platform-external-webrtc

Files

sprang 892dab52b6 Fix incorrect InterframeDelayMaxInMs histogram metrics

Two bugs:

1) The max value should only be reported if the average is also
   reported. Otherwise the max might become lower than average.
   (On average).

2) When reporting that max value, actually use the max value.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/3002593002
Cr-Commit-Position: refs/heads/master@{#19352}

2017-08-15 12:00:33 +00:00

api

Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.

2017-08-10 09:43:14 +00:00

audio

Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"

2017-07-26 14:28:51 +00:00

base

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

call

Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )

2017-08-14 15:03:17 +00:00

common_audio

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

common_video

Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.

2017-08-10 09:43:14 +00:00

examples

AppRTCMobile: Use high resolution app icon on retina screens.

2017-08-15 11:07:12 +00:00

logging

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

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

media

Default enable content type rtp header extension

2017-08-15 08:32:51 +00:00

modules

Roll chromium_revision f156b499f7..f439921f66 (493756:494089)

2017-08-15 10:16:50 +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

Implementation of SSL caching; tests in separate CL.

2017-08-15 00:37:05 +00:00

rtc_tools

Improved UI for event_log_analyzer tool

2017-08-15 09:04:02 +00:00

sdk

Add VideoSink interface to SurfaceViewRenderer.

2017-08-15 08:56:02 +00:00

stats

Audit of kConstants missing the const qualifier

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

system_wrappers

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

test

Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.

2017-08-10 09:43:14 +00:00

video

Fix incorrect InterframeDelayMaxInMs histogram metrics

2017-08-15 12:00:33 +00:00

voice_engine

Add thread annotations to AudioLevel

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

.gitignore

…

BUILD.gn

Removing NO_SOUND_SYSTEM

2017-08-14 07:40:58 +00:00

codereview.settings

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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

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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

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

Removing unused declared arg

2017-08-03 17:45:31 +00:00

whitespace.txt

…

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.