Launch new gain control tuning value for WebRTC's AGC

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_audio_processor.h"
 
 #include <stddef.h>
 #include <stdint.h>
 #include <utility>
 
@@ skipping 103 matching lines @@
       base::CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
           switches::kAgcStartupMinVolume));
   int startup_min_volume;
   if (min_volume_str.empty() ||
       !base::StringToInt(min_volume_str, &startup_min_volume)) {
     return base::Optional<int>();
   }
   return base::Optional<int>(startup_min_volume);
 }
 
-// Features for http://crbug.com/672476. These values will be given to WebRTC's
-// gain control (AGC) as lower bounds for the gain reduction during clipping.
-const base::Feature kTunedClippingLevelMin30{
-    "TunedClippingLevelMin30", base::FEATURE_DISABLED_BY_DEFAULT};
-const base::Feature kTunedClippingLevelMin70{
-    "TunedClippingLevelMin70", base::FEATURE_DISABLED_BY_DEFAULT};
-const base::Feature kTunedClippingLevelMin110{
-    "TunedClippingLevelMin110", base::FEATURE_DISABLED_BY_DEFAULT};
-const base::Feature kTunedClippingLevelMin150{
-    "TunedClippingLevelMin150", base::FEATURE_DISABLED_BY_DEFAULT};
-const base::Feature kTunedClippingLevelMin170{
-    "TunedClippingLevelMin170", base::FEATURE_DISABLED_BY_DEFAULT};
-
-base::Optional<int> GetClippingLevelMin() {
-  if (base::FeatureList::IsEnabled(kTunedClippingLevelMin30))
-    return base::Optional<int>(30);
-  if (base::FeatureList::IsEnabled(kTunedClippingLevelMin70))
-    return base::Optional<int>(70);
-  if (base::FeatureList::IsEnabled(kTunedClippingLevelMin110))
-    return base::Optional<int>(110);
-  if (base::FeatureList::IsEnabled(kTunedClippingLevelMin150))
-    return base::Optional<int>(150);
-  if (base::FeatureList::IsEnabled(kTunedClippingLevelMin170))
-    return base::Optional<int>(170);
-  return base::Optional<int>();
-}
-
 // Checks if the AEC's refined adaptive filter tuning was enabled on the command
 // line.
 bool UseAecRefinedAdaptiveFilter() {
   return base::CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kAecRefinedAdaptiveFilter);
 }
 
 }  // namespace
 
 // Wraps AudioBus to provide access to the array of channel pointers, since this
@@ skipping 505 matching lines @@
         GetArrayGeometryPreferringConstraints(audio_constraints, input_params);
 
     // Only enable beamforming if we have at least two mics.
     config.Set<webrtc::Beamforming>(
         new webrtc::Beamforming(geometry.size() > 1, geometry));
   }
 
   // If the experimental AGC is enabled, check for overridden config params.
   if (audio_constraints.GetGoogExperimentalAutoGainControl()) {
     auto startup_min_volume = GetStartupMinVolumeForAgc();
-    auto clipping_level_min = GetClippingLevelMin();
-    if (startup_min_volume || clipping_level_min) {
-      config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(
-          true, startup_min_volume.value_or(0),
-          clipping_level_min.value_or(webrtc::kClippedLevelMin)));
-    }
+    constexpr int kClippingLevelMin = 70;
+    // TODO(hlundin) Make this value default in WebRTC and clean up here.
+    config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(
+        true, startup_min_volume.value_or(0), kClippingLevelMin));
   }
 
   // Create and configure the webrtc::AudioProcessing.
   audio_processing_.reset(webrtc::AudioProcessing::Create(config));
 
   // Enable the audio processing components.
   webrtc::AudioProcessing::Config apm_config;
 
   if (playout_data_source_) {
     playout_data_source_->AddPlayoutSink(this);
@@ skipping 197 matching lines @@
       0 : agc->stream_analog_level();
 }
 
 void MediaStreamAudioProcessor::UpdateAecStats() {
   DCHECK(main_thread_runner_->BelongsToCurrentThread());
   if (echo_information_)
     echo_information_->UpdateAecStats(audio_processing_->echo_cancellation());
 }
 
 }  // namespace content