
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AirPlay = 5,
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/// <summary>
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/// Output to a Bluetooth Low Energy (LE) peripheral.
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/// </summary>
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BluetoothLE = 6,
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/// <summary>
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/// Output to a Bluetooth A2DP device.
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/// </summary>
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BluetoothA2DP = 7,
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//---Input---
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/// <summary>
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/// Line-level input from the dock connector.
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/// </summary>
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LineIn = 8,
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/// <summary>
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/// The built-in microphone on a device.
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/// </summary>
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BuiltInMic = 9,
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/// <summary>
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/// A microphone that is built-in to a wired headset.
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/// </summary>
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HeadsetMic = 10,
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//---Input-Output---
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/// <summary>
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/// Input or output on a Bluetooth Hands-Free Profile device.
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/// </summary>
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BluetoothHFP = 11,
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/// <summary>
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/// Input or output on a Universal Serial Bus device.
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/// </summary>
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UsbAudio = 12,
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/// <summary>
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/// Input or output via Car Audio.
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/// </summary>
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CarAudio = 13,
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}
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/// <summary>
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/// [iOS] All OUTPUT audio devices currently active.
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/// </summary>
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/// <remarks>
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/// The returned `enum` is native, so on an other platform the available choice completely changes.
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/// Use `#if` directive to make this property compile on multiple platforms.
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///
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/// This is the return value from `[[AVAudioSession sharedInstance] currentRoute]` -> `outputs` -> each item.
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/// </remarks>
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public IosAudioPortType[] audioDevices { get; private set; }
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/// <summary>
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/// [iOS] The latency for audio output, measured in seconds.
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/// </summary>
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/// <remarks>
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/// This is shared with Unity, not just for Native Audio, because `AVAudioSession` is a singleton and is shared.
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///
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/// A result from [this AVAudioSession instance property](https://developer.apple.com/documentation/avfoundation/avaudiosession/1616500-outputlatency).
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///
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/// For reference on my iPhone SE with device speaker, it is 0.0128750000149012.
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/// (Regardless of Project Settings > Audio option selected)
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///
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/// Value is always specified in seconds; it yields sub-millisecond precision over a range of 10,000 years.
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///
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/// I don't know how could iOS know its own latency based on current connected device,
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/// but I got curious and went to audio shop and debug this number on all the things they let me.
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///
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/// Hardware Name String Output Latency Brand
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/// ---------------------------------------------------------
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/// JBL T110BT 0.0825396850705147 JBL
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/// JBL Reflect Mini2 0.0825396850705147 JBL
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/// NW WS-623 0.0149886617437005 Sony Walkman
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/// JBL E45BT 0.111564628779888 JBL
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/// MAJOR III BLUETOOTH 0.149297058582306 Marshall
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/// JBL Flip 4 0.111564628779888 JBL
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/// JBL GO 2 0.0825396850705147 JBL
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/// JBL GO 0.0825396850705147 JBL
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/// JBL JR POP 0.111564628779888 JBL
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/// JBL Charge 4 0.111564628779888 JBL
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///
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/// Looks like the measurement got several suspicious number, like exact same number across devices,
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/// or even some that looks like 10x of the other.
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/// So I think it must be calculated from something rather than measured live with fancy technique.
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///
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/// How is this useful? You could for example, display a warning in the game that your player's device
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/// is not suitable to play the game due to high latency.
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///
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/// However when I test them, most sounds **almost** equal in latency (but all are bad for music games anyways),
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/// but that Marshall MAJOR III has obviously much higher latency (almost 3x) than others.
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/// Feels much higher than what you see in the data.
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///
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/// So it shows this number is somewhat reliable, but may not be 100% accurate of the real latency.
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/// </remarks>
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public double outputLatency { get; private set; }
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/// <summary>
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/// [iOS] The current audio sample rate, in hertz.
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/// </summary>
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/// <remarks>
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/// This is shared with Unity, not just for Native Audio, because `AVAudioSession`
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/// is a singleton and is shared.
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///
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/// A result from [this AVAudioSession instance property](https://developer.apple.com/documentation/avfoundation/avaudiosession/1616499-samplerate).
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///
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/// The available range for hardware sample rate is device dependent. It typically ranges from 8000 through 48000 hertz.
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/// </remarks>
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public double sampleRate { get; private set; }
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/// <summary>
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/// [iOS] The preferred sample rate, in hertz.
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/// </summary>
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/// <remarks>
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/// This is shared with Unity, not just for Native Audio, because `AVAudioSession` is a singleton and is shared.
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///
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/// At native side this is freely specifiable, but iOS might give you something else that becomes <see cref="sampleRate"/>.
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///
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/// A result from [this AVAudioSession instance property](https://developer.apple.com/documentation/avfoundation/avaudiosession/1616543-preferredsamplerate).
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/// </remarks>
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public double preferredSampleRate { get; private set; }
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/// <summary>
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/// [iOS] The current I/O buffer duration, in seconds.
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/// </summary>
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/// <remarks>
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/// A result from [this AVAudioSession instance property](https://developer.apple.com/documentation/avfoundation/avaudiosession/1616498-iobufferduration).
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///
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/// This could be viewed as an another representation of "buffer size" on Android.
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/// This is instead in time unit rather than size. And it depends on the current sample rate in order
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/// to calculate the resulting buffer size.
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///
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/// Value is always specified in seconds; it yields sub-millisecond precision over a range of 10,000 years.
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///
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/// The audio I/O buffer duration is the number of seconds for a single audio input/output cycle.
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/// For example, with an I/O buffer duration of 0.005 s, on each audio I/O cycle:
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///
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/// You receive 0.005 s of audio if obtaining input.
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/// You must provide 0.005 s of audio if providing output.
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///
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/// The typical maximum I/O buffer duration is 0.93 s
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/// (corresponding to 4096 sample frames at a sample rate of 44.1 kHz).
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/// The minimum I/O buffer duration is at least 0.005 s(256 frames)
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/// but might be lower depending on the hardware in use.
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///
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/// For example if this is 0.01s, at sample rate 24000Hz (What Unity use)
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/// it would have to get 0.01s of audio. But compared to 44000Hz rate, that
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/// 0.01s of audio is of much less data. By using higher fidelity 44000Hz,
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/// the same 0.01s could cause buffer underrun if the device is not fast enough.
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///
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/// This is shared with Unity, not just for Native Audio, because `AVAudioSession` is a singleton and is shared.
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///
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/// Here's some behaviour of this number based on my research.
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///
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/// For reference, with varying Project Settings > Audio options :
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/// Best Latency : 0.0106666665524244
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/// Good Latency : 0.0213333331048489
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/// Best Performance : 0.0426666662096977
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///
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/// When connected to external audio device like a bluetooth headphone, the number on Best Latency
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/// drops to 0.005 but sampling rate moved from 24000 to 44100.
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/// This can be interpret as Unity try to make sampling rate compatible with external device,
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/// but now must write half less audio seconds because it now have twice as many data.
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///
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/// By setting <see cref="preferredIOBufferDuration"/> at native side to 0.005
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/// (the limit mentioned in the documentation) this became 0.005333.
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///
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/// When set <see cref="preferredIOBufferDuration"/> back to 0, this became 0.021333. (The same as Good Latency, even though Unity is currently in Best Latency.)
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/// However by benchmarking sometimes 0.005 duration do produce worse latency than 0.01, I wonder why..
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///
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/// </remarks>
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public double ioBufferDuration {get; private set;}
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/// <summary>
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/// [iOS] The preferred I/O buffer duration, in seconds.
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/// </summary>
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/// <remarks>
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/// At native side this is freely specifiable, but iOS might give you something else that becomes <see cref="ioBufferDuration"/>.
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///
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/// This is shared with Unity, not just for Native Audio, because `AVAudioSession` is a singleton and is shared.
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/// This seems to be always 0 in Unity games by default.
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///
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/// A result from [this AVAudioSession instance property](https://developer.apple.com/documentation/avfoundation/avaudiosession/1616464-preferrediobufferduration).
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///
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/// Value is always specified in seconds; it yields sub-millisecond precision over a range of 10,000 years.
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/// </remarks>
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public double preferredIOBufferDuration { get; private set; }
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internal const int interopArrayLength = 5;
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public DeviceAudioInformation(double[] interopDoubleArray, IosAudioPortType[] portArray)
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{
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if (interopDoubleArray.Length != interopArrayLength)
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{
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throw new ArgumentException("The array that fetched data from iOS should be of length " + interopArrayLength);
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}
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this.outputLatency = interopDoubleArray[0];
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this.sampleRate = interopDoubleArray[1];
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this.preferredSampleRate = interopDoubleArray[2];
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this.ioBufferDuration = interopDoubleArray[3];
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this.preferredIOBufferDuration = interopDoubleArray[4];
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this.audioDevices = portArray;
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}
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public override string ToString()
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{
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return string.Format(
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"Audio devices : {0} Output Latency : {1} Sample Rate : {2} Preferred Sample Rate : {3} IO Buffer Duration : {4} Preferred IO Buffer Duration : {5}",
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string.Join(", ", this.audioDevices.Select(x => x.ToString()).ToArray()),
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outputLatency,
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sampleRate,
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preferredSampleRate,
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ioBufferDuration,
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preferredIOBufferDuration
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);
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}
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#endif
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#if UNITY_ANDROID
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/// <summary>
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/// [Android] Only audio matching this sampling rate on a native AudioTrack created with this
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/// sampling rate is eligible for fast track playing.
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/// </summary>
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/// <remarks>
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/// This is NOT the sample rate that Native Audio might be currently working with,
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/// nor what Unity is using for their own audio source.
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/// Just a per-device attribute.
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/// </remarks>
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public int nativeSamplingRate { get; private set; }
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/// <summary>
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/// [Android] How large of a buffer that your phone wants to work with.
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/// </summary>
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/// <remarks>
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/// This is NOT the buffer size that Native Audio might be currently working with,
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/// nor what Unity is using for their own audio source.
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/// Just a per-device attribute.
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/// </remarks>
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public int optimalBufferSize { get; private set; }
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/// <summary>
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/// [Android] Indicates a continuous output latency of 45 ms or less.
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/// Only valid if the device API >= 23 (6.0, Marshmallow), or else it is always `false`.
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/// </summary>
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public bool lowLatencyFeature { get; private set; }
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/// <summary>
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/// [Android] Indicates a continuous round-trip latency of 20 ms or less.
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/// Only valid if the device API >= 23 (6.0, Marshmallow), or else it is always `false`.
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/// </summary>
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public bool proAudioFeature { get; private set; }
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/// <summary>
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/// [Android] All OUTPUT devices currently active.
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/// </summary>
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/// <remarks>
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/// The returned `enum` is native, so on an other platform the available choice completely changes.
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/// Use `#if` directive to make this property compile on multiple platforms.
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///
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/// These are returnd from `audioManager.getDevices(AudioManager.GET_DEVICES_OUTPUTS);` API.
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/// Only valid if the device API >= 23 (6.0, Marshmallow), or else it is always `null`.
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/// </remarks>
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public AndroidAudioDeviceType[] audioDevices { get; private set; }
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/// <summary>
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/// I just copied everything from [AudioDeviceInfo](https://developer.android.com/reference/android/media/AudioDeviceInfo.html#constants_2).
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/// </summary>
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public enum AndroidAudioDeviceType
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{
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TYPE_AUX_LINE = 19,
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TYPE_BLUETOOTH_A2DP = 8,
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TYPE_BLUETOOTH_SCO = 7,
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TYPE_BUILTIN_EARPIECE = 1,
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TYPE_BUILTIN_MIC = 15,
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TYPE_BUILTIN_SPEAKER = 2,
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TYPE_BUS = 21,
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TYPE_DOCK = 13,
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TYPE_FM = 14,
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TYPE_FM_TUNER = 16,
|
|||
|
|
TYPE_HDMI = 9,
|
|||
|
|
TYPE_HDMI_ARC = 10,
|
|||
|
|
TYPE_HEARING_AID = 23,
|
|||
|
|
TYPE_IP = 20,
|
|||
|
|
TYPE_LINE_ANALOG = 5,
|
|||
|
|
TYPE_LINE_DIGITAL = 6,
|
|||
|
|
TYPE_TELEPHONY = 18,
|
|||
|
|
TYPE_TV_TUNER = 17,
|
|||
|
|
TYPE_UNKNOWN = 0,
|
|||
|
|
TYPE_USB_ACCESSORY = 12,
|
|||
|
|
TYPE_USB_DEVICE = 11,
|
|||
|
|
TYPE_USB_HEADSET = 22,
|
|||
|
|
TYPE_WIRED_HEADPHONES = 4,
|
|||
|
|
TYPE_WIRED_HEADSET = 3,
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
public DeviceAudioInformation(AndroidJavaObject jo)
|
|||
|
|
{
|
|||
|
|
AndroidJavaClass versionClass = new AndroidJavaClass("android/os/Build$VERSION");
|
|||
|
|
int sdkLevel = versionClass.GetStatic<int>("SDK_INT");
|
|||
|
|
|
|||
|
|
this.nativeSamplingRate = jo.Get<int>("nativeSamplingRate");
|
|||
|
|
this.optimalBufferSize = jo.Get<int>("optimalBufferSize");
|
|||
|
|
this.lowLatencyFeature = jo.Get<bool>("lowLatencyFeature");
|
|||
|
|
this.proAudioFeature = jo.Get<bool>("proAudioFeature");
|
|||
|
|
|
|||
|
|
if (sdkLevel >= 23)
|
|||
|
|
{
|
|||
|
|
//This one is a Java array, we need to do JNI manually to each elements
|
|||
|
|
AndroidJavaObject outputDevicesJo = jo.Get<AndroidJavaObject>("outputDevices");
|
|||
|
|
|
|||
|
|
IntPtr outputDevicesRaw = outputDevicesJo.GetRawObject();
|
|||
|
|
int outputDeviceAmount = AndroidJNI.GetArrayLength(outputDevicesRaw);
|
|||
|
|
|
|||
|
|
this.audioDevices = new AndroidAudioDeviceType[outputDeviceAmount];
|
|||
|
|
|
|||
|
|
for (int i = 0; i < outputDeviceAmount; i++)
|
|||
|
|
{
|
|||
|
|
IntPtr outputDevice = AndroidJNI.GetObjectArrayElement(outputDevicesRaw, i);
|
|||
|
|
IntPtr audioDeviceInfoClass = AndroidJNI.GetObjectClass(outputDevice);
|
|||
|
|
IntPtr getTypeMethod = AndroidJNIHelper.GetMethodID(audioDeviceInfoClass, "getType");
|
|||
|
|
int type = AndroidJNI.CallIntMethod(outputDevice, getTypeMethod, new jvalue[] { });
|
|||
|
|
this.audioDevices[i] = (AndroidAudioDeviceType)type;
|
|||
|
|
}
|
|||
|
|
}
|
|||
|
|
else
|
|||
|
|
{
|
|||
|
|
this.audioDevices = new AndroidAudioDeviceType[0];
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
//Debug.Log(this.ToString());
|
|||
|
|
}
|
|||
|
|
|
|||
|
|
public override string ToString()
|
|||
|
|
{
|
|||
|
|
return string.Format("Native Sampling Rate: {0} | Optimal Buffer Size: {1} | Low Latency Feature: {2} | Pro Audio Feature: {3} | Output devices : {4}",
|
|||
|
|
nativeSamplingRate, optimalBufferSize, lowLatencyFeature, proAudioFeature,
|
|||
|
|
string.Join(", ", this.audioDevices.Select(x => x.ToString()).ToArray()));
|
|||
|
|
}
|
|||
|
|
#endif
|
|||
|
|
}
|
|||
|
|
}
|