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webrtc_m130/webrtc
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kwiberg d1d79f6866 Remove dead code 
This code became dead when the builtin audio codec factories were
rewritten in https://codereview.webrtc.org/2997713002/.

BUG=webrtc:7821, webrtc:7822

Review-Url: https://codereview.webrtc.org/3003603002
Cr-Commit-Position: refs/heads/master@{#19535}
2017-08-26 05:22:42 +00:00
..
api
Remove dead code
2017-08-26 05:22:42 +00:00
audio
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
call
Reverse |rtx_payload_types| map, and rename.
2017-08-25 11:44:25 +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
Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually
2017-08-25 13:21:52 +00:00
logging
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
media
Reverse |rtx_payload_types| map, and rename.
2017-08-25 11:44:25 +00:00
modules
Remove dead code
2017-08-26 05:22:42 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Renamed dtlstransportchannel.h/.cc/_unittest.cc.
2017-08-24 01:05:50 +00:00
pc
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
rtc_base
Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually
2017-08-25 13:21:52 +00:00
rtc_tools
Move PacedSender ownership to RtpTransportControllerSend.
2017-08-22 14:16:49 +00:00
sdk
Android: Expose EglBase.swapBuffers with presentation time
2017-08-25 13:28:00 +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
Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually
2017-08-25 13:21:52 +00:00
video
Update video_replay tool to be able to dump .jpg files.
2017-08-25 14:24:21 +00:00
voice_engine
Uncomment thread-checkers in ChannelProxy
2017-08-25 08:50:58 +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
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
common_types.h
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +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 video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +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
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_send_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
webrtc.gni
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +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.