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tommi@google.com 53af7595d1 Switch the sample client back to render the videos in the main window 
instead of two popup windows.  This also demonstrates one way of
implementing the VideoRenderer interface.
Review URL: http://webrtc-codereview.appspot.com/51004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@143 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-04 12:47:37 +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
VCM: 1. Updating handling of empty packets. 2. Updating JB test. 3. Removing un-used code.
2011-07-01 18:15:11 +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/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
Route CodecSpecificInfo from encoder to packetizer
2011-07-01 08:32:57 +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

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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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