Danil Chapovalov f3ba6484e3 Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format ...

making it symmetric to AbsoluteSendTime::Parse function.

Bug: None
Change-Id: I9c71d840768064022ebebbbeb2962aeeecc68392
Reviewed-on: https://chromium-review.googlesource.com/531044
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18555}

2017-06-13 09:08:14 +00:00

..

audio_coding

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

audio_conference_mixer

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

audio_device

Fix play block size mismatch in Win audio device.

2017-06-12 23:54:07 +00:00

audio_mixer

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

audio_processing

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

bitrate_controller

Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl

2017-06-12 15:21:59 +00:00

congestion_controller

Reland Refactored incoming bitrate estimator.

2017-06-12 06:57:17 +00:00

desktop_capture

desktopCapture: scale the cursor image according to screen scale factor on OSX

2017-06-01 21:27:41 +00:00

include

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

media_file

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +00:00

pacing

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

remote_bitrate_estimator

Add SafeClamp(), which accepts args of different types

2017-06-12 18:40:47 +00:00

rtp_rtcp

Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format

2017-06-13 09:08:14 +00:00

utility

Remove deprecation warning from JVM::Initialize with the context parameter.

2017-06-01 11:59:52 +00:00

video_capture

Remove outdated warning suppressions.

2017-06-07 11:20:02 +00:00

video_coding

Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.

2017-06-12 12:53:35 +00:00

video_processing

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +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