webrtc_m130

History

bjornv@webrtc.org 8d63d0ee70 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255

Rewritten the test to only check for valid volume when we have actually received a value from the audio device. To check if we have actually received a volume value is out of the scope for this test.

BUG=webrtc:367
TESTED=trybots
R=tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6123 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-05-13 14:14:56 +00:00

build

Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

2014-05-13 08:15:48 +00:00

common_audio

Fix constness of AudioBuffer accessors.

2014-04-30 16:44:13 +00:00

common_video

Added include of assert.h for files calling assert but missing the include.

2014-04-29 17:54:17 +00:00

examples

WebRTCDemo: correct set trace filter operation.

2014-05-05 04:24:47 +00:00

modules

Deleting all NetEq3 files

2014-05-13 10:40:52 +00:00

system_wrappers

Revert "FieldTrial implementation for webrtc." (rev 6089)

2014-05-13 14:09:40 +00:00

test

Move gflags usage to video_loopback.

2014-05-13 11:26:40 +00:00

tools

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

video

Move gflags usage to video_loopback.

2014-05-13 11:26:40 +00:00

video_engine

Reduced kMaxSampleDiffMs (limit to 22fps).

2014-05-13 13:45:13 +00:00

voice_engine

Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255

2014-05-13 14:14:56 +00:00

.gitignore

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

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

call.h

Remove TraceCallback use from Call.

2014-04-24 11:35:33 +00:00

common_types.h

Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65

2014-04-29 17:50:47 +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 parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +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

Roll chromium_revision 260462:266514

2014-04-29 09:36:40 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Remove ALLOW_UNUSED.

2014-05-05 18:18:02 +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

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +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

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +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

Move gflags usage to video_loopback.

2014-05-13 11:26:40 +00:00

webrtc.gyp

Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

2014-05-13 08:15:48 +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.