Index: content/renderer/media/webrtc/media_stream_remote_audio_track.h |
diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.h b/content/renderer/media/webrtc/media_stream_remote_audio_track.h |
deleted file mode 100644 |
index 9e48dfb40d7350b241d8a6db9466aed2272ec5f3..0000000000000000000000000000000000000000 |
--- a/content/renderer/media/webrtc/media_stream_remote_audio_track.h |
+++ /dev/null |
@@ -1,89 +0,0 @@ |
-// Copyright 2015 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_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ |
-#define CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ |
- |
-#include "base/memory/ref_counted.h" |
-#include "base/threading/thread_checker.h" |
-#include "content/renderer/media/media_stream_audio_track.h" |
-#include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
- |
-namespace content { |
- |
-class MediaStreamRemoteAudioSource; |
- |
-// MediaStreamRemoteAudioTrack is a WebRTC specific implementation of an |
-// audio track received from a PeerConnection. |
-// TODO(tommi): Chrome shouldn't have to care about remote vs local so |
-// we should have a single track implementation that delegates to the |
-// sources that do different things depending on the type of source. |
-class MediaStreamRemoteAudioTrack : public MediaStreamAudioTrack { |
- public: |
- explicit MediaStreamRemoteAudioTrack( |
- const blink::WebMediaStreamSource& source, bool enabled); |
- ~MediaStreamRemoteAudioTrack() override; |
- |
- // MediaStreamTrack override. |
- void SetEnabled(bool enabled) override; |
- |
- // MediaStreamAudioTrack overrides. |
- void AddSink(MediaStreamAudioSink* sink) override; |
- void RemoveSink(MediaStreamAudioSink* sink) override; |
- media::AudioParameters GetOutputFormat() const override; |
- |
- webrtc::AudioTrackInterface* GetAudioAdapter() override; |
- |
- private: |
- // MediaStreamAudioTrack override. |
- void OnStop() final; |
- |
- MediaStreamRemoteAudioSource* source() const; |
- |
- blink::WebMediaStreamSource source_; |
- bool enabled_; |
-}; |
- |
-// Inheriting from ExtraData directly since MediaStreamAudioSource has |
-// too much unrelated bloat. |
-// TODO(tommi): MediaStreamAudioSource needs refactoring. |
-// TODO(miu): On it! ;-) |
-class MediaStreamRemoteAudioSource |
- : public blink::WebMediaStreamSource::ExtraData { |
- public: |
- explicit MediaStreamRemoteAudioSource( |
- const scoped_refptr<webrtc::AudioTrackInterface>& track); |
- ~MediaStreamRemoteAudioSource() override; |
- |
- // Controls whether or not the source is included in the main, mixed, audio |
- // output from WebRTC as rendered by WebRtcAudioRenderer (media players). |
- void SetEnabledForMixing(bool enabled); |
- |
- // Adds an audio sink for a track belonging to this source. |
- // |enabled| is the enabled state of the track and can be updated via |
- // a call to SetSinksEnabled. |
- void AddSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, |
- bool enabled); |
- |
- // Removes an audio sink for a track belonging to this source. |
- void RemoveSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track); |
- |
- // Turns audio callbacks on/off for all sinks belonging to a track. |
- void SetSinksEnabled(MediaStreamAudioTrack* track, bool enabled); |
- |
- // Removes all sinks belonging to a track. |
- void RemoveAll(MediaStreamAudioTrack* track); |
- |
- webrtc::AudioTrackInterface* GetAudioAdapter(); |
- |
- private: |
- class AudioSink; |
- std::unique_ptr<AudioSink> sink_; |
- const scoped_refptr<webrtc::AudioTrackInterface> track_; |
- base::ThreadChecker thread_checker_; |
-}; |
- |
-} // namespace content |
- |
-#endif // CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ |