Files
platform-external-webrtc/webrtc/api/rtpreceiver.h
zhihuang 184a3fd648 Forward the SignalFirstPacketReceived to RtpReceiver.
The RtpReceiverObserverInterface is created.
The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession.
WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface.
The application can listen to the Signal by implementing and registering a RtpReceiverObserver.

Review-Url: https://codereview.webrtc.org/1999853002
Cr-Commit-Position: refs/heads/master@{#13139}
2016-06-14 18:47:20 +00:00

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_