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bjornv@webrtc.org 5964fe0f86 audio_processing: DestroyHandle() now returns void 
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.

BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 06:52:28 +00:00
..
build
Set include_internal_video_capture=1 for video_capture_tests
2014-04-16 12:59:49 +00:00
common_audio
common_audio: VADFree() now returns void
2014-04-22 04:45:35 +00:00
common_video
Propagate capture ntp timestamp from rtp to renderer.
2014-04-15 17:46:33 +00:00
examples
VideoCaptureAndroid: stop referencing ViERenderer
2014-04-08 22:55:07 +00:00
modules
audio_processing: DestroyHandle() now returns void
2014-04-22 06:52:28 +00:00
system_wrappers
Add Chromium's ScopedVector.
2014-04-18 21:20:54 +00:00
test
Remove use of tmpnam.
2014-04-16 08:04:26 +00:00
tools
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
video
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
video_engine
Propagate capture ntp timestamp from rtp to renderer.
2014-04-15 17:46:33 +00:00
voice_engine
Resampler modifications in preparation for arbitrary audioproc rates.
2014-04-19 00:32:07 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Remove internal codecs from VideoSendStream.
2014-03-19 08:43:57 +00:00
common_types.h
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-04-17 10:45:01 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-04-14 14:12:50 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +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
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +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
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Fix ARM64 detection.
2014-04-01 01:19:08 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Make VideoReceiveStream::GetStats() const.
2014-02-07 15:32:45 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Simplify pacer interface.
2014-03-19 10:59:52 +00:00
webrtc_examples.gyp
Make WebRTC Android examples build without sourcing envsetup.sh
2014-04-15 08:35:00 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.