| /external/webrtc/webrtc/system_wrappers/include/ |
| rtp_to_ntp.h | 25 uint32_t rtp_timestamp; member in struct:webrtc::RtcpMeasurement
35 uint32_t rtp_timestamp,
41 bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
46 int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator_unittest.cc | 58 uint32_t rtp_timestamp, bool expected_result) {
61 rtp_timestamp));
86 uint32_t rtp_timestamp = GetRemoteTimestamp(); local
91 EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
98 EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
|
| rtcp_receiver_help.h | 86 uint32_t rtp_timestamp; member in class:webrtc::RTCPHelp::RTCPPacketInformation
|
| rtcp_sender.cc | 287 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
290 last_rtp_timestamp_ = rtp_timestamp;
474 uint32_t rtp_timestamp = local
483 report->WithRtpTimestamp(rtp_timestamp);
[all...] |
| /external/webrtc/webrtc/video/ |
| vie_sync_module.cc | 33 uint32_t rtp_timestamp = 0; local
38 &rtp_timestamp)) {
44 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
|
| vie_receiver.cc | 435 uint32_t rtp_timestamp = 0; local
437 &rtp_timestamp)) {
441 ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_single_stream.cc | 75 uint32_t rtp_timestamp = header.timestamp + local
99 if (estimator->inter_arrival.ComputeDeltas(rtp_timestamp, arrival_time_ms,
|
| overuse_detector_unittest.cc | 93 void UpdateDetector(uint32_t rtp_timestamp, int64_t receive_time_ms,
98 if (inter_arrival_->ComputeDeltas(rtp_timestamp,
139 uint32_t rtp_timestamp = 10 * 90; local
143 UpdateDetector(rtp_timestamp, now_ms_, packet_size);
145 rtp_timestamp += frame_duration_ms * 90;
153 uint32_t rtp_timestamp = 10 * 90; local
157 UpdateDetector(rtp_timestamp, now_ms_, packet_size);
158 rtp_timestamp += frame_duration_ms * 90;
171 uint32_t rtp_timestamp = 10 * 90; local
175 UpdateDetector(rtp_timestamp, now_ms_, packet_size)
219 uint32_t rtp_timestamp = frame_duration_ms * 90; local
251 uint32_t rtp_timestamp = frame_duration_ms * 90; local
282 uint32_t rtp_timestamp = frame_duration_ms * 90; local
[all...] |
| remote_bitrate_estimator_unittest_helper.h | 52 uint32_t rtp_timestamp; member in struct:webrtc::testing::RtpStream::RtpPacket
171 uint32_t rtp_timestamp,
|
| /external/webrtc/talk/media/base/ |
| rtpdump.cc | 261 uint32_t rtp_timestamp = 0; local
262 packet.GetRtpTimestamp(&rtp_timestamp);
268 first_rtp_timestamp_ = rtp_timestamp;
271 } else if (rtp_timestamp != prev_rtp_timestamp_) {
277 prev_rtp_timestamp_ = rtp_timestamp;
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| audio_coding_module_impl.cc | 138 uint32_t rtp_timestamp = local
147 last_rtp_timestamp_ = rtp_timestamp;
152 rtp_timestamp, rtc::ArrayView<const int16_t>(
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test_framework_unittest.cc | 793 uint32_t rtp_timestamp = 0; local
810 if (rtp_timestamp > media_packet->header().timestamp) {
813 rtp_timestamp = media_packet->header().timestamp;
[all...] |
| /external/webrtc/webrtc/voice_engine/ |
| channel.cc | 1822 uint32_t rtp_timestamp = 0; local
[all...] |
