mdw@git.distorted.org.uk Git - disorder/blob - clients/playrtp-oss.c

   1 /*
   2  * This file is part of DisOrder.
   3  * Copyright (C) 2007 Richard Kettlewell
   4  * Portions copyright (C) 2007 Ross Younger
   5  *
   6  * This program is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful, but
  12  * WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, write to the Free Software
  18  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
  19  * USA
  20  */
  21 /** @file clients/playrtp-oss.c
  22  * @brief RTP player - OSS and empeg support
  23  */
  24
  25 #include "common.h"
  26
  27 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
  28
  29 #include <poll.h>
  30 #include <sys/ioctl.h>
  31 #if !EMPEG_HOST
  32 #include <sys/soundcard.h>
  33 #endif
  34 #include <pthread.h>
  35 #include <fcntl.h>
  36 #include <unistd.h>
  37 #include <errno.h>
  38 #include <arpa/inet.h>
  39
  40 #include "mem.h"
  41 #include "log.h"
  42 #include "vector.h"
  43 #include "heap.h"
  44 #include "syscalls.h"
  45 #include "playrtp.h"
  46
  47 /** @brief /dev/dsp (or whatever) */
  48 static int playrtp_oss_fd = -1;
  49
  50 /** @brief Audio buffer */
  51 static char *playrtp_oss_buffer;
  52
  53 /** @brief Size of @ref playrtp_oss_buffer in bytes */
  54 static int playrtp_oss_bufsize;
  55
  56 /** @brief Number of bytes used in @ref playrtp_oss_buffer */
  57 static int playrtp_oss_bufused;
  58
  59 /** @brief Open and configure the OSS audio device */
  60 static void playrtp_oss_enable(void) {
  61   if(playrtp_oss_fd == -1) {
  62 #if EMPEG_HOST
  63     /* empeg audio driver only knows /dev/audio, only supports the equivalent
  64      * of AFMT_S16_NE, has a fixed buffer size, and does not support the
  65      * SNDCTL_ ioctls. */
  66     if(!device)
  67       device = "/dev/audio";
  68     if((playrtp_oss_fd = open(device, O_WRONLY)) < 0)
  69       fatal(errno, "error opening %s", device);
  70     playrtp_oss_bufsize = 4608;
  71 #else
  72     int rate = 44100, stereo = 1, format = AFMT_S16_BE;
  73     if(!device) {
  74       if(access("/dev/dsp", W_OK) == 0)
  75         device = "/dev/dsp";
  76       else if(access("/dev/audio", W_OK) == 0)
  77         device = "/dev/audio";
  78       else
  79         fatal(0, "cannot determine default audio device");
  80     }
  81     if((playrtp_oss_fd = open(device, O_WRONLY)) < 0)
  82       fatal(errno, "error opening %s", device);
  83     if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SETFMT, &format) < 0)
  84       fatal(errno, "ioctl SNDCTL_DSP_SETFMT");
  85     if(ioctl(playrtp_oss_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
  86       fatal(errno, "ioctl SNDCTL_DSP_STEREO");
  87     if(ioctl(playrtp_oss_fd, SNDCTL_DSP_SPEED, &rate) < 0)
  88       fatal(errno, "ioctl SNDCTL_DSP_SPEED");
  89     if(rate != 44100)
  90       error(0, "asking for 44100Hz, got %dHz", rate);
  91     if(ioctl(playrtp_oss_fd, SNDCTL_DSP_GETBLKSIZE, &playrtp_oss_bufsize) < 0)
  92       fatal(errno, "ioctl SNDCTL_DSP_GETBLKSIZE");
  93     info("OSS buffer size %d", playrtp_oss_bufsize);
  94 #endif
  95     playrtp_oss_buffer = xmalloc(playrtp_oss_bufsize);
  96     playrtp_oss_bufused = 0;
  97     nonblock(playrtp_oss_fd);
  98   }
  99 }
 100
 101 /** @brief Flush the OSS output buffer
 102  * @return 0 on success, non-0 on error
 103  */
 104 static int playrtp_oss_flush(void) {
 105   int nbyteswritten;
 106
 107   if(!playrtp_oss_bufused)
 108     return 0;                           /* nothing to do */
 109   /* 0 out the unused portion of the buffer */
 110   memset(playrtp_oss_buffer + playrtp_oss_bufused, 0,
 111          playrtp_oss_bufsize - playrtp_oss_bufused);
 112 #if EMPEG_HOST
 113   /* empeg audio driver insists on native-endian samples */
 114   {
 115     uint16_t *ptr,
 116       *const limit = (uint16_t *)(playrtp_oss_buffer + playrtp_oss_bufused);
 117
 118     for(ptr = (uint16_t *)playrtp_oss_buffer; ptr < limit; ++ptr)
 119       *ptr = ntohs(*ptr);
 120   }
 121 #endif
 122   for(;;) {
 123     nbyteswritten = write(playrtp_oss_fd,
 124                           playrtp_oss_buffer, playrtp_oss_bufsize);
 125     if(nbyteswritten < 0) {
 126       switch(errno) {
 127       case EINTR:
 128         break;                          /* try again */
 129       case EAGAIN:
 130         return 0;                       /* try later */
 131       default:
 132         error(errno, "error writing to %s", device);
 133         return -1;
 134       }
 135     } else {
 136       if(nbyteswritten < playrtp_oss_bufsize)
 137         error(0, "%s: short write (%d/%d)",
 138               device, nbyteswritten, playrtp_oss_bufsize);
 139       if(dump_buffer) {
 140         int count;
 141         const int16_t *sp = (const int16_t *)playrtp_oss_buffer;
 142
 143         for(count = 0; count < playrtp_oss_bufsize; count += sizeof(int16_t)) {
 144           dump_buffer[dump_index++] = (int16_t)ntohs(*sp++);
 145           dump_index %= dump_size;
 146         }
 147       }
 148       playrtp_oss_bufused = 0;
 149       return 0;
 150     }
 151   }
 152 }
 153
 154 /** @brief Wait until the audio device can accept more data */
 155 static void playrtp_oss_wait(void) {
 156   struct pollfd fds[1];
 157   int n;
 158
 159   do {
 160     fds[0].fd = playrtp_oss_fd;
 161     fds[0].events = POLLOUT;
 162     while((n = poll(fds, 1, -1)) < 0 && errno == EINTR)
 163       ;
 164     if(n < 0)
 165       fatal(errno, "calling poll");
 166   } while(!(fds[0].revents & (POLLOUT|POLLERR)));
 167 }
 168
 169 /** @brief Close the OSS output device
 170  * @param hard If nonzero, drop pending data
 171  */
 172 static void playrtp_oss_disable(int hard) {
 173   if(hard) {
 174 #if !EMPEG_HOST
 175     /* No SNDCTL_DSP_ ioctls on empeg */
 176     if(ioctl(playrtp_oss_fd, SNDCTL_DSP_RESET, 0) < 0)
 177       error(errno, "ioctl SNDCTL_DSP_RESET");
 178 #endif
 179   } else
 180     playrtp_oss_flush();
 181   xclose(playrtp_oss_fd);
 182   playrtp_oss_fd = -1;
 183   free(playrtp_oss_buffer);
 184   playrtp_oss_buffer = 0;
 185 }
 186
 187 /** @brief Write samples to OSS output device
 188  * @param data Pointer to sample data
 189  * @param samples Number of samples
 190  * @return 0 on success, non-0 on error
 191  */
 192 static int playrtp_oss_write(const char *data, size_t samples) {
 193   long bytes = samples * sizeof(int16_t);
 194   while(bytes > 0) {
 195     int n = playrtp_oss_bufsize - playrtp_oss_bufused;
 196
 197     if(n > bytes)
 198       n = bytes;
 199     memcpy(playrtp_oss_buffer + playrtp_oss_bufused, data, n);
 200     bytes -= n;
 201     data += n;
 202     playrtp_oss_bufused += n;
 203     if(playrtp_oss_bufused == playrtp_oss_bufsize)
 204       if(playrtp_oss_flush())
 205         return -1;
 206   }
 207   next_timestamp += samples;
 208   return 0;
 209 }
 210
 211 /** @brief Play some data from packet @p p
 212  *
 213  * @p p is assumed to contain @ref next_timestamp.
 214  */
 215 static int playrtp_oss_play(const struct packet *p) {
 216   return playrtp_oss_write
 217     ((const char *)(p->samples_raw + next_timestamp - p->timestamp),
 218      (p->timestamp + p->nsamples) - next_timestamp);
 219 }
 220
 221 /** @brief Play some silence before packet @p p
 222  *
 223  * @p p is assumed to be entirely before @ref next_timestamp.
 224  */
 225 static int playrtp_oss_infill(const struct packet *p) {
 226   static const char zeros[INFILL_SAMPLES * sizeof(int16_t)];
 227   size_t samples_available = INFILL_SAMPLES;
 228
 229   if(p && samples_available > p->timestamp - next_timestamp)
 230     samples_available = p->timestamp - next_timestamp;
 231   return playrtp_oss_write(zeros, samples_available);
 232 }
 233
 234 /** @brief OSS backend for playrtp */
 235 void playrtp_oss(void) {
 236   int escape;
 237   const struct packet *p;
 238
 239   pthread_mutex_lock(&lock);
 240   for(;;) {
 241     /* Wait for the buffer to fill up a bit */
 242     playrtp_fill_buffer();
 243     playrtp_oss_enable();
 244     escape = 0;
 245     info("Playing...");
 246     /* Keep playing until the buffer empties out, we get an error */
 247     while((nsamples >= minbuffer
 248            || (nsamples > 0
 249                && contains(pheap_first(&packets), next_timestamp)))
 250           && !escape) {
 251       /* Wait until we can play more */
 252       pthread_mutex_unlock(&lock);
 253       playrtp_oss_wait();
 254       pthread_mutex_lock(&lock);
 255       /* Device is ready for more data, find something to play */
 256       p = playrtp_next_packet();
 257       /* Play it or play some silence */
 258       if(contains(p, next_timestamp))
 259         escape = playrtp_oss_play(p);
 260       else
 261         escape = playrtp_oss_infill(p);
 262     }
 263     active = 0;
 264     /* We stop playing for a bit until the buffer re-fills */
 265     pthread_mutex_unlock(&lock);
 266     playrtp_oss_disable(escape);
 267     pthread_mutex_lock(&lock);
 268   }
 269 }
 270
 271 #endif
 272
 273 /*
 274 Local Variables:
 275 c-basic-offset:2
 276 comment-column:40
 277 fill-column:79
 278 indent-tabs-mode:nil
 279 End:
 280 */