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19f3f71c9873cf5f6d647becd3620ddf8fd6ba7c
platform-external-webrtc/webrtc
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pkasting@chromium.org 19f3f71c98 Fix apparent typo: int -> char. 
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.

BUG=none
TEST=none
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40529004

Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 19:44:42 +00:00
..
base
Control the max IPv6 Networks used by WebRTC.
2015-01-30 00:09:42 +00:00
build
Move use of DEPTH into build_with_chromium==1.
2015-01-30 14:55:20 +00:00
common_audio
Move channel_buffer.{h,cc} to common_audio.
2015-01-28 19:57:44 +00:00
common_video
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
examples
Add arraysize() macro from Chromium, and make use of it in a few places.
2015-01-28 21:37:13 +00:00
libjingle
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
modules
Switched lists of packets to lists of packet pointers. Allows Packet polymorphism.
2015-02-02 14:51:45 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
Adding constness.
2015-01-28 17:33:45 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Rewrite ThreadWindows.
2015-01-30 15:06:44 +00:00
test
Remove ChangeUniqueID.
2015-01-29 12:14:13 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Add tracing for slow paths in new video API.
2015-01-29 12:33:42 +00:00
video_engine
Fix apparent typo: int -> char.
2015-02-02 19:44:42 +00:00
voice_engine
Fixing a VoE test to set correct rate for iSAC
2015-01-30 13:04:47 +00:00
.gitignore
…
BUILD.gn
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Update StreamDataCounter with FEC bytes.
2015-01-27 12:17:29 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
…
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
…
video_send_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
webrtc.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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.