[Ffmpeg-cvslog] r5507 - trunk/libavcodec/flac.c

[Justin Ruggles]

lu_zero wrote:
> -    for (i = pred_order; i < s->blocksize; i++)
> -    {
> -        sum = 0;
> -        for (j = 0; j < pred_order; j++)
> -            sum += coeffs[j] * s->decoded[channel][i-j-1];
> -        s->decoded[channel][i] += sum >> qlevel;
> +    if (s->bps > 16) {
> +        int64_t sum;
> +        for (i = pred_order; i < s->blocksize; i++)
> +        {
> +            sum = 0;
> +            for (j = 0; j < pred_order; j++)
> +                sum += (int64_t)coeffs[j] * s->decoded[channel][i-j-1];
> +            s->decoded[channel][i] += sum >> qlevel;
> +        }
> +    } else {
> +        int sum;
> +        for (i = pred_order; i < s->blocksize; i++)
> +        {
> +            sum = 0;
> +            for (j = 0; j < pred_order; j++)
> +                sum += coeffs[j] * s->decoded[channel][i-j-1];
> +            s->decoded[channel][i] += sum >> qlevel;
> +        }

Hi,

sorry...I do realize that I am speaking up a bit late, but I have a
question about this change.  I don't see the reason to have 2 versions
of this.  I know that having a 32-bit sum for 16-bit audio is not likely
to overflow, but it's not a guarantee, and with lossless audio it's
critical to be 100% sure.  My suggestion would be to always use the
64-bit version.  This would put the overflow limit out of the range of
theoretical possibility.

again...sorry for not bringing up this question until after the commit.

-Justin