| /external/webrtc/webrtc/video/ |
| call_stats.cc | 58 void UpdateAvgRttMs(std::list<CallStats::RttTime>* reports, int64_t* avg_rtt) {
62 *avg_rtt = 0;
65 if (*avg_rtt == 0) {
67 *avg_rtt = cur_rtt_ms;
70 *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl_unittest.cc | 307 int64_t avg_rtt; local
311 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
313 EXPECT_EQ(2 * kOneWayNetworkDelayMs, avg_rtt);
319 sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
|
| rtp_rtcp_impl.h | 157 int64_t* avg_rtt,
|
| rtp_sender.cc | 784 int64_t avg_rtt) {
787 nack_sequence_numbers.size(), "avg_rtt", avg_rtt); local
801 const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
815 if (target_bitrate != 0 && avg_rtt) {
818 (static_cast<size_t>(target_bitrate / 1000) * avg_rtt) >> 3;
[all...] |
| rtp_sender.h | 218 int64_t avg_rtt);
|
| rtp_rtcp_impl.cc | 532 int64_t* avg_rtt,
535 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
|
| /external/webrtc/webrtc/voice_engine/ |
| channel.cc | 4107 int64_t avg_rtt = 0; local
[all...] |
