/external/webrtc/webrtc/system_wrappers/source/ |
D | timestamp_extrapolator.cc | 17 TimestampExtrapolator::TimestampExtrapolator(int64_t start_ms) in TimestampExtrapolator() argument 34 Reset(start_ms); in TimestampExtrapolator() 42 void TimestampExtrapolator::Reset(int64_t start_ms) in Reset() argument 45 _startMs = start_ms; in Reset()
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D | condition_variable_unittest.cc | 194 int64_t start_ms = TickTime::MillisecondTimestamp(); in TEST() local 197 EXPECT_LE(start_ms + kVeryShortWaitMs, end_ms) in TEST() 198 << "actual elapsed:" << end_ms - start_ms; in TEST()
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/external/webrtc/webrtc/system_wrappers/include/ |
D | timestamp_extrapolator.h | 23 explicit TimestampExtrapolator(int64_t start_ms); 27 void Reset(int64_t start_ms);
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/external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/ |
D | audio_provider_mock.cc | 29 int start_ms, int duration_ms, in GetAudioSamples() argument 36 const int start_sample = (start_ms * kAudioSampleFrequency) / 1000; in GetAudioSamples()
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D | audio_provider.h | 33 int start_ms, int duration_ms,
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D | audio_provider.cc | 26 int start_ms, int duration_ms, in GetAudioSamples() argument
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/external/v8/src/heap/ |
D | scavenge-job.cc | 28 double start_ms = heap->MonotonicallyIncreasingTimeInMs(); in RunInternal() local 29 double idle_time_in_ms = deadline_in_ms - start_ms; in RunInternal()
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D | heap.cc | 3457 double start_ms, double deadline_in_ms) { in IdleNotificationEpilogue() argument 3458 double idle_time_in_ms = deadline_in_ms - start_ms; in IdleNotificationEpilogue() 3506 double start_ms = MonotonicallyIncreasingTimeInMs(); in IdleNotification() local 3507 double idle_time_in_ms = deadline_in_ms - start_ms; in IdleNotification() 3509 tracer()->SampleAllocation(start_ms, NewSpaceAllocationCounter(), in IdleNotification() 3519 IdleNotificationEpilogue(action, heap_state, start_ms, deadline_in_ms); in IdleNotification()
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D | heap.h | 1716 GCIdleTimeHeapState heap_state, double start_ms,
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/external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/osx/ |
D | audio_provider.cc | 108 int start_ms, int duration_ms, in GetAudioSamples() argument 126 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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/external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/disco_f746ng/ |
D | audio_provider.cc | 154 int start_ms, int duration_ms, in GetAudioSamples() argument 169 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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/external/webrtc/talk/media/base/ |
D | rtpdump.cc | 48 RtpDumpFileHeader::RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p) in RtpDumpFileHeader() argument 49 : start_sec(start_ms / 1000), in RtpDumpFileHeader() 50 start_usec(start_ms % 1000 * 1000), in RtpDumpFileHeader()
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D | rtpdump.h | 59 RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p);
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/external/u-boot/cmd/ |
D | sf.c | 71 static ulong bytes_per_second(unsigned int len, ulong start_ms) in bytes_per_second() argument 75 return len / (max(get_timer(start_ms) / 1024, 1UL)); in bytes_per_second() 77 return 1024 * len / max(get_timer(start_ms), 1UL); in bytes_per_second()
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/external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/sparkfun_edge/ |
D | audio_provider.cc | 312 int start_ms, int duration_ms, in GetAudioSamples() argument 336 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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/external/tensorflow/tensorflow/core/util/ |
D | stats_calculator.cc | 71 const double start_ms = detail.start_us.avg() / 1000.0; in ColumnString() local 80 InitField(stream, 17) << start_ms; in ColumnString()
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/external/webrtc/webrtc/test/ |
D | rtp_file_reader.cc | 513 uint32_t CalcTimeDelta(uint32_t ts_sec, uint32_t ts_usec, uint32_t start_ms) { in CalcTimeDelta() argument 517 uint64_t t1_ms = static_cast<uint64_t>(start_ms); in CalcTimeDelta()
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/external/webrtc/webrtc/base/ |
D | socket_unittest.cc | 933 uint32_t start_ms = Time(); in UdpReadyToSend() local 936 while (start_ms + kTimeout > Time()) { in UdpReadyToSend()
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