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cduvivier@google.com d0159d8eb0 aec_rdft_128: one entry point for each sign. 
Review URL: http://webrtc-codereview.appspot.com/61007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@153 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-06 23:35:37 +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 Changing JPEG API to to accept rawImage and encodedImage; moved video_image.h from modules/video_coding/codecs to common_video/interface, and some general re-write to JPEG, especially with regard to memory handling. Required VCM/ViE changes are also included. 2011-07-01 01:17:49 +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_rdft_128: one entry point for each sign. 2011-07-06 23:35:37 +00:00
peerconnection Switch the sample client back to render the videos in the main window 2011-07-04 12:47:37 +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 Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. 2011-07-06 17:45:37 +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 Route CodecSpecificInfo from encoder to packetizer 2011-07-01 08:32:57 +00:00
voice_engine Prepares a move all data files required by VoiceEngine into one common place. 2011-07-04 15:52:04 +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 Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. 2011-07-06 17:45:37 +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 Fixed minor type for Mac and Linux target. 2011-07-04 16:07:30 +00:00
webrtc.gyp add command line test app to gyp build 2011-06-20 17:05:14 +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.