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platform-external-webrtc/webrtc
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guoweis@webrtc.org 0c7ec770ff Cleanup: unify rotation to be enum based instead of int for degree. 
Split from https://webrtc-codereview.appspot.com/37029004/

BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8257

Committed: https://code.google.com/p/webrtc/source/detail?r=8276

Review URL: https://webrtc-codereview.appspot.com/37129004

Cr-Commit-Position: refs/heads/master@{#8277}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8277 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 21:01:47 +00:00
..
base
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
2015-02-06 19:47:39 +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
Cleanup: unify rotation to be enum based instead of int for degree.
2015-02-06 21:01:47 +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
audio_processing: Now records mic volume level also when using new AGC
2015-02-06 19:44:46 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
2015-02-06 19:47:39 +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
Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle.
2015-02-06 19:27:12 +00:00
test
Fix the value of the first byte of nal unit
2015-02-05 17:56:44 +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
Implementing a packet router class, used to route RTP packets to the
2015-02-06 13:10:39 +00:00
voice_engine
audio_processing: Now records mic volume level also when using new AGC
2015-02-06 19:44:46 +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
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +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
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +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.