platform-external-webrtc

Files

terelius 2ee076dfa3 Improved UI for event_log_analyzer tool

- Don't plot every graph by default.
- Change --plot_all to --plot_profile=(all|none|default).
- Some other minor cleanups.

BUG=webrtc:8017

Review-Url: https://codereview.webrtc.org/2983983002
Cr-Commit-Position: refs/heads/master@{#19348}

2017-08-15 09:04:02 +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

Add --canary flag to gradle project test and update gradle.

2017-08-11 08:03:13 +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

Change OpenSLES blacklist warning to debug.

2017-08-14 15:39:44 +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

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

voice_engine

Add thread annotations to AudioLevel

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

.gitignore

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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

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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

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.