The current scheme for setting parameters and specifying the behavior
of the audio processing module is quite complex and hard to implement
in a threadsafe and efficient manner. Therefore a new scheme for setting
the parameters in the audio processing module is introduced in this CL.
The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2292863002
Cr-Commit-Position: refs/heads/master@{#14171}
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index 33ea7ca..f1e63a5 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -70,6 +70,7 @@
WEBRTC_STUB(Initialize, (
const webrtc::ProcessingConfig& processing_config));
+ WEBRTC_VOID_STUB(ApplyConfig, (const AudioProcessing::Config& config));
WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 2992411..4343612 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -868,8 +868,9 @@
LOG(LS_INFO) << "Level control: "
<< (!!level_control_ ? *level_control_ : -1);
+ webrtc::AudioProcessing::Config apm_config;
if (level_control_) {
- config.Set<webrtc::LevelControl>(new webrtc::LevelControl(*level_control_));
+ apm_config.level_controller.enabled = *level_control_;
}
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
@@ -877,6 +878,7 @@
webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
if (audioproc) {
audioproc->SetExtraOptions(config);
+ audioproc->ApplyConfig(apm_config);
}
if (options.recording_sample_rate) {
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 222f749..dabc4b8 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -154,15 +154,15 @@
};
AudioProcessing* AudioProcessing::Create() {
- Config config;
+ webrtc::Config config;
return Create(config, nullptr);
}
-AudioProcessing* AudioProcessing::Create(const Config& config) {
+AudioProcessing* AudioProcessing::Create(const webrtc::Config& config) {
return Create(config, nullptr);
}
-AudioProcessing* AudioProcessing::Create(const Config& config,
+AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
@@ -173,10 +173,10 @@
return apm;
}
-AudioProcessingImpl::AudioProcessingImpl(const Config& config)
+AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
: AudioProcessingImpl(config, nullptr) {}
-AudioProcessingImpl::AudioProcessingImpl(const Config& config,
+AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
NonlinearBeamformer* beamformer)
: public_submodules_(new ApmPublicSubmodules()),
private_submodules_(new ApmPrivateSubmodules(beamformer)),
@@ -194,8 +194,7 @@
config.Get<Beamforming>().array_geometry,
config.Get<Beamforming>().target_direction),
capture_nonlocked_(config.Get<Beamforming>().enabled,
- config.Get<Intelligibility>().enabled,
- config.Get<LevelControl>().enabled) {
+ config.Get<Intelligibility>().enabled) {
{
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -432,7 +431,34 @@
return InitializeLocked();
}
-void AudioProcessingImpl::SetExtraOptions(const Config& config) {
+void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
+ AudioProcessing::Config config_to_use = config;
+
+ bool config_ok = LevelController::Validate(config_to_use.level_controller);
+ if (!config_ok) {
+ LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
+ << "level_controller: "
+ << LevelController::ToString(config_to_use.level_controller)
+ << std::endl
+ << "Reverting to default parameter set";
+ config_to_use.level_controller = AudioProcessing::Config::LevelController();
+ }
+
+ // Run in a single-threaded manner when applying the settings.
+ rtc::CritScope cs_render(&crit_render_);
+ rtc::CritScope cs_capture(&crit_capture_);
+
+ if (config.level_controller.enabled !=
+ capture_nonlocked_.level_controller_enabled) {
+ InitializeLevelController();
+ LOG(LS_INFO) << "Level controller activated: "
+ << capture_nonlocked_.level_controller_enabled;
+ capture_nonlocked_.level_controller_enabled =
+ config.level_controller.enabled;
+ }
+}
+
+void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
// Run in a single-threaded manner when setting the extra options.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
@@ -446,16 +472,6 @@
InitializeTransient();
}
- if (capture_nonlocked_.level_controller_enabled !=
- config.Get<LevelControl>().enabled) {
- capture_nonlocked_.level_controller_enabled =
- config.Get<LevelControl>().enabled;
- LOG(LS_INFO) << "Level controller activated: "
- << config.Get<LevelControl>().enabled;
-
- InitializeLevelController();
- }
-
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if(capture_nonlocked_.intelligibility_enabled !=
config.Get<Intelligibility>().enabled) {
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 87e2224..00484c2 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -42,9 +42,10 @@
public:
// Methods forcing APM to run in a single-threaded manner.
// Acquires both the render and capture locks.
- explicit AudioProcessingImpl(const Config& config);
+ explicit AudioProcessingImpl(const webrtc::Config& config);
// AudioProcessingImpl takes ownership of beamformer.
- AudioProcessingImpl(const Config& config, NonlinearBeamformer* beamformer);
+ AudioProcessingImpl(const webrtc::Config& config,
+ NonlinearBeamformer* beamformer);
~AudioProcessingImpl() override;
int Initialize() override;
int Initialize(int input_sample_rate_hz,
@@ -54,7 +55,8 @@
ChannelLayout output_layout,
ChannelLayout reverse_layout) override;
int Initialize(const ProcessingConfig& processing_config) override;
- void SetExtraOptions(const Config& config) override;
+ void ApplyConfig(const AudioProcessing::Config& config) override;
+ void SetExtraOptions(const webrtc::Config& config) override;
void UpdateHistogramsOnCallEnd() override;
int StartDebugRecording(const char filename[kMaxFilenameSize],
int64_t max_log_size_bytes) override;
@@ -312,14 +314,12 @@
struct ApmCaptureNonLockedState {
ApmCaptureNonLockedState(bool beamformer_enabled,
- bool intelligibility_enabled,
- bool level_controller_enabled)
+ bool intelligibility_enabled)
: fwd_proc_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz),
stream_delay_ms(0),
beamformer_enabled(beamformer_enabled),
- intelligibility_enabled(intelligibility_enabled),
- level_controller_enabled(level_controller_enabled) {}
+ intelligibility_enabled(intelligibility_enabled) {}
// Only the rate and samples fields of fwd_proc_format_ are used because the
// forward processing number of channels is mutable and is tracked by the
// capture_audio_.
@@ -328,7 +328,7 @@
int stream_delay_ms;
bool beamformer_enabled;
bool intelligibility_enabled;
- bool level_controller_enabled;
+ bool level_controller_enabled = false;
} capture_nonlocked_;
struct ApmRenderState {
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index 54e0331..d91457c 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -23,8 +23,8 @@
class MockInitialize : public AudioProcessingImpl {
public:
- explicit MockInitialize(const Config& config) : AudioProcessingImpl(config) {
- }
+ explicit MockInitialize(const webrtc::Config& config)
+ : AudioProcessingImpl(config) {}
MOCK_METHOD0(InitializeLocked, int());
int RealInitializeLocked() NO_THREAD_SAFETY_ANALYSIS {
@@ -33,7 +33,7 @@
};
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
- Config config;
+ webrtc::Config config;
MockInitialize mock(config);
ON_CALL(mock, InitializeLocked())
.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index 5a7aef6..9cfa043 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -538,7 +538,7 @@
}
}
-void EchoCancellationImpl::SetExtraOptions(const Config& config) {
+void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
{
rtc::CritScope cs(crit_capture_);
extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.h b/webrtc/modules/audio_processing/echo_cancellation_impl.h
index 0f0945f..8216c50 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -43,7 +43,7 @@
size_t num_reverse_channels_,
size_t num_output_channels_,
size_t num_proc_channels_);
- void SetExtraOptions(const Config& config);
+ void SetExtraOptions(const webrtc::Config& config);
bool is_delay_agnostic_enabled() const;
bool is_extended_filter_enabled() const;
bool is_aec3_enabled() const;
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 44ff732..035fa32 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -91,14 +91,6 @@
bool enabled;
};
-// Enables the adaptive level controller.
-struct LevelControl {
- LevelControl() : enabled(false) {}
- explicit LevelControl(bool enabled) : enabled(enabled) {}
- static const ConfigOptionID identifier = ConfigOptionID::kLevelControl;
- bool enabled;
-};
-
// Enables delay-agnostic echo cancellation. This feature relies on internally
// estimated delays between the process and reverse streams, thus not relying
// on reported system delays. This configuration only applies to
@@ -205,6 +197,10 @@
// Usage example, omitting error checking:
// AudioProcessing* apm = AudioProcessing::Create(0);
//
+// AudioProcessing::Config config;
+// config.level_controller.enabled = true;
+// apm->ApplyConfig(config)
+//
// apm->high_pass_filter()->Enable(true);
//
// apm->echo_cancellation()->enable_drift_compensation(false);
@@ -244,14 +240,29 @@
//
class AudioProcessing {
public:
+ // The struct below constitutes the new parameter scheme for the audio
+ // processing. It is being introduced gradually and until it is fully
+ // introduced, it is prone to change.
+ // TODO(peah): Remove this comment once the new config scheme is fully rolled
+ // out.
+ //
+ // The parameters and behavior of the audio processing module are controlled
+ // by changing the default values in the AudioProcessing::Config struct.
+ // The config is applied by passing the struct to the ApplyConfig method.
+ struct Config {
+ struct LevelController {
+ bool enabled = false;
+ } level_controller;
+ };
+
// TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
enum ChannelLayout {
kMono,
// Left, right.
kStereo,
- // Mono, keyboard mic.
+ // Mono, keyboard, and mic.
kMonoAndKeyboard,
- // Left, right, keyboard mic.
+ // Left, right, keyboard, and mic.
kStereoAndKeyboard
};
@@ -262,9 +273,9 @@
// be one instance for every incoming stream.
static AudioProcessing* Create();
// Allows passing in an optional configuration at create-time.
- static AudioProcessing* Create(const Config& config);
+ static AudioProcessing* Create(const webrtc::Config& config);
// Only for testing.
- static AudioProcessing* Create(const Config& config,
+ static AudioProcessing* Create(const webrtc::Config& config,
NonlinearBeamformer* beamformer);
virtual ~AudioProcessing() {}
@@ -300,9 +311,13 @@
ChannelLayout output_layout,
ChannelLayout reverse_layout) = 0;
+ // TODO(peah): This method is a temporary solution used to take control
+ // over the parameters in the audio processing module and is likely to change.
+ virtual void ApplyConfig(const Config& config) = 0;
+
// Pass down additional options which don't have explicit setters. This
// ensures the options are applied immediately.
- virtual void SetExtraOptions(const Config& config) = 0;
+ virtual void SetExtraOptions(const webrtc::Config& config) = 0;
// TODO(ajm): Only intended for internal use. Make private and friend the
// necessary classes?
diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
index b5ea587..0566e85 100644
--- a/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -190,8 +190,8 @@
ChannelLayout reverse_layout));
MOCK_METHOD1(Initialize,
int(const ProcessingConfig& processing_config));
- MOCK_METHOD1(SetExtraOptions,
- void(const Config& config));
+ MOCK_METHOD1(ApplyConfig, void(const Config& config));
+ MOCK_METHOD1(SetExtraOptions, void(const webrtc::Config& config));
MOCK_METHOD1(set_sample_rate_hz,
int(int rate));
MOCK_CONST_METHOD0(input_sample_rate_hz,
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.cc b/webrtc/modules/audio_processing/level_controller/level_controller.cc
index 2294396..c0edeb9 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.cc
@@ -262,4 +262,17 @@
audio->channels_f()[0], *sample_rate_hz_, 1);
}
+std::string LevelController::ToString(
+ const AudioProcessing::Config::LevelController& config) {
+ std::stringstream ss;
+ ss << "{"
+ << "enabled: " << (config.enabled ? "true" : "false") << "}";
+ return ss.str();
+}
+
+bool LevelController::Validate(
+ const AudioProcessing::Config::LevelController& config) {
+ return true;
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.h b/webrtc/modules/audio_processing/level_controller/level_controller.h
index 65fc360..1d8e043 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.h
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.h
@@ -38,6 +38,12 @@
void Process(AudioBuffer* audio);
float GetLastGain() { return last_gain_; }
+ // Validates a config.
+ static bool Validate(const AudioProcessing::Config::LevelController& config);
+ // Dumps a config to a string.
+ static std::string ToString(
+ const AudioProcessing::Config::LevelController& config);
+
private:
class Metrics {
public:
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
index a90774f..b8033aa 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
@@ -198,16 +198,18 @@
SubmodulePerformanceTimer capture_timer;
SubmodulePerformanceTimer total_timer;
- Config config;
+ webrtc::Config config;
+ AudioProcessing::Config apm_config;
if (include_default_apm_processing) {
config.Set<DelayAgnostic>(new DelayAgnostic(true));
config.Set<ExtendedFilter>(new ExtendedFilter(true));
}
- config.Set<LevelControl>(new LevelControl(true));
+ apm_config.level_controller.enabled = true;
std::unique_ptr<AudioProcessing> apm;
apm.reset(AudioProcessing::Create(config));
ASSERT_TRUE(apm.get());
+ apm->ApplyConfig(apm_config);
ASSERT_EQ(AudioProcessing::kNoError,
apm->gain_control()->Enable(include_default_apm_processing));
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
index c470c2f..f5cea0d 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
@@ -68,6 +68,25 @@
} // namespace
+TEST(LevelControlConfigTest, ToStringEnabled) {
+ AudioProcessing::Config config;
+ config.level_controller.enabled = true;
+ EXPECT_EQ("{enabled: true}",
+ LevelController::ToString(config.level_controller));
+}
+
+TEST(LevelControlConfigTest, ToStringNotEnabled) {
+ AudioProcessing::Config config;
+ config.level_controller.enabled = false;
+ EXPECT_EQ("{enabled: false}",
+ LevelController::ToString(config.level_controller));
+}
+
+TEST(LevelControlConfigTest, DefaultValue) {
+ AudioProcessing::Config config;
+ EXPECT_FALSE(config.level_controller.enabled);
+}
+
TEST(LevelControlBitExactnessTest, DISABLED_Mono8kHz) {
const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f};
RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1,
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
index e21f42a..d2b05ad 100644
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -236,6 +236,7 @@
std::cout << "Setting used in config:" << std::endl;
}
Config config;
+ AudioProcessing::Config apm_config;
if (msg.has_aec_enabled() || settings_.use_aec) {
bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
@@ -442,9 +443,10 @@
}
if (settings_.use_lc) {
- config.Set<LevelControl>(new LevelControl(true));
+ apm_config.level_controller.enabled = true;
}
+ ap_->ApplyConfig(apm_config);
ap_->SetExtraOptions(config);
}
}
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
index a34c3ca..778b347 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -218,6 +218,7 @@
void AudioProcessingSimulator::CreateAudioProcessor() {
Config config;
+ AudioProcessing::Config apm_config;
if (settings_.use_bf && *settings_.use_bf) {
config.Set<Beamforming>(new Beamforming(
true, ParseArrayGeometry(*settings_.microphone_positions),
@@ -234,7 +235,7 @@
config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
}
if (settings_.use_lc) {
- config.Set<LevelControl>(new LevelControl(true));
+ apm_config.level_controller.enabled = *settings_.use_lc;
}
if (settings_.use_refined_adaptive_filter) {
config.Set<RefinedAdaptiveFilter>(
@@ -246,6 +247,9 @@
*settings_.use_delay_agnostic));
ap_.reset(AudioProcessing::Create(config));
+ RTC_CHECK(ap_);
+
+ ap_->ApplyConfig(apm_config);
if (settings_.use_aec) {
RTC_CHECK_EQ(AudioProcessing::kNoError,
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
index 103f8e1..9987342 100644
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -48,7 +48,8 @@
const std::string& dump_file_name);
// Constructor that uses default input files.
- explicit DebugDumpGenerator(const Config& config);
+ explicit DebugDumpGenerator(const Config& config,
+ const AudioProcessing::Config& apm_config);
~DebugDumpGenerator();
@@ -133,11 +134,18 @@
dump_file_name_(dump_file_name) {
}
-DebugDumpGenerator::DebugDumpGenerator(const Config& config)
- : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"), 32000, 2,
- ResourcePath("far32_stereo", "pcm"), 32000, 2,
- config,
- TempFilename(OutputPath(), "debug_aec")) {
+DebugDumpGenerator::DebugDumpGenerator(
+ const Config& config,
+ const AudioProcessing::Config& apm_config)
+ : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"),
+ 32000,
+ 2,
+ ResourcePath("far32_stereo", "pcm"),
+ 32000,
+ 2,
+ config,
+ TempFilename(OutputPath(), "debug_aec")) {
+ apm_->ApplyConfig(apm_config);
}
DebugDumpGenerator::~DebugDumpGenerator() {
@@ -265,7 +273,7 @@
TEST_F(DebugDumpTest, SimpleCase) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -274,7 +282,8 @@
TEST_F(DebugDumpTest, ChangeInputFormat) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
+
generator.StartRecording();
generator.Process(100);
generator.SetInputRate(48000);
@@ -291,7 +300,7 @@
TEST_F(DebugDumpTest, ChangeReverseFormat) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.SetReverseRate(48000);
@@ -303,7 +312,7 @@
TEST_F(DebugDumpTest, ChangeOutputFormat) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.SetOutputRate(48000);
@@ -315,7 +324,7 @@
TEST_F(DebugDumpTest, ToggleAec) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
@@ -330,7 +339,7 @@
TEST_F(DebugDumpTest, ToggleDelayAgnosticAec) {
Config config;
config.Set<DelayAgnostic>(new DelayAgnostic(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
@@ -345,7 +354,7 @@
TEST_F(DebugDumpTest, VerifyRefinedAdaptiveFilterExperimentalString) {
Config config;
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -370,7 +379,7 @@
Config config;
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
config.Set<EchoCanceller3>(new EchoCanceller3(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -396,7 +405,7 @@
TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) {
Config config;
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -422,7 +431,7 @@
TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
Config config;
config.Set<EchoCanceller3>(new EchoCanceller3(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -445,8 +454,9 @@
TEST_F(DebugDumpTest, VerifyLevelControllerExperimentalString) {
Config config;
- config.Set<LevelControl>(new LevelControl(true));
- DebugDumpGenerator generator(config);
+ AudioProcessing::Config apm_config;
+ apm_config.level_controller.enabled = true;
+ DebugDumpGenerator generator(config, apm_config);
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -469,7 +479,7 @@
TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
@@ -491,7 +501,7 @@
TEST_F(DebugDumpTest, ToggleAecLevel) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
EchoCancellation* aec = generator.apm()->echo_cancellation();
EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(true));
EXPECT_EQ(AudioProcessing::kNoError,
@@ -514,7 +524,7 @@
#endif
TEST_F(DebugDumpTest, MAYBE_ToggleAgc) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
@@ -528,7 +538,7 @@
TEST_F(DebugDumpTest, ToggleNs) {
Config config;
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
@@ -543,7 +553,7 @@
TEST_F(DebugDumpTest, TransientSuppressionOn) {
Config config;
config.Set<ExperimentalNs>(new ExperimentalNs(true));
- DebugDumpGenerator generator(config);
+ DebugDumpGenerator generator(config, AudioProcessing::Config());
generator.StartRecording();
generator.Process(100);
generator.StopRecording();
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index ad769ad..f1ac717 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -177,6 +177,7 @@
int extra_delay_ms = 0;
int override_delay_ms = 0;
Config config;
+ AudioProcessing::Config apm_config;
ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
for (int i = 1; i < argc; i++) {
@@ -262,8 +263,7 @@
suppression_level)));
} else if (strcmp(argv[i], "--level_control") == 0) {
- config.Set<LevelControl>(new LevelControl(true));
-
+ apm_config.level_controller.enabled = true;
} else if (strcmp(argv[i], "--extended_filter") == 0) {
config.Set<ExtendedFilter>(new ExtendedFilter(true));
@@ -452,6 +452,7 @@
FAIL() << "Unrecognized argument " << argv[i];
}
}
+ apm->ApplyConfig(apm_config);
apm->SetExtraOptions(config);
// If we're reading a protobuf file, ensure a simulation hasn't also
