webrtc/modules/rtp_rtcp
Niels Möller c936cb6a86 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27190}
2019-03-19 16:59:27 +00:00
..
include Removes rtp level keep alive support. 2019-03-11 14:47:15 +00:00
mocks Move ownership of RTPSenderAudio to ChannelSend. 2019-03-06 17:15:00 +00:00
source Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h 2019-03-19 16:59:27 +00:00
test/testFec (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
BUILD.gn Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h 2019-03-19 16:59:27 +00:00
DEPS Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule 2019-02-21 14:25:34 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00