solenberg@webrtc.org db60434b31 Re-enable the BitrateEstimatorTest cases for the Call API. ...

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6141 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-05-14 09:15:19 +00:00

..

audio_coding

Remove all use of AudioFrame::energy_ from AudioCodingModule

2014-05-14 09:06:52 +00:00

audio_conference_mixer

The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.

2014-05-13 08:50:02 +00:00

audio_device

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

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

audio_processing

Audio processing: Feed each processing step its choice of int or float data

2014-05-14 09:01:35 +00:00

bitrate_controller

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

desktop_capture

Implement the Windows screen capturer using the Magnification API.

2014-05-05 16:08:47 +00:00

interface

The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.

2014-05-13 08:50:02 +00:00

media_file

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

pacing

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

remote_bitrate_estimator

Re-enable the BitrateEstimatorTest cases for the Call API.

2014-05-14 09:15:19 +00:00

rtp_rtcp

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

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

utility

Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()

2014-04-29 17:11:07 +00:00

video_capture

Remove WEBRTC_TRACE use in video_capture/

2014-05-14 08:42:07 +00:00

video_coding

Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.

2014-05-07 17:09:44 +00:00

video_processing/main

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

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

video_render

video_render.gypi: clean up some libraries directives to be more specific.

2014-05-07 00:09:30 +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

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules_unittests.isolate

Support arbitrary input/output rates and downmixing in AudioProcessing.

2014-04-22 21:00:04 +00:00

modules.gyp

Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.

2014-05-07 17:09:44 +00:00

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00