Added a chrome flag to enable Delay Agnostic AEC in WebRTC

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 "base/command_line.h"
 #include "base/debug/trace_event.h"
-#if defined(OS_MACOSX)
 #include "base/metrics/field_trial.h"
-#endif
 #include "base/metrics/histogram.h"
+#include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_converter.h"
 #include "media/base/audio_fifo.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
@@ skipping 48 matching lines @@
   AUDIO_PROCESSING_DISABLED,
   AUDIO_PROCESSING_IN_WEBRTC,
   AUDIO_PROCESSING_MAX
 };
 
 void RecordProcessingState(AudioTrackProcessingStates state) {
   UMA_HISTOGRAM_ENUMERATION("Media.AudioTrackProcessingStates",
                             state, AUDIO_PROCESSING_MAX);
 }
 
+bool isDelayAgnosticAecEnabled() {
+  // Note: It's important to query the field trial state first, to ensure that
+  // UMA reports the correct group.
+  const std::string group_name =
+      base::FieldTrialList::FindFullName("NoReportedDelayOnMac");
+  base::CommandLine* command_line = base::CommandLine::ForCurrentProcess();
+  if (command_line->HasSwitch(switches::kEnableDelayAgnosticAec))
+    return true;
+
+  return (!group_name.empty() && group_name == "Enabled");
+}
 }  // namespace
 
 // Wraps AudioBus to provide access to the array of channel pointers, since this
 // is the type webrtc::AudioProcessing deals in. The array is refreshed on every
 // channel_ptrs() call, and will be valid until the underlying AudioBus pointers
 // are changed, e.g. through calls to SetChannelData() or SwapChannels().
 //
 // All methods are called on one of the capture or render audio threads
 // exclusively.
 class MediaStreamAudioBus {
@@ skipping 371 matching lines @@
     RecordProcessingState(AUDIO_PROCESSING_DISABLED);
     return;
   }
 
   // Experimental options provided at creation.
   webrtc::Config config;
   if (goog_experimental_aec)
     config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
   if (goog_experimental_ns)
     config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true));
-#if defined(OS_MACOSX)
-  if (base::FieldTrialList::FindFullName("NoReportedDelayOnMac") == "Enabled")
+  if (isDelayAgnosticAecEnabled())
     config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false));
-#endif
   if (goog_beamforming) {
     ConfigureBeamforming(&config);
   }
 
   // Create and configure the webrtc::AudioProcessing.
   audio_processing_.reset(webrtc::AudioProcessing::Create(config));
 
   // Enable the audio processing components.
   if (echo_cancellation) {
     EnableEchoCancellation(audio_processing_.get());
@@ skipping 185 matching lines @@
                                               vad->stream_has_voice());
     base::subtle::Release_Store(&typing_detected_, detected);
   }
 
   // Return 0 if the volume hasn't been changed, and otherwise the new volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 }  // namespace content