| /external/webrtc/webrtc/test/ |
| layer_filtering_transport.cc | 41 const PacketOptions& options) {
|
| rtp_rtcp_observer.h | 93 const PacketOptions& options) override {
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| conference_transport.h | 103 const webrtc::PacketOptions& options) override;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
| rtp_rtcp_extensions.cc | 33 const webrtc::PacketOptions& options) override {
|
| /external/webrtc/talk/media/base/ |
| fakenetworkinterface.h | 132 const rtc::PacketOptions& options) {
158 const rtc::PacketOptions& options) {
|
| mediachannel.h | 460 const rtc::PacketOptions& options) = 0;
462 const rtc::PacketOptions& options) = 0;
506 bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) {
510 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) {
542 const rtc::PacketOptions& options) {
[all...] |
| rtpdataengine.cc | 362 MediaChannel::SendPacket(&packet, rtc::PacketOptions());
|
| /external/webrtc/webrtc/base/ |
| asynctcpsocket.cc | 129 const rtc::PacketOptions& options) {
250 const rtc::PacketOptions& options) {
|
| natserver.cc | 176 rtc::PacketOptions options;
203 rtc::PacketOptions options;
|
| /external/webrtc/webrtc/p2p/base/ |
| relayport.cc | 54 int Send(const void* pv, size_t cb, const rtc::PacketOptions& options);
111 const rtc::PacketOptions& options);
157 const rtc::PacketOptions& options);
311 const rtc::PacketOptions& options,
433 rtc::PacketOptions options; // Default dscp set to NO_CHANGE.
443 const rtc::PacketOptions& options) {
561 const rtc::PacketOptions& options) {
767 const rtc::PacketOptions& options) {
|
| turnport.h | 91 const rtc::PacketOptions& options,
232 const rtc::PacketOptions& options);
|
| dtlstransportchannel.cc | 72 rtc::PacketOptions packet_options;
350 const rtc::PacketOptions& options, int flags) {
|
| port.h | 479 const rtc::PacketOptions& options) = 0;
653 const rtc::PacketOptions& options) override;
|
| stunport.cc | 277 const rtc::PacketOptions& options,
499 rtc::PacketOptions options(DefaultDscpValue());
|
| tcpport.cc | 197 const rtc::PacketOptions& options,
332 const rtc::PacketOptions& options) {
|
| asyncstuntcpsocket_unittest.cc | 108 rtc::PacketOptions options;
|
| dtlstransportchannel.h | 108 const rtc::PacketOptions& options,
|
| p2ptransportchannel.h | 98 const rtc::PacketOptions& options,
|
| stunport.h | 134 const rtc::PacketOptions& options,
|
| turnport.cc | 159 const rtc::PacketOptions& options);
510 const rtc::PacketOptions& options,
676 rtc::PacketOptions options(DefaultDscpValue());
897 const rtc::PacketOptions& options) {
[all...] |
| /external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
| after_initialization_fixture.h | 44 const webrtc::PacketOptions& options) override {
|
| /external/webrtc/talk/session/media/ |
| channel.h | 211 const rtc::PacketOptions& options) override;
212 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options)
230 const rtc::PacketOptions& options);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 36 const PacketOptions& options) {
|
| /external/webrtc/webrtc/test/channel_transport/ |
| udp_transport_impl.h | 121 const PacketOptions& packet_options) override;
|
| /external/webrtc/talk/media/sctp/ |
| sctpdataengine_unittest.cc | 67 const rtc::PacketOptions& options) {
96 const rtc::PacketOptions& options) {
|
