Revert "Replace the ExperimentalAgc config with the new config format"
This reverts commit f3aa6326b8e21f627b9fba72040122723251999b.
Reason for revert: Breaks downstream project.
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
TBR=saza@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:5298
Change-Id: I794d2ab4b8caa5330c5ad490ba604646a249a1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164530
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30153}
diff --git a/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
index c5e65ad..b7c569b 100644
--- a/modules/audio_processing/agc/agc_manager_direct_unittest.cc
+++ b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@@ -37,7 +37,7 @@
class MockGainControl : public GainControl {
public:
virtual ~MockGainControl() {}
- MOCK_METHOD0(Initialize, void());
+ MOCK_METHOD1(Enable, int(bool enable));
MOCK_CONST_METHOD0(is_enabled, bool());
MOCK_METHOD1(set_stream_analog_level, int(int level));
MOCK_CONST_METHOD0(stream_analog_level, int());
diff --git a/modules/audio_processing/agc/gain_control.h b/modules/audio_processing/agc/gain_control.h
index f8c706b..f31cbec 100644
--- a/modules/audio_processing/agc/gain_control.h
+++ b/modules/audio_processing/agc/gain_control.h
@@ -20,6 +20,9 @@
// Recommended to be enabled on the client-side.
class GainControl {
public:
+ virtual int Enable(bool enable) = 0;
+ virtual bool is_enabled() const = 0;
+
// When an analog mode is set, this must be called prior to |ProcessStream()|
// to pass the current analog level from the audio HAL. Must be within the
// range provided to |set_analog_level_limits()|.
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 2844311..1c88581 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -334,7 +334,18 @@
std::move(render_pre_processor),
std::move(echo_detector),
std::move(capture_analyzer)),
- constants_(!field_trial::IsEnabled(
+ constants_(config.Get<ExperimentalAgc>().startup_min_volume,
+ config.Get<ExperimentalAgc>().clipped_level_min,
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+ /* enabled= */ false,
+ /* enabled_agc2_level_estimator= */ false,
+ /* digital_adaptive_disabled= */ false,
+#else
+ config.Get<ExperimentalAgc>().enabled,
+ config.Get<ExperimentalAgc>().enabled_agc2_level_estimator,
+ config.Get<ExperimentalAgc>().digital_adaptive_disabled,
+#endif
+ !field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
!field_trial::IsEnabled(
"WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"),
@@ -353,29 +364,18 @@
capture_nonlocked_.echo_controller_enabled =
static_cast<bool>(echo_control_factory_);
+ submodules_.gain_control.reset(new GainControlImpl());
+
// If no echo detector is injected, use the ResidualEchoDetector.
if (!submodules_.echo_detector) {
submodules_.echo_detector =
new rtc::RefCountedObject<ResidualEchoDetector>();
}
-#if !(defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS))
// TODO(webrtc:5298): Remove once the use of ExperimentalNs has been
// deprecated.
+#if !(defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS))
config_.transient_suppression.enabled = config.Get<ExperimentalNs>().enabled;
-
- // TODO(webrtc:5298): Remove once the use of ExperimentalAgc has been
- // deprecated.
- config_.gain_controller1.analog_gain_controller.enabled =
- config.Get<ExperimentalAgc>().enabled;
- config_.gain_controller1.analog_gain_controller.startup_min_volume =
- config.Get<ExperimentalAgc>().startup_min_volume;
- config_.gain_controller1.analog_gain_controller.clipped_level_min =
- config.Get<ExperimentalAgc>().clipped_level_min;
- config_.gain_controller1.analog_gain_controller.enable_agc2_level_estimator =
- config.Get<ExperimentalAgc>().enabled_agc2_level_estimator;
- config_.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- !config.Get<ExperimentalAgc>().digital_adaptive_disabled;
#endif
}
@@ -480,7 +480,34 @@
AllocateRenderQueue();
- InitializeGainController1();
+ submodules_.gain_control->Initialize(num_proc_channels(),
+ proc_sample_rate_hz());
+ if (constants_.use_experimental_agc) {
+ if (!submodules_.agc_manager.get() ||
+ submodules_.agc_manager->num_channels() !=
+ static_cast<int>(num_proc_channels()) ||
+ submodules_.agc_manager->sample_rate_hz() !=
+ capture_nonlocked_.split_rate) {
+ int stream_analog_level = -1;
+ const bool re_creation = !!submodules_.agc_manager;
+ if (re_creation) {
+ stream_analog_level = submodules_.agc_manager->stream_analog_level();
+ }
+ submodules_.agc_manager.reset(new AgcManagerDirect(
+ num_proc_channels(), constants_.agc_startup_min_volume,
+ constants_.agc_clipped_level_min,
+ constants_.use_experimental_agc_agc2_level_estimation,
+ constants_.use_experimental_agc_agc2_digital_adaptive,
+ capture_nonlocked_.split_rate));
+ if (re_creation) {
+ submodules_.agc_manager->set_stream_analog_level(stream_analog_level);
+ }
+ }
+ submodules_.agc_manager->Initialize();
+ submodules_.agc_manager->SetupDigitalGainControl(
+ submodules_.gain_control.get());
+ submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted);
+ }
InitializeTransientSuppressor();
InitializeHighPassFilter(true);
InitializeVoiceDetector();
@@ -623,20 +650,7 @@
config_.gain_controller1.analog_level_minimum !=
config.gain_controller1.analog_level_minimum ||
config_.gain_controller1.analog_level_maximum !=
- config.gain_controller1.analog_level_maximum ||
- config_.gain_controller1.analog_gain_controller.enabled !=
- config.gain_controller1.analog_gain_controller.enabled ||
- config_.gain_controller1.analog_gain_controller.startup_min_volume !=
- config.gain_controller1.analog_gain_controller.startup_min_volume ||
- config_.gain_controller1.analog_gain_controller.clipped_level_min !=
- config.gain_controller1.analog_gain_controller.clipped_level_min ||
- config_.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator !=
- config.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator ||
- config_.gain_controller1.analog_gain_controller.enable_digital_adaptive !=
- config.gain_controller1.analog_gain_controller
- .enable_digital_adaptive;
+ config.gain_controller1.analog_level_maximum;
const bool agc2_config_changed =
config_.gain_controller2.enabled != config.gain_controller2.enabled;
@@ -673,7 +687,7 @@
InitializeHighPassFilter(false);
if (agc1_config_changed) {
- InitializeGainController1();
+ ApplyAgc1Config(config_.gain_controller1);
}
const bool config_ok = GainController2::Validate(config_.gain_controller2);
@@ -708,6 +722,29 @@
}
}
+void AudioProcessingImpl::ApplyAgc1Config(
+ const Config::GainController1& config) {
+ int error = submodules_.gain_control->Enable(config.enabled);
+ RTC_DCHECK_EQ(kNoError, error);
+
+ if (!submodules_.agc_manager) {
+ error = submodules_.gain_control->set_mode(
+ Agc1ConfigModeToInterfaceMode(config.mode));
+ RTC_DCHECK_EQ(kNoError, error);
+ error = submodules_.gain_control->set_target_level_dbfs(
+ config.target_level_dbfs);
+ RTC_DCHECK_EQ(kNoError, error);
+ error = submodules_.gain_control->set_compression_gain_db(
+ config.compression_gain_db);
+ RTC_DCHECK_EQ(kNoError, error);
+ error = submodules_.gain_control->enable_limiter(config.enable_limiter);
+ RTC_DCHECK_EQ(kNoError, error);
+ error = submodules_.gain_control->set_analog_level_limits(
+ config.analog_level_minimum, config.analog_level_maximum);
+ RTC_DCHECK_EQ(kNoError, error);
+ }
+}
+
// TODO(webrtc:5298): Remove.
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {}
@@ -897,11 +934,9 @@
setting.GetFloat(&value);
int int_value = static_cast<int>(value + .5f);
config_.gain_controller1.compression_gain_db = int_value;
- if (submodules_.gain_control) {
- int error =
- submodules_.gain_control->set_compression_gain_db(int_value);
- RTC_DCHECK_EQ(kNoError, error);
- }
+ int error =
+ submodules_.gain_control->set_compression_gain_db(int_value);
+ RTC_DCHECK_EQ(kNoError, error);
}
break;
}
@@ -977,7 +1012,7 @@
}
}
- if (!submodules_.agc_manager && submodules_.gain_control) {
+ if (!submodules_.agc_manager) {
GainControlImpl::PackRenderAudioBuffer(*audio, &agc_render_queue_buffer_);
// Insert the samples into the queue.
if (!agc_render_signal_queue_->Insert(&agc_render_queue_buffer_)) {
@@ -1064,10 +1099,8 @@
}
}
- if (submodules_.gain_control) {
- while (agc_render_signal_queue_->Remove(&agc_capture_queue_buffer_)) {
- submodules_.gain_control->ProcessRenderAudio(agc_capture_queue_buffer_);
- }
+ while (agc_render_signal_queue_->Remove(&agc_capture_queue_buffer_)) {
+ submodules_.gain_control->ProcessRenderAudio(agc_capture_queue_buffer_);
}
while (red_render_signal_queue_->Remove(&red_capture_queue_buffer_)) {
@@ -1188,7 +1221,8 @@
submodules_.echo_controller->AnalyzeCapture(capture_buffer);
}
- if (submodules_.agc_manager) {
+ if (constants_.use_experimental_agc &&
+ submodules_.gain_control->is_enabled()) {
submodules_.agc_manager->AnalyzePreProcess(capture_buffer);
}
@@ -1215,10 +1249,7 @@
/*use_split_band_data=*/true);
}
- if (submodules_.gain_control) {
- RETURN_ON_ERR(
- submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
- }
+ RETURN_ON_ERR(submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
RTC_DCHECK(
!(submodules_.legacy_noise_suppressor && submodules_.noise_suppressor));
@@ -1283,21 +1314,19 @@
capture_.stats.voice_detected = absl::nullopt;
}
- if (submodules_.agc_manager) {
+ if (constants_.use_experimental_agc &&
+ submodules_.gain_control->is_enabled()) {
submodules_.agc_manager->Process(capture_buffer);
absl::optional<int> new_digital_gain =
submodules_.agc_manager->GetDigitalComressionGain();
- if (new_digital_gain && submodules_.gain_control) {
+ if (new_digital_gain) {
submodules_.gain_control->set_compression_gain_db(*new_digital_gain);
}
}
-
- if (submodules_.gain_control) {
- // TODO(peah): Add reporting from AEC3 whether there is echo.
- RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio(
- capture_buffer, /*stream_has_echo*/ false));
- }
+ // TODO(peah): Add reporting from AEC3 whether there is echo.
+ RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio(
+ capture_buffer, /*stream_has_echo*/ false));
if (submodule_states_.CaptureMultiBandProcessingPresent() &&
SampleRateSupportsMultiBand(
@@ -1626,11 +1655,9 @@
submodules_.agc_manager->set_stream_analog_level(level);
data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level",
1, &level);
- } else if (submodules_.gain_control) {
+ } else {
int error = submodules_.gain_control->set_stream_analog_level(level);
RTC_DCHECK_EQ(kNoError, error);
- } else {
- capture_.cached_stream_analog_level_ = level;
}
}
@@ -1638,11 +1665,8 @@
rtc::CritScope cs_capture(&crit_capture_);
if (submodules_.agc_manager) {
return submodules_.agc_manager->stream_analog_level();
- } else if (submodules_.gain_control) {
- return submodules_.gain_control->stream_analog_level();
- } else {
- return capture_.cached_stream_analog_level_;
}
+ return submodules_.gain_control->stream_analog_level();
}
void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
@@ -1699,7 +1723,7 @@
config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile,
config_.residual_echo_detector.enabled,
!!submodules_.legacy_noise_suppressor || !!submodules_.noise_suppressor,
- !!submodules_.gain_control, !!submodules_.gain_controller2,
+ submodules_.gain_control->is_enabled(), !!submodules_.gain_controller2,
config_.pre_amplifier.enabled, capture_nonlocked_.echo_controller_enabled,
config_.voice_detection.enabled, !!submodules_.transient_suppressor);
}
@@ -1830,71 +1854,6 @@
aecm_render_signal_queue_.reset();
}
-void AudioProcessingImpl::InitializeGainController1() {
- if (!config_.gain_controller1.enabled) {
- submodules_.agc_manager.reset();
- submodules_.gain_control.reset();
- return;
- }
-
- if (!submodules_.gain_control) {
- submodules_.gain_control.reset(new GainControlImpl());
- }
-
- submodules_.gain_control->Initialize(num_proc_channels(),
- proc_sample_rate_hz());
-
- if (!config_.gain_controller1.analog_gain_controller.enabled) {
- int error = submodules_.gain_control->set_mode(
- Agc1ConfigModeToInterfaceMode(config_.gain_controller1.mode));
- RTC_DCHECK_EQ(kNoError, error);
- error = submodules_.gain_control->set_target_level_dbfs(
- config_.gain_controller1.target_level_dbfs);
- RTC_DCHECK_EQ(kNoError, error);
- error = submodules_.gain_control->set_compression_gain_db(
- config_.gain_controller1.compression_gain_db);
- RTC_DCHECK_EQ(kNoError, error);
- error = submodules_.gain_control->enable_limiter(
- config_.gain_controller1.enable_limiter);
- RTC_DCHECK_EQ(kNoError, error);
- error = submodules_.gain_control->set_analog_level_limits(
- config_.gain_controller1.analog_level_minimum,
- config_.gain_controller1.analog_level_maximum);
- RTC_DCHECK_EQ(kNoError, error);
-
- submodules_.agc_manager.reset();
- return;
- }
-
- if (!submodules_.agc_manager.get() ||
- submodules_.agc_manager->num_channels() !=
- static_cast<int>(num_proc_channels()) ||
- submodules_.agc_manager->sample_rate_hz() !=
- capture_nonlocked_.split_rate) {
- int stream_analog_level = -1;
- const bool re_creation = !!submodules_.agc_manager;
- if (re_creation) {
- stream_analog_level = submodules_.agc_manager->stream_analog_level();
- }
- submodules_.agc_manager.reset(new AgcManagerDirect(
- num_proc_channels(),
- config_.gain_controller1.analog_gain_controller.startup_min_volume,
- config_.gain_controller1.analog_gain_controller.clipped_level_min,
- config_.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator,
- !config_.gain_controller1.analog_gain_controller
- .enable_digital_adaptive,
- capture_nonlocked_.split_rate));
- if (re_creation) {
- submodules_.agc_manager->set_stream_analog_level(stream_analog_level);
- }
- }
- submodules_.agc_manager->Initialize();
- submodules_.agc_manager->SetupDigitalGainControl(
- submodules_.gain_control.get());
- submodules_.agc_manager->SetCaptureMuted(capture_.output_will_be_muted);
-}
-
void AudioProcessingImpl::InitializeGainController2() {
if (config_.gain_controller2.enabled) {
if (!submodules_.gain_controller2) {
@@ -1998,8 +1957,7 @@
std::string experiments_description = "";
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
- if (config_.gain_controller1.analog_gain_controller.clipped_level_min !=
- kClippedLevelMin) {
+ if (constants_.agc_clipped_level_min != kClippedLevelMin) {
experiments_description += "AgcClippingLevelExperiment;";
}
if (capture_nonlocked_.echo_controller_enabled) {
@@ -2025,14 +1983,10 @@
? static_cast<int>(submodules_.echo_control_mobile->routing_mode())
: 0;
- apm_config.agc_enabled = !!submodules_.gain_control;
-
- apm_config.agc_mode = submodules_.gain_control
- ? static_cast<int>(submodules_.gain_control->mode())
- : GainControl::kAdaptiveAnalog;
+ apm_config.agc_enabled = submodules_.gain_control->is_enabled();
+ apm_config.agc_mode = static_cast<int>(submodules_.gain_control->mode());
apm_config.agc_limiter_enabled =
- submodules_.gain_control ? submodules_.gain_control->is_limiter_enabled()
- : false;
+ submodules_.gain_control->is_limiter_enabled();
apm_config.noise_robust_agc_enabled = !!submodules_.agc_manager;
apm_config.hpf_enabled = config_.high_pass_filter.enabled;
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index af5a0f6..ee3fb4d 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -243,7 +243,6 @@
void InitializeHighPassFilter(bool forced_reset)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeVoiceDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
- void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeTransientSuppressor()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
@@ -264,6 +263,8 @@
void HandleCaptureRuntimeSettings()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void HandleRenderRuntimeSettings() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+ void ApplyAgc1Config(const Config::GainController1& agc_config)
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void EmptyQueuedRenderAudio();
void AllocateRenderQueue()
@@ -380,12 +381,29 @@
// APM constants.
const struct ApmConstants {
- ApmConstants(bool multi_channel_render_support,
+ ApmConstants(int agc_startup_min_volume,
+ int agc_clipped_level_min,
+ bool use_experimental_agc,
+ bool use_experimental_agc_agc2_level_estimation,
+ bool use_experimental_agc_agc2_digital_adaptive,
+ bool multi_channel_render_support,
bool multi_channel_capture_support,
bool enforce_split_band_hpf)
- : multi_channel_render_support(multi_channel_render_support),
+ : agc_startup_min_volume(agc_startup_min_volume),
+ agc_clipped_level_min(agc_clipped_level_min),
+ use_experimental_agc(use_experimental_agc),
+ use_experimental_agc_agc2_level_estimation(
+ use_experimental_agc_agc2_level_estimation),
+ use_experimental_agc_agc2_digital_adaptive(
+ use_experimental_agc_agc2_digital_adaptive),
+ multi_channel_render_support(multi_channel_render_support),
multi_channel_capture_support(multi_channel_capture_support),
enforce_split_band_hpf(enforce_split_band_hpf) {}
+ int agc_startup_min_volume;
+ int agc_clipped_level_min;
+ bool use_experimental_agc;
+ bool use_experimental_agc_agc2_level_estimation;
+ bool use_experimental_agc_agc2_digital_adaptive;
bool multi_channel_render_support;
bool multi_channel_capture_support;
bool enforce_split_band_hpf;
@@ -417,7 +435,6 @@
size_t num_keyboard_frames = 0;
const float* keyboard_data = nullptr;
} keyboard_info;
- int cached_stream_analog_level_ = 0;
} capture_ RTC_GUARDED_BY(crit_capture_);
struct ApmCaptureNonLockedState {
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index ca05f71..8f9e535 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -430,9 +430,10 @@
far_file_(NULL),
near_file_(NULL),
out_file_(NULL) {
- apm_.reset(AudioProcessingBuilder().Create());
+ Config config;
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+ apm_.reset(AudioProcessingBuilder().Create(config));
AudioProcessing::Config apm_config = apm_->GetConfig();
- apm_config.gain_controller1.analog_gain_controller.enabled = false;
apm_config.pipeline.maximum_internal_processing_rate = 48000;
apm_->ApplyConfig(apm_config);
}
@@ -966,49 +967,42 @@
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(ApmTest, GainControlDiesOnTooLowTargetLevelDbfs) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.target_level_dbfs = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighTargetLevelDbfs) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.target_level_dbfs = 32;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowCompressionGainDb) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.compression_gain_db = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighCompressionGainDb) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.compression_gain_db = 91;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooLowAnalogLevelLowerLimit) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = -1;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnTooHighAnalogLevelUpperLimit) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_maximum = 65536;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
}
TEST_F(ApmTest, GainControlDiesOnInvertedAnalogLevelLimits) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 512;
config.gain_controller1.analog_level_maximum = 255;
EXPECT_DEATH(apm_->ApplyConfig(config), "");
@@ -1016,7 +1010,6 @@
TEST_F(ApmTest, ApmDiesOnTooLowAnalogLevel) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 255;
config.gain_controller1.analog_level_maximum = 512;
apm_->ApplyConfig(config);
@@ -1025,7 +1018,6 @@
TEST_F(ApmTest, ApmDiesOnTooHighAnalogLevel) {
auto config = apm_->GetConfig();
- config.gain_controller1.enabled = true;
config.gain_controller1.analog_level_minimum = 255;
config.gain_controller1.analog_level_maximum = 512;
apm_->ApplyConfig(config);
@@ -1541,10 +1533,9 @@
if (test->num_input_channels() != test->num_output_channels())
continue;
- apm_.reset(AudioProcessingBuilder().Create());
- AudioProcessing::Config apm_config = apm_->GetConfig();
- apm_config.gain_controller1.analog_gain_controller.enabled = false;
- apm_->ApplyConfig(apm_config);
+ Config config;
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+ apm_.reset(AudioProcessingBuilder().Create(config));
EnableAllComponents();
@@ -1827,11 +1818,10 @@
size_t num_reverse_input_channels,
size_t num_reverse_output_channels,
const std::string& output_file_prefix) {
- std::unique_ptr<AudioProcessing> ap(AudioProcessingBuilder().Create());
- AudioProcessing::Config apm_config = ap->GetConfig();
- apm_config.gain_controller1.analog_gain_controller.enabled = false;
- ap->ApplyConfig(apm_config);
-
+ Config config;
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+ std::unique_ptr<AudioProcessing> ap(
+ AudioProcessingBuilder().Create(config));
EnableAllAPComponents(ap.get());
ProcessingConfig processing_config = {
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index b5454c0..841d901 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -112,6 +112,10 @@
void GainControlImpl::ProcessRenderAudio(
rtc::ArrayView<const int16_t> packed_render_audio) {
+ if (!enabled_) {
+ return;
+ }
+
for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
WebRtcAgc_AddFarend(mono_agcs_[ch]->state, packed_render_audio.data(),
packed_render_audio.size());
@@ -147,6 +151,10 @@
}
int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) {
+ if (!enabled_) {
+ return AudioProcessing::kNoError;
+ }
+
RTC_DCHECK(num_proc_channels_);
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
RTC_DCHECK_EQ(audio.num_channels(), *num_proc_channels_);
@@ -195,6 +203,10 @@
int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
bool stream_has_echo) {
+ if (!enabled_) {
+ return AudioProcessing::kNoError;
+ }
+
if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
return AudioProcessing::kStreamParameterNotSetError;
}
@@ -297,6 +309,19 @@
return analog_capture_level_;
}
+int GainControlImpl::Enable(bool enable) {
+ if (enable && !enabled_) {
+ enabled_ = enable; // Must be set before Initialize() is called.
+
+ RTC_DCHECK(num_proc_channels_);
+ RTC_DCHECK(sample_rate_hz_);
+ Initialize(*num_proc_channels_, *sample_rate_hz_);
+ } else {
+ enabled_ = enable;
+ }
+ return AudioProcessing::kNoError;
+}
+
int GainControlImpl::set_mode(Mode mode) {
if (MapSetting(mode) == -1) {
return AudioProcessing::kBadParameterError;
@@ -356,6 +381,10 @@
num_proc_channels_ = num_proc_channels;
sample_rate_hz_ = sample_rate_hz;
+ if (!enabled_) {
+ return;
+ }
+
mono_agcs_.resize(*num_proc_channels_);
capture_levels_.resize(*num_proc_channels_);
for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
diff --git a/modules/audio_processing/gain_control_impl.h b/modules/audio_processing/gain_control_impl.h
index b65d697..5ddf5ec 100644
--- a/modules/audio_processing/gain_control_impl.h
+++ b/modules/audio_processing/gain_control_impl.h
@@ -44,9 +44,11 @@
std::vector<int16_t>* packed_buffer);
// GainControl implementation.
+ bool is_enabled() const override { return enabled_; }
int stream_analog_level() const override;
bool is_limiter_enabled() const override { return limiter_enabled_; }
Mode mode() const override { return mode_; }
+ int Enable(bool enable) override;
int set_mode(Mode mode) override;
int compression_gain_db() const override { return compression_gain_db_; }
int set_analog_level_limits(int minimum, int maximum) override;
@@ -68,6 +70,8 @@
std::unique_ptr<ApmDataDumper> data_dumper_;
+ bool enabled_ = false;
+
const bool use_legacy_gain_applier_;
Mode mode_;
int minimum_capture_level_;
@@ -75,7 +79,7 @@
bool limiter_enabled_;
int target_level_dbfs_;
int compression_gain_db_;
- int analog_capture_level_ = 0;
+ int analog_capture_level_;
bool was_analog_level_set_;
bool stream_is_saturated_;
diff --git a/modules/audio_processing/gain_control_unittest.cc b/modules/audio_processing/gain_control_unittest.cc
index 6e01499..c1078b4 100644
--- a/modules/audio_processing/gain_control_unittest.cc
+++ b/modules/audio_processing/gain_control_unittest.cc
@@ -52,6 +52,7 @@
GainControlImpl* gain_controller) {
gain_controller->Initialize(1, sample_rate_hz);
GainControl* gc = static_cast<GainControl*>(gain_controller);
+ gc->Enable(true);
gc->set_mode(mode);
gc->set_stream_analog_level(stream_analog_level);
gc->set_target_level_dbfs(target_level_dbfs);
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index d76d1a8..fe4b0dc 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -60,10 +60,6 @@
static const int kAgcStartupMinVolume = 0;
#endif // defined(WEBRTC_CHROMIUM_BUILD)
static constexpr int kClippedLevelMin = 70;
-
-// To be deprecated: Please instead use the flag in the
-// AudioProcessing::Config::AnalogGainController.
-// TODO(webrtc:5298): Remove.
struct ExperimentalAgc {
ExperimentalAgc() = default;
explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
@@ -318,17 +314,6 @@
// Must be set if an analog mode is used. Limited to [0, 65535].
int analog_level_minimum = 0;
int analog_level_maximum = 255;
-
- // Enables the analog gain controller functionality.
- struct AnalogGainController {
- bool enabled = false;
- int startup_min_volume = kAgcStartupMinVolume;
- // Lowest analog microphone level that will be applied in response to
- // clipping.
- int clipped_level_min = kClippedLevelMin;
- bool enable_agc2_level_estimator = false;
- bool enable_digital_adaptive = true;
- } analog_gain_controller;
} gain_controller1;
// Enables the next generation AGC functionality. This feature replaces the
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 142e707..95a3e37 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -364,10 +364,11 @@
}
}
+ // TODO(peah): Add support for controlling the Experimental AGC from the
+ // command line.
if (msg.has_noise_robust_agc_enabled()) {
- apm_config.gain_controller1.analog_gain_controller.enabled =
- settings_.use_analog_agc ? *settings_.use_analog_agc
- : msg.noise_robust_agc_enabled();
+ config.Set<ExperimentalAgc>(
+ new ExperimentalAgc(msg.noise_robust_agc_enabled()));
if (settings_.use_verbose_logging) {
std::cout << " noise_robust_agc_enabled: "
<< (msg.noise_robust_agc_enabled() ? "true" : "false")
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 84cd9a0..f314732 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -494,20 +494,15 @@
apm_config.gain_controller1.compression_gain_db =
*settings_.agc_compression_gain;
}
- if (settings_.use_analog_agc) {
- apm_config.gain_controller1.analog_gain_controller.enabled =
- *settings_.use_analog_agc;
- }
- if (settings_.use_analog_agc_agc2_level_estimator) {
- apm_config.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator =
- *settings_.use_analog_agc_agc2_level_estimator;
- }
- if (settings_.analog_agc_disable_digital_adaptive) {
- apm_config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- *settings_.analog_agc_disable_digital_adaptive;
- }
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(
+ !settings_.use_experimental_agc || *settings_.use_experimental_agc,
+ !!settings_.use_experimental_agc_agc2_level_estimator &&
+ *settings_.use_experimental_agc_agc2_level_estimator,
+ !!settings_.experimental_agc_disable_digital_adaptive &&
+ *settings_.experimental_agc_disable_digital_adaptive,
+ !!settings_.experimental_agc_analyze_before_aec &&
+ *settings_.experimental_agc_analyze_before_aec));
if (settings_.use_ed) {
apm_config.residual_echo_detector.enabled = *settings_.use_ed;
}
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index c28dd6d..c902d7c 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -57,13 +57,14 @@
absl::optional<bool> use_hpf;
absl::optional<bool> use_ns;
absl::optional<bool> use_ts;
- absl::optional<bool> use_analog_agc;
absl::optional<bool> use_vad;
absl::optional<bool> use_le;
absl::optional<bool> use_all;
absl::optional<bool> use_legacy_ns;
- absl::optional<bool> use_analog_agc_agc2_level_estimator;
- absl::optional<bool> analog_agc_disable_digital_adaptive;
+ absl::optional<bool> use_experimental_agc;
+ absl::optional<bool> use_experimental_agc_agc2_level_estimator;
+ absl::optional<bool> experimental_agc_disable_digital_adaptive;
+ absl::optional<bool> experimental_agc_analyze_before_aec;
absl::optional<int> agc_mode;
absl::optional<int> agc_target_level;
absl::optional<bool> use_agc_limiter;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index ec637c1..c4d2ec2 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -102,10 +102,6 @@
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the transient suppressor");
ABSL_FLAG(int,
- analog_agc,
- kParameterNotSpecifiedValue,
- "Activate (1) or deactivate(0) the transient suppressor");
-ABSL_FLAG(int,
vad,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the voice activity detector");
@@ -123,12 +119,21 @@
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the legacy NS");
ABSL_FLAG(int,
- analog_agc_disable_digital_adaptive,
+ experimental_agc,
+ kParameterNotSpecifiedValue,
+ "Activate (1) or deactivate(0) the experimental AGC");
+ABSL_FLAG(int,
+ experimental_agc_disable_digital_adaptive,
kParameterNotSpecifiedValue,
"Force-deactivate (1) digital adaptation in "
"experimental AGC. Digital adaptation is active by default (0).");
ABSL_FLAG(int,
- analog_agc_agc2_level_estimator,
+ experimental_agc_analyze_before_aec,
+ kParameterNotSpecifiedValue,
+ "Make level estimation happen before AEC"
+ " in the experimental AGC. After AEC is the default (0)");
+ABSL_FLAG(int,
+ experimental_agc_agc2_level_estimator,
kParameterNotSpecifiedValue,
"AGC2 level estimation"
" in the experimental AGC. AGC1 level estimation is the default (0)");
@@ -329,7 +334,6 @@
settings.use_le = true;
settings.use_vad = true;
settings.use_ts = true;
- settings.use_analog_agc = true;
settings.use_ns = true;
settings.use_hpf = true;
settings.use_agc = true;
@@ -373,16 +377,20 @@
SetSettingIfFlagSet(absl::GetFlag(FLAGS_hpf), &settings.use_hpf);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_ns), &settings.use_ns);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_ts), &settings.use_ts);
- SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc),
- &settings.use_analog_agc);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_vad), &settings.use_vad);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_le), &settings.use_le);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_use_legacy_ns),
&settings.use_legacy_ns);
- SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_disable_digital_adaptive),
- &settings.analog_agc_disable_digital_adaptive);
- SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_agc2_level_estimator),
- &settings.use_analog_agc_agc2_level_estimator);
+ SetSettingIfFlagSet(absl::GetFlag(FLAGS_experimental_agc),
+ &settings.use_experimental_agc);
+ SetSettingIfFlagSet(
+ absl::GetFlag(FLAGS_experimental_agc_disable_digital_adaptive),
+ &settings.experimental_agc_disable_digital_adaptive);
+ SetSettingIfFlagSet(absl::GetFlag(FLAGS_experimental_agc_analyze_before_aec),
+ &settings.experimental_agc_analyze_before_aec);
+ SetSettingIfFlagSet(
+ absl::GetFlag(FLAGS_experimental_agc_agc2_level_estimator),
+ &settings.use_experimental_agc_agc2_level_estimator);
SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_mode), &settings.agc_mode);
SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_target_level),
&settings.agc_target_level);
diff --git a/modules/audio_processing/test/debug_dump_replayer.cc b/modules/audio_processing/test/debug_dump_replayer.cc
index 26ca429..d5cf673 100644
--- a/modules/audio_processing/test/debug_dump_replayer.cc
+++ b/modules/audio_processing/test/debug_dump_replayer.cc
@@ -180,6 +180,11 @@
// These configurations cannot be changed on the fly.
Config config;
RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
+
+ RTC_CHECK(msg.has_noise_robust_agc_enabled());
+ config.Set<ExperimentalAgc>(
+ new ExperimentalAgc(msg.noise_robust_agc_enabled()));
+
RTC_CHECK(msg.has_aec_extended_filter_enabled());
// We only create APM once, since changes on these fields should not
@@ -230,9 +235,6 @@
static_cast<AudioProcessing::Config::GainController1::Mode>(
msg.agc_mode());
apm_config.gain_controller1.enable_limiter = msg.agc_limiter_enabled();
- RTC_CHECK(msg.has_noise_robust_agc_enabled());
- apm_config.gain_controller1.analog_gain_controller.enabled =
- msg.noise_robust_agc_enabled();
apm_->ApplyConfig(apm_config);
}
diff --git a/modules/audio_processing/test/debug_dump_test.cc b/modules/audio_processing/test/debug_dump_test.cc
index 71478a9..21458aa 100644
--- a/modules/audio_processing/test/debug_dump_test.cc
+++ b/modules/audio_processing/test/debug_dump_test.cc
@@ -210,7 +210,6 @@
ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
input_->channels());
RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
- apm_->set_stream_analog_level(100);
if (enable_pre_amplifier_) {
apm_->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(1 + i % 10));
@@ -359,10 +358,8 @@
Config config;
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
- apm_config.gain_controller1.analog_gain_controller.enabled = true;
- apm_config.gain_controller1.analog_gain_controller.startup_min_volume = 0;
// Arbitrarily set clipping gain to 17, which will never be the default.
- apm_config.gain_controller1.analog_gain_controller.clipped_level_min = 17;
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(true, 0, 17));
DebugDumpGenerator generator(config, apm_config);
generator.StartRecording();
generator.Process(100);
@@ -439,12 +436,9 @@
TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) {
Config config;
- AudioProcessing::Config apm_config;
- apm_config.gain_controller1.analog_gain_controller.enabled = true;
- apm_config.gain_controller1.analog_gain_controller.startup_min_volume = 0;
// Arbitrarily set clipping gain to 17, which will never be the default.
- apm_config.gain_controller1.analog_gain_controller.clipped_level_min = 17;
- DebugDumpGenerator generator(config, apm_config);
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(true, 0, 17));
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
diff --git a/test/fuzzers/agc_fuzzer.cc b/test/fuzzers/agc_fuzzer.cc
index 890649a..ac3f83b 100644
--- a/test/fuzzers/agc_fuzzer.cc
+++ b/test/fuzzers/agc_fuzzer.cc
@@ -67,7 +67,9 @@
}
gc->set_compression_gain_db(gain);
gc->set_target_level_dbfs(target_level_dbfs);
+ gc->Enable(true);
+ static_cast<void>(gc->is_enabled());
static_cast<void>(gc->mode());
static_cast<void>(gc->analog_level_minimum());
static_cast<void>(gc->analog_level_maximum());
