| /external/webrtc/webrtc/call/ |
| D | rampup_tests.cc | 129 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) { in ModifyVideoConfigs()
196 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) { in ModifyAudioConfigs()
512 RtpExtension::kTransportSequenceNumber, true, false); in TEST_F()
520 RtpExtension::kTransportSequenceNumber, true, false); in TEST_F()
553 RampUpTester test(1, 0, 0, RtpExtension::kTransportSequenceNumber, false, in TEST_F()
559 RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, false, in TEST_F()
565 RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true, in TEST_F()
571 RampUpTester test(3, 1, 0, RtpExtension::kTransportSequenceNumber, true, in TEST_F()
577 RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true, in TEST_F()
584 RtpExtension::kTransportSequenceNumber, false, false); in TEST_F()
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| D | rtc_event_log_unittest.cc | 52 RtpExtension::kTransportSequenceNumber};
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| /external/webrtc/webrtc/ |
| D | config.cc | 38 const char* RtpExtension::kTransportSequenceNumber = member in webrtc::RtpExtension
44 name == webrtc::RtpExtension::kTransportSequenceNumber; in IsSupportedForAudio()
51 name == webrtc::RtpExtension::kTransportSequenceNumber; in IsSupportedForVideo()
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| D | config.h | 67 static const char* kTransportSequenceNumber; member
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| /external/webrtc/webrtc/audio/ |
| D | audio_receive_stream.cc | 41 if (extension.name == RtpExtension::kTransportSequenceNumber) { in UseSendSideBwe()
110 } else if (extension.name == RtpExtension::kTransportSequenceNumber) { in AudioReceiveStream()
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| D | audio_send_stream.cc | 84 } else if (extension.name == RtpExtension::kTransportSequenceNumber) { in AudioSendStream()
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| D | audio_send_stream_unittest.cc | 104 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); in ConfigHelper()
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| D | audio_receive_stream_unittest.cc | 265 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); in TEST()
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| /external/webrtc/webrtc/video/ |
| D | video_receive_stream.cc | 31 if (extension.name == RtpExtension::kTransportSequenceNumber) in UseSendSideBwe()
216 } else if (extension == RtpExtension::kTransportSequenceNumber) { in VideoReceiveStream()
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| D | video_send_stream.cc | 138 if (extension.name == RtpExtension::kTransportSequenceNumber) { in VideoSendStream()
185 } else if (extension == RtpExtension::kTransportSequenceNumber) { in VideoSendStream()
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| D | end_to_end_tests.cc | 1500 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F()
1527 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F()
1606 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in ModifyVideoConfigs()
1616 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in ModifyAudioConfigs()
3314 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F()
3323 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F()
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| D | video_send_stream_tests.cc | 243 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); in TEST_F()
406 RtpExtension(RtpExtension::kTransportSequenceNumber, in ModifyVideoConfigs()
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| D | video_quality_test.cc | 808 RtpExtension(RtpExtension::kTransportSequenceNumber, in SetupCommon()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_unittest.cc | 48 const uint16_t kTransportSequenceNumber = 0xaabbu; variable
559 rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber)); in TEST_F()
606 EXPECT_EQ(kTransportSequenceNumber, in TEST_F()
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| /external/webrtc/talk/media/webrtc/ |
| D | webrtcvideoengine2_unittest.cc | 1226 webrtc::RtpExtension::kTransportSequenceNumber); in TEST_F()
1232 webrtc::RtpExtension::kTransportSequenceNumber); in TEST_F()
|