content/renderer/media/webrtc_audio_renderer.cc in Google Chrome before 24.0.1312.56 on Mac OS X does not use an appropriate buffer size for the 96 kHz sampling rate, which allows remote attackers to cause a denial of service (memory corruption and application crash) or possibly have unspecified other impact via a web site that provides WebRTC audio.
https://codereview.chromium.org/11773017
https://code.google.com/p/chromium/issues/detail?id=166523
http://src.chromium.org/viewvc/chrome?view=rev&revision=175323
http://googlechromereleases.blogspot.com/2013/01/stable-channel-update_22.html