Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(35)

Unified Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
index cfe4a98ef9d291d96e0746833ecd3edcf2dab27e..5adc85a07ea6658ba3cf59428393c742539577c8 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
@@ -11,13 +11,13 @@
#include "base/memory/scoped_vector.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/renderer/media/media_stream_audio_level_calculator.h"
#include "third_party/webrtc/api/mediastreamtrack.h"
-#include "third_party/webrtc/media/base/audiorenderer.h"
+#include "third_party/webrtc/base/refcount.h"
-namespace cricket {
-class AudioRenderer;
+namespace base {
+class WaitableEvent;
}
namespace webrtc {
@@ -28,80 +28,107 @@ class AudioProcessorInterface;
namespace content {
class MediaStreamAudioProcessor;
+class MediaStreamAudioTrack;
class WebRtcAudioSinkAdapter;
-class WebRtcLocalAudioTrack;
+// Provides an implementation of the webrtc::AudioTrackInterface that can be
+// bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an
+// adapter that sits between the media stream object graph and WebRtc's object
+// graph and proxies between the two.
+//
+// TODO(miu): Rename to WebRtcAudioTrackAdapter since this can be used to proxy
+// between any kind (local or remote) of MediaStreamAudioTrack.
class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
: NON_EXPORTED_BASE(
public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
public:
static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
const std::string& label,
- webrtc::AudioSourceInterface* track_source);
-
- WebRtcLocalAudioTrackAdapter(
- const std::string& label,
webrtc::AudioSourceInterface* track_source,
- const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread);
-
- ~WebRtcLocalAudioTrackAdapter() override;
-
- void Initialize(WebRtcLocalAudioTrack* owner);
+ const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner);
- // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
- // level of the audio data.
- void SetSignalLevel(int signal_level);
+ // Set the |track| that manages the MediaStreamAudioSinks. It is the client's
+ // responsibility to call this method with null before the track's destruction
+ // time. This is needed because WebRtcLocalAudioTrackAdapter is ref-counted
+ // and could potentially out-live |track|.
+ //
+ // This method must only be called on the main thread.
+ void SetMediaStreamAudioTrack(MediaStreamAudioTrack* track);
- // Method called by the WebRtcLocalAudioTrack to set the processor that
- // applies signal processing on the data of the track.
+ // Set the processor that applies signal processing on the data of the track.
// This class will keep a reference of the |processor|.
// Called on the main render thread.
void SetAudioProcessor(
const scoped_refptr<MediaStreamAudioProcessor>& processor);
+ // Set the object that provides shared access to the current audio signal
+ // level.
+ void SetReportedLevel(
+ const scoped_refptr<MediaStreamAudioLevelCalculator::ReportedLevel>&
+ reported_level);
+
// webrtc::MediaStreamTrack implementation.
- std::string kind() const override;
- bool set_enabled(bool enable) override;
+ std::string kind() const final;
+ bool set_enabled(bool enable) final;
+
+ protected:
+ friend class rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>;
+
+ WebRtcLocalAudioTrackAdapter(
+ const std::string& label,
+ webrtc::AudioSourceInterface* track_source,
+ const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner);
+
+ ~WebRtcLocalAudioTrackAdapter() override;
private:
- // webrtc::AudioTrackInterface implementation.
- void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
- void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
- bool GetSignalLevel(int* level) override;
- rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
- override;
- webrtc::AudioSourceInterface* GetSource() const override;
+ // Removes the |sink| from |track_| and then signals the |done_event| (if
+ // provided). This is used by RemoveSink() to ensure the audio flow has
+ // halted before it returns (on the signaling thread).
+ void RemoveSinkOnMainThread(webrtc::AudioTrackSinkInterface* sink,
+ base::WaitableEvent* done_event);
- // Weak reference.
- WebRtcLocalAudioTrack* owner_;
+ // webrtc::AudioTrackInterface implementation.
+ void AddSink(webrtc::AudioTrackSinkInterface* sink) final;
+ void RemoveSink(webrtc::AudioTrackSinkInterface* sink) final;
+ bool GetSignalLevel(int* level) final;
+ rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() final;
+ webrtc::AudioSourceInterface* GetSource() const final;
// The source of the audio track which handles the audio constraints.
- // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
- rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
+ const rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
- // Libjingle's signaling thread.
- const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
+ // Task runner for operations that must be done on libjingle's signaling
+ // thread. May be null for single-threaded unit tests.
+ const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
- // The audio processsor that applies audio processing on the data of audio
- // track.
- scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
+ // Task runner for operations that must be done on the main thread. May be
+ // null for single-threaded unit tests.
+ scoped_refptr<base::SingleThreadTaskRunner> main_task_runner_;
- // A vector of WebRtc VoE channels that the capturer sends data to.
- std::vector<int> voe_channels_;
+ // The track to add/remove sinks to/from. When the
+ // webrtc::AudioTrackInterface::Add/RemoveSink() methods are called, they
+ // create a proxy that implements the MediaStreamAudioSink interface to call
+ // into the webrtc::AudioTrackSinkInterface. This must only be accessed on
+ // the main thread.
+ MediaStreamAudioTrack* track_;
// A vector of the peer connection sink adapters which receive the audio data
- // from the audio track.
+ // from the audio track. This must only be accessed on the main thread.
ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
- // The amplitude of the signal.
- int signal_level_;
+ // Protects |audio_processor_|, |voe_channels_|, and |signal_level_|.
+ mutable base::Lock lock_;
+
+ // The audio processsor that applies audio processing on the data of audio
+ // track.
+ scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
- // Thread checker for libjingle's signaling thread.
- base::ThreadChecker signaling_thread_checker_;
- base::ThreadChecker capture_thread_;
+ // Thread-safe accessor to current audio signal level.
+ scoped_refptr<MediaStreamAudioLevelCalculator::ReportedLevel> reported_level_;
- // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
- mutable base::Lock lock_;
+ // A vector of WebRtc VoE channels that the capturer sends data to.
+ std::vector<int> voe_channels_;
};
} // namespace content

Powered by Google App Engine
This is Rietveld 408576698