..
audio_coding
Decouple //rtc_base:rtc_base_tests_utils from gunit.
2018-11-23 12:52:46 +00:00
audio_device
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
audio_mixer
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
audio_processing
Add AudioProcessingCaptureStats and a level estimator replacement
2018-11-26 15:52:14 +00:00
bitrate_controller
Adds stable bandwidth estimate to GoogCC.
2018-11-23 14:55:37 +00:00
congestion_controller
Adds stable bandwidth estimate to GoogCC.
2018-11-23 14:55:37 +00:00
desktop_capture
Disabling ScreenDrawerTest.TwoScreenDrawerLocks
2018-11-21 10:27:52 +00:00
include
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
pacing
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
remote_bitrate_estimator
Moves ProbeBitrateEstimator from DelayBasedBwe.
2018-11-23 13:43:51 +00:00
rtp_rtcp
In ReceiveStatistics use monotonic clock instead of ntp clock
2018-11-26 10:31:44 +00:00
third_party
Fix guards for headers in third party
2018-08-07 09:39:06 +00:00
utility
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
video_capture
Delete _strnicmp. Uses replaced with abseil functions.
2018-11-02 11:03:38 +00:00
video_coding
Reland Profile 2 to default profiles
2018-11-26 07:48:03 +00:00
video_processing
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
BUILD.gn
Export symbols needed by the Chromium component build (part 3).
2018-10-16 12:57:04 +00:00
module_common_types_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
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