/* * 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 #include #include 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(PanningModelHRTF) , 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() { } 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(desiredAzimuthIndexFloat); azimuthBlend = desiredAzimuthIndexFloat - static_cast(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(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(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(sampleRate()); } } // namespace WebCore #endif // ENABLE(WEB_AUDIO)