platform-external-webrtc

Files

deadbeef 19b3a554e8 Fixing incorrect use of erase/remove idiom.

In this case it wasn't an issue, because only one result would be found
by remove_if, but might as well fix it just in case.

BUG=None
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2945723002
Cr-Commit-Position: refs/heads/master@{#18641}

2017-06-17 03:19:08 +00:00

api

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

audio

Also scan stderr for audio files to test, due to change in Android test_runner

2017-06-14 16:35:11 +00:00

base

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

call

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

examples

Move setting switches in AppRTCMobile to Settings screen

2017-06-16 11:16:19 +00:00

logging

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

media

Add cropping to VIEEncoder to match simulcast streams resolution

2017-06-16 13:53:48 +00:00

modules

Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

2017-06-16 14:47:00 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

Fixing incorrect use of erase/remove idiom.

2017-06-17 03:19:08 +00:00

sdk

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +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

Fix Chromium style checker warnings for MockAudioDecoder

2017-06-14 10:31:17 +00:00

tools

Add NetEq delay plotting to event_log_visualizer

2017-06-14 13:09:58 +00:00

video

Add cropping to VIEEncoder to match simulcast streams resolution

2017-06-16 13:53:48 +00:00

voice_engine

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

.gitignore

…

BUILD.gn

Templated AudioEncoderFactory

2017-06-16 17:42:05 +00:00

codereview.settings

…

common_types.cc

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

common_types.h

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Delete old include file webrtc/video_frame.h.

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

LICENSE

…

LICENSE_THIRD_PARTY

…

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

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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.