webrtc_m130

History

Steve Anton c0cde560f2 Fix size_t to int in RtpDemuxer.

Addresses build failures in Chrome.

Bug: 756840
Change-Id: Icfb3b230f05a99d7f29d913180ed6d2601c26f4d
Reviewed-on: https://chromium-review.googlesource.com/623949
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19435}

2017-08-21 20:22:54 +00:00

api

iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-08-21 13:11:18 +00:00

audio

Replace absolute path with relative path for GN files.

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

call

Fix size_t to int in RtpDemuxer.

2017-08-21 20:22:54 +00:00

common_audio

Replace absolute path with relative path for GN files.

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

common_video

Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )

2017-08-18 09:51:12 +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

Fix compile error for the win_msvc_rel bot.

2017-08-21 15:53:47 +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

Fix a delete type mismatch (deleting a sockaddr_in6* as a sockaddr*)

2017-08-18 07:15:19 +00:00

rtc_tools

Delete old CongestionController class

2017-08-21 10:43:33 +00:00

sdk

Android: Fix synchronization problems in VideoFrame release.

2017-08-21 15:02:58 +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

Delete old CongestionController class

2017-08-21 10:43:33 +00:00

video

Revert of Add Jpeg frame writer for test support. (patchset #12 id:220001 of https://codereview.webrtc.org/2990563002/ )

2017-08-20 19:50:56 +00:00

voice_engine

Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ )

2017-08-18 14:22:36 +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

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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

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.