Fix the 'SetConfiguration(RTCConfiguration::use_media_transport)' setting.
In the past, it would incorrectly set up a state for 'use_media_transport' (i.e. it could say "use_media_transport" is true, but jseptransportcontroller wouldn't know about that).
Also, removes unnecessary field (unused).
Bug: webrtc:9719
Change-Id: I7e5c0ce81b3b70f63c49d661d95b95b5bcbb0c68
Reviewed-on: https://webrtc-review.googlesource.com/c/106960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25263}diff --git a/pc/jseptransportcontroller.cc b/pc/jseptransportcontroller.cc
index 19a2d40..991d769 100644
--- a/pc/jseptransportcontroller.cc
+++ b/pc/jseptransportcontroller.cc
@@ -378,6 +378,15 @@
}
}
+void JsepTransportController::SetMediaTransportFactory(
+ MediaTransportFactory* media_transport_factory) {
+ RTC_DCHECK(media_transport_factory == config_.media_transport_factory ||
+ jsep_transports_by_name_.empty())
+ << "You can only call SetMediaTransportFactory before "
+ "JsepTransportController created its first transport.";
+ config_.media_transport_factory = media_transport_factory;
+}
+
std::unique_ptr<cricket::DtlsTransportInternal>
JsepTransportController::CreateDtlsTransport(const std::string& transport_name,
bool rtcp) {
diff --git a/pc/jseptransportcontroller.h b/pc/jseptransportcontroller.h
index 518d310..5d0f5ce 100644
--- a/pc/jseptransportcontroller.h
+++ b/pc/jseptransportcontroller.h
@@ -167,6 +167,12 @@
void SetActiveResetSrtpParams(bool active_reset_srtp_params);
+ // Allows to overwrite the settings from config. You may set or reset the
+ // media transport factory on the jsep transport controller, as long as you
+ // did not call 'GetMediaTransport' or 'MaybeCreateJsepTransport'. Once Jsep
+ // transport is created, you can't change this setting.
+ void SetMediaTransportFactory(MediaTransportFactory* media_transport_factory);
+
// All of these signals are fired on the signaling thread.
// If any transport failed => failed,
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index ca4d13d..ecdec61 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -2981,6 +2981,9 @@
}
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
+ transport_controller_->SetMediaTransportFactory(
+ modified_config.use_media_transport ? factory_->media_transport_factory()
+ : nullptr);
if (configuration_.active_reset_srtp_params !=
modified_config.active_reset_srtp_params) {
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 0cd976a..c1d3d40 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -927,7 +927,10 @@
MediaTransportInterface* GetMediaTransport(const std::string& mid) {
auto media_transport = transport_controller_->GetMediaTransport(mid);
RTC_DCHECK(configuration_.use_media_transport ==
- (media_transport != nullptr));
+ (media_transport != nullptr))
+ << "configuration_.use_media_transport="
+ << configuration_.use_media_transport
+ << ", (media_transport != nullptr)=" << (media_transport != nullptr);
return media_transport;
}
@@ -1043,10 +1046,6 @@
// List of content names for which the remote side triggered an ICE restart.
std::set<std::string> pending_ice_restarts_;
- // Optional media transport for sending / receiving encoded frames.
- // If available, media transport will be used instead of RTP / SRTP.
- std::unique_ptr<MediaTransportFactory> media_transport_factory_;
-
std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
// Member variables for caching global options.
