Danil Chapovalov 61405bcb19 Fix infinite loop in rtp packet parsing ...

when rtp header extension is larger than 2^16 bytes

Bug: chromium:811613
Change-Id: I05b725d734dd628056d603b596d3523e827ddb54
Reviewed-on: https://webrtc-review.googlesource.com/52345
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22003}

2018-02-13 14:42:45 +00:00

..

audio_coding

Don't use gtest-parallel when running webrtc_perf_tests.

2018-02-12 13:10:04 +00:00

audio_device

Add callback when new audio data is ready

2018-02-09 19:31:49 +00:00

audio_mixer

Delete unused MediaFile module.

2018-01-29 11:18:18 +00:00

audio_processing

Removed hysteresis in the delay estimation offset

2018-02-13 14:29:23 +00:00

bitrate_controller

Moved congestion controller to goog_cc folder.

2018-02-07 09:50:48 +00:00

congestion_controller

Moved congestion controller to goog_cc folder.

2018-02-07 09:50:48 +00:00

desktop_capture

Revert "Add ScopedAllowBaseSyncPrimitives for DesktopConfigurationMonitor."

2018-01-31 08:32:19 +00:00

include

Cleanup of unused RTP structs and packetizer for stereo codec

2018-01-25 01:25:56 +00:00

pacing

Renamed constants in unit tests for paced sender.

2018-02-13 13:58:00 +00:00

remote_bitrate_estimator

Moved congestion controller to goog_cc folder.

2018-02-07 09:50:48 +00:00

rtp_rtcp

Fix infinite loop in rtp packet parsing

2018-02-13 14:42:45 +00:00

utility

Delete unused MediaFile module.

2018-01-29 11:18:18 +00:00

video_capture

Update gn files to support Mozilla build

2018-01-16 07:51:23 +00:00

video_coding

Remove unused fields from VideoCodecVP8.

2018-02-09 15:55:59 +00:00

video_processing

Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""

2018-01-10 15:55:04 +00:00

BUILD.gn

Adding simulcast/spatial layering support to VideoProcessor.

2018-02-01 13:28:46 +00:00

module_common_types_unittest.cc

Remove various IDs:

2017-09-28 14:37:11 +00:00

OWNERS

…