webrtc_m130

History

peah ea154106a8 Lowered the allowed jitter in the api calls to a reasonable level

This CL reduces the allowed jitter in the api calls to a reasonable
level in order to ensure a quicker revery from audio path glitches.

BUG=webrtc:8224, chromium:763775

Review-Url: https://codereview.webrtc.org/3009273002
Cr-Commit-Position: refs/heads/master@{#19772}

2017-09-11 13:44:37 +00:00

api

Fix code formating in api/video/video_content_type.*

2017-09-11 09:20:45 +00:00

audio

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

call

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

common_audio

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

common_video

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

examples

Make PeerConnectionClient non-singleton.

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

logging

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

media

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

modules

Lowered the allowed jitter in the api calls to a reasonable level

2017-09-11 13:44:37 +00:00

ortc

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

p2p

Add support for changing the TLS elliptic curve set.

2017-09-08 20:02:03 +00:00

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

rtc_base

Decoupling rtc_base_approved from Obj-C code

2017-09-11 10:43:34 +00:00

rtc_tools

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

sdk

android: add IceServer.urls field

2017-09-11 13:41:38 +00:00

stats

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

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

system_wrappers

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

test

Add |RTCUIApplicationStatusObserver sharedInstance| call in ios test AppDelegate.

2017-09-11 13:43:28 +00:00

video

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

voice_engine

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

.gitignore

…

BUILD.gn

Add VideoProcessorIntegrationTest for MediaCodec implementations.

2017-09-07 14:50:06 +00:00

codereview.settings

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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 optional.h to webrtc/api/

2017-09-05 15:43:13 +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

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LICENSE_THIRD_PARTY

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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

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PRESUBMIT.py

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README.chromium

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typedefs.h

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

webrtc.gni

Revert of Add //build/android/pylib/results/ to Android isolate packaging (patchset #2 id:40001 of https://codereview.webrtc.org/3011023002/ )

2017-09-06 19:11:27 +00:00

whitespace.txt

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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.