webrtc_m130

History

nisse 2013e29df1 Disable automatic scaling in tests.

BUG=webrtc:6990

Review-Url: https://codereview.webrtc.org/2636903004
Cr-Commit-Position: refs/heads/master@{#16119}

2017-01-17 13:45:40 +00:00

androidjunit

Format all Java in WebRTC.

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

api

RTCMediaStreamTrackStats.kind added and collected.

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

audio

Fix for bwe with overhead on audio only calls.

2017-01-17 07:55:07 +00:00

base

Delete unused file faketaskrunner.h.

2017-01-16 08:30:33 +00:00

build

Replace default locale with US locale on Android.

2017-01-16 12:57:32 +00:00

call

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +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

Move file capturer/renderer tests to the correct location.

2017-01-17 11:32:06 +00:00

logging

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

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

media

Disable automatic scaling in tests.

2017-01-17 13:45:40 +00:00

modules

Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

2017-01-17 12:03:53 +00:00

p2p

RTCTransportStats.dtlsState replaces .activeConnection

2017-01-16 15:38:02 +00:00

Provide better message for when RTCP mux "require" policy is triggered.

2017-01-13 19:34:39 +00:00

sdk

Move file capturer/renderer tests to the correct location.

2017-01-17 11:32:06 +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

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

tools

Comparison of videos with reference frame not starting from zero

2017-01-17 11:24:57 +00:00

video

Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

2017-01-17 12:03:53 +00:00

voice_engine

UdpTransport:IsIpAddressValid: Added extra :: check for ipv6

2017-01-17 10:31:37 +00:00

.gitignore

…

BUILD.gn

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +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

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PRESUBMIT.py

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README.chromium

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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.