platform-external-webrtc

Files

tommi c45d6d9c85 Remove dependency on rtc::Thread and rtc_base from audio_mixer_unittests.

Instead, use a TaskQueue in the only test that required it.

BUG=none

Review-Url: https://codereview.webrtc.org/2975883002
Cr-Commit-Position: refs/heads/master@{#18969}

2017-07-11 13:17:10 +00:00

api

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

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

base

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

call

Call should allow pass through of keep-alive packets.

2017-07-11 10:56:21 +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

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

examples

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

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

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

modules

Remove dependency on rtc::Thread and rtc_base from audio_mixer_unittests.

2017-07-11 13:17:10 +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

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

Don't call CreateDtlsTransport_n from non-network thread in WebRtcSession

2017-07-11 03:06:59 +00:00

rtc_base

Revert of Make "set_ignore_non_default_routes" actually use its argument. (patchset #1 id:1 of https://codereview.webrtc.org/2974873002/ )

2017-07-11 10:45:59 +00:00

rtc_tools

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

sdk

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

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

Fix video_replay tool to respect RTX stream and fix default parameters.

2017-07-11 09:38:36 +00:00

video

Move RTP keep-alive config from VideoSendStream::Config to Call::Config

2017-07-11 10:44:17 +00:00

voice_engine

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

.gitignore

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BUILD.gn

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

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

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

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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

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.