webrtc_m130

History

sakal 07a3bd7c4b Bindings for injectable Java video encoders.

BUG=webrtc:7760

Review-Url: https://codereview.webrtc.org/3003873002
Cr-Commit-Position: refs/heads/master@{#19651}

2017-09-04 10:57:21 +00:00

api

Change reporting of timing frames conditions in GetStats on receive side

2017-09-04 10:35:40 +00:00

audio

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

call

Change reporting of timing frames conditions in GetStats on receive side

2017-09-04 10:35:40 +00:00

common_audio

[MIPS]: Fix compiler error for mips32r2 dspr2

2017-08-31 12:48:24 +00:00

common_video

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

examples

Make PeerConnectionClient non-singleton.

2017-08-31 15:03:46 +00:00

logging

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

media

Change reporting of timing frames conditions in GetStats on receive side

2017-09-04 10:35:40 +00:00

modules

Remove VideoProcessorIntegrationTest::SetTestConfig.

2017-09-04 10:03:40 +00:00

ortc

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

p2p

Completed the functionalities of SrtpTransport.

2017-08-31 23:00:07 +00:00

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

rtc_base

Allow PostTask() to take unique_ptr to classes derived of QueuedTask

2017-08-31 11:36:05 +00:00

rtc_tools

Add a readme for the network tester.

2017-08-31 13:24:56 +00:00

sdk

Bindings for injectable Java video encoders.

2017-09-04 10:57:21 +00:00

stats

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

system_wrappers

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

test

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

video

Change reporting of timing frames conditions in GetStats on receive side

2017-09-04 10:35:40 +00:00

voice_engine

Reland "Supporting 48kHz PCM file."

2017-09-01 19:49:24 +00:00

.gitignore

…

BUILD.gn

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

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

common_types.cc

Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )

2017-08-28 14:16:43 +00:00

common_types.h

Reland "Supporting 48kHz PCM file."

2017-09-01 19:49:24 +00:00

config.h

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

DEPS

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +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

webrtc.gni

Removing EXPAT_RELATIVE_PATH and rtc_relative_path.

2017-08-25 15:14:04 +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.