| /external/webrtc/modules/video_coding/timing/ |
| D | timestamp_extrapolator_unittest.cc | 45 uint32_t rtp = 90000; in TEST() local
66 uint32_t rtp = 90000; in TEST() local
88 uint32_t rtp = std::numeric_limits<uint32_t>::max(); in TEST() local
130 uint32_t rtp = 90000; in TEST() local
157 uint32_t rtp = 90000; in TEST() local
183 uint32_t rtp = 90000; in TEST() local
|
| D | inter_frame_delay_unittest.cc | 49 uint32_t rtp = 1500; in TEST() local
64 uint32_t rtp = 90000; in TEST() local
107 uint32_t rtp = std::numeric_limits<uint32_t>::max() - 1500; in TEST() local
129 uint32_t rtp = 0; in TEST() local
171 uint32_t rtp = std::numeric_limits<uint32_t>::max() - 1500; in TEST() local
|
| /external/webrtc/video/ |
| D | task_queue_frame_decode_scheduler_unittest.cc | 39 const uint32_t rtp = 90000; in TEST() local
64 const uint32_t rtp = 90000; in TEST() local
84 const uint32_t rtp = 90000; in TEST() local
|
| D | task_queue_frame_decode_scheduler.cc | 37 uint32_t rtp, in ScheduleFrame()
|
| D | decode_synchronizer.cc | 82 uint32_t rtp, in ScheduleFrame()
|
| /external/webrtc/media/base/ |
| D | rtp_utils.cc | 74 void UpdateRtpAuthTag(uint8_t* rtp, in UpdateRtpAuthTag()
186 bool ValidateRtpHeader(const uint8_t* rtp, in ValidateRtpHeader()
240 bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp, in UpdateRtpAbsSendTimeExtension()
|
| /external/webrtc/audio/voip/test/ |
| D | audio_egress_unittest.cc | 125 RtpPacketReceived rtp; in TEST_F() local
163 RtpPacketReceived rtp; in TEST_F() local
263 auto is_dtmf = [&](RtpPacketReceived& rtp) { in TEST_F()
274 RtpPacketReceived rtp; in TEST_F() local
|
| D | audio_channel_unittest.cc | 130 RtpPacketReceived rtp; in TEST_F() local
|
| /external/webrtc/rtc_tools/rtp_generator/configs/ |
| D | vp9.json | 8 "rtp" : { object
|
| D | vp8.json | 8 "rtp" : { object
|
| /external/webrtc/call/ |
| D | flexfec_receive_stream.h | 46 ReceiveStreamRtpConfig rtp; member
|
| D | audio_send_stream.h | 109 } rtp; member
|
| /external/webrtc/test/fuzzers/configs/replay_packet_fuzzer/ |
| D | vp8_config.json | 10 "rtp" : { object
|
| D | vp9_config.json | 10 "rtp" : { object
|
| D | h264_non_interleaved_config.json | 36 "rtp" : { object
|
| D | h264_single_nal_config.json | 36 "rtp" : { object
|
| D | vp8_fec_config.json | 11 "rtp" : { object
|
| D | vp9_fec_config.json | 16 "rtp" : { object
|
| D | h264_fec_config.json | 76 "rtp" : { object
|
| /external/webrtc/rtc_tools/rtp_generator/ |
| D | rtp_generator.h | 51 RtpConfig rtp; member
|
| /external/tensorflow/tensorflow/compiler/xla/mlir_hlo/lib/Dialect/mhlo/transforms/ |
| D | legalize_to_linalg_utils.cc | 92 Value preSparsify(Operation* op, llvm::SmallVector<Value, 2>& values, Type rtp, in preSparsify()
|
| /external/skia/tests/ |
| D | WrappedSurfaceCopyOnWriteTest.cpp | 77 GrRenderTargetProxy* rtp = skgpu::ganesh::TopDeviceTargetProxy(surf->getCanvas()); in DEF_GANESH_TEST_FOR_ALL_CONTEXTS() local
|
| /external/webrtc/logging/rtc_event_log/events/ |
| D | logged_rtp_rtcp.h | 64 LoggedRtpPacket rtp; member
77 LoggedRtpPacket rtp; member
|
| /external/libsrtp2/test/ |
| D | rtp_decoder.c | 689 int rtp; in rtp_decoder_handle_pkt() local
|
| /external/webrtc/modules/pacing/ |
| D | packet_router_unittest.cc | 404 NiceMock<MockRtpRtcpInterface> rtp; in TEST_F() local
566 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
|