15583c19d7
The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545}
97 lines
3.3 KiB
C++
97 lines
3.3 KiB
C++
/*
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* libjingle
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* Copyright 2014 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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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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