webrtc_m130

History

stefan@webrtc.org 5b91464edf Allow an aggregated partition to spill over to a new packet.

Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-03 10:26:12 +00:00

build

refactor the gyp file to gypi file.

2011-09-12 12:24:39 +00:00

common_audio

In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7.

2011-09-26 16:35:25 +00:00

common_video

refactor the gyp file to gypi file.

2011-09-12 12:24:39 +00:00

modules

Allow an aggregated partition to spill over to a new packet.

2011-10-03 10:26:12 +00:00

system_wrappers

Added compare methods for TickInterval class.

2011-09-23 11:33:31 +00:00

video_engine

Reverting changelist 666 since it broke the build on Mac.

2011-10-03 07:37:41 +00:00

voice_engine

Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected.

2011-09-28 17:45:03 +00:00

common_settings.gypi

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

common_types.h

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

engine_configurations.h

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

LICENSE

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +00:00

LICENSE_THIRD_PARTY

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +00:00

PATENTS

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +00:00

README.chromium

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

typedefs.h

Add missing intrinsic casts for VS 2005.

2011-09-19 18:48:25 +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.