Michael Froman
c22b3b6e19
Upstream commit: https://webrtc.googlesource.com/src/+/eac26aa4a210b14e7556bddf9c18a14cc5711872 [146] Check maximum buffer size in ResamplerHelper::MaybeResample Verify that the target sample count does not exceed the maximum allowed size for an AudioFrame. Previously, requesting a resample operation with a high number of channels or a high sample rate could result in a target data size that exceeded the internal limits of the AudioFrame class. This change adds a validation check before starting the resampling process. If the calculated target size—based on the desired sample rate and channel count—surpasses kMaxDataSizeSamples, the operation now safely aborts. In such cases, the error is logged, the audio frame is muted to avoid undefined behavior, and the function returns false. (cherry picked from commit ecde302f3f4e4f4149ad7eda697bb9309955e57d) Bug: chromium:486349161 Fixes: chromium:486349161 Change-Id: Ia0d8abcd390f90a590c07c0606f9b6c968f663e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454000 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#47076} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/456340 Reviewed-by: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/branch-heads/7680@{#2} Cr-Branched-From: d1972add2a63b2a528a6471d447f82e0010b5215-refs/heads/main@{#46853}
9 lines
184 B
Python
9 lines
184 B
Python
include_rules = [
|
|
"+audio/utility",
|
|
"+call",
|
|
"+common_audio",
|
|
"+logging/rtc_event_log",
|
|
"+audio_coding/neteq/neteq_unittest.pb.h", # Different path.
|
|
"+system_wrappers",
|
|
]
|