webrtc_m130

History

kjellander 3c1e558449 Extract iOS SDK helpers to separate target.

This will reduce the redundant objects built
in Chromium when opening up test_support for the fuzzer
tests.

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2849823002
Cr-Commit-Position: refs/heads/master@{#17972}

2017-05-02 11:51:40 +00:00

api

Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )

2017-04-28 13:03:40 +00:00

audio

Have AudioSendStream register CNG payload types with the RtpRtcpModule.

2017-04-27 15:03:42 +00:00

base

Relanding: Fixing crash that can occur if signal is modified while firing.

2017-05-02 05:24:20 +00:00

call

Injectable audio encoders: WebRtcVoiceEngine and company

2017-04-27 09:08:52 +00:00

common_audio

Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )

2017-04-28 12:24:50 +00:00

common_video

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

examples

Disable video quality test due to unexpected failures

2017-05-02 07:09:32 +00:00

logging

Create tool to print statistics about the file size usage of an RTC event log.

2017-04-28 14:48:17 +00:00

media

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

modules

Extract iOS SDK helpers to separate target.

2017-05-02 11:51:40 +00:00

ortc

Enabling 'gn check' on webrtc/video.

2017-04-25 07:35:35 +00:00

p2p

Don't crash if STUN error message is missing ERROR-CODE attribute.

2017-04-26 16:21:22 +00:00

Run some peer connection end-to-end tests with an empty audio decoder factory

2017-04-27 10:55:57 +00:00

sdk

Extract iOS SDK helpers to separate target.

2017-05-02 11:51:40 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )

2017-04-28 12:24:50 +00:00

test

Extract iOS SDK helpers to separate target.

2017-05-02 11:51:40 +00:00

tools

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

video

Fix video_loopback to work with -duration flag: add missing ntp_timestamp to frames

2017-05-02 07:48:41 +00:00

voice_engine

Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )

2017-04-28 12:24:50 +00:00

.gitignore

…

BUILD.gn

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

call.h

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codereview.settings

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common_types.cc

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +00:00

common_types.h

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )

2017-04-05 10:02:20 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

…

OWNERS

…

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_frame.h

Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )

2017-04-28 13:03:40 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

video_send_stream.h

Add number of quality adapt changes to VideoSendStream stats.

2017-04-04 12:51:49 +00:00

webrtc.gni

Remove the expat and jsoncpp dependencies to reduce binary size.

2017-04-24 19:05:06 +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.