/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
H A D | rate_statistics_unittest.cc | 25 int64_t now_ms = 0; local 27 EXPECT_EQ(0u, stats_.Rate(now_ms)); 28 stats_.Update(1500, now_ms); 30 EXPECT_EQ(24000u, stats_.Rate(now_ms)); 33 EXPECT_EQ(0u, stats_.Rate(now_ms)); 35 if (now_ms % 10 == 0) { 36 stats_.Update(1500, now_ms); 40 if (now_ms > 0 && now_ms % 500 == 0) { 41 EXPECT_NEAR(1200000u, stats_.Rate(now_ms), 2400 52 int64_t now_ms = 0; local [all...] |
H A D | rate_statistics.cc | 38 void RateStatistics::Update(uint32_t count, int64_t now_ms) { argument 39 if (now_ms < oldest_time_) { 44 EraseOld(now_ms); 46 int now_offset = static_cast<int>(now_ms - oldest_time_); 56 uint32_t RateStatistics::Rate(int64_t now_ms) { argument 57 EraseOld(now_ms); 61 void RateStatistics::EraseOld(int64_t now_ms) { argument 62 int64_t new_oldest_time = now_ms - num_buckets_ + 1;
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H A D | remote_bitrate_estimator_single_stream.cc | 82 void UpdateEstimate(int64_t now_ms); 115 int64_t now_ms = clock_->TimeInMilliseconds(); local 127 std::make_pair(OveruseDetector(OverUseDetectorOptions()), now_ms))); 130 SetPacketTimeMs(it, now_ms); 132 incoming_bitrate_.Update(payload_size, now_ms); 136 unsigned int incoming_bitrate = incoming_bitrate_.Rate(now_ms); 138 remote_rate_.TimeToReduceFurther(now_ms, incoming_bitrate)) { 142 UpdateEstimate(now_ms); 151 int64_t now_ms = clock_->TimeInMilliseconds(); local 152 UpdateEstimate(now_ms); 164 UpdateEstimate(int64_t now_ms) argument [all...] |
H A D | remote_rate_control.cc | 86 uint32_t RemoteRateControl::UpdateBandwidthEstimate(int64_t now_ms) { argument 90 now_ms); 99 int64_t now_ms) { 107 time_first_incoming_estimate_ = now_ms; 109 } else if (now_ms - time_first_incoming_estimate_ > 500 && 131 int64_t now_ms) { 136 UpdateChangePeriod(now_ms); 137 ChangeState(current_input_, now_ms); 161 double alpha = RateIncreaseFactor(now_ms, last_bit_rate_change_, 172 last_bit_rate_change_ = now_ms; 98 Update(const RateControlInput* input, int64_t now_ms) argument 128 ChangeBitRate(uint32_t current_bit_rate, uint32_t incoming_bit_rate, double noise_var, int64_t now_ms) argument 217 RateIncreaseFactor(int64_t now_ms, int64_t last_ms, uint32_t reaction_time_ms, double noise_var) const argument 253 UpdateChangePeriod(int64_t now_ms) argument 286 ChangeState(const RateControlInput& input, int64_t now_ms) argument [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | network_predictor.cc | 23 int64_t now_ms = clock_->TimeInMilliseconds(); local 26 static_cast<float>(now_ms - last_loss_rate_update_time_ms_), 28 last_loss_rate_update_time_ms_ = now_ms;
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/external/chromium_org/third_party/android_crazy_linker/src/tests/ |
H A D | bench_load_library.cpp | 29 static double now_ms() { function 50 start_ms_ = now_ms(); 54 double elapsed_ms = now_ms() - start_ms_;
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/external/chromium_org/net/quic/ |
H A D | quic_server_session.cc | 77 int64 now_ms = now.Subtract(last_server_config_update_time_).ToMilliseconds(); local 78 if (now_ms < (kMinIntervalBetweenServerConfigUpdatesRTTs * srtt_ms) || 79 now_ms < kMinIntervalBetweenServerConfigUpdatesMs) {
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/external/chromium_org/net/tools/quic/ |
H A D | quic_server_session.cc | 77 int64 now_ms = now.Subtract(last_server_config_update_time_).ToMilliseconds(); local 78 if (now_ms < (kMinIntervalBetweenServerConfigUpdatesRTTs * srtt_ms) || 79 now_ms < kMinIntervalBetweenServerConfigUpdatesMs) {
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
H A D | send_side_bandwidth_estimation.cc | 97 uint32_t now_ms) { 122 time_last_receiver_block_ms_ = now_ms; 123 UpdateEstimate(now_ms); 126 void SendSideBandwidthEstimation::UpdateEstimate(uint32_t now_ms) { argument 127 UpdateMinHistory(now_ms); 155 if ((now_ms - time_last_decrease_ms_) >= 158 time_last_decrease_ms_ = now_ms; 178 void SendSideBandwidthEstimation::UpdateMinHistory(uint32_t now_ms) { argument 183 now_ms - min_bitrate_history_.front().first + 1 > 195 min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate 94 UpdateReceiverBlock(uint8_t fraction_loss, uint32_t rtt, int number_of_packets, uint32_t now_ms) argument [all...] |
H A D | bitrate_controller_impl.cc | 37 int64_t now_ms) OVERRIDE { 72 total_number_of_packets, now_ms); 250 const uint32_t now_ms) { 253 fraction_loss, rtt, number_of_packets, now_ms); 246 OnReceivedRtcpReceiverReport( const uint8_t fraction_loss, const uint32_t rtt, const int number_of_packets, const uint32_t now_ms) argument
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/external/chromium_org/third_party/webrtc/modules/video_render/ |
H A D | incoming_video_stream.cc | 124 int64_t now_ms = TickTime::MillisecondTimestamp(); local 125 if (now_ms >= last_rate_calculation_time_ms_ + KFrameRatePeriodMs) { 128 (now_ms - last_rate_calculation_time_ms_)); 130 last_rate_calculation_time_ms_ = now_ms;
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/external/chromium_org/third_party/webrtc/test/ |
H A D | fake_audio_device.cc | 118 int64_t now_ms = clock_->TimeInMilliseconds(); local 119 uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | timing.cc | 142 int64_t now_ms) { 144 int32_t time_diff_ms = codec_timer_.StopTimer(start_time_ms, now_ms); 150 void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) { argument 152 ts_extrapolator_->Update(now_ms, time_stamp); 155 int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) 158 const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms); 163 int64_t now_ms) const { 167 estimated_complete_time_ms = now_ms; 183 uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) 187 const int64_t max_wait_time_ms = render_time_ms - now_ms 140 StopDecodeTimer(uint32_t time_stamp, int64_t start_time_ms, int64_t now_ms) argument [all...] |
H A D | receiver.cc | 129 const int64_t now_ms = clock_->TimeInMilliseconds(); local 131 next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms); 137 } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) { 138 int frame_delay = static_cast<int>(std::abs(next_render_time_ms - now_ms)); 319 const int64_t now_ms = clock_->TimeInMilliseconds(); local 322 uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms); 323 uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
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H A D | media_optimization.cc | 358 const int64_t now_ms = clock_->TimeInMilliseconds(); local 359 PurgeOldFrameSamples(now_ms); 360 UpdateSentBitrate(now_ms); 376 const int64_t now_ms = clock_->TimeInMilliseconds(); local 377 PurgeOldFrameSamples(now_ms); 384 encoded_frame_samples_.back().time_complete_ms = now_ms; 387 EncodedFrameSample(encoded_length, timestamp, now_ms)); 389 UpdateSentBitrate(now_ms); 519 void MediaOptimization::PurgeOldFrameSamples(int64_t now_ms) { argument 521 if (now_ms 530 UpdateSentBitrate(int64_t now_ms) argument [all...] |
H A D | jitter_buffer.cc | 612 int64_t now_ms = clock_->TimeInMilliseconds(); local 618 inter_frame_delay_.Reset(now_ms); 645 waiting_for_completion_.latest_packet_time = now_ms; 647 waiting_for_completion_.latest_packet_time + 2000 <= now_ms) { 665 now_ms,
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/external/chromium_org/third_party/webrtc/video/ |
H A D | rampup_tests.cc | 245 int64_t now_ms = clock_->TimeInMilliseconds(); local 246 if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass. 249 8 * 1000 / (now_ms - interval_start_ms_); 255 last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000; 258 interval_start_ms_ = now_ms;
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H A D | call_perf_tests.cc | 140 int64_t now_ms = clock_->TimeInMilliseconds(); variable 150 std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); 161 int64_t time_since_creation = now_ms - creation_time_ms_; 168 first_time_in_sync_ = now_ms; 343 int64_t now_ms = clock_->TimeInMilliseconds(); local 344 int64_t time_since_creation = now_ms - creation_time_ms_;
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/external/chromium_org/third_party/webrtc/modules/pacing/ |
H A D | paced_sender.cc | 227 int64_t now_ms = clock_->TimeInMilliseconds(); local 228 int64_t oldest_packet_enqueue_time = now_ms; 244 return now_ms - oldest_packet_enqueue_time;
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/external/chromium_org/third_party/webrtc/system_wrappers/source/ |
H A D | clock.cc | 285 int64_t now_ms = TimeInMilliseconds(); local 286 seconds = (now_ms / 1000) + kNtpJan1970; 288 static_cast<uint32_t>((now_ms % 1000) * kMagicNtpFractionalUnit / 1000);
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/test/ |
H A D | bwe_test_framework_unittest.cc | 445 int64_t now_ms = 0; local 454 packets.push_back(Packet(now_ms * 1000, sequence_number++)); 455 now_ms += 5 * stddev_jitter_ms; 508 int64_t now_ms = 0; local 510 for (uint32_t i = 0; i < kPacketCount; ++i, now_ms += 10) { 511 packets.push_back(Packet(now_ms * 1000, sequence_number++)); 519 filter.RunFor(now_ms, &packets);
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | rtp_player.cc | 109 int64_t now_ms = clock_->TimeInMilliseconds(); local 114 packet->resend_time_ms() + 10 < now_ms) { 152 int64_t now_ms = clock_->TimeInMilliseconds(); local 156 MaskWord64ToUWord32(now_ms));
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender.cc | 575 int64_t now_ms = clock_->TimeInMilliseconds(); local 583 padding_packet, length, rtp_header, now_ms - capture_time_ms); 586 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms); 832 int64_t now_ms = clock_->TimeInMilliseconds(); local 833 int64_t diff_ms = now_ms - capture_time_ms; 836 UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms); 919 int64_t now_ms = clock_->TimeInMilliseconds(); local 926 rtp_header, now_ms - capture_time_ms); 930 rtp_header, now_ms); 952 UpdateDelayStatistics(capture_time_ms, now_ms); 966 UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) argument [all...] |
/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_encoder.cc | 875 int64_t now_ms = TickTime::MillisecondTimestamp(); local 876 if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
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