| /external/webrtc/modules/audio_processing/transient/ |
| D | transient_suppressor_unittest.cc | 30 for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) { in TEST() local
44 for (int time_ms = 0; time_ms < 990; time_ms += ts::kChunkSizeMs) { in TEST() local
61 for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) { in TEST() local
73 for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) { in TEST() local
78 for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) { in TEST() local
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| /external/webrtc/video/ |
| D | encoder_overshoot_detector.cc | 37 int64_t time_ms) { in SetTargetRate()
55 void EncoderOvershootDetector::OnEncodedFrame(size_t bytes, int64_t time_ms) { in OnEncodedFrame()
82 int64_t time_ms, in HandleEncodedFrame()
119 EncoderOvershootDetector::GetNetworkRateUtilizationFactor(int64_t time_ms) { in GetNetworkRateUtilizationFactor()
133 int64_t time_ms) { in GetMediaRateUtilizationFactor()
166 void EncoderOvershootDetector::LeakBits(int64_t time_ms) { in LeakBits()
188 void EncoderOvershootDetector::CullOldUpdates(int64_t time_ms) { in CullOldUpdates()
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| D | rtp_streams_synchronizer2.cc | 169 int64_t time_ms; in GetStreamSyncOffsetInMs() local
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| D | send_delay_stats.cc | 91 bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) { in OnSentPacket()
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| D | rtp_streams_synchronizer.cc | 166 int64_t time_ms; in GetStreamSyncOffsetInMs() local
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | packet.cc | 28 double time_ms, in Packet()
41 double time_ms) in Packet()
48 Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms) in Packet()
58 double time_ms) in Packet()
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| D | neteq_input.h | 36 int64_t time_ms; member
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| D | packet.h | 99 double time_ms() const { return time_ms_; } in time_ms() function
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| /external/tensorflow/tensorflow/lite/micro/tools/make/targets/ecm3531/ |
| D | _main.c | 56 uint64_t time_ms; in _main() local
70 void EtaPrintExecutionTime(uint64_t time_ms) { in EtaPrintExecutionTime()
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| /external/pdfium/fxjs/ |
| D | fx_date_helpers_unittest.cpp | 17 double time_ms; in TEST() member
41 double time_ms; in TEST() member
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| /external/webrtc/common_audio/ |
| D | smoothing_filter.cc | 79 void SmoothingFilterImpl::ExtrapolateLastSample(int64_t time_ms) { in ExtrapolateLastSample()
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| /external/webrtc/modules/audio_coding/codecs/tools/ |
| D | audio_codec_speed_test.cc | 100 float time_ms; in EncodeDecode() local
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| /external/libchrome/base/trace_event/ |
| D | memory_dump_scheduler_unittest.cc | 72 const double time_ms = (TimeTicks::Now() - tstart).InMillisecondsF(); in TEST_F() local
154 const double time_ms = (TimeTicks::Now() - tstart).InMillisecondsF(); in TEST_F() local
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| /external/webrtc/test/ |
| D | rtp_file_reader.h | 29 uint32_t time_ms; member
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| /external/webrtc/api/video/ |
| D | video_timing.cc | 20 uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) { in GetDeltaCappedMs()
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| /external/webrtc/modules/desktop_capture/ |
| D | desktop_frame.h | 77 void set_capture_time_ms(int64_t time_ms) { capture_time_ms_ = time_ms; } in set_capture_time_ms()
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| /external/webrtc/modules/video_coding/test/ |
| D | stream_generator.cc | 35 int64_t time_ms) { in GenerateFrame()
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | alr_detector_unittest.cc | 35 SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) { in ForTimeMs()
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| /external/webrtc/modules/pacing/ |
| D | interval_budget_unittest.cc | 22 size_t TimeToBytes(int bitrate_kbps, int time_ms) { in TimeToBytes()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_audio.cc | 326 uint16_t time_ms, in SendTelephoneEvent()
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| D | rtp_header_extensions.h | 40 static constexpr uint32_t MsTo24Bits(int64_t time_ms) { in MsTo24Bits()
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| /external/grpc-grpc/test/core/util/ |
| D | test_config.cc | 374 gpr_timespec grpc_timeout_milliseconds_to_deadline(int64_t time_ms) { in grpc_timeout_milliseconds_to_deadline()
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| /external/ltp/testcases/kernel/syscalls/perf_event_open/ |
| D | perf_event_open02.c | 109 static void bench_work(int time_ms) in bench_work()
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| /external/jemalloc_new/include/jemalloc/internal/ |
| D | arena_structs_b.h | 30 atomic_zd_t time_ms; member
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| /external/webrtc/audio/ |
| D | audio_receive_stream.cc | 326 int64_t time_ms) { in SetEstimatedPlayoutNtpTimestampMs()
|