holopy3/Assets/Normal/Realtime/Audio/AudioOutput.cs

using UnityEngine;
using Normal.Realtime.Native;
using Normal.Utility;

namespace Normal.Realtime {
    public class AudioOutput : MonoBehaviour {
        private volatile bool _mute;
        public           bool  mute { get { return _mute; } set { _mute = mute; } }
        public  volatile float _dbLevel = -42.0f;
        public           float  dbLevel { get { return _dbLevel; } }

        private int               _systemSampleRate;
        private AudioOutputStream _audioOutputStream;

        void Awake() {
            _systemSampleRate = AudioSettings.outputSampleRate;
        }

        void OnDestroy() {
            Stop();
        }

        void OnEnable() {
            if (_audioOutputStream != null) {
                AudioSource audioSource = GetComponent<AudioSource>();
                if (audioSource != null) {
                    if (!audioSource.isPlaying)
                        audioSource.Play();
                } else {
                    Debug.LogError("Realtime: AudioOutput has audio output stream, but no AudioSource. Was it destroyed on accident?");
                }
            }
        }

        public void StartWithAudioOutputStream(AudioOutputStream audioOutputStream) {
            _audioOutputStream = audioOutputStream;

            // Speaker
            AudioSource audioSource = GetComponent<AudioSource>();

            // Create audio source if needed.
            if (audioSource == null) {
                audioSource = gameObject.AddComponent<AudioSource>();
                audioSource.spatialize = true;
                audioSource.spatialBlend = 1.0f;
            }

            // TODO: Do we want AudioClip's sample rate to match OPUS? That means Unity is left with doing any resampling. We might be able to do the resampling better ourselves.
            // TODO: We can probably specify a shorter clip length here since it's autogenerated now.
            audioSource.enabled = true;
            audioSource.loop = true;
            audioSource.clip = AudioClip.Create("Normcore Audio Stream", 48000, 1, 48000, true, (float[] data) => { for (int i = 0; i < data.Length; i++) data[i] = 1.0f; });
            audioSource.pitch = 1.0f;
            audioSource.spatializePostEffects = true;
            audioSource.Play();
        }

        public void Stop() {
            AudioSource audioSource = GetComponent<AudioSource>();
            if (audioSource != null) {
                audioSource.Stop();
            }

            _audioOutputStream = null;
        }

        void OnAudioFilterRead(float[] data, int channels) {
            if (_audioOutputStream == null || _audioOutputStream.nativePointerIsNull) {

                // Zero the data back out.
                for (int i = 0; i < data.Length; i++)
                    data[i] = 0.0f;

                // Zero db level
                _dbLevel = -42.0f;

                // Bail
                return;
            }

            // Configure the AudioOutputStream to resample to our desired sample rate
            _audioOutputStream.SetSampleRate(_systemSampleRate);
            
            int incomingNumberOfChannels = _audioOutputStream.Channels();
            int numberOfFramesNeeded     = data.Length / channels;

            int numberOfIncomingSamplesNeeded = numberOfFramesNeeded * incomingNumberOfChannels;
            float[] audioData = new float[numberOfIncomingSamplesNeeded];
            if (_audioOutputStream.GetAudioData(audioData)) {
                // Mix incoming audio data into buffer buffer
                for (int f = 0; f < numberOfFramesNeeded; f++) {
                    for (int c = 0; c < channels; c++) {
                        int cIn = c;
                        if (cIn >= incomingNumberOfChannels)
                            cIn = incomingNumberOfChannels-1;
                        int sIn  = f*incomingNumberOfChannels + cIn;
                        int sOut = f*channels + c;

                        // TODO: If there's no spatializer, we need to do this, but if there is a spatializer, we can just copy the value.
                        // TODO: Why is the input signal we're getting not 1.0 when spatialization is turned off??
                        data[sOut] = !_mute ? audioData[sIn] : 0.0f;
                    }
                }

                // Calculate db level using the last 256 frames
                int firstFrame = numberOfFramesNeeded - 256;
                if (firstFrame < 0)
                    firstFrame = 0;
                int firstSample = firstFrame * incomingNumberOfChannels;
                _dbLevel = StaticFunctions.CalculateAverageDbForAudioBuffer(audioData, firstSample);
            } else {
                // Failed to retrieve audio samples. zero the data back out.
                // TODO: Maybe we should fade in/out here? Maybe the native interface can do that for us?
                for (int i = 0; i < data.Length; i++)
                    data[i] = 0.0f;

                // Zero db level
                _dbLevel = -42.0f;
            }
        }
    }
}