webrtc_m130

History

Henrik Kjellander c03627683f Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.

BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org

Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}

2017-06-29 06:04:25 +00:00

api

Expose ILBC codec in webrtc/api/audio_codecs/

2017-06-28 09:05:04 +00:00

audio

Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky

2017-06-26 12:04:12 +00:00

base

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

call

nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_

2017-06-28 15:18:51 +00:00

common_audio

Put attribute before function name instead of after, as required by GCC

2017-06-26 08:32:40 +00:00

common_video

Return WrappedI444Buffer in VP9Impl

2017-06-23 03:28:06 +00:00

examples

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

logging

Limit the number of simultaneous event logs.

2017-06-28 13:40:51 +00:00

media

Reland of Periodically update codec bit/frame rate settings.

2017-06-27 14:06:52 +00:00

modules

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

ortc

Fixing bad use of std::sort in test method.

2017-06-27 14:52:50 +00:00

p2p

Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.

2017-06-28 16:37:23 +00:00

Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.

2017-06-28 16:37:23 +00:00

rtc_base

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

sdk

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

tools

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

video

Fix deadlock in webrtc_perf_tests

2017-06-27 14:21:01 +00:00

voice_engine

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

2017-06-21 08:05:22 +00:00

.gitignore

…

BUILD.gn

Add webrtc/rtc_base skeleton.

2017-06-21 12:50:22 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

common_types.h

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

config.cc

Implement timing frames.

2017-06-19 14:18:55 +00:00

config.h

Implement timing frames.

2017-06-19 14:18:55 +00:00

DEPS

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

video_send_stream.h

Add field trial for balanced degradation preference.

2017-06-14 12:52:21 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.