webrtc/modules/audio_coding/codecs/isac
Karl Wiberg 338f58d95c iSAC decoder: Don't read past the end of the buffer of encoded bytes
Bug: chromium:825524
Change-Id: Iff40a9fd62a34474af71b51dd3831a16412fbf3b
Reviewed-on: https://webrtc-review.googlesource.com/66361
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22748}
2018-04-05 13:22:53 +00:00
..
fix Move asm_defines.h to rtc_base/system/ 2018-03-22 03:12:13 +00:00
main iSAC decoder: Don't read past the end of the buffer of encoded bytes 2018-04-05 13:22:53 +00:00
audio_decoder_isac_t.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_decoder_isac_t_impl.h Add missing iSAC headers. 2018-01-02 13:01:11 +00:00
audio_encoder_isac_t.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder_isac_t_impl.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
bandwidth_info.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
empty.cc Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
locked_bandwidth_info.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
locked_bandwidth_info.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00