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

content/renderer/media/webrtc_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_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 113 matching lines @@
 
  private:
   base::ThreadChecker thread_checker_;
   const scoped_refptr<MediaStreamAudioRenderer> delegate_;
   const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
   bool started_;
   WebRtcAudioRenderer::PlayingState playing_state_;
   OnPlayStateChanged on_play_state_changed_;
 };
 
-// Returns either AudioParameters::NO_EFFECTS or AudioParameters::DUCKING
-// depending on whether or not an input element is currently open with
-// ducking enabled.
-int GetCurrentDuckingFlag(int render_frame_id) {
-  RenderFrameImpl* const frame =
-      RenderFrameImpl::FromRoutingID(render_frame_id);
-  MediaStreamDispatcher* const dispatcher = frame ?
-      frame->GetMediaStreamDispatcher() : NULL;
-  if (dispatcher && dispatcher->IsAudioDuckingActive()) {
-    return media::AudioParameters::DUCKING;
-  }
-
-  return media::AudioParameters::NO_EFFECTS;
-}
-
 }  // namespace
 
 int WebRtcAudioRenderer::GetOptimalBufferSize(int sample_rate,
                                               int hardware_buffer_size) {
   // Use native hardware buffer size as default. On Windows, we strive to open
   // up using this native hardware buffer size to achieve best
   // possible performance and to ensure that no FIFO is needed on the browser
   // side to match the client request. That is why there is no #if case for
   // Windows below.
   int frames_per_buffer = hardware_buffer_size;
@@ skipping 34 matching lines @@
       session_id_(session_id),
       signaling_thread_(signaling_thread),
       media_stream_(media_stream),
       source_(NULL),
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
       fifo_delay_milliseconds_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
-                   frames_per_buffer,
-                   GetCurrentDuckingFlag(source_render_frame_id)),
+                   frames_per_buffer),
       render_callback_count_(0) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_frame_id=%d"
       ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i",
       source_render_frame_id, session_id, sample_rate, frames_per_buffer,
       sink_params_.effects()));
 }
 
 WebRtcAudioRenderer::~WebRtcAudioRenderer() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 354 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content