webrtc_m130

History

Zijie He 5af2af36ee Remove resolution_tracker_ from dxgi_texture

DxgiTexture now does not rely on a fixed resolution, so the ResolutionTracker
can be removed from it.

This change does not have logic impact, the upper component
(DxgiDuplicatorController) always reinitializes itself once the screen
resolution changes. And this check is also a legacy one: DxgiFrame now can take
care of the resolution change itself without needing to return false in
DxgiTexture.

Bug: webrtc:8044
Change-Id: I3ad9ce175f2bc9bf03b0a3985efa2681aa55d14b
Reviewed-on: https://chromium-review.googlesource.com/592247
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19191}

2017-07-31 18:59:12 +00:00

api

G711 implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-07-31 18:34:57 +00:00

audio

Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"

2017-07-26 14:28:51 +00:00

base

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

call

Only one implementation of MockRtpPacketSink once

2017-07-28 17:05:45 +00:00

common_audio

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

common_video

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

examples

Update native plugin dll for turn servers and video.

2017-07-25 23:04:31 +00:00

logging

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

media

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +00:00

modules

Remove resolution_tracker_ from dxgi_texture

2017-07-31 18:59:12 +00:00

ortc

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

p2p

Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.

2017-07-26 23:09:33 +00:00

Make RTCStatsReport::ToString() return JSON-parseable string.

2017-07-28 14:29:12 +00:00

rtc_base

If adapter type is unknown and interface name is "ipsec", treat as VPN.

2017-07-27 22:05:29 +00:00

rtc_tools

Print general usage information for event_log_analyzer

2017-07-20 15:05:09 +00:00

sdk

ObjC RTCEAGLVideoVideo: Check GL context is non-nil in constructor

2017-07-31 16:11:46 +00:00

stats

Make RTCStatsReport::ToString() return JSON-parseable string.

2017-07-28 14:29:12 +00:00

system_wrappers

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

test

Rename isolated_output to test_output and add a method to get the test_output directory.

2017-07-25 14:31:18 +00:00

video

Only one implementation of MockRtpPacketSink once

2017-07-28 17:05:45 +00:00

voice_engine

Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.

2017-07-20 16:57:42 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

common_types.h

Add parsing/serializing for MID RTP header extension.

2017-07-21 17:33:25 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Report interframe delay sum in old GetStats

2017-07-07 08:26:24 +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

Turning on Opus 120ms frame length switch.

2017-07-27 17:23:35 +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.