1a7162dbc9
Reason for revert: Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland. Original issue's description: > Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. > > This eliminates the need for the extra layer of indirection provided by > mediastreamprovider.h. It will thus make it easier to implement new > functionality in RtpSender/RtpReceiver. > > It also brings us one step closer to the end goal of combining "senders" > and "send streams". Currently the sender still needs to go through the > BaseChannel and MediaChannel, using an SSRC as a key. > > R=pthatcher@webrtc.org > > Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c > Cr-Commit-Position: refs/heads/master@{#13285} TBR=pthatcher@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2099843003 Cr-Commit-Position: refs/heads/master@{#13286}
141 lines
4.4 KiB
C++
141 lines
4.4 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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// 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 WEBRTC_API_RTPRECEIVER_H_
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#define WEBRTC_API_RTPRECEIVER_H_
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#include <string>
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#include "webrtc/api/mediastreamprovider.h"
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#include "webrtc/api/rtpreceiverinterface.h"
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#include "webrtc/api/remoteaudiosource.h"
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#include "webrtc/api/videotracksource.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/media/base/videobroadcaster.h"
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namespace webrtc {
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// Internal class used by PeerConnection.
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class RtpReceiverInternal : public RtpReceiverInterface {
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public:
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virtual void Stop() = 0;
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};
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class AudioRtpReceiver : public ObserverInterface,
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public AudioSourceInterface::AudioObserver,
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public rtc::RefCountedObject<RtpReceiverInternal>,
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public sigslot::has_slots<> {
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public:
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AudioRtpReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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uint32_t 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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rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
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return track_.get();
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}
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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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RtpParameters GetParameters() const override;
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bool SetParameters(const RtpParameters& parameters) override;
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// RtpReceiverInternal implementation.
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void Stop() override;
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void SetObserver(RtpReceiverObserverInterface* observer) override;
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cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; }
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private:
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void Reconfigure();
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void OnFirstAudioPacketReceived();
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const std::string id_;
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const uint32_t ssrc_;
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AudioProviderInterface* provider_; // Set to null in Stop().
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const rtc::scoped_refptr<AudioTrackInterface> track_;
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bool cached_track_enabled_;
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RtpReceiverObserverInterface* observer_ = nullptr;
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bool received_first_packet_ = false;
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};
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class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
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public sigslot::has_slots<> {
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public:
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VideoRtpReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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rtc::Thread* worker_thread,
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uint32_t ssrc,
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VideoProviderInterface* provider);
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virtual ~VideoRtpReceiver();
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rtc::scoped_refptr<VideoTrackInterface> video_track() const {
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return track_.get();
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}
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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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RtpParameters GetParameters() const override;
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bool SetParameters(const RtpParameters& parameters) override;
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// RtpReceiverInternal implementation.
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void Stop() override;
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void SetObserver(RtpReceiverObserverInterface* observer) override;
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cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
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private:
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void OnFirstVideoPacketReceived();
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std::string id_;
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uint32_t ssrc_;
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VideoProviderInterface* provider_;
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// |broadcaster_| is needed since the decoder can only handle one sink.
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// It might be better if the decoder can handle multiple sinks and consider
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// the VideoSinkWants.
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rtc::VideoBroadcaster broadcaster_;
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// |source_| is held here to be able to change the state of the source when
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// the VideoRtpReceiver is stopped.
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rtc::scoped_refptr<VideoTrackSource> source_;
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rtc::scoped_refptr<VideoTrackInterface> track_;
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RtpReceiverObserverInterface* observer_ = nullptr;
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bool received_first_packet_ = false;
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};
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} // namespace webrtc
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#endif // WEBRTC_API_RTPRECEIVER_H_
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