[webkit-reviews] review granted: [Bug 195436] WebRTC: don't use stack buffer : [Attachment 363958] Patch

[bugzilla-daemon at webkit.org]

youenn fablet <youennf at gmail.com> has granted JF Bastien
<jfbastien at apple.com>'s request for review:
Bug 195436: WebRTC: don't use stack buffer
https://bugs.webkit.org/show_bug.cgi?id=195436

Attachment 363958: Patch

https://bugs.webkit.org/attachment.cgi?id=363958&action=review




--- Comment #11 from youenn fablet <youennf at gmail.com> ---
Comment on attachment 363958
  --> https://bugs.webkit.org/attachment.cgi?id=363958
Patch

View in context: https://bugs.webkit.org/attachment.cgi?id=363958&action=review

> Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp:99
> +	   m_data.reset(std::make_unique<char[]>(bytesPerSample * channels *
samplesPerFrame));

Would use m_data = std::make_unique.

If we go down that road, we could also allocate m_data in
StartPlayoutOnAudioThread and delete it when exiting of
StartPlayoutOnAudioThread.
That would remove the if check and deallocate it when no longer needed.