Add AudioProcessingCaptureStats and a level estimator replacement
This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.
Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 3764647..2937c06 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -259,6 +259,7 @@
std::unique_ptr<CustomProcessing> render_pre_processor;
std::unique_ptr<GainApplier> pre_amplifier;
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
+ std::unique_ptr<LevelEstimatorImpl> output_level_estimator;
};
AudioProcessingBuilder::AudioProcessingBuilder() = default;
@@ -673,6 +674,13 @@
<< config_.gain_controller2.enabled;
RTC_LOG(LS_INFO) << "Pre-amplifier activated: "
<< config_.pre_amplifier.enabled;
+
+ if (config_.level_estimation.enabled &&
+ !private_submodules_->output_level_estimator) {
+ private_submodules_->output_level_estimator.reset(
+ new LevelEstimatorImpl(&crit_capture_));
+ private_submodules_->output_level_estimator->Enable(true);
+ }
}
void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
@@ -1336,6 +1344,13 @@
// The level estimator operates on the recombined data.
public_submodules_->level_estimator->ProcessStream(capture_buffer);
+ if (config_.level_estimation.enabled) {
+ private_submodules_->output_level_estimator->ProcessStream(capture_buffer);
+ capture_.stats.output_rms_dbfs =
+ private_submodules_->output_level_estimator->RMS();
+ } else {
+ capture_.stats.output_rms_dbfs = absl::nullopt;
+ }
capture_output_rms_.Analyze(rtc::ArrayView<const int16_t>(
capture_buffer->channels_const()[0],
@@ -1587,49 +1602,50 @@
AudioProcessingStats AudioProcessingImpl::GetStatistics(
bool has_remote_tracks) const {
- AudioProcessingStats stats;
- if (has_remote_tracks) {
- EchoCancellationImpl::Metrics metrics;
- rtc::CritScope cs_capture(&crit_capture_);
- if (private_submodules_->echo_controller) {
- auto ec_metrics = private_submodules_->echo_controller->GetMetrics();
- stats.echo_return_loss = ec_metrics.echo_return_loss;
+ rtc::CritScope cs_capture(&crit_capture_);
+ if (!has_remote_tracks) {
+ return capture_.stats;
+ }
+ AudioProcessingStats stats = capture_.stats;
+ EchoCancellationImpl::Metrics metrics;
+ if (private_submodules_->echo_controller) {
+ auto ec_metrics = private_submodules_->echo_controller->GetMetrics();
+ stats.echo_return_loss = ec_metrics.echo_return_loss;
+ stats.echo_return_loss_enhancement =
+ ec_metrics.echo_return_loss_enhancement;
+ stats.delay_ms = ec_metrics.delay_ms;
+ } else if (private_submodules_->echo_cancellation->GetMetrics(&metrics) ==
+ Error::kNoError) {
+ if (metrics.divergent_filter_fraction != -1.0f) {
+ stats.divergent_filter_fraction =
+ absl::optional<double>(metrics.divergent_filter_fraction);
+ }
+ if (metrics.echo_return_loss.instant != -100) {
+ stats.echo_return_loss =
+ absl::optional<double>(metrics.echo_return_loss.instant);
+ }
+ if (metrics.echo_return_loss_enhancement.instant != -100) {
stats.echo_return_loss_enhancement =
- ec_metrics.echo_return_loss_enhancement;
- stats.delay_ms = ec_metrics.delay_ms;
- } else if (private_submodules_->echo_cancellation->GetMetrics(&metrics) ==
- Error::kNoError) {
- if (metrics.divergent_filter_fraction != -1.0f) {
- stats.divergent_filter_fraction =
- absl::optional<double>(metrics.divergent_filter_fraction);
- }
- if (metrics.echo_return_loss.instant != -100) {
- stats.echo_return_loss =
- absl::optional<double>(metrics.echo_return_loss.instant);
- }
- if (metrics.echo_return_loss_enhancement.instant != -100) {
- stats.echo_return_loss_enhancement = absl::optional<double>(
- metrics.echo_return_loss_enhancement.instant);
- }
+ absl::optional<double>(metrics.echo_return_loss_enhancement.instant);
}
- if (config_.residual_echo_detector.enabled) {
- RTC_DCHECK(private_submodules_->echo_detector);
- auto ed_metrics = private_submodules_->echo_detector->GetMetrics();
- stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
- stats.residual_echo_likelihood_recent_max =
- ed_metrics.echo_likelihood_recent_max;
+ }
+ if (config_.residual_echo_detector.enabled) {
+ RTC_DCHECK(private_submodules_->echo_detector);
+ auto ed_metrics = private_submodules_->echo_detector->GetMetrics();
+ stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
+ stats.residual_echo_likelihood_recent_max =
+ ed_metrics.echo_likelihood_recent_max;
+ }
+ int delay_median, delay_std;
+ float fraction_poor_delays;
+ if (private_submodules_->echo_cancellation->GetDelayMetrics(
+ &delay_median, &delay_std, &fraction_poor_delays) ==
+ Error::kNoError) {
+ if (delay_median >= 0) {
+ stats.delay_median_ms = absl::optional<int32_t>(delay_median);
}
- int delay_median, delay_std;
- float fraction_poor_delays;
- if (private_submodules_->echo_cancellation->GetDelayMetrics(
- &delay_median, &delay_std, &fraction_poor_delays) ==
- Error::kNoError) {
- if (delay_median >= 0) {
- stats.delay_median_ms = absl::optional<int32_t>(delay_median);
- }
- if (delay_std >= 0) {
- stats.delay_standard_deviation_ms = absl::optional<int32_t>(delay_std);
- }
+ if (delay_std >= 0) {
+ stats.delay_standard_deviation_ms = absl::optional<int32_t>(delay_std);
}
}
return stats;
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index e376a74..2f946c5 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -18,6 +18,7 @@
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/render_queue_item_verifier.h"
#include "modules/audio_processing/rms_level.h"
#include "rtc_base/criticalsection.h"
@@ -390,6 +391,7 @@
bool echo_path_gain_change;
int prev_analog_mic_level;
float prev_pre_amp_gain;
+ AudioProcessingStats stats;
} capture_ RTC_GUARDED_BY(crit_capture_);
struct ApmCaptureNonLockedState {
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 18e669f..6809ab9 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2801,4 +2801,42 @@
EXPECT_FALSE(stats.delay_median_ms);
EXPECT_FALSE(stats.delay_standard_deviation_ms);
}
+
+TEST(ApmStatistics, ReportOutputRmsDbfs) {
+ ProcessingConfig processing_config = {
+ {{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
+ AudioProcessing::Config config;
+
+ // Set up an audioframe.
+ AudioFrame frame;
+ frame.num_channels_ = 1;
+ SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate48kHz);
+
+ // Fill the audio frame with a sawtooth pattern.
+ int16_t* ptr = frame.mutable_data();
+ for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
+ ptr[i] = 10000 * ((i % 3) - 1);
+ }
+
+ std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
+ apm->Initialize(processing_config);
+
+ // If not enabled, no metric should be reported.
+ EXPECT_EQ(apm->ProcessStream(&frame), 0);
+ EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
+
+ // If enabled, metrics should be reported.
+ config.level_estimation.enabled = true;
+ apm->ApplyConfig(config);
+ EXPECT_EQ(apm->ProcessStream(&frame), 0);
+ auto stats = apm->GetStatistics(false);
+ EXPECT_TRUE(stats.output_rms_dbfs);
+ EXPECT_GE(*stats.output_rms_dbfs, 0);
+
+ // If re-disabled, the value is again not reported.
+ config.level_estimation.enabled = false;
+ apm->ApplyConfig(config);
+ EXPECT_EQ(apm->ProcessStream(&frame), 0);
+ EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
+}
} // namespace webrtc
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 9a1a03c..df51313 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -283,6 +283,11 @@
} adaptive_digital;
} gain_controller2;
+ // Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
+ struct LevelEstimation {
+ bool enabled = false;
+ } level_estimation;
+
// Explicit copy assignment implementation to avoid issues with memory
// sanitizer complaints in case of self-assignment.
// TODO(peah): Add buildflag to ensure that this is only included for memory
diff --git a/modules/audio_processing/include/audio_processing_statistics.h b/modules/audio_processing/include/audio_processing_statistics.h
index 2ff2009..683db05 100644
--- a/modules/audio_processing/include/audio_processing_statistics.h
+++ b/modules/audio_processing/include/audio_processing_statistics.h
@@ -24,6 +24,14 @@
AudioProcessingStats(const AudioProcessingStats& other);
~AudioProcessingStats();
+ // The root mean square (RMS) level in dBFS (decibels from digital
+ // full-scale) of the last capture frame, after processing. It is
+ // constrained to [-127, 0].
+ // The computation follows: https://tools.ietf.org/html/rfc6465
+ // with the intent that it can provide the RTP audio level indication.
+ // Only reported if level estimation is enabled in AudioProcessing::Config.
+ absl::optional<int> output_rms_dbfs;
+
// AEC Statistics.
// ERL = 10log_10(P_far / P_echo)
absl::optional<double> echo_return_loss;