Sebastian Jansson f222d2823d Adds srte@webrtc.org as modules/pacing/ OWNER. ...

Bug: webrtc:8415
Change-Id: I5ef199825dbb061ae91baa7f8781238433d72d67
Reviewed-on: https://webrtc-review.googlesource.com/86129
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23790}

2018-06-29 15:25:24 +00:00

..

audio_coding

Implement PacketDuration() for FakeDecoderFromFile.

2018-06-29 08:32:36 +00:00

audio_device

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

audio_mixer

Calculate all audio samples in AudioMixerCalculateEnergy.

2018-06-29 14:47:13 +00:00

audio_processing

AEC3: Misadjustment estimator of the linear filter.

2018-06-29 15:05:14 +00:00

bitrate_controller

Removing usage of //build/config/compiler:no_size_t_to_int_warning.

2018-06-20 13:44:26 +00:00

congestion_controller

Clean up RateControlInput struct, used by bandwidth estimation.

2018-06-29 10:47:37 +00:00

desktop_capture

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

include

Add RTPVideoHeader const accessor.

2018-06-21 09:49:40 +00:00

pacing

Adds srte@webrtc.org as modules/pacing/ OWNER.

2018-06-29 15:25:24 +00:00

remote_bitrate_estimator

Clean up RateControlInput struct, used by bandwidth estimation.

2018-06-29 10:47:37 +00:00

rtp_rtcp

Add Generic frame descritpor header extension

2018-06-29 15:02:44 +00:00

utility

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

video_capture

Delete unused file.

2018-06-28 12:53:17 +00:00

video_coding

Revert "Unit test for case where the number of active and configured spatial"

2018-06-29 12:01:38 +00:00

video_processing

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

BUILD.gn

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

module_common_types_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

OWNERS

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