webrtc_m130

History

henrik.lundin d56ef80857 RtpToNtpEstimator:: Add a DCHECK to avoid div-by-0

The division that follows the added DCHECK should not be div-by-0,
since when params_.calculated is true, params_.frequency_khz would
have been updated (with something non-zero).

BUG=none

Review-Url: https://codereview.webrtc.org/3006003002
Cr-Commit-Position: refs/heads/master@{#19671}

2017-09-04 16:02:15 +00:00

api

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

audio

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

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

call

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

common_audio

[MIPS]: Fix compiler error for mips32r2 dspr2

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

common_video

Delete static method TaskQueue::IsCurrent.

2017-09-04 12:18:21 +00:00

examples

Make PeerConnectionClient non-singleton.

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

logging

Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread

2017-09-04 14:26:14 +00:00

media

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

modules

Make UBSan warnings fatal and fix the existing ones

2017-09-04 15:32:43 +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

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

rtc_base

Revert of Decoupling rtc_base from Obj-C code (patchset #8 id:140001 of https://codereview.webrtc.org/3001623003/ )

2017-09-04 14:19:15 +00:00

rtc_tools

Add a readme for the network tester.

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

sdk

Tightening visibility of video_toolbox_cc.

2017-09-04 12:51:34 +00:00

stats

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

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

system_wrappers

RtpToNtpEstimator:: Add a DCHECK to avoid div-by-0

2017-09-04 16:02:15 +00:00

test

Move array_view.h to webrtc/api/

2017-09-04 12:43:17 +00:00

video

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

voice_engine

Move array_view.h to webrtc/api/

2017-09-04 12:43:17 +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

…

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

Move array_view.h to webrtc/api/

2017-09-04 12:43:17 +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.