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-rw-r--r--libavcodec/flacdsp_lpc_template.c26
1 files changed, 22 insertions, 4 deletions
diff --git a/libavcodec/flacdsp_lpc_template.c b/libavcodec/flacdsp_lpc_template.c
index 269e64b..5d532e0 100644
--- a/libavcodec/flacdsp_lpc_template.c
+++ b/libavcodec/flacdsp_lpc_template.c
@@ -1,18 +1,18 @@
/*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -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.
+ */
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