webrtc_m130

History

solenberg 08b19dfc67 Remove VoEVideoSync interface.

The removed tests are covered by cases in call_perf_tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}

2017-02-15 08:42:31 +00:00

api

Changing some PeerConnection-related comments.

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

audio

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

base

Delete httpclient.c and related files.

2017-02-14 08:23:10 +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

Revert of Add the url attribute to the IceCandidate (Java Wrapper) (patchset #4 id:120001 of https://codereview.webrtc.org/2690593002/ )

2017-02-14 22:13:56 +00:00

logging

Move AudioDecoder and related stuff to the api/ directory

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

media

Handle InitDecode and reset in fallback decoder.

2017-02-13 22:11:08 +00:00

modules

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

p2p

Fix the build break by keeping the old Port::AddAddress method since the

2017-02-14 01:33:27 +00:00

Changing some PeerConnection-related comments.

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

sdk

Revert of Add the url attribute to the IceCandidate (Java Wrapper) (patchset #4 id:120001 of https://codereview.webrtc.org/2690593002/ )

2017-02-14 22:13:56 +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

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

tools

Move AudioDecoder and related stuff to the api/ directory

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

video

Fix flaky ViEEncoder unit test.

2017-02-14 19:57:25 +00:00

voice_engine

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +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

Remove VoEExternalMedia interface.

2017-02-14 12:55:00 +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

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

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

Move iOS tests to XCTest from gtest.

2017-02-14 12:58:56 +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.