webrtc_m130

History

xians@webrtc.org 2f84afad30 Merge r4326 from stable to trunk.

r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.

Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-07-31 16:23:37 +00:00

build

Don't set clang_use_chrome_plugins in common.gypi

2013-07-26 00:55:46 +00:00

common_audio

Downstream latest Chromium SincResampler changes.

2013-07-25 22:04:30 +00:00

common_video

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

modules

Merge r4326 from stable to trunk.

2013-07-31 16:23:37 +00:00

system_wrappers

Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.

2013-07-16 16:37:22 +00:00

test

Added libjingle_peerconnection_java_unittest to buildbot_tests.py

2013-07-31 10:50:30 +00:00

tools

Fix some chromium-style warnings in webrtc/tools/

2013-07-30 12:50:59 +00:00

video_engine

Updated WebRTC version to 3.37

2013-07-29 16:57:21 +00:00

voice_engine

Merge r4326 from stable to trunk.

2013-07-31 16:23:37 +00:00

.gitignore

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

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

common_types.h

webrtc/common_types.h: Document bitrate fields' units.

2013-07-23 18:32:10 +00:00

common_unittest.cc

Fix unnamed-type-template-args warnings on clang.

2013-06-27 10:18:09 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

engine_configurations.h

Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.

2013-06-19 17:03:47 +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

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +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

webrtc.gyp

Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle

2013-07-23 18:15:11 +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.