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  /external/webrtc/webrtc/modules/video_coding/
rtt_filter.cc 56 void VCMRttFilter::Update(int64_t rttMs) {
58 if (rttMs == 0) {
65 if (rttMs > 3000) {
66 rttMs = 3000;
82 _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
84 (1 - filtFactor) * (rttMs - _avgRtt) * (rttMs - _avgRtt);
85 _maxRtt = VCM_MAX(rttMs, _maxRtt);
86 if (!JumpDetection(rttMs) || !DriftDetection(rttMs)) {
    [all...]
rtt_filter.h 27 void Update(int64_t rttMs);
29 int64_t RttMs() const;
40 bool JumpDetection(int64_t rttMs);
45 bool DriftDetection(int64_t rttMs);
jitter_estimator.h 61 // - rttMs : RTT in ms
62 void UpdateRtt(int64_t rttMs);
jitter_estimator.cc 375 void VCMJitterEstimator::UpdateRtt(int64_t rttMs) {
376 _rttFilter.Update(rttMs);
392 jitterMS += _rttFilter.RttMs() * rttMultiplier;
  /external/webrtc/webrtc/voice_engine/test/auto_test/
voe_conference_test.cc 95 if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
96 IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
105 stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
  /external/webrtc/webrtc/modules/video_coding/test/
rtp_player.h 94 int64_t rttMs,
  /external/webrtc/webrtc/voice_engine/include/
voe_rtp_rtcp.h 68 int64_t rttMs;
  /external/webrtc/webrtc/audio/
audio_send_stream.cc 138 if (call_stats.rttMs > 0) {
139 stats.rtt_ms = call_stats.rttMs;
audio_send_stream_unittest.cc 218 EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
  /external/webrtc/webrtc/voice_engine/
channel.cc     [all...]

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