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webrtc_m130/webrtc
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zstein 83dc6b6f53 Remove default implementation of PeerConnectionInterface::SetBitrate. 
This was included to avoid breaking chromium, which now includes its own implementation (725cb26dab).

BUG=webrtc:7395

Review-Url: https://codereview.webrtc.org/2924243003
Cr-Commit-Position: refs/heads/master@{#19063}
2017-07-17 22:09:30 +00:00
..
api
Remove default implementation of PeerConnectionInterface::SetBitrate.
2017-07-17 22:09:30 +00:00
audio
Don't run NoBandwidthDropAfterDtx test on andriod because it's flaky.
2017-07-17 15:19:27 +00:00
base
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
2017-07-11 23:56:05 +00:00
call
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
common_audio
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
common_video
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
examples
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
2017-07-11 23:56:05 +00:00
logging
nit: Avoid pointer-to-unique_ptr in RtcEventLogImpl
2017-07-16 23:44:08 +00:00
media
Revert "Prefer external video codecs over internal in SDP"
2017-07-17 15:45:17 +00:00
modules
Expose IsCurrentSessionSupported() from ScreenCapturerWinDirectx
2017-07-17 18:58:03 +00:00
ortc
Disabling test on iOS64 debug bot
2017-07-14 22:19:03 +00:00
p2p
Reinstate "Add additional check when setting RTCConfiguration"
2017-07-14 23:40:53 +00:00
pc
Reinstate "Add additional check when setting RTCConfiguration"
2017-07-14 23:40:53 +00:00
rtc_base
Reinstate "API for periodically regathering ICE candidates"
2017-07-14 22:26:05 +00:00
rtc_tools
Added -show_detector_state which show the detector state in the total bitrate graph.
2017-07-14 13:30:03 +00:00
sdk
Android bindings for ice_regather_interval_range RTCConfiguration option
2017-07-17 20:09:43 +00:00
stats
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
system_wrappers
Fix DetectNumberOfCores for 32-bit processes
2017-07-13 17:06:12 +00:00
test
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
video
Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
2017-07-13 07:47:03 +00:00
voice_engine
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
.gitignore
…
BUILD.gn
Reimplemeted "Test and fix for huge bwe drop after alr state"
2017-07-17 08:41:41 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
…

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.