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webrtc_m130/webrtc
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henrik.lundin 06863c9ce5 Add TimestampUnwrapper and generalize the code 
BUG=webrtc:7467
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device;master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng

Review-Url: https://codereview.webrtc.org/2813593003
Cr-Commit-Position: refs/heads/master@{#17833}
2017-04-24 06:54:13 +00:00
..
api
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
audio
Fix lint errors to enable stricter PyLint rules
2017-04-12 19:06:13 +00:00
base
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
call
Replace first_packet_sent_ms_ in Call.
2017-04-20 06:28:53 +00:00
common_audio
Enable GN check for webrtc/base
2017-04-19 15:37:36 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Removed duplicated annotation in webrtc/examples/androidtests/AndroidManifest.xml
2017-04-20 12:31:32 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
modules
Add TimestampUnwrapper and generalize the code
2017-04-24 06:54:13 +00:00
ortc
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
p2p
Fixing SignalSentPacket for TCP connections.
2017-04-21 21:22:23 +00:00
pc
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
sdk
Android: Prepare moving EglBase10/EglBase14 from to API to src
2017-04-21 08:34:12 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
test
Revert of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #3 id:40001 of https://codereview.webrtc.org/2781713004/ )
2017-04-20 10:54:22 +00:00
tools
Add network tester client [android] to be able to test mobile networks in terms of packet size and sending rates.
2017-04-20 13:56:27 +00:00
video
Only increment scale counter and adaptation stats (and store last_adaptation_request_) if sink_wants_ is updated.
2017-04-21 08:47:31 +00:00
voice_engine
Fix invalid output buffer usage
2017-04-20 22:19:10 +00:00
.gitignore
…
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +00:00
call.h
…
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
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_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +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 number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Revert of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #3 id:40001 of https://codereview.webrtc.org/2781713004/ )
2017-04-20 10:54:22 +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.