Drop legacy AEC metrics interface from ApmTest.Process
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index a894817..dae5db2 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -225,24 +225,6 @@
return max_data;
}
-#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
-void TestStats(const AudioProcessing::Statistic& test,
- const audioproc::Test::Statistic& reference) {
- EXPECT_EQ(reference.instant(), test.instant);
- EXPECT_EQ(reference.average(), test.average);
- EXPECT_EQ(reference.maximum(), test.maximum);
- EXPECT_EQ(reference.minimum(), test.minimum);
-}
-
-void WriteStatsMessage(const AudioProcessing::Statistic& output,
- audioproc::Test::Statistic* msg) {
- msg->set_instant(output.instant);
- msg->set_average(output.average);
- msg->set_maximum(output.maximum);
- msg->set_minimum(output.minimum);
-}
-#endif
-
void OpenFileAndWriteMessage(const std::string& filename,
const MessageLite& msg) {
FILE* file = fopen(filename.c_str(), "wb");
@@ -2047,7 +2029,6 @@
true);
int frame_count = 0;
- int has_echo_count = 0;
int has_voice_count = 0;
int is_saturated_count = 0;
int analog_level = 127;
@@ -2076,10 +2057,6 @@
max_output_average += MaxAudioFrame(*frame_);
- if (apm_->echo_cancellation()->stream_has_echo()) {
- has_echo_count++;
- }
-
analog_level = apm_->gain_control()->stream_analog_level();
analog_level_average += analog_level;
if (apm_->gain_control()->stream_is_saturated()) {
@@ -2108,18 +2085,25 @@
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kStatsAggregationFrameNum = 100; // 1 second.
if (frame_count % kStatsAggregationFrameNum == 0) {
- // Get echo metrics.
- EchoCancellation::Metrics echo_metrics;
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->GetMetrics(&echo_metrics));
+ // Get echo and delay metrics.
+ AudioProcessingStats stats =
+ apm_->GetStatistics(true /* has_remote_tracks */);
- // Get delay metrics.
- int median = 0;
- int std = 0;
- float fraction_poor_delays = 0;
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->GetDelayMetrics(
- &median, &std, &fraction_poor_delays));
+ // Echo metrics.
+ 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);
// Get RMS.
int rms_level = apm_->level_estimator()->RMS();
@@ -2129,46 +2113,40 @@
if (!write_ref_data) {
const audioproc::Test::EchoMetrics& reference =
test->echo_metrics(stats_index);
- TestStats(echo_metrics.residual_echo_return_loss,
- reference.residual_echo_return_loss());
- TestStats(echo_metrics.echo_return_loss,
- reference.echo_return_loss());
- TestStats(echo_metrics.echo_return_loss_enhancement,
- reference.echo_return_loss_enhancement());
- TestStats(echo_metrics.a_nlp,
- reference.a_nlp());
- EXPECT_EQ(echo_metrics.divergent_filter_fraction,
- reference.divergent_filter_fraction());
+ constexpr float kEpsilon = 0.01;
+ 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(), median);
- EXPECT_EQ(reference_delay.std(), std);
- EXPECT_EQ(reference_delay.fraction_poor_delays(),
- fraction_poor_delays);
+ EXPECT_EQ(reference_delay.median(), delay_median_ms);
+ EXPECT_EQ(reference_delay.std(), delay_standard_deviation_ms);
EXPECT_EQ(test->rms_level(stats_index), rms_level);
++stats_index;
} else {
- audioproc::Test::EchoMetrics* message =
- test->add_echo_metrics();
- WriteStatsMessage(echo_metrics.residual_echo_return_loss,
- message->mutable_residual_echo_return_loss());
- WriteStatsMessage(echo_metrics.echo_return_loss,
- message->mutable_echo_return_loss());
- WriteStatsMessage(echo_metrics.echo_return_loss_enhancement,
- message->mutable_echo_return_loss_enhancement());
- WriteStatsMessage(echo_metrics.a_nlp,
- message->mutable_a_nlp());
- message->set_divergent_filter_fraction(
- echo_metrics.divergent_filter_fraction);
-
+ 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(median);
- message_delay->set_std(std);
- message_delay->set_fraction_poor_delays(fraction_poor_delays);
+ message_delay->set_median(delay_median_ms);
+ message_delay->set_std(delay_standard_deviation_ms);
test->add_rms_level(rms_level);
}
@@ -2198,7 +2176,6 @@
const int kMaxOutputAverageOffset = 0;
const int kMaxOutputAverageNear = kIntNear;
#endif
- EXPECT_NEAR(test->has_echo_count(), has_echo_count, kIntNear);
EXPECT_NEAR(test->has_voice_count(),
has_voice_count - kHasVoiceCountOffset,
kHasVoiceCountNear);
@@ -2215,7 +2192,6 @@
kFloatNear);
#endif
} else {
- test->set_has_echo_count(has_echo_count);
test->set_has_voice_count(has_voice_count);
test->set_is_saturated_count(is_saturated_count);