webrtc_m130/webrtc/pc/rtpreceiver.h
hbos 8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 14:39:05 +00:00

170 lines
5.4 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 cricket::VoiceChannel/cricket::VideoChannel)
#ifndef WEBRTC_PC_RTPRECEIVER_H_
#define WEBRTC_PC_RTPRECEIVER_H_
#include <stdint.h>
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/media/base/videobroadcaster.h"
#include "webrtc/pc/channel.h"
#include "webrtc/pc/remoteaudiosource.h"
#include "webrtc/pc/videotracksource.h"
namespace webrtc {
// Internal class used by PeerConnection.
class RtpReceiverInternal : public RtpReceiverInterface {
public:
virtual void Stop() = 0;
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine2, WebRtcVoiceEngine layer.
virtual uint32_t ssrc() const = 0;
};
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public rtc::RefCountedObject<RtpReceiverInternal>,
public sigslot::has_slots<> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
// TODO(deadbeef): Use rtc::Optional, or have another constructor that
// doesn't take an SSRC, and make this one DCHECK(ssrc != 0).
AudioRtpReceiver(const std::string& track_id,
uint32_t ssrc,
cricket::VoiceChannel* channel);
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();
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
// RtpReceiverInternal implementation.
void Stop() override;
uint32_t ssrc() const override { return ssrc_; }
void SetObserver(RtpReceiverObserverInterface* observer) override;
// Does not take ownership.
// Should call SetChannel(nullptr) before |channel| is destroyed.
void SetChannel(cricket::VoiceChannel* channel);
std::vector<RtpSource> GetSources() const override;
private:
void Reconfigure();
void OnFirstPacketReceived(cricket::BaseChannel* channel);
const std::string id_;
const uint32_t ssrc_;
cricket::VoiceChannel* channel_;
const rtc::scoped_refptr<AudioTrackInterface> track_;
bool cached_track_enabled_;
double cached_volume_ = 1;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
};
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
public sigslot::has_slots<> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
VideoRtpReceiver(const std::string& track_id,
rtc::Thread* worker_thread,
uint32_t ssrc,
cricket::VideoChannel* channel);
virtual ~VideoRtpReceiver();
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
bool SetParameters(const RtpParameters& parameters) override;
// RtpReceiverInternal implementation.
void Stop() override;
uint32_t ssrc() const override { return ssrc_; }
void SetObserver(RtpReceiverObserverInterface* observer) override;
// Does not take ownership.
// Should call SetChannel(nullptr) before |channel| is destroyed.
void SetChannel(cricket::VideoChannel* channel);
private:
void OnFirstPacketReceived(cricket::BaseChannel* channel);
std::string id_;
uint32_t ssrc_;
cricket::VideoChannel* channel_;
// |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_;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
};
} // namespace webrtc
#endif // WEBRTC_PC_RTPRECEIVER_H_