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Diffstat (limited to 'chromium/third_party/ffmpeg/libavcodec/flacdsp_lpc_template.c')
-rw-r--r--chromium/third_party/ffmpeg/libavcodec/flacdsp_lpc_template.c18
1 files changed, 18 insertions, 0 deletions
diff --git a/chromium/third_party/ffmpeg/libavcodec/flacdsp_lpc_template.c b/chromium/third_party/ffmpeg/libavcodec/flacdsp_lpc_template.c
index 0c453aee8e2..acdac042b6e 100644
--- a/chromium/third_party/ffmpeg/libavcodec/flacdsp_lpc_template.c
+++ b/chromium/third_party/ffmpeg/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.
+ */