tests/audiogen.c - nest-android-app/ffmpeg - Git at Google

 /*
  * Generate a synthetic stereo sound.
  * NOTE: No floats are used to guarantee bitexact 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 <stdint.h>
 #include <stdio.h>
 #include <string.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 cosine */
 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 put16(int16_t v)
 {
     fputc( v       & 0xff, outfile);
     fputc((v >> 8) & 0xff, outfile);
 }

 static void put32(uint32_t v)
 {
     fputc( v        & 0xff, outfile);
     fputc((v >>  8) & 0xff, outfile);
     fputc((v >> 16) & 0xff, outfile);
     fputc((v >> 24) & 0xff, outfile);
 }

 #define HEADER_SIZE      46
 #define FMT_SIZE         18
 #define SAMPLE_SIZE       2
 #define WFORMAT_PCM  0x0001

 static void put_wav_header(int sample_rate, int channels, int nb_samples)
 {
     int block_align = SAMPLE_SIZE * channels;
     int data_size   = block_align * nb_samples;

     fputs("RIFF", outfile);
     put32(HEADER_SIZE + data_size);
     fputs("WAVEfmt ", outfile);
     put32(FMT_SIZE);
     put16(WFORMAT_PCM);
     put16(channels);
     put32(sample_rate);
     put32(block_align * sample_rate);
     put16(block_align);
     put16(SAMPLE_SIZE * 8);
     put16(0);
     fputs("data", outfile);
     put32(data_size);
 }

 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;
     char *ext;

     if (argc < 2 || argc > 5) {
         printf("usage: %s file [<sample rate> [<channels>] [<random seed>]]\n"
                "generate a test raw 16 bit audio stream\n"
                "If the file extension is .wav a WAVE header will be added.\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;
         }
     }

     if (argc > 4)
         seed = atoi(argv[4]);

     outfile = fopen(argv[1], "wb");
     if (!outfile) {
         perror(argv[1]);
         return 1;
     }

     if ((ext = strrchr(argv[1], '.')) && !strcmp(ext, ".wav"))
         put_wav_header(sample_rate, nb_channels, 6 * sample_rate);

     /* 1 second of single freq sine 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++)
             put16(v);
         a += (1000 * FRAC_ONE) / sample_rate;
     }

     /* 1 second of varying 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++)
             put16(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++)
             put16(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++)
             put16(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;
             put16(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;
             put16(v);
             a    += (500 * FRAC_ONE) / sample_rate;
             ampa += (2 * FRAC_ONE) / sample_rate;
         }
     }

     fclose(outfile);
     return 0;
 }
	/*
	* Generate a synthetic stereo sound.
	* NOTE: No floats are used to guarantee bitexact 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 <stdint.h>
	#include <stdio.h>
	#include <string.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 cosine */
	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 put16(int16_t v)
	{
	fputc( v & 0xff, outfile);
	fputc((v >> 8) & 0xff, outfile);
	}

	static void put32(uint32_t v)
	{
	fputc( v & 0xff, outfile);
	fputc((v >> 8) & 0xff, outfile);
	fputc((v >> 16) & 0xff, outfile);
	fputc((v >> 24) & 0xff, outfile);
	}

	#define HEADER_SIZE 46
	#define FMT_SIZE 18
	#define SAMPLE_SIZE 2
	#define WFORMAT_PCM 0x0001

	static void put_wav_header(int sample_rate, int channels, int nb_samples)
	{
	int block_align = SAMPLE_SIZE * channels;
	int data_size = block_align * nb_samples;

	fputs("RIFF", outfile);
	put32(HEADER_SIZE + data_size);
	fputs("WAVEfmt ", outfile);
	put32(FMT_SIZE);
	put16(WFORMAT_PCM);
	put16(channels);
	put32(sample_rate);
	put32(block_align * sample_rate);
	put16(block_align);
	put16(SAMPLE_SIZE * 8);
	put16(0);
	fputs("data", outfile);
	put32(data_size);
	}

	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;
	char *ext;

	if (argc < 2 \|\| argc > 5) {
	printf("usage: %s file [<sample rate> [<channels>] [<random seed>]]\n"
	"generate a test raw 16 bit audio stream\n"
	"If the file extension is .wav a WAVE header will be added.\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;
	}
	}

	if (argc > 4)
	seed = atoi(argv[4]);

	outfile = fopen(argv[1], "wb");
	if (!outfile) {
	perror(argv[1]);
	return 1;
	}

	if ((ext = strrchr(argv[1], '.')) && !strcmp(ext, ".wav"))
	put_wav_header(sample_rate, nb_channels, 6 * sample_rate);

	/* 1 second of single freq sine 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++)
	put16(v);
	a += (1000 * FRAC_ONE) / sample_rate;
	}

	/* 1 second of varying 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++)
	put16(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++)
	put16(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++)
	put16(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;
	put16(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;
	put16(v);
	a += (500 * FRAC_ONE) / sample_rate;
	ampa += (2 * FRAC_ONE) / sample_rate;
	}
	}

	fclose(outfile);
	return 0;
	}