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bjornv@webrtc.org aafd7a88c5 The correct fix of workaround in r6261. 
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c

BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org, glaznev@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:53:51 +00:00
..
base Rebase webrtc/base with r6250: 2014-05-29 15:53:39 +00:00
build Make it possible to build webrtc for arm64. 2014-06-04 17:15:42 +00:00
common_audio The correct fix of workaround in r6261. 2014-06-05 08:53:51 +00:00
common_video Android: cleanup gtest_target_type conditions. 2014-06-04 20:46:50 +00:00
examples WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker 2014-05-21 03:37:45 +00:00
modules The correct fix of workaround in r6261. 2014-06-05 08:53:51 +00:00
overrides/webrtc/base Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 2014-05-21 16:52:14 +00:00
system_wrappers Android: cleanup gtest_target_type conditions. 2014-06-04 20:46:50 +00:00
test Android: cleanup gtest_target_type conditions. 2014-06-04 20:46:50 +00:00
tools Android: cleanup gtest_target_type conditions. 2014-06-04 20:46:50 +00:00
video Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 2014-06-05 08:25:29 +00:00
video_engine Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 2014-06-05 08:25:29 +00:00
voice_engine Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 2014-06-05 08:25:29 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
call.h Add DeliveryStatus enum to DeliverPacket(). 2014-05-14 13:57:12 +00:00
common_types.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.cc Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
config.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-05-12 12:19:19 +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
OWNERS Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 2014-04-14 20:08:03 +00:00
PATENTS Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
supplement.gypi Roll chromium_revision 260462:266514 2014-04-29 09:36:40 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +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 Rename Start/Stop in Video{Send,Receive}Streams. 2014-04-24 11:13:21 +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 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 2014-06-05 08:25:29 +00:00
webrtc_examples.gyp Add webrtc field trials API. 2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate Move realtime tests to webrtc_perf_tests. 2013-12-13 12:48:05 +00:00
webrtc_tests.gypi Android: cleanup gtest_target_type conditions. 2014-06-04 20:46:50 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.