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andrew@webrtc.org fff3fd35a6 Add arm64 to typedefs.h 
This is the first step to get a buildable chrome_shell_apk for arm64.

BUG=chromium:354405
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10819004

Patch from Primiano Tucci <primiano@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-27 19:48:53 +00:00
..
build
Enable VS2013 for Windows compilation by default.
2014-03-12 14:41:58 +00:00
common_audio
Add fir_filter to common_audio
2014-03-24 10:16:11 +00:00
common_video
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
examples
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
2014-01-24 18:42:12 +00:00
modules
Don't disable experimental AGC in audioproc.
2014-03-27 00:11:11 +00:00
system_wrappers
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
test
VoE changes to allow forwarding of packets from VoE to ViE BWE.
2014-03-24 10:38:25 +00:00
tools
Allow loopback tests to do TURN when served from webrtc.googlecode.com.
2014-03-27 19:24:45 +00:00
video
Re-submit: rev5775
2014-03-26 21:00:21 +00:00
video_engine
Re-submit: rev5775
2014-03-26 21:00:21 +00:00
voice_engine
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
2014-03-25 19:57:07 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Remove internal codecs from VideoSendStream.
2014-03-19 08:43:57 +00:00
common_types.h
Removing VideoCodecDerived and moving methods inside VideoCodec.
2014-03-25 18:44:58 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Remove external encryption API for VoE.
2014-02-18 11:27:22 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add arm64 to typedefs.h
2014-03-27 19:48:53 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Make VideoReceiveStream::GetStats() const.
2014-02-07 15:32:45 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Simplify pacer interface.
2014-03-19 10:59:52 +00:00
webrtc_examples.gyp
Android example apps: fixes issue where useful failure information was suppressed.
2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.