Searched refs:RtpRtcpInterface (Results 1 – 25 of 48) sorted by relevance
12
| /external/webrtc/modules/pacing/ |
| D | packet_router.h | 35 class RtpRtcpInterface; variable
50 void AddSendRtpModule(RtpRtcpInterface* rtp_module, bool remb_candidate);
51 void RemoveSendRtpModule(RtpRtcpInterface* rtp_module);
93 void AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module, uint32_t ssrc)
100 std::unordered_map<uint32_t, RtpRtcpInterface*> send_modules_map_
102 std::list<RtpRtcpInterface*> send_modules_list_
105 RtpRtcpInterface* last_send_module_ RTC_GUARDED_BY(modules_mutex_);
|
| D | packet_router.cc | 56 void PacketRouter::AddSendRtpModule(RtpRtcpInterface* rtp_module, in AddSendRtpModule()
77 void PacketRouter::AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module, in AddSendRtpModuleToMap()
99 void PacketRouter::RemoveSendRtpModule(RtpRtcpInterface* rtp_module) { in RemoveSendRtpModule()
163 RtpRtcpInterface* rtp_module = kv->second; in SendPacket()
210 for (RtpRtcpInterface* rtp_module : send_modules_list_) { in GeneratePadding()
313 for (RtpRtcpInterface* rtp_module : send_modules_list_) { in SendCombinedRtcpPacket()
|
| /external/webrtc/audio/voip/ |
| D | audio_egress.h | 47 AudioEgress(RtpRtcpInterface* rtp_rtcp,
113 RtpRtcpInterface* const rtp_rtcp_;
|
| D | audio_ingress.h | 47 AudioIngress(RtpRtcpInterface* rtp_rtcp,
126 RtpRtcpInterface* const rtp_rtcp_;
|
| D | audio_channel.cc | 46 RtpRtcpInterface::Configuration rtp_config; in AudioChannel()
|
| D | audio_egress.cc | 20 AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp, in AudioEgress()
|
| D | audio_ingress.cc | 38 RtpRtcpInterface* rtp_rtcp, in AudioIngress()
|
| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_rtcp_impl2.h | 49 class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
54 const RtpRtcpInterface::Configuration& configuration);
277 explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
|
| D | nack_rtx_unittest.cc | 66 void SetSendModule(RtpRtcpInterface* rtpRtcpModule) { in SetSendModule()
114 RtpRtcpInterface* module_;
130 RtpRtcpInterface::Configuration configuration; in SetUp()
|
| D | rtcp_sender_unittest.cc | 79 RtpRtcpInterface::Configuration configuration = GetDefaultConfig(); in RtcpSenderTest()
88 RtpRtcpInterface::Configuration GetDefaultConfig() { in GetDefaultConfig()
89 RtpRtcpInterface::Configuration configuration; in GetDefaultConfig()
194 RtpRtcpInterface::Configuration config; in TEST_F()
216 RtpRtcpInterface::Configuration config; in TEST_F()
525 RtpRtcpInterface::Configuration config; in TEST_F()
653 RtpRtcpInterface::Configuration config; in TEST_F()
|
| D | rtp_rtcp_impl.h | 49 const RtpRtcpInterface::Configuration& configuration);
317 explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
|
| D | rtp_sender_unittest.cc | 248 RtpSenderContext(const RtpRtcpInterface::Configuration& config, in RtpSenderContext()
358 RtpRtcpInterface::Configuration config; in SetUpRtpSender()
556 RtpRtcpInterface::Configuration config; in TEST_P()
607 RtpRtcpInterface::Configuration config; in TEST_P()
643 RtpRtcpInterface::Configuration config; in TEST_P()
683 RtpRtcpInterface::Configuration config; in TEST_P()
739 RtpRtcpInterface::Configuration config; in TEST_P()
827 RtpRtcpInterface::Configuration config; in TEST_P()
1321 RtpRtcpInterface::Configuration config; in TEST_P()
1425 RtpRtcpInterface::Configuration config; in TEST_P()
[all …]
|
| D | rtp_sender_egress.h | 59 RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
|
| D | rtp_sender_audio_unittest.cc | 72 RtpRtcpInterface::Configuration config; in RtpSenderAudioTest()
|
| /external/webrtc/test/fuzzers/ |
| D | rtcp_receiver_fuzzer.cc | 43 RtpRtcpInterface::Configuration config; in FuzzOneInput()
|
| /external/webrtc/modules/rtp_rtcp/include/ |
| D | rtp_rtcp.h | 25 class RtpRtcp : public Module, public RtpRtcpInterface {
|
| /external/webrtc/audio/ |
| D | channel_send.h | 99 virtual RtpRtcpInterface* GetRtpRtcp() const = 0;
|
| D | mock_voe_channel_proxy.h | 155 MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
|
| D | audio_send_stream.h | 179 RtpRtcpInterface* const rtp_rtcp_module_;
|
| D | channel_send.cc | 111 RtpRtcpInterface* GetRtpRtcp() const override;
480 RtpRtcpInterface::Configuration configuration; in ChannelSend()
858 RtpRtcpInterface* ChannelSend::GetRtpRtcp() const { in GetRtpRtcp()
|
| /external/webrtc/modules/rtp_rtcp/source/deprecated/ |
| D | deprecated_rtp_sender_egress.h | 50 DEPRECATED_RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
|
| /external/webrtc/call/ |
| D | flexfec_receive_stream_impl.cc | 126 RtpRtcpInterface::Configuration configuration; in CreateRtpRtcpModule()
|
| D | rtp_video_sender.h | 219 std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;
|
| /external/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 30 class MockRtpRtcpInterface : public RtpRtcpInterface {
|
| /external/webrtc/audio/voip/test/ |
| D | audio_ingress_unittest.cc | 42 RtpRtcpInterface::Configuration rtp_config; in AudioIngressTest()
|
12