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Searched
refs:capture_start_rtp_time_stamp_
(Results
1 - 3
of
3
) sorted by null
/external/chromium_org/third_party/webrtc/voice_engine/
channel.h
566
int64_t
capture_start_rtp_time_stamp_
;
member in class:webrtc::voe::Channel
channel.cc
686
if (
capture_start_rtp_time_stamp_
< 0 && audioFrame.timestamp_ != 0) {
688
capture_start_rtp_time_stamp_
= audioFrame.timestamp_;
691
if (
capture_start_rtp_time_stamp_
>= 0) {
698
(unwrap_timestamp -
capture_start_rtp_time_stamp_
) /
891
capture_start_rtp_time_stamp_
(-1),
[
all
...]
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
webrtcvideoengine.cc
188
capture_start_rtp_time_stamp_
(-1),
238
if (
capture_start_rtp_time_stamp_
< 0) {
239
capture_start_rtp_time_stamp_
= rtp_time_stamp;
246
capture_start_rtp_time_stamp_
) / kVideoCodecClockratekHz;
331
int64
capture_start_rtp_time_stamp_
;
member in class:cricket::WebRtcRenderAdapter
[
all
...]
Completed in 38 milliseconds