[SCM] WebKit Debian packaging branch, debian/experimental, updated. upstream/1.3.3-9427-gc2be6fc
crogers at google.com
crogers at google.com
Wed Dec 22 14:27:26 UTC 2010
The following commit has been merged in the debian/experimental branch:
commit ce231f362639fb4ac2591dd204250cd6000e3ae6
Author: crogers at google.com <crogers at google.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc>
Date: Sat Oct 9 02:30:25 2010 +0000
2010-10-08 Chris Rogers <crogers at google.com>
Reviewed by James Robinson.
Add HRTFPanner files
https://bugs.webkit.org/show_bug.cgi?id=46299
No new tests since audio API is not yet implemented.
* platform/audio/HRTFPanner.cpp: Added.
(WebCore::HRTFPanner::HRTFPanner):
(WebCore::HRTFPanner::~HRTFPanner):
(WebCore::HRTFPanner::fftSizeForSampleRate):
(WebCore::HRTFPanner::reset):
(WebCore::wrapDistance):
(WebCore::HRTFPanner::calculateDesiredAzimuthIndexAndBlend):
(WebCore::HRTFPanner::pan):
* platform/audio/HRTFPanner.h: Added.
(WebCore::HRTFPanner::fftSize):
(WebCore::HRTFPanner::sampleRate):
git-svn-id: http://svn.webkit.org/repository/webkit/trunk@69440 268f45cc-cd09-0410-ab3c-d52691b4dbfc
diff --git a/WebCore/ChangeLog b/WebCore/ChangeLog
index bd57853..51055f6 100644
--- a/WebCore/ChangeLog
+++ b/WebCore/ChangeLog
@@ -1,3 +1,24 @@
+2010-10-08 Chris Rogers <crogers at google.com>
+
+ Reviewed by James Robinson.
+
+ Add HRTFPanner files
+ https://bugs.webkit.org/show_bug.cgi?id=46299
+
+ No new tests since audio API is not yet implemented.
+
+ * platform/audio/HRTFPanner.cpp: Added.
+ (WebCore::HRTFPanner::HRTFPanner):
+ (WebCore::HRTFPanner::~HRTFPanner):
+ (WebCore::HRTFPanner::fftSizeForSampleRate):
+ (WebCore::HRTFPanner::reset):
+ (WebCore::wrapDistance):
+ (WebCore::HRTFPanner::calculateDesiredAzimuthIndexAndBlend):
+ (WebCore::HRTFPanner::pan):
+ * platform/audio/HRTFPanner.h: Added.
+ (WebCore::HRTFPanner::fftSize):
+ (WebCore::HRTFPanner::sampleRate):
+
2010-10-08 James Robinson <jamesr at chromium.org>
Reviewed by Adam Barth.
diff --git a/WebCore/platform/audio/HRTFPanner.cpp b/WebCore/platform/audio/HRTFPanner.cpp
new file mode 100644
index 0000000..56f06f1
--- /dev/null
+++ b/WebCore/platform/audio/HRTFPanner.cpp
@@ -0,0 +1,229 @@
+/*
+ * 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 <math.h>
+#include <wtf/RefPtr.h>
+
+using namespace std;
+
+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;
+
+HRTFPanner::HRTFPanner(double sampleRate)
+ : Panner(PanningModelHRTF)
+ , m_sampleRate(sampleRate)
+ , m_isFirstRender(true)
+ , m_azimuthIndex(0)
+ , m_convolverL(fftSizeForSampleRate(sampleRate))
+ , m_convolverR(fftSizeForSampleRate(sampleRate))
+ , m_delayLineL(MaxDelayTimeSeconds, sampleRate)
+ , m_delayLineR(MaxDelayTimeSeconds, sampleRate)
+{
+}
+
+HRTFPanner::~HRTFPanner()
+{
+}
+
+size_t HRTFPanner::fftSizeForSampleRate(double 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 that size for higher sample rates.
+ ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0);
+ return (sampleRate <= 48000.0) ? 512 : 1024;
+}
+
+void HRTFPanner::reset()
+{
+ m_isFirstRender = true;
+ m_convolverL.reset();
+ m_convolverR.reset();
+ m_delayLineL.reset();
+ m_delayLineR.reset();
+}
+
+static bool wrapDistance(int i, int j, int length)
+{
+ int directDistance = abs(i - j);
+ int indirectDistance = length - directDistance;
+
+ return indirectDistance < directDistance;
+}
+
+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 = HRTFDatabaseLoader::defaultHRTFDatabase();
+ 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 = max(0, desiredAzimuthIndex);
+ desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex);
+ return desiredAzimuthIndex;
+}
+
+void HRTFPanner::pan(double desiredAzimuth, double elevation, 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 = HRTFDatabaseLoader::defaultHRTFDatabase();
+ 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.
+ AudioChannel* inputChannelL = inputBus->channelByType(AudioBus::ChannelLeft);
+ AudioChannel* inputChannelR = numInputChannels > 1 ? inputBus->channelByType(AudioBus::ChannelRight) : 0;
+
+ // Get source and destination pointers.
+ float* sourceL = inputChannelL->data();
+ float* sourceR = numInputChannels > 1 ? inputChannelR->data() : sourceL;
+ float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->data();
+ float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->data();
+
+ double azimuthBlend;
+ int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
+
+ // This algorithm currently requires that we process in power-of-two size chunks at least 128.
+ ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
+ ASSERT(framesToProcess >= 128);
+
+ const unsigned framesPerSegment = 128;
+ const unsigned numberOfSegments = framesToProcess / framesPerSegment;
+
+ for (unsigned segment = 0; segment < numberOfSegments; ++segment) {
+ if (m_isFirstRender) {
+ // Snap exactly to desired position (first time and after reset()).
+ m_azimuthIndex = desiredAzimuthIndex;
+ m_isFirstRender = false;
+ } else {
+ // Each segment renders with an azimuth index closer by one to the desired azimuth index.
+ // Because inter-aural time delay is mostly a factor of azimuth and the delay is where the clicks and graininess come from,
+ // we don't bother smoothing the elevations.
+ int numberOfAzimuths = database->numberOfAzimuths();
+ bool wrap = wrapDistance(m_azimuthIndex, desiredAzimuthIndex, numberOfAzimuths);
+ if (wrap) {
+ if (m_azimuthIndex < desiredAzimuthIndex)
+ m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths;
+ else if (m_azimuthIndex > desiredAzimuthIndex)
+ m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths;
+ } else {
+ if (m_azimuthIndex < desiredAzimuthIndex)
+ m_azimuthIndex = (m_azimuthIndex + 1) % numberOfAzimuths;
+ else if (m_azimuthIndex > desiredAzimuthIndex)
+ m_azimuthIndex = (m_azimuthIndex - 1 + numberOfAzimuths) % numberOfAzimuths;
+ }
+ }
+
+ // Get the HRTFKernels and interpolated delays.
+ HRTFKernel* kernelL;
+ HRTFKernel* kernelR;
+ double frameDelayL;
+ double frameDelayR;
+ database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex, elevation, kernelL, kernelR, frameDelayL, frameDelayR);
+
+ ASSERT(kernelL && kernelR);
+ if (!kernelL || !kernelR) {
+ outputBus->zero();
+ return;
+ }
+
+ ASSERT(frameDelayL / sampleRate() < MaxDelayTimeSeconds && frameDelayR / sampleRate() < MaxDelayTimeSeconds);
+
+ // Calculate the source and destination pointers for the current segment.
+ unsigned offset = segment * framesPerSegment;
+ float* segmentSourceL = sourceL + offset;
+ 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);
+
+ // Now do the convolutions in-place.
+ m_convolverL.process(kernelL->fftFrame(), segmentDestinationL, segmentDestinationL, framesPerSegment);
+ m_convolverR.process(kernelR->fftFrame(), segmentDestinationR, segmentDestinationR, framesPerSegment);
+ }
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/platform/audio/HRTFPanner.h b/WebCore/platform/audio/HRTFPanner.h
new file mode 100644
index 0000000..6c13d48
--- /dev/null
+++ b/WebCore/platform/audio/HRTFPanner.h
@@ -0,0 +1,68 @@
+/*
+ * 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.
+ */
+
+#ifndef HRTFPanner_h
+#define HRTFPanner_h
+
+#include "DelayDSPKernel.h"
+#include "FFTConvolver.h"
+#include "Panner.h"
+
+namespace WebCore {
+
+class HRTFPanner : public Panner {
+public:
+ explicit HRTFPanner(double sampleRate);
+ virtual ~HRTFPanner();
+
+ // Panner
+ virtual void pan(double azimuth, double elevation, AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess);
+ virtual void reset();
+
+ size_t fftSize() { return fftSizeForSampleRate(m_sampleRate); }
+ static size_t fftSizeForSampleRate(double sampleRate);
+
+ double sampleRate() const { return m_sampleRate; }
+
+private:
+ // Given an azimuth angle in the range -180 -> +180, returns the corresponding azimuth index for the database,
+ // and azimuthBlend which is an interpolation value from 0 -> 1.
+ int calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend);
+
+ double m_sampleRate;
+
+ // m_isFirstRender and m_azimuthIndex are used to avoid harshly changing from rendering at one azimuth angle to another angle very far away.
+ // Changing the azimuth gradually produces a smoother sound.
+ bool m_isFirstRender;
+ int m_azimuthIndex;
+
+ FFTConvolver m_convolverL;
+ FFTConvolver m_convolverR;
+ DelayDSPKernel m_delayLineL;
+ DelayDSPKernel m_delayLineR;
+};
+
+} // namespace WebCore
+
+#endif // HRTFPanner_h
--
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