mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 14:50:39 +01:00
bd09a46aa1
Moves OnSendSideDelayUpdated and OnSendPacketUpdated out from rtp_sender_unittest and into rtp_sender_egress_unittest and rtp_rtcp_impl2_unittest. The former test now only tests the logic for updating send-side-delay stats. The latter is now on a proper RtpRtcp-level and also verifies that frame timestamps makes it to the egress (as assumed by the first test). Bug: webrtc:11340 Change-Id: I784042ad91eb66a4d1eebdbbc625f9522528bfb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218502 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33996} |
||
|---|---|---|
| .. | ||
| async_audio_processing | ||
| audio_coding | ||
| audio_device | ||
| audio_mixer | ||
| audio_processing | ||
| congestion_controller | ||
| desktop_capture | ||
| include | ||
| pacing | ||
| remote_bitrate_estimator | ||
| rtp_rtcp | ||
| third_party | ||
| utility | ||
| video_capture | ||
| video_coding | ||
| video_processing | ||
| BUILD.gn | ||
| module_common_types_unittest.cc | ||