1 files changed, 94 insertions, 0 deletions
diff --git a/libavcodec/flacdsp.c b/libavcodec/flacdsp.c
new file mode 100644
index 0000000000..6c90e89d9b
--- /dev/null
+++ b/libavcodec/flacdsp.c
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/samplefmt.h"
+#include "flacdsp.h"
+
+#define SAMPLE_SIZE 16
+#include "flacdsp_template.c"
+
+#undef  SAMPLE_SIZE
+#define SAMPLE_SIZE 32
+#include "flacdsp_template.c"
+
+static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
+                          int pred_order, int qlevel, int len)
+{
+    int i, j;
+
+    for (i = pred_order; i < len - 1; i += 2) {
+        int c;
+        int d = decoded[i-pred_order];
+        int s0 = 0, s1 = 0;
+        for (j = pred_order-1; j > 0; j--) {
+            c = coeffs[j];
+            s0 += c*d;
+            d = decoded[i-j];
+            s1 += c*d;
+        }
+        c = coeffs[0];
+        s0 += c*d;
+        d = decoded[i] += s0 >> qlevel;
+        s1 += c*d;
+        decoded[i+1] += s1 >> qlevel;
+    }
+    if (i < len) {
+        int sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += coeffs[j] * decoded[i-j-1];
+        decoded[i] += sum >> qlevel;
+    }
+}
+
+static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
+                          int pred_order, int qlevel, int len)
+{
+    int i, j;
+
+    for (i = pred_order; i < len; i++) {
+        int64_t sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += (int64_t)coeffs[j] * decoded[i-j-1];
+        decoded[i] += sum >> qlevel;
+    }
+
+}
+
+av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
+{
+    switch (fmt) {
+    case AV_SAMPLE_FMT_S32:
+        c->decorrelate[0] = flac_decorrelate_indep_c_32;
+        c->decorrelate[1] = flac_decorrelate_ls_c_32;
+        c->decorrelate[2] = flac_decorrelate_rs_c_32;
+        c->decorrelate[3] = flac_decorrelate_ms_c_32;
+        c->lpc            = flac_lpc_32_c;
+        break;
+
+    case AV_SAMPLE_FMT_S16:
+        c->decorrelate[0] = flac_decorrelate_indep_c_16;
+        c->decorrelate[1] = flac_decorrelate_ls_c_16;
+        c->decorrelate[2] = flac_decorrelate_rs_c_16;
+        c->decorrelate[3] = flac_decorrelate_ms_c_16;
+        c->lpc            = flac_lpc_16_c;
+        break;
+    }
+}