Sebastian Jansson 9e3f1e4ca2 Fixed a miscalculation of sent bitrate caused by mixup of time units ...

Bug: webrtc:7949
Change-Id: Ia57fdd3d1de0952b80e77c30b0a6cfe44515eff2
Reviewed-on: https://chromium-review.googlesource.com/561460
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18917}

2017-07-06 15:22:58 +00:00

..

audio_coding

Let NetEq reset the AudioFrame during muted state

2017-07-06 12:23:53 +00:00

audio_conference_mixer

Changing AudioConferenceMixer logging to base/logging.h

2017-07-03 12:35:46 +00:00

audio_device

Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat

2017-07-05 09:34:31 +00:00

audio_mixer

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

audio_processing

Changed default value for the duration of the echo in echocanceller 3

2017-07-04 11:33:11 +00:00

bitrate_controller

Fix issue with zero rtt reports when using FlexFEC and add perf test.

2017-07-05 10:03:02 +00:00

congestion_controller

Reduce send rate to 50% if overusing before we have an acknowledged bitrate.

2017-07-04 11:52:58 +00:00

desktop_capture

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

include

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

media_file

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

pacing

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

remote_bitrate_estimator

Added implementation of four functions in the BBR congestion controller:

2017-07-05 12:00:46 +00:00

rtp_rtcp

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

utility

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

video_capture

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

video_coding

Fixed a miscalculation of sent bitrate caused by mixup of time units

2017-07-06 15:22:58 +00:00

video_processing

Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

2017-07-01 14:42:25 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00