)]}'
{
  "commit": "ecde302f3f4e4f4149ad7eda697bb9309955e57d",
  "tree": "5d30341b4f16058c89f6b216b5212a95447a2f13",
  "parents": [
    "7f5a8b656cf69708359a8179d226dda8d2206205"
  ],
  "author": {
    "name": "Tommi",
    "email": "tommi@webrtc.org",
    "time": "Fri Mar 06 08:37:17 2026"
  },
  "committer": {
    "name": "WebRTC LUCI CQ",
    "email": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "time": "Fri Mar 06 13:08:34 2026"
  },
  "message": "Check maximum buffer size in ResamplerHelper::MaybeResample\n\nVerify that the target sample count does not exceed the maximum allowed\nsize for an AudioFrame. Previously, requesting a resample operation with\na high number of channels or a high sample rate could result in a target\ndata size that exceeded the internal limits of the AudioFrame class.\n\nThis change adds a validation check before starting the resampling\nprocess. If the calculated target size—based on the desired sample rate\nand channel count—surpasses kMaxDataSizeSamples, the operation now\nsafely aborts. In such cases, the error is logged, the audio frame is\nmuted to avoid undefined behavior, and the function returns false.\n\nBug: chromium:486349161\nFixes: chromium:486349161\nChange-Id: Ia0d8abcd390f90a590c07c0606f9b6c968f663e6\nReviewed-on: https://webrtc-review.googlesource.com/c/src/+/454000\nReviewed-by: Per Åhgren \u003cpeah@webrtc.org\u003e\nReviewed-by: Henrik Lundin \u003chenrik.lundin@webrtc.org\u003e\nCommit-Queue: Tomas Gunnarsson \u003ctommi@webrtc.org\u003e\nCr-Commit-Position: refs/heads/main@{#47076}\n",
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