| /external/perfetto/src/trace_processor/importers/ninja/ |
| D | ninja_log_parser.cc | 110 *t_start == jobs_.back().start_ms && *t_end == jobs_.back().end_ms) { in Parse()
126 [](const Job& x, const Job& y) { return x.start_ms < y.start_ms; }); in NotifyEndOfFile()
158 if (cur.busy_until <= job.start_ms) { in NotifyEndOfFile()
180 const int64_t start_ns = job.start_ms * kMsToNs; in NotifyEndOfFile()
181 const int64_t dur_ns = (job.end_ms - job.start_ms) * kMsToNs; in NotifyEndOfFile()
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| D | ninja_log_parser.h | 58 : build_id(b), start_ms(s), end_ms(e), hash(h), names(n) {} in Job()
61 int64_t start_ms; member
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| /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/micro/examples/micro_speech/ |
| D | audio_provider_mock.cc | 27 int start_ms, int duration_ms, in GetAudioSamples() argument
34 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/tensorflow/tensorflow/lite/micro/examples/micro_speech/arduino/ |
| D | audio_provider.cc | 80 int start_ms, int duration_ms, in GetAudioSamples() argument
99 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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| /external/tensorflow/tensorflow/core/profiler/utils/ |
| D | timespan.h | 113 inline Timespan MilliSpan(double start_ms, double end_ms) { in MilliSpan() argument
114 return PicoSpan(MillisToPicos(start_ms), MillisToPicos(end_ms)); in MilliSpan()
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| /external/tensorflow/tensorflow/lite/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/micro/examples/micro_speech/disco_f746ng/ |
| D | audio_provider.cc | 153 int start_ms, int duration_ms, in GetAudioSamples() argument
168 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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| /external/tensorflow/tensorflow/lite/micro/examples/micro_speech/apollo3evb/ |
| D | audio_provider.cc | 478 int start_ms, int duration_ms, in GetAudioSamples() argument
506 (start_ms < 0) ? 0 : 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 | 73 static ulong bytes_per_second(unsigned int len, ulong start_ms) in bytes_per_second() argument
77 return len / (max(get_timer(start_ms) / 1024, 1UL)); in bytes_per_second()
79 return 1024 * len / max(get_timer(start_ms), 1UL); in bytes_per_second()
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| /external/tensorflow/tensorflow/lite/micro/examples/micro_speech/sparkfun_edge/ |
| D | audio_provider.cc | 311 int start_ms, int duration_ms, in GetAudioSamples() argument
335 const int start_offset = start_ms * (kAudioSampleFrequency / 1000); in GetAudioSamples()
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| /external/tensorflow/tensorflow/lite/micro/examples/micro_speech/nxp_k66f/ |
| D | audio_provider.cc | 353 int start_ms, int duration_ms, in GetAudioSamples() argument
368 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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| /external/v8/src/heap/ |
| D | heap.cc | 3566 double start_ms, double deadline_in_ms) { in IdleNotificationEpilogue() argument
3567 double idle_time_in_ms = deadline_in_ms - start_ms; in IdleNotificationEpilogue()
3625 double start_ms = MonotonicallyIncreasingTimeInMs(); in IdleNotification() local
3626 double idle_time_in_ms = deadline_in_ms - start_ms; in IdleNotification()
3628 tracer()->SampleAllocation(start_ms, NewSpaceAllocationCounter(), in IdleNotification()
3639 IdleNotificationEpilogue(action, heap_state, start_ms, deadline_in_ms); in IdleNotification()
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| D | heap.h | 1746 GCIdleTimeHeapState heap_state, double start_ms,
|