platform-external-webrtc

Files

henrik.lundin 02739d9149 NetEqTest: Extend the callback structure

This change allows more callbacks to be registered to the test object.
The callbacks are used to give the user of the test object the ability
to instrument the test object. This CL specifically adds
instrumentation points just after a packet is inserted into NetEq, and
just after audio is pulled out of NetEq.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2851383004
Cr-Commit-Position: refs/heads/master@{#18014}

2017-05-04 13:09:06 +00:00

api

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

audio

Add --quick flag to low bandwidth audio test

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

base

fixing braces around initialization of subobject

2017-05-03 18:31:38 +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

Initial implementation of Android audio playback error handling.

2017-05-03 18:32:39 +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

NetEqTest: Extend the callback structure

2017-05-04 13:09:06 +00:00

ortc

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

2017-05-02 13:46:30 +00:00

p2p

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

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

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

2017-05-02 13:46:30 +00:00

sdk

Add functionality to clear surface to a specific color in EglRenderer.

2017-05-04 13:06:56 +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

NetEqTest: Extend the callback structure

2017-05-04 13:09:06 +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.