| 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/
|
|
|