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platform-external-webrtc/webrtc
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solenberg 2324b35890 Remove unused voe_output_test.cc. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2684933007
Cr-Commit-Position: refs/heads/master@{#16506}
2017-02-09 09:23:25 +00:00
..
api
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
audio
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
base
Use fake clock in some more networks tests.
2017-02-08 21:18:00 +00:00
call
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
2017-02-07 09:18:43 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Delete VideoReceiveStream::Config::pre_render_callback.
2017-02-07 17:37:41 +00:00
examples
Move Android video quality loopback script.
2017-02-09 09:05:54 +00:00
logging
Add option to print information about configured SSRCs from RTC event logs.
2017-02-08 17:37:30 +00:00
media
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
modules
Log information (at level LS_INFO) about which overuse estimator is used.
2017-02-08 16:45:20 +00:00
p2p
Increase STUN RTOs to work better on poor networks, such as 2G networks.
2017-02-08 22:42:22 +00:00
pc
Increase STUN RTOs to work better on poor networks, such as 2G networks.
2017-02-08 22:42:22 +00:00
sdk
Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230)
2017-02-09 07:56:43 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Add new target direct_transport and remove fake_network and direct_transport from test_common.
2017-02-08 13:55:51 +00:00
tools
Move Android video quality loopback script.
2017-02-09 09:05:54 +00:00
video
Fix for flaky RemoveOverheadFromBandwidth test.
2017-02-08 16:21:52 +00:00
voice_engine
Remove unused voe_output_test.cc.
2017-02-09 09:23:25 +00:00
.gitignore
…
BUILD.gn
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +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.