webrtc/modules/rtp_rtcp
Tony Herre 5f3ac43551 Ensure cloning and then sending audio encoded frames propagates CSRCs
Bug: chromium:1508337
Change-Id: I9f28fc0958d28bc97f9378a46fbec3e45148736f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330260
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41337}
2023-12-07 15:09:01 +00:00
..
include Delete SendDelayObserver interface 2023-09-15 14:59:23 +00:00
mocks Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types 2023-07-21 10:36:49 +00:00
source Ensure cloning and then sending audio encoded frames propagates CSRCs 2023-12-07 15:09:01 +00:00
test/testFec Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED 2021-11-15 21:44:59 +00:00
BUILD.gn Add rtp packetizer for H265 2023-11-08 15:49:37 +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