webrtc_m130

History

sakal 4dee344493 Implement QP parsing in VideoDecoderWrapper.

BUG=webrtc:7760
NOTRY=True

Review-Url: https://codereview.webrtc.org/3000993002
Cr-Commit-Position: refs/heads/master@{#19378}

2017-08-17 09:18:04 +00:00

api

L16 implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-08-16 13:12:57 +00:00

audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

call

Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )

2017-08-16 15:16:25 +00:00

common_audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

common_video

Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )

2017-08-15 19:31:23 +00:00

examples

Remove dep on system_wrappers for apprtc and use only framework.

2017-08-16 14:28:03 +00:00

logging

Replace CHECK(x && y) with two separate CHECK() calls

2017-08-10 00:22:01 +00:00

media

Delete unneeded Start and Stop methods on FlexfecReceiveStream.

2017-08-16 09:41:27 +00:00

modules

Reland of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2999893002/ )

2017-08-17 09:13:54 +00:00

ortc

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

p2p

Replace CHECK(x && y) with two separate CHECK() calls

2017-08-10 00:22:01 +00:00

Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )

2017-08-15 06:48:03 +00:00

rtc_base

Check for prior thread annotation macros before defining.

2017-08-16 17:31:58 +00:00

rtc_tools

Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.

2017-08-16 09:48:33 +00:00

sdk

Implement QP parsing in VideoDecoderWrapper.

2017-08-17 09:18:04 +00:00

stats

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

system_wrappers

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

test

Delete unneeded Start and Stop methods on FlexfecReceiveStream.

2017-08-16 09:41:27 +00:00

video

Reland of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2999893002/ )

2017-08-17 09:13:54 +00:00

voice_engine

Add thread annotations to AudioLevel

2017-08-10 12:01:21 +00:00

.gitignore

…

BUILD.gn

When using clang, switch on -Wc++11-narrowing

2017-08-16 11:52:35 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

common_types.h

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.cc

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

config.h

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Protected streams report RTP messages directly to the FlexFec streams

2017-08-02 14:39:07 +00:00

video_send_stream.h

Move RTP keep-alive config from VideoSendStream::Config to Call::Config

2017-07-11 10:44:17 +00:00

webrtc.gni

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +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.