7324eb9e62
Reason for revert: Breaks GN in chromium. Original issue's description: > Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. > > webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is > depending on voice engine, resulting in a cyclic dependency (which we > don't detect since we have that check turned off, see webrtc:4243). > > BUG=webrtc:4243, webrtc:5589 > R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org > TBR=tommi@webrtc.org > > Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2 > Cr-Commit-Position: refs/heads/master@{#11766} TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4243, webrtc:5589 Review URL: https://codereview.webrtc.org/1739783002 Cr-Commit-Position: refs/heads/master@{#11769}
80 lines
2.3 KiB
C++
80 lines
2.3 KiB
C++
/*
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* Copyright 2014 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 WEBRTC_API_REMOTEAUDIOSOURCE_H_
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#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
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#include <list>
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#include <string>
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/notifier.h"
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#include "webrtc/audio/audio_sink.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/media/base/audiorenderer.h"
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namespace rtc {
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struct Message;
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class Thread;
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} // namespace rtc
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namespace webrtc {
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class AudioProviderInterface;
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// This class implements the audio source used by the remote audio track.
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class RemoteAudioSource : public Notifier<AudioSourceInterface> {
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public:
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// Creates an instance of RemoteAudioSource.
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static rtc::scoped_refptr<RemoteAudioSource> Create(
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uint32_t ssrc,
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AudioProviderInterface* provider);
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// MediaSourceInterface implementation.
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MediaSourceInterface::SourceState state() const override;
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bool remote() const override;
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void AddSink(AudioTrackSinkInterface* sink) override;
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void RemoveSink(AudioTrackSinkInterface* sink) override;
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protected:
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RemoteAudioSource();
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~RemoteAudioSource() override;
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// Post construction initialize where we can do things like save a reference
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// to ourselves (need to be fully constructed).
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void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
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private:
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typedef std::list<AudioObserver*> AudioObserverList;
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// AudioSourceInterface implementation.
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void SetVolume(double volume) override;
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void RegisterAudioObserver(AudioObserver* observer) override;
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void UnregisterAudioObserver(AudioObserver* observer) override;
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class Sink;
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void OnData(const AudioSinkInterface::Data& audio);
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void OnAudioProviderGone();
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class MessageHandler;
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void OnMessage(rtc::Message* msg);
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AudioObserverList audio_observers_;
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rtc::CriticalSection sink_lock_;
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std::list<AudioTrackSinkInterface*> sinks_;
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rtc::Thread* const main_thread_;
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SourceState state_;
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};
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} // namespace webrtc
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#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_
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