Mirrors/overte
3
0
Fork 0
mirror of https://github.com/overte-org/overte.git synced 2025-04-21 19:04:32 +02:00
Code Issues Projects Releases Packages Wiki Activity Actions 2

Merge pull request #3368 from chansensturm/audio-noise

Browse source 
New Audio Buffer, Format, Tone Generator, Pink Noise Generator, and Gain objects
This commit is contained in:
Brad Hefta-Gaub 2014-09-08 11:28:02 -07:00
commit 8068117f0c
15 changed files with 687 additions and 42 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

82
interface/src/Audio.cpp
View file

@ -40,6 +40,7 @@
#include <glm/glm.hpp>
#include "Audio.h"
#include "Menu.h"
#include "Util.h"
#include "PositionalAudioStream.h"
@ -82,7 +83,7 @@ Audio::Audio(QObject* parent) :
_noiseGateSampleCounter(0),
_noiseGateOpen(false),
_noiseGateEnabled(true),
_toneInjectionEnabled(false),
_audioSourceInjectEnabled(false),
_noiseGateFramesToClose(0),
_totalInputAudioSamples(0),
_collisionSoundMagnitude(0.0f),
@ -102,6 +103,8 @@ Audio::Audio(QObject* parent) :
_framesPerScope(DEFAULT_FRAMES_PER_SCOPE),
_samplesPerScope(NETWORK_SAMPLES_PER_FRAME * _framesPerScope),
_peqEnabled(false),
_noiseSourceEnabled(false),
_toneSourceEnabled(true),
_scopeInput(0),
_scopeOutputLeft(0),
_scopeOutputRight(0),
@ -137,6 +140,10 @@ void Audio::reset() {
_receivedAudioStream.reset();
resetStats();
_peq.reset();
_noiseSource.reset();
_toneSource.reset();
_sourceGain.reset();
_inputGain.reset();
}
void Audio::resetStats() {
@ -424,14 +431,25 @@ void Audio::start() {
qDebug() << "Unable to set up audio output because of a problem with output format.";
}
_peq.initialize( _inputFormat.sampleRate(), _audioInput->bufferSize() );
_inputFrameBuffer.initialize( _inputFormat.channelCount(), _audioInput->bufferSize() * 2 );
_peq.initialize( _inputFormat.sampleRate() );
_inputGain.initialize();
_sourceGain.initialize();
_noiseSource.initialize();
_toneSource.initialize();
_sourceGain.setParameters(0.25f,0.0f);
_inputGain.setParameters(1.0f,0.0f);
}
void Audio::stop() {
_inputFrameBuffer.finalize();
_peq.finalize();
_inputGain.finalize();
_sourceGain.finalize();
_noiseSource.finalize();
_toneSource.finalize();
// "switch" to invalid devices in order to shut down the state
switchInputToAudioDevice(QAudioDeviceInfo());
switchOutputToAudioDevice(QAudioDeviceInfo());
@ -477,14 +495,30 @@ void Audio::handleAudioInput() {
QByteArray inputByteArray = _inputDevice->readAll();
int16_t* inputFrameData = (int16_t*)inputByteArray.data();
const int inputFrameCount = inputByteArray.size() / sizeof(int16_t);
_inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, false /*copy in*/);
_inputGain.render(_inputFrameBuffer); // input/mic gain+mute
// Add audio source injection if enabled
if (_audioSourceInjectEnabled && !_muted) {
if (_toneSourceEnabled) { // sine generator
_toneSource.render(_inputFrameBuffer);
}
else if(_noiseSourceEnabled) { // pink noise generator
_noiseSource.render(_inputFrameBuffer);
}
_sourceGain.render(_inputFrameBuffer); // post gain
}
if (_peqEnabled && !_muted) {
// we wish to pre-filter our captured input, prior to loopback
int16_t* ioBuffer = (int16_t*)inputByteArray.data();
_peq.render(ioBuffer, ioBuffer, inputByteArray.size() / sizeof(int16_t));
_peq.render(_inputFrameBuffer); // 3-band parametric eq
}
_inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, true /*copy out*/);
if (Menu::getInstance()->isOptionChecked(MenuOption::EchoLocalAudio) && !_muted && _audioOutput) {
// if this person wants local loopback add that to the locally injected audio
@ -522,7 +556,7 @@ void Audio::handleAudioInput() {
int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
const int numNetworkBytes = _isStereoInput ? NETWORK_BUFFER_LENGTH_BYTES_STEREO : NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL;
const int numNetworkSamples = _isStereoInput ? NETWORK_BUFFER_LENGTH_SAMPLES_STEREO : NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
@ -599,20 +633,8 @@ void Audio::handleAudioInput() {
_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.0f - DC_OFFSET_AVERAGING) * measuredDcOffset;
}
// Add tone injection if enabled
const float TONE_FREQ = 220.0f / SAMPLE_RATE * TWO_PI;
const float QUARTER_VOLUME = 8192.0f;
if (_toneInjectionEnabled) {
loudness = 0.0f;
for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
networkAudioSamples[i] = QUARTER_VOLUME * sinf(TONE_FREQ * (float)(i + _proceduralEffectSample));
loudness += fabsf(networkAudioSamples[i]);
}
}
_lastInputLoudness = fabs(loudness / NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
// If Noise Gate is enabled, check and turn the gate on and off
if (!_toneInjectionEnabled && _noiseGateEnabled) {
if (!_audioSourceInjectEnabled && _noiseGateEnabled) {
float averageOfAllSampleFrames = 0.0f;
_noiseSampleFrames[_noiseGateSampleCounter++] = _lastInputLoudness;
if (_noiseGateSampleCounter == NUMBER_OF_NOISE_SAMPLE_FRAMES) {
@ -1041,8 +1063,18 @@ void Audio::processProceduralAudio(int16_t* monoInput, int numSamples) {
}
}
void Audio::toggleToneInjection() {
_toneInjectionEnabled = !_toneInjectionEnabled;
void Audio::toggleAudioSourceInject() {
_audioSourceInjectEnabled = !_audioSourceInjectEnabled;
}
void Audio::selectAudioSourcePinkNoise() {
_noiseSourceEnabled = Menu::getInstance()->isOptionChecked(MenuOption::AudioSourcePinkNoise);
_toneSourceEnabled = !_noiseSourceEnabled;
}
void Audio::selectAudioSourceSine440() {
_toneSourceEnabled = Menu::getInstance()->isOptionChecked(MenuOption::AudioSourceSine440);
_noiseSourceEnabled = !_toneSourceEnabled;
}
void Audio::toggleAudioSpatialProcessing() {

36
interface/src/Audio.h
View file

@ -20,6 +20,12 @@
#include "Recorder.h"
#include "RingBufferHistory.h"
#include "MovingMinMaxAvg.h"
#include "AudioRingBuffer.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioSourceTone.h"
#include "AudioSourceNoise.h"
#include "AudioGain.h"
#include "AudioFilter.h"
#include "AudioFilterBank.h"
@ -116,7 +122,9 @@ public slots:
void audioMixerKilled();
void toggleMute();
void toggleAudioNoiseReduction();
void toggleToneInjection();
void toggleAudioSourceInject();
void selectAudioSourcePinkNoise();
void selectAudioSourceSine440();
void toggleScope();
void toggleScopePause();
void toggleStats();
@ -199,7 +207,8 @@ private:
int _noiseGateSampleCounter;
bool _noiseGateOpen;
bool _noiseGateEnabled;
bool _toneInjectionEnabled;
bool _audioSourceInjectEnabled;
int _noiseGateFramesToClose;
int _totalInputAudioSamples;
@ -282,10 +291,27 @@ private:
int _framesPerScope;
int _samplesPerScope;
// Multi-band parametric EQ
bool _peqEnabled;
AudioFilterPEQ3m _peq;
// Input framebuffer
AudioBufferFloat32 _inputFrameBuffer;
// Input gain
AudioGain _inputGain;
// Post tone/pink noise generator gain
AudioGain _sourceGain;
// Pink noise source
bool _noiseSourceEnabled;
AudioSourcePinkNoise _noiseSource;
// Tone source
bool _toneSourceEnabled;
AudioSourceTone _toneSource;
// Multi-band parametric EQ
bool _peqEnabled;
AudioFilterPEQ3m _peq;
QMutex _guard;
QByteArray* _scopeInput;
QByteArray* _scopeOutputLeft;

24
interface/src/Menu.cpp
View file

@ -545,11 +545,31 @@ Menu::Menu() :
0,
this,
SLOT(muteEnvironment()));
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::AudioToneInjection,
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::AudioSourceInject,
0,
false,
appInstance->getAudio(),
SLOT(toggleToneInjection()));
SLOT(toggleAudioSourceInject()));
QMenu* audioSourceMenu = audioDebugMenu->addMenu("Generated Audio Source");
{
QAction *pinkNoise = addCheckableActionToQMenuAndActionHash(audioSourceMenu, MenuOption::AudioSourcePinkNoise,
0,
false,
appInstance->getAudio(),
SLOT(selectAudioSourcePinkNoise()));
QAction *sine440 = addCheckableActionToQMenuAndActionHash(audioSourceMenu, MenuOption::AudioSourceSine440,
0,
true,
appInstance->getAudio(),
SLOT(selectAudioSourceSine440()));
QActionGroup* audioSourceGroup = new QActionGroup(audioSourceMenu);
audioSourceGroup->addAction(pinkNoise);
audioSourceGroup->addAction(sine440);
}
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::AudioScope,
Qt::CTRL | Qt::Key_P, false,
appInstance->getAudio(),

4
interface/src/Menu.h
View file

@ -347,7 +347,9 @@ namespace MenuOption {
const QString AudioSpatialProcessingSlightlyRandomSurfaces = "Slightly Random Surfaces";
const QString AudioSpatialProcessingStereoSource = "Stereo Source";
const QString AudioSpatialProcessingWithDiffusions = "With Diffusions";
const QString AudioToneInjection = "Inject Test Tone";
const QString AudioSourceInject = "Generated Audio";
const QString AudioSourcePinkNoise = "Pink Noise";
const QString AudioSourceSine440 = "Sine 440hz";
const QString Avatars = "Avatars";
const QString AvatarsReceiveShadows = "Avatars Receive Shadows";
const QString Bandwidth = "Bandwidth Display";

174
libraries/audio/src/AudioBuffer.h Normal file
View file

@ -0,0 +1,174 @@
//
// AudioBuffer.h
// hifi
//
// Created by Craig Hansen-Sturm on 8/29/14.
// Copyright 2014 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
//
#ifndef hifi_AudioBuffer_h
#define hifi_AudioBuffer_h
#include <typeinfo>
template< typename T >
class AudioFrameBuffer {
uint16_t _channelCount;
uint16_t _frameCount;
uint16_t _frameCountMax;
T** _frameBuffer;
void allocateFrames() {
_frameBuffer = new T*[_channelCount];
if (_frameBuffer) {
for (uint16_t i = 0; i < _channelCount; ++i) {
_frameBuffer[i] = new T[_frameCountMax];
}
}
}
void deallocateFrames() {
if (_frameBuffer) {
for (uint16_t i = 0; i < _channelCount; ++i) {
delete _frameBuffer[i];
}
delete _frameBuffer;
}
_frameBuffer = NULL;
}
public:
AudioFrameBuffer() :
_channelCount(0),
_frameCount(0),
_frameCountMax(0),
_frameBuffer(NULL) {
}
AudioFrameBuffer(const uint16_t channelCount, const uint16_t frameCount) :
_channelCount(channelCount),
_frameCount(frameCount),
_frameCountMax(frameCount),
_frameBuffer(NULL) {
allocateFrames();
}
~AudioFrameBuffer() {
finalize();
}
void initialize(const uint16_t channelCount, const uint16_t frameCount) {
if (_frameBuffer) {
finalize();
}
_channelCount = channelCount;
_frameCount = frameCount;
_frameCountMax = frameCount;
allocateFrames();
}
void finalize() {
deallocateFrames();
_channelCount = 0;
_frameCount = 0;
}
T**& getFrameData() {
return _frameBuffer;
}
uint16_t getChannelCount() {
return _channelCount;
}
uint16_t getFrameCount() {
return _frameCount;
}
void zeroFrames() {
if (!_frameBuffer) {
return;
}
for (uint16_t i = 0; i < _channelCount; ++i) {
memset(_frameBuffer[i], 0, sizeof(T)*_frameCountMax);
}
}
template< typename S >
void copyFrames(uint16_t channelCount, const uint16_t frameCount, S* frames, const bool copyOut = false) {
if ( !_frameBuffer || !frames) {
return;
}
assert(channelCount == _channelCount);
assert(frameCount <= _frameCountMax);
_frameCount = frameCount; // we allow copying fewer frames than we've allocated
if (copyOut) {
S* dst = frames;
if(typeid(T) == typeid(S)) { // source and destination types are the same
for (int i = 0; i < _frameCount; ++i) {
for (int j = 0; j < _channelCount; ++j) {
*dst++ = _frameBuffer[j][i];
}
}
}
else {
if(typeid(T) == typeid(float32_t) &&
typeid(S) == typeid(int16_t)) {
const int scale = (2 << ((8 * sizeof(S)) - 1));
for (int i = 0; i < _frameCount; ++i) {
for (int j = 0; j < _channelCount; ++j) {
*dst++ = (S)(_frameBuffer[j][i] * scale);
}
}
}
else {
assert(0); // currently unsupported conversion
}
}
}
else { // copyIn
S* src = frames;
if(typeid(T) == typeid(S)) { // source and destination types are the same
for (int i = 0; i < _frameCount; ++i) {
for (int j = 0; j < _channelCount; ++j) {
_frameBuffer[j][i] = *src++;
}
}
}
else {
if(typeid(T) == typeid(float32_t) &&
typeid(S) == typeid(int16_t)) {
const int scale = (2 << ((8 * sizeof(S)) - 1));
for (int i = 0; i < _frameCount; ++i) {
for (int j = 0; j < _channelCount; ++j) {
_frameBuffer[j][i] = ((T)(*src++)) / scale;
}
}
}
else {
assert(0); // currently unsupported conversion
}
}
}
}
};
typedef AudioFrameBuffer< float32_t > AudioBufferFloat32;
typedef AudioFrameBuffer< int32_t > AudioBufferSInt32;
#endif // hifi_AudioBuffer_h

10
libraries/audio/src/AudioFilter.h
View file

@ -37,11 +37,11 @@ public:
//
// ctor/dtor
//
AudioBiquad()
: _xm1(0.)
, _xm2(0.)
, _ym1(0.)
, _ym2(0.) {
AudioBiquad() :
_xm1(0.),
_xm2(0.),
_ym1(0.),
_ym2(0.) {
setParameters(0.,0.,0.,0.,0.);
}

3
libraries/audio/src/AudioFilterBank.cpp
View file

@ -9,9 +9,12 @@
// See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//
#include <assert.h>
#include <math.h>
#include <SharedUtil.h>
#include "AudioRingBuffer.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioFilter.h"
#include "AudioFilterBank.h"

18
libraries/audio/src/AudioFilterBank.h
View file

@ -49,9 +49,9 @@ public:
//
// ctor/dtor
//
AudioFilterBank()
: _sampleRate(0.)
, _frameCount(0) {
AudioFilterBank() :
_sampleRate(0.0f),
_frameCount(0) {
for (int i = 0; i < _channelCount; ++i) {
_buffer[ i ] = NULL;
}
@ -64,7 +64,7 @@ public:
//
// public interface
//
void initialize(const float sampleRate, const int frameCount) {
void initialize(const float sampleRate, const int frameCount = 0) {
finalize();
for (int i = 0; i < _channelCount; ++i) {
@ -141,6 +141,16 @@ public:
}
}
void render(AudioBufferFloat32& frameBuffer) {
float32_t** samples = frameBuffer.getFrameData();
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
for (int i = 0; i < _filterCount; ++i) {
_filters[i][j].render( samples[j], samples[j], frameBuffer.getFrameCount() );
}
}
}
void reset() {
for (int i = 0; i < _filterCount; ++i) {
for (int j = 0; j < _channelCount; ++j) {

85
libraries/audio/src/AudioFormat.h Normal file
View file

@ -0,0 +1,85 @@
//
// AudioFormat.h
// hifi
//
// Created by Craig Hansen-Sturm on 8/28/14.
// Copyright 2014 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
//
#ifndef hifi_AudioFormat_h
#define hifi_AudioFormat_h
#ifndef _FLOAT32_T
#define _FLOAT32_T
typedef float float32_t;
#endif
#ifndef _FLOAT64_T
#define _FLOAT64_T
typedef double float64_t;
#endif
//
// Audio format structure (currently for uncompressed streams only)
//
struct AudioFormat {
struct Flags {
uint32_t _isFloat : 1;
uint32_t _isSigned : 1;
uint32_t _isInterleaved : 1;
uint32_t _isBigEndian : 1;
uint32_t _isPacked : 1;
uint32_t _reserved : 27;
} _flags;
uint32_t _bytesPerFrame;
uint32_t _channelsPerFrame;
uint32_t _bitsPerChannel;
float64_t _sampleRate;
AudioFormat() {
memset(this, 0, sizeof(*this));
}
~AudioFormat() { }
AudioFormat& operator=(const AudioFormat& fmt) {
memcpy(this, &fmt, sizeof(*this));
return *this;
}
bool operator==(const AudioFormat& fmt) {
return memcmp(this, &fmt, sizeof(*this)) == 0;
}
bool operator!=(const AudioFormat& fmt) {
return memcmp(this, &fmt, sizeof(*this)) != 0;
}
void setCanonicalFloat32(uint32_t channels) {
assert(channels > 0 && channels <= 2);
_sampleRate = SAMPLE_RATE; // todo: create audio constants header
_bitsPerChannel = sizeof(float32_t) * 8;
_channelsPerFrame = channels;
_bytesPerFrame = _channelsPerFrame * _bitsPerChannel / 8;
_flags._isFloat = true;
_flags._isInterleaved = _channelsPerFrame > 1;
}
void setCanonicalInt16(uint32_t channels) {
assert(channels > 0 && channels <= 2);
_sampleRate = SAMPLE_RATE; // todo: create audio constants header
_bitsPerChannel = sizeof(int16_t) * 8;
_channelsPerFrame = channels;
_bytesPerFrame = _channelsPerFrame * _bitsPerChannel / 8;
_flags._isSigned = true;
_flags._isInterleaved = _channelsPerFrame > 1;
}
};
#endif // hifi_AudioFormat_h

75
libraries/audio/src/AudioGain.h Normal file
View file

@ -0,0 +1,75 @@
//
// AudioGain.h
// hifi
//
// Created by Craig Hansen-Sturm on 9/1/14.
// Copyright 2014 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
//
#ifndef hifi_AudioGain_h
#define hifi_AudioGain_h
class AudioGain
{
float32_t _gain;
bool _mute;
public:
AudioGain() {
initialize();
}
~AudioGain() {
finalize();
}
void initialize() {
setParameters(1.0f,0.0f);
}
void finalize() {
}
void reset() {
initialize();
}
void setParameters(const float gain, const float mute) {
_gain = std::min(std::max(gain, 0.0f), 1.0f);
_mute = mute != 0.0f;
}
void getParameters(float& gain, float& mute) {
gain = _gain;
mute = _mute ? 1.0f : 0.0f;
}
void render(AudioBufferFloat32& frameBuffer) {
if (_mute) {
frameBuffer.zeroFrames();
return;
}
float32_t** samples = frameBuffer.getFrameData();
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
for (uint16_t i = 0; i < frameBuffer.getFrameCount(); i += 8) {
samples[j][i + 0] *= _gain;
samples[j][i + 1] *= _gain;
samples[j][i + 2] *= _gain;
samples[j][i + 3] *= _gain;
samples[j][i + 4] *= _gain;
samples[j][i + 5] *= _gain;
samples[j][i + 6] *= _gain;
samples[j][i + 7] *= _gain;
}
}
}
};
#endif // AudioGain_h

21
libraries/audio/src/AudioSourceNoise.cpp Normal file
View file

@ -0,0 +1,21 @@
//
// AudioSourceNoise.cpp
// hifi
//
// Created by Craig Hansen-Sturm on 8/10/14.
// Copyright 2014 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
//
#include <assert.h>
#include <math.h>
#include <SharedUtil.h>
#include "AudioRingBuffer.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioSourceNoise.h"
template<>
uint32_t AudioSourcePinkNoise::_randomSeed = 1974; // a truly random number

103
libraries/audio/src/AudioSourceNoise.h Normal file
View file

@ -0,0 +1,103 @@
//
// AudioSourceNoise.h
// hifi
//
// Created by Craig Hansen-Sturm on 9/1/14.
// Copyright 2014 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
//
// Adapted from code by Phil Burk http://www.firstpr.com.au/dsp/pink-noise/
//
#ifndef hifi_AudioSourceNoise_h
#define hifi_AudioSourceNoise_h
template< const uint16_t N = 30>
class AudioSourceNoise
{
static const uint16_t _randomRows = N;
static const uint16_t _randomBits = 24;
static const uint16_t _randomShift = (sizeof(int32_t) * 8) - _randomBits;
static uint32_t _randomSeed;
int32_t _rows[_randomRows];
int32_t _runningSum; // used to optimize summing of generators.
uint16_t _index; // incremented each sample.
uint16_t _indexMask; // index wrapped by ANDing with this mask.
float32_t _scale; // used to scale within range of -1.0 to +1.0
static uint32_t generateRandomNumber() {
_randomSeed = (_randomSeed * 196314165) + 907633515;
return _randomSeed >> _randomShift;
}
public:
AudioSourceNoise() {
initialize();
}
~AudioSourceNoise() {
finalize();
}
void initialize() {
memset(_rows, 0, _randomRows * sizeof(int32_t));
_runningSum = 0;
_index = 0;
_indexMask = (1 << _randomRows) - 1;
_scale = 1.0f / ((_randomRows + 1) * (1 << (_randomBits - 1)));
}
void finalize() {
}
void reset() {
initialize();
}
void setParameters(void) {
}
void getParameters(void) {
}
void render(AudioBufferFloat32& frameBuffer) {
uint32_t randomNumber;
float32_t** samples = frameBuffer.getFrameData();
for (uint16_t i = 0; i < frameBuffer.getFrameCount(); ++i) {
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
_index = (_index + 1) & _indexMask; // increment and mask index.
if (_index != 0) { // if index is zero, don't update any random values.
uint32_t numZeros = 0; // determine how many trailing zeros in _index
uint32_t tmp = _index;
while ((tmp & 1) == 0) {
tmp >>= 1;
numZeros++;
}
// replace the indexed _rows random value. subtract and add back to _runningSum instead
// of adding all the random values together. only one value changes each time.
_runningSum -= _rows[numZeros];
randomNumber = generateRandomNumber();
_runningSum += randomNumber;
_rows[numZeros] = randomNumber;
}
// add extra white noise value and scale between -1.0 and +1.0
samples[j][i] = (_runningSum + generateRandomNumber()) * _scale;
}
}
}
};
typedef AudioSourceNoise<> AudioSourcePinkNoise;
#endif // AudioSourceNoise_h

20
libraries/audio/src/AudioSourceTone.cpp Normal file
View file

@ -0,0 +1,20 @@
//
// AudioSourceTone.cpp
// hifi
//
// Created by Craig Hansen-Sturm on 8/10/14.
// Copyright 2014 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
//
#include <assert.h>
#include <math.h>
#include <SharedUtil.h>
#include "AudioRingBuffer.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioSourceTone.h"
uint32_t AudioSourceTone::_frameOffset = 0;

72
libraries/audio/src/AudioSourceTone.h Normal file
View file

@ -0,0 +1,72 @@
//
// AudioSourceTone.h
// hifi
//
// Created by Craig Hansen-Sturm on 9/1/14.
// Copyright 2014 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
//
#ifndef hifi_AudioSourceTone_h
#define hifi_AudioSourceTone_h
class AudioSourceTone
{
static uint32_t _frameOffset;
float32_t _frequency;
float32_t _amplitude;
float32_t _sampleRate;
float32_t _omega;
public:
AudioSourceTone() {
initialize();
}
~AudioSourceTone() {
finalize();
}
void initialize() {
_frameOffset = 0;
setParameters(SAMPLE_RATE, 220.0f, 0.9f);
}
void finalize() {
}
void reset() {
_frameOffset = 0;
}
void setParameters(const float32_t sampleRate, const float32_t frequency, const float32_t amplitude) {
_sampleRate = std::max(sampleRate, 1.0f);
_frequency = std::max(frequency, 1.0f);
_amplitude = std::max(amplitude, 1.0f);
_omega = _frequency / _sampleRate * TWO_PI;
}
void getParameters(float32_t& sampleRate, float32_t& frequency, float32_t& amplitude) {
sampleRate = _sampleRate;
frequency = _frequency;
amplitude = _amplitude;
}
void render(AudioBufferFloat32& frameBuffer) {
// note: this is a placeholder implementation. final version will not include any transcendental ops in our render loop
float32_t** samples = frameBuffer.getFrameData();
for (uint16_t i = 0; i < frameBuffer.getFrameCount(); ++i) {
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
samples[j][i] = sinf((i + _frameOffset) * _omega);
}
}
_frameOffset += frameBuffer.getFrameCount();
}
};
#endif

2
libraries/audio/src/PositionalAudioStream.h
View file

@ -16,6 +16,8 @@
#include <AABox.h>
#include "InboundAudioStream.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioFilter.h"
#include "AudioFilterBank.h"