MediaStream audio rendering: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_local_audio_renderer.h

Index: content/renderer/media/webrtc_local_audio_renderer.h
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h
deleted file mode 100644
index d33c384975002ab70473339e02d6d543f27103b8..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_renderer.h
+++ /dev/null
@@ -1,183 +0,0 @@
-// 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_LOCAL_AUDIO_RENDERER_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
-
-#include <stdint.h>
-
-#include <string>
-#include <vector>
-
-#include "base/callback.h"
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/single_thread_task_runner.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "content/common/content_export.h"
-#include "content/public/renderer/media_stream_audio_renderer.h"
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "media/base/output_device.h"
-#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
-
-namespace media {
-class AudioBus;
-class AudioShifter;
-class AudioOutputDevice;
-class AudioParameters;
-}
-
-namespace content {
-
-class WebRtcAudioCapturer;
-
-// WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
-// local audio media stream tracks,
-// http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
-// It also implements media::AudioRendererSink::RenderCallback to render audio
-// data provided from a WebRtcLocalAudioTrack source.
-// When the audio layer in the browser process asks for data to render, this
-// class provides the data by implementing the MediaStreamAudioSink
-// interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
-// TODO(henrika): improve by using similar principles as in
-// MediaStreamVideoRendererSink which register itself to the video track when
-// the provider is started and deregisters itself when it is stopped. Tracking
-// this at http://crbug.com/164813.
-class CONTENT_EXPORT WebRtcLocalAudioRenderer
-    : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
-      NON_EXPORTED_BASE(public MediaStreamAudioSink),
-      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
-      NON_EXPORTED_BASE(public media::OutputDevice) {
- public:
-  // Creates a local renderer and registers a capturing |source| object.
-  // The |source| is owned by the WebRtcAudioDeviceImpl.
-  // Called on the main thread.
-  WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track,
-                           int source_render_frame_id,
-                           int session_id,
-                           const std::string& device_id,
-                           const url::Origin& security_origin);
-
-  // MediaStreamAudioRenderer implementation.
-  // Called on the main thread.
-  void Start() override;
-  void Stop() override;
-  void Play() override;
-  void Pause() override;
-  void SetVolume(float volume) override;
-  media::OutputDevice* GetOutputDevice() 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 base::TimeDelta& total_render_time() const {
-    return total_render_time_;
-  }
-
- protected:
-  ~WebRtcLocalAudioRenderer() override;
-
- private:
-  // MediaStreamAudioSink implementation.
-
-  // Called on the AudioInputDevice worker thread.
-  void OnData(const media::AudioBus& audio_bus,
-              base::TimeTicks estimated_capture_time) override;
-
-  // Called on the AudioInputDevice worker thread.
-  void OnSetFormat(const media::AudioParameters& params) override;
-
-  // media::AudioRendererSink::RenderCallback implementation.
-  // Render() is called on the AudioOutputDevice thread and OnRenderError()
-  // on the IO thread.
-  int Render(media::AudioBus* audio_bus,
-             uint32_t audio_delay_milliseconds,
-             uint32_t frames_skipped) override;
-  void OnRenderError() override;
-
-  // Initializes and starts the |sink_| if
-  //  we have received valid |source_params_| &&
-  //  |playing_| has been set to true &&
-  //  |volume_| is not zero.
-  void MaybeStartSink();
-
-  // Sets new |source_params_| and then re-initializes and restarts |sink_|.
-  void ReconfigureSink(const media::AudioParameters& params);
-
-  // The audio track which provides data to render. Given that this class
-  // implements local loopback, the audio track is getting data from a capture
-  // instance like a selected microphone and forwards the recorded data to its
-  // sinks. The recorded data is stored in a FIFO and consumed
-  // by this class when the sink asks for new data.
-  // This class is calling MediaStreamAudioSink::AddToAudioTrack() and
-  // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect
-  // with the audio track.
-  blink::WebMediaStreamTrack audio_track_;
-
-  // The render view and frame in which the audio is rendered into |sink_|.
-  const int source_render_frame_id_;
-  const int session_id_;
-
-  // MessageLoop associated with the single thread that performs all control
-  // tasks.  Set to the MessageLoop that invoked the ctor.
-  const scoped_refptr<base::SingleThreadTaskRunner> task_runner_;
-
-  // The sink (destination) for rendered audio.
-  scoped_refptr<media::AudioOutputDevice> sink_;
-
-  // This does all the synchronization/resampling/smoothing.
-  scoped_ptr<media::AudioShifter> audio_shifter_;
-
-  // Stores last time a render callback was received. The time difference
-  // between a new time stamp and this value can be used to derive the
-  // total render time.
-  base::TimeTicks last_render_time_;
-
-  // Keeps track of total time audio has been rendered.
-  base::TimeDelta total_render_time_;
-
-  // The audio parameters of the capture source.
-  // Must only be touched on the main thread.
-  media::AudioParameters source_params_;
-
-  // The audio parameters used by the sink.
-  // Must only be touched on the main thread.
-  media::AudioParameters sink_params_;
-
-  // Set when playing, cleared when paused.
-  bool playing_;
-
-  // Protects |audio_shifter_|, |playing_|, |last_render_time_|,
-  // |total_render_time_| and |volume_|.
-  mutable base::Lock thread_lock_;
-
-  // The preferred device id of the output device or empty for the default
-  // output device.
-  std::string output_device_id_;
-  url::Origin security_origin_;
-
-  // Cache value for the volume.
-  float volume_;
-
-  // Flag to indicate whether |sink_| has been started yet.
-  bool sink_started_;
-
-  // Used to DCHECK that some methods are called on the capture audio thread.
-  base::ThreadChecker capture_thread_checker_;
-
-  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_