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![]() This CL refactors RtpSenderTest.SendPacketHandlesRetransmissionHistory, moves some testing to rtp_ender_egress_unittest and adds test coverage for a few cases. Bug: webrtc:11340 Change-Id: Ic225d2af43c3926f69fe3ea45f41b18c29b8b4fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219796 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34111} |
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.. | ||
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 |