db0cd9e774
This is similar to how a "receive" method is used to apply RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters doesn't allow changing the parameters, so the main use of the API is to retrieve the set of configured codecs. But other uses will likely be made possible in the future. R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1917193008 . Cr-Commit-Position: refs/heads/master@{#12761}
115 lines
3.5 KiB
C++
115 lines
3.5 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/media/base/videobroadcaster.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(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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void Stop() override;
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RtpParameters GetParameters() const override;
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bool SetParameters(const RtpParameters& parameters) override;
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private:
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void Reconfigure();
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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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};
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class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
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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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void Stop() override;
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RtpParameters GetParameters() const override;
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bool SetParameters(const RtpParameters& parameters) override;
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private:
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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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};
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} // namespace webrtc
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#endif // WEBRTC_API_RTPRECEIVER_H_
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