Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
49 lines
1.5 KiB
C++
49 lines
1.5 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_MEDIA_STREAM_OBSERVER_H_
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#define PC_MEDIA_STREAM_OBSERVER_H_
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#include "api/media_stream_interface.h"
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#include "rtc_base/scoped_ref_ptr.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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namespace webrtc {
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// Helper class which will listen for changes to a stream and emit the
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// corresponding signals.
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class MediaStreamObserver : public ObserverInterface {
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public:
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explicit MediaStreamObserver(MediaStreamInterface* stream);
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~MediaStreamObserver() override;
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const MediaStreamInterface* stream() const { return stream_; }
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void OnChanged() override;
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sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
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SignalAudioTrackAdded;
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sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
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SignalAudioTrackRemoved;
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sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
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SignalVideoTrackAdded;
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sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
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SignalVideoTrackRemoved;
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private:
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rtc::scoped_refptr<MediaStreamInterface> stream_;
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AudioTrackVector cached_audio_tracks_;
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VideoTrackVector cached_video_tracks_;
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};
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} // namespace webrtc
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#endif // PC_MEDIA_STREAM_OBSERVER_H_
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