platform-external-webrtc

Files

sakal fd6c94d311 Allow more config changes for CallActivity.

CallActivity doesn't survive recreation so we have to handle these config
changes. Fortunately, they don't seem to require any special handling.

BUG=webrtc:7018

Review-Url: https://codereview.webrtc.org/2645853002
Cr-Commit-Position: refs/heads/master@{#16180}

2017-01-20 10:36:49 +00:00

androidjunit

Format all Java in WebRTC.

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

api

Remove the dependency of TransportChannel and TransportChannelImpl.

2017-01-20 00:54:25 +00:00

audio

Pass SdpAudioFormat through Channel, without converting to CodecInst

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

base

Reduce the log verbosity in sslstreamadapter_unittest

2017-01-18 23:29:35 +00:00

build

GN: Refactor modules_unittests to eliminate package boundary violations.

2017-01-19 16:27:11 +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

Allow more config changes for CallActivity.

2017-01-20 10:36:49 +00:00

logging

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

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

media

Clear out cached codecs when calculating new codec lists.

2017-01-20 09:07:26 +00:00

modules

Allow GCC 4.9 to compile Chromium

2017-01-20 04:20:45 +00:00

p2p

Remove the dependency of TransportChannel and TransportChannelImpl.

2017-01-20 00:54:25 +00:00

Remove the dependency of TransportChannel and TransportChannelImpl.

2017-01-20 00:54:25 +00:00

sdk

Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release.

2017-01-19 17:02:29 +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

…

BUILD.gn

Refactor OveruseFrameDetector to use timing in us units

2017-01-18 10:16:20 +00:00

call.h

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

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

codereview.settings

…

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.