webrtc/modules/rtp_rtcp
Ilya Nikolaevskiy 00a1bcb441 Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.

Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
2019-12-17 12:03:24 +00:00
..
include Makes padding prefer video SSRCs instead of audio. 2019-12-11 16:32:14 +00:00
mocks Makes padding prefer video SSRCs instead of audio. 2019-12-11 16:32:14 +00:00
source Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps 2019-12-17 12:03:24 +00:00
test/testFec Reland "Refactor FEC code to use COW buffers" 2019-09-09 16:20:33 +00:00
BUILD.gn Move vp9 rtp depacketization to VideoRtpDepacketizerVp9 2019-12-16 17:11:13 +00:00
DEPS Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule 2019-02-21 14:25:34 +00:00
OWNERS Add sprang@ as owner in modules/rtp_rtcp 2019-07-24 17:06:36 +00:00