adding tests folder

Author: roman10

tests/audiogen.c

@@ -0,0 +1,205 @@
+/*
+ * Generates a synthetic stereo sound
+ * NOTE: No floats are used to guarantee a bit exact output.
+ *
+ * Copyright (c) 2002 Fabrice Bellard
+ *
+ * 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 <stdlib.h>
+#include <stdio.h>
+
+#define MAX_CHANNELS 8
+
+static unsigned int myrnd(unsigned int *seed_ptr, int n)
+{
+    unsigned int seed, val;
+
+    seed = *seed_ptr;
+    seed = (seed * 314159) + 1;
+    if (n == 256) {
+        val = seed >> 24;
+    } else {
+        val = seed % n;
+    }
+    *seed_ptr = seed;
+    return val;
+}
+
+#define FRAC_BITS 16
+#define FRAC_ONE (1 << FRAC_BITS)
+
+#define COS_TABLE_BITS 7
+
+/* integer cosinus */
+static const unsigned short cos_table[(1 << COS_TABLE_BITS) + 2] = {
+ 0x8000, 0x7ffe, 0x7ff6, 0x7fea, 0x7fd9, 0x7fc2, 0x7fa7, 0x7f87,
+ 0x7f62, 0x7f38, 0x7f0a, 0x7ed6, 0x7e9d, 0x7e60, 0x7e1e, 0x7dd6,
+ 0x7d8a, 0x7d3a, 0x7ce4, 0x7c89, 0x7c2a, 0x7bc6, 0x7b5d, 0x7aef,
+ 0x7a7d, 0x7a06, 0x798a, 0x790a, 0x7885, 0x77fb, 0x776c, 0x76d9,
+ 0x7642, 0x75a6, 0x7505, 0x7460, 0x73b6, 0x7308, 0x7255, 0x719e,
+ 0x70e3, 0x7023, 0x6f5f, 0x6e97, 0x6dca, 0x6cf9, 0x6c24, 0x6b4b,
+ 0x6a6e, 0x698c, 0x68a7, 0x67bd, 0x66d0, 0x65de, 0x64e9, 0x63ef,
+ 0x62f2, 0x61f1, 0x60ec, 0x5fe4, 0x5ed7, 0x5dc8, 0x5cb4, 0x5b9d,
+ 0x5a82, 0x5964, 0x5843, 0x571e, 0x55f6, 0x54ca, 0x539b, 0x5269,
+ 0x5134, 0x4ffb, 0x4ec0, 0x4d81, 0x4c40, 0x4afb, 0x49b4, 0x486a,
+ 0x471d, 0x45cd, 0x447b, 0x4326, 0x41ce, 0x4074, 0x3f17, 0x3db8,
+ 0x3c57, 0x3af3, 0x398d, 0x3825, 0x36ba, 0x354e, 0x33df, 0x326e,
+ 0x30fc, 0x2f87, 0x2e11, 0x2c99, 0x2b1f, 0x29a4, 0x2827, 0x26a8,
+ 0x2528, 0x23a7, 0x2224, 0x209f, 0x1f1a, 0x1d93, 0x1c0c, 0x1a83,
+ 0x18f9, 0x176e, 0x15e2, 0x1455, 0x12c8, 0x113a, 0x0fab, 0x0e1c,
+ 0x0c8c, 0x0afb, 0x096b, 0x07d9, 0x0648, 0x04b6, 0x0324, 0x0192,
+ 0x0000, 0x0000,
+};
+
+#define CSHIFT (FRAC_BITS - COS_TABLE_BITS - 2)
+
+static int int_cos(int a)
+{
+    int neg, v, f;
+    const unsigned short *p;
+
+    a = a & (FRAC_ONE - 1); /* modulo 2 * pi */
+    if (a >= (FRAC_ONE / 2))
+        a = FRAC_ONE - a;
+    neg = 0;
+    if (a > (FRAC_ONE / 4)) {
+        neg = -1;
+        a = (FRAC_ONE / 2) - a;
+    }
+    p = cos_table + (a >> CSHIFT);
+    /* linear interpolation */
+    f = a & ((1 << CSHIFT) - 1);
+    v = p[0] + (((p[1] - p[0]) * f + (1 << (CSHIFT - 1))) >> CSHIFT);
+    v = (v ^ neg) - neg;
+    v = v << (FRAC_BITS - 15);
+    return v;
+}
+
+FILE *outfile;
+
+static void put_sample(int v)
+{
+    fputc(v & 0xff, outfile);
+    fputc((v >> 8) & 0xff, outfile);
+}
+
+int main(int argc, char **argv)
+{
+    int i, a, v, j, f, amp, ampa;
+    unsigned int seed = 1;
+    int tabf1[MAX_CHANNELS], tabf2[MAX_CHANNELS];
+    int taba[MAX_CHANNELS];
+    int sample_rate = 44100;
+    int nb_channels = 2;
+
+    if (argc < 2 || argc > 4) {
+        printf("usage: %s file [<sample rate> [<channels>]]\n"
+               "generate a test raw 16 bit audio stream\n"
+               "default: 44100 Hz stereo\n", argv[0]);
+        exit(1);
+    }
+
+    if (argc > 2) {
+        sample_rate = atoi(argv[2]);
+        if (sample_rate <= 0) {
+            fprintf(stderr, "invalid sample rate: %d\n", sample_rate);
+            return 1;
+        }
+    }
+
+    if (argc > 3) {
+        nb_channels = atoi(argv[3]);
+        if (nb_channels < 1 || nb_channels > MAX_CHANNELS) {
+            fprintf(stderr, "invalid number of channels: %d\n", nb_channels);
+            return 1;
+        }
+    }
+
+    outfile = fopen(argv[1], "wb");
+    if (!outfile) {
+        perror(argv[1]);
+        return 1;
+    }
+
+    /* 1 second of single freq sinus at 1000 Hz */
+    a = 0;
+    for(i=0;i<1 * sample_rate;i++) {
+        v = (int_cos(a) * 10000) >> FRAC_BITS;
+        for(j=0;j<nb_channels;j++)
+            put_sample(v);
+        a += (1000 * FRAC_ONE) / sample_rate;
+    }
+
+    /* 1 second of varing frequency between 100 and 10000 Hz */
+    a = 0;
+    for(i=0;i<1 * sample_rate;i++) {
+        v = (int_cos(a) * 10000) >> FRAC_BITS;
+        for(j=0;j<nb_channels;j++)
+            put_sample(v);
+        f = 100 + (((10000 - 100) * i) / sample_rate);
+        a += (f * FRAC_ONE) / sample_rate;
+    }
+
+    /* 0.5 second of low amplitude white noise */
+    for(i=0;i<sample_rate / 2;i++) {
+        v = myrnd(&seed, 20000) - 10000;
+        for(j=0;j<nb_channels;j++)
+            put_sample(v);
+    }
+
+    /* 0.5 second of high amplitude white noise */
+    for(i=0;i<sample_rate / 2;i++) {
+        v = myrnd(&seed, 65535) - 32768;
+        for(j=0;j<nb_channels;j++)
+            put_sample(v);
+    }
+
+    /* 1 second of unrelated ramps for each channel */
+    for(j=0;j<nb_channels;j++) {
+        taba[j] = 0;
+        tabf1[j] = 100 + myrnd(&seed, 5000);
+        tabf2[j] = 100 + myrnd(&seed, 5000);
+    }
+    for(i=0;i<1 * sample_rate;i++) {
+        for(j=0;j<nb_channels;j++) {
+            v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
+            put_sample(v);
+            f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
+            taba[j] += (f * FRAC_ONE) / sample_rate;
+        }
+    }
+
+    /* 2 seconds of 500 Hz with varying volume */
+    a = 0;
+    ampa = 0;
+    for(i=0;i<2 * sample_rate;i++) {
+        for(j=0;j<nb_channels;j++) {
+            amp = ((FRAC_ONE + int_cos(ampa)) * 5000) >> FRAC_BITS;
+            if (j & 1)
+                amp = 10000 - amp;
+            v = (int_cos(a) * amp) >> FRAC_BITS;
+            put_sample(v);
+            a += (500 * FRAC_ONE) / sample_rate;
+            ampa += (2 * FRAC_ONE) / sample_rate;
+        }
+    }
+
+    fclose(outfile);
+    return 0;
+}

tests/base64.c

@@ -0,0 +1,53 @@
+/*
+ * 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
+ */
+
+/*
+ * Based on libavutil/base64.c
+ */
+
+#include <stdio.h>
+
+int main(void)
+{
+    static const char b64[] =
+        "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
+    unsigned i_bits = 0;
+    int i_shift = 0;
+    int out_len = 0;
+    int in;
+
+#define putb64() do {                                   \
+        putchar(b64[(i_bits << 6 >> i_shift) & 0x3f]);  \
+        out_len++;                                      \
+        i_shift -= 6;                                   \
+    } while (0)
+
+    while ((in = getchar()) != EOF) {
+        i_bits = (i_bits << 8) + in;
+        i_shift += 8;
+        while (i_shift > 6)
+            putb64();
+    }
+    while (i_shift > 0)
+        putb64();
+    while (out_len++ & 3)
+        putchar('=');
+    putchar('\n');
+
+    return 0;
+}