Sat Nov 1 06:28:27 2008

Asterisk developer's documentation

---

chan_oss.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
 * note-this code best seen with ts=8 (8-spaces tabs) in the editor
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief Channel driver for OSS sound cards
 *
 * \author Mark Spencer <markster@digium.com>
 * \author Luigi Rizzo
 *
 * \par See also
 * \arg \ref Config_oss
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
   <depend>ossaudio</depend>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 108796 $")

#include <stdio.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <sys/time.h>
#include <stdlib.h>
#include <errno.h>

#ifdef __linux
#include <linux/soundcard.h>
#elif defined(__FreeBSD__)
#include <sys/soundcard.h>
#else
#include <soundcard.h>
#endif

#include "asterisk/lock.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/callerid.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/cli.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/endian.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"

/* ringtones we use */
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"

/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
   .flags = 0,
   .max_size = -1,
   .resync_threshold = -1,
   .impl = "",
};
static struct ast_jb_conf global_jbconf;

/*
 * Basic mode of operation:
 *
 * we have one keyboard (which receives commands from the keyboard)
 * and multiple headset's connected to audio cards.
 * Cards/Headsets are named as the sections of oss.conf.
 * The section called [general] contains the default parameters.
 *
 * At any time, the keyboard is attached to one card, and you
 * can switch among them using the command 'console foo'
 * where 'foo' is the name of the card you want.
 *
 * oss.conf parameters are
START_CONFIG

[general]
    ; General config options, with default values shown.
    ; You should use one section per device, with [general] being used
    ; for the first device and also as a template for other devices.
    ;
    ; All but 'debug' can go also in the device-specific sections.
    ;
    ; debug = 0x0    ; misc debug flags, default is 0

    ; Set the device to use for I/O
    ; device = /dev/dsp

    ; Optional mixer command to run upon startup (e.g. to set
    ; volume levels, mutes, etc.
    ; mixer =

    ; Software mic volume booster (or attenuator), useful for sound
    ; cards or microphones with poor sensitivity. The volume level
    ; is in dB, ranging from -20.0 to +20.0
    ; boost = n         ; mic volume boost in dB

    ; Set the callerid for outgoing calls
    ; callerid = John Doe <555-1234>

    ; autoanswer = no      ; no autoanswer on call
    ; autohangup = yes     ; hangup when other party closes
    ; extension = s     ; default extension to call
    ; context = default    ; default context for outgoing calls
    ; language = ""     ; default language

    ; Default Music on Hold class to use when this channel is placed on hold in
    ; the case that the music class is not set on the channel with
    ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
    ; putting this one on hold did not suggest a class to use.
    ;
    ; mohinterpret=default

    ; If you set overridecontext to 'yes', then the whole dial string
    ; will be interpreted as an extension, which is extremely useful
    ; to dial SIP, IAX and other extensions which use the '@' character.
    ; The default is 'no' just for backward compatibility, but the
    ; suggestion is to change it.
    ; overridecontext = no ; if 'no', the last @ will start the context
            ; if 'yes' the whole string is an extension.

    ; low level device parameters in case you have problems with the
    ; device driver on your operating system. You should not touch these
    ; unless you know what you are doing.
    ; queuesize = 10    ; frames in device driver
    ; frags = 8         ; argument to SETFRAGMENT

    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
                                  ; OSS channel. Defaults to "no". An enabled jitterbuffer will
                                  ; be used only if the sending side can create and the receiving
                                  ; side can not accept jitter. The OSS channel can't accept jitter,
                                  ; thus an enabled jitterbuffer on the receive OSS side will always
                                  ; be used if the sending side can create jitter.

    ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

    ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                                  ; resynchronized. Useful to improve the quality of the voice, with
                                  ; big jumps in/broken timestamps, usualy sent from exotic devices
                                  ; and programs. Defaults to 1000.

    ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of an OSS
                                  ; channel. Two implementations are currenlty available - "fixed"
                                  ; (with size always equals to jbmax-size) and "adaptive" (with
                                  ; variable size, actually the new jb of IAX2). Defaults to fixed.

    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
    ;-----------------------------------------------------------------------------------

[card1]
    ; device = /dev/dsp1   ; alternate device

END_CONFIG

.. and so on for the other cards.

 */

/*
 * Helper macros to parse config arguments. They will go in a common
 * header file if their usage is globally accepted. In the meantime,
 * we define them here. Typical usage is as below.
 * Remember to open a block right before M_START (as it declares
 * some variables) and use the M_* macros WITHOUT A SEMICOLON:
 *
 * {
 *    M_START(v->name, v->value) 
 *
 *    M_BOOL("dothis", x->flag1)
 *    M_STR("name", x->somestring)
 *    M_F("bar", some_c_code)
 *    M_END(some_final_statement)
 *    ... other code in the block
 * }
 *
 * XXX NOTE these macros should NOT be replicated in other parts of asterisk. 
 * Likely we will come up with a better way of doing config file parsing.
 */
#define M_START(var, val) \
        char *__s = var; char *__val = val;
#define M_END(x)   x;
#define M_F(tag, f)        if (!strcasecmp((__s), tag)) { f; } else
#define M_BOOL(tag, dst)   M_F(tag, (dst) = ast_true(__val) )
#define M_UINT(tag, dst)   M_F(tag, (dst) = strtoul(__val, NULL, 0) )
#define M_STR(tag, dst)    M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))

/*
 * The following parameters are used in the driver:
 *
 *  FRAME_SIZE the size of an audio frame, in samples.
 *    160 is used almost universally, so you should not change it.
 *
 *  FRAGS   the argument for the SETFRAGMENT ioctl.
 *    Overridden by the 'frags' parameter in oss.conf
 *
 *    Bits 0-7 are the base-2 log of the device's block size,
 *    bits 16-31 are the number of blocks in the driver's queue.
 *    There are a lot of differences in the way this parameter
 *    is supported by different drivers, so you may need to
 *    experiment a bit with the value.
 *    A good default for linux is 30 blocks of 64 bytes, which
 *    results in 6 frames of 320 bytes (160 samples).
 *    FreeBSD works decently with blocks of 256 or 512 bytes,
 *    leaving the number unspecified.
 *    Note that this only refers to the device buffer size,
 *    this module will then try to keep the lenght of audio
 *    buffered within small constraints.
 *
 *  QUEUE_SIZE The max number of blocks actually allowed in the device
 *    driver's buffer, irrespective of the available number.
 *    Overridden by the 'queuesize' parameter in oss.conf
 *
 *    Should be >=2, and at most as large as the hw queue above
 *    (otherwise it will never be full).
 */

#define FRAME_SIZE   160
#define  QUEUE_SIZE  10

#if defined(__FreeBSD__)
#define  FRAGS 0x8
#else
#define  FRAGS ( ( (6 * 5) << 16 ) | 0x6 )
#endif

/*
 * XXX text message sizes are probably 256 chars, but i am
 * not sure if there is a suitable definition anywhere.
 */
#define TEXT_SIZE 256

#if 0
#define  TRYOPEN  1           /* try to open on startup */
#endif
#define  O_CLOSE  0x444       /* special 'close' mode for device */
/* Which device to use */
#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
#define DEV_DSP "/dev/audio"
#else
#define DEV_DSP "/dev/dsp"
#endif

#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif

static char *config = "oss.conf";   /* default config file */

static int oss_debug;

/*
 * Each sound is made of 'datalen' samples of sound, repeated as needed to
 * generate 'samplen' samples of data, then followed by 'silencelen' samples
 * of silence. The loop is repeated if 'repeat' is set.
 */
struct sound {
   int ind;
   char *desc;
   short *data;
   int datalen;
   int samplen;
   int silencelen;
   int repeat;
};

static struct sound sounds[] = {
   { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
   { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
   { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
   { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
   { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
   { -1, NULL, 0, 0, 0, 0 },  /* end marker */
};


/*
 * descriptor for one of our channels.
 * There is one used for 'default' values (from the [general] entry in
 * the configuration file), and then one instance for each device
 * (the default is cloned from [general], others are only created
 * if the relevant section exists).
 */
struct chan_oss_pvt {
   struct chan_oss_pvt *next;

   char *name;
   /*
    * cursound indicates which in struct sound we play. -1 means nothing,
    * any other value is a valid sound, in which case sampsent indicates
    * the next sample to send in [0..samplen + silencelen]
    * nosound is set to disable the audio data from the channel
    * (so we can play the tones etc.).
    */
   int sndcmd[2];          /* Sound command pipe */
   int cursound;           /* index of sound to send */
   int sampsent;           /* # of sound samples sent  */
   int nosound;            /* set to block audio from the PBX */

   int total_blocks;       /* total blocks in the output device */
   int sounddev;
   enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
   int autoanswer;
   int autohangup;
   int hookstate;
   char *mixer_cmd;        /* initial command to issue to the mixer */
   unsigned int queuesize;    /* max fragments in queue */
   unsigned int frags;        /* parameter for SETFRAGMENT */

   int warned;             /* various flags used for warnings */
#define WARN_used_blocks   1
#define WARN_speed      2
#define WARN_frag    4
   int w_errors;           /* overfull in the write path */
   struct timeval lastopen;

   int overridecontext;
   int mute;

   /* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
    * be representable in 16 bits to avoid overflows.
    */
#define  BOOST_SCALE (1<<9)
#define  BOOST_MAX   40       /* slightly less than 7 bits */
   int boost;              /* input boost, scaled by BOOST_SCALE */
   char device[64];        /* device to open */

   pthread_t sthread;

   struct ast_channel *owner;
   char ext[AST_MAX_EXTENSION];
   char ctx[AST_MAX_CONTEXT];
   char language[MAX_LANGUAGE];
   char cid_name[256];        /*XXX */
   char cid_num[256];         /*XXX */
   char mohinterpret[MAX_MUSICCLASS];

   /* buffers used in oss_write */
   char oss_write_buf[FRAME_SIZE * 2];
   int oss_write_dst;
   /* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
    * plus enough room for a full frame
    */
   char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
   int readpos;            /* read position above */
   struct ast_frame read_f;   /* returned by oss_read */
};

static struct chan_oss_pvt oss_default = {
   .cursound = -1,
   .sounddev = -1,
   .duplex = M_UNSET,         /* XXX check this */
   .autoanswer = 1,
   .autohangup = 1,
   .queuesize = QUEUE_SIZE,
   .frags = FRAGS,
   .ext = "s",
   .ctx = "default",
   .readpos = AST_FRIENDLY_OFFSET,  /* start here on reads */
   .lastopen = { 0, 0 },
   .boost = BOOST_SCALE,
};

static char *oss_active;    /* the active device */

static int setformat(struct chan_oss_pvt *o, int mode);

static struct ast_channel *oss_request(const char *type, int format, void *data
, int *cause);
static int oss_digit_begin(struct ast_channel *c, char digit);
static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
static int oss_text(struct ast_channel *c, const char *text);
static int oss_hangup(struct ast_channel *c);
static int oss_answer(struct ast_channel *c);
static struct ast_frame *oss_read(struct ast_channel *chan);
static int oss_call(struct ast_channel *c, char *dest, int timeout);
static int oss_write(struct ast_channel *chan, struct ast_frame *f);
static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static char tdesc[] = "OSS Console Channel Driver";

static const struct ast_channel_tech oss_tech = {
   .type = "Console",
   .description = tdesc,
   .capabilities = AST_FORMAT_SLINEAR,
   .requester = oss_request,
   .send_digit_begin = oss_digit_begin,
   .send_digit_end = oss_digit_end,
   .send_text = oss_text,
   .hangup = oss_hangup,
   .answer = oss_answer,
   .read = oss_read,
   .call = oss_call,
   .write = oss_write,
   .indicate = oss_indicate,
   .fixup = oss_fixup,
};

/*
 * returns a pointer to the descriptor with the given name
 */
static struct chan_oss_pvt *find_desc(char *dev)
{
   struct chan_oss_pvt *o = NULL;

   if (!dev)
      ast_log(LOG_WARNING, "null dev\n");

   for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);

   if (!o)
      ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");

   return o;
}

/*
 * split a string in extension-context, returns pointers to malloc'ed
 * strings.
 * If we do not have 'overridecontext' then the last @ is considered as
 * a context separator, and the context is overridden.
 * This is usually not very necessary as you can play with the dialplan,
 * and it is nice not to need it because you have '@' in SIP addresses.
 * Return value is the buffer address.
 */
static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
{
   struct chan_oss_pvt *o = find_desc(oss_active);

   if (ext == NULL || ctx == NULL)
      return NULL;         /* error */

   *ext = *ctx = NULL;

   if (src && *src != '\0')
      *ext = ast_strdup(src);

   if (*ext == NULL)
      return NULL;

   if (!o->overridecontext) {
      /* parse from the right */
      *ctx = strrchr(*ext, '@');
      if (*ctx)
         *(*ctx)++ = '\0';
   }

   return *ext;
}

/*
 * Returns the number of blocks used in the audio output channel
 */
static int used_blocks(struct chan_oss_pvt *o)
{
   struct audio_buf_info info;

   if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
      if (!(o->warned & WARN_used_blocks)) {
         ast_log(LOG_WARNING, "Error reading output space\n");
         o->warned |= WARN_used_blocks;
      }
      return 1;
   }

   if (o->total_blocks == 0) {
      if (0)               /* debugging */
         ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
      o->total_blocks = info.fragments;
   }

   return o->total_blocks - info.fragments;
}

/* Write an exactly FRAME_SIZE sized frame */
static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
{
   int res;

   if (o->sounddev < 0)
      setformat(o, O_RDWR);
   if (o->sounddev < 0)
      return 0;            /* not fatal */
   /*
    * Nothing complex to manage the audio device queue.
    * If the buffer is full just drop the extra, otherwise write.
    * XXX in some cases it might be useful to write anyways after
    * a number of failures, to restart the output chain.
    */
   res = used_blocks(o);
   if (res > o->queuesize) {  /* no room to write a block */
      if (o->w_errors++ == 0 && (oss_debug & 0x4))
         ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
      return 0;
   }
   o->w_errors = 0;
   return write(o->sounddev, ((void *) data), FRAME_SIZE * 2);
}

/*
 * Handler for 'sound writable' events from the sound thread.
 * Builds a frame from the high level description of the sounds,
 * and passes it to the audio device.
 * The actual sound is made of 1 or more sequences of sound samples
 * (s->datalen, repeated to make s->samplen samples) followed by
 * s->silencelen samples of silence. The position in the sequence is stored
 * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen.
 * In case we fail to write a frame, don't update o->sampsent.
 */
static void send_sound(struct chan_oss_pvt *o)
{
   short myframe[FRAME_SIZE];
   int ofs, l, start;
   int l_sampsent = o->sampsent;
   struct sound *s;

   if (o->cursound < 0)    /* no sound to send */
      return;

   s = &sounds[o->cursound];

   for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
      l = s->samplen - l_sampsent;  /* # of available samples */
      if (l > 0) {
         start = l_sampsent % s->datalen; /* source offset */
         if (l > FRAME_SIZE - ofs)  /* don't overflow the frame */
            l = FRAME_SIZE - ofs;
         if (l > s->datalen - start)   /* don't overflow the source */
            l = s->datalen - start;
         bcopy(s->data + start, myframe + ofs, l * 2);
         if (0)
            ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", l_sampsent, l, s->samplen, ofs);
         l_sampsent += l;
      } else {          /* end of samples, maybe some silence */
         static const short silence[FRAME_SIZE] = { 0, };

         l += s->silencelen;
         if (l > 0) {
            if (l > FRAME_SIZE - ofs)
               l = FRAME_SIZE - ofs;
            bcopy(silence, myframe + ofs, l * 2);
            l_sampsent += l;
         } else {       /* silence is over, restart sound if loop */
            if (s->repeat == 0) {   /* last block */
               o->cursound = -1;
               o->nosound = 0;   /* allow audio data */
               if (ofs < FRAME_SIZE)   /* pad with silence */
                  bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs) * 2);
            }
            l_sampsent = 0;
         }
      }
   }
   l = soundcard_writeframe(o, myframe);
   if (l > 0)
      o->sampsent = l_sampsent;  /* update status */
}

static void *sound_thread(void *arg)
{
   char ign[4096];
   struct chan_oss_pvt *o = (struct chan_oss_pvt *) arg;

   /*
    * Just in case, kick the driver by trying to read from it.
    * Ignore errors - this read is almost guaranteed to fail.
    */
   read(o->sounddev, ign, sizeof(ign));
   for (;;) {
      fd_set rfds, wfds;
      int maxfd, res;

      FD_ZERO(&rfds);
      FD_ZERO(&wfds);
      FD_SET(o->sndcmd[0], &rfds);
      maxfd = o->sndcmd[0];   /* pipe from the main process */
      if (o->cursound > -1 && o->sounddev < 0)
         setformat(o, O_RDWR);   /* need the channel, try to reopen */
      else if (o->cursound == -1 && o->owner == NULL)
         setformat(o, O_CLOSE);  /* can close */
      if (o->sounddev > -1) {
         if (!o->owner) {  /* no one owns the audio, so we must drain it */
            FD_SET(o->sounddev, &rfds);
            maxfd = MAX(o->sounddev, maxfd);
         }
         if (o->cursound > -1) {
            FD_SET(o->sounddev, &wfds);
            maxfd = MAX(o->sounddev, maxfd);
         }
      }
      /* ast_select emulates linux behaviour in terms of timeout handling */
      res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL);
      if (res < 1) {
         ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
         sleep(1);
         continue;
      }
      if (FD_ISSET(o->sndcmd[0], &rfds)) {
         /* read which sound to play from the pipe */
         int i, what = -1;

         read(o->sndcmd[0], &what, sizeof(what));
         for (i = 0; sounds[i].ind != -1; i++) {
            if (sounds[i].ind == what) {
               o->cursound = i;
               o->sampsent = 0;
               o->nosound = 1;   /* block audio from pbx */
               break;
            }
         }
         if (sounds[i].ind == -1)
            ast_log(LOG_WARNING, "invalid sound index: %d\n", what);
      }
      if (o->sounddev > -1) {
         if (FD_ISSET(o->sounddev, &rfds))   /* read and ignore errors */
            read(o->sounddev, ign, sizeof(ign));
         if (FD_ISSET(o->sounddev, &wfds))
            send_sound(o);
      }
   }
   return NULL;            /* Never reached */
}

/*
 * reset and close the device if opened,
 * then open and initialize it in the desired mode,
 * trigger reads and writes so we can start using it.
 */
static int setformat(struct chan_oss_pvt *o, int mode)
{
   int fmt, desired, res, fd;

   if (o->sounddev >= 0) {
      ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
      close(o->sounddev);
      o->duplex = M_UNSET;
      o->sounddev = -1;
   }
   if (mode == O_CLOSE)    /* we are done */
      return 0;
   if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
      return -1;           /* don't open too often */
   o->lastopen = ast_tvnow();
   fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
   if (fd < 0) {
      ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
      return -1;
   }
   if (o->owner)
      o->owner->fds[0] = fd;

#if __BYTE_ORDER == __LITTLE_ENDIAN
   fmt = AFMT_S16_LE;
#else
   fmt = AFMT_S16_BE;
#endif
   res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
   if (res < 0) {
      ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
      return -1;
   }
   switch (mode) {
      case O_RDWR:
         res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
         /* Check to see if duplex set (FreeBSD Bug) */
         res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
         if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
            if (option_verbose > 1)
               ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
            o->duplex = M_FULL;
         };
         break;
      case O_WRONLY:
         o->duplex = M_WRITE;
         break;
      case O_RDONLY:
         o->duplex = M_READ;
         break;
   }

   fmt = 0;
   res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
   if (res < 0) {
      ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
      return -1;
   }
   fmt = desired = DEFAULT_SAMPLE_RATE;   /* 8000 Hz desired */
   res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);

   if (res < 0) {
      ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
      return -1;
   }
   if (fmt != desired) {
      if (!(o->warned & WARN_speed)) {
         ast_log(LOG_WARNING,
             "Requested %d Hz, got %d Hz -- sound may be choppy\n",
             desired, fmt);
         o->warned |= WARN_speed;
      }
   }
   /*
    * on Freebsd, SETFRAGMENT does not work very well on some cards.
    * Default to use 256 bytes, let the user override
    */
   if (o->frags) {
      fmt = o->frags;
      res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
      if (res < 0) {
         if (!(o->warned & WARN_frag)) {
            ast_log(LOG_WARNING,
               "Unable to set fragment size -- sound may be choppy\n");
            o->warned |= WARN_frag;
         }
      }
   }
   /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
   res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
   res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
   /* it may fail if we are in half duplex, never mind */
   return 0;
}

/*
 * some of the standard methods supported by channels.
 */
static int oss_digit_begin(struct ast_channel *c, char digit)
{
   return 0;
}

static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
{
   /* no better use for received digits than print them */
   ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
      digit, duration);
   return 0;
}

static int oss_text(struct ast_channel *c, const char *text)
{
   /* print received messages */
   ast_verbose(" << Console Received text %s >> \n", text);
   return 0;
}

/* Play ringtone 'x' on device 'o' */
static void ring(struct chan_oss_pvt *o, int x)
{
   write(o->sndcmd[1], &x, sizeof(x));
}


/*
 * handler for incoming calls. Either autoanswer, or start ringing
 */
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
   struct chan_oss_pvt *o = c->tech_pvt;
   struct ast_frame f = { 0, };

   ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n", dest, c->cid.cid_dnid, c->cid.cid_rdnis, c->cid.cid_name, c->cid.cid_num);
   if (o->autoanswer) {
      ast_verbose(" << Auto-answered >> \n");
      f.frametype = AST_FRAME_CONTROL;
      f.subclass = AST_CONTROL_ANSWER;
      ast_queue_frame(c, &f);
   } else {
      ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
      f.frametype = AST_FRAME_CONTROL;
      f.subclass = AST_CONTROL_RINGING;
      ast_queue_frame(c, &f);
      ring(o, AST_CONTROL_RING);
   }
   return 0;
}

/*
 * remote side answered the phone
 */
static int oss_answer(struct ast_channel *c)
{
   struct chan_oss_pvt *o = c->tech_pvt;

   ast_verbose(" << Console call has been answered >> \n");
#if 0
   /* play an answer tone (XXX do we really need it ?) */
   ring(o, AST_CONTROL_ANSWER);
#endif
   ast_setstate(c, AST_STATE_UP);
   o->cursound = -1;
   o->nosound = 0;
   return 0;
}

static int oss_hangup(struct ast_channel *c)
{
   struct chan_oss_pvt *o = c->tech_pvt;

   o->cursound = -1;
   o->nosound = 0;
   c->tech_pvt = NULL;
   o->owner = NULL;
   ast_verbose(" << Hangup on console >> \n");
   ast_module_unref(ast_module_info->self);
   if (o->hookstate) {
      if (o->autoanswer || o->autohangup) {
         /* Assume auto-hangup too */
         o->hookstate = 0;
         setformat(o, O_CLOSE);
      } else {
         /* Make congestion noise */
         ring(o, AST_CONTROL_CONGESTION);
      }
   }
   return 0;
}

/* used for data coming from the network */
static int oss_write(struct ast_channel *c, struct ast_frame *f)
{
   int src;
   struct chan_oss_pvt *o = c->tech_pvt;

   /* Immediately return if no sound is enabled */
   if (o->nosound)
      return 0;
   /* Stop any currently playing sound */
   o->cursound = -1;
   /*
    * we could receive a block which is not a multiple of our
    * FRAME_SIZE, so buffer it locally and write to the device
    * in FRAME_SIZE chunks.
    * Keep the residue stored for future use.
    */
   src = 0;             /* read position into f->data */
   while (src < f->datalen) {
      /* Compute spare room in the buffer */
      int l = sizeof(o->oss_write_buf) - o->oss_write_dst;

      if (f->datalen - src >= l) {  /* enough to fill a frame */
         memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
         soundcard_writeframe(o, (short *) o->oss_write_buf);
         src += l;
         o->oss_write_dst = 0;
      } else {          /* copy residue */
         l = f->datalen - src;
         memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
         src += l;         /* but really, we are done */
         o->oss_write_dst += l;
      }
   }
   return 0;
}

static struct ast_frame *oss_read(struct ast_channel *c)
{
   int res;
   struct chan_oss_pvt *o = c->tech_pvt;
   struct ast_frame *f = &o->read_f;

   /* XXX can be simplified returning &ast_null_frame */
   /* prepare a NULL frame in case we don't have enough data to return */
   bzero(f, sizeof(struct ast_frame));
   f->frametype = AST_FRAME_NULL;
   f->src = oss_tech.type;

   res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
   if (res < 0)            /* audio data not ready, return a NULL frame */
      return f;

   o->readpos += res;
   if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */
      return f;

   if (o->mute)
      return f;

   o->readpos = AST_FRIENDLY_OFFSET;   /* reset read pointer for next frame */
   if (c->_state != AST_STATE_UP)   /* drop data if frame is not up */
      return f;
   /* ok we can build and deliver the frame to the caller */
   f->frametype = AST_FRAME_VOICE;
   f->subclass = AST_FORMAT_SLINEAR;
   f->samples = FRAME_SIZE;
   f->datalen = FRAME_SIZE * 2;
   f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET;
   if (o->boost != BOOST_SCALE) {   /* scale and clip values */
      int i, x;
      int16_t *p = (int16_t *) f->data;
      for (i = 0; i < f->samples; i++) {
         x = (p[i] * o->boost) / BOOST_SCALE;
         if (x > 32767)
            x = 32767;
         else if (x < -32768)
            x = -32768;
         p[i] = x;
      }
   }

   f->offset = AST_FRIENDLY_OFFSET;
   return f;
}

static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
   struct chan_oss_pvt *o = newchan->tech_pvt;
   o->owner = newchan;
   return 0;
}

static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
{
   struct chan_oss_pvt *o = c->tech_pvt;
   int res = -1;

   switch (cond) {
   case AST_CONTROL_BUSY:
   case AST_CONTROL_CONGESTION:
   case AST_CONTROL_RINGING:
         res = cond;
         break;
         
   case -1:
      o->cursound = -1;
      o->nosound = 0;      /* when cursound is -1 nosound must be 0 */
      return 0;
      
   case AST_CONTROL_VIDUPDATE:
      res = -1;
      break;
   case AST_CONTROL_HOLD:
      ast_verbose(" << Console Has Been Placed on Hold >> \n");
      ast_moh_start(c, data, o->mohinterpret);
         break;
   case AST_CONTROL_UNHOLD:
      ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
      ast_moh_stop(c);
      break;
   case AST_CONTROL_SRCUPDATE:
      break;
   default: