stefan@webrtc.org 946ad76f7e Switched lists of packets to lists of packet pointers. Allows Packet polymorphism. ...

This allows for different packet types in a follow-up CL, so that feedback can be passed through the network instead being fed directly into senders. It also made the whole simulator faster.

BUG=4173
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39679004

Cr-Commit-Position: refs/heads/master@{#8227}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8227 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-02 14:51:45 +00:00

..

audio_coding

Change name for local CriticalSectionScoped variable

2015-02-02 10:36:39 +00:00

audio_conference_mixer

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

audio_device

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

audio_processing

Add arbitrary microphone geometry input to audioproc_f test utility.

2015-01-29 22:40:13 +00:00

bitrate_controller

Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.

2015-01-19 15:44:47 +00:00

desktop_capture

GN: Prepare to remove webrtc_base target

2015-01-21 20:22:33 +00:00

interface

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

media_file

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

pacing

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

remote_bitrate_estimator

Switched lists of packets to lists of packet pointers. Allows Packet polymorphism.

2015-02-02 14:51:45 +00:00

rtp_rtcp

Fixed potential crash if rtp packet history is completely full.

2015-02-02 13:08:14 +00:00

utility

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

video_capture

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

video_coding

Using << on an int8_t or uint8_t will output a character rather than a number.

2015-01-30 19:54:19 +00:00

video_processing

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

video_render

Remove ChangeUniqueID.

2015-01-29 12:14:13 +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

Removes unused test files by audio_processing/transient

2014-12-15 16:13:05 +00:00

modules.gyp

Re-land "Remove <(webrtc_root) from source file entries."

2015-01-29 14:30:41 +00:00

OWNERS

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

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