Moved the GainControlForNewAGC class to be a separate file.
Apart from being motivated in order to make the source files shorter
this is needed when separating the APM submodules structs into
separate files.
BUG=
Review URL: https://codereview.webrtc.org/1678813002
Cr-Commit-Position: refs/heads/master@{#11611}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 13da8d6..5834793 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -68,6 +68,8 @@
"echo_cancellation_impl.h",
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
+ "gain_control_for_experimental_agc.cc",
+ "gain_control_for_experimental_agc.h",
"gain_control_impl.cc",
"gain_control_impl.h",
"high_pass_filter_impl.cc",
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index c665f5b..eaee644 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -78,6 +78,8 @@
'echo_cancellation_impl.h',
'echo_control_mobile_impl.cc',
'echo_control_mobile_impl.h',
+ 'gain_control_for_experimental_agc.cc',
+ 'gain_control_for_experimental_agc.h',
'gain_control_impl.cc',
'gain_control_impl.h',
'high_pass_filter_impl.cc',
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index b3f38f4..f385612 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -29,6 +29,7 @@
#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"
+#include "webrtc/modules/audio_processing/gain_control_for_experimental_agc.h"
#include "webrtc/modules/audio_processing/gain_control_impl.h"
#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
@@ -80,72 +81,6 @@
// Throughout webrtc, it's assumed that success is represented by zero.
static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
-// This class has two main functionalities:
-//
-// 1) It is returned instead of the real GainControl after the new AGC has been
-// enabled in order to prevent an outside user from overriding compression
-// settings. It doesn't do anything in its implementation, except for
-// delegating the const methods and Enable calls to the real GainControl, so
-// AGC can still be disabled.
-//
-// 2) It is injected into AgcManagerDirect and implements volume callbacks for
-// getting and setting the volume level. It just caches this value to be used
-// in VoiceEngine later.
-class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
- public:
- explicit GainControlForNewAgc(GainControlImpl* gain_control)
- : real_gain_control_(gain_control), volume_(0) {}
-
- // GainControl implementation.
- int Enable(bool enable) override {
- return real_gain_control_->Enable(enable);
- }
- bool is_enabled() const override { return real_gain_control_->is_enabled(); }
- int set_stream_analog_level(int level) override {
- volume_ = level;
- return AudioProcessing::kNoError;
- }
- int stream_analog_level() override { return volume_; }
- int set_mode(Mode mode) override { return AudioProcessing::kNoError; }
- Mode mode() const override { return GainControl::kAdaptiveAnalog; }
- int set_target_level_dbfs(int level) override {
- return AudioProcessing::kNoError;
- }
- int target_level_dbfs() const override {
- return real_gain_control_->target_level_dbfs();
- }
- int set_compression_gain_db(int gain) override {
- return AudioProcessing::kNoError;
- }
- int compression_gain_db() const override {
- return real_gain_control_->compression_gain_db();
- }
- int enable_limiter(bool enable) override { return AudioProcessing::kNoError; }
- bool is_limiter_enabled() const override {
- return real_gain_control_->is_limiter_enabled();
- }
- int set_analog_level_limits(int minimum, int maximum) override {
- return AudioProcessing::kNoError;
- }
- int analog_level_minimum() const override {
- return real_gain_control_->analog_level_minimum();
- }
- int analog_level_maximum() const override {
- return real_gain_control_->analog_level_maximum();
- }
- bool stream_is_saturated() const override {
- return real_gain_control_->stream_is_saturated();
- }
-
- // VolumeCallbacks implementation.
- void SetMicVolume(int volume) override { volume_ = volume; }
- int GetMicVolume() override { return volume_; }
-
- private:
- GainControl* real_gain_control_;
- int volume_;
-};
-
struct AudioProcessingImpl::ApmPublicSubmodules {
ApmPublicSubmodules()
: echo_cancellation(nullptr),
@@ -159,7 +94,8 @@
rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
rtc::scoped_ptr<VoiceDetectionImpl> voice_detection;
- rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
+ rtc::scoped_ptr<GainControlForExperimentalAgc>
+ gain_control_for_experimental_agc;
// Accessed internally from both render and capture.
rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
@@ -248,8 +184,9 @@
new NoiseSuppressionImpl(&crit_capture_));
public_submodules_->voice_detection.reset(
new VoiceDetectionImpl(&crit_capture_));
- public_submodules_->gain_control_for_new_agc.reset(
- new GainControlForNewAgc(public_submodules_->gain_control));
+ public_submodules_->gain_control_for_experimental_agc.reset(
+ new GainControlForExperimentalAgc(public_submodules_->gain_control,
+ &crit_capture_));
private_submodules_->component_list.push_back(
public_submodules_->echo_cancellation);
@@ -264,10 +201,10 @@
AudioProcessingImpl::~AudioProcessingImpl() {
// Depends on gain_control_ and
- // public_submodules_->gain_control_for_new_agc.
+ // public_submodules_->gain_control_for_experimental_agc.
private_submodules_->agc_manager.reset();
// Depends on gain_control_.
- public_submodules_->gain_control_for_new_agc.reset();
+ public_submodules_->gain_control_for_experimental_agc.reset();
while (!private_submodules_->component_list.empty()) {
ProcessingComponent* component =
private_submodules_->component_list.front();
@@ -759,7 +696,7 @@
AudioBuffer* ca = capture_.capture_audio.get(); // For brevity.
- if (constants_.use_new_agc &&
+ if (constants_.use_experimental_agc &&
public_submodules_->gain_control->is_enabled()) {
private_submodules_->agc_manager->AnalyzePreProcess(
ca->channels()[0], ca->num_channels(),
@@ -796,7 +733,7 @@
public_submodules_->echo_control_mobile->ProcessCaptureAudio(ca));
public_submodules_->voice_detection->ProcessCaptureAudio(ca);
- if (constants_.use_new_agc &&
+ if (constants_.use_experimental_agc &&
public_submodules_->gain_control->is_enabled() &&
(!capture_nonlocked_.beamformer_enabled ||
private_submodules_->beamformer->is_target_present())) {
@@ -998,7 +935,7 @@
RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
RETURN_ON_ERR(
public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
- if (!constants_.use_new_agc) {
+ if (!constants_.use_experimental_agc) {
RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
}
@@ -1164,8 +1101,8 @@
GainControl* AudioProcessingImpl::gain_control() const {
// Adding a lock here has no effect as it allows any access to the submodule
// from the returned pointer.
- if (constants_.use_new_agc) {
- return public_submodules_->gain_control_for_new_agc.get();
+ if (constants_.use_experimental_agc) {
+ return public_submodules_->gain_control_for_experimental_agc.get();
}
return public_submodules_->gain_control;
}
@@ -1284,11 +1221,11 @@
}
void AudioProcessingImpl::InitializeExperimentalAgc() {
- if (constants_.use_new_agc) {
+ if (constants_.use_experimental_agc) {
if (!private_submodules_->agc_manager.get()) {
private_submodules_->agc_manager.reset(new AgcManagerDirect(
public_submodules_->gain_control,
- public_submodules_->gain_control_for_new_agc.get(),
+ public_submodules_->gain_control_for_experimental_agc.get(),
constants_.agc_startup_min_volume));
}
private_submodules_->agc_manager->Initialize();
@@ -1519,7 +1456,7 @@
static_cast<int>(public_submodules_->gain_control->mode()));
config.set_agc_limiter_enabled(
public_submodules_->gain_control->is_limiter_enabled());
- config.set_noise_robust_agc_enabled(constants_.use_new_agc);
+ config.set_noise_robust_agc_enabled(constants_.use_experimental_agc);
config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index a19d3fc..b3f43fa 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -274,14 +274,14 @@
// APM constants.
const struct ApmConstants {
ApmConstants(int agc_startup_min_volume,
- bool use_new_agc,
+ bool use_experimental_agc,
bool intelligibility_enabled)
: // Format of processing streams at input/output call sites.
agc_startup_min_volume(agc_startup_min_volume),
- use_new_agc(use_new_agc),
+ use_experimental_agc(use_experimental_agc),
intelligibility_enabled(intelligibility_enabled) {}
int agc_startup_min_volume;
- bool use_new_agc;
+ bool use_experimental_agc;
bool intelligibility_enabled;
} constants_;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index 2e22b2c..7ce6a56 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -600,7 +600,6 @@
(test_config_->aec_type ==
AecType::BasicWebRtcAecSettingsWithAecMobile));
EXPECT_TRUE(apm_->gain_control()->is_enabled());
- apm_->gain_control()->stream_analog_level();
EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
// The below return values are not testable.
@@ -713,9 +712,12 @@
// Prepare a proper capture side processing API call input.
PrepareFrame();
- // Set the stream delay
+ // Set the stream delay.
apm_->set_stream_delay_ms(30);
+ // Set the analog level.
+ apm_->gain_control()->set_stream_analog_level(80);
+
// Call the specified capture side API processing method.
int result = AudioProcessing::kNoError;
switch (test_config_->capture_api_function) {
@@ -738,6 +740,9 @@
FAIL();
}
+ // Retrieve the new analog level.
+ apm_->gain_control()->stream_analog_level();
+
// Check the return code for error.
ASSERT_EQ(AudioProcessing::kNoError, result);
}
diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
new file mode 100644
index 0000000..2ef88c0
--- /dev/null
+++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
@@ -0,0 +1,104 @@
+/*
+ * Copyright (c) 2016 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.
+ */
+
+#include "webrtc/modules/audio_processing/gain_control_for_experimental_agc.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+GainControlForExperimentalAgc::GainControlForExperimentalAgc(
+ GainControl* gain_control,
+ rtc::CriticalSection* crit_capture)
+ : real_gain_control_(gain_control),
+ volume_(0),
+ crit_capture_(crit_capture) {}
+
+int GainControlForExperimentalAgc::Enable(bool enable) {
+ return real_gain_control_->Enable(enable);
+}
+
+bool GainControlForExperimentalAgc::is_enabled() const {
+ return real_gain_control_->is_enabled();
+}
+
+int GainControlForExperimentalAgc::set_stream_analog_level(int level) {
+ rtc::CritScope cs_capture(crit_capture_);
+ volume_ = level;
+ return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::stream_analog_level() {
+ rtc::CritScope cs_capture(crit_capture_);
+ return volume_;
+}
+
+int GainControlForExperimentalAgc::set_mode(Mode mode) {
+ return AudioProcessing::kNoError;
+}
+
+GainControl::Mode GainControlForExperimentalAgc::mode() const {
+ return GainControl::kAdaptiveAnalog;
+}
+
+int GainControlForExperimentalAgc::set_target_level_dbfs(int level) {
+ return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::target_level_dbfs() const {
+ return real_gain_control_->target_level_dbfs();
+}
+
+int GainControlForExperimentalAgc::set_compression_gain_db(int gain) {
+ return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::compression_gain_db() const {
+ return real_gain_control_->compression_gain_db();
+}
+
+int GainControlForExperimentalAgc::enable_limiter(bool enable) {
+ return AudioProcessing::kNoError;
+}
+
+bool GainControlForExperimentalAgc::is_limiter_enabled() const {
+ return real_gain_control_->is_limiter_enabled();
+}
+
+int GainControlForExperimentalAgc::set_analog_level_limits(int minimum,
+ int maximum) {
+ return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::analog_level_minimum() const {
+ return real_gain_control_->analog_level_minimum();
+}
+
+int GainControlForExperimentalAgc::analog_level_maximum() const {
+ return real_gain_control_->analog_level_maximum();
+}
+
+bool GainControlForExperimentalAgc::stream_is_saturated() const {
+ return real_gain_control_->stream_is_saturated();
+}
+
+void GainControlForExperimentalAgc::SetMicVolume(int volume) {
+ rtc::CritScope cs_capture(crit_capture_);
+ volume_ = volume;
+}
+
+int GainControlForExperimentalAgc::GetMicVolume() {
+ rtc::CritScope cs_capture(crit_capture_);
+ return volume_;
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
new file mode 100644
index 0000000..4fbd05c
--- /dev/null
+++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2016 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_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// This class has two main purposes:
+//
+// 1) It is returned instead of the real GainControl after the new AGC has been
+// enabled in order to prevent an outside user from overriding compression
+// settings. It doesn't do anything in its implementation, except for
+// delegating the const methods and Enable calls to the real GainControl, so
+// AGC can still be disabled.
+//
+// 2) It is injected into AgcManagerDirect and implements volume callbacks for
+// getting and setting the volume level. It just caches this value to be used
+// in VoiceEngine later.
+class GainControlForExperimentalAgc : public GainControl,
+ public VolumeCallbacks {
+ public:
+ explicit GainControlForExperimentalAgc(GainControl* gain_control,
+ rtc::CriticalSection* crit_capture);
+
+ // GainControl implementation.
+ int Enable(bool enable) override;
+ bool is_enabled() const override;
+ int set_stream_analog_level(int level) override;
+ int stream_analog_level() override;
+ int set_mode(Mode mode) override;
+ Mode mode() const override;
+ int set_target_level_dbfs(int level) override;
+ int target_level_dbfs() const override;
+ int set_compression_gain_db(int gain) override;
+ int compression_gain_db() const override;
+ int enable_limiter(bool enable) override;
+ bool is_limiter_enabled() const override;
+ int set_analog_level_limits(int minimum, int maximum) override;
+ int analog_level_minimum() const override;
+ int analog_level_maximum() const override;
+ bool stream_is_saturated() const override;
+
+ // VolumeCallbacks implementation.
+ void SetMicVolume(int volume) override;
+ int GetMicVolume() override;
+
+ private:
+ GainControl* real_gain_control_;
+ int volume_;
+ rtc::CriticalSection* crit_capture_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(GainControlForExperimentalAgc);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
