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authorAnton Khirnov <anton@khirnov.net>2016-08-09 18:10:26 +0200
committerAnton Khirnov <anton@khirnov.net>2016-09-22 09:47:52 +0200
commite9ef6171396dc4106526aaa86b620c61ca3d1017 (patch)
treec244145877956b24bb560801709c6b9ba0893f97 /tests/checkasm
parent2eb97af66af90ca3978229da151f0b8b3a5d9370 (diff)
checkasm: add tests for audiodsp
Diffstat (limited to 'tests/checkasm')
-rw-r--r--tests/checkasm/Makefile1
-rw-r--r--tests/checkasm/audiodsp.c146
-rw-r--r--tests/checkasm/checkasm.c3
-rw-r--r--tests/checkasm/checkasm.h1
4 files changed, 151 insertions, 0 deletions
diff --git a/tests/checkasm/Makefile b/tests/checkasm/Makefile
index 7862633997..f66c8b934e 100644
--- a/tests/checkasm/Makefile
+++ b/tests/checkasm/Makefile
@@ -1,5 +1,6 @@
# libavcodec tests
# subsystems
+AVCODECOBJS-$(CONFIG_AUDIODSP) += audiodsp.o
AVCODECOBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
AVCODECOBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
AVCODECOBJS-$(CONFIG_FMTCONVERT) += fmtconvert.o
diff --git a/tests/checkasm/audiodsp.c b/tests/checkasm/audiodsp.c
new file mode 100644
index 0000000000..456b90bfec
--- /dev/null
+++ b/tests/checkasm/audiodsp.c
@@ -0,0 +1,146 @@
+/*
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * Libav 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with Libav; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <math.h>
+#include <string.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavcodec/audiodsp.h"
+
+#include "libavutil/common.h"
+#include "libavutil/intreadwrite.h"
+
+#include "checkasm.h"
+
+#define MAX_SIZE (32 * 128)
+
+#define randomize_float(buf, len) \
+ do { \
+ int i; \
+ for (i = 0; i < len; i++) { \
+ float f = (float)rnd() / (UINT_MAX >> 5) - 16.0f; \
+ buf[i] = f; \
+ } \
+ } while (0)
+
+#define randomize_int(buf, len, size, bits) \
+ do { \
+ int i; \
+ for (i = 0; i < len; i++) { \
+ uint ## size ## _t r = rnd() & ((1LL << bits) - 1); \
+ AV_WN ## size ## A(buf + i, -(1LL << (bits - 1)) + r); \
+ } \
+ } while (0)
+
+void checkasm_check_audiodsp(void)
+{
+ AudioDSPContext adsp;
+
+ ff_audiodsp_init(&adsp);
+
+ if (check_func(adsp.scalarproduct_int16, "audiodsp.scalarproduct_int16")) {
+ LOCAL_ALIGNED(32, int16_t, v1, [MAX_SIZE]);
+ LOCAL_ALIGNED(32, int16_t, v2, [MAX_SIZE]);
+ unsigned int len_bits_minus4, v1_bits, v2_bits, len;
+ int32_t res0, res1;
+
+ declare_func_emms(AV_CPU_FLAG_MMX, int32_t, const int16_t *v1, const int16_t *v2, int len);
+
+ // generate random 5-12bit vector length
+ len_bits_minus4 = rnd() % 8;
+ len = rnd() & ((1 << len_bits_minus4) - 1);
+ len = 16 * FFMAX(len, 1);
+
+ // generate the bit counts for each of the vectors such that the result
+ // fits into int32
+ v1_bits = 1 + rnd() % 15;
+ v2_bits = FFMIN(32 - (len_bits_minus4 + 4) - v1_bits - 1, 15);
+
+ randomize_int(v1, MAX_SIZE, 16, v1_bits + 1);
+ randomize_int(v2, MAX_SIZE, 16, v2_bits + 1);
+
+ res0 = call_ref(v1, v2, len);
+ res1 = call_new(v1, v2, len);
+ if (res0 != res1)
+ fail();
+ bench_new(v1, v2, MAX_SIZE);
+ }
+
+ if (check_func(adsp.vector_clip_int32, "audiodsp.vector_clip_int32")) {
+ LOCAL_ALIGNED(32, int32_t, src, [MAX_SIZE]);
+ LOCAL_ALIGNED(32, int32_t, dst0, [MAX_SIZE]);
+ LOCAL_ALIGNED(32, int32_t, dst1, [MAX_SIZE]);
+ int32_t val1, val2, min, max;
+ int len;
+
+ declare_func_emms(AV_CPU_FLAG_MMX, void, int32_t *dst, const int32_t *src,
+ int32_t min, int32_t max, unsigned int len);
+
+ val1 = ((int32_t)rnd());
+ val1 = FFSIGN(val1) * (val1 & ((1 << 24) - 1));
+ val2 = ((int32_t)rnd());
+ val2 = FFSIGN(val2) * (val2 & ((1 << 24) - 1));
+
+ min = FFMIN(val1, val2);
+ max = FFMAX(val1, val2);
+
+ randomize_int(src, MAX_SIZE, 32, 32);
+
+ len = rnd() % 128;
+ len = 32 * FFMAX(len, 1);
+
+ call_ref(dst0, src, min, max, len);
+ call_new(dst1, src, min, max, len);
+ if (memcmp(dst0, dst1, len * sizeof(*dst0)))
+ fail();
+ bench_new(dst1, src, min, max, MAX_SIZE);
+ }
+
+ if (check_func(adsp.vector_clipf, "audiodsp.vector_clipf")) {
+ LOCAL_ALIGNED(32, float, src, [MAX_SIZE]);
+ LOCAL_ALIGNED(32, float, dst0, [MAX_SIZE]);
+ LOCAL_ALIGNED(32, float, dst1, [MAX_SIZE]);
+ float val1, val2, min, max;
+ int i, len;
+
+ declare_func_emms(AV_CPU_FLAG_MMX, void, float *dst, const float *src,
+ float min, float max, unsigned int len);
+
+ val1 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
+ val2 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
+
+ min = FFMIN(val1, val2);
+ max = FFMAX(val1, val2);
+
+ randomize_float(src, MAX_SIZE);
+
+ len = rnd() % 128;
+ len = 16 * FFMAX(len, 1);
+
+ call_ref(dst0, src, min, max, len);
+ call_new(dst1, src, min, max, len);
+ for (i = 0; i < len; i++) {
+ if (!float_near_ulp_array(dst0, dst1, 3, len))
+ fail();
+ }
+ bench_new(dst1, src, min, max, MAX_SIZE);
+ }
+
+ report("audiodsp");
+}
diff --git a/tests/checkasm/checkasm.c b/tests/checkasm/checkasm.c
index 525284ad4b..c279ed17fc 100644
--- a/tests/checkasm/checkasm.c
+++ b/tests/checkasm/checkasm.c
@@ -64,6 +64,9 @@ static const struct {
const char *name;
void (*func)(void);
} tests[] = {
+#if CONFIG_AUDIODSP
+ { "audiodsp", checkasm_check_audiodsp },
+#endif
#if CONFIG_BLOCKDSP
{ "blockdsp", checkasm_check_blockdsp },
#endif
diff --git a/tests/checkasm/checkasm.h b/tests/checkasm/checkasm.h
index c1141aab74..169aa2a600 100644
--- a/tests/checkasm/checkasm.h
+++ b/tests/checkasm/checkasm.h
@@ -31,6 +31,7 @@
#include "libavutil/lfg.h"
#include "libavutil/timer.h"
+void checkasm_check_audiodsp(void);
void checkasm_check_blockdsp(void);
void checkasm_check_bswapdsp(void);
void checkasm_check_dcadsp(void);