Slightly improve AudioBufferSourceNode resampling

[Chris Dumez]

Slightly improve AudioBufferSourceNode resampling. Use simple linear extrapolation to resample the data when we reach the end of the buffer. Previously, the last sample would just be repeated enough times.

[Chris Dumez]

Created attachment 409423 [details]
Patch

[Chris Dumez]

Created attachment 409424 [details]
Patch

[Eric Carlson]

Comment on attachment 409424 [details]
Patch

View in context: https://bugs.webkit.org/attachment.cgi?id=409424&action=review

> Source/WebCore/Modules/webaudio/AudioBufferSourceNode.cpp:416
> +                float sample;
> +                if (readIndex == readIndex2 && readIndex >= 1) {
> +                    // We're at the end of the buffer, so just linearly extrapolate from the last two samples.
> +                    float sample1 = source[readIndex - 1];
> +                    float sample2 = source[readIndex];
> +                    sample = sample2 + (sample2 - sample1) * interpolationFactor;
> +                } else {
> +                    float sample1 = source[readIndex];
> +                    float sample2 = source[readIndex2];
> +                    sample = sample1 + interpolationFactor * (sample2 - sample1);
> +                }
>  
> -                destination[writeIndex] = narrowPrecisionToFloat(sample);
> +                destination[writeIndex] = sample;

It looks like this is an identical copy of the code above. Can it be moved to a shared function?

[Chris Dumez]

Created attachment 409428 [details]
Patch

[Chris Dumez]

Created attachment 409432 [details]
Patch

[EWS]

Committed r267453: <https://trac.webkit.org/changeset/267453>

All reviewed patches have been landed. Closing bug and clearing flags on attachment 409432 [details].

[Radar WebKit Bug Importer]

<rdar://problem/69406153>