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henrik.lundin@webrtc.org 2a26734f04 Partial revert of r7396 
This change reverts a small part of what was done in r7396. It seems
like that change uncovered another issue with NEON.

BUG=4177,chrome-os-partner:31534
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8043 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 20:52:21 +00:00
..
base
Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS.
2015-01-07 18:01:07 +00:00
build
Use Java based audio as default for WebRTC.
2015-01-12 14:27:23 +00:00
common_audio
common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin()
2015-01-12 05:53:43 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
Added support for an Origin header in STUN messages.
2015-01-10 00:47:02 +00:00
modules
Partial revert of r7396
2015-01-12 20:52:21 +00:00
overrides
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
p2p
Add support for trying alternate server (STUN 300 error message) on TCP
2015-01-10 02:41:32 +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
Hack clock_unittest fix for parallel execution.
2015-01-05 19:08:58 +00:00
test
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
tools
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +00:00
video
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
video_engine
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
voice_engine
GN: Fix Windows build.
2015-01-08 21:22:01 +00:00
.gitignore
…
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
common.gyp
…
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
…
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
…
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +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
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +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
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
…
video_send_stream.h
Enable payload-based padding by default and remove the API.
2014-12-19 15:33:17 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
webrtc.gyp
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +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.