webrtc_m130

History

eladalon bd09ebc71e Remove unused #include "libyuv/compare.h"

BUG=None

Review-Url: https://codereview.webrtc.org/2934423003
Cr-Commit-Position: refs/heads/master@{#18627}

2017-06-16 11:54:22 +00:00

api

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +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

Delete SignalSrtpError.

2017-06-16 07:57:21 +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

Remove unused #include "libyuv/compare.h"

2017-06-16 11:54:22 +00:00

modules

Validate references of frames inserted into FrameBuffer2.

2017-06-15 16:06:21 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Adding PortAllocator option to support cases where sockets can't be bound.

2017-06-13 22:49:45 +00:00

Delete SignalSrtpError.

2017-06-16 07:57:21 +00:00

sdk

Create AndroidVideoBuffer and allow renderers to consume it.

2017-06-16 08:31:37 +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

Use the same QP max for tests as in production

2017-06-15 13:34:42 +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

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Delete old include file webrtc/video_frame.h.

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

codereview.settings

Give a more specific URL for creating WebRTC checkout

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

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

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.