webrtc_m130

History

ilnik 26e5cbd6bb Add Jpeg frame writer for test support.

Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.

BUG=webrtc:8030

Review-Url: https://codereview.webrtc.org/2990563002
Cr-Commit-Position: refs/heads/master@{#19414}

2017-08-18 16:00:04 +00:00

api

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

2017-08-18 11:09:40 +00:00

audio

Replace absolute path with relative path for GN files.

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

call

Change ThreadChecker to SequencedTaskChecker in internal::Call

2017-08-18 09:47:08 +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

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

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

Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.

2017-08-16 09:48:33 +00:00

sdk

Make a friendlier RTCVideoCodecInfo initializer for when only name and

2017-08-18 13:34:09 +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

Add Jpeg frame writer for test support.

2017-08-18 16:00:04 +00:00

video

Add Jpeg frame writer for test support.

2017-08-18 16:00:04 +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.