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 PC_RTPRECEIVER_H_
#define 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 <vector>
#include "api/mediastreaminterface.h"
#include "api/rtpreceiverinterface.h"
#include "media/base/videobroadcaster.h"
#include "pc/remoteaudiosource.h"
#include "pc/videotracksource.h"
namespace webrtc {
// Internal class used by PeerConnection.
class RtpReceiverInternal : public RtpReceiverInterface {
public:
virtual void Stop() = 0;
// Sets the underlying MediaEngine channel associated with this RtpSender.
// SetVoiceMediaChannel should be used for audio RtpSenders and
// SetVideoMediaChannel should be used for video RtpSenders. Must call the
// appropriate SetXxxMediaChannel(nullptr) before the media channel is
// destroyed.
virtual void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) = 0;
virtual void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) = 0;
// Configures the RtpReceiver with the underlying media channel, with the
// given SSRC as the stream identifier. If |ssrc| is 0, the receiver will
// receive packets on unsignaled SSRCs.
virtual void SetupMediaChannel(uint32_t ssrc) = 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;
// Call this to notify the RtpReceiver when the first packet has been received
// on the corresponding channel.
virtual void NotifyFirstPacketReceived() = 0;
// Set the associated remote media streams for this receiver. The remote track
// will be removed from any streams that are no longer present and added to
// any new streams.
virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
// TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
// set_stream_ids() as soon as downstream projects are no longer dependent on
// stream objects.
virtual void SetStreams(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
// Returns an ID that changes if the attached track changes, but
// otherwise remains constant. Used to generate IDs for stats.
// The special value zero means that no track is attached.
virtual int AttachmentId() const = 0;
};
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public rtc::RefCountedObject<RtpReceiverInternal> {
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
rtc::Thread* worker_thread,
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
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::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
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;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override;
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override {
RTC_NOTREACHED();
}
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;
int AttachmentId() const override { return attachment_id_; }
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
private:
void Reconfigure();
bool SetOutputVolume(double volume);
rtc::Thread* const worker_thread_;
const std::string id_;
const rtc::scoped_refptr<RemoteAudioSource> source_;
const rtc::scoped_refptr<AudioTrackInterface> track_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool cached_track_enabled_;
double cached_volume_ = 1;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
};
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees.
VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> streams_ids);
// TODO(hbos): Remove this when streams() is removed.
// https://crbug.com/webrtc/9480
VideoRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
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::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
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;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetVoiceMediaChannel(
cricket::VoiceMediaChannel* voice_media_channel) override {
RTC_NOTREACHED();
}
void SetVideoMediaChannel(
cricket::VideoMediaChannel* video_media_channel) override;
int AttachmentId() const override { return attachment_id_; }
std::vector<RtpSource> GetSources() const override;
private:
class VideoRtpTrackSource : public VideoTrackSource {
public:
VideoRtpTrackSource() : VideoTrackSource(true /* remote */) {}
rtc::VideoSourceInterface<VideoFrame>* source() override {
return &broadcaster_;
}
rtc::VideoSinkInterface<VideoFrame>* sink() { return &broadcaster_; }
private:
// |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_;
};
bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
rtc::Thread* const worker_thread_;
const std::string id_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
// |source_| is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
rtc::scoped_refptr<VideoRtpTrackSource> source_;
rtc::scoped_refptr<VideoTrackInterface> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool stopped_ = false;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
};
} // namespace webrtc
#endif // PC_RTPRECEIVER_H_