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Rename Beamformer to NonlinearBeamformer.

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R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42359004

Cr-Commit-Position: refs/heads/master@{#8710}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8710 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mgraczyk@chromium.org 2015-03-13 00:13:32 +00:00
parent 86639737b8
commit 0f663de2ec
13 changed files with 66 additions and 58 deletions
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4
webrtc/modules/audio_processing/BUILD.gn
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@ -170,8 +170,8 @@ source_set("audio_processing") {
if (rtc_use_openmax_dl) {
defines += [ "WEBRTC_BEAMFORMER" ]
sources += [
"beamformer/beamformer.cc",
"beamformer/beamformer.h",
"beamformer/nonlinear_beamformer.cc",
"beamformer/nonlinear_beamformer.h",
]
}

4
webrtc/modules/audio_processing/audio_processing.gypi
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@ -179,8 +179,8 @@
['rtc_use_openmax_dl==1', {
'defines': ['WEBRTC_BEAMFORMER'],
'sources': [
'beamformer/beamformer.cc',
'beamformer/beamformer.h',
'beamformer/nonlinear_beamformer.cc',
'beamformer/nonlinear_beamformer.h',
],
}],
['target_arch=="ia32" or target_arch=="x64"', {

10
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -17,7 +17,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
@ -45,7 +45,7 @@
#define RETURN_ON_ERR(expr) \
do { \
int err = expr; \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
@ -134,7 +134,7 @@ AudioProcessing* AudioProcessing::Create(const Config& config) {
}
AudioProcessing* AudioProcessing::Create(const Config& config,
Beamformer* beamformer) {
NonlinearBeamformer* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
@ -148,7 +148,7 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
: AudioProcessingImpl(config, nullptr) {}
AudioProcessingImpl::AudioProcessingImpl(const Config& config,
Beamformer* beamformer)
NonlinearBeamformer* beamformer)
: echo_cancellation_(NULL),
echo_control_mobile_(NULL),
gain_control_(NULL),
@ -988,7 +988,7 @@ void AudioProcessingImpl::InitializeBeamformer() {
if (beamformer_enabled_) {
#ifdef WEBRTC_BEAMFORMER
if (!beamformer_) {
beamformer_.reset(new Beamformer(array_geometry_));
beamformer_.reset(new NonlinearBeamformer(array_geometry_));
}
beamformer_->Initialize(kChunkSizeMs, split_rate_);
#else

6
webrtc/modules/audio_processing/audio_processing_impl.h
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@ -23,7 +23,7 @@ namespace webrtc {
class AgcManagerDirect;
class AudioBuffer;
class Beamformer;
class NonlinearBeamformer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
@ -87,7 +87,7 @@ class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
// Only for testing.
AudioProcessingImpl(const Config& config, Beamformer* beamformer);
AudioProcessingImpl(const Config& config, NonlinearBeamformer* beamformer);
virtual ~AudioProcessingImpl();
// AudioProcessing methods.
@ -218,7 +218,7 @@ class AudioProcessingImpl : public AudioProcessing {
bool transient_suppressor_enabled_;
rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
const bool beamformer_enabled_;
rtc::scoped_ptr<Beamformer> beamformer_;
rtc::scoped_ptr<NonlinearBeamformer> beamformer_;
const std::vector<Point> array_geometry_;
const bool supports_48kHz_;

6
webrtc/modules/audio_processing/audio_processing_tests.gypi
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@ -88,18 +88,18 @@
['rtc_use_openmax_dl==1', {
'targets': [
{
'target_name': 'beamformer_test',
'target_name': 'nonlinear_beamformer_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
],
'sources': [
'beamformer/beamformer_test.cc',
'beamformer/nonlinear_beamformer_test.cc',
'beamformer/pcm_utils.cc',
'beamformer/pcm_utils.h',
],
}, # beamformer_test
}, # nonlinear_beamformer_test
],
}],
],

10
webrtc/modules/audio_processing/beamformer/mock_beamformer.cc → webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
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@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/beamformer/mock_beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
#include <vector>
namespace webrtc {
MockBeamformer::MockBeamformer(const std::vector<Point>& array_geometry)
: Beamformer(array_geometry) {}
MockNonlinearBeamformer::MockNonlinearBeamformer(
const std::vector<Point>& array_geometry)
: NonlinearBeamformer(array_geometry) {
}
MockBeamformer::~MockBeamformer() {}
MockNonlinearBeamformer::~MockNonlinearBeamformer() {}
} // namespace webrtc

8
webrtc/modules/audio_processing/beamformer/mock_beamformer.h → webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
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@ -14,14 +14,14 @@
#include <vector>
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
namespace webrtc {
class MockBeamformer : public Beamformer {
class MockNonlinearBeamformer : public NonlinearBeamformer {
public:
explicit MockBeamformer(const std::vector<Point>& array_geometry);
~MockBeamformer() override;
explicit MockNonlinearBeamformer(const std::vector<Point>& array_geometry);
~MockNonlinearBeamformer() override;
MOCK_METHOD2(Initialize, void(int chunk_size_ms, int sample_rate_hz));
MOCK_METHOD2(ProcessChunk, void(const ChannelBuffer<float>* input,

41
webrtc/modules/audio_processing/beamformer/beamformer.cc → webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
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@ -10,10 +10,11 @@
#define _USE_MATH_DEFINES
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
#include <algorithm>
#include <cmath>
#include <vector>
#include "webrtc/base/arraysize.h"
#include "webrtc/common_audio/window_generator.h"
@ -174,13 +175,14 @@ std::vector<Point> GetCenteredArray(std::vector<Point> array_geometry) {
} // namespace
Beamformer::Beamformer(const std::vector<Point>& array_geometry)
: num_input_channels_(array_geometry.size()),
NonlinearBeamformer::NonlinearBeamformer(
const std::vector<Point>& array_geometry)
: num_input_channels_(array_geometry.size()),
array_geometry_(GetCenteredArray(array_geometry)) {
WindowGenerator::KaiserBesselDerived(kAlpha, kFftSize, window_);
}
void Beamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
void NonlinearBeamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
chunk_length_ = sample_rate_hz / (1000.f / chunk_size_ms);
sample_rate_hz_ = sample_rate_hz;
low_average_start_bin_ =
@ -230,7 +232,7 @@ void Beamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
}
}
void Beamformer::InitDelaySumMasks() {
void NonlinearBeamformer::InitDelaySumMasks() {
for (int f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
delay_sum_masks_[f_ix].Resize(1, num_input_channels_);
CovarianceMatrixGenerator::PhaseAlignmentMasks(f_ix,
@ -250,7 +252,7 @@ void Beamformer::InitDelaySumMasks() {
}
}
void Beamformer::InitTargetCovMats() {
void NonlinearBeamformer::InitTargetCovMats() {
for (int i = 0; i < kNumFreqBins; ++i) {
target_cov_mats_[i].Resize(num_input_channels_, num_input_channels_);
TransposedConjugatedProduct(delay_sum_masks_[i], &target_cov_mats_[i]);
@ -259,7 +261,7 @@ void Beamformer::InitTargetCovMats() {
}
}
void Beamformer::InitInterfCovMats() {
void NonlinearBeamformer::InitInterfCovMats() {
for (int i = 0; i < kNumFreqBins; ++i) {
interf_cov_mats_[i].Resize(num_input_channels_, num_input_channels_);
ComplexMatrixF uniform_cov_mat(num_input_channels_, num_input_channels_);
@ -291,7 +293,7 @@ void Beamformer::InitInterfCovMats() {
}
}
void Beamformer::ProcessChunk(const ChannelBuffer<float>* input,
void NonlinearBeamformer::ProcessChunk(const ChannelBuffer<float>* input,
ChannelBuffer<float>* output) {
DCHECK_EQ(input->num_channels(), num_input_channels_);
DCHECK_EQ(input->num_frames_per_band(), chunk_length_);
@ -321,7 +323,7 @@ void Beamformer::ProcessChunk(const ChannelBuffer<float>* input,
}
}
void Beamformer::ProcessAudioBlock(const complex_f* const* input,
void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
int num_input_channels,
int num_freq_bins,
int num_output_channels,
@ -371,11 +373,12 @@ void Beamformer::ProcessAudioBlock(const complex_f* const* input,
EstimateTargetPresence();
}
float Beamformer::CalculatePostfilterMask(const ComplexMatrixF& interf_cov_mat,
float rpsiw,
float ratio_rxiw_rxim,
float rmw_r,
float mask_threshold) {
float NonlinearBeamformer::CalculatePostfilterMask(
const ComplexMatrixF& interf_cov_mat,
float rpsiw,
float ratio_rxiw_rxim,
float rmw_r,
float mask_threshold) {
float rpsim = Norm(interf_cov_mat, eig_m_);
// Find lambda.
@ -394,7 +397,7 @@ float Beamformer::CalculatePostfilterMask(const ComplexMatrixF& interf_cov_mat,
return mask;
}
void Beamformer::ApplyMasks(const complex_f* const* input,
void NonlinearBeamformer::ApplyMasks(const complex_f* const* input,
complex_f* const* output) {
complex_f* output_channel = output[0];
for (int f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
@ -410,14 +413,14 @@ void Beamformer::ApplyMasks(const complex_f* const* input,
}
}
void Beamformer::ApplyMaskSmoothing() {
void NonlinearBeamformer::ApplyMaskSmoothing() {
for (int i = 0; i < kNumFreqBins; ++i) {
postfilter_mask_[i] = kMaskSmoothAlpha * new_mask_[i] +
(1.f - kMaskSmoothAlpha) * postfilter_mask_[i];
}
}
void Beamformer::ApplyLowFrequencyCorrection() {
void NonlinearBeamformer::ApplyLowFrequencyCorrection() {
float low_frequency_mask = 0.f;
for (int i = low_average_start_bin_; i < low_average_end_bin_; ++i) {
low_frequency_mask += postfilter_mask_[i];
@ -430,7 +433,7 @@ void Beamformer::ApplyLowFrequencyCorrection() {
}
}
void Beamformer::ApplyHighFrequencyCorrection() {
void NonlinearBeamformer::ApplyHighFrequencyCorrection() {
high_pass_postfilter_mask_ = 0.f;
for (int i = high_average_start_bin_; i < high_average_end_bin_; ++i) {
high_pass_postfilter_mask_ += postfilter_mask_[i];
@ -443,7 +446,7 @@ void Beamformer::ApplyHighFrequencyCorrection() {
}
}
void Beamformer::EstimateTargetPresence() {
void NonlinearBeamformer::EstimateTargetPresence() {
const int quantile = (1.f - kMaskQuantile) * high_average_end_bin_ +
kMaskQuantile * low_average_start_bin_;
std::nth_element(new_mask_ + low_average_start_bin_,

15
webrtc/modules/audio_processing/beamformer/beamformer.h → webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
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@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
#include <vector>
#include "webrtc/common_audio/lapped_transform.h"
#include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
@ -25,15 +27,14 @@ namespace webrtc {
// Beamforming Postprocessor" by Bastiaan Kleijn.
//
// TODO: Target angle assumed to be 0. Parameterize target angle.
class Beamformer : public LappedTransform::Callback {
class NonlinearBeamformer : public LappedTransform::Callback {
public:
// At the moment it only accepts uniform linear microphone arrays. Using the
// first microphone as a reference position [0, 0, 0] is a natural choice.
explicit Beamformer(const std::vector<Point>& array_geometry);
virtual ~Beamformer() {};
explicit NonlinearBeamformer(const std::vector<Point>& array_geometry);
// Sample rate corresponds to the lower band.
// Needs to be called before the Beamformer can be used.
// Needs to be called before the NonlinearBeamformer can be used.
virtual void Initialize(int chunk_size_ms, int sample_rate_hz);
// Process one time-domain chunk of audio. The audio is expected to be split
@ -160,4 +161,4 @@ class Beamformer : public LappedTransform::Callback {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_

5
webrtc/modules/audio_processing/beamformer/beamformer_test.cc → webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
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@ -9,9 +9,10 @@
*/
#include <iostream>
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/pcm_utils.h"
DEFINE_int32(sample_rate,
@ -59,7 +60,7 @@ int main(int argc, char* argv[]) {
for (int i = 0; i < FLAGS_num_input_channels; ++i) {
array_geometry.push_back(webrtc::Point(i * FLAGS_mic_spacing, 0.f, 0.f));
}
webrtc::Beamformer bf(array_geometry);
webrtc::NonlinearBeamformer bf(array_geometry);
bf.Initialize(kChunkTimeMilliseconds, FLAGS_sample_rate);
while (true) {
size_t samples_read = webrtc::PcmReadToFloat(read_file,

5
webrtc/modules/audio_processing/include/audio_processing.h
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@ -25,7 +25,7 @@ struct AecCore;
namespace webrtc {
class AudioFrame;
class Beamformer;
class NonlinearBeamformer;
class EchoCancellation;
class EchoControlMobile;
class GainControl;
@ -201,7 +201,8 @@ class AudioProcessing {
// Allows passing in an optional configuration at create-time.
static AudioProcessing* Create(const Config& config);
// Only for testing.
static AudioProcessing* Create(const Config& config, Beamformer* beamformer);
static AudioProcessing* Create(const Config& config,
NonlinearBeamformer* beamformer);
virtual ~AudioProcessing() {}
// Initializes internal states, while retaining all user settings. This

6
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
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@ -19,7 +19,7 @@
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/beamformer/mock_beamformer.h"
#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
@ -1228,8 +1228,8 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
geometry.push_back(webrtc::Point(0.f, 0.f, 0.f));
geometry.push_back(webrtc::Point(0.05f, 0.f, 0.f));
config.Set<Beamforming>(new Beamforming(true, geometry));
testing::NiceMock<MockBeamformer>* beamformer =
new testing::NiceMock<MockBeamformer>(geometry);
testing::NiceMock<MockNonlinearBeamformer>* beamformer =
new testing::NiceMock<MockNonlinearBeamformer>(geometry);
rtc::scoped_ptr<AudioProcessing> apm(
AudioProcessing::Create(config, beamformer));
EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true));

4
webrtc/modules/modules.gyp
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@ -170,8 +170,8 @@
'audio_processing/beamformer/complex_matrix_unittest.cc',
'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
'audio_processing/beamformer/matrix_unittest.cc',
'audio_processing/beamformer/mock_beamformer.cc',
'audio_processing/beamformer/mock_beamformer.h',
'audio_processing/beamformer/mock_nonlinear_beamformer.cc',
'audio_processing/beamformer/mock_nonlinear_beamformer.h',
'audio_processing/beamformer/pcm_utils.cc',
'audio_processing/beamformer/pcm_utils.h',
'audio_processing/echo_cancellation_impl_unittest.cc',