webrtc/modules/rtp_rtcp
Tomas Lundqvist b50599b7b5 Expose jitter in time in addition to in samples.
RFC 3550 specifies samples to be the unit while https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict* specifies time. This avoids the need to convert to time in code that reads the jitter value from RtpReceiveStats.

Bug: webrtc:13757
Change-Id: I972996971c58b686babd621ff4e0f5790fdf2cb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279281
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#38419}
2022-10-17 16:27:57 +00:00
..
include Expose jitter in time in addition to in samples. 2022-10-17 16:27:57 +00:00
mocks Add ability to abort retransmissions. 2022-09-08 16:34:40 +00:00
source Expose jitter in time in addition to in samples. 2022-10-17 16:27:57 +00:00
test/testFec Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED 2021-11-15 21:44:59 +00:00
BUILD.gn Add missing dependencies. 2022-10-10 13:50:03 +00:00
DEPS WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
OWNERS Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00