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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/ffvpx/libavcodec/flacdsp.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/ffvpx/libavcodec/flacdsp.c')
-rw-r--r--media/ffvpx/libavcodec/flacdsp.c130
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 000000000..30b66484e
--- /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) {
+ int c = coeffs[0];
+ int 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] += s0 >> qlevel;
+ s1 += c*d;
+ decoded[j + 1] += s1 >> qlevel;
+ }
+ if (i < len) {
+ int sum = 0;
+ for (j = 0; j < pred_order; j++)
+ sum += coeffs[j] * decoded[j];
+ decoded[j] += 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);
+}