Alex Loiko 9f789a4500 LowCutFilter::BiqueadFilter::Process: Fix UBSan fuzzer bug ...

(left shift of negative value)


Bug: chromium:735593
Change-Id: I9f1165370d850456480fbb22ce2434bf933a420b
Reviewed-on: https://chromium-review.googlesource.com/552136
Commit-Queue: Alex Loiko <aleloi@google.com>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18812}

2017-06-28 14:55:20 +00:00

..

audio_coding

Expose ILBC codec in webrtc/api/audio_codecs/

2017-06-28 09:05:04 +00:00

audio_conference_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_device

Improved thread checking scheme for iOS.

2017-06-28 14:20:30 +00:00

audio_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_processing

LowCutFilter::BiqueadFilter::Process: Fix UBSan fuzzer bug

2017-06-28 14:55:20 +00:00

bitrate_controller

Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

2017-06-16 14:47:00 +00:00

congestion_controller

Only use 95% of the link capacity if the true link capacity is found by probing.

2017-06-27 14:50:31 +00:00

desktop_capture

Ensure Dxgi duplicator works correctly in session 0

2017-06-28 05:04:21 +00:00

include

Implement timing frames.

2017-06-19 14:18:55 +00:00

media_file

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pacing

Use RaceChecker instead of ThreadChecker in a few places.

2017-06-16 12:02:05 +00:00

remote_bitrate_estimator

Remove unneeded enum forward declaration

2017-06-24 20:04:29 +00:00

rtp_rtcp

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

utility

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video_capture

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

video_coding

Workaround for hardware encoders crashing timing frames processing

2017-06-28 10:53:19 +00:00

video_processing

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

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