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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /dom/media/webaudio/blink/HRTFPanner.cpp | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/webaudio/blink/HRTFPanner.cpp')
-rw-r--r-- | dom/media/webaudio/blink/HRTFPanner.cpp | 328 |
1 files changed, 328 insertions, 0 deletions
diff --git a/dom/media/webaudio/blink/HRTFPanner.cpp b/dom/media/webaudio/blink/HRTFPanner.cpp new file mode 100644 index 0000000000..3d01fb5b39 --- /dev/null +++ b/dom/media/webaudio/blink/HRTFPanner.cpp @@ -0,0 +1,328 @@ +/* + * 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 "HRTFPanner.h" +#include "HRTFDatabaseLoader.h" + +#include "FFTConvolver.h" +#include "HRTFDatabase.h" +#include "AudioBlock.h" + +using namespace mozilla; +using dom::ChannelInterpretation; + +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 float MaxDelayTimeSeconds = 0.002f; + +const int UninitializedAzimuth = -1; + +HRTFPanner::HRTFPanner(float sampleRate, + already_AddRefed<HRTFDatabaseLoader> databaseLoader) + : m_databaseLoader(databaseLoader), + m_sampleRate(sampleRate), + m_crossfadeSelection(CrossfadeSelection1), + m_azimuthIndex1(UninitializedAzimuth), + m_azimuthIndex2(UninitializedAzimuth) + // m_elevation1 and m_elevation2 are initialized in pan() + , + m_crossfadeX(0), + m_crossfadeIncr(0), + m_convolverL1(HRTFElevation::fftSizeForSampleRate(sampleRate)), + m_convolverR1(m_convolverL1.fftSize()), + m_convolverL2(m_convolverL1.fftSize()), + m_convolverR2(m_convolverL1.fftSize()), + m_delayLine(MaxDelayTimeSeconds * sampleRate) { + MOZ_ASSERT(m_databaseLoader); + MOZ_COUNT_CTOR(HRTFPanner); +} + +HRTFPanner::~HRTFPanner() { MOZ_COUNT_DTOR(HRTFPanner); } + +size_t HRTFPanner::sizeOfIncludingThis( + mozilla::MallocSizeOf aMallocSizeOf) const { + size_t amount = aMallocSizeOf(this); + + // NB: m_databaseLoader can be shared, so it is not measured here + amount += m_convolverL1.sizeOfExcludingThis(aMallocSizeOf); + amount += m_convolverR1.sizeOfExcludingThis(aMallocSizeOf); + amount += m_convolverL2.sizeOfExcludingThis(aMallocSizeOf); + amount += m_convolverR2.sizeOfExcludingThis(aMallocSizeOf); + amount += m_delayLine.SizeOfExcludingThis(aMallocSizeOf); + + return amount; +} + +void HRTFPanner::reset() { + m_azimuthIndex1 = UninitializedAzimuth; + m_azimuthIndex2 = UninitializedAzimuth; + // m_elevation1 and m_elevation2 are initialized in pan() + m_crossfadeSelection = CrossfadeSelection1; + m_crossfadeX = 0.0f; + m_crossfadeIncr = 0.0f; + m_convolverL1.reset(); + m_convolverR1.reset(); + m_convolverL2.reset(); + m_convolverR2.reset(); + m_delayLine.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; + + int numberOfAzimuths = HRTFDatabase::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 AudioBlock* inputBus, AudioBlock* outputBus) { +#ifdef DEBUG + unsigned numInputChannels = inputBus->IsNull() ? 0 : inputBus->ChannelCount(); + + MOZ_ASSERT(numInputChannels <= 2); + MOZ_ASSERT(inputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE); +#endif + + bool isOutputGood = outputBus && outputBus->ChannelCount() == 2 && + outputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE; + MOZ_ASSERT(isOutputGood); + + if (!isOutputGood) { + if (outputBus) outputBus->SetNull(outputBus->GetDuration()); + return; + } + + HRTFDatabase* database = m_databaseLoader->database(); + if (!database) { // not yet loaded + outputBus->SetNull(outputBus->GetDuration()); + 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; + MOZ_ASSERT(isAzimuthGood); + if (!isAzimuthGood) { + outputBus->SetNull(outputBus->GetDuration()); + 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. + + // Get destination pointers. + float* destinationL = + static_cast<float*>(const_cast<void*>(outputBus->mChannelData[0])); + float* destinationR = + static_cast<float*>(const_cast<void*>(outputBus->mChannelData[1])); + + 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; + } + } + + // 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; + MOZ_ASSERT(areKernelsGood); + if (!areKernelsGood) { + outputBus->SetNull(outputBus->GetDuration()); + return; + } + + MOZ_ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds && + frameDelayR1 / sampleRate() < MaxDelayTimeSeconds); + MOZ_ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds && + frameDelayR2 / sampleRate() < MaxDelayTimeSeconds); + + // Crossfade inter-aural delays based on transitions. + float frameDelaysL[WEBAUDIO_BLOCK_SIZE]; + float frameDelaysR[WEBAUDIO_BLOCK_SIZE]; + { + float x = m_crossfadeX; + float incr = m_crossfadeIncr; + for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) { + frameDelaysL[i] = (1 - x) * frameDelayL1 + x * frameDelayL2; + frameDelaysR[i] = (1 - x) * frameDelayR1 + x * frameDelayR2; + x += incr; + } + } + + // First run through delay lines for inter-aural time difference. + m_delayLine.Write(*inputBus); + // "Speakers" means a mono input is read into both outputs (with possibly + // different delays). + m_delayLine.ReadChannel(frameDelaysL, outputBus, 0, + ChannelInterpretation::Speakers); + m_delayLine.ReadChannel(frameDelaysR, outputBus, 1, + ChannelInterpretation::Speakers); + m_delayLine.NextBlock(); + + bool needsCrossfading = m_crossfadeIncr; + + const float* convolutionDestinationL1; + const float* convolutionDestinationR1; + const float* convolutionDestinationL2; + const float* convolutionDestinationR2; + + // 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) { + convolutionDestinationL1 = + m_convolverL1.process(kernelL1->fftFrame(), destinationL); + convolutionDestinationR1 = + m_convolverR1.process(kernelR1->fftFrame(), destinationR); + } + + if (m_crossfadeSelection == CrossfadeSelection2 || needsCrossfading) { + convolutionDestinationL2 = + m_convolverL2.process(kernelL2->fftFrame(), destinationL); + convolutionDestinationR2 = + m_convolverR2.process(kernelR2->fftFrame(), destinationR); + } + + if (needsCrossfading) { + // Apply linear cross-fade. + float x = m_crossfadeX; + float incr = m_crossfadeIncr; + for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) { + destinationL[i] = (1 - x) * convolutionDestinationL1[i] + + x * convolutionDestinationL2[i]; + destinationR[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; + } + } else { + const float* sourceL; + const float* sourceR; + if (m_crossfadeSelection == CrossfadeSelection1) { + sourceL = convolutionDestinationL1; + sourceR = convolutionDestinationR1; + } else { + sourceL = convolutionDestinationL2; + sourceR = convolutionDestinationR2; + } + PodCopy(destinationL, sourceL, WEBAUDIO_BLOCK_SIZE); + PodCopy(destinationR, sourceR, WEBAUDIO_BLOCK_SIZE); + } +} + +int HRTFPanner::maxTailFrames() const { + // Although the ideal tail time would be the length of the impulse + // response, there is additional tail time from the approximations in the + // implementation. 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. The + // FFTs of the convolver are fftSize(), half of which is latency, but this + // is aligned with blocks and so is reduced by the one block which is + // processed immediately. + return m_delayLine.MaxDelayTicks() + m_convolverL1.fftSize() / 2 + + m_convolverL1.latencyFrames(); +} + +} // namespace WebCore |