webrtc_m130

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hlundin@google.com 6b04739e04 Route CodecSpecificInfo from encoder to packetizer

Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-01 08:32:57 +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

Route CodecSpecificInfo from encoder to packetizer

2011-07-01 08:32:57 +00:00

peerconnection

Switch use of wsprintfW out for the libjingle equivalent.

2011-06-30 07:30:13 +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

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.

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