--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2007 Richard Kettlewell
+ *
+ * This program 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.
+ *
+ * This program 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, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ * USA
+ */
+/** @file clients/playrtp-oss.c
+ * @brief RTP player - OSS support
+ */
+
+#include <config.h>
+
+#if HAVE_SYS_SOUNDCARD_H
+#include "types.h"
+
+#include <poll.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+#include <assert.h>
+#include <pthread.h>
+#include <string.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+
+#include "mem.h"
+#include "log.h"
+#include "vector.h"
+#include "heap.h"
+#include "syscalls.h"
+#include "playrtp.h"
+
+/** @brief /dev/dsp (or whatever) */
+static int playrtp_oss_fd = -1;
+
+/** @brief Open and configure the OSS audio device */
+static void playrtp_oss_enable(void) {
+ if(playrtp_oss_fd == -1) {
+ int rate = 44100, stereo = 1, format = AFMT_S16_BE;
+ if(!device) {
+ if(access("/dev/dsp", W_OK) == 0)
+ device = "/dev/dsp";
+ else if(access("/dev/audio", W_OK) == 0)
+ device = "/dev/audio";
+ else
+ fatal(0, "cannot determine default audio device");
+ }
+ if((playrtp_oss_fd = open(device, O_WRONLY)) < 0)
+ fatal(errno, "error opening %s", device);
+ if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SETFMT, &format) < 0)
+ fatal(errno, "ioctl SNDCTL_DSP_SETFMT");
+ if(ioctl(playrtp_oss_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
+ fatal(errno, "ioctl SNDCTL_DSP_STEREO");
+ if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SPEED, &rate) < 0)
+ fatal(errno, "ioctl SNDCTL_DSP_SPEED");
+ if(rate != 44100)
+ error(0, "asking for 44100Hz, got %dHz", rate);
+ nonblock(playrtp_oss_fd);
+ }
+}
+
+/** @brief Wait until the audio device can accept more data */
+static void playrtp_oss_wait(void) {
+ struct pollfd fds[1];
+ int n;
+
+ do {
+ fds[0].fd = playrtp_oss_fd;
+ fds[0].events = POLLOUT;
+ while((n = poll(fds, 1, -1)) < 0 && errno == EINTR)
+ ;
+ if(n < 0)
+ fatal(errno, "calling poll");
+ } while(!(fds[0].revents & (POLLOUT|POLLERR)));
+}
+
+/** @brief Close the OSS output device
+ * @param hard If nonzero, drop pending data
+ */
+static void playrtp_oss_disable(int hard) {
+ if(hard)
+ if(ioctl(playrtp_oss_fd, SNDCTL_DSP_RESET, 0) < 0)
+ error(errno, "ioctl SNDCTL_DSP_RESET");
+ xclose(playrtp_oss_fd);
+ playrtp_oss_fd = -1;
+}
+
+/** @brief Write samples to OSS output device
+ * @param data Pointer to sample data
+ * @param nsamples Number of samples
+ * @return 0 on success, non-0 on error
+ */
+static int playrtp_oss_write(const void *data, size_t samples) {
+ const ssize_t nbyteswritten = write(playrtp_oss_fd, data,
+ samples * sizeof (int16_t));
+
+ if(nbyteswritten < 0) {
+ switch(errno) {
+ case EAGAIN:
+ case EINTR:
+ return 0;
+ default:
+ error(errno, "error writing to %s", device);
+ return -1;
+ }
+ } else {
+ next_timestamp += nbyteswritten / 2;
+ return 0;
+ }
+}
+
+/** @brief Play some data from packet @p p
+ *
+ * @p p is assumed to contain @ref next_timestamp.
+ */
+static int playrtp_oss_play(const struct packet *p) {
+ return playrtp_oss_write(p->samples_raw + next_timestamp - p->timestamp,
+ (p->timestamp + p->nsamples) - next_timestamp);
+}
+
+/** @brief Play some silence before packet @p p
+ *
+ * @p p is assumed to be entirely before @ref next_timestamp.
+ */
+static int playrtp_oss_infill(const struct packet *p) {
+ static const uint16_t zeros[INFILL_SAMPLES];
+ size_t samples_available = INFILL_SAMPLES;
+
+ if(p && samples_available > p->timestamp - next_timestamp)
+ samples_available = p->timestamp - next_timestamp;
+ return playrtp_oss_write(zeros, samples_available);
+}
+
+/** @brief OSS backend for playrtp */
+void playrtp_oss(void) {
+ int escape;
+ const struct packet *p;
+
+ pthread_mutex_lock(&lock);
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ playrtp_fill_buffer();
+ playrtp_oss_enable();
+ escape = 0;
+ info("Playing...");
+ /* Keep playing until the buffer empties out, we get an error */
+ while((nsamples >= minbuffer
+ || (nsamples > 0
+ && contains(pheap_first(&packets), next_timestamp)))
+ && !escape) {
+ /* Wait until we can play more */
+ pthread_mutex_unlock(&lock);
+ playrtp_oss_wait();
+ pthread_mutex_lock(&lock);
+ /* Device is ready for more data, find something to play */
+ p = playrtp_next_packet();
+ /* Play it or play some silence */
+ if(contains(p, next_timestamp))
+ escape = playrtp_oss_play(p);
+ else
+ escape = playrtp_oss_infill(p);
+ }
+ active = 0;
+ /* We stop playing for a bit until the buffer re-fills */
+ pthread_mutex_unlock(&lock);
+ playrtp_oss_disable(escape);
+ pthread_mutex_lock(&lock);
+ }
+}
+
+#endif
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/