webrtc_m130/pc/rtpreceiver.h

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/*
* 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_
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ ) Reason for revert: Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used. Original issue's description: > Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ ) > > Reason for revert: > Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why. > > Original issue's description: > > Replace basictypes.h with stdint.h for int_t types. > > > > Removes basictypes.h for types that only makes use of it for fixed-size-int > > typedefs and replaces it with stdint.h. > > > > BUG=webrtc:6853 > > R=tommi@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2604043002 > > Cr-Commit-Position: refs/heads/master@{#15867} > > Committed: https://chromium.googlesource.com/external/webrtc/+/7fd1a753005ca93e8bd934a55808a2137b0ad84f > > TBR=tommi@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6853 > > Review-Url: https://codereview.webrtc.org/2603203003 > Cr-Commit-Position: refs/heads/master@{#15869} > Committed: https://chromium.googlesource.com/external/webrtc/+/7eb0e23bcf675635ef339a519a10563ebc9d93dc BUG=webrtc:6853 TBR=tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2609783002 Cr-Commit-Position: refs/heads/master@{#15873}
2017-01-02 08:42:32 -08:00
#include <stdint.h>
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/media/base/videobroadcaster.h"
#include "webrtc/pc/channel.h"
#include "webrtc/pc/remoteaudiosource.h"
#include "webrtc/pc/videotracksource.h"
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/rtc_base/sigslot.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 WebRtcVideoEngine, 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);
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: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > 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: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 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 07:39:05 -07:00
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_