generic-library/webrtc
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ajm@google.com 7c4469bf61 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. 
Review URL: http://webrtc-codereview.appspot.com/56002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@151 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-06 17:45:37 +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
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
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.
Description
No description provided
Readme 108 MiB
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