platform-external-webrtc

Files

andrew@webrtc.org 8c845cb623 Relax VoE's max packet length threshold.

The earlier threshold would cause packets from a currently available
codec (L16, 32 kHz, stereo) to be discarded.

TESTED=voe_cmd_test using L16, 32 kHz, stereo now works properly.
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1305008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3936 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-02 15:28:02 +00:00

build

Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi .

2013-04-30 22:53:51 +00:00

common_audio

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

common_video

WebRtc_Word32 -> int32_t in common_video.

2013-04-09 13:36:08 +00:00

modules

Revert 3933 "Remove traces of deprecated WebRtc_Word types."

2013-05-02 09:30:12 +00:00

system_wrappers

Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi .

2013-04-30 22:53:51 +00:00

test

Revert "Updating test file contents to emmastjernloef"

2013-04-17 14:59:28 +00:00

tools

Fix frame_editing_unittest reference file handling.

2013-03-11 15:30:33 +00:00

video_engine

WebRTCDemo Android app to route audio to headphone when it's plugged in.

2013-05-02 03:18:00 +00:00

voice_engine

Relax VoE's max packet length threshold.

2013-05-02 15:28:02 +00:00

.gitignore

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

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

common_types.h

Revert 3933 "Remove traces of deprecated WebRtc_Word types."

2013-05-02 09:30:12 +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

Move src/ -> webrtc/

2012-10-22 18:19:23 +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

Revert 3933 "Remove traces of deprecated WebRtc_Word types."

2013-05-02 09:30:12 +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.