Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 130 insertions, 0 deletions

diff --git a/media/ffvpx/libavcodec/flacdsp.c b/media/ffvpx/libavcodec/flacdsp.c
new file mode 100644
index 0000000000..bc9a5dbed9
--- /dev/null
+++ b/media/ffvpx/libavcodec/flacdsp.c
@@ -0,0 +1,130 @@
+/*
+ * 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"
+#include "config.h"
+
+#define SAMPLE_SIZE 16
+#define PLANAR 0
+#include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
+
+#undef  PLANAR
+#define PLANAR 1
+#include "flacdsp_template.c"
+
+#undef  SAMPLE_SIZE
+#undef  PLANAR
+#define SAMPLE_SIZE 32
+#define PLANAR 0
+#include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
+
+#undef  PLANAR
+#define PLANAR 1
+#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, decoded += 2) {
+        SUINT c = coeffs[0];
+        SUINT d = decoded[0];
+        int s0 = 0, s1 = 0;
+        for (j = 1; j < pred_order; j++) {
+            s0 += c*d;
+            d = decoded[j];
+            s1 += c*d;
+            c = coeffs[j];
+        }
+        s0 += c*d;
+        d = decoded[j] += (SUINT)(s0 >> qlevel);
+        s1 += c*d;
+        decoded[j + 1] += (SUINT)(s1 >> qlevel);
+    }
+    if (i < len) {
+        int sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += coeffs[j] * (SUINT)decoded[j];
+        decoded[j] = decoded[j] + (unsigned)(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++, decoded++) {
+        int64_t sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += (int64_t)coeffs[j] * decoded[j];
+        decoded[j] += sum >> qlevel;
+    }
+
+}
+
+av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels,
+                             int bps)
+{
+    c->lpc16        = flac_lpc_16_c;
+    c->lpc32        = flac_lpc_32_c;
+    c->lpc16_encode = flac_lpc_encode_c_16;
+    c->lpc32_encode = flac_lpc_encode_c_32;
+
+    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;
+        break;
+
+    case AV_SAMPLE_FMT_S32P:
+        c->decorrelate[0] = flac_decorrelate_indep_c_32p;
+        c->decorrelate[1] = flac_decorrelate_ls_c_32p;
+        c->decorrelate[2] = flac_decorrelate_rs_c_32p;
+        c->decorrelate[3] = flac_decorrelate_ms_c_32p;
+        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;
+        break;
+
+    case AV_SAMPLE_FMT_S16P:
+        c->decorrelate[0] = flac_decorrelate_indep_c_16p;
+        c->decorrelate[1] = flac_decorrelate_ls_c_16p;
+        c->decorrelate[2] = flac_decorrelate_rs_c_16p;
+        c->decorrelate[3] = flac_decorrelate_ms_c_16p;
+        break;
+    }
+
+    if (ARCH_ARM)
+        ff_flacdsp_init_arm(c, fmt, channels, bps);
+    if (ARCH_X86)
+        ff_flacdsp_init_x86(c, fmt, channels, bps);
+}