Stefan Holmer a51e8f490c Fix some simulation issues. ...

Don't default to an infinite queue.
Make sure the computation of missing packets is correct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49059004

Cr-Commit-Position: refs/heads/master@{#9028}

2015-04-17 13:48:58 +00:00

..

audio_coding

Let rtp_analyze parse absolute sender time

2015-04-17 09:46:56 +00:00

audio_conference_mixer

Remove henrike@ from OWNERS

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

audio_device

Removes use of AudioManager.setSpeakerphoneOn in audio manager

2015-04-14 07:19:49 +00:00

audio_processing

Fix unknown option '-msse2' warning

2015-04-16 06:57:12 +00:00

bitrate_controller

Moving the pacer and the pacer thread to ChannelGroup.

2015-03-26 10:11:22 +00:00

desktop_capture

Fix unknown option '-msse2' warning

2015-04-16 06:57:12 +00:00

interface

Fix clang style warnings in webrtc/modules/audio_coding

2015-04-07 14:13:10 +00:00

media_file

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

pacing

Moving the pacer and the pacer thread to ChannelGroup.

2015-03-26 10:11:22 +00:00

remote_bitrate_estimator

Fix some simulation issues.

2015-04-17 13:48:58 +00:00

rtp_rtcp

Remove rtc::Buffer::length(), since no one uses it anymore

2015-04-16 19:48:52 +00:00

utility

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +00:00

video_capture

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +00:00

video_coding

Modified the simulcast encoder adapter to correctly handle encoded frames from sub encoders even if the encoder is unable to (temporarily or permanently) produce frames of the exactly matching resolution. This is done by using a different EncodedImageCallback for each encoder, which remembers which VideoEncoder it is registered to and forwards that on to SimulcastEncoderAdapter::Encoded.

2015-04-15 16:24:16 +00:00

video_processing

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +00:00

video_render

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +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

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

Add a BWE fairness test.

2015-04-16 18:20:26 +00:00

OWNERS

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

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