Searched refs:start_usec (Results 1 – 6 of 6) sorted by relevance
| /external/tensorflow/tensorflow/core/distributed_runtime/rpc/ |
| D | grpc_remote_worker.cc | 130 int64 start_usec = Env::Default()->NowMicros(); in RecvBufAsync() local
138 wrapper_done = [this, request, response, done, start_usec](Status s) { in RecvBufAsync()
148 int64 send_start_usec = start_usec; in RecvBufAsync()
152 start_usec, static_cast<int64>(response->send_start_micros())); in RecvBufAsync()
194 int64 start_usec = Env::Default()->NowMicros(); in RecvTensorAsync() local
202 wrapper_done = [this, request, response, done, start_usec](Status s) { in RecvTensorAsync()
207 int64 send_start_usec = start_usec; in RecvTensorAsync()
223 start_usec, in RecvTensorAsync()
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| /external/webrtc/talk/media/base/ |
| D | rtpdump.cc | 50 start_usec(start_ms % 1000 * 1000), in RtpDumpFileHeader()
58 buf->WriteUInt32(start_usec); in WriteToByteBuffer()
179 uint32_t start_usec; in ReadFileHeader() local
181 buf.ReadUInt32(&start_usec); in ReadFileHeader()
182 start_time_ms_ = start_sec * 1000 + start_usec / 1000; in ReadFileHeader()
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| D | rtpdump.h | 65 uint32_t start_usec; // start of recording, the microseconds part. member
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| /external/webrtc/webrtc/modules/audio_coding/test/ |
| D | RTPFile.cc | 153 uint32_t start_sec, start_usec, source; in ReadHeader() local
159 EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile)); in ReadHeader()
160 start_usec = ntohl(start_usec); in ReadHeader()
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| /external/kmod/testsuite/ |
| D | testsuite.c | 576 unsigned long long end_usec, start_usec; in test_run_parent_check_outputs() local
609 start_usec = now_usec(); in test_run_parent_check_outputs()
610 end_usec = start_usec + TEST_TIMEOUT_USEC; in test_run_parent_check_outputs()
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| /external/webrtc/webrtc/test/ |
| D | rtp_file_reader.cc | 161 uint32_t start_usec; in Init() local
166 TRY(ReadUint32(&start_usec, file_)); in Init()
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