| /external/webrtc/webrtc/modules/video_coding/ |
| timing_unittest.cc | 41 timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
47 timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
48 clock.TimeInMilliseconds());
59 timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
60 clock.TimeInMilliseconds());
68 timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
69 clock.TimeInMilliseconds());
77 timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
81 timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
82 clock.TimeInMilliseconds());
[all...] |
| media_optimization.cc | 85 new VCMLossProtectionLogic(clock_->TimeInMilliseconds())),
121 loss_prot_logic_->Reset(clock_->TimeInMilliseconds());
126 loss_prot_logic_->Reset(clock_->TimeInMilliseconds());
166 last_change_time_ = clock_->TimeInMilliseconds();
221 clock_->TimeInMilliseconds(), filter_mode, fraction_lost);
313 ProcessIncomingFrameRate(clock_->TimeInMilliseconds());
323 PurgeOldFrameSamples(clock_->TimeInMilliseconds());
330 const int64_t now_ms = clock_->TimeInMilliseconds();
341 const int64_t now_ms = clock_->TimeInMilliseconds();
365 clock_->TimeInMilliseconds());
[all...] |
| receiver.cc | 89 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
102 const int64_t start_time_ms = clock_->TimeInMilliseconds();
116 const int64_t now_ms = clock_->TimeInMilliseconds();
149 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
153 *next_render_time_ms, clock_->TimeInMilliseconds());
256 const int64_t now_ms = clock_->TimeInMilliseconds();
|
| receiver_unittest.cc | 39 new StreamGenerator(0, clock_->TimeInMilliseconds()));
66 (frame_type == kEmptyFrame) ? 1 : 0, clock_->TimeInMilliseconds());
201 int64_t key_frame_inserted = clock_->TimeInMilliseconds();
210 clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
231 int64_t key_frame_inserted = clock_->TimeInMilliseconds();
241 clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
263 int64_t key_frame_inserted = clock_->TimeInMilliseconds();
273 clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
295 int64_t key_frame_inserted = clock_->TimeInMilliseconds();
305 clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds()
[all...] |
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_abs_send_time_unittest.cc | 97 IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0, true);
108 int64_t now_ms = clock_.TimeInMilliseconds();
112 now_ms = clock_.TimeInMilliseconds();
120 now_ms = clock_.TimeInMilliseconds();
133 int64_t now_ms = clock_.TimeInMilliseconds();
138 now_ms = clock_.TimeInMilliseconds();
157 int64_t now_ms = clock_.TimeInMilliseconds();
162 now_ms = clock_.TimeInMilliseconds();
172 now_ms = clock_.TimeInMilliseconds();
186 int64_t now_ms = clock_.TimeInMilliseconds();
[all...] |
| send_time_history.cc | 38 sequence_number, PacketInfo(clock_->TimeInMilliseconds(), 0, -1,
56 if (clock_->TimeInMilliseconds() - it->second.creation_time_ms <=
|
| remote_bitrate_estimator_unittest_helper.cc | 325 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
334 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
362 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
368 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
385 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
387 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
482 int64_t overuse_start_time = clock_.TimeInMilliseconds();
490 bitrate_drop_time = clock_.TimeInMilliseconds();
529 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
550 IncomingPacket(kDefaultSsrc, 100, clock_.TimeInMilliseconds(), timestamp
[all...] |
| remote_estimator_proxy.cc | 72 int64_t now = clock_->TimeInMilliseconds();
86 last_process_time_ms_ = clock_->TimeInMilliseconds();
|
| /external/webrtc/webrtc/voice_engine/ |
| network_predictor.cc | 18 last_loss_rate_update_time_ms_(clock_->TimeInMilliseconds()),
29 int64_t now_ms = clock_->TimeInMilliseconds();
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator.cc | 25 ts_extrapolator_(new TimestampExtrapolator(clock_->TimeInMilliseconds())),
45 // The extrapolator assumes the TimeInMilliseconds time.
46 int64_t receiver_arrival_time_ms = clock_->TimeInMilliseconds();
66 clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds();
68 int64_t now_ms = clock_->TimeInMilliseconds();
|
| rtp_packet_history_unittest.cc | 70 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
83 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
102 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
131 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
149 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
167 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
198 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
228 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
273 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
|
| /external/webrtc/webrtc/modules/pacing/ |
| paced_sender_unittest.cc | 88 int64_t now_ms = clock_->TimeInMilliseconds();
147 clock_.TimeInMilliseconds(),
153 clock_.TimeInMilliseconds(),
159 clock_.TimeInMilliseconds(),
162 int64_t queued_packet_timestamp = clock_.TimeInMilliseconds();
183 clock_.TimeInMilliseconds(),
189 clock_.TimeInMilliseconds(),
193 sequence_number++, clock_.TimeInMilliseconds(),
207 clock_.TimeInMilliseconds(),
213 sequence_number++, clock_.TimeInMilliseconds(),
[all...] |
| paced_sender.cc | 95 time_last_updated_(clock_->TimeInMilliseconds()) {}
308 int64_t now_ms = clock_->TimeInMilliseconds();
335 return clock_->TimeInMilliseconds() - oldest_packet;
340 packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
347 int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
369 packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
437 prober_->PacketSent(clock_->TimeInMilliseconds(), packet.bytes);
451 prober_->PacketSent(clock_->TimeInMilliseconds(), bytes_sent);
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| packet_sender.cc | 48 metric_recorder_->UpdateTimeMs(clock_.TimeInMilliseconds());
99 in_out, clock_.TimeInMilliseconds() + time_ms, source_->flow_id());
112 feedbacks->front()->send_time_ms() - clock_.TimeInMilliseconds();
177 int64_t end_time_ms = clock_.TimeInMilliseconds() + time_ms;
189 feedbacks.front()->send_time_ms() - clock_.TimeInMilliseconds(), 0);
198 if (clock_.TimeInMilliseconds() + time_until_next_event_ms > end_time_ms)
199 time_until_next_event_ms = end_time_ms - clock_.TimeInMilliseconds();
235 } while (clock_.TimeInMilliseconds() < end_time_ms);
288 int64_t pace_out_time_ms = clock_.TimeInMilliseconds();
348 if (clock_.TimeInMilliseconds() + time_ms < offset_ms_)
[all...] |
| /external/webrtc/webrtc/system_wrappers/include/ |
| clock.h | 33 virtual int64_t TimeInMilliseconds() const = 0;
60 int64_t TimeInMilliseconds() const override;
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
| bwe_rtp_play.cc | 30 static_cast<uint32_t>(clock_->TimeInMilliseconds()),
69 if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
84 rbe->IncomingPacket(clock.TimeInMilliseconds(),
100 next_rtp_time_ms - clock.TimeInMilliseconds());
104 clock.TimeInMilliseconds());
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/estimators/ |
| remb.cc | 48 report_blocks, 0, clock_->TimeInMilliseconds());
92 while ((clock_.TimeInMilliseconds() + step_ms) < arrival_time_ms) {
99 clock_.AdvanceTimeMilliseconds(arrival_time_ms - clock_.TimeInMilliseconds());
100 ASSERT_TRUE(arrival_time_ms == clock_.TimeInMilliseconds());
125 clock_.TimeInMilliseconds(), estimated_kbps);
|
| send_side.cc | 60 clock_->TimeInMilliseconds() - feedback.latest_send_time_ms();
62 BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
79 report_blocks, rtt_ms, clock_->TimeInMilliseconds());
|
| /external/webrtc/webrtc/video/ |
| vie_remb.cc | 33 last_remb_time_(clock_->TimeInMilliseconds()),
109 last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
115 int64_t now = clock_->TimeInMilliseconds();
|
| call_stats.cc | 98 last_process_time_(clock_->TimeInMilliseconds()),
107 return last_process_time_ + kUpdateIntervalMs - clock_->TimeInMilliseconds();
112 int64_t now = clock_->TimeInMilliseconds();
165 reports_.push_back(RttTime(rtt, clock_->TimeInMilliseconds()));
|
| overuse_frame_detector.cc | 178 next_process_time_(clock_->TimeInMilliseconds()),
217 return next_process_time_ - clock_->TimeInMilliseconds();
248 int64_t now = clock_->TimeInMilliseconds();
264 clock_->TimeInMilliseconds());
271 int64_t now = clock_->TimeInMilliseconds();
283 int64_t now = clock_->TimeInMilliseconds();
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_receive_test_oldapi.cc | 157 while (clock_.TimeInMilliseconds() < packet->time_ms()) {
209 last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
213 if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
217 last_toggle_time_ms_ = clock_.TimeInMilliseconds();
|
| /external/webrtc/webrtc/modules/video_coding/test/ |
| video_rtp_play.cc | 63 while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
66 clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
|
| /external/webrtc/webrtc/modules/bitrate_controller/ |
| bitrate_controller_impl.cc | 89 last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
140 bandwidth_estimation_.UpdateReceiverEstimate(clock_->TimeInMilliseconds(),
150 clock_->TimeInMilliseconds() - last_bitrate_update_ms_;
160 bandwidth_estimation_.UpdateEstimate(clock_->TimeInMilliseconds());
163 last_bitrate_update_ms_ = clock_->TimeInMilliseconds();
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_performance_test.cc | 76 int64_t start_time_ms = clock->TimeInMilliseconds();
128 int64_t end_time_ms = clock->TimeInMilliseconds();
|
