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Fix AudioProcessingImplTest tests on analog gain changes

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`EchoControllerObservesAnalogAgc1EchoPathGainChange` is incorrect
since it does not call `set_stream_analog_level()`,
`ProcessCapture()` and `recommended_stream_analog_level()`
according to the contract.

`EchoControllerObservesNoDigitalAgc2EchoPathGainChange` is
useless since AGC2 doesn't have any analog controller at the
moment and the test is not written to explictly trigger digital
gain adaptations.

Bug: webrtc:7494, b/241923537
Change-Id: I56203c736448ec060077b00b57e98cd4c29fa737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38052}
This commit is contained in:
Alessio Bazzica
2022-09-06 16:39:07 +02:00
committed by WebRTC LUCI CQ
parent 3153b363cd
commit a5aaedb327
1 changed files with 45 additions and 43 deletions
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88
modules/audio_processing/audio_processing_impl_unittest.cc
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@ -450,8 +450,7 @@ TEST(AudioProcessingImplTest,
webrtc::AudioProcessing::Config apm_config;
// Enable AGC1.
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveAnalog;
apm_config.gain_controller1.analog_gain_controller.enabled = true;
apm_config.gain_controller2.enabled = false;
apm_config.pre_amplifier.enabled = false;
apm->ApplyConfig(apm_config);
@ -465,34 +464,43 @@ TEST(AudioProcessingImplTest,
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
const int initial_analog_gain = apm->recommended_stream_analog_level();
constexpr int kInitialStreamAnalogLevel = 123;
apm->set_stream_analog_level(kInitialStreamAnalogLevel);
// When the first fame is processed, no echo path gain change must be
// detected.
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, false))
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
// Force an analog gain change if it did not happen.
if (initial_analog_gain == apm->recommended_stream_analog_level()) {
apm->set_stream_analog_level(initial_analog_gain + 1);
// Simulate the application of the recommended analog level.
int recommended_analog_level = apm->recommended_stream_analog_level();
if (recommended_analog_level == kInitialStreamAnalogLevel) {
// Force an analog gain change if it did not happen.
recommended_analog_level++;
}
apm->set_stream_analog_level(recommended_analog_level);
// After the first fame and with a stream analog level change, the echo path
// gain change must be detected.
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, true))
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
}
// Tests that a stream is successfully processed when AGC2 adaptive digital is
// used and when the field trial
// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/` is set.
TEST(AudioProcessingImplTest,
EchoControllerObservesNoDigitalAgc2EchoPathGainChange) {
// Tests that the echo controller doesn't observe an echo path gain change
// when the AGC2 digital submodule changes the digital gain without analog
// gain changes.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const auto* echo_control_factory_ptr = echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/");
rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
webrtc::AudioProcessing::Config apm_config;
// Disable AGC1 analog.
apm_config.gain_controller1.enabled = false;
@ -500,33 +508,27 @@ TEST(AudioProcessingImplTest,
apm_config.gain_controller2.enabled = true;
apm_config.gain_controller2.adaptive_digital.enabled = true;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 1000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig stream_config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_,
/*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_,
/*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
/*num_channels=*/kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
for (int i = 0; i < kFramesToProcess; ++i) {
RandomizeSampleVector(&random_generator, buffer);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
}
TEST(AudioProcessingImplTest, ProcessWithAgc2InjectedSpeechProbability) {
// Tests that a stream is successfully processed for the field trial
// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` using
// injected speech probability in AGC2 digital.
// Tests that a stream is successfully processed when AGC2 adaptive digital is
// used and when the field trial
// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` is set.
TEST(AudioProcessingImplTest,
ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/");
rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();