webrtc_m130

History

Steve Anton 786de70a59 Add TLS TURN tests.

This change extends the TurnPort tests to cover connections to
TURN servers over TLS.
As part of this, the TestTurnServer is extended to support
connections from clients over TLS.

Note that this also fixes the remaining bugs in webrtc:7562

Bug: webrtc:7584
Change-Id: If89ceae49d33417625464b5892d20eee4de7c3b5
Reviewed-on: https://chromium-review.googlesource.com/611520
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19397}

2017-08-17 23:03:04 +00:00

api

Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.

2017-08-17 15:58:54 +00:00

audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

call

Rename RsidResolutionObserver to SsrcBindingObserver.

2017-08-17 23:01:59 +00:00

common_audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

common_video

Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )

2017-08-15 19:31:23 +00:00

examples

Remove dep on system_wrappers for apprtc and use only framework.

2017-08-16 14:28:03 +00:00

logging

Add rtc_event_log_unittest_helper.h to relevant BUILD.gn

2017-08-17 15:32:39 +00:00

media

Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.

2017-08-17 15:58:54 +00:00

modules

Share ScreenDrawerLockPosix implementation

2017-08-17 20:47:38 +00:00

ortc

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

p2p

Add TLS TURN tests.

2017-08-17 23:03:04 +00:00

Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.

2017-08-17 21:10:50 +00:00

rtc_base

Add TLS TURN tests.

2017-08-17 23:03:04 +00:00

rtc_tools

Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.

2017-08-16 09:48:33 +00:00

sdk

Fix a reference leak when converting I420 VideoFrame to I420Frame.

2017-08-17 10:56:58 +00:00

stats

Audit of kConstants missing the const qualifier

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

system_wrappers

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

test

Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names

2017-08-17 12:31:02 +00:00

video

Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ )

2017-08-17 18:01:46 +00:00

voice_engine

Add thread annotations to AudioLevel

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

.gitignore

…

BUILD.gn

Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )

2017-08-17 14:59:54 +00:00

codereview.settings

…

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

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

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

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +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.