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andrew@webrtc.org a22485eaf0 MIPS optimizations for AEC audio processing module 
Added new optimizations for MIPS that were removed in r6797.
For more information about this see https://code.google.com/p/webrtc/source/detail?r=6797

R=andrew@webrtc.org, djordje.pesut@imgtec.com

Review URL: https://webrtc-codereview.appspot.com/15259004

Patch from Ljubomir Papuga <ljubomir.papuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7010 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:51:28 +00:00
..
base
Precompile out our standalone CHECK macros in a Chromium build.
2014-08-28 19:00:15 +00:00
build
Add LTO support for Android Chromium.
2014-08-29 17:41:13 +00:00
common_audio
Add LTO support for Android Chromium.
2014-08-29 17:41:13 +00:00
common_video
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
examples
Remove former team members from OWNERS and WATCHLISTS
2014-08-26 06:12:08 +00:00
modules
MIPS optimizations for AEC audio processing module
2014-08-29 17:51:28 +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
sound
Add talk owners to migrated talk folders
2014-08-28 16:03:58 +00:00
system_wrappers
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
test
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
GN: Implement video_engine, video_capture and video_render.
2014-08-28 13:51:08 +00:00
video_engine
GN: Implement video_engine, video_capture and video_render.
2014-08-28 13:51:08 +00:00
voice_engine
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
.gitignore
…
BUILD.gn
GN: Disable Chromium clang plugins for standalone build.
2014-08-25 14:15:35 +00:00
call.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Check before send/receive rtp header extensions.
2014-07-20 15:27:35 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +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
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
…
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_receive_stream.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +00:00
video_renderer.h
…
video_send_stream.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
webrtc_examples.gyp
WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.
2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
RTP video playback tool using Call APIs.
2014-08-06 16:26:56 +00:00
webrtc.gyp
Create a copy of talk/sound under webrtc/sound.
2014-08-26 22:04:04 +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.