Stefan Holmer 11324b9561 Wait for a longer time (5 seconds) before establishing the first bandwidth estimate. ...

This reduces the risk of getting a small initial estimate when doing combined a/v BWE, and the audio stream is received earlier than the video stream.

In addition a check is added to make sure a probe can't reduce the BWE.

R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1219303002 .

Cr-Commit-Position: refs/heads/master@{#9560}

2015-07-09 15:28:07 +00:00

..

audio_coding

Targets should not depend on protobuf when enable_protobuf=0.

2015-07-03 15:53:22 +00:00

audio_conference_mixer

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

audio_device

Enabling AudioDeviceTest.StartStopPlayout on Nexus 9

2015-06-24 11:27:35 +00:00

audio_processing

Remove empty-string comparisons.

2015-07-09 14:48:27 +00:00

bitrate_controller

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

desktop_capture

Removed some unused variables in Windows code.

2015-07-09 07:54:02 +00:00

interface

Reformat existing code. There should be no functional effects.

2015-06-11 21:31:48 +00:00

media_file

Ensures that modules_unittests runs on iOS

2015-06-10 11:24:58 +00:00

pacing

Make the BWE threshold adaptive.

2015-07-07 11:20:40 +00:00

remote_bitrate_estimator

Wait for a longer time (5 seconds) before establishing the first bandwidth estimate.

2015-07-09 15:28:07 +00:00

rtp_rtcp

Remove ResetStatistics from RTP feedback.

2015-07-07 15:32:56 +00:00

utility

Use strcmp instead of == operator for c.name and name to find appropriate classes for WebRtcAudio*.java

2015-07-07 08:10:21 +00:00

video_capture

Use one scoped_refptr.

2015-06-04 13:18:28 +00:00

video_coding

Remove empty-string comparisons.

2015-07-09 14:48:27 +00:00

video_processing

Ensures that modules_unittests runs on iOS

2015-06-10 11:24:58 +00:00

video_render

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

module_common_types_unittest.cc

Testing and improving NADA algorithm.

2015-05-30 15:49:32 +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

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

modules_unittests.isolate

Adds C++/JNI/Java unit test for audio device module on Android.

2015-03-09 12:40:43 +00:00

modules.gyp

iSAC: Functions for importing and exporting bandwidth est. info

2015-07-03 02:04:46 +00:00

OWNERS

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

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