2015-09-24 16:47:53 -07:00
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/*
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* libjingle
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* Copyright 2015 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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2015-09-28 16:53:55 -07:00
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// This file contains classes that implement RtpReceiverInterface.
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// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
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// transport (provided by AudioProviderInterface/VideoProviderInterface)
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#ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_
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#define TALK_APP_WEBRTC_RTPRECEIVER_H_
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#include <string>
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#include "talk/app/webrtc/mediastreamprovider.h"
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#include "talk/app/webrtc/rtpreceiverinterface.h"
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#include "webrtc/base/basictypes.h"
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namespace webrtc {
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class AudioRtpReceiver : public ObserverInterface,
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public AudioSourceInterface::AudioObserver,
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public rtc::RefCountedObject<RtpReceiverInterface> {
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public:
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AudioRtpReceiver(AudioTrackInterface* track,
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uint32 ssrc,
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AudioProviderInterface* provider);
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virtual ~AudioRtpReceiver();
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// ObserverInterface implementation
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void OnChanged() override;
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// AudioSourceInterface::AudioObserver implementation
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void OnSetVolume(double volume) override;
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// RtpReceiverInterface implementation
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_.get();
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}
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std::string id() const override { return id_; }
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void Stop() override;
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private:
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void Reconfigure();
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std::string id_;
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rtc::scoped_refptr<AudioTrackInterface> track_;
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uint32 ssrc_;
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AudioProviderInterface* provider_;
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bool cached_track_enabled_;
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};
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class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
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public:
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VideoRtpReceiver(VideoTrackInterface* track,
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uint32 ssrc,
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VideoProviderInterface* provider);
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virtual ~VideoRtpReceiver();
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// RtpReceiverInterface implementation
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_.get();
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}
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std::string id() const override { return id_; }
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void Stop() override;
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private:
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std::string id_;
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rtc::scoped_refptr<VideoTrackInterface> track_;
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uint32 ssrc_;
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VideoProviderInterface* provider_;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_RTPRECEIVER_H_
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