blob: 6c874a3a5adb05c4e25012ffbf3e03b0e7c4551c [file] [log] [blame]
/*
* Copyright (C) 2010, Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "HRTFPanner.h"
#include "AudioBus.h"
#include "FFTConvolver.h"
#include "HRTFDatabase.h"
#include "HRTFDatabaseLoader.h"
#include <algorithm>
#include <wtf/MathExtras.h>
namespace WebCore {
// The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds).
// We ASSERT the delay values used in process() with this value.
const double MaxDelayTimeSeconds = 0.002;
const int UninitializedAzimuth = -1;
const unsigned RenderingQuantum = 128;
HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader)
: Panner(PanningModelType::HRTF)
, m_databaseLoader(databaseLoader)
, m_sampleRate(sampleRate)
, m_crossfadeSelection(CrossfadeSelection1)
, m_azimuthIndex1(UninitializedAzimuth)
, m_elevation1(0)
, m_azimuthIndex2(UninitializedAzimuth)
, m_elevation2(0)
, m_crossfadeX(0)
, m_crossfadeIncr(0)
, m_convolverL1(fftSizeForSampleRate(sampleRate))
, m_convolverR1(fftSizeForSampleRate(sampleRate))
, m_convolverL2(fftSizeForSampleRate(sampleRate))
, m_convolverR2(fftSizeForSampleRate(sampleRate))
, m_delayLineL(MaxDelayTimeSeconds, sampleRate)
, m_delayLineR(MaxDelayTimeSeconds, sampleRate)
, m_tempL1(RenderingQuantum)
, m_tempR1(RenderingQuantum)
, m_tempL2(RenderingQuantum)
, m_tempR2(RenderingQuantum)
{
ASSERT(databaseLoader);
}
HRTFPanner::~HRTFPanner() = default;
size_t HRTFPanner::fftSizeForSampleRate(float sampleRate)
{
// The HRTF impulse responses (loaded as audio resources) are 512 sample-frames @44.1KHz.
// Currently, we truncate the impulse responses to half this size, but an FFT-size of twice impulse response size is needed (for convolution).
// So for sample rates around 44.1KHz an FFT size of 512 is good. We double the FFT-size only for sample rates at least double this.
ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0);
return (sampleRate < 88200.0) ? 512 : 1024;
}
void HRTFPanner::reset()
{
m_convolverL1.reset();
m_convolverR1.reset();
m_convolverL2.reset();
m_convolverR2.reset();
m_delayLineL.reset();
m_delayLineR.reset();
}
int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend)
{
// Convert the azimuth angle from the range -180 -> +180 into the range 0 -> 360.
// The azimuth index may then be calculated from this positive value.
if (azimuth < 0)
azimuth += 360.0;
HRTFDatabase* database = m_databaseLoader->database();
ASSERT(database);
int numberOfAzimuths = database->numberOfAzimuths();
const double angleBetweenAzimuths = 360.0 / numberOfAzimuths;
// Calculate the azimuth index and the blend (0 -> 1) for interpolation.
double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths;
int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat);
azimuthBlend = desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex);
// We don't immediately start using this azimuth index, but instead approach this index from the last index we rendered at.
// This minimizes the clicks and graininess for moving sources which occur otherwise.
desiredAzimuthIndex = std::max(0, desiredAzimuthIndex);
desiredAzimuthIndex = std::min(numberOfAzimuths - 1, desiredAzimuthIndex);
return desiredAzimuthIndex;
}
void HRTFPanner::pan(double desiredAzimuth, double elevation, const AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess)
{
unsigned numInputChannels = inputBus ? inputBus->numberOfChannels() : 0;
bool isInputGood = inputBus && numInputChannels >= 1 && numInputChannels <= 2;
ASSERT(isInputGood);
bool isOutputGood = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length();
ASSERT(isOutputGood);
if (!isInputGood || !isOutputGood) {
if (outputBus)
outputBus->zero();
return;
}
// This code only runs as long as the context is alive and after database has been loaded.
HRTFDatabase* database = m_databaseLoader->database();
ASSERT(database);
if (!database) {
outputBus->zero();
return;
}
// IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth.
double azimuth = -desiredAzimuth;
bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0;
ASSERT(isAzimuthGood);
if (!isAzimuthGood) {
outputBus->zero();
return;
}
// Normally, we'll just be dealing with mono sources.
// If we have a stereo input, implement stereo panning with left source processed by left HRTF, and right source by right HRTF.
const AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft);
const AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0;
// Get source and destination pointers.
const float* sourceL = inputChannelL->data();
const float* sourceR = numInputChannels > 1 ? inputChannelR->data() : sourceL;
float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->mutableData();
float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->mutableData();
double azimuthBlend;
int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
// Initially snap azimuth and elevation values to first values encountered.
if (m_azimuthIndex1 == UninitializedAzimuth) {
m_azimuthIndex1 = desiredAzimuthIndex;
m_elevation1 = elevation;
}
if (m_azimuthIndex2 == UninitializedAzimuth) {
m_azimuthIndex2 = desiredAzimuthIndex;
m_elevation2 = elevation;
}
// Cross-fade / transition over a period of around 45 milliseconds.
// This is an empirical value tuned to be a reasonable trade-off between
// smoothness and speed.
const double fadeFrames = sampleRate() <= 48000 ? 2048 : 4096;
// Check for azimuth and elevation changes, initiating a cross-fade if needed.
if (!m_crossfadeX && m_crossfadeSelection == CrossfadeSelection1) {
if (desiredAzimuthIndex != m_azimuthIndex1 || elevation != m_elevation1) {
// Cross-fade from 1 -> 2
m_crossfadeIncr = 1 / fadeFrames;
m_azimuthIndex2 = desiredAzimuthIndex;
m_elevation2 = elevation;
}
}
if (m_crossfadeX == 1 && m_crossfadeSelection == CrossfadeSelection2) {
if (desiredAzimuthIndex != m_azimuthIndex2 || elevation != m_elevation2) {
// Cross-fade from 2 -> 1
m_crossfadeIncr = -1 / fadeFrames;
m_azimuthIndex1 = desiredAzimuthIndex;
m_elevation1 = elevation;
}
}
// This algorithm currently requires that we process in power-of-two size chunks at least RenderingQuantum.
ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
ASSERT(framesToProcess >= RenderingQuantum);
const unsigned framesPerSegment = RenderingQuantum;
const unsigned numberOfSegments = framesToProcess / framesPerSegment;
for (unsigned segment = 0; segment < numberOfSegments; ++segment) {
// Get the HRTFKernels and interpolated delays.
HRTFKernel* kernelL1;
HRTFKernel* kernelR1;
HRTFKernel* kernelL2;
HRTFKernel* kernelR2;
double frameDelayL1;
double frameDelayR1;
double frameDelayL2;
double frameDelayR2;
database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex1, m_elevation1, kernelL1, kernelR1, frameDelayL1, frameDelayR1);
database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex2, m_elevation2, kernelL2, kernelR2, frameDelayL2, frameDelayR2);
bool areKernelsGood = kernelL1 && kernelR1 && kernelL2 && kernelR2;
ASSERT(areKernelsGood);
if (!areKernelsGood) {
outputBus->zero();
return;
}
ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds && frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds && frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
// Crossfade inter-aural delays based on transitions.
double frameDelayL = (1 - m_crossfadeX) * frameDelayL1 + m_crossfadeX * frameDelayL2;
double frameDelayR = (1 - m_crossfadeX) * frameDelayR1 + m_crossfadeX * frameDelayR2;
// Calculate the source and destination pointers for the current segment.
unsigned offset = segment * framesPerSegment;
const float* segmentSourceL = sourceL + offset;
const float* segmentSourceR = sourceR + offset;
float* segmentDestinationL = destinationL + offset;
float* segmentDestinationR = destinationR + offset;
// First run through delay lines for inter-aural time difference.
m_delayLineL.setDelayFrames(frameDelayL);
m_delayLineR.setDelayFrames(frameDelayR);
m_delayLineL.process(segmentSourceL, segmentDestinationL, framesPerSegment);
m_delayLineR.process(segmentSourceR, segmentDestinationR, framesPerSegment);
bool needsCrossfading = m_crossfadeIncr;
// Have the convolvers render directly to the final destination if we're not cross-fading.
float* convolutionDestinationL1 = needsCrossfading ? m_tempL1.data() : segmentDestinationL;
float* convolutionDestinationR1 = needsCrossfading ? m_tempR1.data() : segmentDestinationR;
float* convolutionDestinationL2 = needsCrossfading ? m_tempL2.data() : segmentDestinationL;
float* convolutionDestinationR2 = needsCrossfading ? m_tempR2.data() : segmentDestinationR;
// Now do the convolutions.
// Note that we avoid doing convolutions on both sets of convolvers if we're not currently cross-fading.
if (m_crossfadeSelection == CrossfadeSelection1 || needsCrossfading) {
m_convolverL1.process(kernelL1->fftFrame(), segmentDestinationL, convolutionDestinationL1, framesPerSegment);
m_convolverR1.process(kernelR1->fftFrame(), segmentDestinationR, convolutionDestinationR1, framesPerSegment);
}
if (m_crossfadeSelection == CrossfadeSelection2 || needsCrossfading) {
m_convolverL2.process(kernelL2->fftFrame(), segmentDestinationL, convolutionDestinationL2, framesPerSegment);
m_convolverR2.process(kernelR2->fftFrame(), segmentDestinationR, convolutionDestinationR2, framesPerSegment);
}
if (needsCrossfading) {
// Apply linear cross-fade.
float x = m_crossfadeX;
float incr = m_crossfadeIncr;
for (unsigned i = 0; i < framesPerSegment; ++i) {
segmentDestinationL[i] = (1 - x) * convolutionDestinationL1[i] + x * convolutionDestinationL2[i];
segmentDestinationR[i] = (1 - x) * convolutionDestinationR1[i] + x * convolutionDestinationR2[i];
x += incr;
}
// Update cross-fade value from local.
m_crossfadeX = x;
if (m_crossfadeIncr > 0 && fabs(m_crossfadeX - 1) < m_crossfadeIncr) {
// We've fully made the crossfade transition from 1 -> 2.
m_crossfadeSelection = CrossfadeSelection2;
m_crossfadeX = 1;
m_crossfadeIncr = 0;
} else if (m_crossfadeIncr < 0 && fabs(m_crossfadeX) < -m_crossfadeIncr) {
// We've fully made the crossfade transition from 2 -> 1.
m_crossfadeSelection = CrossfadeSelection1;
m_crossfadeX = 0;
m_crossfadeIncr = 0;
}
}
}
}
double HRTFPanner::tailTime() const
{
// Because HRTFPanner is implemented with a DelayKernel and a FFTConvolver, the tailTime of the HRTFPanner
// is the sum of the tailTime of the DelayKernel and the tailTime of the FFTConvolver, which is MaxDelayTimeSeconds
// and fftSize() / 2, respectively.
return MaxDelayTimeSeconds + (fftSize() / 2) / static_cast<double>(sampleRate());
}
double HRTFPanner::latencyTime() const
{
// The latency of a FFTConvolver is also fftSize() / 2, and is in addition to its tailTime of the
// same value.
return (fftSize() / 2) / static_cast<double>(sampleRate());
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)