rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/api/rtpreceiver.h

141 lines
4.4 KiB
C++

/*
* 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 <string>
#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<RtpReceiverInternal>,
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<AudioTrackInterface> audio_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> 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<AudioTrackInterface> track_;
bool cached_track_enabled_;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
};
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
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<VideoTrackInterface> video_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> 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<VideoTrackSource> source_;
rtc::scoped_refptr<VideoTrackInterface> track_;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
};
} // namespace webrtc
#endif // WEBRTC_API_RTPRECEIVER_H_