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);
diff --git a/modules/audio_processing/test/unittest.proto b/modules/audio_processing/test/unittest.proto
index 3c93bfd..7cd1deb 100644
--- a/modules/audio_processing/test/unittest.proto
+++ b/modules/audio_processing/test/unittest.proto
@@ -15,24 +15,15 @@
optional int32 analog_level_average = 6;
optional int32 max_output_average = 7;
-
- optional int32 has_echo_count = 8;
optional int32 has_voice_count = 9;
optional int32 is_saturated_count = 10;
- message Statistic {
- optional int32 instant = 1;
- optional int32 average = 2;
- optional int32 maximum = 3;
- optional int32 minimum = 4;
- }
-
message EchoMetrics {
- optional Statistic residual_echo_return_loss = 1;
- optional Statistic echo_return_loss = 2;
- optional Statistic echo_return_loss_enhancement = 3;
- optional Statistic a_nlp = 4;
- optional float divergent_filter_fraction = 5;
+ optional float echo_return_loss = 1;
+ optional float echo_return_loss_enhancement = 2;
+ optional float divergent_filter_fraction = 3;
+ optional float residual_echo_likelihood = 4;
+ optional float residual_echo_likelihood_recent_max = 5;
}
repeated EchoMetrics echo_metrics = 11;
@@ -40,7 +31,6 @@
message DelayMetrics {
optional int32 median = 1;
optional int32 std = 2;
- optional float fraction_poor_delays = 3;
}
repeated DelayMetrics delay_metrics = 12;
diff --git a/resources/audio_processing/output_data_fixed.pb.sha1 b/resources/audio_processing/output_data_fixed.pb.sha1
index 1e2d75a..072dc4f 100644
--- a/resources/audio_processing/output_data_fixed.pb.sha1
+++ b/resources/audio_processing/output_data_fixed.pb.sha1
@@ -1 +1 @@
-75a835fd9e641531989b7ee1de20e84b553d3bc9
\ No newline at end of file
+7481cf57b2ade2f600d91e8bc77fd9780a56b62e
\ No newline at end of file
diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1
index e86cb37..da3afc0 100644
--- a/resources/audio_processing/output_data_float.pb.sha1
+++ b/resources/audio_processing/output_data_float.pb.sha1
@@ -1 +1 @@
-64802f0a924f23b86a91d8d9d181bffe3f02c308
\ No newline at end of file
+82e9600c82f03c21e9feb33f82792d8d17908a5f
\ No newline at end of file
diff --git a/resources/audio_processing/output_data_mac.pb.sha1 b/resources/audio_processing/output_data_mac.pb.sha1
index cf4a896..56c4bd6 100644
--- a/resources/audio_processing/output_data_mac.pb.sha1
+++ b/resources/audio_processing/output_data_mac.pb.sha1
@@ -1 +1 @@
-98249662a7ba79579a199a8b9ef9f6fc26ff33aa
\ No newline at end of file
+3f2550064d3e71c6428c3759dcce18bddec36690
\ No newline at end of file