Add RtpTransceiverInterface and implementing class
Introduces the public API interface corresponding to the standardized RtpTransceiver object in the WebRTC spec. https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver The RtpTransceiver will be the internal representation for both Plan B and Unified Plan SDP, but the public API interface will only support Unified Plan (existing users should continue to use GetSenders/GetReceivers, which will still be supported). Bug: webrtc:7600 Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7 Reviewed-on: https://webrtc-review.googlesource.com/21460 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20659}
This commit is contained in:
Steve Anton
@ -61,6 +61,7 @@ rtc_static_library("libjingle_peerconnection_api") {
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"rtpparameters.h",
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"rtpreceiverinterface.h",
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"rtpsenderinterface.h",
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"rtptransceiverinterface.h",
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"statstypes.cc",
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"statstypes.h",
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"turncustomizer.h",
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107
api/rtptransceiverinterface.h
Normal file
107
api/rtptransceiverinterface.h
Normal file
@ -0,0 +1,107 @@
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/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_RTPTRANSCEIVERINTERFACE_H_
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#define API_RTPTRANSCEIVERINTERFACE_H_
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#include <string>
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#include "api/optional.h"
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#include "api/rtpreceiverinterface.h"
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#include "api/rtpsenderinterface.h"
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#include "rtc_base/refcount.h"
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namespace webrtc {
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enum class RtpTransceiverDirection {
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kSendRecv,
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kSendOnly,
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kRecvOnly,
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kInactive
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};
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// The RtpTransceiverInterface maps to the RTCRtpTransceiver defined by the
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// WebRTC specification. A transceiver represents a combination of an RtpSender
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// and an RtpReceiver than share a common mid. As defined in JSEP, an
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// RtpTransceiver is said to be associated with a media description if its mid
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// property is non-null; otherwise, it is said to be disassociated.
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// JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
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//
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// Note that RtpTransceivers are only supported when using PeerConnection with
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// Unified Plan SDP.
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//
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// This class is thread-safe.
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//
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// WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
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class RtpTransceiverInterface : public rtc::RefCountInterface {
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public:
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// The mid attribute is the mid negotiated and present in the local and
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// remote descriptions. Before negotiation is complete, the mid value may be
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// null. After rollbacks, the value may change from a non-null value to null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
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virtual rtc::Optional<std::string> mid() const = 0;
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// The sender attribute exposes the RtpSender corresponding to the RTP media
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// that may be sent with the transceiver's mid. The sender is always present,
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// regardless of the direction of media.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
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virtual rtc::scoped_refptr<RtpSenderInterface> sender() const = 0;
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// The receiver attribute exposes the RtpReceiver corresponding to the RTP
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// media that may be received with the transceiver's mid. The receiver is
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// always present, regardless of the direction of media.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
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virtual rtc::scoped_refptr<RtpReceiverInterface> receiver() const = 0;
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// The stopped attribute indicates that the sender of this transceiver will no
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// longer send, and that the receiver will no longer receive. It is true if
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// either stop has been called or if setting the local or remote description
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// has caused the RtpTransceiver to be stopped.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
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virtual bool stopped() const = 0;
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// The direction attribute indicates the preferred direction of this
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// transceiver, which will be used in calls to CreateOffer and CreateAnswer.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
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virtual RtpTransceiverDirection direction() const = 0;
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// Sets the preferred direction of this transceiver. An update of
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// directionality does not take effect immediately. Instead, future calls to
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// CreateOffer and CreateAnswer mark the corresponding media descriptions as
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// sendrecv, sendonly, recvonly, or inactive.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
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virtual void SetDirection(RtpTransceiverDirection new_direction) = 0;
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// The current_direction attribute indicates the current direction negotiated
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// for this transceiver. If this transceiver has never been represented in an
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// offer/answer exchange, or if the transceiver is stopped, the value is null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
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virtual rtc::Optional<RtpTransceiverDirection> current_direction() const = 0;
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// The Stop method irreversibly stops the RtpTransceiver. The sender of this
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// transceiver will no longer send, the receiver will no longer receive.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
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virtual void Stop() = 0;
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// The SetCodecPreferences method overrides the default codec preferences used
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// by WebRTC for this transceiver.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
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// TODO(steveanton): Not implemented.
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virtual void SetCodecPreferences(
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rtc::ArrayView<RtpCodecCapability> codecs) = 0;
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protected:
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virtual ~RtpTransceiverInterface() = default;
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};
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} // namespace webrtc
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#endif // API_RTPTRANSCEIVERINTERFACE_H_
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@ -137,6 +137,8 @@ rtc_static_library("peerconnection") {
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"rtpreceiver.h",
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"rtpsender.cc",
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"rtpsender.h",
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"rtptransceiver.cc",
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"rtptransceiver.h",
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"sctputils.cc",
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"sctputils.h",
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"sdputils.cc",
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154
pc/rtptransceiver.cc
Normal file
154
pc/rtptransceiver.cc
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@ -0,0 +1,154 @@
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/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/rtptransceiver.h"
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#include <string>
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namespace webrtc {
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RtpTransceiver::RtpTransceiver(cricket::MediaType media_type)
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: unified_plan_(false), media_type_(media_type) {
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RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
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media_type == cricket::MEDIA_TYPE_VIDEO);
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}
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RtpTransceiver::~RtpTransceiver() {
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Stop();
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}
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void RtpTransceiver::SetChannel(cricket::BaseChannel* channel) {
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if (channel) {
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RTC_DCHECK_EQ(media_type(), channel->media_type());
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}
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channel_ = channel;
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for (auto sender : senders_) {
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if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
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static_cast<AudioRtpSender*>(sender->internal())
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->SetChannel(static_cast<cricket::VoiceChannel*>(channel));
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} else {
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static_cast<VideoRtpSender*>(sender->internal())
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->SetChannel(static_cast<cricket::VideoChannel*>(channel));
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}
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}
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for (auto receiver : receivers_) {
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if (!channel) {
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receiver->internal()->Stop();
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}
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if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
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static_cast<AudioRtpReceiver*>(receiver->internal())
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->SetChannel(static_cast<cricket::VoiceChannel*>(channel));
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} else {
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static_cast<VideoRtpReceiver*>(receiver->internal())
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->SetChannel(static_cast<cricket::VideoChannel*>(channel));
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}
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}
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}
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void RtpTransceiver::AddSender(
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rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender) {
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RTC_DCHECK(!unified_plan_);
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RTC_DCHECK(sender);
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RTC_DCHECK_EQ(media_type(), sender->internal()->media_type());
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RTC_DCHECK(std::find(senders_.begin(), senders_.end(), sender) ==
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senders_.end());
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senders_.push_back(sender);
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}
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bool RtpTransceiver::RemoveSender(RtpSenderInterface* sender) {
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RTC_DCHECK(!unified_plan_);
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if (sender) {
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RTC_DCHECK_EQ(media_type(), sender->media_type());
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}
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auto it = std::find(senders_.begin(), senders_.end(), sender);
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if (it == senders_.end()) {
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return false;
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}
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(*it)->internal()->Stop();
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senders_.erase(it);
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return true;
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}
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void RtpTransceiver::AddReceiver(
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
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receiver) {
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RTC_DCHECK(!unified_plan_);
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RTC_DCHECK(receiver);
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RTC_DCHECK_EQ(media_type(), receiver->internal()->media_type());
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RTC_DCHECK(std::find(receivers_.begin(), receivers_.end(), receiver) ==
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receivers_.end());
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receivers_.push_back(receiver);
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}
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bool RtpTransceiver::RemoveReceiver(RtpReceiverInterface* receiver) {
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RTC_DCHECK(!unified_plan_);
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if (receiver) {
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RTC_DCHECK_EQ(media_type(), receiver->media_type());
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}
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auto it = std::find(receivers_.begin(), receivers_.end(), receiver);
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if (it == receivers_.end()) {
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return false;
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}
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(*it)->internal()->Stop();
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receivers_.erase(it);
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return true;
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}
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rtc::Optional<std::string> RtpTransceiver::mid() const {
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return mid_;
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}
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rtc::scoped_refptr<RtpSenderInterface> RtpTransceiver::sender() const {
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RTC_DCHECK(unified_plan_);
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RTC_CHECK_EQ(1u, senders_.size());
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return senders_[0];
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}
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rtc::scoped_refptr<RtpReceiverInterface> RtpTransceiver::receiver() const {
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RTC_DCHECK(unified_plan_);
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RTC_CHECK_EQ(1u, receivers_.size());
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return receivers_[0];
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}
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bool RtpTransceiver::stopped() const {
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return stopped_;
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}
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RtpTransceiverDirection RtpTransceiver::direction() const {
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return direction_;
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}
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void RtpTransceiver::SetDirection(RtpTransceiverDirection new_direction) {
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// TODO(steveanton): This should fire OnNegotiationNeeded.
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direction_ = new_direction;
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}
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rtc::Optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
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const {
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return current_direction_;
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}
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void RtpTransceiver::Stop() {
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for (auto sender : senders_) {
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sender->internal()->Stop();
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}
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for (auto receiver : receivers_) {
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receiver->internal()->Stop();
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}
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stopped_ = true;
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}
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void RtpTransceiver::SetCodecPreferences(
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rtc::ArrayView<RtpCodecCapability> codecs) {
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// TODO(steveanton): Implement this.
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RTC_NOTREACHED() << "Not implemented";
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}
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} // namespace webrtc
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147
pc/rtptransceiver.h
Normal file
147
pc/rtptransceiver.h
Normal file
@ -0,0 +1,147 @@
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/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTPTRANSCEIVER_H_
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#define PC_RTPTRANSCEIVER_H_
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#include <string>
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#include <vector>
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#include "api/rtptransceiverinterface.h"
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#include "pc/rtpreceiver.h"
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#include "pc/rtpsender.h"
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namespace webrtc {
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// Implementation of the public RtpTransceiverInterface.
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//
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// The RtpTransceiverInterface is only intended to be used with a PeerConnection
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// that enables Unified Plan SDP. Thus, the methods that only need to implement
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// public API features and are not used internally can assume exactly one sender
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// and receiver.
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//
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// Since the RtpTransceiver is used internally by PeerConnection for tracking
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// RtpSenders, RtpReceivers, and BaseChannels, and PeerConnection needs to be
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// backwards compatible with Plan B SDP, this implementation is more flexible
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// than that required by the WebRTC specification.
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//
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// With Plan B SDP, an RtpTransceiver can have any number of senders and
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// receivers which map to a=ssrc lines in the m= section.
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// With Unified Plan SDP, an RtpTransceiver will have exactly one sender and one
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// receiver which are encapsulated by the m= section.
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//
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// This class manages the RtpSenders, RtpReceivers, and BaseChannel associated
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// with this m= section. Since the transceiver, senders, and receivers are
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// reference counted and can be referenced from JavaScript (in Chromium), these
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// objects must be ready to live for an arbitrary amount of time. The
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// BaseChannel is not reference counted and is owned by the ChannelManager, so
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// the PeerConnection must take care of creating/deleting the BaseChannel and
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// setting the channel reference in the transceiver to null when it has been
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// deleted.
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//
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// The RtpTransceiver is specialized to either audio or video according to the
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// MediaType specified in the constructor. Audio RtpTransceivers will have
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// AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers
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// will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel.
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class RtpTransceiver final
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: public rtc::RefCountedObject<RtpTransceiverInterface> {
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public:
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// Construct an RtpTransceiver with no senders, receivers, or channel set.
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// |media_type| specifies the type of RtpTransceiver (and, by transitivity,
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// the type of senders, receivers, and channel). Can either by audio or video.
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explicit RtpTransceiver(cricket::MediaType media_type);
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~RtpTransceiver() override;
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cricket::MediaType media_type() const { return media_type_; }
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// Returns the Voice/VideoChannel set for this transceiver. May be null if
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// the transceiver is not in the currently set local/remote description.
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cricket::BaseChannel* channel() const { return channel_; }
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// Sets the Voice/VideoChannel. The caller must pass in the correct channel
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// implementation based on the type of the transceiver.
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void SetChannel(cricket::BaseChannel* channel);
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// Adds an RtpSender of the appropriate type to be owned by this transceiver.
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// Must not be null.
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void AddSender(
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rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender);
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// Removes the given RtpSender. Returns false if the sender is not owned by
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// this transceiver.
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bool RemoveSender(RtpSenderInterface* sender);
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// Returns a vector of the senders owned by this transceiver.
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std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
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senders() const {
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return senders_;
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}
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// Adds an RtpReceiver of the appropriate type to be owned by this
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// transceiver. Must not be null.
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void AddReceiver(
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
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receiver);
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// Removes the given RtpReceiver. Returns false if the sender is not owned by
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// this transceiver.
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bool RemoveReceiver(RtpReceiverInterface* receiver);
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// Returns a vector of the receivers owned by this transceiver.
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std::vector<
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
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receivers() const {
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return receivers_;
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}
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// RtpTransceiverInterface implementation.
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rtc::Optional<std::string> mid() const override;
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rtc::scoped_refptr<RtpSenderInterface> sender() const override;
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rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
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bool stopped() const override;
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RtpTransceiverDirection direction() const override;
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void SetDirection(RtpTransceiverDirection new_direction) override;
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rtc::Optional<RtpTransceiverDirection> current_direction() const override;
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void Stop() override;
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void SetCodecPreferences(rtc::ArrayView<RtpCodecCapability> codecs) override;
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private:
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const bool unified_plan_;
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const cricket::MediaType media_type_;
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std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
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senders_;
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std::vector<
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
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receivers_;
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bool stopped_ = false;
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RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive;
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rtc::Optional<RtpTransceiverDirection> current_direction_;
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rtc::Optional<std::string> mid_;
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cricket::BaseChannel* channel_ = nullptr;
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};
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BEGIN_SIGNALING_PROXY_MAP(RtpTransceiver)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_CONSTMETHOD0(rtc::Optional<std::string>, mid);
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender);
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver);
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PROXY_CONSTMETHOD0(bool, stopped);
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PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction);
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PROXY_METHOD1(void, SetDirection, RtpTransceiverDirection);
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PROXY_CONSTMETHOD0(rtc::Optional<RtpTransceiverDirection>, current_direction);
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PROXY_METHOD0(void, Stop);
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PROXY_METHOD1(void, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>);
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END_PROXY_MAP();
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} // namespace webrtc
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|
||||
#endif // PC_RTPTRANSCEIVER_H_
|
||||
Reference in New Issue
Block a user