platform-external-webrtc

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marpan@google.com 0d7e5bc712 Fix bug on key frame boost allocation, and some update/cleanup to same function.

Review URL: http://webrtc-codereview.appspot.com/50001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@123 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-06-23 16:36:33 +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

Fix bug on key frame boost allocation, and some update/cleanup to same function.

2011-06-23 16:36:33 +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

Fixes Valgrind warnings in system_wrapper.

2011-06-21 20:23:23 +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

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

There where multiple lines on the same row. I also alphabetized it.

2011-06-07 11:40:02 +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.

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