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zhihuang ab97e18fa9 Fix the binary size regression on Chromium Windows. 
There is a dependency chain from Chromium windows main_dll to Opus
which should never exist. We used to rely on rtc_static_library
to break this chain. So this CL replaced some rtc_source_set
with rtc_static_library.

libvpx fix (https://chromium-review.googlesource.com/c/544107/) for
ios-simulator linking issue is landed and this CL can be sumbitted once the new
Chromium is rolled into WebRTC.

BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2947273002
Cr-Commit-Position: refs/heads/master@{#18709}
2017-06-22 08:28:59 +00:00
..
api
Implement timing frames.
2017-06-19 14:18:55 +00:00
audio
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
base
Upgrade to (D)TLS1.2 using the new BoringSSL (D)TLSv1_2_method functions
2017-06-22 00:30:04 +00:00
call
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )
2017-06-21 08:02:59 +00:00
examples
Use information about blacklisted devices in video_quality_loopback_test
2017-06-19 08:16:45 +00:00
logging
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
media
Fix the binary size regression on Chromium Windows.
2017-06-22 08:28:59 +00:00
modules
Fix timing frames and FEC conflict
2017-06-21 15:23:19 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
pc
Fix the binary size regression on Chromium Windows.
2017-06-22 08:28:59 +00:00
rtc_base
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
sdk
Fix AVFoundation framework import
2017-06-21 10:44:05 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Added new AudioProcessing fuzzer
2017-06-20 12:26:55 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
What can't loopback test be more like full stack test?
2017-06-21 16:00:17 +00:00
voice_engine
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
.gitignore
…
BUILD.gn
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
codereview.settings
…
common_types.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
common_types.h
Implement timing frames.
2017-06-19 14:18:55 +00:00
config.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
config.h
Implement timing frames.
2017-06-19 14:18:55 +00:00
DEPS
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +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
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +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.