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_ |