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webrtc/modules
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Alessio Bazzica ba68aabb06 Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 
This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.

This change has been tested on mobile platforms.

Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
2017-10-23 14:25:37 +00:00
..
audio_coding NetEq: Fix an UBSan error 2017-10-23 11:56:47 +00:00
audio_device Removes StaticAccessedFromInstance warning in Android audio track. 2017-10-20 11:02:03 +00:00
audio_mixer Add explicit includes of refcountedobject.h where it is used. 2017-10-06 13:00:14 +00:00
audio_processing Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process 2017-10-23 14:25:37 +00:00
bitrate_controller Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead 2017-10-03 15:26:56 +00:00
congestion_controller Add field trials to configure the backoff factor and the trendline window of the BWE. 2017-10-06 07:10:04 +00:00
desktop_capture New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
include Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." 2017-10-04 11:31:18 +00:00
media_file Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
pacing New PacketQueue2 behind WebRTC-RoundRobinPacing field trial. 2017-10-23 11:39:57 +00:00
remote_bitrate_estimator Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
rtp_rtcp Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
utility Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
video_capture Stop using ANDROID macro in favour of WEBRTC_ANDROID. 2017-10-16 11:37:08 +00:00
video_coding Revert "Add fine grained dropped video frames counters on sending side" 2017-10-21 09:23:54 +00:00
video_processing Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Remove voe::OutputMixer and AudioConferenceMixer. 2017-09-22 13:48:10 +00:00
module_common_types_unittest.cc Remove various IDs: 2017-09-28 14:37:11 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00