webrtc/modules/audio_processing/agc
Sam Zackrisson 71729eb0a8 Fix fuzzer-found flow-over in AGC1
This CL changes a constant from an approximately correct limit
of 2^25.5.

The new limit is the largest x such that z = 10 satisfies:
((x >> z) + 1)^2 <= 2^31 - 1.
If gains[k + 1] > x, then z >= 11 and needs to be computed.

Bug: chromium:860638
Change-Id: If17f257dacd94806e59e4f32b345a5fb15b4e32b
Reviewed-on: https://webrtc-review.googlesource.com/87583
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23908}
2018-07-10 14:02:49 +00:00
..
legacy Fix fuzzer-found flow-over in AGC1 2018-07-10 14:02:49 +00:00
agc.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
agc.h Streamline error handling and logging in the audio processing module 2018-02-15 15:06:26 +00:00
agc_manager_direct.cc Allow AGC2 level estimation in AgcManagerDirect. 2018-07-06 14:18:18 +00:00
agc_manager_direct.h Allow AGC2 level estimation in AgcManagerDirect. 2018-07-06 14:18:18 +00:00
agc_manager_direct_unittest.cc Streamline error handling and logging in the audio processing module 2018-02-15 15:06:26 +00:00
BUILD.gn Allow AGC2 level estimation in AgcManagerDirect. 2018-07-06 14:18:18 +00:00
gain_map_internal.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
loudness_histogram.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
loudness_histogram.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
loudness_histogram_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_agc.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
utility.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
utility.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00