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/*
* Copyright (C) 2024 Mark Hills <mark@xwax.org>
*
* This file is part of "xwax".
*
* "xwax" is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License, version 3 as
* published by the Free Software Foundation.
*
* "xwax" 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 this program; if not, see <https://www.gnu.org/licenses/>.
*
*/
#include <fcntl.h>
#include <poll.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "oss.h"
#define FRAME 32 /* maximum read size */
struct oss {
int fd;
struct pollfd *pe;
unsigned int rate;
};
static void clear(struct device *dv)
{
int r;
struct oss *oss = (struct oss*)dv->local;
r = close(oss->fd);
if (r == -1) {
perror("close");
abort();
}
free(dv->local);
}
/* Push audio into the device's buffer, for playback */
static int push(int fd, signed short *pcm, int samples)
{
int r, bytes;
bytes = samples * DEVICE_CHANNELS * sizeof(short);
r = write(fd, pcm, bytes);
if (r == -1) {
perror("write");
return -1;
}
if (r < bytes)
fprintf(stderr, "Device output overrun.\n");
return r / DEVICE_CHANNELS / sizeof(short);
}
/* Pull audio from the device, for recording */
static int pull(int fd, signed short *pcm, int samples)
{
int r, bytes;
bytes = samples * DEVICE_CHANNELS * sizeof(short);
r = read(fd, pcm, bytes);
if (r == -1) {
perror("read");
return -1;
}
if (r < bytes)
fprintf(stderr, "Device input underrun.\n");
return r / DEVICE_CHANNELS / sizeof(short);
}
static int handle(struct device *dv)
{
signed short pcm[FRAME * DEVICE_CHANNELS];
int samples;
struct oss *oss = (struct oss*)dv->local;
/* Check input buffer for recording */
if (oss->pe->revents & POLLIN) {
samples = pull(oss->fd, pcm, FRAME);
if (samples == -1)
return -1;
device_submit(dv, pcm, samples);
}
/* Check the output buffer for playback */
if (oss->pe->revents & POLLOUT) {
device_collect(dv, pcm, FRAME);
samples = push(oss->fd, pcm, FRAME);
if (samples == -1)
return -1;
}
return 0;
}
static ssize_t pollfds(struct device *dv, struct pollfd *pe, size_t z)
{
struct oss *oss = (struct oss*)dv->local;
if (z < 1)
return -1;
pe->fd = oss->fd;
pe->events = POLLIN | POLLOUT;
oss->pe = pe;
return 1;
}
static unsigned int sample_rate(struct device *dv)
{
struct oss *oss = (struct oss*)dv->local;
return oss->rate;
}
static struct device_ops oss_ops = {
.pollfds = pollfds,
.handle = handle,
.sample_rate = sample_rate,
.clear = clear
};
int oss_init(struct device *dv, const char *filename, unsigned int rate,
unsigned short buffers, unsigned short fragment)
{
int p, fd;
struct oss *oss;
fd = open(filename, O_RDWR, 0);
if (fd == -1) {
perror("open");
return -1;
}
/* Ideally would set independent settings for the record and
* playback buffers. Recording needs to buffer where necessary, as
* lost audio results in zero-crossings which corrupt the
* timecode. Playback buffer needs to be short to avoid high
* latency */
p = (buffers << 16) | fragment;
if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &p) == -1) {
perror("SNDCTL_DSP_SETFRAGMENT");
goto fail;
}
p = AFMT_S16_LE;
if (ioctl(fd, SNDCTL_DSP_SETFMT, &p) == -1) {
perror("SNDCTL_DSP_SETFMT");
goto fail;
}
p = DEVICE_CHANNELS;
if (ioctl(fd, SNDCTL_DSP_CHANNELS, &p) == -1) {
perror("SNDCTL_DSP_CHANNELS");
goto fail;
}
p = rate;
if (ioctl(fd, SNDCTL_DSP_SPEED, &p) == -1) {
perror("SNDCTL_DSP_SPEED");
goto fail;
}
p = fcntl(fd, F_GETFL);
if (p == -1) {
perror("F_GETFL");
goto fail;
}
p |= O_NONBLOCK;
if (fcntl(fd, F_SETFL, p) == -1) {
perror("fcntl");
return -1;
}
oss = malloc(sizeof *oss);
if (oss == NULL) {
perror("malloc");
goto fail;
}
oss->fd = fd;
oss->pe = NULL;
oss->rate = rate;
device_init(dv, &oss_ops);
dv->local = oss;
return 0;
fail:
close(fd);
return -1;
}
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