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/*
* 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.
*/
#ifdef UNSAFE_BUFFERS_BUILD
// TODO(crbug.com/351564777): Remove this and convert code to safer constructs.
#pragma allow_unsafe_buffers
#endif
#include "third_party/blink/renderer/platform/audio/hrtf_panner.h"
#include "base/containers/span.h"
#include "base/memory/scoped_refptr.h"
#include "third_party/blink/renderer/platform/audio/audio_bus.h"
#include "third_party/blink/renderer/platform/audio/audio_utilities.h"
#include "third_party/blink/renderer/platform/audio/fft_frame.h"
#include "third_party/blink/renderer/platform/audio/hrtf_database.h"
#include "third_party/blink/renderer/platform/audio/hrtf_database_loader.h"
#include "third_party/blink/renderer/platform/wtf/math_extras.h"
namespace blink {
namespace {
// 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.
constexpr double kMaxDelayTimeSeconds = 0.002;
constexpr int kUninitializedAzimuth = -1;
// 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& azimuth_blend) {
// 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;
}
const int number_of_azimuths = HRTFDatabase::NumberOfAzimuths();
const double angle_between_azimuths = 360.0 / number_of_azimuths;
// Calculate the azimuth index and the blend (0 -> 1) for interpolation.
const double desired_azimuth_index_float = azimuth / angle_between_azimuths;
int desired_azimuth_index = static_cast<int>(desired_azimuth_index_float);
azimuth_blend =
desired_azimuth_index_float - static_cast<double>(desired_azimuth_index);
// 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.
desired_azimuth_index =
ClampTo(desired_azimuth_index, 0, number_of_azimuths - 1);
return desired_azimuth_index;
}
} // namespace
HRTFPanner::HRTFPanner(float sample_rate,
unsigned render_quantum_frames,
HRTFDatabaseLoader* database_loader)
: database_loader_(database_loader),
sample_rate_(sample_rate),
azimuth_index1_(kUninitializedAzimuth),
azimuth_index2_(kUninitializedAzimuth),
convolver_l1_(FftSizeForSampleRate(sample_rate)),
convolver_r1_(FftSizeForSampleRate(sample_rate)),
convolver_l2_(FftSizeForSampleRate(sample_rate)),
convolver_r2_(FftSizeForSampleRate(sample_rate)),
delay_line_l_(kMaxDelayTimeSeconds, sample_rate, render_quantum_frames),
delay_line_r_(kMaxDelayTimeSeconds, sample_rate, render_quantum_frames),
temp_l1_(render_quantum_frames),
temp_r1_(render_quantum_frames),
temp_l2_(render_quantum_frames),
temp_r2_(render_quantum_frames),
render_quantum_frames_(render_quantum_frames) {
DCHECK(database_loader);
}
HRTFPanner::~HRTFPanner() = default;
unsigned HRTFPanner::FftSizeForSampleRate(float sample_rate) {
// 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.
DCHECK(audio_utilities::IsValidAudioBufferSampleRate(sample_rate));
constexpr int truncated_impulse_length = 256;
const double sample_rate_ratio = sample_rate / 44100;
const double resampled_length = truncated_impulse_length * sample_rate_ratio;
// This is the size used for analysis frames in the HRTF kernel. The
// convolvers used by the kernel are twice this size.
unsigned analysis_fft_size = 1u
<< static_cast<unsigned>(log2(resampled_length));
// Don't let the analysis size be smaller than the supported size
analysis_fft_size = std::max(analysis_fft_size, FFTFrame::MinFFTSize());
const unsigned convolver_fft_size = 2 * analysis_fft_size;
// Make sure this size of convolver is supported.
DCHECK_LE(convolver_fft_size, FFTFrame::MaxFFTSize());
return convolver_fft_size;
}
void HRTFPanner::Reset() {
convolver_l1_.Reset();
convolver_r1_.Reset();
convolver_l2_.Reset();
convolver_r2_.Reset();
delay_line_l_.Reset();
delay_line_r_.Reset();
}
void HRTFPanner::Pan(double desired_azimuth,
double elevation,
const AudioBus* input_bus,
AudioBus* output_bus,
uint32_t frames_to_process,
AudioBus::ChannelInterpretation channel_interpretation) {
const unsigned num_input_channels =
input_bus ? input_bus->NumberOfChannels() : 0;
DCHECK(input_bus);
DCHECK_GE(num_input_channels, 1u);
DCHECK_LE(num_input_channels, 2u);
DCHECK(output_bus);
DCHECK_EQ(output_bus->NumberOfChannels(), 2u);
DCHECK_LE(frames_to_process, output_bus->length());
const HRTFDatabase* const database = database_loader_->Database();
if (!database) {
output_bus->CopyFrom(*input_bus, channel_interpretation);
return;
}
// IRCAM HRTF azimuths values from the loaded database is reversed from the
// panner's notion of azimuth.
const double azimuth = -desired_azimuth;
DCHECK_GE(azimuth, -180.0);
DCHECK_LE(azimuth, 180.0);
// 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* input_channel_l =
input_bus->ChannelByType(AudioBus::kChannelLeft);
const AudioChannel* input_channel_r =
num_input_channels > 1 ? input_bus->ChannelByType(AudioBus::kChannelRight)
: nullptr;
// Get source and destination pointers.
const float* source_l = input_channel_l->Data();
const float* source_r =
num_input_channels > 1 ? input_channel_r->Data() : source_l;
float* destination_l =
output_bus->ChannelByType(AudioBus::kChannelLeft)->MutableData();
float* destination_r =
output_bus->ChannelByType(AudioBus::kChannelRight)->MutableData();
double azimuth_blend;
const int desired_azimuth_index =
CalculateDesiredAzimuthIndexAndBlend(azimuth, azimuth_blend);
// Initially snap azimuth and elevation values to first values encountered.
if (azimuth_index1_ == kUninitializedAzimuth) {
azimuth_index1_ = desired_azimuth_index;
elevation1_ = elevation;
}
if (azimuth_index2_ == kUninitializedAzimuth) {
azimuth_index2_ = desired_azimuth_index;
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 fade_frames = SampleRate() <= 48000 ? 2048 : 4096;
// Check for azimuth and elevation changes, initiating a cross-fade if needed.
if (!crossfade_x_ && crossfade_selection_ == kCrossfadeSelection1) {
if (desired_azimuth_index != azimuth_index1_ || elevation != elevation1_) {
// Cross-fade from 1 -> 2
crossfade_incr_ = 1 / fade_frames;
azimuth_index2_ = desired_azimuth_index;
elevation2_ = elevation;
}
}
if (crossfade_x_ == 1 && crossfade_selection_ == kCrossfadeSelection2) {
if (desired_azimuth_index != azimuth_index2_ || elevation != elevation2_) {
// Cross-fade from 2 -> 1
crossfade_incr_ = -1 / fade_frames;
azimuth_index1_ = desired_azimuth_index;
elevation1_ = elevation;
}
}
// This algorithm currently requires that we process in power-of-two size
// chunks of at least `RenderQuantumFrames()`.
DCHECK_EQ(1UL << static_cast<int>(log2(frames_to_process)),
frames_to_process);
DCHECK_GE(frames_to_process, RenderQuantumFrames());
const unsigned kFramesPerSegment = RenderQuantumFrames();
const unsigned number_of_segments = frames_to_process / kFramesPerSegment;
for (unsigned segment = 0; segment < number_of_segments; ++segment) {
// Get the HRTFKernels and interpolated delays.
HRTFKernel* kernel_l1;
HRTFKernel* kernel_r1;
HRTFKernel* kernel_l2;
HRTFKernel* kernel_r2;
double frame_delay_l1;
double frame_delay_r1;
double frame_delay_l2;
double frame_delay_r2;
database->GetKernelsFromAzimuthElevation(azimuth_blend, azimuth_index1_,
elevation1_, kernel_l1, kernel_r1,
frame_delay_l1, frame_delay_r1);
database->GetKernelsFromAzimuthElevation(azimuth_blend, azimuth_index2_,
elevation2_, kernel_l2, kernel_r2,
frame_delay_l2, frame_delay_r2);
DCHECK(kernel_l1);
DCHECK(kernel_r1);
DCHECK(kernel_l2);
DCHECK(kernel_r2);
DCHECK_LT(frame_delay_l1 / SampleRate(), kMaxDelayTimeSeconds);
DCHECK_LT(frame_delay_r1 / SampleRate(), kMaxDelayTimeSeconds);
DCHECK_LT(frame_delay_l2 / SampleRate(), kMaxDelayTimeSeconds);
DCHECK_LT(frame_delay_r2 / SampleRate(), kMaxDelayTimeSeconds);
// Crossfade inter-aural delays based on transitions.
const double frame_delay_l =
(1 - crossfade_x_) * frame_delay_l1 + crossfade_x_ * frame_delay_l2;
const double frame_delay_r =
(1 - crossfade_x_) * frame_delay_r1 + crossfade_x_ * frame_delay_r2;
// Calculate the source and destination pointers for the current segment.
const unsigned offset = segment * kFramesPerSegment;
const float* segment_source_l = source_l + offset;
const float* segment_source_r = source_r + offset;
float* segment_destination_l = destination_l + offset;
float* segment_destination_r = destination_r + offset;
// First run through delay lines for inter-aural time difference.
delay_line_l_.SetDelayFrames(frame_delay_l);
delay_line_r_.SetDelayFrames(frame_delay_r);
delay_line_l_.ProcessKRate(segment_source_l, segment_destination_l,
kFramesPerSegment);
delay_line_r_.ProcessKRate(segment_source_r, segment_destination_r,
kFramesPerSegment);
const bool needs_crossfading = crossfade_incr_;
// Have the convolvers render directly to the final destination if we're not
// cross-fading.
float* convolution_destination_l1 =
needs_crossfading ? temp_l1_.Data() : segment_destination_l;
float* convolution_destination_r1 =
needs_crossfading ? temp_r1_.Data() : segment_destination_r;
float* convolution_destination_l2 =
needs_crossfading ? temp_l2_.Data() : segment_destination_l;
float* convolution_destination_r2 =
needs_crossfading ? temp_r2_.Data() : segment_destination_r;
// Now do the convolutions.
// Note that we avoid doing convolutions on both sets of convolvers if we're
// not currently cross-fading.
if (crossfade_selection_ == kCrossfadeSelection1 || needs_crossfading) {
convolver_l1_.Process(kernel_l1->FftFrame(), segment_destination_l,
convolution_destination_l1, kFramesPerSegment);
convolver_r1_.Process(kernel_r1->FftFrame(), segment_destination_r,
convolution_destination_r1, kFramesPerSegment);
}
if (crossfade_selection_ == kCrossfadeSelection2 || needs_crossfading) {
convolver_l2_.Process(kernel_l2->FftFrame(), segment_destination_l,
convolution_destination_l2, kFramesPerSegment);
convolver_r2_.Process(kernel_r2->FftFrame(), segment_destination_r,
convolution_destination_r2, kFramesPerSegment);
}
if (needs_crossfading) {
// Apply linear cross-fade.
float x = crossfade_x_;
const float incr = crossfade_incr_;
for (unsigned i = 0; i < kFramesPerSegment; ++i) {
segment_destination_l[i] = (1 - x) * convolution_destination_l1[i] +
x * convolution_destination_l2[i];
segment_destination_r[i] = (1 - x) * convolution_destination_r1[i] +
x * convolution_destination_r2[i];
x += incr;
}
// Update cross-fade value from local.
crossfade_x_ = x;
if (crossfade_incr_ > 0 && fabs(crossfade_x_ - 1) < crossfade_incr_) {
// We've fully made the crossfade transition from 1 -> 2.
crossfade_selection_ = kCrossfadeSelection2;
crossfade_x_ = 1;
crossfade_incr_ = 0;
} else if (crossfade_incr_ < 0 && fabs(crossfade_x_) < -crossfade_incr_) {
// We've fully made the crossfade transition from 2 -> 1.
crossfade_selection_ = kCrossfadeSelection1;
crossfade_x_ = 0;
crossfade_incr_ = 0;
}
}
}
}
void HRTFPanner::PanWithSampleAccurateValues(
base::span<double> desired_azimuth,
base::span<double> elevation,
const AudioBus* input_bus,
AudioBus* output_bus,
uint32_t frames_to_process,
AudioBus::ChannelInterpretation channel_interpretation) {
// 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 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(desired_azimuth[0], elevation[0], input_bus, output_bus,
frames_to_process, channel_interpretation);
}
bool HRTFPanner::RequiresTailProcessing() const {
// Always return true since the tail and latency are never zero.
return true;
}
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 kMaxDelayTimeSeconds +
(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 blink
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