admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
magjed b8853ca8d3 ObjC: Add support for injectable native audio and video codecs 
The native interface is added in the private file
webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h.

BUG=webrtc:8093

Review-Url: https://codereview.webrtc.org/3012443002
Cr-Commit-Position: refs/heads/master@{#19576}
2017-08-29 10:57:22 +00:00
..
api
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
audio
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
call
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Change logging method from WEBRTC_TRACE macro to LOG macro.
2017-08-22 07:29:13 +00:00
examples
Android: Replace webrtc_jni namespace with nested jni namespace
2017-08-28 23:24:06 +00:00
logging
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
media
Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder
2017-08-29 07:58:52 +00:00
modules
Now uses builder class for AudioRecord objects from SDK 23.
2017-08-29 07:24:32 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
pc
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
rtc_base
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
rtc_tools
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
sdk
ObjC: Add support for injectable native audio and video codecs
2017-08-29 10:57:22 +00:00
stats
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
video
Removing dependencies on stub headers within WebRTC.
2017-08-28 13:46:48 +00:00
voice_engine
Add logging host lookups made by TurnPort to the RtcEventLog.
2017-08-29 10:20:58 +00:00
.gitignore
…
BUILD.gn
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +00:00
codereview.settings
…
common_types.cc
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
common_types.h
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +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
Removing video_{receive,send}_stream.h stub headers.
2017-08-29 10:35:02 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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
webrtc.gni
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +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.