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webrtc_m130/webrtc
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danilchap ef8d773d26 Add read support of RtpStreamId/RepairedRtpStreamId header extensions. 
BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2805023002
Cr-Commit-Position: refs/heads/master@{#17759}
2017-04-19 09:59:48 +00:00
..
api
Modified the rtp_receiver_unittests.
2017-04-11 18:28:10 +00:00
audio
Fix lint errors to enable stricter PyLint rules
2017-04-12 19:06:13 +00:00
base
base/testutils.h: Use namespace webrtc::testing instead of plain testing
2017-04-18 10:18:22 +00:00
call
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
common_audio
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Add error codes for Android audio recording start errors.
2017-04-18 17:30:29 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
modules
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
ortc
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
p2p
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
pc
Adding integration test for unsignaled inbound RTP stream stats.
2017-04-18 23:01:17 +00:00
sdk
Add Java binding for new getStats implementation.
2017-04-18 17:27:51 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
test
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
tools
Reland "Add first part of the network_tester functionality".
2017-04-19 08:58:38 +00:00
video
Do not report cpu limited resolution stats when degradation preference is disabled and no scaling is done.
2017-04-19 09:01:06 +00:00
voice_engine
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +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
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
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
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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_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
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.