platform-external-webrtc

Files

deadbeef 7145280954 Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver.

The test attempted to call InsertDtmf before verifying that the DTLS
handshake was complete. Unlike a data channel, the DTMF sender doesn't
do any buffering, so this isn't reliable.

BUG=webrtc:7547
NOTRY=True

Review-Url: https://codereview.webrtc.org/2855573004
Cr-Commit-Position: refs/heads/master@{#18041}

2017-05-08 00:21:01 +00:00

api

Add myself as OWNER of webrtc/api/ and webrtc/base/

2017-05-05 21:04:54 +00:00

audio

Add --quick flag to low bandwidth audio test

2017-05-04 10:06:18 +00:00

base

Add myself as OWNER of webrtc/api/ and webrtc/base/

2017-05-05 21:04:54 +00:00

call

Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.

2017-05-02 13:46:30 +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

Reland of Move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/ (patchset #1 id:1 of https://codereview.webrtc.org/2862663003/ )

2017-05-04 10:41:33 +00:00

examples

Make fps NSInteger in startCaptureWithDevice.

2017-05-05 07:45:30 +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

Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ )

2017-05-04 09:11:12 +00:00

modules

Prevent residual echo likelihood values greater than 1.0.

2017-05-05 12:50:10 +00:00

ortc

Delete MediaController class, move Call ownership to PeerConnection.

2017-05-05 09:23:02 +00:00

p2p

Fixing DCHECK in turnport.cc that was broken by refactoring.

2017-05-03 16:48:35 +00:00

Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver.

2017-05-08 00:21:01 +00:00

sdk

When a data channel fails to be created, return nil instead of crashing.

2017-05-05 17:00:18 +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

Fixing pseudotcp_parser_fuzzer crash with NO_MAIN_THREAD_WRAPPING.

2017-05-04 19:16:28 +00:00

tools

Delete RawVideoType enum, use the VideoType enum instead.

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

video

Update decode/render fps stats when calling VideoReceiveStream::GetStats

2017-05-04 09:47:13 +00:00

voice_engine

Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()

2017-05-03 15:30:15 +00:00

.gitignore

…

BUILD.gn

Remove config.h from //webrtc:webrtc target.

2017-05-03 12:28:46 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

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

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +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_frame.h

Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ )

2017-05-04 09:11:12 +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

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.