Unify AGC2 experiment field trials into one
In order to experiment with AGC2 and TS at the same time, 3 field
trials are removed and merged into `WebRTC-Audio-GainController2`,
which is existing.
New parameters for the `WebRTC-Audio-GainController2` field trial:
- `switch_to_agc2`: true by default; when true, the gain control
switches to AGC2 (both for the input volume and for the adaptive
digital gain);
- `min_input_volume`: minimum input volume enforced by the input
volume controller when the applied input volume is not zero;
- `disallow_transient_suppressor_usage`: when true, TS is never
created.
Removed field trials:
- `WebRTC-Audio-Agc2-MinInputVolume`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-ApmTransientSuppressorKillSwitch`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-Audio-TransientSuppressorVadMode`: automatically inferred
from `WebRTC-Audio-GainController2`.
Bug: webrtc:7494
Change-Id: I452798c0862d71f9adae6d163fe841df05ca44d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287861
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38890}
diff --git a/modules/audio_processing/agc2/input_volume_controller.cc b/modules/audio_processing/agc2/input_volume_controller.cc
index fe1b3d9..bcc650f 100644
--- a/modules/audio_processing/agc2/input_volume_controller.cc
+++ b/modules/audio_processing/agc2/input_volume_controller.cc
@@ -50,29 +50,6 @@
return config;
}
-// Returns the minimum input volume to recommend.
-// If the "WebRTC-Audio-Agc2-MinInputVolume" field trial is specified, parses it
-// and returns the value specified after "Enabled-" if valid - i.e., in the
-// range 0-255. Otherwise returns the default value.
-// Example:
-// "WebRTC-Audio-Agc2-MinInputVolume/Enabled-80" => returns 80.
-int GetMinInputVolume() {
- constexpr int kDefaultMinInputVolume = 12;
- constexpr char kFieldTrial[] = "WebRTC-Audio-Agc2-MinInputVolume";
- if (!webrtc::field_trial::IsEnabled(kFieldTrial)) {
- return kDefaultMinInputVolume;
- }
- std::string field_trial_str = webrtc::field_trial::FindFullName(kFieldTrial);
- int min_input_volume = -1;
- sscanf(field_trial_str.c_str(), "Enabled-%d", &min_input_volume);
- if (min_input_volume >= 0 && min_input_volume <= 255) {
- return min_input_volume;
- }
- RTC_LOG(LS_WARNING) << "[AGC2] Invalid volume for " << kFieldTrial
- << ", ignored.";
- return kDefaultMinInputVolume;
-}
-
// Returns an input volume in the [`min_input_volume`, `kMaxInputVolume`] range
// that reduces `gain_error_db`, which is a gain error estimated when
// `input_volume` was applied, according to a fixed gain map.
@@ -377,7 +354,7 @@
InputVolumeController::InputVolumeController(int num_capture_channels,
const Config& config)
: num_capture_channels_(num_capture_channels),
- min_input_volume_(GetMinInputVolume()),
+ min_input_volume_(config.min_input_volume),
capture_output_used_(true),
clipped_level_step_(config.clipped_level_step),
clipped_ratio_threshold_(config.clipped_ratio_threshold),
diff --git a/modules/audio_processing/agc2/input_volume_controller.h b/modules/audio_processing/agc2/input_volume_controller.h
index 95ed160..40eae88 100644
--- a/modules/audio_processing/agc2/input_volume_controller.h
+++ b/modules/audio_processing/agc2/input_volume_controller.h
@@ -35,6 +35,9 @@
public:
// Config for the constructor.
struct Config {
+ // Minimum input volume that can be recommended. Not enforced when the
+ // applied input volume is zero outside startup.
+ int min_input_volume = 20;
// Lowest input volume level that will be applied in response to clipping.
int clipped_level_min = 70;
// Amount input volume level is lowered with every clipping event. Limited
@@ -52,13 +55,9 @@
// [`target_range_min_dbfs`, `target_range_max_dbfs`], no input volume
// adjustments are done based on the speech level. For speech levels below
// and above the range, the targets `target_range_min_dbfs` and
- // `target_range_max_dbfs` are used, respectively. The example values
- // `target_range_max_dbfs` -18 and `target_range_min_dbfs` -48 refer to a
- // configuration where the zero-digital-gain target is -18 dBFS and the
- // digital gain control is expected to compensate for speech level errors
- // up to -30 dB.
- int target_range_max_dbfs = -18;
- int target_range_min_dbfs = -48;
+ // `target_range_max_dbfs` are used, respectively.
+ int target_range_max_dbfs = -30;
+ int target_range_min_dbfs = -50;
// Number of wait frames between the recommended input volume updates.
int update_input_volume_wait_frames = 100;
// Speech probability threshold: speech probabilities below the threshold
diff --git a/modules/audio_processing/agc2/input_volume_controller_unittest.cc b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
index f1ce5c4..3979b2d 100644
--- a/modules/audio_processing/agc2/input_volume_controller_unittest.cc
+++ b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
@@ -38,7 +38,7 @@
constexpr int kInitialInputVolume = 128;
constexpr int kClippedMin = 165; // Arbitrary, but different from the default.
constexpr float kAboveClippedThreshold = 0.2f;
-constexpr int kMinMicLevel = 12;
+constexpr int kMinMicLevel = 20;
constexpr int kClippedLevelStep = 15;
constexpr float kClippedRatioThreshold = 0.1f;
constexpr int kClippedWaitFrames = 300;
@@ -56,7 +56,6 @@
using InputVolumeControllerConfig = InputVolumeController::Config;
-constexpr InputVolumeControllerConfig kDefaultInputVolumeControllerConfig{};
constexpr ClippingPredictorConfig kDefaultClippingPredictorConfig{};
std::unique_ptr<InputVolumeController> CreateInputVolumeController(
@@ -66,6 +65,7 @@
bool enable_clipping_predictor = false,
int update_input_volume_wait_frames = 0) {
InputVolumeControllerConfig config{
+ .min_input_volume = kMinMicLevel,
.clipped_level_min = kClippedMin,
.clipped_level_step = clipped_level_step,
.clipped_ratio_threshold = clipped_ratio_threshold,
@@ -82,34 +82,6 @@
config);
}
-constexpr char kMinInputVolumeFieldTrial[] = "WebRTC-Audio-Agc2-MinInputVolume";
-
-std::string GetAgcMinInputVolumeFieldTrial(const std::string& value) {
- char field_trial_buffer[64];
- rtc::SimpleStringBuilder builder(field_trial_buffer);
- builder << kMinInputVolumeFieldTrial << "/" << value << "/";
- return builder.str();
-}
-
-std::string GetAgcMinInputVolumeFieldTrialEnabled(
- int enabled_value,
- const std::string& suffix = "") {
- RTC_DCHECK_GE(enabled_value, 0);
- RTC_DCHECK_LE(enabled_value, 255);
- char field_trial_buffer[64];
- rtc::SimpleStringBuilder builder(field_trial_buffer);
- builder << kMinInputVolumeFieldTrial << "/Enabled-" << enabled_value << suffix
- << "/";
- return builder.str();
-}
-
-std::string GetAgcMinInputVolumeFieldTrial(absl::optional<int> volume) {
- if (volume.has_value()) {
- return GetAgcMinInputVolumeFieldTrialEnabled(*volume);
- }
- return GetAgcMinInputVolumeFieldTrial("Disabled");
-}
-
// (Over)writes `samples_value` for the samples in `audio_buffer`.
// When `clipped_ratio`, a value in [0, 1], is greater than 0, the corresponding
// fraction of the frame is set to a full scale value to simulate clipping.
@@ -150,31 +122,6 @@
}
}
-// Deprecated.
-// TODO(bugs.webrtc.org/7494): Delete this helper, use
-// `InputVolumeControllerTestHelper::CallAgcSequence()` instead.
-int CallAnalyzeAndRecommend(int num_calls,
- int initial_volume,
- const AudioBuffer& audio_buffer,
- float speech_probability,
- absl::optional<float> speech_level_dbfs,
- InputVolumeController& controller) {
- RTC_DCHECK(controller.capture_output_used());
- int volume = initial_volume;
- for (int n = 0; n < num_calls; ++n) {
- controller.AnalyzeInputAudio(volume, audio_buffer);
- const auto recommended_input_volume =
- controller.RecommendInputVolume(speech_probability, speech_level_dbfs);
-
- // Expect no errors: Applied volume set for every frame;
- // `RecommendInputVolume()` returns a non-empty value.
- EXPECT_TRUE(recommended_input_volume.has_value());
-
- volume = *recommended_input_volume;
- }
- return volume;
-}
-
// Reads a given number of 10 ms chunks from a PCM file and feeds them to
// `InputVolumeController`.
class SpeechSamplesReader {
@@ -379,24 +326,12 @@
};
class InputVolumeControllerParametrizedTest
- : public ::testing::TestWithParam<absl::optional<int>> {
- protected:
- InputVolumeControllerParametrizedTest()
- : field_trials_(GetAgcMinInputVolumeFieldTrial(GetParam())) {}
-
- int GetMinInputVolume() const { return GetParam().value_or(kMinMicLevel); }
-
- private:
- test::ScopedFieldTrials field_trials_;
-};
-
-INSTANTIATE_TEST_SUITE_P(,
- InputVolumeControllerParametrizedTest,
- ::testing::Values(absl::nullopt, 12, 20));
+ : public ::testing::TestWithParam<int> {};
TEST_P(InputVolumeControllerParametrizedTest,
StartupMinVolumeConfigurationIsRespected) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
EXPECT_EQ(*helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel),
@@ -404,7 +339,9 @@
}
TEST_P(InputVolumeControllerParametrizedTest, MicVolumeResponseToRmsError) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -446,7 +383,10 @@
}
TEST_P(InputVolumeControllerParametrizedTest, MicVolumeIsLimited) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ const int min_input_volume = GetParam();
+ config.min_input_volume = min_input_volume;
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -492,16 +432,18 @@
// Won't go lower than the minimum.
volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
kHighSpeechProbability, 22.0f);
- EXPECT_EQ(volume, std::max(18, GetMinInputVolume()));
+ EXPECT_EQ(volume, std::max(18, min_input_volume));
volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
kHighSpeechProbability, 22.0f);
- EXPECT_EQ(volume, std::max(12, GetMinInputVolume()));
+ EXPECT_EQ(volume, std::max(12, min_input_volume));
}
TEST_P(InputVolumeControllerParametrizedTest, NoActionWhileMuted) {
- InputVolumeControllerTestHelper helper_1;
- InputVolumeControllerTestHelper helper_2;
+ InputVolumeControllerTestHelper helper_1(
+ /*config=*/{.min_input_volume = GetParam()});
+ InputVolumeControllerTestHelper helper_2(
+ /*config=*/{.min_input_volume = GetParam()});
int volume_1 = *helper_1.CallAgcSequence(/*applied_input_volume=*/255,
kHighSpeechProbability, kSpeechLevel,
@@ -531,7 +473,8 @@
TEST_P(InputVolumeControllerParametrizedTest,
UnmutingChecksVolumeWithoutRaising) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel);
@@ -548,7 +491,9 @@
}
TEST_P(InputVolumeControllerParametrizedTest, UnmutingRaisesTooLowVolume) {
- InputVolumeControllerTestHelper helper;
+ const int min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = min_input_volume});
helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel);
@@ -560,12 +505,14 @@
EXPECT_EQ(
helper.CallRecommendInputVolume(/*num_calls=*/1, kInputVolume,
kHighSpeechProbability, kSpeechLevel),
- GetMinInputVolume());
+ min_input_volume);
}
TEST_P(InputVolumeControllerParametrizedTest,
ManualLevelChangeResultsInNoSetMicCall) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -589,7 +536,9 @@
TEST_P(InputVolumeControllerParametrizedTest,
RecoveryAfterManualLevelChangeFromMax) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -621,7 +570,10 @@
// of the input volume.
TEST_P(InputVolumeControllerParametrizedTest,
EnforceMinInputVolumeDuringUpwardsAdjustment) {
- InputVolumeControllerTestHelper helper;
+ const int min_input_volume = GetParam();
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = min_input_volume;
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -631,19 +583,19 @@
/*num_calls=*/1, /*initial_volume=*/1, kHighSpeechProbability, -17.0f);
// Trigger an upward adjustment of the input volume.
- EXPECT_EQ(volume, GetMinInputVolume());
+ EXPECT_EQ(volume, min_input_volume);
volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
kHighSpeechProbability, -29.0f);
- EXPECT_EQ(volume, GetMinInputVolume());
+ EXPECT_EQ(volume, min_input_volume);
volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
kHighSpeechProbability, -30.0f);
- EXPECT_EQ(volume, GetMinInputVolume());
+ EXPECT_EQ(volume, min_input_volume);
// After a number of consistently low speech level observations, the input
// volume is eventually raised above the minimum.
volume = helper.CallRecommendInputVolume(/*num_calls=*/10, volume,
kHighSpeechProbability, -38.0f);
- EXPECT_GT(volume, GetMinInputVolume());
+ EXPECT_GT(volume, min_input_volume);
}
// Checks that, when the min mic level override is specified, AGC immediately
@@ -651,7 +603,9 @@
// minimum gain to enforce.
TEST_P(InputVolumeControllerParametrizedTest,
RecoveryAfterManualLevelChangeBelowMin) {
- InputVolumeControllerTestHelper helper;
+ const int min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = min_input_volume});
int volume = *helper.CallAgcSequence(kInitialInputVolume,
kHighSpeechProbability, kSpeechLevel);
@@ -659,11 +613,12 @@
// AGC won't take any action.
volume = helper.CallRecommendInputVolume(
/*num_calls=*/1, /*initial_volume=*/1, kHighSpeechProbability, -17.0f);
- EXPECT_EQ(volume, GetMinInputVolume());
+ EXPECT_EQ(volume, min_input_volume);
}
TEST_P(InputVolumeControllerParametrizedTest, NoClippingHasNoImpact) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel);
@@ -673,7 +628,8 @@
TEST_P(InputVolumeControllerParametrizedTest,
ClippingUnderThresholdHasNoImpact) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel);
@@ -682,7 +638,8 @@
}
TEST_P(InputVolumeControllerParametrizedTest, ClippingLowersVolume) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
kSpeechLevel);
@@ -692,7 +649,8 @@
TEST_P(InputVolumeControllerParametrizedTest,
WaitingPeriodBetweenClippingChecks) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
kSpeechLevel);
@@ -710,7 +668,9 @@
}
TEST_P(InputVolumeControllerParametrizedTest, ClippingLoweringIsLimited) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
helper.CallAgcSequence(/*applied_input_volume=*/180, kHighSpeechProbability,
kSpeechLevel);
@@ -725,7 +685,8 @@
TEST_P(InputVolumeControllerParametrizedTest,
ClippingMaxIsRespectedWhenEqualToLevel) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
kSpeechLevel);
@@ -740,7 +701,9 @@
TEST_P(InputVolumeControllerParametrizedTest,
ClippingMaxIsRespectedWhenHigherThanLevel) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
helper.CallAgcSequence(/*applied_input_volume=*/200, kHighSpeechProbability,
kSpeechLevel);
@@ -758,7 +721,9 @@
}
TEST_P(InputVolumeControllerParametrizedTest, UserCanRaiseVolumeAfterClipping) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
helper.CallAgcSequence(/*applied_input_volume=*/225, kHighSpeechProbability,
kSpeechLevel);
@@ -787,7 +752,9 @@
TEST_P(InputVolumeControllerParametrizedTest,
ClippingDoesNotPullLowVolumeBackUp) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
helper.CallAgcSequence(/*applied_input_volume=*/80, kHighSpeechProbability,
kSpeechLevel);
@@ -798,7 +765,8 @@
}
TEST_P(InputVolumeControllerParametrizedTest, TakesNoActionOnZeroMicVolume) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = GetParam()});
helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
kSpeechLevel);
@@ -809,7 +777,9 @@
}
TEST_P(InputVolumeControllerParametrizedTest, ClippingDetectionLowersVolume) {
- InputVolumeControllerTestHelper helper;
+ InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+ config.min_input_volume = GetParam();
+ InputVolumeControllerTestHelper helper(config);
int volume = *helper.CallAgcSequence(/*applied_input_volume=*/255,
kHighSpeechProbability, kSpeechLevel,
/*num_calls=*/1);
@@ -829,298 +799,6 @@
EXPECT_EQ(volume, 240);
}
-TEST(InputVolumeControllerTest, MinInputVolumeDefault) {
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
- EXPECT_EQ(controller->channel_controllers_[0]->min_input_volume(),
- kMinMicLevel);
-}
-
-TEST(InputVolumeControllerTest, MinInputVolumeDisabled) {
- for (const std::string& field_trial_suffix : {"", "_20220210"}) {
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrial("Disabled" + field_trial_suffix));
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
-
- EXPECT_EQ(controller->channel_controllers_[0]->min_input_volume(),
- kMinMicLevel);
- }
-}
-
-// Checks that a field-trial parameter outside of the valid range [0,255] is
-// ignored.
-TEST(InputVolumeControllerTest, MinInputVolumeOutOfRangeAbove) {
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrial("Enabled-256"));
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
- EXPECT_EQ(controller->channel_controllers_[0]->min_input_volume(),
- kMinMicLevel);
-}
-
-// Checks that a field-trial parameter outside of the valid range [0,255] is
-// ignored.
-TEST(InputVolumeControllerTest, MinInputVolumeOutOfRangeBelow) {
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrial("Enabled--1"));
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
- EXPECT_EQ(controller->channel_controllers_[0]->min_input_volume(),
- kMinMicLevel);
-}
-
-// Verifies that a valid experiment changes the minimum microphone level. The
-// start volume is larger than the min level and should therefore not be
-// changed.
-TEST(InputVolumeControllerTest, MinInputVolumeEnabled50) {
- constexpr int kMinInputVolume = 50;
- for (const std::string& field_trial_suffix : {"", "_20220210"}) {
- SCOPED_TRACE(field_trial_suffix);
- test::ScopedFieldTrials field_trial(GetAgcMinInputVolumeFieldTrialEnabled(
- kMinInputVolume, field_trial_suffix));
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
-
- EXPECT_EQ(controller->channel_controllers_[0]->min_input_volume(),
- kMinInputVolume);
- }
-}
-
-// Checks that, when the "WebRTC-Audio-Agc2-MinInputVolume" field trial is
-// specified with a valid value, the mic level never gets lowered beyond the
-// override value in the presence of clipping.
-TEST(InputVolumeControllerTest, MinInputVolumeCheckMinLevelWithClipping) {
- constexpr int kMinInputVolume = 250;
-
- // Create and initialize two AGCs by specifying and leaving unspecified the
- // relevant field trial.
- const auto factory = []() {
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
- controller->Initialize();
- return controller;
- };
- std::unique_ptr<InputVolumeController> controller = factory();
- std::unique_ptr<InputVolumeController> controller_with_override;
- {
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrialEnabled(kMinInputVolume));
- controller_with_override = factory();
- }
-
- // Create a test input signal which containts 80% of clipped samples.
- AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
- 1);
- WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
- audio_buffer);
-
- // Simulate 4 seconds of clipping; it is expected to trigger a downward
- // adjustment of the analog gain. Use low speech probability to limit the
- // volume changes to clipping handling.
- const int volume = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kLowSpeechProbability, /*speech_level_dbfs=*/-42.0f, *controller);
- const int volume_with_override = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kLowSpeechProbability, /*speech_level_dbfs=*/-42.0f,
- *controller_with_override);
-
- // Make sure that an adaptation occurred.
- ASSERT_GT(volume, 0);
-
- // Check that the test signal triggers a larger downward adaptation for
- // `controller`, which is allowed to reach a lower gain.
- EXPECT_GT(volume_with_override, volume);
- // Check that the gain selected by `controller_with_override` equals the
- // minimum value overridden via field trial.
- EXPECT_EQ(volume_with_override, kMinInputVolume);
-}
-
-// Checks that, when the "WebRTC-Audio-Agc2-MinInputVolume" field trial is
-// specified with a valid value, the mic level never gets lowered beyond the
-// override value in the presence of clipping when RMS error is not empty.
-// TODO(webrtc:7494): Revisit the test after moving the number of update wait
-// frames to APM config. The test passes but internally the gain update timing
-// differs.
-TEST(InputVolumeControllerTest,
- MinInputVolumeCheckMinLevelWithClippingWithRmsError) {
- constexpr int kMinInputVolume = 250;
-
- // Create and initialize two AGCs by specifying and leaving unspecified the
- // relevant field trial.
- const auto factory = []() {
- std::unique_ptr<InputVolumeController> controller =
- CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
- kClippedWaitFrames);
- controller->Initialize();
- return controller;
- };
- std::unique_ptr<InputVolumeController> controller = factory();
- std::unique_ptr<InputVolumeController> controller_with_override;
- {
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrialEnabled(kMinInputVolume));
- controller_with_override = factory();
- }
-
- // Create a test input signal which containts 80% of clipped samples.
- AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
- 1);
- WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
- audio_buffer);
-
- // Simulate 4 seconds of clipping; it is expected to trigger a downward
- // adjustment of the analog gain.
- const int volume = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kHighSpeechProbability,
- /*speech_level_dbfs=*/-18.0f, *controller);
- const int volume_with_override = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kHighSpeechProbability,
- /*speech_level_dbfs=*/-18.0f, *controller_with_override);
-
- // Make sure that an adaptation occurred.
- ASSERT_GT(volume, 0);
-
- // Check that the test signal triggers a larger downward adaptation for
- // `controller`, which is allowed to reach a lower gain.
- EXPECT_GT(volume_with_override, volume);
-
- // Check that the gain selected by `controller_with_override` equals the
- // minimum value overridden via field trial.
- EXPECT_EQ(volume_with_override, kMinInputVolume);
-}
-
-// Checks that, when the "WebRTC-Audio-Agc2-MinInputVolume" field trial is
-// specified with a value lower than the `clipped_level_min`, the behavior of
-// the analog gain controller is the same as that obtained when the field trial
-// is not specified.
-TEST(InputVolumeControllerTest, MinInputVolumeCompareMicLevelWithClipping) {
- // Create and initialize two AGCs by specifying and leaving unspecified the
- // relevant field trial.
- const auto factory = []() {
- // Use a large clipped level step to more quickly decrease the analog gain
- // with clipping.
- InputVolumeControllerConfig config = kDefaultInputVolumeControllerConfig;
- config.clipped_level_step = 64;
- config.clipped_ratio_threshold = kClippedRatioThreshold;
- config.clipped_wait_frames = kClippedWaitFrames;
- auto controller = std::make_unique<InputVolumeController>(
- /*num_capture_channels=*/1, config);
- controller->Initialize();
- return controller;
- };
- std::unique_ptr<InputVolumeController> controller = factory();
- std::unique_ptr<InputVolumeController> controller_with_override;
- {
- constexpr int kMinInputVolume = 20;
- static_assert(kDefaultInputVolumeControllerConfig.clipped_level_min >=
- kMinInputVolume,
- "Use a lower override value.");
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrialEnabled(kMinInputVolume));
- controller_with_override = factory();
- }
-
- // Create a test input signal which containts 80% of clipped samples.
- AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
- 1);
- WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
- audio_buffer);
-
- // Simulate 4 seconds of clipping; it is expected to trigger a downward
- // adjustment of the analog gain. Use low speech probability to limit the
- // volume changes to clipping handling.
- const int volume = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kLowSpeechProbability, /*speech_level_dbfs=*/-18, *controller);
- const int volume_with_override = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- kLowSpeechProbability, /*speech_level_dbfs=*/-18,
- *controller_with_override);
-
- // Make sure that an adaptation occurred.
- ASSERT_GT(volume, 0);
-
- // Check that the selected analog gain is the same for both controllers and
- // that it equals the minimum level reached when clipping is handled. That is
- // expected because the minimum microphone level override is less than the
- // minimum level used when clipping is detected.
- EXPECT_EQ(volume, volume_with_override);
- EXPECT_EQ(volume_with_override,
- kDefaultInputVolumeControllerConfig.clipped_level_min);
-}
-
-// Checks that, when the "WebRTC-Audio-Agc2-MinInputVolume" field trial is
-// specified with a value lower than the `clipped_level_min`, the behavior of
-// the analog gain controller is the same as that obtained when the field trial
-// is not specified.
-// TODO(webrtc:7494): Revisit the test after moving the number of update wait
-// frames to APM config. The test passes but internally the gain update timing
-// differs.
-TEST(InputVolumeControllerTest,
- MinInputVolumeCompareMicLevelWithClippingWithRmsError) {
- // Create and initialize two AGCs by specifying and leaving unspecified the
- // relevant field trial.
- const auto factory = []() {
- // Use a large clipped level step to more quickly decrease the analog gain
- // with clipping.
- InputVolumeControllerConfig config = kDefaultInputVolumeControllerConfig;
- config.clipped_level_step = 64;
- config.clipped_ratio_threshold = kClippedRatioThreshold;
- config.clipped_wait_frames = kClippedWaitFrames;
- auto controller = std::make_unique<InputVolumeController>(
- /*num_capture_channels=*/1, config);
- controller->Initialize();
- return controller;
- };
- std::unique_ptr<InputVolumeController> controller = factory();
- std::unique_ptr<InputVolumeController> controller_with_override;
- {
- constexpr int kMinInputVolume = 20;
- static_assert(kDefaultInputVolumeControllerConfig.clipped_level_min >=
- kMinInputVolume,
- "Use a lower override value.");
- test::ScopedFieldTrials field_trial(
- GetAgcMinInputVolumeFieldTrialEnabled(kMinInputVolume));
- controller_with_override = factory();
- }
-
- // Create a test input signal which containts 80% of clipped samples.
- AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
- 1);
- WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
- audio_buffer);
-
- const int volume = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- /*speech_probability=*/0.7f,
- /*speech_level_dbfs=*/-18.0f, *controller);
- const int volume_with_override = CallAnalyzeAndRecommend(
- /*num_calls=*/400, kInitialInputVolume, audio_buffer,
- /*speech_probability=*/0.7f,
- /*speech_level_dbfs=*/-18.0f, *controller_with_override);
-
- // Make sure that an adaptation occurred.
- ASSERT_GT(volume, 0);
-
- // Check that the selected analog gain is the same for both controllers and
- // that it equals the minimum level reached when clipping is handled. That is
- // expected because the minimum microphone level override is less than the
- // minimum level used when clipping is detected.
- EXPECT_EQ(volume, volume_with_override);
- EXPECT_EQ(volume_with_override,
- kDefaultInputVolumeControllerConfig.clipped_level_min);
-}
-
// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_level_step`.
// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_ratio_threshold`.
// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_wait_frames`.
@@ -1416,6 +1094,42 @@
ASSERT_GT(volume_wait_100, kInputVolume);
}
+INSTANTIATE_TEST_SUITE_P(,
+ InputVolumeControllerParametrizedTest,
+ ::testing::Values(12, 20));
+
+TEST(InputVolumeControllerTest,
+ MinInputVolumeEnforcedWithClippingWhenAboveClippedLevelMin) {
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = 80, .clipped_level_min = 70});
+
+ // Trigger a downward adjustment caused by clipping input. Use a low speech
+ // probability to limit the volume changes to clipping handling.
+ WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+ helper.audio_buffer);
+ constexpr int kNumCalls = 800;
+ helper.CallAgcSequence(/*applied_input_volume=*/100, kLowSpeechProbability,
+ /*speech_level_dbfs=*/-18.0f, kNumCalls);
+
+ EXPECT_EQ(helper.controller.recommended_input_volume(), 80);
+}
+
+TEST(InputVolumeControllerTest,
+ ClippedlevelMinEnforcedWithClippingWhenAboveMinInputVolume) {
+ InputVolumeControllerTestHelper helper(
+ /*config=*/{.min_input_volume = 70, .clipped_level_min = 80});
+
+ // Trigger a downward adjustment caused by clipping input. Use a low speech
+ // probability to limit the volume changes to clipping handling.
+ WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+ helper.audio_buffer);
+ constexpr int kNumCalls = 800;
+ helper.CallAgcSequence(/*applied_input_volume=*/100, kLowSpeechProbability,
+ /*speech_level_dbfs=*/-18.0f, kNumCalls);
+
+ EXPECT_EQ(helper.controller.recommended_input_volume(), 80);
+}
+
TEST(InputVolumeControllerTest, SpeechRatioThresholdIsEffective) {
constexpr int kInputVolume = kInitialInputVolume;
// Create two input volume controllers with 10 frames between volume updates
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index d28a44b..e92ae6d 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -69,29 +69,6 @@
"WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch");
}
-// If the "WebRTC-Audio-TransientSuppressorVadMode" field trial is unspecified,
-// returns `TransientSuppressor::VadMode::kDefault`, otherwise parses the field
-// trial and returns the specified mode:
-// - WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default returns `kDefault`;
-// - WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad returns `kRnnVad`;
-// - WebRTC-Audio-TransientSuppressorVadMode/Enabled-NoVad returns `kNoVad`.
-TransientSuppressor::VadMode GetTransientSuppressorVadMode() {
- constexpr char kFieldTrial[] = "WebRTC-Audio-TransientSuppressorVadMode";
- std::string full_name = webrtc::field_trial::FindFullName(kFieldTrial);
- if (full_name.empty() || absl::EndsWith(full_name, "-Default")) {
- return TransientSuppressor::VadMode::kDefault;
- }
- if (absl::EndsWith(full_name, "-RnnVad")) {
- return TransientSuppressor::VadMode::kRnnVad;
- }
- if (absl::EndsWith(full_name, "-NoVad")) {
- return TransientSuppressor::VadMode::kNoVad;
- }
- // Fallback to default.
- RTC_LOG(LS_WARNING) << "Invalid parameter for " << kFieldTrial;
- return TransientSuppressor::VadMode::kDefault;
-}
-
// Identify the native processing rate that best handles a sample rate.
int SuitableProcessRate(int minimum_rate,
int max_splitting_rate,
@@ -325,17 +302,44 @@
return error_code;
}
-const absl::optional<AudioProcessingImpl::GainController2ConfigOverride>
-GetGainController2ConfigOverride() {
+using DownmixMethod = AudioProcessing::Config::Pipeline::DownmixMethod;
+
+void SetDownmixMethod(AudioBuffer& buffer, DownmixMethod method) {
+ switch (method) {
+ case DownmixMethod::kAverageChannels:
+ buffer.set_downmixing_by_averaging();
+ break;
+ case DownmixMethod::kUseFirstChannel:
+ buffer.set_downmixing_to_specific_channel(/*channel=*/0);
+ break;
+ }
+}
+
+constexpr int kUnspecifiedDataDumpInputVolume = -100;
+
+} // namespace
+
+// Throughout webrtc, it's assumed that success is represented by zero.
+static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
+
+absl::optional<AudioProcessingImpl::GainController2ExperimentParams>
+AudioProcessingImpl::GetGainController2ExperimentParams() {
constexpr char kFieldTrialName[] = "WebRTC-Audio-GainController2";
if (!field_trial::IsEnabled(kFieldTrialName)) {
return absl::nullopt;
}
- constexpr InputVolumeController::Config kDefaultInputVolumeControllerConfig;
-
FieldTrialFlag enabled("Enabled", false);
+
+ // Whether the gain control should switch to AGC2. Enabled by default.
+ FieldTrialParameter<bool> switch_to_agc2("switch_to_agc2", true);
+
+ // AGC2 input volume controller configuration.
+ constexpr InputVolumeController::Config kDefaultInputVolumeControllerConfig;
+ FieldTrialConstrained<int> min_input_volume(
+ "min_input_volume", kDefaultInputVolumeControllerConfig.min_input_volume,
+ 0, 255);
FieldTrialConstrained<int> clipped_level_min(
"clipped_level_min",
kDefaultInputVolumeControllerConfig.clipped_level_min, 0, 255);
@@ -369,9 +373,9 @@
"speech_ratio_threshold",
kDefaultInputVolumeControllerConfig.speech_ratio_threshold, 0, 1);
+ // AGC2 adaptive digital controller configuration.
constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
kDefaultAdaptiveDigitalConfig;
-
FieldTrialConstrained<double> headroom_db(
"headroom_db", kDefaultAdaptiveDigitalConfig.headroom_db, 0,
absl::nullopt);
@@ -390,83 +394,102 @@
kDefaultAdaptiveDigitalConfig.max_output_noise_level_dbfs, absl::nullopt,
0);
+ // Transient suppressor.
+ FieldTrialParameter<bool> disallow_transient_suppressor_usage(
+ "disallow_transient_suppressor_usage", false);
+
// Field-trial based override for the input volume controller and adaptive
// digital configs.
- const std::string field_trial_name =
- field_trial::FindFullName(kFieldTrialName);
-
ParseFieldTrial(
- {&enabled, &clipped_level_min, &clipped_level_step,
- &clipped_ratio_threshold, &clipped_wait_frames,
+ {&enabled, &switch_to_agc2, &min_input_volume, &clipped_level_min,
+ &clipped_level_step, &clipped_ratio_threshold, &clipped_wait_frames,
&enable_clipping_predictor, &target_range_max_dbfs,
&target_range_min_dbfs, &update_input_volume_wait_frames,
&speech_probability_threshold, &speech_ratio_threshold, &headroom_db,
&max_gain_db, &initial_gain_db, &max_gain_change_db_per_second,
- &max_output_noise_level_dbfs},
- field_trial_name);
-
+ &max_output_noise_level_dbfs, &disallow_transient_suppressor_usage},
+ field_trial::FindFullName(kFieldTrialName));
// Checked already by `IsEnabled()` before parsing, therefore always true.
RTC_DCHECK(enabled);
- return AudioProcessingImpl::GainController2ConfigOverride{
- .input_volume_controller_config =
- {
- .clipped_level_min = static_cast<int>(clipped_level_min.Get()),
- .clipped_level_step = static_cast<int>(clipped_level_step.Get()),
- .clipped_ratio_threshold =
- static_cast<float>(clipped_ratio_threshold.Get()),
- .clipped_wait_frames =
- static_cast<int>(clipped_wait_frames.Get()),
- .enable_clipping_predictor =
- static_cast<bool>(enable_clipping_predictor.Get()),
- .target_range_max_dbfs =
- static_cast<int>(target_range_max_dbfs.Get()),
- .target_range_min_dbfs =
- static_cast<int>(target_range_min_dbfs.Get()),
- .update_input_volume_wait_frames =
- static_cast<int>(update_input_volume_wait_frames.Get()),
- .speech_probability_threshold =
- static_cast<float>(speech_probability_threshold.Get()),
- .speech_ratio_threshold =
- static_cast<float>(speech_ratio_threshold.Get()),
- },
- .adaptive_digital_config =
- {
- .headroom_db = static_cast<float>(headroom_db.Get()),
- .max_gain_db = static_cast<float>(max_gain_db.Get()),
- .initial_gain_db = static_cast<float>(initial_gain_db.Get()),
- .max_gain_change_db_per_second =
- static_cast<float>(max_gain_change_db_per_second.Get()),
- .max_output_noise_level_dbfs =
- static_cast<float>(max_output_noise_level_dbfs.Get()),
- },
- };
+ const bool do_not_change_agc_config = !switch_to_agc2.Get();
+ if (do_not_change_agc_config && !disallow_transient_suppressor_usage.Get()) {
+ // Return an unspecifed value since, in this case, both the AGC2 and TS
+ // configurations won't be adjusted.
+ return absl::nullopt;
+ }
+ using Params = AudioProcessingImpl::GainController2ExperimentParams;
+ if (do_not_change_agc_config) {
+ // Return a value that leaves the AGC2 config unchanged and that always
+ // disables TS.
+ return Params{.agc2_config = absl::nullopt,
+ .disallow_transient_suppressor_usage = true};
+ }
+ // Return a value that switches all the gain control to AGC2.
+ return Params{
+ .agc2_config =
+ Params::Agc2Config{
+ .input_volume_controller =
+ {
+ .min_input_volume = min_input_volume.Get(),
+ .clipped_level_min = clipped_level_min.Get(),
+ .clipped_level_step = clipped_level_step.Get(),
+ .clipped_ratio_threshold =
+ static_cast<float>(clipped_ratio_threshold.Get()),
+ .clipped_wait_frames = clipped_wait_frames.Get(),
+ .enable_clipping_predictor =
+ enable_clipping_predictor.Get(),
+ .target_range_max_dbfs = target_range_max_dbfs.Get(),
+ .target_range_min_dbfs = target_range_min_dbfs.Get(),
+ .update_input_volume_wait_frames =
+ update_input_volume_wait_frames.Get(),
+ .speech_probability_threshold = static_cast<float>(
+ speech_probability_threshold.Get()),
+ .speech_ratio_threshold =
+ static_cast<float>(speech_ratio_threshold.Get()),
+ },
+ .adaptive_digital_controller =
+ {
+ .headroom_db = static_cast<float>(headroom_db.Get()),
+ .max_gain_db = static_cast<float>(max_gain_db.Get()),
+ .initial_gain_db =
+ static_cast<float>(initial_gain_db.Get()),
+ .max_gain_change_db_per_second = static_cast<float>(
+ max_gain_change_db_per_second.Get()),
+ .max_output_noise_level_dbfs =
+ static_cast<float>(max_output_noise_level_dbfs.Get()),
+ }},
+ .disallow_transient_suppressor_usage =
+ disallow_transient_suppressor_usage.Get()};
}
-// If `disallow_transient_supporessor_usage` is true, disables transient
-// suppression. When `gain_controller2_config_override` is specified,
-// switches all gain control to AGC2.
-AudioProcessing::Config AdjustConfig(
+AudioProcessing::Config AudioProcessingImpl::AdjustConfig(
const AudioProcessing::Config& config,
- bool disallow_transient_supporessor_usage,
- const absl::optional<AudioProcessingImpl::GainController2ConfigOverride>&
- gain_controller2_config_override) {
+ const absl::optional<AudioProcessingImpl::GainController2ExperimentParams>&
+ experiment_params) {
+ if (!experiment_params.has_value() ||
+ (!experiment_params->agc2_config.has_value() &&
+ !experiment_params->disallow_transient_suppressor_usage)) {
+ // When the experiment parameters are unspecified or when the AGC and TS
+ // configuration are not overridden, return the unmodified configuration.
+ return config;
+ }
+
AudioProcessing::Config adjusted_config = config;
// Override the transient suppressor configuration.
- if (disallow_transient_supporessor_usage) {
+ if (experiment_params->disallow_transient_suppressor_usage) {
adjusted_config.transient_suppression.enabled = false;
}
// Override the auto gain control configuration if the AGC1 analog gain
- // controller is active and `gain_controller2_config_override` is
- // specified.
+ // controller is active and `experiment_params->agc2_config` is specified.
const bool agc1_analog_enabled =
config.gain_controller1.enabled &&
(config.gain_controller1.mode ==
AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
config.gain_controller1.analog_gain_controller.enabled);
- if (agc1_analog_enabled && gain_controller2_config_override.has_value()) {
+ if (agc1_analog_enabled && experiment_params->agc2_config.has_value()) {
// Check that the unadjusted AGC config meets the preconditions.
const bool hybrid_agc_config_detected =
config.gain_controller1.enabled &&
@@ -499,7 +522,7 @@
adjusted_config.gain_controller2.enabled = true;
adjusted_config.gain_controller2.input_volume_controller.enabled = true;
adjusted_config.gain_controller2.adaptive_digital =
- gain_controller2_config_override->adaptive_digital_config;
+ experiment_params->agc2_config->adaptive_digital_controller;
adjusted_config.gain_controller2.adaptive_digital.enabled = true;
}
}
@@ -507,26 +530,21 @@
return adjusted_config;
}
-using DownmixMethod = AudioProcessing::Config::Pipeline::DownmixMethod;
-
-void SetDownmixMethod(AudioBuffer& buffer, DownmixMethod method) {
- switch (method) {
- case DownmixMethod::kAverageChannels:
- buffer.set_downmixing_by_averaging();
- break;
- case DownmixMethod::kUseFirstChannel:
- buffer.set_downmixing_to_specific_channel(/*channel=*/0);
- break;
+TransientSuppressor::VadMode AudioProcessingImpl::GetTransientSuppressorVadMode(
+ const absl::optional<AudioProcessingImpl::GainController2ExperimentParams>&
+ params) {
+ if (params.has_value() && params->agc2_config.has_value() &&
+ !params->disallow_transient_suppressor_usage) {
+ // When the experiment is active, the gain control switches to AGC2 and TS
+ // can be active, use the RNN VAD to control TS. This choice will also
+ // disable the internal RNN VAD in AGC2.
+ return TransientSuppressor::VadMode::kRnnVad;
}
+ // If TS is disabled, the returned value does not matter. If enabled, use the
+ // default VAD.
+ return TransientSuppressor::VadMode::kDefault;
}
-constexpr int kUnspecifiedDataDumpInputVolume = -100;
-
-} // namespace
-
-// Throughout webrtc, it's assumed that success is represented by zero.
-static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
-
AudioProcessingImpl::SubmoduleStates::SubmoduleStates(
bool capture_post_processor_enabled,
bool render_pre_processor_enabled,
@@ -644,20 +662,17 @@
: data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
use_setup_specific_default_aec3_config_(
UseSetupSpecificDefaultAec3Congfig()),
- gain_controller2_config_override_(GetGainController2ConfigOverride()),
+ gain_controller2_experiment_params_(GetGainController2ExperimentParams()),
use_denormal_disabler_(
!field_trial::IsEnabled("WebRTC-ApmDenormalDisablerKillSwitch")),
- disallow_transient_supporessor_usage_(
- field_trial::IsEnabled("WebRTC-ApmTransientSuppressorKillSwitch")),
- transient_suppressor_vad_mode_(GetTransientSuppressorVadMode()),
+ transient_suppressor_vad_mode_(
+ GetTransientSuppressorVadMode(gain_controller2_experiment_params_)),
capture_runtime_settings_(RuntimeSettingQueueSize()),
render_runtime_settings_(RuntimeSettingQueueSize()),
capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
render_runtime_settings_enqueuer_(&render_runtime_settings_),
echo_control_factory_(std::move(echo_control_factory)),
- config_(AdjustConfig(config,
- disallow_transient_supporessor_usage_,
- gain_controller2_config_override_)),
+ config_(AdjustConfig(config, gain_controller2_experiment_params_)),
submodule_states_(!!capture_post_processor,
!!render_pre_processor,
!!capture_analyzer),
@@ -893,8 +908,7 @@
MutexLock lock_capture(&mutex_capture_);
const auto adjusted_config =
- AdjustConfig(config, disallow_transient_supporessor_usage_,
- gain_controller2_config_override_);
+ AdjustConfig(config, gain_controller2_experiment_params_);
RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: "
<< adjusted_config.ToString();
@@ -2340,11 +2354,16 @@
if (!submodules_.gain_controller2 || config_has_changed) {
const bool use_internal_vad =
transient_suppressor_vad_mode_ != TransientSuppressor::VadMode::kRnnVad;
+ const bool input_volume_controller_config_overridden =
+ gain_controller2_experiment_params_.has_value() &&
+ gain_controller2_experiment_params_->agc2_config.has_value();
+ const InputVolumeController::Config input_volume_controller_config =
+ input_volume_controller_config_overridden
+ ? gain_controller2_experiment_params_->agc2_config
+ ->input_volume_controller
+ : InputVolumeController::Config{};
submodules_.gain_controller2 = std::make_unique<GainController2>(
- config_.gain_controller2,
- gain_controller2_config_override_.has_value()
- ? gain_controller2_config_override_->input_volume_controller_config
- : InputVolumeController::Config{},
+ config_.gain_controller2, input_volume_controller_config,
proc_fullband_sample_rate_hz(), num_proc_channels(), use_internal_vad);
submodules_.gain_controller2->SetCaptureOutputUsed(
capture_.capture_output_used);
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 0f74c30..8ee07ed 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -138,14 +138,6 @@
AudioProcessing::Config GetConfig() const override;
- // TODO(bugs.webrtc.org/7494): Remove when the related field trial is
- // removed.
- struct GainController2ConfigOverride {
- InputVolumeController::Config input_volume_controller_config;
- AudioProcessing::Config::GainController2::AdaptiveDigital
- adaptive_digital_config;
- };
-
protected:
// Overridden in a mock.
virtual void InitializeLocked()
@@ -199,19 +191,47 @@
static std::atomic<int> instance_count_;
const bool use_setup_specific_default_aec3_config_;
- // TODO(bugs.webrtc.org/7494): Remove when the linked field trial is removed.
- // Override based on the "WebRTC-Audio-GainController2" field trial for the
- // AGC2 input volume controller and adaptive digital controller configuration.
- const absl::optional<GainController2ConfigOverride>
- gain_controller2_config_override_;
+ // Parameters for the "GainController2" experiment which determines whether
+ // the following APM sub-modules are created and, if so, their configurations:
+ // AGC2 (`gain_controller2`), AGC1 (`gain_control`, `agc_manager`) and TS
+ // (`transient_suppressor`).
+ // TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
+ // field trial is removed.
+ struct GainController2ExperimentParams {
+ struct Agc2Config {
+ InputVolumeController::Config input_volume_controller;
+ AudioProcessing::Config::GainController2::AdaptiveDigital
+ adaptive_digital_controller;
+ };
+ // When `agc2_config` is specified, all gain control switches to AGC2 and
+ // the configuration is overridden.
+ absl::optional<Agc2Config> agc2_config;
+ // When true, the transient suppressor submodule is never created regardless
+ // of the APM configuration.
+ bool disallow_transient_suppressor_usage;
+ };
+ // Specified when the "WebRTC-Audio-GainController2" field trial is specified.
+ // TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
+ // field trial is removed.
+ const absl::optional<GainController2ExperimentParams>
+ gain_controller2_experiment_params_;
+
+ // Parses the "WebRTC-Audio-GainController2" field trial. If disabled, returns
+ // an unspecified value.
+ static absl::optional<GainController2ExperimentParams>
+ GetGainController2ExperimentParams();
+
+ // When `experiment_params` is specified, returns an APM configuration
+ // modified according to the experiment parameters. Otherwise returns
+ // `config`.
+ static AudioProcessing::Config AdjustConfig(
+ const AudioProcessing::Config& config,
+ const absl::optional<GainController2ExperimentParams>& experiment_params);
+ static TransientSuppressor::VadMode GetTransientSuppressorVadMode(
+ const absl::optional<GainController2ExperimentParams>& experiment_params);
const bool use_denormal_disabler_;
- // When true, the transient suppressor submodule is never created regardless
- // of the APM configuration.
- // TODO(bugs.webrtc.org/13663): Remove when the linked field trial is removed.
- const bool disallow_transient_supporessor_usage_;
-
const TransientSuppressor::VadMode transient_suppressor_vad_mode_;
SwapQueue<RuntimeSetting> capture_runtime_settings_;
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 65bda71..e48a5d8 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -10,6 +10,7 @@
#include "modules/audio_processing/audio_processing_impl.h"
+#include <algorithm>
#include <array>
#include <memory>
#include <tuple>
@@ -131,32 +132,6 @@
static constexpr float ProcessSample(float x) { return 2.f * x; }
};
-// Creates a simple `AudioProcessing` instance for APM input volume testing
-// with AGC1 analog and/or AGC2 input volume controller enabled and AGC2
-// digital controller enabled.
-rtc::scoped_refptr<AudioProcessing> CreateApmForInputVolumeTest(
- bool agc1_analog_gain_controller_enabled,
- bool agc2_input_volume_controller_enabled) {
- webrtc::AudioProcessing::Config config;
- // Enable AGC1 analog controller.
- config.gain_controller1.enabled = agc1_analog_gain_controller_enabled;
- config.gain_controller1.analog_gain_controller.enabled =
- agc1_analog_gain_controller_enabled;
- // Enable AG2 input volume controller
- config.gain_controller2.input_volume_controller.enabled =
- agc2_input_volume_controller_enabled;
- // Enable AGC2 adaptive digital controller.
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- false;
- config.gain_controller2.enabled = true;
- config.gain_controller2.adaptive_digital.enabled = true;
-
- auto apm(AudioProcessingBuilder().Create());
- apm->ApplyConfig(config);
-
- return apm;
-}
-
// Runs `apm` input processing for volume adjustments for `num_frames` random
// frames starting from the volume `initial_volume`. This includes three steps:
// 1) Set the input volume 2) Process the stream 3) Set the new recommended
@@ -183,99 +158,6 @@
return recommended_input_volume;
}
-constexpr char kMinMicLevelFieldTrial[] =
- "WebRTC-Audio-2ndAgcMinMicLevelExperiment";
-constexpr char kMinInputVolumeFieldTrial[] = "WebRTC-Audio-Agc2-MinInputVolume";
-constexpr int kMinInputVolume = 12;
-
-std::string GetMinMicLevelExperimentFieldTrial(absl::optional<int> value) {
- char field_trial_buffer[128];
- rtc::SimpleStringBuilder builder(field_trial_buffer);
- if (value.has_value()) {
- RTC_DCHECK_GE(*value, 0);
- RTC_DCHECK_LE(*value, 255);
- builder << kMinMicLevelFieldTrial << "/Enabled-" << *value << "/";
- builder << kMinInputVolumeFieldTrial << "/Enabled-" << *value << "/";
- } else {
- builder << kMinMicLevelFieldTrial << "/Disabled/";
- builder << kMinInputVolumeFieldTrial << "/Disabled/";
- }
- return builder.str();
-}
-
-// TODO(webrtc:7494): Remove the fieldtrial from the input volume tests when
-// "WebRTC-Audio-2ndAgcMinMicLevelExperiment" and
-// "WebRTC-Audio-Agc2-MinInputVolume" are removed.
-class InputVolumeStartupParameterizedTest
- : public ::testing::TestWithParam<
- std::tuple<int, absl::optional<int>, bool, bool>> {
- protected:
- InputVolumeStartupParameterizedTest()
- : field_trials_(
- GetMinMicLevelExperimentFieldTrial(std::get<1>(GetParam()))) {}
- int GetStartupVolume() const { return std::get<0>(GetParam()); }
- int GetMinVolume() const {
- return std::get<1>(GetParam()).value_or(kMinInputVolume);
- }
- bool GetAgc1AnalogControllerEnabled() const {
- return std::get<2>(GetParam());
- }
- bool GetAgc2InputVolumeControllerEnabled() const {
- return std::get<3>(GetParam());
- }
-
- private:
- test::ScopedFieldTrials field_trials_;
-};
-
-class InputVolumeNotZeroParameterizedTest
- : public ::testing::TestWithParam<
- std::tuple<int, int, absl::optional<int>, bool, bool>> {
- protected:
- InputVolumeNotZeroParameterizedTest()
- : field_trials_(
- GetMinMicLevelExperimentFieldTrial(std::get<2>(GetParam()))) {}
- int GetStartupVolume() const { return std::get<0>(GetParam()); }
- int GetVolume() const { return std::get<1>(GetParam()); }
- int GetMinVolume() const {
- return std::get<2>(GetParam()).value_or(kMinInputVolume);
- }
- bool GetMinMicLevelExperimentEnabled() {
- return std::get<2>(GetParam()).has_value();
- }
- bool GetAgc1AnalogControllerEnabled() const {
- return std::get<3>(GetParam());
- }
- bool GetAgc2InputVolumeControllerEnabled() const {
- return std::get<4>(GetParam());
- }
-
- private:
- test::ScopedFieldTrials field_trials_;
-};
-
-class InputVolumeZeroParameterizedTest
- : public ::testing::TestWithParam<
- std::tuple<int, absl::optional<int>, bool, bool>> {
- protected:
- InputVolumeZeroParameterizedTest()
- : field_trials_(
- GetMinMicLevelExperimentFieldTrial(std::get<1>(GetParam()))) {}
- int GetStartupVolume() const { return std::get<0>(GetParam()); }
- int GetMinVolume() const {
- return std::get<1>(GetParam()).value_or(kMinInputVolume);
- }
- bool GetAgc1AnalogControllerEnabled() const {
- return std::get<2>(GetParam());
- }
- bool GetAgc2InputVolumeControllerEnabled() const {
- return std::get<3>(GetParam());
- }
-
- private:
- test::ScopedFieldTrials field_trials_;
-};
-
} // namespace
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
@@ -644,8 +526,10 @@
TEST(AudioProcessingImplTest,
ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/");
- rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
+ "WebRTC-Audio-GainController2/Disabled/");
+ auto apm = AudioProcessingBuilder()
+ .SetConfig({.gain_controller1{.enabled = false}})
+ .Create();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
@@ -675,7 +559,7 @@
TEST(AudioProcessingImplTest,
ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/");
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
webrtc::AudioProcessing::Config apm_config;
@@ -958,224 +842,116 @@
}
}
-TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) {
- // Do not explicitly disable "WebRTC-ApmTransientSuppressorKillSwitch" since
- // to check that, by default, it is disabled.
- auto apm = AudioProcessingBuilder()
- .SetConfig({.transient_suppression = {.enabled = false}})
- .Create();
- EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
-}
-
-TEST(AudioProcessingImplTest, CanEnableTransientSuppressor) {
- // Do not explicitly disable "WebRTC-ApmTransientSuppressorKillSwitch" since
- // to check that, by default, it is disabled.
- auto apm = AudioProcessingBuilder()
- .SetConfig({.transient_suppression = {.enabled = true}})
- .Create();
- EXPECT_TRUE(apm->GetConfig().transient_suppression.enabled);
-}
-
-TEST(AudioProcessingImplTest, CanDisableTransientSuppressorIfUsageAllowed) {
- // Disable the field trial that disallows to enable transient suppression.
- test::ScopedFieldTrials field_trials(
- "WebRTC-ApmTransientSuppressorKillSwitch/Disabled/");
- auto apm = AudioProcessingBuilder()
- .SetConfig({.transient_suppression = {.enabled = false}})
- .Create();
- EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
-}
-
-TEST(AudioProcessingImplTest, CanEnableTransientSuppressorIfUsageAllowed) {
- // Disable the field trial that disallows to enable transient suppression.
- test::ScopedFieldTrials field_trials(
- "WebRTC-ApmTransientSuppressorKillSwitch/Disabled/");
- auto apm = AudioProcessingBuilder()
- .SetConfig({.transient_suppression = {.enabled = true}})
- .Create();
- EXPECT_TRUE(apm->GetConfig().transient_suppression.enabled);
-}
-
-TEST(AudioProcessingImplTest,
- CannotEnableTransientSuppressorIfUsageDisallowed) {
- // Enable the field trial that disallows to enable transient suppression.
- test::ScopedFieldTrials field_trials(
- "WebRTC-ApmTransientSuppressorKillSwitch/Enabled/");
- auto apm = AudioProcessingBuilder()
- .SetConfig({.transient_suppression = {.enabled = true}})
- .Create();
- EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
-}
-
-// TODO(bugs.webrtc.org/7494): Test AGCs with different multi-channel configs.
-
-// Tests that the minimum startup volume is applied at the startup.
-TEST_P(InputVolumeStartupParameterizedTest,
- VerifyStartupMinVolumeAppliedAtStartup) {
- const int applied_startup_input_volume = GetStartupVolume();
- const int expected_volume =
- std::max(applied_startup_input_volume, GetMinVolume());
- const bool agc1_analog_controller_enabled = GetAgc1AnalogControllerEnabled();
- const bool agc2_input_volume_controller_enabled =
- GetAgc2InputVolumeControllerEnabled();
- auto apm = CreateApmForInputVolumeTest(agc1_analog_controller_enabled,
- agc2_input_volume_controller_enabled);
-
- const int recommended_input_volume =
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_startup_input_volume);
-
- if (!agc1_analog_controller_enabled &&
- !agc2_input_volume_controller_enabled) {
- // No input volume changes if none of the analog controllers is enabled.
- ASSERT_EQ(recommended_input_volume, applied_startup_input_volume);
- } else {
- ASSERT_EQ(recommended_input_volume, expected_volume);
- }
-}
-
-// Tests that the minimum input volume is applied if the volume is manually
-// adjusted to a non-zero value 1) always for AGC2 input volume controller and
-// 2) only if "WebRTC-Audio-2ndAgcMinMicLevelExperiment" is enabled for AGC1
-// analog controller.
-TEST_P(InputVolumeNotZeroParameterizedTest,
- VerifyMinVolumeMaybeAppliedAfterManualVolumeAdjustments) {
- const int applied_startup_input_volume = GetStartupVolume();
- const int applied_input_volume = GetVolume();
- const int expected_volume = std::max(applied_input_volume, GetMinVolume());
- const bool agc1_analog_controller_enabled = GetAgc1AnalogControllerEnabled();
- const bool agc2_input_volume_controller_enabled =
- GetAgc2InputVolumeControllerEnabled();
- auto apm = CreateApmForInputVolumeTest(agc1_analog_controller_enabled,
- agc2_input_volume_controller_enabled);
-
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_startup_input_volume);
- const int recommended_input_volume =
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_input_volume);
-
- ASSERT_NE(applied_input_volume, 0);
-
- if (!agc1_analog_controller_enabled &&
- !agc2_input_volume_controller_enabled) {
- // No input volume changes if none of the analog controllers is enabled.
- ASSERT_EQ(recommended_input_volume, applied_input_volume);
- } else {
- if (GetMinMicLevelExperimentEnabled() ||
- (!agc1_analog_controller_enabled &&
- agc2_input_volume_controller_enabled)) {
- ASSERT_EQ(recommended_input_volume, expected_volume);
- } else {
- ASSERT_EQ(recommended_input_volume, applied_input_volume);
+class ApmInputVolumeControllerParametrizedTest
+ : public ::testing::TestWithParam<
+ std::tuple<int, int, AudioProcessing::Config>> {
+ protected:
+ ApmInputVolumeControllerParametrizedTest()
+ : sample_rate_hz_(std::get<0>(GetParam())),
+ num_channels_(std::get<1>(GetParam())),
+ channels_(num_channels_),
+ channel_pointers_(num_channels_) {
+ const int frame_size = sample_rate_hz_ / 100;
+ for (int c = 0; c < num_channels_; ++c) {
+ channels_[c].resize(frame_size);
+ channel_pointers_[c] = channels_[c].data();
+ std::fill(channels_[c].begin(), channels_[c].end(), 0.0f);
}
}
+
+ int sample_rate_hz() const { return sample_rate_hz_; }
+ int num_channels() const { return num_channels_; }
+ AudioProcessing::Config GetConfig() const { return std::get<2>(GetParam()); }
+
+ float* const* channel_pointers() { return channel_pointers_.data(); }
+
+ private:
+ const int sample_rate_hz_;
+ const int num_channels_;
+ std::vector<std::vector<float>> channels_;
+ std::vector<float*> channel_pointers_;
+};
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+ EnforceMinInputVolumeAtStartupWithZeroVolume) {
+ const StreamConfig stream_config(sample_rate_hz(), num_channels());
+ auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+ apm->set_stream_analog_level(0);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ EXPECT_GT(apm->recommended_stream_analog_level(), 0);
}
-// Tests that the minimum input volume is not applied if the volume is manually
-// adjusted to zero.
-TEST_P(InputVolumeZeroParameterizedTest,
- VerifyMinVolumeNotAppliedAfterManualVolumeAdjustments) {
- constexpr int kZeroVolume = 0;
- const int applied_startup_input_volume = GetStartupVolume();
- const bool agc1_analog_controller_enabled = GetAgc1AnalogControllerEnabled();
- const bool agc2_input_volume_controller_enabled =
- GetAgc2InputVolumeControllerEnabled();
- auto apm = CreateApmForInputVolumeTest(agc1_analog_controller_enabled,
- agc2_input_volume_controller_enabled);
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+ EnforceMinInputVolumeAtStartupWithNonZeroVolume) {
+ const StreamConfig stream_config(sample_rate_hz(), num_channels());
+ auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
- const int recommended_input_volume_after_startup =
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_startup_input_volume);
- const int recommended_input_volume =
- ProcessInputVolume(*apm, /*num_frames=*/1, kZeroVolume);
+ constexpr int kStartupVolume = 3;
+ apm->set_stream_analog_level(kStartupVolume);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ EXPECT_GT(apm->recommended_stream_analog_level(), kStartupVolume);
+}
- if (!agc1_analog_controller_enabled &&
- !agc2_input_volume_controller_enabled) {
- // No input volume changes if none of the analog controllers is enabled.
- ASSERT_EQ(recommended_input_volume, kZeroVolume);
- } else {
- ASSERT_NE(recommended_input_volume, recommended_input_volume_after_startup);
- ASSERT_EQ(recommended_input_volume, kZeroVolume);
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+ EnforceMinInputVolumeAfterManualVolumeAdjustment) {
+ const auto config = GetConfig();
+ if (config.gain_controller1.enabled) {
+ // After a downward manual adjustment, AGC1 slowly converges to the minimum
+ // input volume.
+ GTEST_SKIP() << "Does not apply to AGC1";
}
+ const StreamConfig stream_config(sample_rate_hz(), num_channels());
+ auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+ apm->set_stream_analog_level(20);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ constexpr int kManuallyAdjustedVolume = 3;
+ apm->set_stream_analog_level(kManuallyAdjustedVolume);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ EXPECT_GT(apm->recommended_stream_analog_level(), kManuallyAdjustedVolume);
}
-// Tests that the minimum input volume is applied if the volume is not zero
-// before it is automatically adjusted.
-TEST_P(InputVolumeNotZeroParameterizedTest,
- VerifyMinVolumeAppliedAfterAutomaticVolumeAdjustments) {
- const int applied_startup_input_volume = GetStartupVolume();
- const int applied_input_volume = GetVolume();
- const bool agc1_analog_controller_enabled = GetAgc1AnalogControllerEnabled();
- const bool agc2_input_volume_controller_enabled =
- GetAgc2InputVolumeControllerEnabled();
- auto apm = CreateApmForInputVolumeTest(agc1_analog_controller_enabled,
- agc2_input_volume_controller_enabled);
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+ DoNotEnforceMinInputVolumeAfterManualVolumeAdjustmentToZero) {
+ const StreamConfig stream_config(sample_rate_hz(), num_channels());
+ auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_startup_input_volume);
- const int recommended_input_volume =
- ProcessInputVolume(*apm, /*num_frames=*/400, applied_input_volume);
-
- ASSERT_NE(applied_input_volume, 0);
-
- if (!agc1_analog_controller_enabled &&
- !agc2_input_volume_controller_enabled) {
- // No input volume changes if none of the analog controllers is enabled.
- ASSERT_EQ(recommended_input_volume, applied_input_volume);
- } else {
- if (recommended_input_volume != applied_input_volume) {
- ASSERT_GE(recommended_input_volume, GetMinVolume());
- }
- }
+ apm->set_stream_analog_level(100);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ apm->set_stream_analog_level(0);
+ apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+ channel_pointers());
+ EXPECT_EQ(apm->recommended_stream_analog_level(), 0);
}
-// Tests that the minimum input volume is not applied if the volume is zero
-// before it is automatically adjusted.
-TEST_P(InputVolumeZeroParameterizedTest,
- VerifyMinVolumeNotAppliedAfterAutomaticVolumeAdjustments) {
- constexpr int kZeroVolume = 0;
- const int applied_startup_input_volume = GetStartupVolume();
- const bool agc1_analog_controller_enabled = GetAgc1AnalogControllerEnabled();
- const bool agc2_input_volume_controller_enabled =
- GetAgc2InputVolumeControllerEnabled();
- auto apm = CreateApmForInputVolumeTest(agc1_analog_controller_enabled,
- agc2_input_volume_controller_enabled);
-
- const int recommended_input_volume_after_startup =
- ProcessInputVolume(*apm, /*num_frames=*/1, applied_startup_input_volume);
- const int recommended_input_volume =
- ProcessInputVolume(*apm, /*num_frames=*/400, kZeroVolume);
-
- if (!agc1_analog_controller_enabled &&
- !agc2_input_volume_controller_enabled) {
- // No input volume changes if none of the analog controllers is enabled.
- ASSERT_EQ(recommended_input_volume, kZeroVolume);
- } else {
- ASSERT_NE(recommended_input_volume, recommended_input_volume_after_startup);
- ASSERT_EQ(recommended_input_volume, kZeroVolume);
- }
-}
-
-INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest,
- InputVolumeStartupParameterizedTest,
- ::testing::Combine(::testing::Values(0, 5, 30),
- ::testing::Values(absl::nullopt,
- 20),
- ::testing::Bool(),
- ::testing::Bool()));
-
-INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest,
- InputVolumeNotZeroParameterizedTest,
- ::testing::Combine(::testing::Values(0, 5, 15),
- ::testing::Values(1, 5, 30),
- ::testing::Values(absl::nullopt,
- 20),
- ::testing::Bool(),
- ::testing::Bool()));
-
-INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest,
- InputVolumeZeroParameterizedTest,
- ::testing::Combine(::testing::Values(0, 5, 15),
- ::testing::Values(absl::nullopt,
- 20),
- ::testing::Bool(),
- ::testing::Bool()));
+INSTANTIATE_TEST_SUITE_P(
+ AudioProcessingImplTest,
+ ApmInputVolumeControllerParametrizedTest,
+ ::testing::Combine(
+ ::testing::Values(8000, 16000, 32000, 48000), // Sample rates.
+ ::testing::Values(1, 2), // Number of channels.
+ ::testing::Values(
+ // Full AGC1.
+ AudioProcessing::Config{
+ .gain_controller1 = {.enabled = true,
+ .analog_gain_controller =
+ {.enabled = true,
+ .enable_digital_adaptive = true}},
+ .gain_controller2 = {.enabled = false}},
+ // Hybrid AGC.
+ AudioProcessing::Config{
+ .gain_controller1 = {.enabled = true,
+ .analog_gain_controller =
+ {.enabled = true,
+ .enable_digital_adaptive = false}},
+ .gain_controller2 = {.enabled = true,
+ .adaptive_digital = {.enabled = true}}})));
// When the input volume is not emulated and no input volume controller is
// active, the recommended volume must always be the applied volume.
@@ -1237,53 +1013,336 @@
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
-class GainController2FieldTrialParametrizedTest
+TEST(AudioProcessingImplTest,
+ Agc2FieldTrialDoNotSwitchToFullAgc2WhenNoAgcIsActive) {
+ constexpr AudioProcessing::Config kOriginal{
+ .gain_controller1{.enabled = false},
+ .gain_controller2{.enabled = false},
+ };
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+}
+
+TEST(AudioProcessingImplTest,
+ Agc2FieldTrialDoNotSwitchToFullAgc2WithAgc1Agc2InputVolumeControllers) {
+ constexpr AudioProcessing::Config kOriginal{
+ .gain_controller1{.enabled = true,
+ .analog_gain_controller{.enabled = true}},
+ .gain_controller2{.enabled = true,
+ .input_volume_controller{.enabled = true}},
+ };
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+}
+
+class Agc2FieldTrialParametrizedTest
: public ::testing::TestWithParam<AudioProcessing::Config> {};
-TEST_P(GainController2FieldTrialParametrizedTest,
- CheckAgc2AdaptiveDigitalOverridesApplied) {
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfDisabled) {
+ const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-GainController2/"
- "Enabled,"
- "enable_clipping_predictor:true,"
+ "WebRTC-Audio-GainController2/Disabled/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfNoOverride) {
+ const AudioProcessing::Config original = GetParam();
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,"
+ "switch_to_agc2:false,"
+ "disallow_transient_suppressor_usage:false/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotSwitchToFullAgc2) {
+ const AudioProcessing::Config original = GetParam();
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:false/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+ EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, SwitchToFullAgc2) {
+ const AudioProcessing::Config original = GetParam();
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+ SwitchToFullAgc2AndOverrideInputVolumeControllerParameters) {
+ const AudioProcessing::Config original = GetParam();
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
+ "min_input_volume:123,"
"clipped_level_min:20,"
"clipped_level_step:30,"
"clipped_ratio_threshold:0.4,"
"clipped_wait_frames:50,"
+ "enable_clipping_predictor:true,"
"target_range_max_dbfs:-6,"
"target_range_min_dbfs:-70,"
"update_input_volume_wait_frames:80,"
"speech_probability_threshold:0.9,"
- "speech_ratio_threshold:1.0,"
+ "speech_ratio_threshold:1.0/");
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+ SwitchToFullAgc2AndOverrideAdaptiveDigitalControllerParameters) {
+ const AudioProcessing::Config original = GetParam();
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
"headroom_db:10,"
"max_gain_db:20,"
"initial_gain_db:7,"
"max_gain_change_db_per_second:5,"
"max_output_noise_level_dbfs:-40/");
- auto adjusted_config =
- AudioProcessingBuilder().SetConfig(GetParam()).Create()->GetConfig();
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+ ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
+ original.gain_controller2.adaptive_digital);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
+ EXPECT_EQ(
+ adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
+ 5);
+ EXPECT_EQ(
+ adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
+ -40);
- EXPECT_FALSE(adjusted_config.gain_controller1.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.input_volume_controller.enabled);
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(original);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(adjusted.gain_controller1.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+ EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+ ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
+ original.gain_controller2.adaptive_digital);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
+ EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
+ EXPECT_EQ(
+ adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
+ 5);
+ EXPECT_EQ(
+ adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
+ -40);
+}
- EXPECT_EQ(adjusted_config.gain_controller2.adaptive_digital.headroom_db, 10);
- EXPECT_EQ(adjusted_config.gain_controller2.adaptive_digital.max_gain_db, 20);
- EXPECT_EQ(adjusted_config.gain_controller2.adaptive_digital.initial_gain_db,
- 7);
- EXPECT_EQ(adjusted_config.gain_controller2.adaptive_digital
- .max_gain_change_db_per_second,
- 5);
- EXPECT_EQ(adjusted_config.gain_controller2.adaptive_digital
- .max_output_noise_level_dbfs,
- -40);
+TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithTs) {
+ AudioProcessing::Config config = GetParam();
+ config.transient_suppression.enabled = true;
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Disabled/");
+ auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+ constexpr int kSampleRateHz = 48000;
+ constexpr int kNumChannels = 1;
+ std::array<float, kSampleRateHz / 100> buffer;
+ float* channel_pointers[] = {buffer.data()};
+ StreamConfig stream_config(kSampleRateHz, kNumChannels);
+ Random random_generator(2341U);
+ constexpr int kFramesToProcess = 10;
+ int volume = 100;
+ for (int i = 0; i < kFramesToProcess; ++i) {
+ SCOPED_TRACE(i);
+ RandomizeSampleVector(&random_generator, buffer);
+ apm->set_stream_analog_level(volume);
+ ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+ channel_pointers),
+ kNoErr);
+ volume = apm->recommended_stream_analog_level();
+ }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithoutTs) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,"
+ "switch_to_agc2:false,"
+ "disallow_transient_suppressor_usage:true/");
+ auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
+
+ constexpr int kSampleRateHz = 48000;
+ constexpr int kNumChannels = 1;
+ std::array<float, kSampleRateHz / 100> buffer;
+ float* channel_pointers[] = {buffer.data()};
+ StreamConfig stream_config(kSampleRateHz, kNumChannels);
+ Random random_generator(2341U);
+ constexpr int kFramesToProcess = 10;
+ int volume = 100;
+ for (int i = 0; i < kFramesToProcess; ++i) {
+ SCOPED_TRACE(i);
+ RandomizeSampleVector(&random_generator, buffer);
+ apm->set_stream_analog_level(volume);
+ ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+ channel_pointers),
+ kNoErr);
+ volume = apm->recommended_stream_analog_level();
+ }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+ ProcessSucceedsWhenSwitchToFullAgc2WithTs) {
+ AudioProcessing::Config config = GetParam();
+ config.transient_suppression.enabled = true;
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,"
+ "switch_to_agc2:true,"
+ "disallow_transient_suppressor_usage:false/");
+ auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+ constexpr int kSampleRateHz = 48000;
+ constexpr int kNumChannels = 1;
+ std::array<float, kSampleRateHz / 100> buffer;
+ float* channel_pointers[] = {buffer.data()};
+ StreamConfig stream_config(kSampleRateHz, kNumChannels);
+ Random random_generator(2341U);
+ constexpr int kFramesToProcess = 10;
+ int volume = 100;
+ for (int i = 0; i < kFramesToProcess; ++i) {
+ SCOPED_TRACE(i);
+ RandomizeSampleVector(&random_generator, buffer);
+ apm->set_stream_analog_level(volume);
+ ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+ channel_pointers),
+ kNoErr);
+ volume = apm->recommended_stream_analog_level();
+ }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+ ProcessSucceedsWhenSwitchToFullAgc2WithoutTs) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,"
+ "switch_to_agc2:true,"
+ "disallow_transient_suppressor_usage:true/");
+ auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
+
+ constexpr int kSampleRateHz = 48000;
+ constexpr int kNumChannels = 1;
+ std::array<float, kSampleRateHz / 100> buffer;
+ float* channel_pointers[] = {buffer.data()};
+ StreamConfig stream_config(kSampleRateHz, kNumChannels);
+ Random random_generator(2341U);
+ constexpr int kFramesToProcess = 10;
+ int volume = 100;
+ for (int i = 0; i < kFramesToProcess; ++i) {
+ SCOPED_TRACE(i);
+ RandomizeSampleVector(&random_generator, buffer);
+ apm->set_stream_analog_level(volume);
+ ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+ channel_pointers),
+ kNoErr);
+ volume = apm->recommended_stream_analog_level();
+ }
}
INSTANTIATE_TEST_SUITE_P(
AudioProcessingImplTest,
- GainController2FieldTrialParametrizedTest,
+ Agc2FieldTrialParametrizedTest,
::testing::Values(
// Full AGC1.
AudioProcessing::Config{
@@ -1301,314 +1360,132 @@
.gain_controller2 = {.enabled = true,
.adaptive_digital = {.enabled = true}}}));
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigAdjustedWhenExperimentEnabledAndAgc1AnalogEnabled) {
- constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
- kDefaultAdaptiveDigitalConfig;
+TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = false}};
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.transient_suppression.enabled);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest, CanEnableTs) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = true}};
+
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
+
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest, CanDisableTsWithAgc2FieldTrialDisabled) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = false}};
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-GainController2/"
- "Enabled,"
- "enable_clipping_predictor:true,"
- "clipped_level_min:20,"
- "clipped_level_step:30,"
- "clipped_ratio_threshold:0.4,"
- "clipped_wait_frames:50,"
- "target_range_max_dbfs:-6,"
- "target_range_min_dbfs:-70,"
- "update_input_volume_wait_frames:80,"
- "speech_probability_threshold:0.9,"
- "speech_ratio_threshold:1.0,"
- "headroom_db:10,"
- "max_gain_db:20,"
- "initial_gain_db:7,"
- "max_gain_change_db_per_second:3,"
- "max_output_noise_level_dbfs:-40/");
+ "WebRTC-Audio-GainController2/Disabled/");
- AudioProcessingBuilderForTesting apm_builder;
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.transient_suppression.enabled);
- // Set a config with analog AGC1 enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = true;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
- config.gain_controller2.enabled = false;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- // Expect the config to be adjusted.
- EXPECT_FALSE(adjusted_config.gain_controller1.enabled);
- EXPECT_FALSE(adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.input_volume_controller.enabled);
- EXPECT_NE(adjusted_config.gain_controller2.adaptive_digital,
- kDefaultAdaptiveDigitalConfig);
-
- // Change config back and compare.
- adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
- adjusted_config.gain_controller1.analog_gain_controller.enabled =
- config.gain_controller1.analog_gain_controller.enabled;
- adjusted_config.gain_controller2.enabled = config.gain_controller2.enabled;
- adjusted_config.gain_controller2.adaptive_digital.enabled =
- config.gain_controller2.adaptive_digital.enabled;
- adjusted_config.gain_controller2.input_volume_controller.enabled =
- config.gain_controller2.input_volume_controller.enabled;
- adjusted_config.gain_controller2.adaptive_digital =
- config.gain_controller2.adaptive_digital;
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
}
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigAdjustedWhenExperimentEnabledAndHybridAgcEnabled) {
- constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
- kDefaultAdaptiveDigitalConfig;
+TEST(AudioProcessingImplTest, CanEnableTsWithAgc2FieldTrialDisabled) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = true}};
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-GainController2/"
- "Enabled,"
- "enable_clipping_predictor:true,"
- "clipped_level_min:20,"
- "clipped_level_step:30,"
- "clipped_ratio_threshold:0.4,"
- "clipped_wait_frames:50,"
- "target_range_max_dbfs:-6,"
- "target_range_min_dbfs:-70,"
- "update_input_volume_wait_frames:80,"
- "speech_probability_threshold:0.9,"
- "speech_ratio_threshold:1.0,"
- "headroom_db:10,"
- "max_gain_db:20,"
- "initial_gain_db:7,"
- "max_gain_change_db_per_second:3,"
- "max_output_noise_level_dbfs:-40/");
+ "WebRTC-Audio-GainController2/Disabled/");
- AudioProcessingBuilderForTesting apm_builder;
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
- // Set a config with hybrid AGC enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = true;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- false;
- config.gain_controller2.enabled = true;
- config.gain_controller2.adaptive_digital.enabled = true;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- // Expect the config to be adjusted.
- EXPECT_FALSE(adjusted_config.gain_controller1.enabled);
- EXPECT_FALSE(adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_TRUE(adjusted_config.gain_controller2.input_volume_controller.enabled);
- EXPECT_NE(adjusted_config.gain_controller2.adaptive_digital,
- kDefaultAdaptiveDigitalConfig);
-
- // Change config back and compare.
- adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
- adjusted_config.gain_controller1.analog_gain_controller.enabled =
- config.gain_controller1.analog_gain_controller.enabled;
- adjusted_config.gain_controller2.enabled = config.gain_controller2.enabled;
- adjusted_config.gain_controller2.adaptive_digital.enabled =
- config.gain_controller2.adaptive_digital.enabled;
- adjusted_config.gain_controller2.input_volume_controller.enabled =
- config.gain_controller2.input_volume_controller.enabled;
- adjusted_config.gain_controller2.adaptive_digital =
- config.gain_controller2.adaptive_digital;
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
}
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigNotAdjustedWhenExperimentEnabledAndAgc1AnalogNotEnabled) {
+TEST(AudioProcessingImplTest,
+ CanDisableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = false}};
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-GainController2/"
- "Enabled,"
- "enable_clipping_predictor:true,"
- "clipped_level_min:20,"
- "clipped_level_step:30,"
- "clipped_ratio_threshold:0.4,"
- "clipped_wait_frames:50,"
- "target_range_max_dbfs:-6,"
- "target_range_min_dbfs:-70,"
- "update_input_volume_wait_frames:80,"
- "speech_probability_threshold:0.9,"
- "speech_ratio_threshold:1.0,"
- "headroom_db:10,"
- "max_gain_db:20,"
- "initial_gain_db:7,"
- "max_gain_change_db_per_second:3,"
- "max_output_noise_level_dbfs:-40/");
+ "WebRTC-Audio-GainController2/Enabled,"
+ "disallow_transient_suppressor_usage:false/");
- AudioProcessingBuilderForTesting apm_builder;
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.transient_suppression.enabled);
- // Set a config with analog AGC1 not enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = false;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
- config.gain_controller2.enabled = false;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- EXPECT_EQ(config.gain_controller1.enabled,
- adjusted_config.gain_controller1.enabled);
- EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
- adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_EQ(config.gain_controller2.enabled,
- adjusted_config.gain_controller2.enabled);
- EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
- adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_FALSE(
- adjusted_config.gain_controller2.input_volume_controller.enabled);
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(adjusted.transient_suppression.enabled);
}
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigNotAdjustedWhenExperimentEnabledAndHybridAgcNotEnabled) {
+TEST(AudioProcessingImplTest,
+ CanEnableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = true}};
webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-GainController2/"
- "Enabled,"
- "enable_clipping_predictor:true,"
- "clipped_level_min:20,"
- "clipped_level_step:30,"
- "clipped_ratio_threshold:0.4,"
- "clipped_wait_frames:50,"
- "target_range_max_dbfs:-6,"
- "target_range_min_dbfs:-70,"
- "update_input_volume_wait_frames:80,"
- "speech_probability_threshold:0.9,"
- "speech_ratio_threshold:1.0,"
- "headroom_db:10,"
- "max_gain_db:20,"
- "initial_gain_db:7,"
- "max_gain_change_db_per_second:3,"
- "max_output_noise_level_dbfs:-40/");
+ "WebRTC-Audio-GainController2/Enabled,"
+ "disallow_transient_suppressor_usage:false/");
- AudioProcessingBuilderForTesting apm_builder;
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
- // Set a config with hybrid AGC analog not enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = false;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- false;
- config.gain_controller2.enabled = true;
- config.gain_controller2.adaptive_digital.enabled = true;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- EXPECT_EQ(config.gain_controller1.enabled,
- adjusted_config.gain_controller1.enabled);
- EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
- adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_EQ(config.gain_controller2.enabled,
- adjusted_config.gain_controller2.enabled);
- EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
- adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_FALSE(
- adjusted_config.gain_controller2.input_volume_controller.enabled);
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_TRUE(adjusted.transient_suppression.enabled);
}
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigNotAdjustedWhenExperimentNotEnabledAndAgc1AnalogEnabled) {
- AudioProcessingBuilderForTesting apm_builder;
+TEST(AudioProcessingImplTest,
+ CannotEnableTsWithAgc2FieldTrialEnabledAndUsageDisallowed) {
+ constexpr AudioProcessing::Config kOriginal = {
+ .transient_suppression = {.enabled = true}};
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-GainController2/Enabled,"
+ "disallow_transient_suppressor_usage:true/");
- // Set a config with analog AGC1 analog enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = true;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
- config.gain_controller2.enabled = false;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ // Test config application via `AudioProcessing` ctor.
+ auto adjusted =
+ AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+ EXPECT_FALSE(adjusted.transient_suppression.enabled);
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- EXPECT_EQ(config.gain_controller1.enabled,
- adjusted_config.gain_controller1.enabled);
- EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
- adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_EQ(config.gain_controller2.enabled,
- adjusted_config.gain_controller2.enabled);
- EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
- adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_FALSE(
- adjusted_config.gain_controller2.input_volume_controller.enabled);
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
-}
-
-TEST(AudioProcessingImplGainController2FieldTrialTest,
- ConfigNotAdjustedWhenExperimentNotEnabledAndHybridAgcEnabled) {
- AudioProcessingBuilderForTesting apm_builder;
-
- // Set a config with hybrid AGC enabled.
- AudioProcessing::Config config;
- config.gain_controller1.enabled = true;
- config.gain_controller1.analog_gain_controller.enabled = true;
- config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
- false;
- config.gain_controller2.enabled = true;
- config.gain_controller2.adaptive_digital.enabled = true;
- config.gain_controller1.mode =
- AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-
- EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
-
- apm_builder.SetConfig(config);
-
- auto apm = apm_builder.Create();
- auto adjusted_config = apm->GetConfig();
-
- EXPECT_EQ(config.gain_controller1.enabled,
- adjusted_config.gain_controller1.enabled);
- EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
- adjusted_config.gain_controller1.analog_gain_controller.enabled);
- EXPECT_EQ(config.gain_controller2.enabled,
- adjusted_config.gain_controller2.enabled);
- EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
- adjusted_config.gain_controller2.adaptive_digital.enabled);
- EXPECT_FALSE(
- adjusted_config.gain_controller2.input_volume_controller.enabled);
-
- EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+ // Test config application via `AudioProcessing::ApplyConfig()`.
+ auto apm = AudioProcessingBuilder().Create();
+ apm->ApplyConfig(kOriginal);
+ adjusted = apm->GetConfig();
+ EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
}
} // namespace webrtc