| /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 | send_delay_stats.cc | 91 bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) {  in OnSentPacket()
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| D | rtp_streams_synchronizer2.cc | 180   int64_t time_ms;  in GetStreamSyncOffsetInMs()  local
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| /external/webrtc/modules/audio_processing/transient/ | 
| D | transient_suppressor_unittest.cc | 51   for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) {  in TEST_P()  local65     for (int time_ms = 0; time_ms < 990; time_ms += ts::kChunkSizeMs) {  in TEST_P()  local
 82     for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) {  in TEST_P()  local
 94   for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) {  in TEST_P()  local
 99   for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) {  in TEST_P()  local
 
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| /external/webrtc/modules/audio_coding/neteq/tools/ | 
| D | packet.cc | 23                double time_ms,  in Packet()33                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 | 85   double time_ms() const { return time_ms_; }  in time_ms()  function
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| /external/pdfium/fxjs/ | 
| D | fx_date_helpers_unittest.cpp | 20     double time_ms;  in TEST()  member44     double time_ms;  in TEST()  member
 
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| /external/cronet/third_party/metrics_proto/ | 
| D | perf_stat.proto | 38     optional uint64 time_ms = 1;  field
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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/webrtc/common_audio/ | 
| D | smoothing_filter.cc | 79 void SmoothingFilterImpl::ExtrapolateLastSample(int64_t time_ms) {  in ExtrapolateLastSample()
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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()  local154   const double time_ms = (TimeTicks::Now() - tstart).InMillisecondsF();  in TEST_F()  local
 
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| /external/cronet/base/trace_event/ | 
| D | memory_dump_scheduler_unittest.cc | 73   const double time_ms = (TimeTicks::Now() - tstart).InMillisecondsF();  in TEST_F()  local157   const double time_ms = (TimeTicks::Now() - tstart).InMillisecondsF();  in TEST_F()  local
 
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| /external/webrtc/test/ | 
| D | rtp_file_reader.h | 31   uint32_t time_ms;  member
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| /external/webrtc/api/video/ | 
| D | video_timing.cc | 21 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 | 88   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/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/congestion_controller/goog_cc/ | 
| D | alr_detector_unittest.cc | 35   SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {  in ForTimeMs()
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| /external/ltp/testcases/kernel/syscalls/perf_event_open/ | 
| D | perf_event_open02.c | 78 static void bench_work(int time_ms)  in bench_work()
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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/webrtc/modules/rtp_rtcp/source/ | 
| D | rtp_sender_audio.cc | 330                                            uint16_t time_ms,  in SendTelephoneEvent()
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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/AFLplusplus/src/ | 
| D | afl-forkserver.c | 788     u32 time_ms = read_s32_timed(fsrv->fsrv_st_fd, &status, fsrv->init_tmout,  in afl_fsrv_start()  local
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| /external/webrtc/audio/ | 
| D | audio_receive_stream.cc | 449     int64_t time_ms) {  in SetEstimatedPlayoutNtpTimestampMs()
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