CreateProxied(const cricket::MediaConfig &config, cricket::ChannelManager *channel_manager, webrtc::RtcEventLog *event_log, rtc::Thread *signaling_thread, rtc::Thread *worker_thread) | webrtc::RtpTransportControllerAdapter | static |
CreateProxied(const cricket::MediaConfig &config, cricket::ChannelManager *channel_manager, webrtc::RtcEventLog *event_log, rtc::Thread *signaling_thread, rtc::Thread *worker_thread) | webrtc::RtpTransportControllerAdapter | static |
CreateProxiedRtpReceiver(cricket::MediaType kind, RtpTransportInterface *transport_proxy) | webrtc::RtpTransportControllerAdapter | |
CreateProxiedRtpReceiver(cricket::MediaType kind, RtpTransportInterface *transport_proxy) | webrtc::RtpTransportControllerAdapter | |
CreateProxiedRtpSender(cricket::MediaType kind, RtpTransportInterface *transport_proxy) | webrtc::RtpTransportControllerAdapter | |
CreateProxiedRtpSender(cricket::MediaType kind, RtpTransportInterface *transport_proxy) | webrtc::RtpTransportControllerAdapter | |
CreateProxiedRtpTransport(const RtcpParameters &rtcp_parameters, PacketTransportInterface *rtp, PacketTransportInterface *rtcp) | webrtc::RtpTransportControllerAdapter | |
CreateProxiedRtpTransport(const RtcpParameters &rtcp_parameters, PacketTransportInterface *rtp, PacketTransportInterface *rtcp) | webrtc::RtpTransportControllerAdapter | |
disconnect_all() | sigslot::has_slots_interface | inline |
disconnect_all() | sigslot::has_slots_interface | inline |
GetInternal() override | webrtc::RtpTransportControllerAdapter | inlineprotectedvirtual |
GetInternal() override | webrtc::RtpTransportControllerAdapter | inlineprotectedvirtual |
GetTransports() const override | webrtc::RtpTransportControllerAdapter | virtual |
GetTransports() const override | webrtc::RtpTransportControllerAdapter | virtual |
has_slots() | sigslot::has_slots<> | inline |
has_slots() | sigslot::has_slots<> | inline |
has_slots_interface(signal_connect_t conn, signal_disconnect_t disc, disconnect_all_t disc_all) | sigslot::has_slots_interface | inlineprotected |
has_slots_interface(signal_connect_t conn, signal_disconnect_t disc, disconnect_all_t disc_all) | sigslot::has_slots_interface | inlineprotected |
SetRtcpParameters(const RtcpParameters ¶meters, RtpTransportInterface *inner_transport) | webrtc::RtpTransportControllerAdapter | |
SetRtcpParameters(const RtcpParameters ¶meters, RtpTransportInterface *inner_transport) | webrtc::RtpTransportControllerAdapter | |
signal_connect(_signal_base_interface *sender) | sigslot::has_slots_interface | inline |
signal_connect(_signal_base_interface *sender) | sigslot::has_slots_interface | inline |
signal_disconnect(_signal_base_interface *sender) | sigslot::has_slots_interface | inline |
signal_disconnect(_signal_base_interface *sender) | sigslot::has_slots_interface | inline |
signaling_thread() const | webrtc::RtpTransportControllerAdapter | inline |
signaling_thread() const | webrtc::RtpTransportControllerAdapter | inline |
ValidateAndApplyAudioReceiverParameters(const RtpParameters ¶meters) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyAudioReceiverParameters(const RtpParameters ¶meters) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyAudioSenderParameters(const RtpParameters ¶meters, uint32_t *primary_ssrc) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyAudioSenderParameters(const RtpParameters ¶meters, uint32_t *primary_ssrc) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyVideoReceiverParameters(const RtpParameters ¶meters) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyVideoReceiverParameters(const RtpParameters ¶meters) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyVideoSenderParameters(const RtpParameters ¶meters, uint32_t *primary_ssrc) | webrtc::RtpTransportControllerAdapter | |
ValidateAndApplyVideoSenderParameters(const RtpParameters ¶meters, uint32_t *primary_ssrc) | webrtc::RtpTransportControllerAdapter | |
video_channel() | webrtc::RtpTransportControllerAdapter | inline |
video_channel() | webrtc::RtpTransportControllerAdapter | inline |
voice_channel() | webrtc::RtpTransportControllerAdapter | inline |
voice_channel() | webrtc::RtpTransportControllerAdapter | inline |
worker_thread() const | webrtc::RtpTransportControllerAdapter | inline |
worker_thread() const | webrtc::RtpTransportControllerAdapter | inline |
~has_slots() | sigslot::has_slots<> | inline |
~has_slots() | sigslot::has_slots<> | inline |
~has_slots_interface() | sigslot::has_slots_interface | inlineprotectedvirtual |
~has_slots_interface() | sigslot::has_slots_interface | inlineprotectedvirtual |
~RtpTransportControllerAdapter() override | webrtc::RtpTransportControllerAdapter | |
~RtpTransportControllerAdapter() override | webrtc::RtpTransportControllerAdapter | |
~RtpTransportControllerInterface() | webrtc::RtpTransportControllerInterface | inlinevirtual |
~RtpTransportControllerInterface() | webrtc::RtpTransportControllerInterface | inlinevirtual |