Sindbad~EG File Manager

/*
 * SPDX-License-Identifier: BSD-2-Clause-FreeBSD
 *
 * Copyright (c) 2021 Goran Mekić
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
 * ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
 * SUCH DAMAGE.
 */

#include <sys/soundcard.h>
#include <errno.h>
#include <fcntl.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>


#ifndef SAMPLE_SIZE
#define SAMPLE_SIZE 16
#endif

/* Format can be unsigned, in which case replace S with U */
#if SAMPLE_SIZE == 32
typedef int32_t sample_t;
int	format = AFMT_S32_NE;		/* Signed 32bit native endian format */
#elif SAMPLE_SIZE == 16
typedef int16_t sample_t;
int	format = AFMT_S16_NE;		/* Signed 16bit native endian format */
#elif SAMPLE_SIZE == 8
typedef int8_t sample_t;
int	format = AFMT_S8_NE;		/* Signed 8bit native endian format */
#else
#error Unsupported sample format!
typedef int32_t sample_t;
int	format = AFMT_S32_NE;		/* Not a real value, just silencing
					 * compiler errors */
#endif



/*
 * Minimal configuration for OSS
 * For real world applications, this structure will probably contain many
 * more fields
 */
typedef struct config {
	char   *device;
	int	channels;
	int	fd;
	int	format;
	int	frag;
	int	sample_count;
	int	sample_rate;
	int	sample_size;
	int	chsamples;
	int	mmap;
	oss_audioinfo audio_info;
	audio_buf_info buffer_info;
} config_t;


/*
 * Error state is indicated by value=-1 in which case application exits
 * with error
 */
static inline void
check_error(const int value, const char *message)
{
	if (value == -1) {
		fprintf(stderr, "OSS error: %s %s\n", message, strerror(errno));
		exit(1);
	}
}



/* Calculate frag by giving it minimal size of buffer */
static inline int
size2frag(int x)
{
	int frag = 0;

	while ((1 << frag) < x) {
		++frag;
	}
	return frag;
}


/*
 * Split input buffer into channels. Input buffer is in interleaved format
 * which means if we have 2 channels (L and R), this is what the buffer of
 * 8 samples would contain: L,R,L,R,L,R,L,R. The result are two channels
 * containing: L,L,L,L and R,R,R,R.
 */
void
oss_split(config_t *config, sample_t *input, sample_t *output)
{
	int channel;
	int index;

	for (int i = 0; i < config->sample_count; ++i) {
		channel = i % config->channels;
		index = i / config->channels;
		output[channel * index] = input[i];
	}
}


/*
 * Convert channels into interleaved format and place it in output
 * buffer
 */
void
oss_merge(config_t *config, sample_t *input, sample_t *output)
{
	for (int channel = 0; channel < config->channels; ++channel) {
		for (int index = 0; index < config->chsamples; ++index) {
			output[index * config->channels + channel] = input[channel * index];
		}
	}
}

void
oss_init(config_t *config)
{
	int error;
	int tmp;

	/* Open the device for read and write */
	config->fd = open(config->device, O_RDWR);
	check_error(config->fd, "open");

	/* Get device information */
	config->audio_info.dev = -1;
	error = ioctl(config->fd, SNDCTL_ENGINEINFO, &(config->audio_info));
	check_error(error, "SNDCTL_ENGINEINFO");
	printf("min_channels: %d\n", config->audio_info.min_channels);
	printf("max_channels: %d\n", config->audio_info.max_channels);
	printf("latency: %d\n", config->audio_info.latency);
	printf("handle: %s\n", config->audio_info.handle);
	if (config->audio_info.min_rate > config->sample_rate || config->sample_rate > config->audio_info.max_rate) {
		fprintf(stderr, "%s doesn't support chosen ", config->device);
		fprintf(stderr, "samplerate of %dHz!\n", config->sample_rate);
		exit(1);
	}
	if (config->channels < 1) {
		config->channels = config->audio_info.max_channels;
	}

	/*
         * If device is going to be used in mmap mode, disable all format
         * conversions. Official OSS documentation states error code should not be
         * checked. http://manuals.opensound.com/developer/mmap_test.c.html#LOC10
         */
	if (config->mmap) {
		tmp = 0;
		ioctl(config->fd, SNDCTL_DSP_COOKEDMODE, &tmp);
	}

	/*
         * Set number of channels. If number of channels is chosen to the value
         * near the one wanted, save it in config
         */
	tmp = config->channels;
	error = ioctl(config->fd, SNDCTL_DSP_CHANNELS, &tmp);
	check_error(error, "SNDCTL_DSP_CHANNELS");
	if (tmp != config->channels) {	/* or check if tmp is close enough? */
		fprintf(stderr, "%s doesn't support chosen ", config->device);
		fprintf(stderr, "channel count of %d", config->channels);
		fprintf(stderr, ", set to %d!\n", tmp);
	}
	config->channels = tmp;

	/* Set format, or bit size: 8, 16, 24 or 32 bit sample */
	tmp = config->format;
	error = ioctl(config->fd, SNDCTL_DSP_SETFMT, &tmp);
	check_error(error, "SNDCTL_DSP_SETFMT");
	if (tmp != config->format) {
		fprintf(stderr, "%s doesn't support chosen sample format!\n", config->device);
		exit(1);
	}

	/* Most common values for samplerate (in kHz): 44.1, 48, 88.2, 96 */
	tmp = config->sample_rate;
	error = ioctl(config->fd, SNDCTL_DSP_SPEED, &tmp);
	check_error(error, "SNDCTL_DSP_SPEED");

	/* Get and check device capabilities */
	error = ioctl(config->fd, SNDCTL_DSP_GETCAPS, &(config->audio_info.caps));
	check_error(error, "SNDCTL_DSP_GETCAPS");
	if (!(config->audio_info.caps & PCM_CAP_DUPLEX)) {
		fprintf(stderr, "Device doesn't support full duplex!\n");
		exit(1);
	}
	if (config->mmap) {
		if (!(config->audio_info.caps & PCM_CAP_TRIGGER)) {
			fprintf(stderr, "Device doesn't support triggering!\n");
			exit(1);
		}
		if (!(config->audio_info.caps & PCM_CAP_MMAP)) {
			fprintf(stderr, "Device doesn't support mmap mode!\n");
			exit(1);
		}
	}

	/*
         * If desired frag is smaller than minimum, based on number of channels
         * and format (size in bits: 8, 16, 24, 32), set that as frag. Buffer size
         * is 2^frag, but the real size of the buffer will be read when the
         * configuration of the device is successfull
         */
	int min_frag = size2frag(config->sample_size * config->channels);

	if (config->frag < min_frag) {
		config->frag = min_frag;
	}

	/*
         * Allocate buffer in fragments. Total buffer will be split in number
         * of fragments (2 by default)
         */
	if (config->buffer_info.fragments < 0) {
		config->buffer_info.fragments = 2;
	}
	tmp = ((config->buffer_info.fragments) << 16) | config->frag;
	error = ioctl(config->fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
	check_error(error, "SNDCTL_DSP_SETFRAGMENT");

	/* When all is set and ready to go, get the size of buffer */
	error = ioctl(config->fd, SNDCTL_DSP_GETOSPACE, &(config->buffer_info));
	check_error(error, "SNDCTL_DSP_GETOSPACE");
	if (config->buffer_info.bytes < 1) {
		fprintf(
		    stderr,
		    "OSS buffer error: buffer size can not be %d\n",
		    config->buffer_info.bytes
		    );
		exit(1);
	}
	config->sample_count = config->buffer_info.bytes / config->sample_size;
	config->chsamples = config->sample_count / config->channels;
}