Index: third_party/WebKit/Source/platform/audio/HRTFPanner.cpp |
diff --git a/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp b/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp |
index a2ad4a13cbd240ac1846fcf0a68ce74daa77fd94..d1240e0f5328d247b9502c52dee15fd52c84967e 100644 |
--- a/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp |
+++ b/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp |
@@ -10,16 +10,17 @@ |
* 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. |
+ * 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 "platform/audio/HRTFPanner.h" |
@@ -31,7 +32,8 @@ |
namespace blink { |
-// The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds). |
+// 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; |
@@ -65,13 +67,14 @@ HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* 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. |
- // For different sample rates, the truncated response is resampled. |
- // The resampled length is used to compute the FFT size by choosing a power of two that is |
- // greater than or equal the resampled length. This power of two is doubled to get the actual FFT size. |
+ // 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. For different sample rates, the truncated response is resampled. |
+ // The resampled length is used to compute the FFT size by choosing a power |
+ // of two that is greater than or equal the resampled length. This power of |
+ // two is doubled to get the actual FFT size. |
ASSERT(AudioUtilities::isValidAudioBufferSampleRate(sampleRate)); |
@@ -93,8 +96,8 @@ void HRTFPanner::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. |
+ // 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; |
@@ -107,8 +110,9 @@ int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, |
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. |
+ // 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 = clampTo(desiredAzimuthIndex, 0, numberOfAzimuths - 1); |
return desiredAzimuthIndex; |
} |
@@ -140,7 +144,8 @@ void HRTFPanner::pan(double desiredAzimuth, |
return; |
} |
- // IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth. |
+ // 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; |
@@ -151,7 +156,8 @@ void HRTFPanner::pan(double desiredAzimuth, |
} |
// 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. |
+ // 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 = |
@@ -203,7 +209,8 @@ void HRTFPanner::pan(double desiredAzimuth, |
} |
} |
- // This algorithm currently requires that we process in power-of-two size chunks at least RenderingQuantum. |
+ // 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); |
@@ -260,7 +267,8 @@ void HRTFPanner::pan(double desiredAzimuth, |
bool needsCrossfading = m_crossfadeIncr; |
- // Have the convolvers render directly to the final destination if we're not cross-fading. |
+ // Have the convolvers render directly to the final destination if we're not |
+ // cross-fading. |
float* convolutionDestinationL1 = |
needsCrossfading ? m_tempL1.data() : segmentDestinationL; |
float* convolutionDestinationR1 = |
@@ -271,7 +279,8 @@ void HRTFPanner::pan(double desiredAzimuth, |
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. |
+ // 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, |
@@ -323,29 +332,31 @@ void HRTFPanner::panWithSampleAccurateValues( |
AudioBus* outputBus, |
size_t framesToProcess, |
AudioBus::ChannelInterpretation channelInterpretation) { |
- // Sample-accurate (a-rate) HRTF panner is not implemented, just k-rate. Just grab the current |
- // azimuth/elevation and use that. |
+ // Sample-accurate (a-rate) HRTF panner is not implemented, just k-rate. Just |
+ // grab the current azimuth/elevation and use that. |
// |
- // We are assuming that the inherent smoothing in the HRTF processing is good enough, and we |
- // don't want to increase the complexity of the HRTF panner by 15-20 times. (We need to cmopute |
- // one output sample for each possibly different impulse response. That N^2. Previously, we |
- // used an FFT to do them all at once for a complexity of N/log2(N). Hence, N/log2(N) times |
+ // We are assuming that the inherent smoothing in the HRTF processing is good |
+ // enough, and we don't want to increase the complexity of the HRTF panner by |
+ // 15-20 times. (We need to compute one output sample for each possibly |
+ // different impulse response. That N^2. Previously, we used an FFT to do |
+ // them all at once for a complexity of N/log2(N). Hence, N/log2(N) times |
// more complex.) |
pan(desiredAzimuth[0], elevation[0], inputBus, outputBus, framesToProcess, |
channelInterpretation); |
} |
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. |
+ // 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. |
+ // 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()); |
} |