webrtc_m130

History

andrew@webrtc.org 6f69eb78dd Allow audio devices with up to 64 channels on Mac.

Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.

BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.

R=henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1628004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-06-07 17:56:50 +00:00

build

Fix relative path to .gitignore and other minor changes.

2013-06-07 15:43:04 +00:00

common_audio

Replace the old resampler with SincResampler in the voice engine signal path.

2013-06-03 19:00:29 +00:00

common_video

Restore relative include paths to libyuv.

2013-06-04 23:53:48 +00:00

modules

Allow audio devices with up to 64 channels on Mac.

2013-06-07 17:56:50 +00:00

system_wrappers

Implement DetectNumberOfCores on Android and make it consistent on Linux and Android

2013-06-05 16:37:42 +00:00

test

Remove #pragma once

2013-05-29 13:19:09 +00:00

tools

Include files from webrtc/.. paths in tools/

2013-06-04 08:14:10 +00:00

video_engine

Removing functionality for inserting pre-encoded frames instead of raw

2013-06-07 13:57:57 +00:00

voice_engine

Fixed Rtp/Rtcp tests

2013-06-07 16:03:19 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Replace ExtraCodecOptions with new Config class that supports multiple settings at once.

2013-05-14 08:02:25 +00:00

common_unittest.cc

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

common.h

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Disable clang C++11 warnings to permit OVERRIDE keyword.

2013-05-08 08:34:34 +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.