Include default communication devices in audio device enumerations. This removes heuristic that pic…

content/renderer/media/webrtc_local_audio_renderer.cc

 // Copyright (c) 2012 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/webrtc_local_audio_renderer.h"
 
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/synchronization/lock.h"
@@ skipping 237 matching lines @@
   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                             kSinkStarted, kSinkStatesMax);
 }
 
 void WebRtcLocalAudioRenderer::ReconfigureSink(
     const media::AudioParameters& params) {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()";
 
-  int implicit_ducking_effect = 0;
-  RenderFrameImpl* const frame =
-      RenderFrameImpl::FromRoutingID(source_render_frame_id_);
-  MediaStreamDispatcher* const dispatcher = frame ?
-      frame->GetMediaStreamDispatcher() : NULL;
-  if (dispatcher && dispatcher->IsAudioDuckingActive()) {
-    DVLOG(1) << "Forcing DUCKING to be ON for output";
-    implicit_ducking_effect = media::AudioParameters::DUCKING;
-  } else {
-    DVLOG(1) << "DUCKING not forced ON for output";
-  }
-
   if (source_params_.Equals(params))
     return;
 
   // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
   // the new format.
 
   source_params_ = params;
 
   sink_params_ = media::AudioParameters(source_params_.format(),
       source_params_.channel_layout(), source_params_.sample_rate(),
       source_params_.bits_per_sample(),
       WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
                                                 frames_per_buffer_),
-      // If DUCKING is enabled on the source, it needs to be enabled on the
-      // sink as well.
-      source_params_.effects() | implicit_ducking_effect);
+      source_params_.effects());
 
   {
     // Note: The max buffer is fairly large, but will rarely be used.
     // Cast needs the buffer to hold at least one second of audio.
     // The clock accuracy is set to 20ms because clock accuracy is
     // ~15ms on windows.
     media::AudioShifter* const new_shifter = new media::AudioShifter(
         base::TimeDelta::FromSeconds(2),
         base::TimeDelta::FromMilliseconds(20),
         base::TimeDelta::FromSeconds(20),
@@ skipping 12 matching lines @@
   if (sink_started_) {
     sink_->Stop();
     sink_started_ = false;
   }
 
   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
   MaybeStartSink();
 }
 
 }  // namespace content