From 91126dc481a48400dd00cc45e98fb25520344874 Mon Sep 17 00:00:00 2001 From: James Darnley Date: Fri, 14 Feb 2014 12:40:10 +0100 Subject: [PATCH] flacdsp_lpc_template: add comment to explain the CONFIG_SMALL code I found the optimisation of 2 samples per iteration obscured the underlying algorithm. I had to write it out on paper and translate into a mathematical sum to see that the two samples are unconnected. I hope that if anyone else is struggling to understand the code that this will be useful. Reviewed-by: Christophe Gisquet Signed-off-by: Michael Niedermayer --- libavcodec/flacdsp_lpc_template.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/libavcodec/flacdsp_lpc_template.c b/libavcodec/flacdsp_lpc_template.c index 0c453aee8e..acdac042b6 100644 --- a/libavcodec/flacdsp_lpc_template.c +++ b/libavcodec/flacdsp_lpc_template.c @@ -139,3 +139,21 @@ static void FUNC(flac_lpc_encode_c)(int32_t *res, const int32_t *smp, int len, } #endif } + +/* Comment for clarity/de-obfuscation. + * + * for (int i = order; i < len; i++) { + * int32_t p = 0; + * for (int j = 0; j < order; j++) { + * int c = coefs[j]; + * int s = smp[(i-1)-j]; + * p += c*s; + * } + * res[i] = smp[i] - (p >> shift); + * } + * + * The CONFIG_SMALL code above simplifies to this, in the case of SAMPLE_SIZE + * not being equal to 32 (at the present time that means for 16-bit audio). The + * code above does 2 samples per iteration. Commit bfdd5bc ( made all the way + * back in 2007) says that way is faster. + */