Index: webrtc/ortc/rtptransportcontrolleradapter.h |
diff --git a/webrtc/ortc/rtptransportcontrolleradapter.h b/webrtc/ortc/rtptransportcontrolleradapter.h |
index 9728c3968aaf94fb232e6e205354fa96c906ac99..d4494d0d6ce5bda54c6241c0703cf1faeb970e1f 100644 |
--- a/webrtc/ortc/rtptransportcontrolleradapter.h |
+++ b/webrtc/ortc/rtptransportcontrolleradapter.h |
@@ -21,6 +21,7 @@ |
#include "webrtc/api/ortc/rtptransportcontrollerinterface.h" |
#include "webrtc/api/ortc/srtptransportinterface.h" |
#include "webrtc/call/call.h" |
+#include "webrtc/call/rtp_transport_controller_send.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/media/base/mediachannel.h" // For MediaConfig. |
#include "webrtc/pc/channelmanager.h" |
@@ -77,12 +78,12 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface, |
// these methods return proxies that will safely call methods on the correct |
// thread. |
RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxiedRtpTransport( |
- const RtcpParameters& rtcp_parameters, |
+ const RtpTransportParameters& rtcp_parameters, |
PacketTransportInterface* rtp, |
PacketTransportInterface* rtcp); |
RTCErrorOr<std::unique_ptr<SrtpTransportInterface>> |
- CreateProxiedSrtpTransport(const RtcpParameters& rtcp_parameters, |
+ CreateProxiedSrtpTransport(const RtpTransportParameters& rtcp_parameters, |
PacketTransportInterface* rtp, |
PacketTransportInterface* rtcp); |
@@ -100,8 +101,10 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface, |
rtc::Thread* signaling_thread() const { return signaling_thread_; } |
rtc::Thread* worker_thread() const { return worker_thread_; } |
- RTCError SetRtcpParameters(const RtcpParameters& parameters, |
- RtpTransportInterface* inner_transport); |
+ // |parameters.keepalive| will be set for ALL RTP transports in the call. |
+ RTCError SetRtpTransportParameters(const RtpTransportParameters& parameters, |
+ RtpTransportInterface* inner_transport); |
+ void SetRtpTransportParameters_w(const RtpTransportParameters& parameters); |
cricket::VoiceChannel* voice_channel() { return voice_channel_; } |
cricket::VideoChannel* video_channel() { return video_channel_; } |
@@ -193,9 +196,11 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface, |
RtpTransportInterface* inner_audio_transport_ = nullptr; |
RtpTransportInterface* inner_video_transport_ = nullptr; |
const cricket::MediaConfig media_config_; |
+ RtpKeepAliveConfig keepalive_; |
cricket::ChannelManager* channel_manager_; |
webrtc::RtcEventLog* event_log_; |
std::unique_ptr<Call> call_; |
+ webrtc::RtpTransportControllerSend* call_send_rtp_transport_controller_; |
// BaseChannel takes content descriptions as input, so we store them here |
// such that they can be updated when a new RtpSenderAdapter/ |