Moving SwitchOutputDevice out of OutputDevice interface, eliminating OutputDevice

content/renderer/media/webrtc_audio_renderer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/threading/thread_checker.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_decoder.h"
 #include "media/base/audio_pull_fifo.h"
 #include "media/base/audio_renderer_sink.h"
 #include "media/base/channel_layout.h"
-#include "media/base/output_device.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 
 namespace webrtc {
 class AudioSourceInterface;
 }  // namespace webrtc
 
 namespace content {
 
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
 // for connecting WebRtc MediaStream with the audio pipeline.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
-      NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
-      NON_EXPORTED_BASE(public media::OutputDevice) {
+      NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
  public:
   // This is a little utility class that holds the configured state of an audio
   // stream.
   // It is used by both WebRtcAudioRenderer and SharedAudioRenderer (see cc
   // file) so a part of why it exists is to avoid code duplication and track
   // the state in the same way in WebRtcAudioRenderer and SharedAudioRenderer.
   class PlayingState : public base::NonThreadSafe {
    public:
     PlayingState() : playing_(false), volume_(1.0f) {}
 
@@ skipping 60 matching lines @@
 
  private:
   // MediaStreamAudioRenderer implementation.  This is private since we want
   // callers to use proxy objects.
   // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
   void Start() override;
   void Play() override;
   void Pause() override;
   void Stop() override;
   void SetVolume(float volume) override;
-  media::OutputDevice* GetOutputDevice() override;
+  media::OutputDeviceInfo GetOutputDeviceInfo() override;
   base::TimeDelta GetCurrentRenderTime() const override;
   bool IsLocalRenderer() const override;
-
-  // media::OutputDevice implementation
   void SwitchOutputDevice(const std::string& device_id,
                           const url::Origin& security_origin,
-                          const media::SwitchOutputDeviceCB& callback) override;
-  media::AudioParameters GetOutputParameters() override;
-  media::OutputDeviceStatus GetDeviceStatus() override;
+                          const media::OutputDeviceStatusCB& callback) override;
 
   // Called when an audio renderer, either the main or a proxy, starts playing.
   // Here we maintain a reference count of how many renderers are currently
   // playing so that the shared play state of all the streams can be reflected
   // correctly.
   void EnterPlayState();
 
   // Called when an audio renderer, either the main or a proxy, is paused.
   // See EnterPlayState for more details.
   void EnterPauseState();
@@ skipping 118 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_