/* * Copyright 2015 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ // This file contains classes that implement RtpReceiverInterface. // An RtpReceiver associates a MediaStreamTrackInterface with an underlying // transport (provided by AudioProviderInterface/VideoProviderInterface) #ifndef WEBRTC_API_RTPRECEIVER_H_ #define WEBRTC_API_RTPRECEIVER_H_ #include #include "webrtc/api/mediastreamprovider.h" #include "webrtc/api/rtpreceiverinterface.h" #include "webrtc/api/remoteaudiosource.h" #include "webrtc/api/videotracksource.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/sigslot.h" #include "webrtc/media/base/videobroadcaster.h" namespace webrtc { // Internal class used by PeerConnection. class RtpReceiverInternal : public RtpReceiverInterface { public: virtual void Stop() = 0; }; class AudioRtpReceiver : public ObserverInterface, public AudioSourceInterface::AudioObserver, public rtc::RefCountedObject, public sigslot::has_slots<> { public: AudioRtpReceiver(MediaStreamInterface* stream, const std::string& track_id, uint32_t ssrc, AudioProviderInterface* provider); virtual ~AudioRtpReceiver(); // ObserverInterface implementation void OnChanged() override; // AudioSourceInterface::AudioObserver implementation void OnSetVolume(double volume) override; rtc::scoped_refptr audio_track() const { return track_.get(); } // RtpReceiverInterface implementation rtc::scoped_refptr track() const override { return track_.get(); } std::string id() const override { return id_; } RtpParameters GetParameters() const override; bool SetParameters(const RtpParameters& parameters) override; // RtpReceiverInternal implementation. void Stop() override; void SetObserver(RtpReceiverObserverInterface* observer) override; cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; } private: void Reconfigure(); void OnFirstAudioPacketReceived(); const std::string id_; const uint32_t ssrc_; AudioProviderInterface* provider_; // Set to null in Stop(). const rtc::scoped_refptr track_; bool cached_track_enabled_; RtpReceiverObserverInterface* observer_ = nullptr; bool received_first_packet_ = false; }; class VideoRtpReceiver : public rtc::RefCountedObject, public sigslot::has_slots<> { public: VideoRtpReceiver(MediaStreamInterface* stream, const std::string& track_id, rtc::Thread* worker_thread, uint32_t ssrc, VideoProviderInterface* provider); virtual ~VideoRtpReceiver(); rtc::scoped_refptr video_track() const { return track_.get(); } // RtpReceiverInterface implementation rtc::scoped_refptr track() const override { return track_.get(); } std::string id() const override { return id_; } RtpParameters GetParameters() const override; bool SetParameters(const RtpParameters& parameters) override; // RtpReceiverInternal implementation. void Stop() override; void SetObserver(RtpReceiverObserverInterface* observer) override; cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; } private: void OnFirstVideoPacketReceived(); std::string id_; uint32_t ssrc_; VideoProviderInterface* provider_; // |broadcaster_| is needed since the decoder can only handle one sink. // It might be better if the decoder can handle multiple sinks and consider // the VideoSinkWants. rtc::VideoBroadcaster broadcaster_; // |source_| is held here to be able to change the state of the source when // the VideoRtpReceiver is stopped. rtc::scoped_refptr source_; rtc::scoped_refptr track_; RtpReceiverObserverInterface* observer_ = nullptr; bool received_first_packet_ = false; }; } // namespace webrtc #endif // WEBRTC_API_RTPRECEIVER_H_