webrtc_m130

History

saza 10b0d03673 Removes internal buffer memory check in AEC module.

The check triggered in 30 / 1000 cases of running PeerConnectionIntegrationTest.CallTransferredForCaller locally, far more often than expected.

It will soon be replaced by more graceful handling.

BUG=webrtc:7845

Review-Url: https://codereview.webrtc.org/2975043002
Cr-Commit-Position: refs/heads/master@{#18983}

2017-07-12 07:29:36 +00:00

api

API for periodically regathering ICE candidates

2017-07-11 21:49:38 +00:00

audio

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

base

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

call

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

common_audio

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

common_video

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

examples

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

logging

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

media

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

modules

Removes internal buffer memory check in AEC module.

2017-07-12 07:29:36 +00:00

ortc

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

p2p

API for periodically regathering ICE candidates

2017-07-11 21:49:38 +00:00

API for periodically regathering ICE candidates

2017-07-11 21:49:38 +00:00

rtc_base

Reland of Make "set_ignore_non_default_routes" actually use its argument. (patchset #1 id:1 of https://codereview.webrtc.org/2974193002/ )

2017-07-12 03:07:37 +00:00

rtc_tools

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

sdk

Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ )

2017-07-11 22:46:31 +00:00

stats

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

system_wrappers

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

test

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

video

Let alr detector use a budged to detect underuse.

2017-07-11 13:56:04 +00:00

voice_engine

TransmitMixer: Check GetSendCodec return value.

2017-07-11 15:19:31 +00:00

.gitignore

…

BUILD.gn

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

codereview.settings

…

common_types.cc

Rename webrtc/base -> webrtc/rtc_base

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

common_types.h

Rename webrtc/base -> webrtc/rtc_base

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

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

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

config.cc

Rename webrtc/base -> webrtc/rtc_base

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

config.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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

Report interframe delay sum in old GetStats

2017-07-07 08:26:24 +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

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +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.