Reparent Nonlinear beamformer under beamforming interface.
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41269004
Cr-Original-Commit-Position: refs/heads/master@{#8862}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: dfa36058c945cf2ef9932a566987f648c24fa632
diff --git a/common_audio/lapped_transform.h b/common_audio/lapped_transform.h
index 3ed9528..9f6b302 100644
--- a/common_audio/lapped_transform.h
+++ b/common_audio/lapped_transform.h
@@ -13,7 +13,6 @@
#include <complex>
-#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/blocker.h"
#include "webrtc/common_audio/real_fourier.h"
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index cc11603..016c684 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -72,6 +72,7 @@
"audio_buffer.h",
"audio_processing_impl.cc",
"audio_processing_impl.h",
+ "beamformer/beamformer.h",
"beamformer/complex_matrix.h",
"beamformer/covariance_matrix_generator.cc",
"beamformer/covariance_matrix_generator.h",
diff --git a/modules/audio_processing/audio_processing.gypi b/modules/audio_processing/audio_processing.gypi
index 0b19fd9..3ceeed8 100644
--- a/modules/audio_processing/audio_processing.gypi
+++ b/modules/audio_processing/audio_processing.gypi
@@ -82,6 +82,7 @@
'audio_buffer.h',
'audio_processing_impl.cc',
'audio_processing_impl.h',
+ 'beamformer/beamformer.h',
'beamformer/complex_matrix.h',
'beamformer/covariance_matrix_generator.cc',
'beamformer/covariance_matrix_generator.h',
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index e989708..eeb9a79 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -14,11 +14,11 @@
#include "webrtc/base/platform_file.h"
#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/channel_buffer.h"
#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/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"
#include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
@@ -134,7 +134,7 @@
}
AudioProcessing* AudioProcessing::Create(const Config& config,
- NonlinearBeamformer* beamformer) {
+ Beamformer<float>* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
@@ -148,7 +148,7 @@
: AudioProcessingImpl(config, nullptr) {}
AudioProcessingImpl::AudioProcessingImpl(const Config& config,
- NonlinearBeamformer* beamformer)
+ Beamformer<float>* beamformer)
: echo_cancellation_(NULL),
echo_control_mobile_(NULL),
gain_control_(NULL),
@@ -600,7 +600,7 @@
}
if (beamformer_enabled_) {
- beamformer_->ProcessChunk(ca->split_data_f(), ca->split_data_f());
+ beamformer_->ProcessChunk(*ca->split_data_f(), ca->split_data_f());
ca->set_num_channels(1);
}
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index b5114cf..765cde7 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -11,19 +11,21 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-
#include <list>
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class AgcManagerDirect;
class AudioBuffer;
-class NonlinearBeamformer;
+
+template<typename T>
+class Beamformer;
+
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
@@ -86,8 +88,9 @@
class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
- // Only for testing.
- AudioProcessingImpl(const Config& config, NonlinearBeamformer* beamformer);
+
+ // AudioProcessingImpl takes ownership of beamformer.
+ AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
virtual ~AudioProcessingImpl();
// AudioProcessing methods.
@@ -218,7 +221,7 @@
bool transient_suppressor_enabled_;
rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
const bool beamformer_enabled_;
- rtc::scoped_ptr<NonlinearBeamformer> beamformer_;
+ rtc::scoped_ptr<Beamformer<float>> beamformer_;
const std::vector<Point> array_geometry_;
const bool supports_48kHz_;
diff --git a/modules/audio_processing/beamformer/beamformer.h b/modules/audio_processing/beamformer/beamformer.h
new file mode 100644
index 0000000..04cb659
--- /dev/null
+++ b/modules/audio_processing/beamformer/beamformer.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * 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_
+
+#include "webrtc/common_audio/channel_buffer.h"
+
+namespace webrtc {
+
+template<typename T>
+class Beamformer {
+ public:
+ virtual ~Beamformer() {}
+
+ // Process one time-domain chunk of audio. The audio is expected to be split
+ // into frequency bands inside the ChannelBuffer. The number of frames and
+ // channels must correspond to the constructor parameters. The same
+ // ChannelBuffer can be passed in as |input| and |output|.
+ virtual void ProcessChunk(const ChannelBuffer<T>& input,
+ ChannelBuffer<T>* output) = 0;
+
+ // Sample rate corresponds to the lower band.
+ // Needs to be called before the the Beamformer can be used.
+ virtual void Initialize(int chunk_size_ms, int sample_rate_hz) = 0;
+
+ // Returns true if the current data contains the target signal.
+ // Which signals are considered "targets" is implementation dependent.
+ virtual bool is_target_present() = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
diff --git a/modules/audio_processing/beamformer/complex_matrix.h b/modules/audio_processing/beamformer/complex_matrix.h
index 391050b..f5be2b2 100644
--- a/modules/audio_processing/beamformer/complex_matrix.h
+++ b/modules/audio_processing/beamformer/complex_matrix.h
@@ -15,7 +15,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/beamformer/matrix.h"
namespace webrtc {
diff --git a/modules/audio_processing/beamformer/matrix.h b/modules/audio_processing/beamformer/matrix.h
index 9e485ef..990f6a4 100644
--- a/modules/audio_processing/beamformer/matrix.h
+++ b/modules/audio_processing/beamformer/matrix.h
@@ -12,13 +12,13 @@
#define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_H_
#include <algorithm>
+#include <cstring>
#include <string>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_audio/channel_buffer.h"
namespace {
diff --git a/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc b/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
index 4a1936e..aecb0ec 100644
--- a/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
+++ b/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
@@ -19,6 +19,4 @@
: NonlinearBeamformer(array_geometry) {
}
-MockNonlinearBeamformer::~MockNonlinearBeamformer() {}
-
} // namespace webrtc
diff --git a/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h b/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
index 56e647b..eb05ecd 100644
--- a/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
+++ b/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
@@ -21,10 +21,9 @@
class MockNonlinearBeamformer : public NonlinearBeamformer {
public:
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,
+ MOCK_METHOD2(ProcessChunk, void(const ChannelBuffer<float>& input,
ChannelBuffer<float>* output));
MOCK_METHOD0(is_target_present, bool());
};
diff --git a/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/modules/audio_processing/beamformer/nonlinear_beamformer.cc
index 9630b7d..8fd6c68 100644
--- a/modules/audio_processing/beamformer/nonlinear_beamformer.cc
+++ b/modules/audio_processing/beamformer/nonlinear_beamformer.cc
@@ -293,32 +293,32 @@
}
}
-void NonlinearBeamformer::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_);
+ DCHECK_EQ(input.num_channels(), num_input_channels_);
+ DCHECK_EQ(input.num_frames_per_band(), chunk_length_);
float old_high_pass_mask = high_pass_postfilter_mask_;
- lapped_transform_->ProcessChunk(input->channels(0), output->channels(0));
+ lapped_transform_->ProcessChunk(input.channels(0), output->channels(0));
// Ramp up/down for smoothing. 1 mask per 10ms results in audible
// discontinuities.
const float ramp_increment =
(high_pass_postfilter_mask_ - old_high_pass_mask) /
- input->num_frames_per_band();
+ input.num_frames_per_band();
// Apply delay and sum and post-filter in the time domain. WARNING: only works
// because delay-and-sum is not frequency dependent.
- for (int i = 1; i < input->num_bands(); ++i) {
+ for (int i = 1; i < input.num_bands(); ++i) {
float smoothed_mask = old_high_pass_mask;
- for (int j = 0; j < input->num_frames_per_band(); ++j) {
+ for (int j = 0; j < input.num_frames_per_band(); ++j) {
smoothed_mask += ramp_increment;
// Applying the delay and sum (at zero degrees, this is equivalent to
// averaging).
float sum = 0.f;
- for (int k = 0; k < input->num_channels(); ++k) {
- sum += input->channels(i)[k][j];
+ for (int k = 0; k < input.num_channels(); ++k) {
+ sum += input.channels(i)[k][j];
}
- output->channels(i)[0][j] = sum / input->num_channels() * smoothed_mask;
+ output->channels(i)[0][j] = sum / input.num_channels() * smoothed_mask;
}
}
}
diff --git a/modules/audio_processing/beamformer/nonlinear_beamformer.h b/modules/audio_processing/beamformer/nonlinear_beamformer.h
index 91e47cd..bebfad8 100644
--- a/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/modules/audio_processing/beamformer/nonlinear_beamformer.h
@@ -14,8 +14,10 @@
#include <vector>
#include "webrtc/common_audio/lapped_transform.h"
-#include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
+#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/beamformer/array_util.h"
+#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
+#include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
namespace webrtc {
@@ -27,7 +29,9 @@
// Beamforming Postprocessor" by Bastiaan Kleijn.
//
// TODO: Target angle assumed to be 0. Parameterize target angle.
-class NonlinearBeamformer : public LappedTransform::Callback {
+class NonlinearBeamformer
+ : public Beamformer<float>,
+ 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.
@@ -35,19 +39,20 @@
// Sample rate corresponds to the lower band.
// Needs to be called before the NonlinearBeamformer can be used.
- virtual void Initialize(int chunk_size_ms, int sample_rate_hz);
+ void Initialize(int chunk_size_ms, int sample_rate_hz) override;
// Process one time-domain chunk of audio. The audio is expected to be split
// into frequency bands inside the ChannelBuffer. The number of frames and
// channels must correspond to the constructor parameters. The same
// ChannelBuffer can be passed in as |input| and |output|.
- virtual void ProcessChunk(const ChannelBuffer<float>* input,
- ChannelBuffer<float>* output);
+ void ProcessChunk(const ChannelBuffer<float>& input,
+ ChannelBuffer<float>* output) override;
+
// After processing each block |is_target_present_| is set to true if the
// target signal es present and to false otherwise. This methods can be called
// to know if the data is target signal or interference and process it
// accordingly.
- virtual bool is_target_present() { return is_target_present_; }
+ bool is_target_present() override { return is_target_present_; }
protected:
// Process one frequency-domain block of audio. This is where the fun
diff --git a/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
index 9d85ec5..48d7c2b 100644
--- a/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
+++ b/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
@@ -72,7 +72,7 @@
break;
}
- bf.ProcessChunk(&captured_audio_cb, &captured_audio_cb);
+ bf.ProcessChunk(captured_audio_cb, &captured_audio_cb);
webrtc::PcmWriteFromFloat(
write_file, kChunkSize, 1, captured_audio_cb.channels());
}
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 7c230d3..72553ff 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -25,7 +25,10 @@
namespace webrtc {
class AudioFrame;
-class NonlinearBeamformer;
+
+template<typename T>
+class Beamformer;
+
class EchoCancellation;
class EchoControlMobile;
class GainControl;
@@ -202,7 +205,7 @@
static AudioProcessing* Create(const Config& config);
// Only for testing.
static AudioProcessing* Create(const Config& config,
- NonlinearBeamformer* beamformer);
+ Beamformer<float>* beamformer);
virtual ~AudioProcessing() {}
// Initializes internal states, while retaining all user settings. This
