webrtc_m130

History

deadbeef eaa826c2ee Removing #defines previously used for building without BoringSSL/OpenSSL.

These defines don't work any more, so they only cause confusion:

FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL

BUG=webrtc:7025

Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}

2017-01-20 23:15:58 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

Removing #defines previously used for building without BoringSSL/OpenSSL.

2017-01-20 23:15:58 +00:00

audio

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

base

Removing #defines previously used for building without BoringSSL/OpenSSL.

2017-01-20 23:15:58 +00:00

build

Finalize the support for building WebRTC library for iOS with bitcode

2017-01-20 16:01:36 +00:00

call

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

common_audio

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

common_video

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

examples

Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )

2017-01-20 14:48:03 +00:00

logging

GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.

2017-01-17 10:37:34 +00:00

media

RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.

2017-01-20 11:56:50 +00:00

modules

Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )

2017-01-20 14:48:03 +00:00

p2p

Removing #defines previously used for building without BoringSSL/OpenSSL.

2017-01-20 23:15:58 +00:00

Removing #defines previously used for building without BoringSSL/OpenSSL.

2017-01-20 23:15:58 +00:00

sdk

Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS

2017-01-20 14:56:44 +00:00

stats

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +00:00

system_wrappers

Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )

2017-01-09 09:22:16 +00:00

test

GN: Refactor modules_unittests to eliminate package boundary violations.

2017-01-19 16:27:11 +00:00

tools

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

video

Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )

2017-01-19 08:06:17 +00:00

voice_engine

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

.gitignore

Flesh out webrtc/.gitignore

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

BUILD.gn

Removing #defines previously used for building without BoringSSL/OpenSSL.

2017-01-20 23:15:58 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

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

common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

config.cc

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

config.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

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

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +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.