bjornv@webrtc.org 84f8ec1f9c Changes to tests and tools in audio_processing. ...

- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.

- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.

BUG=N/A
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-19 12:14:33 +00:00

..

audio_coding

Adding test::AudioSink interface and derived classes

2014-06-19 10:02:11 +00:00

audio_conference_mixer

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

audio_device

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

audio_processing

Changes to tests and tools in audio_processing.

2014-06-19 12:14:33 +00:00

bitrate_controller

Ensure that the start bitrate can be set multiple times.

2014-06-19 12:13:00 +00:00

desktop_capture

Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed.

2014-06-06 21:42:00 +00:00

interface

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

media_file

Adding missing break in media_file_utility.cc.

2014-06-04 05:21:56 +00:00

pacing

Enable pacing by default and remove the option to disable it from the new API.

2014-06-12 15:12:25 +00:00

remote_bitrate_estimator

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

rtp_rtcp

Add round-robin selection of send stream to pad on.

2014-06-17 17:32:05 +00:00

utility

Add support of texture frames for video capturer.

2014-05-28 07:00:51 +00:00

video_capture

Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.

2014-06-13 23:31:35 +00:00

video_coding

Add high perf mode to VP8

2014-06-17 16:32:08 +00:00

video_processing/main

Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps.

2014-05-29 19:43:26 +00:00

video_render

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Rename neteq4 folder to neteq

2014-06-09 08:10:28 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00