Remove AEC interface usage in MediaStreamAudioProcessor and AudioProcessor

echo_control_mobile() and echo_cancellation() interfaces are replaced
with AudioProcessing::Config.

Some configuration is now hardcoded defaults in APM, and hence should
not (can not) be configured in these processors.

Statistics are now accessed through AudioProcessing::GetStatistics.

A UMA metric disappears.

Bug: webrtc:9535
Change-Id: Ia23d58cf1a1315605beef3ec72e080844bb94e5d
Reviewed-on: https://chromium-review.googlesource.com/1243123Reviewed-by: Oskar Sundbom <ossu@chromium.org>
Reviewed-by: Max Morin <maxmorin@chromium.org>
Commit-Queue: Sam Zackrisson <saza@chromium.org>
Cr-Commit-Position: refs/heads/master@{#595806}

content/renderer/media/stream/media_stream_audio_processor.cc

@@ -471,11 +471,6 @@ void MediaStreamAudioProcessor::OnPlayoutData(media::AudioBus* audio_bus,
                                               int sample_rate,
                                               int audio_delay_milliseconds) {
   DCHECK(render_thread_checker_.CalledOnValidThread());
-#if defined(OS_ANDROID)
-  DCHECK(!audio_processing_->echo_cancellation()->is_enabled());
-#else
-  DCHECK(!audio_processing_->echo_control_mobile()->is_enabled());
-#endif
   DCHECK_GE(audio_bus->channels(), 1);
   DCHECK_LE(audio_bus->channels(), 2);
   int frames_per_10_ms = sample_rate / 100;
@@ -792,7 +787,8 @@ int MediaStreamAudioProcessor::ProcessData(const float* const* process_ptrs,
 void MediaStreamAudioProcessor::UpdateAecStats() {
   DCHECK(main_thread_runner_->BelongsToCurrentThread());
   if (echo_information_)
-    echo_information_->UpdateAecStats(audio_processing_->echo_cancellation());
+    echo_information_->UpdateAecStats(
+        audio_processing_->GetStatistics(true /* has_remote_tracks */));
 }
 
 }  // namespace content

content/renderer/media/stream/media_stream_audio_processor_options.cc

@@ -91,25 +91,11 @@ AudioProcessingProperties::ToAudioProcessingSettings() const {
 }
 
 void EnableEchoCancellation(AudioProcessing* audio_processing) {
-  // TODO(bugs.webrtc.org/9535): Remove double-booking AEC toggle when the
-  // config applies (from 2018-08-16).
   webrtc::AudioProcessing::Config apm_config = audio_processing->GetConfig();
   apm_config.echo_canceller.enabled = true;
 #if defined(OS_ANDROID)
-  // Mobile devices are using AECM.
-  CHECK_EQ(0, audio_processing->echo_control_mobile()->set_routing_mode(
-                  webrtc::EchoControlMobile::kSpeakerphone));
-  CHECK_EQ(0, audio_processing->echo_control_mobile()->Enable(true));
   apm_config.echo_canceller.mobile_mode = true;
 #else
-  int err = audio_processing->echo_cancellation()->set_suppression_level(
-      webrtc::EchoCancellation::kHighSuppression);
-
-  // Enable the metrics for AEC.
-  err |= audio_processing->echo_cancellation()->enable_metrics(true);
-  err |= audio_processing->echo_cancellation()->enable_delay_logging(true);
-  err |= audio_processing->echo_cancellation()->Enable(true);
-  CHECK_EQ(err, 0);
   apm_config.echo_canceller.mobile_mode = false;
 #endif
   audio_processing->ApplyConfig(apm_config);

content/renderer/media/stream/media_stream_audio_processor_unittest.cc

@@ -117,13 +117,7 @@ class MediaStreamAudioProcessorTest : public ::testing::Test {
       // |audio_processor| does nothing when the audio processing is off in
       // the processor.
       webrtc::AudioProcessing* ap = audio_processor->audio_processing_.get();
-#if defined(OS_ANDROID)
-      const bool is_aec_enabled = ap && ap->echo_control_mobile()->is_enabled();
-      // AEC should be turned off for mobiles.
-      DCHECK(!ap || !ap->echo_cancellation()->is_enabled());
-#else
-      const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
-#endif
+      const bool is_aec_enabled = ap && ap->GetConfig().echo_canceller.enabled;
       if (is_aec_enabled) {
         if (params.channels() > kMaxNumberOfPlayoutDataChannels) {
           for (int i = 0; i < kMaxNumberOfPlayoutDataChannels; ++i) {
@@ -159,22 +153,15 @@ class MediaStreamAudioProcessorTest : public ::testing::Test {
   void VerifyDefaultComponents(MediaStreamAudioProcessor* audio_processor) {
     webrtc::AudioProcessing* audio_processing =
         audio_processor->audio_processing_.get();
-    const webrtc::AudioProcessing::Config apm_config =
+    const webrtc::AudioProcessing::Config config =
         audio_processing->GetConfig();
-    EXPECT_TRUE(apm_config.high_pass_filter.enabled);
+    EXPECT_TRUE(config.echo_canceller.enabled);
 #if defined(OS_ANDROID)
-    EXPECT_TRUE(audio_processing->echo_control_mobile()->is_enabled());
-    EXPECT_TRUE(audio_processing->echo_control_mobile()->routing_mode() ==
-        webrtc::EchoControlMobile::kSpeakerphone);
-    EXPECT_FALSE(audio_processing->echo_cancellation()->is_enabled());
+    EXPECT_TRUE(config.echo_canceller.mobile_mode);
 #else
-    EXPECT_TRUE(audio_processing->echo_cancellation()->is_enabled());
-    EXPECT_TRUE(audio_processing->echo_cancellation()->suppression_level() ==
-        webrtc::EchoCancellation::kHighSuppression);
-    EXPECT_TRUE(audio_processing->echo_cancellation()->are_metrics_enabled());
-    EXPECT_TRUE(
-        audio_processing->echo_cancellation()->is_delay_logging_enabled());
+    EXPECT_FALSE(config.echo_canceller.mobile_mode);
 #endif
+    EXPECT_TRUE(config.high_pass_filter.enabled);
 
     EXPECT_TRUE(audio_processing->noise_suppression()->is_enabled());
     EXPECT_TRUE(audio_processing->noise_suppression()->level() ==

media/webrtc/audio_processor.cc

@@ -129,7 +129,8 @@ void AudioProcessor::AnalyzePlayout(const AudioBus& audio,
 
 void AudioProcessor::UpdateInternalStats() {
   if (audio_processing_)
-    echo_information_.UpdateAecStats(audio_processing_->echo_cancellation());
+    echo_information_.UpdateAecStats(
+        audio_processing_->GetStatistics(has_reverse_stream_));
 }
 
 void AudioProcessor::GetStats(GetStatsCB callback) {
@@ -250,22 +251,6 @@ void AudioProcessor::InitializeAPM() {
   // Audio processing module construction.
   audio_processing_ = base::WrapUnique(ap_builder.Create(ap_config));
 
-  // AEC setup part 2.
-  switch (settings_.echo_cancellation) {
-    case EchoCancellationType::kSystemAec:
-    case EchoCancellationType::kDisabled:
-      break;
-    case EchoCancellationType::kAec2:
-    case EchoCancellationType::kAec3:
-      int err = audio_processing_->echo_cancellation()->set_suppression_level(
-          webrtc::EchoCancellation::kHighSuppression);
-      err |= audio_processing_->echo_cancellation()->enable_metrics(true);
-      err |= audio_processing_->echo_cancellation()->enable_delay_logging(true);
-      err |= audio_processing_->echo_cancellation()->Enable(true);
-      DCHECK_EQ(err, 0);
-      break;
-  }
-
   // Noise suppression setup part 2.
   if (settings_.noise_suppression != NoiseSuppressionType::kDisabled) {
     int err = audio_processing_->noise_suppression()->set_level(
@@ -302,8 +287,7 @@ void AudioProcessor::InitializeAPM() {
 
   webrtc::AudioProcessing::Config apm_config = audio_processing_->GetConfig();
 
-  // AEC setup part 3.
-  // New-fangled echo cancellation setup. (see: https://bugs.webrtc.org/9535).
+  // AEC setup part 2.
   apm_config.echo_canceller.enabled =
       settings_.echo_cancellation == EchoCancellationType::kAec2 ||
       settings_.echo_cancellation == EchoCancellationType::kAec3;

media/webrtc/audio_processor_unittest.cc

@@ -105,7 +105,7 @@ class WebRtcAudioProcessorTest : public ::testing::Test {
       // |audio_processor| does nothing when the audio processing is off in
       // the processor.
       webrtc::AudioProcessing* ap = audio_processor->audio_processing_.get();
-      const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
+      const bool is_aec_enabled = ap && ap->GetConfig().echo_canceller.enabled;
       if (is_aec_enabled) {
         if (params.channels() > kMaxNumberOfPlayoutDataChannels) {
           for (int i = 0; i < kMaxNumberOfPlayoutDataChannels; ++i) {
@@ -128,12 +128,9 @@ class WebRtcAudioProcessorTest : public ::testing::Test {
   void VerifyEnabledComponents(AudioProcessor* audio_processor) {
     webrtc::AudioProcessing* audio_processing =
         audio_processor->audio_processing_.get();
-    EXPECT_TRUE(audio_processing->echo_cancellation()->is_enabled());
-    EXPECT_TRUE(audio_processing->echo_cancellation()->suppression_level() ==
-                webrtc::EchoCancellation::kHighSuppression);
-    EXPECT_TRUE(audio_processing->echo_cancellation()->are_metrics_enabled());
-    EXPECT_TRUE(
-        audio_processing->echo_cancellation()->is_delay_logging_enabled());
+    webrtc::AudioProcessing::Config ap_config = audio_processing->GetConfig();
+    EXPECT_TRUE(ap_config.echo_canceller.enabled);
+    EXPECT_FALSE(ap_config.echo_canceller.mobile_mode);
 
     EXPECT_TRUE(audio_processing->noise_suppression()->is_enabled());
     EXPECT_TRUE(audio_processing->noise_suppression()->level() ==

media/webrtc/echo_information.cc

@@ -9,44 +9,8 @@
 
 namespace media {
 
-namespace {
-
-// Used to log echo quality based on delay estimates.
-enum DelayBasedEchoQuality {
-  DELAY_BASED_ECHO_QUALITY_GOOD = 0,
-  DELAY_BASED_ECHO_QUALITY_SPURIOUS,
-  DELAY_BASED_ECHO_QUALITY_BAD,
-  DELAY_BASED_ECHO_QUALITY_INVALID,
-  DELAY_BASED_ECHO_QUALITY_MAX = DELAY_BASED_ECHO_QUALITY_INVALID
-};
-
-DelayBasedEchoQuality EchoDelayFrequencyToQuality(float delay_frequency) {
-  const float kEchoDelayFrequencyLowerLimit = 0.1f;
-  const float kEchoDelayFrequencyUpperLimit = 0.8f;
-  // DELAY_BASED_ECHO_QUALITY_GOOD
-  //   delay is out of bounds during at most 10 % of the time.
-  // DELAY_BASED_ECHO_QUALITY_SPURIOUS
-  //   delay is out of bounds 10-80 % of the time.
-  // DELAY_BASED_ECHO_QUALITY_BAD
-  //   delay is mostly out of bounds >= 80 % of the time.
-  // DELAY_BASED_ECHO_QUALITY_INVALID
-  //   delay_frequency is negative which happens if we have insufficient data.
-  if (delay_frequency < 0)
-    return DELAY_BASED_ECHO_QUALITY_INVALID;
-  else if (delay_frequency <= kEchoDelayFrequencyLowerLimit)
-    return DELAY_BASED_ECHO_QUALITY_GOOD;
-  else if (delay_frequency < kEchoDelayFrequencyUpperLimit)
-    return DELAY_BASED_ECHO_QUALITY_SPURIOUS;
-  else
-    return DELAY_BASED_ECHO_QUALITY_BAD;
-}
-
-}  // namespace
-
 EchoInformation::EchoInformation()
-    : delay_stats_time_ms_(0),
-      echo_frames_received_(false),
-      divergent_filter_stats_time_ms_(0),
+    : divergent_filter_stats_time_ms_(0),
       num_divergent_filter_fraction_(0),
       num_non_zero_divergent_filter_fraction_(0) {}
 
@@ -56,85 +20,30 @@ EchoInformation::~EchoInformation() {
 }
 
 void EchoInformation::UpdateAecStats(
-    webrtc::EchoCancellation* echo_cancellation) {
-  DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
-
-  if (!echo_cancellation->is_enabled())
-    return;
-
-  UpdateAecDelayStats(echo_cancellation);
-  UpdateAecDivergentFilterStats(echo_cancellation);
-}
-
-void EchoInformation::UpdateAecDelayStats(
-    webrtc::EchoCancellation* echo_cancellation) {
+    const webrtc::AudioProcessingStats& audio_processing_stats) {
   DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
 
-  // Only start collecting stats if we know echo cancellation has measured an
-  // echo. Otherwise we clutter the stats with for example cases where only the
-  // microphone is used.
-  if (!echo_frames_received_ & !echo_cancellation->stream_has_echo())
-    return;
-
-  echo_frames_received_ = true;
-
-  // In WebRTC, three echo delay metrics are calculated and updated every
-  // five seconds. We use one of them, |fraction_poor_delays| to log in a UMA
-  // histogram an Echo Cancellation quality metric. The stat in WebRTC has a
-  // fixed aggregation window of five seconds, so we use the same query
-  // frequency to avoid logging old values.
-  if (!echo_cancellation->is_delay_logging_enabled())
-    return;
-
-  delay_stats_time_ms_ += webrtc::AudioProcessing::kChunkSizeMs;
-  if (delay_stats_time_ms_ <
-      500 * webrtc::AudioProcessing::kChunkSizeMs) {  // 5 seconds
+  if (!audio_processing_stats.divergent_filter_fraction) {
     return;
   }
 
-  int dummy_median = 0, dummy_std = 0;
-  float fraction_poor_delays = 0;
-  if (echo_cancellation->GetDelayMetrics(&dummy_median, &dummy_std,
-                                         &fraction_poor_delays) ==
-      webrtc::AudioProcessing::kNoError) {
-    delay_stats_time_ms_ = 0;
-    // Map |fraction_poor_delays| to an Echo Cancellation quality and log in UMA
-    // histogram. See DelayBasedEchoQuality for information on histogram
-    // buckets.
-    UMA_HISTOGRAM_ENUMERATION("WebRTC.AecDelayBasedQuality",
-                              EchoDelayFrequencyToQuality(fraction_poor_delays),
-                              DELAY_BASED_ECHO_QUALITY_MAX + 1);
-  }
-}
-
-void EchoInformation::UpdateAecDivergentFilterStats(
-    webrtc::EchoCancellation* echo_cancellation) {
-  DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
-
-  if (!echo_cancellation->are_metrics_enabled())
-    return;
-
   divergent_filter_stats_time_ms_ += webrtc::AudioProcessing::kChunkSizeMs;
   if (divergent_filter_stats_time_ms_ <
       100 * webrtc::AudioProcessing::kChunkSizeMs) {  // 1 second
     return;
   }
 
-  webrtc::EchoCancellation::Metrics metrics;
-  if (echo_cancellation->GetMetrics(&metrics) ==
-      webrtc::AudioProcessing::kNoError) {
+  double divergent_filter_fraction =
+      *audio_processing_stats.divergent_filter_fraction;
   // If not yet calculated, |metrics.divergent_filter_fraction| is -1.0. After
   // being calculated the first time, it is updated periodically.
-    if (metrics.divergent_filter_fraction < 0.0f) {
+  if (divergent_filter_fraction < 0.0f) {
     DCHECK_EQ(num_divergent_filter_fraction_, 0);
     return;
   }
-    if (metrics.divergent_filter_fraction > 0.0f) {
+  if (divergent_filter_fraction > 0.0f) {
     ++num_non_zero_divergent_filter_fraction_;
   }
-  } else {
-    DLOG(WARNING) << "Get echo cancellation metrics failed.";
-  }
   ++num_divergent_filter_fraction_;
   divergent_filter_stats_time_ms_ = 0;
 }

media/webrtc/echo_information.h

@@ -11,32 +11,24 @@
 
 namespace media {
 
-// A helper class to log echo information in general and Echo Cancellation
+// A helper class to log echo information in general and AEC2
 // quality in particular.
 class COMPONENT_EXPORT(MEDIA_WEBRTC) EchoInformation {
  public:
   EchoInformation();
   virtual ~EchoInformation();
 
-  // Updates stats, and reports delay metrics as UMA stats every 5 seconds.
+  // Updates stats, and reports metrics as UMA stats every 5 seconds.
   // Must be called every time AudioProcessing::ProcessStream() is called.
-  void UpdateAecStats(webrtc::EchoCancellation* echo_cancellation);
+  void UpdateAecStats(
+      const webrtc::AudioProcessingStats& audio_processing_stats);
 
   // Reports AEC divergent filter metrics as UMA and resets the associated data.
   void ReportAndResetAecDivergentFilterStats();
 
  private:
-  void UpdateAecDelayStats(webrtc::EchoCancellation* echo_cancellation);
-  void UpdateAecDivergentFilterStats(
-      webrtc::EchoCancellation* echo_cancellation);
-
-  // Counter to track 5 seconds of data in order to query a new metric from
-  // webrtc::EchoCancellation::GetEchoDelayMetrics().
-  int delay_stats_time_ms_;
-  bool echo_frames_received_;
-
-  // Counter to track 1 second of data in order to query a new divergent filter
-  // fraction metric from webrtc::EchoCancellation::GetMetrics().
+  // Counter to store a new value for the divergent filter fraction metric in
+  // AEC2, once every second.
   int divergent_filter_stats_time_ms_;
 
   // Total number of times we queried for the divergent filter fraction metric.