| /external/webrtc/webrtc/voice_engine/ |
| voe_base_impl.h | 73 int64_t* elapsed_time_ms,
102 int64_t* elapsed_time_ms,
134 void* audio_data, int64_t* elapsed_time_ms,
|
| voe_base_impl.cc | 104 int64_t* elapsed_time_ms,
107 audioSamples, elapsed_time_ms, ntp_time_ms); local
169 void* audio_data, int64_t* elapsed_time_ms,
175 audio_data, elapsed_time_ms, ntp_time_ms);
769 void* audio_data, int64_t* elapsed_time_ms,
792 *elapsed_time_ms = audioFrame_.elapsed_time_ms_;
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| mock_delay_manager.h | 42 void(int elapsed_time_ms));
|
| /external/webrtc/webrtc/modules/pacing/ |
| bitrate_prober.cc | 88 int64_t elapsed_time_ms = now_ms - time_last_send_ms_; local
96 time_until_probe_ms = next_delta_ms - elapsed_time_ms;
|
| paced_sender.cc | 352 int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000; local
353 return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
359 int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000; local
363 if (!paused_ && elapsed_time_ms > 0) {
380 int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
|
| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 66 int64_t* elapsed_time_ms,
131 int64_t* elapsed_time_ms,
|
| /external/webrtc/webrtc/test/ |
| fake_audio_device.cc | 118 int64_t elapsed_time_ms = -1; local
127 &elapsed_time_ms,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| unittest.cc | 142 int elapsed_time_ms = 0; local
143 for (int i = 0; elapsed_time_ms < 10000; ++i) {
163 const int send_time = elapsed_time_ms * (sample_rate_hz / 1000);
179 elapsed_time_ms += duration1_ms;
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| delay_manager.cc | 341 void DelayManager::UpdateCounters(int elapsed_time_ms) {
342 packet_iat_count_ms_ += elapsed_time_ms;
343 peak_detector_.IncrementCounter(elapsed_time_ms);
344 max_timer_ms_ += elapsed_time_ms;
|
| delay_manager.h | 79 // |elapsed_time_ms| have elapsed.
80 virtual void UpdateCounters(int elapsed_time_ms);
|
| decision_logic.cc | 155 const int elapsed_time_ms = local
157 delay_manager_->UpdateCounters(elapsed_time_ms);
|
| /external/webrtc/talk/app/webrtc/test/ |
| fakeaudiocapturemodule_unittest.cc | 89 int64_t* elapsed_time_ms,
98 *elapsed_time_ms = 0;
|
| fakeaudiocapturemodule.cc | 702 int64_t elapsed_time_ms = 0; local
707 &elapsed_time_ms, &ntp_time_ms) != 0) {
|
| /external/webrtc/webrtc/modules/audio_device/test/ |
| func_test_manager.h | 105 int64_t* elapsed_time_ms,
|
| audio_device_test_api.cc | 117 int64_t* elapsed_time_ms,
[all...] |
| func_test_manager.cc | 346 int64_t* elapsed_time_ms,
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ |
| audio_device_buffer.cc | 537 int64_t elapsed_time_ms = -1; local
545 &elapsed_time_ms,
|
| /external/webrtc/talk/media/base/ |
| testutils.cc | 146 uint32_t elapsed_time_ms = 0; local
155 RtpDumpPacket dump_packet(buf.Data(), buf.Length(), elapsed_time_ms, rtcp);
156 elapsed_time_ms += kElapsedTimeInterval;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| conference_transport.cc | 208 int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_); local
209 int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_device_unittest.cc | 400 int64_t* elapsed_time_ms,
452 int64_t* elapsed_time_ms,
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 390 int64_t* elapsed_time_ms,
444 int64_t* elapsed_time_ms,
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvideoengine2.cc | 2445 int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame \/ local
[all...] |
