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webrtc_m130/webrtc
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deadbeef 82215872f8 Add Java binding for new getStats implementation. 
Very similar to the current interface, but matches the new C++ structure, and
exposes the stats values as Objects which can be downcast to more specific
types (where the previous API only exposed the values as strings).

BUG=webrtc:6871

Review-Url: https://codereview.webrtc.org/2807933003
Cr-Commit-Position: refs/heads/master@{#17746}
2017-04-18 17:27:51 +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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +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
Address denicija's comments for AppRTCMobile video codec setting.
2017-04-11 12:36:43 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Fix SDP stream ID mismatch issue when a track's stream changes.
2017-04-14 17:41:32 +00:00
modules
Added audioSessionDidActivate and audioSessionDidDectivate so that RTCAudioSession can be notified when the audio session is activated or deactivated by iOS outside of the app.
2017-04-18 16:09:35 +00:00
ortc
Enabling 'gn check' on //webrtc/ortc.
2017-04-11 09:52:49 +00:00
p2p
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
pc
Fix SDP stream ID mismatch issue when a track's stream changes.
2017-04-14 17:41:32 +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
Revert of Enabling 'gn check' on //webrtc/test. (patchset #10 id:180001 of https://codereview.webrtc.org/2796363003/ )
2017-04-13 09:13:09 +00:00
tools
Revert of "Add first part of the network_tester functionality" (patchset #1 id:1 of https://codereview.webrtc.org/2811253005/ )
2017-04-13 10:03:51 +00:00
video
Make EndToEndTest.ContentTypeSwitches more robust by sending more frames before test ends
2017-04-18 16:17:53 +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
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
common_types.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
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
…
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
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +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.