/external/webrtc/webrtc/modules/audio_processing/transient/ |
transient_suppressor_unittest.cc | 30 for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) { 44 for (int time_ms = 0; time_ms < 990; time_ms += ts::kChunkSizeMs) { 61 for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) { 73 for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) [all...] |
/external/v8/src/ |
date.h | 56 // Computes floor(time_ms / kMsPerDay). 57 static int DaysFromTime(int64_t time_ms) { 58 if (time_ms < 0) time_ms -= (kMsPerDay - 1); 59 return static_cast<int>(time_ms / kMsPerDay); 63 // Computes modulo(time_ms, kMsPerDay) given that 64 // days = floor(time_ms / kMsPerDay). 65 static int TimeInDay(int64_t time_ms, int days) { 66 return static_cast<int>(time_ms - days * kMsPerDay); 92 const char* LocalTimezone(int64_t time_ms) { 206 double time_ms = static_cast<double>(time_sec * 1000); local [all...] |
/hardware/qcom/gps/msm8998/utils/platform_lib_abstractions/loc_stub/src/ |
loc_stub_time.cpp | 45 int64_t time_ms = 0; local 47 time_ms += (ts.tv_sec * 1000000000LL); /* Seconds to nanoseconds */ 48 time_ms += ts.tv_nsec; /* Add Nanoseconds */ 49 return time_ms;
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/hardware/qcom/gps/sdm845/utils/platform_lib_abstractions/loc_stub/src/ |
loc_stub_time.cpp | 45 int64_t time_ms = 0; local 47 time_ms += (ts.tv_sec * 1000000000LL); /* Seconds to nanoseconds */ 48 time_ms += ts.tv_nsec; /* Add Nanoseconds */ 49 return time_ms;
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/external/webrtc/webrtc/modules/bitrate_controller/ |
bitrate_controller_unittest.cc | 105 int64_t time_ms = 1001; local 112 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms); 114 time_ms += 2000; 121 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms); 125 time_ms += 1000; 129 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms); 133 time_ms += 1000; 137 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms); 139 time_ms += 1000; 143 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms); 177 int64_t time_ms = 1; local 280 int64_t time_ms = 1001; local [all...] |
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
bwe_test_baselinefile.h | 26 virtual void Estimate(int64_t time_ms, uint32_t estimate_bps) = 0;
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packet_sender.h | 74 void RunFor(int64_t time_ms, Packets* in_out) override; 89 void ProcessFeedbackAndGeneratePackets(int64_t time_ms, 110 void RunFor(int64_t time_ms, Packets* in_out) override; 145 void RunFor(int64_t time_ms, Packets* in_out) override; 155 time_ms(packet.send_time_ms()) {} 158 : sequence_number(seq_num), time_ms(now_ms) {} 166 int64_t time_ms; // Time of when the packet left the sender, or when the member in struct:webrtc::testing::bwe::TcpSender::InFlight
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packet_sender.cc | 97 void VideoSender::RunFor(int64_t time_ms, Packets* in_out) { 99 in_out, clock_.TimeInMilliseconds() + time_ms, source_->flow_id()); 100 ProcessFeedbackAndGeneratePackets(time_ms, &feedbacks, in_out); 104 int64_t time_ms, 109 int64_t time_to_run_ms = std::min<int64_t>(time_ms, 100); 114 std::max<int64_t>(std::min(time_ms, time_until_feedback_ms), 0); 136 time_ms -= time_to_run_ms; 137 } while (time_ms > 0); 176 void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) { 177 int64_t end_time_ms = clock_.TimeInMilliseconds() + time_ms; [all...] |
bwe_test_baselinefile.cc | 67 virtual void Estimate(int64_t time_ms, uint32_t estimate_bps) { 71 if (reader_->Read(&read_ms) && read_ms == time_ms && 75 static_cast<uint32_t>(time_ms), estimate_bps, read_bps); 112 virtual void Estimate(int64_t time_ms, uint32_t estimate_bps) { 113 verifier_->Estimate(time_ms, estimate_bps); 114 output_content_.push_back(static_cast<uint32_t>(time_ms));
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metric_recorder.h | 49 time_ms(0), 54 time_ms = now_ms; 60 int64_t time_ms; member in struct:webrtc::testing::bwe::PlotInformation 79 int64_t time_ms, 85 void UpdateTimeMs(int64_t time_ms);
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/external/v8/src/heap/ |
memory-reducer.cc | 31 double time_ms = heap->MonotonicallyIncreasingTimeInMs(); local 32 heap->tracer()->SampleAllocation(time_ms, heap->NewSpaceAllocationCounter(), 43 event.time_ms = time_ms; 86 ScheduleTimer(event.time_ms, state_.next_gc_start_ms - event.time_ms); 90 state_.next_gc_start_ms - event.time_ms); 102 ScheduleTimer(event.time_ms, state_.next_gc_start_ms - event.time_ms); 119 ScheduleTimer(event.time_ms, state_.next_gc_start_ms - event.time_ms) [all...] |
memory-reducer.h | 106 double time_ms; member in struct:v8::internal::MemoryReducer::Event 126 void ScheduleTimer(double time_ms, double delay_ms);
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
packet.h | 34 // when the Packet object is deleted. The |time_ms| is an extra time 39 double time_ms, 51 double time_ms, 58 Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms); 63 double time_ms); 98 double time_ms() const { return time_ms_; } function in class:webrtc::test::Packet
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packet.cc | 23 double time_ms, 31 time_ms_(time_ms) { 38 double time_ms, 46 time_ms_(time_ms) { 50 Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms) 57 time_ms_(time_ms) { 65 double time_ms) 72 time_ms_(time_ms) {
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/hardware/invensense/6515/libsensors_iio/software/core/mpl/ |
motion_no_motion.h | 23 inv_error_t inv_set_no_motion_time(long time_ms);
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shake.h | 84 void inv_set_shake_time_min_ms(long time_ms); 85 void inv_set_shake_time_max_ms(long time_ms);
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/hardware/invensense/65xx/libsensors_iio/software/core/mpl/ |
motion_no_motion.h | 23 inv_error_t inv_set_no_motion_time(long time_ms);
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shake.h | 84 void inv_set_shake_time_min_ms(long time_ms); 85 void inv_set_shake_time_max_ms(long time_ms);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
bwe_rtp_play.cc | 66 first_rtp_time_ms = packet.time_ms; 67 packet.time_ms = packet.time_ms - first_rtp_time_ms; 91 packet.time_ms = packet.time_ms - first_rtp_time_ms; 92 next_rtp_time_ms = packet.time_ms;
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rtp_to_text.cc | 46 ss << static_cast<int64_t>(packet.time_ms) * 1000000; 55 packet.time_ms,
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
loudest_filter.h | 40 void RemoveTimeoutStreams(uint32_t time_ms);
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loudest_filter.cc | 17 void LoudestFilter::RemoveTimeoutStreams(uint32_t time_ms) { 20 if (rtc::TimeDiff(time_ms, it->second.last_time_ms) >
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/external/webrtc/webrtc/modules/audio_coding/codecs/tools/ |
audio_codec_speed_test.cc | 101 float time_ms; local 108 time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0], 110 encoding_time_ms_ += time_ms; 111 time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]); 112 decoding_time_ms_ += time_ms;
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/external/webrtc/webrtc/test/ |
rtp_file_reader.h | 31 uint32_t time_ms; member in struct:webrtc::test::RtpPacket
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/external/webrtc/webrtc/modules/desktop_capture/ |
desktop_frame.h | 55 void set_capture_time_ms(int64_t time_ms) { capture_time_ms_ = time_ms; }
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