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zstein 634977b611 SignalPacketReceived should pass packet as a pointer instead of a non-const reference. 
See https://google.github.io/styleguide/cppguide.html#Reference_Arguments.
The Bind to ProcessPacket in OnPacketReceived is safe because Bind captures arguments by value.

BUG=webrtc:7013

Review-Url: https://codereview.webrtc.org/2980923002
Cr-Commit-Position: refs/heads/master@{#19028}
2017-07-14 19:30:04 +00:00
..
api
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
audio
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
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
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
media
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
modules
Patch set 1:
2017-07-14 12:22:33 +00:00
ortc
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
p2p
Revert "API for periodically regathering ICE candidates"
2017-07-14 15:42:02 +00:00
pc
SignalPacketReceived should pass packet as a pointer instead of a non-const reference.
2017-07-14 19:30:04 +00:00
rtc_base
Revert "API for periodically regathering ICE candidates"
2017-07-14 15:42:02 +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
Revert of iOS - Add iceRegatherIntervalRange. (patchset #1 id:1 of https://codereview.webrtc.org/2976953002/ )
2017-07-14 14:59:05 +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
Use relative paths in GN files.
2017-07-11 13:20:45 +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
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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
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.