platform-external-webrtc

Files

Zijie He 09f16c6a0a Add new constructors for all DesktopFrame inheritances

This change adds constructors for all DesktopFrame inheritances to pass in
DesktopRect instead of DesktopSize.
Because the newly added constructors and DesktopFrame::top_left() function are
not actively used, this change should have no logic impact.

Bug: webrtc:7950
Change-Id: If78187865c991211dfc28d3723403ce6e6fe0290
Reviewed-on: https://chromium-review.googlesource.com/590508
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19204}

2017-08-02 01:07:44 +00:00

api

Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt

2017-08-01 06:22:01 +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

Add new constructors for all DesktopFrame inheritances

2017-08-02 01:07:44 +00:00

ortc

Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt

2017-08-01 06:22:01 +00:00

p2p

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

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

Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt

2017-08-01 06:22:01 +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: Fix quality scaling for injected encoders

2017-08-01 15:07:59 +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

Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates

2017-08-01 13:30:28 +00:00

voice_engine

Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates

2017-08-01 13:30:28 +00:00

.gitignore

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BUILD.gn

Remove remains of webrtc/base

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

codereview.settings

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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

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LICENSE_THIRD_PARTY

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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

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

…

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.