webrtc_m130

History

deadbeef 8e814d7906 Provide better message for when RTCP mux "require" policy is triggered.

Previously: Failed to setup RTCP mux filter.
Now: rtcpMuxPolicy is 'require', but media description does not
     contain 'a=rtcp-mux'.

BUG=webrtc:6966

Review-Url: https://codereview.webrtc.org/2622553003
Cr-Commit-Position: refs/heads/master@{#16062}

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

androidjunit

Format all Java in WebRTC.

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

api

Replace left-over ASSERTs in httpcommon.h and webrtcsession.cc.

2017-01-13 15:08:34 +00:00

audio

GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.

2017-01-05 14:03:24 +00:00

base

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +00:00

build

Remove lint ignores and add WrongCall ignore directly to the test.

2017-01-12 14:19:40 +00:00

call

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +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

Update ice server provider response format in iOS AppRTCMobile

2017-01-13 13:59:46 +00:00

logging

Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )

2017-01-13 14:52:12 +00:00

media

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +00:00

modules

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +00:00

p2p

Remove BaseChannel's dependency on TransportController.

2017-01-13 03:37:48 +00:00

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

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

sdk

Remove use of PCH files in webrtc/sdk/objc

2017-01-13 15:10:16 +00:00

stats

Update smoothed bitrate.

2017-01-12 18:17:38 +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

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +00:00

tools

Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )

2017-01-13 14:52:12 +00:00

video

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +00:00

voice_engine

Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )

2017-01-13 14:52:12 +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

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

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

config.h

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +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

Implement periodic bandwidth probing in application-limited region.

2016-11-28 21:11:24 +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.