webrtc_m130

History

zhihuang 8586c8ee88 Add the url attribute to the IceCandidate (Java Wrapper)

The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.

BUG=webrtc:7128

Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}

2017-02-13 22:04:50 +00:00

api

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

audio

Clean out platform specific things from voice_engine_defines.h.

2017-02-13 12:42:52 +00:00

base

Delete unused classes AutoDetectProxy and SslSocketFactory.

2017-02-13 12:33:28 +00:00

call

Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

2017-02-13 13:59:46 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-02-09 11:05:59 +00:00

common_video

Remove henrik.lundin from webrtc/common_video/OWNERS

2017-02-10 16:44:31 +00:00

examples

Add the url attribute to the IceCandidate (Java Wrapper)

2017-02-13 22:04:50 +00:00

logging

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

media

WebRtcVoiceEngineTest: Changed a static_cast to a checked_cast.

2017-02-13 15:04:05 +00:00

modules

Reduce the BWE with 50% when feedback is received too late.

2017-02-13 17:08:22 +00:00

p2p

Add the URL attribute to cricket::Candiate.

2017-02-13 20:47:27 +00:00

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

sdk

Add the url attribute to the IceCandidate (Java Wrapper)

2017-02-13 22:04:50 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

2017-01-27 14:35:16 +00:00

system_wrappers

replace NtpTime->Clock with Clock->NtpTime dependency

2017-02-09 19:15:25 +00:00

test

Do not encode frames in MultithreadedFakeH264Encoder after Release().

2017-02-11 08:25:18 +00:00

tools

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

video

Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

2017-02-13 13:59:46 +00:00

voice_engine

Clean out platform specific things from voice_engine_defines.h.

2017-02-13 12:42:52 +00:00

.gitignore

Flesh out webrtc/.gitignore

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

BUILD.gn

Added gn target for rtc_event_log2rtp_dump.

2017-02-08 08:28:09 +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

no_op_function.cc

Moving no_op_function.cc out of webrtc/build

2017-01-24 13:25:12 +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

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +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

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

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

webrtc.gni

Check for use_x11 before runnig desktop_capture_modules_tests on linux.

2017-02-10 09:38:23 +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.