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cduvivier@google.com 181f543de4 AEC specific version of " Real Discrete Fourier Transform".
Lots of AEC CPU usage is coming from calls to 'rdft'. To optimize this,
deep changes (modification of memory layout, ...) have to be done and it
is not practical to do them in an utility library. Most of these changes
will occur in subsequent CLs.

The new file 'aec_core_rdft.c' is a copy of 'modules/audio_processing/
utility/fft4g.c' whose size has been significantly reduced by removing
all code non-necessary to compute rdft. The main entry point and utility
functions have also been modified to take into account the fact that all
'rdft' calls performed by AEC have a length of 128. This yields:
* 1.8% AEC overall speedup for the straight C path.
* 2.3% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/44008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@126 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-24 18:22:47 +00:00
build Proof-of-concept proposal for a standalone webrtc build without using gyp_chromium etc. This adds the necessary scripts and gyp files. The idea is to assume that we are building within Chromium; in that case common.gypi (which every gyp file includes) provides the necessary logic to build webrtc. 2011-06-08 23:09:32 +00:00
common_audio enable optimized code for android 2011-06-17 17:39:05 +00:00
common_video VPLIB: Fixing a bug in ConvertYUY2TOI420 + some code clean-up 2011-06-20 18:03:34 +00:00
interface git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:22:19 +00:00
modules AEC specific version of " Real Discrete Fourier Transform". 2011-06-24 18:22:47 +00:00
peerconnection/samples If this gives you problems, delete the third_party/libjingle directory and sync again 2011-06-08 11:24:32 +00:00
system_wrappers Minor update that fixes crash in system wrappers unittest. (the crash was in the test of map_wrapper). 2011-06-23 17:30:17 +00:00
test/data/audio_processing Creating a new directory for test data files, and moving audio_processing files there. 2011-06-23 11:45:12 +00:00
third_party_mods add android makefile, some modification in vpx makefile to build encoder from c source for now 2011-06-07 17:24:39 +00:00
tools git-svn-id: http://webrtc.googlecode.com/svn/trunk@8 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:42:35 +00:00
video_engine Correcting two auto test errors. 2011-06-23 08:43:37 +00:00
voice_engine add command line test app to gyp build 2011-06-20 17:05:14 +00:00
android-webrtc.mk add android makefile, some modification in vpx makefile to build encoder from c source for now 2011-06-07 17:24:39 +00:00
Android.mk add command line test app to gyp build 2011-06-20 17:05:14 +00:00
AUTHORS git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
codereview.settings git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
common_settings.gypi Proof-of-concept proposal for a standalone webrtc build without using gyp_chromium etc. This adds the necessary scripts and gyp files. The idea is to assume that we are building within Chromium; in that case common.gypi (which every gyp file includes) provides the necessary logic to build webrtc. 2011-06-08 23:09:32 +00:00
common_types.h Replacing kTraceVqe with kTraceAudioProcessing. 2011-05-31 22:15:52 +00:00
DEPS If this gives you problems, delete the third_party/libjingle directory and sync again 2011-06-08 11:24:32 +00:00
engine_configurations.h git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
libvpx.mk add android makefile, some modification in vpx makefile to build encoder from c source for now 2011-06-07 17:24:39 +00:00
LICENSE git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
license_template.txt git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
LICENSE_THIRD_PARTY AEC specific version of " Real Discrete Fourier Transform". 2011-06-24 18:22:47 +00:00
OWNERS Global OWNERS. 2011-06-21 08:09:52 +00:00
PATENTS Modified patent grant 2011-05-31 22:47:37 +00:00
PRESUBMIT.py Adding owners check in presubmit script. 2011-06-09 07:07:24 +00:00
README.chromium Adding README.chromium 2011-06-21 14:12:46 +00:00
typedefs.h git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
video_engine.gyp git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00
voice_engine.gyp add command line test app to gyp build 2011-06-20 17:05:14 +00:00
webrtc.gyp add command line test app to gyp build 2011-06-20 17:05:14 +00:00

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.