/external/webrtc/webrtc/ |
D | config.cc | 24 std::string RtpExtension::ToString() const { in ToString() 32 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset"; 33 const char* RtpExtension::kAbsSendTime = 35 const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation"; 36 const char* RtpExtension::kAudioLevel = 38 const char* RtpExtension::kTransportSequenceNumber = 41 bool RtpExtension::IsSupportedForAudio(const std::string& name) { in IsSupportedForAudio() 42 return name == webrtc::RtpExtension::kAbsSendTime || in IsSupportedForAudio() 43 name == webrtc::RtpExtension::kAudioLevel || in IsSupportedForAudio() 44 name == webrtc::RtpExtension::kTransportSequenceNumber; in IsSupportedForAudio() [all …]
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D | config.h | 54 struct RtpExtension { struct 55 RtpExtension(const std::string& name, int id) : name(name), id(id) {} in RtpExtension() function 57 bool operator==(const RtpExtension& rhs) const {
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D | audio_send_stream.h | 68 std::vector<RtpExtension> extensions;
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D | audio_receive_stream.h | 83 std::vector<RtpExtension> extensions;
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D | video_receive_stream.h | 137 std::vector<RtpExtension> extensions;
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D | video_send_stream.h | 112 std::vector<RtpExtension> extensions;
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/external/webrtc/webrtc/call/ |
D | rampup_tests.cc | 124 if (extension_type_ == RtpExtension::kAbsSendTime) { in ModifyVideoConfigs() 128 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); in ModifyVideoConfigs() 129 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) { in ModifyVideoConfigs() 132 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs() 137 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs() 185 EXPECT_NE(RtpExtension::kTOffset, extension_type_) in ModifyAudioConfigs() 192 if (extension_type_ == RtpExtension::kAbsSendTime) { in ModifyAudioConfigs() 195 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); in ModifyAudioConfigs() 196 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) { in ModifyAudioConfigs() 198 send_config->rtp.extensions.push_back(RtpExtension( in ModifyAudioConfigs() [all …]
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D | bitrate_estimator_tests.cc | 137 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId)); in SetUp() 139 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); in SetUp() 192 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); in Stream() 269 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId)); in TEST_F() 278 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); in TEST_F() 289 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); in TEST_F() 305 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); in TEST_F() 314 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId)); in TEST_F() 321 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId); in TEST_F() 330 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId)); in TEST_F() [all …]
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D | rtc_event_log_unittest.cc | 48 const char* kExtensionNames[] = {RtpExtension::kTOffset, 49 RtpExtension::kAudioLevel, 50 RtpExtension::kAbsSendTime, 51 RtpExtension::kVideoRotation, 52 RtpExtension::kTransportSequenceNumber}; 390 RtpExtension(kExtensionNames[i], prng->Rand<int>())); in GenerateVideoReceiveConfig() 410 RtpExtension(kExtensionNames[i], prng->Rand<int>())); in GenerateVideoSendConfig()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcmediaengine.cc | 77 std::vector<webrtc::RtpExtension>* extensions, in DiscardRedundantExtensions() 83 [name](const webrtc::RtpExtension& rhs) { in DiscardRedundantExtensions() 112 std::vector<webrtc::RtpExtension> FilterRtpExtensions( in FilterRtpExtensions() 118 std::vector<webrtc::RtpExtension> result; in FilterRtpExtensions() 132 [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) { in FilterRtpExtensions() 139 [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) { in FilterRtpExtensions()
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D | webrtcmediaengine_unittest.cc | 66 bool IsSorted(const std::vector<webrtc::RtpExtension>& extensions) { in IsSorted() 114 std::vector<webrtc::RtpExtension> filtered = in TEST() 121 std::vector<webrtc::RtpExtension> filtered = in TEST() 130 std::vector<webrtc::RtpExtension> filtered = in TEST() 138 std::vector<webrtc::RtpExtension> filtered = in TEST() 146 std::vector<webrtc::RtpExtension> filtered = in TEST() 155 std::vector<webrtc::RtpExtension> filtered = in TEST() 174 std::vector<webrtc::RtpExtension> filtered = in TEST() 188 std::vector<webrtc::RtpExtension> filtered = in TEST() 200 std::vector<webrtc::RtpExtension> filtered = in TEST()
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D | webrtcvideoengine2.h | 247 const std::vector<webrtc::RtpExtension>& rtp_extensions, 254 const std::vector<webrtc::RtpExtension>& rtp_extensions); 406 void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions); 522 std::vector<webrtc::RtpExtension> send_rtp_extensions_; 527 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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D | webrtcmediaengine.h | 63 std::vector<webrtc::RtpExtension> FilterRtpExtensions(
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D | webrtcvoiceengine.h | 279 std::vector<webrtc::RtpExtension> send_rtp_extensions_; 283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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D | webrtcvideoengine2_unittest.cc | 1181 webrtc::RtpExtension::kTOffset); in TEST_F() 1185 webrtc::RtpExtension::kTOffset); in TEST_F() 1191 webrtc::RtpExtension::kAbsSendTime); in TEST_F() 1195 webrtc::RtpExtension::kAbsSendTime); in TEST_F() 1226 webrtc::RtpExtension::kTransportSequenceNumber); in TEST_F() 1232 webrtc::RtpExtension::kTransportSequenceNumber); in TEST_F() 1238 webrtc::RtpExtension::kVideoRotation); in TEST_F() 1242 webrtc::RtpExtension::kVideoRotation); in TEST_F() 1316 cricket::RtpHeaderExtension(webrtc::RtpExtension::kTOffset, kTOffsetId)); in TEST_F() 1324 EXPECT_STREQ(webrtc::RtpExtension::kTOffset, in TEST_F() [all …]
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D | webrtcvoiceengine.cc | 1079 const std::vector<webrtc::RtpExtension>& extensions, in WebRtcAudioSendStream() 1102 const std::vector<webrtc::RtpExtension>& extensions) { in RecreateAudioSendStream() 1208 const std::vector<webrtc::RtpExtension>& extensions, in WebRtcAudioReceiveStream() 1227 const std::vector<webrtc::RtpExtension>& extensions) { in RecreateAudioReceiveStream() 1254 const std::vector<webrtc::RtpExtension>& extensions) { in RecreateAudioReceiveStream() 1317 std::vector<webrtc::RtpExtension> filtered_extensions = in SetSendParameters() 1319 webrtc::RtpExtension::IsSupportedForAudio, true); in SetSendParameters() 1348 std::vector<webrtc::RtpExtension> filtered_extensions = in SetRecvParameters() 1350 webrtc::RtpExtension::IsSupportedForAudio, false); in SetRecvParameters()
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/external/webrtc/webrtc/audio/ |
D | audio_send_stream_unittest.cc | 100 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId)); in ConfigHelper() 102 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); in ConfigHelper() 103 stream_config_.rtp.extensions.push_back(RtpExtension( in ConfigHelper() 104 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); in ConfigHelper() 172 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); in TEST()
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D | audio_receive_stream_unittest.cc | 107 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); in ConfigHelper() 109 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId)); in ConfigHelper() 209 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); in TEST() 264 helper.config().rtp.extensions.push_back(RtpExtension( in TEST() 265 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); in TEST()
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D | audio_receive_stream.cc | 41 if (extension.name == RtpExtension::kTransportSequenceNumber) { in UseSendSideBwe() 100 if (extension.name == RtpExtension::kAudioLevel) { in AudioReceiveStream() 105 } else if (extension.name == RtpExtension::kAbsSendTime) { in AudioReceiveStream() 110 } else if (extension.name == RtpExtension::kTransportSequenceNumber) { in AudioReceiveStream()
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D | audio_send_stream.cc | 80 if (extension.name == RtpExtension::kAbsSendTime) { in AudioSendStream() 82 } else if (extension.name == RtpExtension::kAudioLevel) { in AudioSendStream() 84 } else if (extension.name == RtpExtension::kTransportSequenceNumber) { in AudioSendStream()
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/external/webrtc/webrtc/video/ |
D | video_receive_stream.cc | 29 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) { in UseSendSideBwe() 31 if (extension.name == RtpExtension::kTransportSequenceNumber) in UseSendSideBwe() 210 if (extension == RtpExtension::kTOffset) { in VideoReceiveStream() 212 } else if (extension == RtpExtension::kAbsSendTime) { in VideoReceiveStream() 214 } else if (extension == RtpExtension::kVideoRotation) { in VideoReceiveStream() 216 } else if (extension == RtpExtension::kTransportSequenceNumber) { in VideoReceiveStream()
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D | replay.cc | 228 RtpExtension(RtpExtension::kTOffset, flags::TransmissionOffsetId())); in RtpReplay() 232 RtpExtension(RtpExtension::kAbsSendTime, flags::AbsSendTimeId())); in RtpReplay()
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D | video_send_stream.cc | 137 for (const RtpExtension& extension : config.rtp.extensions) { in VideoSendStream() 138 if (extension.name == RtpExtension::kTransportSequenceNumber) { in VideoSendStream() 179 if (extension == RtpExtension::kTOffset) { in VideoSendStream() 181 } else if (extension == RtpExtension::kAbsSendTime) { in VideoSendStream() 183 } else if (extension == RtpExtension::kVideoRotation) { in VideoSendStream() 185 } else if (extension == RtpExtension::kTransportSequenceNumber) { in VideoSendStream()
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D | video_send_stream_tests.cc | 155 send_config->rtp.extensions.push_back(RtpExtension( in TEST_F() 156 RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId)); in TEST_F() 199 RtpExtension(RtpExtension::kTOffset, test::kTOffsetExtensionId)); in TEST_F() 243 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F() 403 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs() 404 RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId)); in ModifyVideoConfigs() 406 RtpExtension(RtpExtension::kTransportSequenceNumber, in ModifyVideoConfigs()
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/external/webrtc/webrtc/test/ |
D | call_test.cc | 191 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); in CreateSendConfig() 195 video_send_config_.rtp.extensions.push_back(RtpExtension( in CreateSendConfig() 196 RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId)); in CreateSendConfig() 214 for (const RtpExtension& extension : video_send_config_.rtp.extensions) in CreateMatchingReceiveConfigs()
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