Index: content/renderer/media/track_audio_renderer.h |
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/track_audio_renderer.h |
similarity index 52% |
rename from content/renderer/media/webrtc_local_audio_renderer.h |
rename to content/renderer/media/track_audio_renderer.h |
index d33c384975002ab70473339e02d6d543f27103b8..0703746be2383c92495ca3e48455c5d92b67c431 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.h |
+++ b/content/renderer/media/track_audio_renderer.h |
@@ -2,8 +2,8 @@ |
// 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_ |
+#ifndef CONTENT_RENDERER_MEDIA_TRACK_AUDIO_RENDERER_H_ |
+#define CONTENT_RENDERER_MEDIA_TRACK_AUDIO_RENDERER_H_ |
#include <stdint.h> |
@@ -19,8 +19,7 @@ |
#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/audio_renderer_sink.h" |
#include "media/base/output_device.h" |
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
@@ -33,34 +32,43 @@ 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 |
+// TrackAudioRenderer is a MediaStreamAudioRenderer for plumbing audio data |
+// generated from either local or remote (but not PeerConnection/WebRTC-sourced) |
+// MediaStreamAudioTracks to an audio output device, reconciling differences in |
+// the rates of production and consumption of the audio data. Note that remote |
+// PeerConnection-sourced tracks are NOT rendered by this implementation (see |
+// MediaStreamRendererFactoryImpl). |
+// |
+// This class uses AudioDeviceFactory to create media::AudioOutputDevices and |
+// owns/manages their lifecycles. Output devices are automatically re-created |
+// in response to audio format changes, or use of the SwitchOutputDevice() API |
+// by client code. |
+// |
+// Audio data is feed-in from the source via calls to OnData(). The |
+// internally-owned media::AudioOutputDevice calls Render() to pull-out that |
+// audio data. However, because of clock differences and other environmental |
+// factors, the audio will inevitably feed-in at a rate different from the rate |
+// it is being rendered-out. media::AudioShifter is used to buffer, stretch |
+// and skip audio to maintain time synchronization between the producer and |
+// consumer. |
+class CONTENT_EXPORT TrackAudioRenderer |
: 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. |
+ // Creates a renderer for the given |audio_track|. |playout_render_frame_id| |
+ // refers to the RenderFrame that owns this instance (e.g., it contains the |
+ // DOM widget representing the player). |session_id| and |device_id| are |
+ // optional, and are used to direct audio output to a pre-selected device; |
+ // otherwise, audio is output to the default device for the system. |
+ // |
// 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); |
+ TrackAudioRenderer(const blink::WebMediaStreamTrack& audio_track, |
+ int playout_render_frame_id, |
+ int session_id, |
+ const std::string& device_id, |
+ const url::Origin& security_origin); |
// MediaStreamAudioRenderer implementation. |
// Called on the main thread. |
@@ -80,19 +88,15 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
media::AudioParameters GetOutputParameters() override; |
media::OutputDeviceStatus GetDeviceStatus() override; |
- const base::TimeDelta& total_render_time() const { |
- return total_render_time_; |
- } |
- |
protected: |
- ~WebRtcLocalAudioRenderer() override; |
+ ~TrackAudioRenderer() override; |
private: |
// MediaStreamAudioSink implementation. |
// Called on the AudioInputDevice worker thread. |
void OnData(const media::AudioBus& audio_bus, |
- base::TimeTicks estimated_capture_time) override; |
+ base::TimeTicks reference_time) override; |
// Called on the AudioInputDevice worker thread. |
void OnSetFormat(const media::AudioParameters& params) override; |
@@ -107,25 +111,31 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
// Initializes and starts the |sink_| if |
// we have received valid |source_params_| && |
- // |playing_| has been set to true && |
- // |volume_| is not zero. |
+ // |playing_| has been set to true. |
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. |
+ // Creates a new AudioShifter, destroying the old one (if any). This is |
+ // called any time playback is started/stopped, or the sink changes. |
+ void CreateAudioShifter(); |
+ |
+ // Called when either the source or sink has changed somehow, or audio has |
+ // been paused. Drops the AudioShifter and updates |
+ // |prior_elapsed_render_time_|. May be called from either the main thread or |
+ // the audio thread. Assumption: |thread_lock_| is already acquired. |
+ void HaltAudioFlowWhileLockHeld(); |
+ |
+ // The audio track which provides access to the source data to render. |
+ // |
// 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 playout_render_frame_id_; |
const int session_id_; |
// MessageLoop associated with the single thread that performs all control |
@@ -138,27 +148,22 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
// 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_; |
+ // These track the time duration of all the audio rendered so far by this |
+ // instance. |prior_elapsed_render_time_| tracks the time duration of all |
+ // audio rendered before the last format change. |num_samples_rendered_| |
+ // tracks the number of audio samples rendered since the last format change. |
+ base::TimeDelta prior_elapsed_render_time_; |
+ int64_t num_samples_rendered_; |
- // Keeps track of total time audio has been rendered. |
- base::TimeDelta total_render_time_; |
- |
- // The audio parameters of the capture source. |
+ // The audio parameters of the track's 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_|. |
+ // Protects |audio_shifter_|, |prior_elapsed_render_time_|, and |
+ // |num_samples_rendered_|. |
mutable base::Lock thread_lock_; |
// The preferred device id of the output device or empty for the default |
@@ -166,18 +171,19 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
std::string output_device_id_; |
url::Origin security_origin_; |
- // Cache value for the volume. |
+ // Cache value for the volume. Whenever |sink_| is re-created, its volume |
+ // should be set to this. |
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_; |
+ // Used to DCHECK that some methods are called on the audio thread. |
+ base::ThreadChecker audio_thread_checker_; |
- DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); |
+ DISALLOW_COPY_AND_ASSIGN(TrackAudioRenderer); |
}; |
} // namespace content |
-#endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ |
+#endif // CONTENT_RENDERER_MEDIA_TRACK_AUDIO_RENDERER_H_ |