Add SDP semantics option to RTCConfiguration
This setting allows the user of PeerConnection to choose whether
to use Plan B (current) or Unified Plan (future) semantics.
Unified Plan semantics are not yet supported.
Bug: chromium:465349
Change-Id: I77a5c376c83f335f734488e11e619582a314bffe
Reviewed-on: https://webrtc-review.googlesource.com/22766
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20806}diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 60f1635..c95cc7d 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -141,6 +141,10 @@
virtual ~StatsObserver() {}
};
+// For now, kDefault is interpreted as kPlanB.
+// TODO(bugs.webrtc.org/8530): Switch default to kUnifiedPlan.
+enum class SdpSemantics { kDefault, kPlanB, kUnifiedPlan };
+
class PeerConnectionInterface : public rtc::RefCountInterface {
public:
// See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
@@ -476,6 +480,34 @@
// called.
webrtc::TurnCustomizer* turn_customizer = nullptr;
+ // Configure the SDP semantics used by this PeerConnection. Note that the
+ // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
+ // RtpTransceiver API is only available with kUnifiedPlan semantics.
+ //
+ // kPlanB will cause PeerConnection to create offers and answers with at
+ // most one audio and one video m= section with multiple RtpSenders and
+ // RtpReceivers specified as multiple a=ssrc lines within the section. This
+ // will also cause PeerConnection to reject offers/answers with multiple m=
+ // sections of the same media type.
+ //
+ // kUnifiedPlan will cause PeerConnection to create offers and answers with
+ // multiple m= sections where each m= section maps to one RtpSender and one
+ // RtpReceiver (an RtpTransceiver), either both audio or both video. Plan B
+ // style offers or answers will be rejected in calls to SetLocalDescription
+ // or SetRemoteDescription.
+ //
+ // For users who only send at most one audio and one video track, this
+ // choice does not matter and should be left as kDefault.
+ //
+ // For users who wish to send multiple audio/video streams and need to stay
+ // interoperable with legacy WebRTC implementations, specify kPlanB.
+ //
+ // For users who wish to send multiple audio/video streams and/or wish to
+ // use the new RtpTransceiver API, specify kUnifiedPlan.
+ //
+ // TODO(steveanton): Implement support for kUnifiedPlan.
+ SdpSemantics sdp_semantics = SdpSemantics::kDefault;
+
//
// Don't forget to update operator== if adding something.
//
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index d617c71..53c891e 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -675,6 +675,7 @@
rtc::Optional<int> ice_check_min_interval;
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
webrtc::TurnCustomizer* turn_customizer;
+ SdpSemantics sdp_semantics;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
"Did you add something to RTCConfiguration and forget to "
@@ -710,7 +711,8 @@
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
ice_check_min_interval == o.ice_check_min_interval &&
ice_regather_interval_range == o.ice_regather_interval_range &&
- turn_customizer == o.turn_customizer;
+ turn_customizer == o.turn_customizer &&
+ sdp_semantics == o.sdp_semantics;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index ae6b537..0b00588 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -458,9 +458,11 @@
// semantics for creating offers/answers and setting local/remote
// descriptions. If this is true the RtpTransceiver API will also be available
// to the user. If this is false, Plan B semantics are assumed.
- // TODO(steveanton): Flip the default to be Unified Plan once sufficient time
- // has passed.
- bool IsUnifiedPlan() const { return false; }
+ // TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
+ // sufficient time has passed.
+ bool IsUnifiedPlan() const {
+ return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
+ }
// Is there an RtpSender of the given type?
bool HasRtpSender(cricket::MediaType type) const;
diff --git a/pc/rtptransceiver.cc b/pc/rtptransceiver.cc
index 7f6ed6d..8ca29fc 100644
--- a/pc/rtptransceiver.cc
+++ b/pc/rtptransceiver.cc
@@ -20,6 +20,18 @@
media_type == cricket::MEDIA_TYPE_VIDEO);
}
+RtpTransceiver::RtpTransceiver(
+ rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
+ rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
+ receiver)
+ : unified_plan_(true), media_type_(sender->media_type()) {
+ RTC_DCHECK(media_type_ == cricket::MEDIA_TYPE_AUDIO ||
+ media_type_ == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(sender->media_type(), receiver->media_type());
+ senders_.push_back(sender);
+ receivers_.push_back(receiver);
+}
+
RtpTransceiver::~RtpTransceiver() {
Stop();
}
diff --git a/pc/rtptransceiver.h b/pc/rtptransceiver.h
index 9feaac7..7053fad 100644
--- a/pc/rtptransceiver.h
+++ b/pc/rtptransceiver.h
@@ -53,10 +53,18 @@
class RtpTransceiver final
: public rtc::RefCountedObject<RtpTransceiverInterface> {
public:
- // Construct an RtpTransceiver with no senders, receivers, or channel set.
+ // Construct a Plan B-style RtpTransceiver with no senders, receivers, or
+ // channel set.
// |media_type| specifies the type of RtpTransceiver (and, by transitivity,
// the type of senders, receivers, and channel). Can either by audio or video.
explicit RtpTransceiver(cricket::MediaType media_type);
+ // Construct a Unified Plan-style RtpTransceiver with the given sender and
+ // receiver. The media type will be derived from the media types of the sender
+ // and receiver. The sender and receiver should have the same media type.
+ RtpTransceiver(
+ rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
+ rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
+ receiver);
~RtpTransceiver() override;
cricket::MediaType media_type() const { return media_type_; }
