Change unittests to use AEC3 instead of AEC2
This CL changes the APM unittests to use AEC3 instead of
AEC2.
Bug: webrtc:8671
Change-Id: I80f88dbafb7c31696abd8b7efb5a187a9fb30d1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129420
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27607}diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 7ff37cb..ec2c58b 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -110,10 +110,6 @@
return 0;
}
-int TruncateToMultipleOf10(int value) {
- return (value / 10) * 10;
-}
-
void MixStereoToMono(const float* stereo, float* mono,
size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; ++i)
@@ -188,10 +184,7 @@
apm_config.gain_controller1.mode =
AudioProcessing::Config::GainController1::kAdaptiveDigital;
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- // TODO(peah): Update tests to instead use AEC3.
- apm_config.echo_canceller.use_legacy_aec = true;
apm_config.echo_canceller.mobile_mode = false;
- apm_config.echo_canceller.legacy_moderate_suppression_level = true;
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode =
@@ -421,16 +414,9 @@
ref_filename_(test::ResourcePath("audio_processing/output_data_fixed",
"pb")),
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
-#if defined(WEBRTC_MAC)
- // A different file for Mac is needed because on this platform the AEC
- // constant |kFixedDelayMs| value is 20 and not 50 as it is on the rest.
- ref_filename_(test::ResourcePath("audio_processing/output_data_mac",
- "pb")),
-#else
ref_filename_(test::ResourcePath("audio_processing/output_data_float",
"pb")),
#endif
-#endif
frame_(NULL),
revframe_(NULL),
output_sample_rate_hz_(0),
@@ -870,83 +856,6 @@
}
}
-TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
- // TODO(bjornv): Fix this test to work with DA-AEC.
- // Enable AEC only.
- AudioProcessing::Config apm_config = apm_->GetConfig();
- apm_config.echo_canceller.enabled = true;
- // TODO(peah): Update tests to instead use AEC3.
- apm_config.echo_canceller.use_legacy_aec = true;
- apm_config.echo_canceller.mobile_mode = false;
- apm_->ApplyConfig(apm_config);
- Config config;
- config.Set<DelayAgnostic>(new DelayAgnostic(false));
- apm_->SetExtraOptions(config);
-
- // Internally in the AEC the amount of lookahead the delay estimation can
- // handle is 15 blocks and the maximum delay is set to 60 blocks.
- const int kLookaheadBlocks = 15;
- const int kMaxDelayBlocks = 60;
- // The AEC has a startup time before it actually starts to process. This
- // procedure can flush the internal far-end buffer, which of course affects
- // the delay estimation. Therefore, we set a system_delay high enough to
- // avoid that. The smallest system_delay you can report without flushing the
- // buffer is 66 ms in 8 kHz.
- //
- // It is known that for 16 kHz (and 32 kHz) sampling frequency there is an
- // additional stuffing of 8 ms on the fly, but it seems to have no impact on
- // delay estimation. This should be noted though. In case of test failure,
- // this could be the cause.
- const int kSystemDelayMs = 66;
- // Test a couple of corner cases and verify that the estimated delay is
- // within a valid region (set to +-1.5 blocks). Note that these cases are
- // sampling frequency dependent.
- for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
- Init(kProcessSampleRates[i],
- kProcessSampleRates[i],
- kProcessSampleRates[i],
- 2,
- 2,
- 2,
- false);
- // Sampling frequency dependent variables.
- const int num_ms_per_block =
- std::max(4, static_cast<int>(640 / frame_->samples_per_channel_));
- const int delay_min_ms = -kLookaheadBlocks * num_ms_per_block;
- const int delay_max_ms = (kMaxDelayBlocks - 1) * num_ms_per_block;
-
- // 1) Verify correct delay estimate at lookahead boundary.
- int delay_ms = TruncateToMultipleOf10(kSystemDelayMs + delay_min_ms);
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- // 2) A delay less than maximum lookahead should give an delay estimate at
- // the boundary (= -kLookaheadBlocks * num_ms_per_block).
- delay_ms -= 20;
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- // 3) Three values around zero delay. Note that we need to compensate for
- // the fake system_delay.
- delay_ms = TruncateToMultipleOf10(kSystemDelayMs - 10);
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- delay_ms = TruncateToMultipleOf10(kSystemDelayMs);
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- delay_ms = TruncateToMultipleOf10(kSystemDelayMs + 10);
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- // 4) Verify correct delay estimate at maximum delay boundary.
- delay_ms = TruncateToMultipleOf10(kSystemDelayMs + delay_max_ms);
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- // 5) A delay above the maximum delay should give an estimate at the
- // boundary (= (kMaxDelayBlocks - 1) * num_ms_per_block).
- delay_ms += 20;
- ProcessDelayVerificationTest(delay_ms, kSystemDelayMs, delay_min_ms,
- delay_max_ms);
- }
-}
-
TEST_F(ApmTest, GainControl) {
// Testing gain modes
EXPECT_EQ(apm_->kNoError,
@@ -1452,8 +1361,6 @@
// Check the test is valid. We should have distortion from the filter
// when AEC is enabled (which won't affect the audio).
apm_config.echo_canceller.enabled = true;
- // TODO(peah): Update tests to instead use AEC3.
- apm_config.echo_canceller.use_legacy_aec = true;
apm_config.echo_canceller.mobile_mode = false;
apm_->ApplyConfig(apm_config);
frame_->samples_per_channel_ = 320;
@@ -1703,114 +1610,6 @@
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
-TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
- audioproc::OutputData ref_data;
- OpenFileAndReadMessage(ref_filename_, &ref_data);
-
- Config config;
- config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
- std::unique_ptr<AudioProcessing> fapm(
- AudioProcessingBuilder().Create(config));
- EnableAllComponents();
- EnableAllAPComponents(fapm.get());
- for (int i = 0; i < ref_data.test_size(); i++) {
- printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
-
- audioproc::Test* test = ref_data.mutable_test(i);
- // TODO(ajm): Restore downmixing test cases.
- if (test->num_input_channels() != test->num_output_channels())
- continue;
-
- const size_t num_render_channels =
- static_cast<size_t>(test->num_reverse_channels());
- const size_t num_input_channels =
- static_cast<size_t>(test->num_input_channels());
- const size_t num_output_channels =
- static_cast<size_t>(test->num_output_channels());
- const size_t samples_per_channel = static_cast<size_t>(
- test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000);
-
- Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
- num_input_channels, num_output_channels, num_render_channels, true);
- Init(fapm.get());
-
- ChannelBuffer<int16_t> output_cb(samples_per_channel, num_input_channels);
- ChannelBuffer<int16_t> output_int16(samples_per_channel,
- num_input_channels);
-
- int analog_level = 127;
- size_t num_bad_chunks = 0;
- while (ReadFrame(far_file_, revframe_, revfloat_cb_.get()) &&
- ReadFrame(near_file_, frame_, float_cb_.get())) {
- frame_->vad_activity_ = AudioFrame::kVadUnknown;
-
- EXPECT_NOERR(apm_->ProcessReverseStream(revframe_));
- EXPECT_NOERR(fapm->AnalyzeReverseStream(
- revfloat_cb_->channels(),
- samples_per_channel,
- test->sample_rate(),
- LayoutFromChannels(num_render_channels)));
-
- EXPECT_NOERR(apm_->set_stream_delay_ms(0));
- EXPECT_NOERR(fapm->set_stream_delay_ms(0));
- EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(analog_level));
- EXPECT_NOERR(fapm->gain_control()->set_stream_analog_level(analog_level));
-
- EXPECT_NOERR(apm_->ProcessStream(frame_));
- Deinterleave(frame_->data(), samples_per_channel, num_output_channels,
- output_int16.channels());
-
- EXPECT_NOERR(fapm->ProcessStream(
- float_cb_->channels(),
- samples_per_channel,
- test->sample_rate(),
- LayoutFromChannels(num_input_channels),
- test->sample_rate(),
- LayoutFromChannels(num_output_channels),
- float_cb_->channels()));
- for (size_t j = 0; j < num_output_channels; ++j) {
- FloatToS16(float_cb_->channels()[j],
- samples_per_channel,
- output_cb.channels()[j]);
- float variance = 0;
- float snr = ComputeSNR(output_int16.channels()[j],
- output_cb.channels()[j],
- samples_per_channel, &variance);
-
- const float kVarianceThreshold = 20;
- const float kSNRThreshold = 20;
-
- // Skip frames with low energy.
- if (std::sqrt(variance) > kVarianceThreshold && snr < kSNRThreshold) {
- ++num_bad_chunks;
- }
- }
-
- analog_level = fapm->gain_control()->stream_analog_level();
- EXPECT_EQ(apm_->gain_control()->stream_analog_level(),
- fapm->gain_control()->stream_analog_level());
- EXPECT_NEAR(apm_->noise_suppression()->speech_probability(),
- fapm->noise_suppression()->speech_probability(),
- 0.01);
-
- // Reset in case of downmixing.
- frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
- }
-
-#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- const size_t kMaxNumBadChunks = 0;
-#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- // There are a few chunks in the fixed-point profile that give low SNR.
- // Listening confirmed the difference is acceptable.
- const size_t kMaxNumBadChunks = 60;
-#endif
- EXPECT_LE(num_bad_chunks, kMaxNumBadChunks);
-
- rewind(far_file_);
- rewind(near_file_);
- }
-}
-
// TODO(andrew): Add a test to process a few frames with different combinations
// of enabled components.
@@ -1938,18 +1737,11 @@
const float echo_return_loss = stats.echo_return_loss.value_or(-1.0f);
const float echo_return_loss_enhancement =
stats.echo_return_loss_enhancement.value_or(-1.0f);
- const float divergent_filter_fraction =
- stats.divergent_filter_fraction.value_or(-1.0f);
const float residual_echo_likelihood =
stats.residual_echo_likelihood.value_or(-1.0f);
const float residual_echo_likelihood_recent_max =
stats.residual_echo_likelihood_recent_max.value_or(-1.0f);
- // Delay metrics.
- const int32_t delay_median_ms = stats.delay_median_ms.value_or(-1.0);
- const int32_t delay_standard_deviation_ms =
- stats.delay_standard_deviation_ms.value_or(-1.0);
-
if (!write_ref_data) {
const audioproc::Test::EchoMetrics& reference =
test->echo_metrics(stats_index);
@@ -1957,34 +1749,20 @@
EXPECT_NEAR(echo_return_loss, reference.echo_return_loss(), kEpsilon);
EXPECT_NEAR(echo_return_loss_enhancement,
reference.echo_return_loss_enhancement(), kEpsilon);
- EXPECT_NEAR(divergent_filter_fraction,
- reference.divergent_filter_fraction(), kEpsilon);
EXPECT_NEAR(residual_echo_likelihood,
reference.residual_echo_likelihood(), kEpsilon);
EXPECT_NEAR(residual_echo_likelihood_recent_max,
reference.residual_echo_likelihood_recent_max(),
kEpsilon);
-
- const audioproc::Test::DelayMetrics& reference_delay =
- test->delay_metrics(stats_index);
- EXPECT_EQ(reference_delay.median(), delay_median_ms);
- EXPECT_EQ(reference_delay.std(), delay_standard_deviation_ms);
-
++stats_index;
} else {
audioproc::Test::EchoMetrics* message_echo = test->add_echo_metrics();
message_echo->set_echo_return_loss(echo_return_loss);
message_echo->set_echo_return_loss_enhancement(
echo_return_loss_enhancement);
- message_echo->set_divergent_filter_fraction(
- divergent_filter_fraction);
message_echo->set_residual_echo_likelihood(residual_echo_likelihood);
message_echo->set_residual_echo_likelihood_recent_max(
residual_echo_likelihood_recent_max);
- audioproc::Test::DelayMetrics* message_delay =
- test->add_delay_metrics();
- message_delay->set_median(delay_median_ms);
- message_delay->set_std(delay_standard_deviation_ms);
}
}
#endif // defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE).
@@ -2689,12 +2467,8 @@
apm->ProcessStream(&audio);
}
-std::unique_ptr<AudioProcessing> CreateApm(bool use_AEC2) {
+std::unique_ptr<AudioProcessing> CreateApm(bool mobile_aec) {
Config old_config;
- if (use_AEC2) {
- old_config.Set<ExtendedFilter>(new ExtendedFilter(true));
- old_config.Set<DelayAgnostic>(new DelayAgnostic(true));
- }
std::unique_ptr<AudioProcessing> apm(
AudioProcessingBuilder().Create(old_config));
if (!apm) {
@@ -2709,14 +2483,12 @@
}
// Disable all components except for an AEC and the residual echo detector.
- // TODO(peah): Update this to also work on AEC3.
AudioProcessing::Config apm_config;
apm_config.residual_echo_detector.enabled = true;
apm_config.high_pass_filter.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.echo_canceller.enabled = true;
- apm_config.echo_canceller.mobile_mode = !use_AEC2;
- apm_config.echo_canceller.use_legacy_aec = use_AEC2;
+ apm_config.echo_canceller.mobile_mode = mobile_aec;
apm->ApplyConfig(apm_config);
EXPECT_EQ(apm->gain_control()->Enable(false), 0);
EXPECT_EQ(apm->level_estimator()->Enable(false), 0);
@@ -2731,14 +2503,12 @@
#define MAYBE_ApmStatistics ApmStatistics
#endif
-TEST(MAYBE_ApmStatistics, AEC2EnabledTest) {
- // Set up APM with AEC2 and process some audio.
- std::unique_ptr<AudioProcessing> apm = CreateApm(true);
+TEST(MAYBE_ApmStatistics, AECEnabledTest) {
+ // Set up APM with AEC3 and process some audio.
+ std::unique_ptr<AudioProcessing> apm = CreateApm(false);
ASSERT_TRUE(apm);
AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
- // TODO(peah): Update tests to instead use AEC3.
- apm_config.echo_canceller.use_legacy_aec = true;
apm->ApplyConfig(apm_config);
// Set up an audioframe.
@@ -2772,13 +2542,6 @@
EXPECT_NE(*stats.echo_return_loss, -100.0);
ASSERT_TRUE(stats.echo_return_loss_enhancement);
EXPECT_NE(*stats.echo_return_loss_enhancement, -100.0);
- ASSERT_TRUE(stats.divergent_filter_fraction);
- EXPECT_NE(*stats.divergent_filter_fraction, -1.0);
- ASSERT_TRUE(stats.delay_standard_deviation_ms);
- EXPECT_GE(*stats.delay_standard_deviation_ms, 0);
- // We don't check stats.delay_median_ms since it takes too long to settle to a
- // value. At least 20 seconds of data need to be processed before it will get
- // a value, which would make this test take too much time.
// If there are no receive streams, we expect the stats not to be set. The
// 'false' argument signals to APM that no receive streams are currently
@@ -2790,14 +2553,11 @@
EXPECT_FALSE(stats.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
- EXPECT_FALSE(stats.divergent_filter_fraction);
- EXPECT_FALSE(stats.delay_median_ms);
- EXPECT_FALSE(stats.delay_standard_deviation_ms);
}
TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
// Set up APM with AECM and process some audio.
- std::unique_ptr<AudioProcessing> apm = CreateApm(false);
+ std::unique_ptr<AudioProcessing> apm = CreateApm(true);
ASSERT_TRUE(apm);
// Set up an audioframe.
@@ -2834,9 +2594,6 @@
}
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
- EXPECT_FALSE(stats.divergent_filter_fraction);
- EXPECT_FALSE(stats.delay_median_ms);
- EXPECT_FALSE(stats.delay_standard_deviation_ms);
// If there are no receive streams, we expect the stats not to be set.
stats = apm->GetStatistics(false);
@@ -2844,9 +2601,6 @@
EXPECT_FALSE(stats.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.echo_return_loss);
EXPECT_FALSE(stats.echo_return_loss_enhancement);
- EXPECT_FALSE(stats.divergent_filter_fraction);
- EXPECT_FALSE(stats.delay_median_ms);
- EXPECT_FALSE(stats.delay_standard_deviation_ms);
}
TEST(ApmStatistics, ReportOutputRmsDbfs) {
diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1
index 6cadef9..907ebfd 100644
--- a/resources/audio_processing/output_data_float.pb.sha1
+++ b/resources/audio_processing/output_data_float.pb.sha1
@@ -1 +1 @@
-f3f7efa512900b06a30af8e3c92aa7863fbc96f8
\ No newline at end of file
+169276fe22bbeb1c06e0ed1a9df8149c5dbf8f80
\ No newline at end of file
