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-rw-r--r--third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc8
1 files changed, 4 insertions, 4 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 0e5e40f502..1dc56bb96f 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -18,6 +18,7 @@
#include "api/units/time_delta.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ -159,16 +160,15 @@ void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
int32_t time_diff_samples =
receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
- time_diff_samples = std::abs(time_diff_samples);
-
ReviseFrequencyAndJitter(packet.payload_type_frequency());
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
- if (time_diff_samples < 450000) {
+ if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
+ time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
// Note we calculate in Q4 to avoid using float.
- int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
+ int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
}