webrtc_m130

History

ilnik bfb124596e Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )

Reason for revert:
webrtc_perf_tests crashes on android and windows due to too large test.

Original issue's description:
> Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
>
> Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2730073002
> Cr-Commit-Position: refs/heads/master@{#17068}
> Committed: d8bd1b1d82

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2734753004
Cr-Commit-Position: refs/heads/master@{#17071}

2017-03-06 15:35:13 +00:00

api

Reland "Enable GN check for webrtc/examples"

2017-03-06 08:29:21 +00:00

audio

Adding placeholder for low bandwidth audio test

2017-03-06 12:01:16 +00:00

base

Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker.

2017-03-03 18:33:18 +00:00

call

Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.

2017-03-03 11:20:24 +00:00

common_audio

Enable GN check for webrtc/common_audio

2017-03-01 15:07:10 +00:00

common_video

Ignore aud and sei NALus when parsing bitstream.

2017-03-06 10:49:36 +00:00

examples

Add kjellander to OWNERS for *.py in examples/android{app,tests}

2017-03-06 11:56:57 +00:00

logging

Remove unused include from rtc_event_log_parser.cc

2017-02-28 09:46:19 +00:00

media

Enable GN check for webrtc/media

2017-03-06 14:04:55 +00:00

modules

Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate.

2017-03-06 14:49:27 +00:00

ortc

Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )

2017-03-04 23:08:44 +00:00

p2p

Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )

2017-03-04 23:08:44 +00:00

Reland: Improve testing of SRTP external auth code paths.

2017-03-06 08:14:10 +00:00

sdk

Android HW decoder: Don't check slice_height for texture output

2017-03-06 13:20:49 +00:00

stats

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

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

system_wrappers

Check __GLIBC_PREREQ availability before use.

2017-03-06 15:34:06 +00:00

test

Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )

2017-03-06 15:35:13 +00:00

tools

Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.

2017-03-06 13:32:21 +00:00

video

Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )

2017-03-06 15:35:13 +00:00

voice_engine

Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.

2017-03-06 13:32:21 +00:00

.gitignore

…

BUILD.gn

Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )

2017-02-26 15:12:50 +00:00

call.h

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

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

codereview.settings

…

common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Remove the Windows Wave audio device implementation.

2017-02-17 22:48:07 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

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

no_op_function.cc

Moving no_op_function.cc out of webrtc/build

2017-01-24 13:25:12 +00:00

OWNERS

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

2016-11-29 10:52:22 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

webrtc.gni

Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )

2017-03-04 03:41:59 +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.