| /external/webrtc/webrtc/voice_engine/ |
| D | 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,
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| D | voe_base_impl.cc | 104 int64_t* elapsed_time_ms, in NeedMorePlayData() argument
107 audioSamples, elapsed_time_ms, ntp_time_ms); in NeedMorePlayData()
169 void* audio_data, int64_t* elapsed_time_ms, in PullRenderData() argument
175 audio_data, elapsed_time_ms, ntp_time_ms); in PullRenderData()
769 void* audio_data, int64_t* elapsed_time_ms, in GetPlayoutData() argument
792 *elapsed_time_ms = audioFrame_.elapsed_time_ms_; in GetPlayoutData()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | delay_manager.cc | 341 void DelayManager::UpdateCounters(int elapsed_time_ms) { in UpdateCounters() argument
342 packet_iat_count_ms_ += elapsed_time_ms; in UpdateCounters()
343 peak_detector_.IncrementCounter(elapsed_time_ms); in UpdateCounters()
344 max_timer_ms_ += elapsed_time_ms; in UpdateCounters()
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| D | decision_logic.cc | 155 const int elapsed_time_ms = in FilterBufferLevel() local
157 delay_manager_->UpdateCounters(elapsed_time_ms); in FilterBufferLevel()
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| D | delay_manager.h | 80 virtual void UpdateCounters(int elapsed_time_ms);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| D | unittest.cc | 142 int elapsed_time_ms = 0; in TestGetSetBandwidthInfo() local
143 for (int i = 0; elapsed_time_ms < 10000; ++i) { in TestGetSetBandwidthInfo()
163 const int send_time = elapsed_time_ms * (sample_rate_hz / 1000); in TestGetSetBandwidthInfo()
179 elapsed_time_ms += duration1_ms; in TestGetSetBandwidthInfo()
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| /external/webrtc/webrtc/modules/pacing/ |
| D | paced_sender.cc | 352 int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000; in TimeUntilNextProcess() local
353 return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0); in TimeUntilNextProcess()
359 int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000; in Process() local
363 if (!paused_ && elapsed_time_ms > 0) { in Process()
380 int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms); in Process()
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| D | bitrate_prober.cc | 88 int64_t elapsed_time_ms = now_ms - time_last_send_ms_; in TimeUntilNextProbe() local
96 time_until_probe_ms = next_delta_ms - elapsed_time_ms; in TimeUntilNextProbe()
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| /external/webrtc/webrtc/test/ |
| D | fake_audio_device.cc | 118 int64_t elapsed_time_ms = -1; in CaptureAudio() local
127 &elapsed_time_ms, in CaptureAudio()
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| /external/webrtc/webrtc/modules/audio_device/include/ |
| D | audio_device_defines.h | 66 int64_t* elapsed_time_ms,
131 int64_t* elapsed_time_ms, in PullRenderData() argument
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| /external/webrtc/talk/app/webrtc/test/ |
| D | fakeaudiocapturemodule_unittest.cc | 89 int64_t* elapsed_time_ms, in NeedMorePlayData() argument
98 *elapsed_time_ms = 0; in NeedMorePlayData()
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| D | fakeaudiocapturemodule.cc | 702 int64_t elapsed_time_ms = 0; in ReceiveFrameP() local
707 &elapsed_time_ms, &ntp_time_ms) != 0) { in ReceiveFrameP()
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| /external/webrtc/talk/media/base/ |
| D | testutils.cc | 146 uint32_t elapsed_time_ms = 0; in WriteTestPackets() local
155 RtpDumpPacket dump_packet(buf.Data(), buf.Length(), elapsed_time_ms, rtcp); in WriteTestPackets()
156 elapsed_time_ms += kElapsedTimeInterval; in WriteTestPackets()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| D | mock_delay_manager.h | 42 void(int elapsed_time_ms));
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| /external/webrtc/webrtc/modules/audio_device/ |
| D | audio_device_buffer.cc | 537 int64_t elapsed_time_ms = -1; in RequestPlayoutData() local
545 &elapsed_time_ms, in RequestPlayoutData()
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| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| D | conference_transport.cc | 208 int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_); in DispatchPackets() local
209 int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms; in DispatchPackets()
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| /external/webrtc/webrtc/modules/audio_device/test/ |
| D | func_test_manager.h | 105 int64_t* elapsed_time_ms,
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| D | func_test_manager.cc | 346 int64_t* elapsed_time_ms, in NeedMorePlayData() argument
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| D | audio_device_test_api.cc | 117 int64_t* elapsed_time_ms, in NeedMorePlayData() argument
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| /external/webrtc/webrtc/modules/audio_device/ios/ |
| D | audio_device_unittest_ios.cc | 390 int64_t* elapsed_time_ms,
444 int64_t* elapsed_time_ms, in RealNeedMorePlayData() argument
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_device_unittest.cc | 400 int64_t* elapsed_time_ms,
452 int64_t* elapsed_time_ms, in RealNeedMorePlayData() argument
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| /external/webrtc/talk/media/webrtc/ |
| D | webrtcvideoengine2.cc | 2445 int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame / in RenderFrame() local
2448 estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms; in RenderFrame()
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