webrtc_m130

History

jbudorick 58f1725ff1 Add gn dependency between ana_debug_dump_proto and ana_config_proto.

BUG=chromium:746106

Review-Url: https://codereview.webrtc.org/2985853002
Cr-Commit-Position: refs/heads/master@{#19158}

2017-07-26 21:49:20 +00:00

api

Remove remains of webrtc/base

2017-07-19 17:40:47 +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

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +00:00

common_audio

Remove remains of webrtc/base

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

common_video

Remove remains of webrtc/base

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

examples

Update native plugin dll for turn servers and video.

2017-07-25 23:04:31 +00:00

logging

Remove remains of webrtc/base

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

media

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +00:00

modules

Add gn dependency between ana_debug_dump_proto and ana_config_proto.

2017-07-26 21:49:20 +00:00

ortc

Remove remains of webrtc/base

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

p2p

Revert of Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. (patchset #2 id:20001 of https://codereview.webrtc.org/2983213002/ )

2017-07-21 19:59:46 +00:00

When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded

2017-07-25 00:00:25 +00:00

rtc_base

Implemented a new sequence number unwrapper in sequence_number_util.h.

2017-07-26 14:48:15 +00:00

rtc_tools

Print general usage information for event_log_analyzer

2017-07-20 15:05:09 +00:00

sdk

Fix issues with NetworkMonitor singleton when used by multiple clients.

2017-07-26 18:56:49 +00:00

stats

Remove remains of webrtc/base

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

system_wrappers

Remove remains of webrtc/base

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

test

Rename isolated_output to test_output and add a method to get the test_output directory.

2017-07-25 14:31:18 +00:00

video

Report minimum PSNR in VideoQualityTest and save corresponding frame to file

2017-07-25 12:45:03 +00:00

voice_engine

Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.

2017-07-20 16:57:42 +00:00

.gitignore

…

BUILD.gn

Remove remains of webrtc/base

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

codereview.settings

…

common_types.cc

Rename webrtc/base -> webrtc/rtc_base

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

common_types.h

Add parsing/serializing for MID RTP header extension.

2017-07-21 17:33:25 +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

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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.