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 |
deleted file mode 100644 |
index 72b80194b08ed09a01673c98c4ed9816aa4e6d74..0000000000000000000000000000000000000000 |
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h |
+++ /dev/null |
@@ -1,107 +0,0 @@ |
-// Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
-#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
- |
-#include <vector> |
- |
-#include "base/memory/ref_counted.h" |
-#include "base/memory/scoped_vector.h" |
-#include "base/single_thread_task_runner.h" |
-#include "base/synchronization/lock.h" |
-#include "content/common/content_export.h" |
-#include "content/renderer/media/media_stream_audio_level_calculator.h" |
-#include "content/renderer/media/media_stream_audio_processor.h" |
-#include "third_party/webrtc/api/mediastreamtrack.h" |
-#include "third_party/webrtc/media/base/audiorenderer.h" |
- |
-namespace cricket { |
-class AudioRenderer; |
-} |
- |
-namespace webrtc { |
-class AudioSourceInterface; |
-class AudioProcessorInterface; |
-} |
- |
-namespace content { |
- |
-class MediaStreamAudioProcessor; |
-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. |
-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, |
- scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner); |
- |
- ~WebRtcLocalAudioTrackAdapter() override; |
- |
- void Initialize(WebRtcLocalAudioTrack* owner); |
- |
- // Set the object that provides shared access to the current audio signal |
- // level. This method may only be called once, before the audio data flow |
- // starts, and before any calls to GetSignalLevel() might be made. |
- void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level); |
- |
- // Method called by the WebRtcLocalAudioTrack to 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. |
- // This method may only be called once, before the audio data flow starts, and |
- // before any calls to GetAudioProcessor() might be made. |
- void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor); |
- |
- // webrtc::MediaStreamTrack implementation. |
- std::string kind() const override; |
- bool set_enabled(bool enable) 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; |
- |
- // Weak reference. |
- WebRtcLocalAudioTrack* owner_; |
- |
- // 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_; |
- |
- // Libjingle's signaling thread. |
- const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_; |
- |
- // The audio processsor that applies audio processing on the data of audio |
- // track. This must be set before calls to GetAudioProcessor() are made. |
- scoped_refptr<MediaStreamAudioProcessor> audio_processor_; |
- |
- // A vector of the peer connection sink adapters which receive the audio data |
- // from the audio track. |
- ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_; |
- |
- // Thread-safe accessor to current audio signal level. This must be set |
- // before calls to GetSignalLevel() are made. |
- scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_; |
-}; |
- |
-} // namespace content |
- |
-#endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |