Merge pull request #6416 from kencooke/audio-reverb

New reverb

examples/utilities/tools/reverbTest.js

@@ -0,0 +1,69 @@
+//
+//  reverbTest.js
+//  examples
+//
+//  Created by Ken Cooke on 11/23/2015.
+//  Copyright 2015 High Fidelity, Inc.
+//
+//  Distributed under the Apache License, Version 2.0.
+//  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
+//
+
+Script.include("cookies.js");
+
+var audioOptions = new AudioEffectOptions({
+    maxRoomSize: 50,
+    roomSize: 50,
+    reverbTime: 4,
+    damping: 0.50,
+    inputBandwidth: 0.8,
+    earlyLevel: 0,
+    tailLevel: 0,
+    dryLevel: -6,
+    wetLevel: -6
+});
+
+AudioDevice.setReverbOptions(audioOptions);
+AudioDevice.setReverb(true);
+print("Reverb is ON.");
+
+var panel = new Panel(10, 200);
+
+var parameters = [
+    { name: "roomSize", min: 0, max: 100, units: " feet" },
+    { name: "reverbTime", min: 0, max: 10, units: " sec" },
+    { name: "damping", min: 0, max: 1, units: " " },
+    { name: "inputBandwidth", min: 0, max: 1, units: " " },
+    { name: "earlyLevel", min: -48, max: 0, units: " dB" },
+    { name: "tailLevel", min: -48, max: 0, units: " dB" },
+    { name: "wetLevel", min: -48, max: 0, units: " dB" },    
+]
+
+function setter(name) {
+    return function(value) { audioOptions[name] = value; AudioDevice.setReverbOptions(audioOptions); }
+}
+
+function getter(name) {
+    return function() { return audioOptions[name]; }
+}
+
+function displayer(units) {
+    return function(value) { return (value).toFixed(1) + units; };
+}
+
+// create a slider for each parameter
+for (var i = 0; i < parameters.length; i++) {
+    var p = parameters[i];
+    panel.newSlider(p.name, p.min, p.max, setter(p.name), getter(p.name), displayer(p.units));
+}
+
+Controller.mouseMoveEvent.connect(function panelMouseMoveEvent(event) { return panel.mouseMoveEvent(event); });
+Controller.mousePressEvent.connect( function panelMousePressEvent(event) { return panel.mousePressEvent(event); });
+Controller.mouseReleaseEvent.connect(function(event) { return panel.mouseReleaseEvent(event); });
+
+function scriptEnding() {
+    panel.destroy();
+    AudioDevice.setReverb(false);
+    print("Reverb is OFF.");
+}
+Script.scriptEnding.connect(scriptEnding);

libraries/audio-client/src/AudioClient.cpp

@@ -33,29 +33,6 @@
 #include <QtMultimedia/QAudioInput>
 #include <QtMultimedia/QAudioOutput>
 
-#if defined(__GNUC__) && !defined(__clang__)
-#pragma GCC diagnostic push
-#pragma GCC diagnostic ignored "-Wdouble-promotion"
-#endif
-
-#ifdef WIN32
-#pragma warning (push)
-#pragma warning (disable: 4273 4305)
-#endif
-
-extern "C" {
-    #include <gverb/gverb.h>
-    #include <gverb/gverbdsp.h>
-}
-
-#ifdef WIN32
-#pragma warning (pop)
-#endif
-
-#if defined(__GNUC__) && !defined(__clang__)
-#pragma GCC diagnostic pop
-#endif
-
 #include <NodeList.h>
 #include <udt/PacketHeaders.h>
 #include <PositionalAudioStream.h>
@@ -120,7 +97,6 @@ AudioClient::AudioClient() :
     _audioSourceInjectEnabled(false),
     _reverb(false),
     _reverbOptions(&_scriptReverbOptions),
-    _gverb(NULL),
     _inputToNetworkResampler(NULL),
     _networkToOutputResampler(NULL),
     _loopbackResampler(NULL),
@@ -145,9 +121,7 @@ AudioClient::AudioClient() :
     connect(updateTimer, &QTimer::timeout, this, &AudioClient::checkDevices);
     updateTimer->start(DEVICE_CHECK_INTERVAL_MSECS);
 
-    // create GVerb filter
+    configureReverb();
-    _gverb = createGverbFilter();
-    configureGverbFilter(_gverb);
 
     auto& packetReceiver = DependencyManager::get<NodeList>()->getPacketReceiver();
     packetReceiver.registerListener(PacketType::AudioStreamStats, &_stats, "processStreamStatsPacket");
@@ -160,10 +134,6 @@ AudioClient::AudioClient() :
 
 AudioClient::~AudioClient() {
     stop();
-
-    if (_gverb) {
-        gverb_free(_gverb);
-    }
 }
 
 void AudioClient::reset() {
@@ -173,8 +143,8 @@ void AudioClient::reset() {
     _toneSource.reset();
     _sourceGain.reset();
     _inputGain.reset();
-
+    _sourceReverb.reset();
-    gverb_flush(_gverb);
+    _listenerReverb.reset();
 }
 
 void AudioClient::audioMixerKilled() {
@@ -569,27 +539,32 @@ bool AudioClient::switchOutputToAudioDevice(const QString& outputDeviceName) {
     return switchOutputToAudioDevice(getNamedAudioDeviceForMode(QAudio::AudioOutput, outputDeviceName));
 }
 
-ty_gverb* AudioClient::createGverbFilter() {
+void AudioClient::configureReverb() {
-    // Initialize a new gverb instance
+    ReverbParameters p;
-    ty_gverb* filter = gverb_new(_outputFormat.sampleRate(), _reverbOptions->getMaxRoomSize(), _reverbOptions->getRoomSize(),
+    _listenerReverb.getParameters(&p);
-                                 _reverbOptions->getReverbTime(), _reverbOptions->getDamping(), _reverbOptions->getSpread(),
-                                 _reverbOptions->getInputBandwidth(), _reverbOptions->getEarlyLevel(),
-                                 _reverbOptions->getTailLevel());
 
-    return filter;
+    // for now, reuse the gverb parameters
+    p.sampleRate = _outputFormat.sampleRate();
+    p.roomSize = _reverbOptions->getRoomSize();
+    p.reverbTime = _reverbOptions->getReverbTime();
+    p.highGain = -24.0f * (1.0f - _reverbOptions->getDamping());
+    p.bandwidth = 10000.0f * _reverbOptions->getInputBandwidth();
+    p.earlyGain = _reverbOptions->getEarlyLevel();
+    p.lateGain = _reverbOptions->getTailLevel();
+    p.wetDryMix = 100.0f * powf(10.0f, _reverbOptions->getWetLevel() * (1/20.0f));
+    _listenerReverb.setParameters(&p);
+
+    // used for adding self-reverb to loopback audio
+    p.wetDryMix = 100.0f;
+    p.preDelay = 0.0f;
+    p.earlyGain = -96.0f;   // disable ER
+    p.lateGain -= 12.0f;     // quieter than listener reverb
+    p.lateMixLeft = 0.0f;
+    p.lateMixRight = 0.0f;
+    _sourceReverb.setParameters(&p);
 }
 
-void AudioClient::configureGverbFilter(ty_gverb* filter) {
+void AudioClient::updateReverbOptions() {
-    // Configure the instance (these functions are not super well named - they actually set several internal variables)
-    gverb_set_roomsize(filter, _reverbOptions->getRoomSize());
-    gverb_set_revtime(filter, _reverbOptions->getReverbTime());
-    gverb_set_damping(filter, _reverbOptions->getDamping());
-    gverb_set_inputbandwidth(filter, _reverbOptions->getInputBandwidth());
-    gverb_set_earlylevel(filter, DB_CO(_reverbOptions->getEarlyLevel()));
-    gverb_set_taillevel(filter, DB_CO(_reverbOptions->getTailLevel()));
-}
-
-void AudioClient::updateGverbOptions() {
     bool reverbChanged = false;
     if (_receivedAudioStream.hasReverb()) {
 
@@ -599,7 +574,7 @@ void AudioClient::updateGverbOptions() {
         }
         if (_zoneReverbOptions.getWetLevel() != _receivedAudioStream.getWetLevel()) {
             _zoneReverbOptions.setWetLevel(_receivedAudioStream.getWetLevel());
-            // Not part of actual filter config, no need to set reverbChanged to true
+            reverbChanged = true;
         }
 
         if (_reverbOptions != &_zoneReverbOptions) {
@@ -612,9 +587,7 @@ void AudioClient::updateGverbOptions() {
     }
 
     if (reverbChanged) {
-        gverb_free(_gverb);
+        configureReverb();
-        _gverb = createGverbFilter();
-        configureGverbFilter(_gverb);
     }
 }
 
@@ -622,7 +595,8 @@ void AudioClient::setReverb(bool reverb) {
     _reverb = reverb;
 
     if (!_reverb) {
-        gverb_flush(_gverb);
+        _sourceReverb.reset();
+        _listenerReverb.reset();
     }
 }
 
@@ -642,47 +616,7 @@ void AudioClient::setReverbOptions(const AudioEffectOptions* options) {
 
     if (_reverbOptions == &_scriptReverbOptions) {
         // Apply them to the reverb instances
-        gverb_free(_gverb);
+        configureReverb();
-        _gverb = createGverbFilter();
-        configureGverbFilter(_gverb);
-    }
-}
-
-void AudioClient::addReverb(ty_gverb* gverb, int16_t* samplesData, int16_t* reverbAlone, int numSamples,
-                            QAudioFormat& audioFormat, bool noEcho) {
-    float wetFraction = DB_CO(_reverbOptions->getWetLevel());
-    float dryFraction = 1.0f - wetFraction;
-
-    float lValue,rValue;
-    for (int sample = 0; sample < numSamples; sample += audioFormat.channelCount()) {
-        // Run GVerb
-        float value = (float)samplesData[sample];
-        gverb_do(gverb, value, &lValue, &rValue);
-
-        // Mix, accounting for clipping, the left and right channels. Ignore the rest.
-        for (int j = sample; j < sample + audioFormat.channelCount(); j++) {
-            if (j == sample) {
-                // left channel
-                int lResult = glm::clamp((int)(samplesData[j] * dryFraction + lValue * wetFraction),
-                                         AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
-                samplesData[j] = (int16_t)lResult;
-
-                if (noEcho) {
-                    reverbAlone[j] = (int16_t)lValue * wetFraction;
-                }
-            } else if (j == (sample + 1)) {
-                // right channel
-                int rResult = glm::clamp((int)(samplesData[j] * dryFraction + rValue * wetFraction),
-                                         AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
-                samplesData[j] = (int16_t)rResult;
-
-                if (noEcho) {
-                    reverbAlone[j] = (int16_t)rValue * wetFraction;
-                }
-            } else {
-                // ignore channels above 2
-            }
-        }
     }
 }
 
@@ -716,30 +650,28 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
         _loopbackResampler = new AudioSRC(_inputFormat.sampleRate(), _outputFormat.sampleRate(), channelCount);
     }
 
-    static QByteArray reverbAlone;  // Intermediary for local reverb with no echo
     static QByteArray loopBackByteArray;
 
     int numInputSamples = inputByteArray.size() / sizeof(int16_t);
     int numLoopbackSamples = numDestinationSamplesRequired(_inputFormat, _outputFormat, numInputSamples);
 
-    reverbAlone.resize(numInputSamples * sizeof(int16_t));
     loopBackByteArray.resize(numLoopbackSamples * sizeof(int16_t));
 
     int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
-    int16_t* reverbAloneSamples = reinterpret_cast<int16_t*>(reverbAlone.data());
     int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
 
-    if (hasReverb) {
-        updateGverbOptions();
-        addReverb(_gverb, inputSamples, reverbAloneSamples, numInputSamples,
-                  _inputFormat, !_shouldEchoLocally);
-    }
-
     possibleResampling(_loopbackResampler,
-                       (_shouldEchoLocally) ? inputSamples : reverbAloneSamples, loopbackSamples,
+                       inputSamples, loopbackSamples,
                        numInputSamples, numLoopbackSamples,
                        _inputFormat, _outputFormat);
 
+    // apply stereo reverb at the source, to the loopback audio
+    if (!_shouldEchoLocally && hasReverb) {
+        assert(_outputFormat.channelCount() == 2);
+        updateReverbOptions();
+        _sourceReverb.render(loopbackSamples, loopbackSamples, numLoopbackSamples/2);
+    }
+
     _loopbackOutputDevice->write(loopBackByteArray);
 }
 
@@ -871,12 +803,20 @@ void AudioClient::processReceivedSamples(const QByteArray& inputBuffer, QByteArr
     outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
 
     const int16_t* receivedSamples = reinterpret_cast<const int16_t*>(inputBuffer.data());
+    int16_t* outputSamples = reinterpret_cast<int16_t*>(outputBuffer.data());
 
     // copy the packet from the RB to the output
-    possibleResampling(_networkToOutputResampler, receivedSamples,
+    possibleResampling(_networkToOutputResampler, receivedSamples, outputSamples,
-                       reinterpret_cast<int16_t*>(outputBuffer.data()),
                        numNetworkOutputSamples, numDeviceOutputSamples,
                        _desiredOutputFormat, _outputFormat);
+
+    // apply stereo reverb at the listener, to the received audio
+    bool hasReverb = _reverb || _receivedAudioStream.hasReverb();
+    if (hasReverb) {
+        assert(_outputFormat.channelCount() == 2);
+        updateReverbOptions();
+        _listenerReverb.render(outputSamples, outputSamples, numDeviceOutputSamples/2);
+    }
 }
 
 void AudioClient::sendMuteEnvironmentPacket() {

libraries/audio-client/src/AudioClient.h

@@ -46,6 +46,7 @@
 #include "AudioIOStats.h"
 #include "AudioNoiseGate.h"
 #include "AudioSRC.h"
+#include "AudioReverb.h"
 
 #ifdef _WIN32
 #pragma warning( push )
@@ -75,8 +76,6 @@ class QAudioOutput;
 class QIODevice;
 
 
-typedef struct ty_gverb ty_gverb;
-
 class Transform;
 class NLPacket;
 
@@ -263,7 +262,8 @@ private:
     AudioEffectOptions _scriptReverbOptions;
     AudioEffectOptions _zoneReverbOptions;
     AudioEffectOptions* _reverbOptions;
-    ty_gverb* _gverb;
+    AudioReverb _sourceReverb { AudioConstants::SAMPLE_RATE };
+    AudioReverb _listenerReverb { AudioConstants::SAMPLE_RATE };
 
     // possible streams needed for resample
     AudioSRC* _inputToNetworkResampler;
@@ -271,10 +271,8 @@ private:
     AudioSRC* _loopbackResampler;
 
     // Adds Reverb
-    ty_gverb* createGverbFilter();
+    void configureReverb();
-    void configureGverbFilter(ty_gverb* filter);
+    void updateReverbOptions();
-    void updateGverbOptions();
-    void addReverb(ty_gverb* gverb, int16_t* samples, int16_t* reverbAlone, int numSamples, QAudioFormat& format, bool noEcho = false);
 
     void handleLocalEchoAndReverb(QByteArray& inputByteArray);
 

libraries/audio/src/AudioEffectOptions.cpp

@@ -28,10 +28,10 @@ AudioEffectOptions::AudioEffectOptions(QScriptValue arguments) :
     _damping(0.5f),
     _spread(15.0f),
     _inputBandwidth(0.75f),
-    _earlyLevel(-22.0f),
+    _earlyLevel(-12.0f),
-    _tailLevel(-28.0f),
+    _tailLevel(-18.0f),
     _dryLevel(0.0f),
-    _wetLevel(6.0f) {
+    _wetLevel(0.0f) {
         if (arguments.property(MAX_ROOM_SIZE_HANDLE).isNumber()) {
             _maxRoomSize = arguments.property(MAX_ROOM_SIZE_HANDLE).toNumber();
         }

libraries/audio/src/AudioEffectOptions.h

@@ -15,6 +15,8 @@
 #include <QtScript/QScriptContext>
 #include <QtScript/QScriptEngine>
 
+#include "AudioReverb.h"
+
 class AudioEffectOptions : public QObject {
     Q_OBJECT
 

libraries/audio/src/AudioReverb.h

@@ -0,0 +1,76 @@
+//
+//  AudioReverb.h
+//  libraries/audio/src
+//
+//  Created by Ken Cooke on 10/11/15.
+//  Copyright 2015 High Fidelity, Inc.
+//
+
+#ifndef hifi_AudioReverb_h
+#define hifi_AudioReverb_h
+
+#include <stdint.h>
+
+typedef struct ReverbParameters {
+
+    float sampleRate;       // [24000, 48000] Hz
+    float bandwidth;        // [20, 24000] Hz
+
+    float preDelay;         // [0, 333] ms
+    float lateDelay;        // [0, 166] ms
+
+    float reverbTime;       // [0.1, 100] seconds
+
+    float earlyDiffusion;   // [0, 100] percent
+    float lateDiffusion;    // [0, 100] percent
+
+    float roomSize;         // [0, 100] percent
+    float density;          // [0, 100] percent
+
+    float bassMult;         // [0.1, 10] ratio
+    float bassFreq;         // [10, 500] Hz
+    float highGain;         // [-24, 0] dB
+    float highFreq;         // [1000, 12000] Hz
+
+    float modRate;          // [0.1, 10] Hz
+    float modDepth;         // [0, 100] percent
+
+    float earlyGain;        // [-96, +24] dB
+    float lateGain;         // [-96, +24] dB
+
+    float earlyMixLeft;     // [0, 100] percent
+    float earlyMixRight;    // [0, 100] percent
+    float lateMixLeft;      // [0, 100] percent
+    float lateMixRight;     // [0, 100] percent
+
+    float wetDryMix;        // [0, 100] percent
+
+} ReverbParameters;
+
+class ReverbImpl;
+
+class AudioReverb {
+public:
+    AudioReverb(float sampleRate);
+    ~AudioReverb();
+
+    void setParameters(ReverbParameters *p);
+    void getParameters(ReverbParameters *p);
+    void reset();
+
+    // deinterleaved float input/output (native format)
+    void render(float** inputs, float** outputs, int numFrames);
+
+    // interleaved int16_t input/output
+    void render(const int16_t* input, int16_t* output, int numFrames);
+
+private:
+    ReverbImpl *_impl;
+    ReverbParameters _params;
+
+    float* _inout[2];
+    void convertInputFromInt16(const int16_t* input, float** outputs, int numFrames);
+    void convertOutputToInt16(float** inputs, int16_t* output, int numFrames);
+};
+
+#endif // hifi_AudioReverb_h

libraries/audio/src/AudioSRC.cpp

@@ -1218,7 +1218,7 @@ static inline float dither() {
     rz = rz * 69069 + 1;
     int32_t r0 = rz & 0xffff;
     int32_t r1 = rz >> 16;
-    return (r0 - r1) * (1/65536.0f);
+    return (int32_t)(r0 - r1) * (1/65536.0f);
 }
 
 // convert float to int16_t, interleave stereo

libraries/audio/src/AudioSRC.h

@@ -12,7 +12,7 @@
 #ifndef hifi_AudioSRC_h
 #define hifi_AudioSRC_h
 
-#include "stdint.h"
+#include <stdint.h>
 
 class AudioSRC {