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asterisk/channels/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__, "$Revision$")
#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"
#include "asterisk/app.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 */
};
/*!
* \brief 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,
};
/*!
* \brief 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;
}
/* !
* \brief 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 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;
}
/*!
* \brief 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);
}
/*!
* \brief 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 */
l = MIN(l, FRAME_SIZE - ofs); /* don't overflow the frame */
l = MIN(l, s->datalen - start); /* don't overflow the source */
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) {
l = MIN(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;
struct timeval *to = NULL, t;
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) {
/*
* We would like to use select here, but the device
* is always writable, so this would become busy wait.
* So we rather set a timeout to 1/2 of the frame size.
*/
t.tv_sec = 0;
t.tv_usec = (1000000 * FRAME_SIZE) / (5 * DEFAULT_SAMPLE_RATE);
to = &t;
}
}
/* ast_select emulates linux behaviour in terms of timeout handling */
res = ast_select(maxfd + 1, &rfds, &wfds, NULL, to);
if (res < 0) {
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 (to != NULL) /* maybe it is possible to write */
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)
ast_channel_set_fd(o->owner, 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)) {
ast_verb(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;
}
/*! \brief Play ringtone 'x' on device 'o' */
static void ring(struct chan_oss_pvt *o, int x)
{
write(o->sndcmd[1], &x, sizeof(x));
}
/*!
* \brief 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_DECLARE_APP_ARGS(args,
AST_APP_ARG(name);
AST_APP_ARG(flags);
);
char *parse = ast_strdupa(dest);
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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 (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "answer") == 0) {
f.frametype = AST_FRAME_CONTROL;
f.subclass = AST_CONTROL_ANSWER;
ast_queue_frame(c, &f);
} else if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "noanswer") == 0) {
f.frametype = AST_FRAME_CONTROL;
f.subclass = AST_CONTROL_RINGING;
ast_queue_frame(c, &f);
ring(o, AST_CONTROL_RING);
} else 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;
}
/*!
* \brief 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;
}
/*! \brief 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;
default:
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name);
return -1;
}
if (res > -1)
ring(o, res);
return 0;
}
/*!
* \brief allocate a new channel.
*/
static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state)
{
struct ast_channel *c;
c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, 0, "OSS/%s", o->device + 5);
if (c == NULL)
return NULL;
c->tech = &oss_tech;
if (o->sounddev < 0)
setformat(o, O_RDWR);
ast_channel_set_fd(c, 0, o->sounddev); /* -1 if device closed, override later */
c->nativeformats = AST_FORMAT_SLINEAR;
c->readformat = AST_FORMAT_SLINEAR;
c->writeformat = AST_FORMAT_SLINEAR;
c->tech_pvt = o;
if (!ast_strlen_zero(o->language))
ast_string_field_set(c, language, o->language);
/* Don't use ast_set_callerid() here because it will
* generate a needless NewCallerID event */
c->cid.cid_ani = ast_strdup(o->cid_num);
if (!ast_strlen_zero(ext))
c->cid.cid_dnid = ast_strdup(ext);
o->owner = c;
ast_module_ref(ast_module_info->self);
ast_jb_configure(c, &global_jbconf);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(c)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
ast_hangup(c);
o->owner = c = NULL;
/* XXX what about the channel itself ? */
}
}
return c;
}
static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
{
struct ast_channel *c;
struct chan_oss_pvt *o;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(name);
AST_APP_ARG(flags);
);
char *parse = ast_strdupa(data);
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
o = find_desc(args.name);
ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, (char *) data);
if (o == NULL) {
ast_log(LOG_NOTICE, "Device %s not found\n", args.name);
/* XXX we could default to 'dsp' perhaps ? */
return NULL;
}
if ((format & AST_FORMAT_SLINEAR) == 0) {
ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format);
return NULL;
}
if (o->owner) {
ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
*cause = AST_CAUSE_BUSY;
return NULL;
}
c = oss_new(o, NULL, NULL, AST_STATE_DOWN);
if (c == NULL) {
ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
return NULL;
}
return c;
}
static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
switch (cmd) {
case CLI_INIT:
e->command = "console autoanswer [on|off]";
e->usage =
"Usage: console autoanswer [on|off]\n"
" Enables or disables autoanswer feature. If used without\n"
" argument, displays the current on/off status of autoanswer.\n"
" The default value of autoanswer is in 'oss.conf'.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == e->args - 1) {
ast_cli(a->fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
return CLI_SUCCESS;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (o == NULL) {
ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
oss_active);
return CLI_FAILURE;
}
if (!strcasecmp(a->argv[e->args-1], "on"))
o->autoanswer = 1;
else if (!strcasecmp(a->argv[e->args - 1], "off"))
o->autoanswer = 0;
else
return CLI_SHOWUSAGE;
return CLI_SUCCESS;
}
/*!
* \brief answer command from the console
*/
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
struct chan_oss_pvt *o = find_desc(oss_active);
switch (cmd) {
case CLI_INIT:
e->command = "console answer";
e->usage =
"Usage: console answer\n"
" Answers an incoming call on the console (OSS) channel.\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!o->owner) {
ast_cli(a->fd, "No one is calling us\n");
return CLI_FAILURE;
}
o->hookstate = 1;
o->cursound = -1;
o->nosound = 0;
ast_queue_frame(o->owner, &f);
return CLI_SUCCESS;
}
/*!
* \brief Console send text CLI command
*
* \note concatenate all arguments into a single string. argv is NULL-terminated
* so we can use it right away
*/
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
char buf[TEXT_SIZE];
if (cmd == CLI_INIT) {
e->command = "console send text";
e->usage =
"Usage: console send text <message>\n"
" Sends a text message for display on the remote terminal.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc < e->args + 1)
return CLI_SHOWUSAGE;
if (!o->owner) {
ast_cli(a->fd, "Not in a call\n");
return CLI_FAILURE;
}
ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
if (!ast_strlen_zero(buf)) {
struct ast_frame f = { 0, };
int i = strlen(buf);
buf[i] = '\n';
f.frametype = AST_FRAME_TEXT;
f.subclass = 0;
f.data = buf;
f.datalen = i + 1;
ast_queue_frame(o->owner, &f);
}
return CLI_SUCCESS;
}
static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console hangup";
e->usage =
"Usage: console hangup\n"
" Hangs up any call currently placed on the console.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
o->cursound = -1;
o->nosound = 0;
if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
ast_cli(a->fd, "No call to hang up\n");
return CLI_FAILURE;
}
o->hookstate = 0;
if (o->owner)
ast_queue_hangup(o->owner);
setformat(o, O_CLOSE);
return CLI_SUCCESS;
}
static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console flash";
e->usage =
"Usage: console flash\n"
" Flashes the call currently placed on the console.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
o->cursound = -1;
o->nosound = 0; /* when cursound is -1 nosound must be 0 */
if (!o->owner) { /* XXX maybe !o->hookstate too ? */
ast_cli(a->fd, "No call to flash\n");
return CLI_FAILURE;
}
o->hookstate = 0;
if (o->owner) /* XXX must be true, right ? */
ast_queue_frame(o->owner, &f);
return CLI_SUCCESS;
}
static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char *s = NULL, *mye = NULL, *myc = NULL;
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console dial";
e->usage =
"Usage: console dial [extension[@context]]\n"
" Dials a given extension (and context if specified)\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc > e->args + 1)
return CLI_SHOWUSAGE;
if (o->owner) { /* already in a call */
int i;
struct ast_frame f = { AST_FRAME_DTMF, 0 };
if (a->argc == e->args) { /* argument is mandatory here */
ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
return CLI_FAILURE;
}
s = a->argv[e->args];
/* send the string one char at a time */
for (i = 0; i < strlen(s); i++) {
f.subclass = s[i];
ast_queue_frame(o->owner, &f);
}
return CLI_SUCCESS;
}
/* if we have an argument split it into extension and context */
if (a->argc == e->args + 1)
s = ast_ext_ctx(a->argv[e->args], &mye, &myc);
/* supply default values if needed */
if (mye == NULL)
mye = o->ext;
if (myc == NULL)
myc = o->ctx;
if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
o->hookstate = 1;
oss_new(o, mye, myc, AST_STATE_RINGING);
} else
ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
if (s)
ast_free(s);
return CLI_SUCCESS;
}
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
char *s;
if (cmd == CLI_INIT) {
e->command = "console {mute|unmute}";
e->usage =
"Usage: console {mute|unmute}\n"
" Mute/unmute the microphone.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
s = a->argv[e->args-1];
if (!strcasecmp(s, "mute"))
o->mute = 1;
else if (!strcasecmp(s, "unmute"))
o->mute = 0;
else
return CLI_SHOWUSAGE;
return CLI_SUCCESS;
}
static int console_transfer(int fd, int argc, char *argv[])
{
struct chan_oss_pvt *o = find_desc(oss_active);
struct ast_channel *b = NULL;
char *tmp, *ext, *ctx;
if (argc != 3)
return RESULT_SHOWUSAGE;
if (o == NULL)
return RESULT_FAILURE;
if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
ast_cli(fd, "There is no call to transfer\n");
return RESULT_SUCCESS;
}
tmp = ast_ext_ctx(argv[2], &ext, &ctx);
if (ctx == NULL) /* supply default context if needed */
ctx = o->owner->context;
if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
ast_cli(fd, "No such extension exists\n");
else {
ast_cli(fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx);
if (ast_async_goto(b, ctx, ext, 1))
ast_cli(fd, "Failed to transfer :(\n");
}
if (tmp)
ast_free(tmp);
return RESULT_SUCCESS;
}
static const char transfer_usage[] =
"Usage: console transfer <extension>[@context]\n"
" Transfers the currently connected call to the given extension (and\n"
"context if specified)\n";
static int console_active(int fd, int argc, char *argv[])
{
if (argc == 2)
ast_cli(fd, "active console is [%s]\n", oss_active);
else if (argc != 3)
return RESULT_SHOWUSAGE;
else {
struct chan_oss_pvt *o;
if (strcmp(argv[2], "show") == 0) {
for (o = oss_default.next; o; o = o->next)
ast_cli(fd, "device [%s] exists\n", o->name);
return RESULT_SUCCESS;
}
o = find_desc(argv[2]);
if (o == NULL)
ast_cli(fd, "No device [%s] exists\n", argv[2]);
else
oss_active = o->name;
}
return RESULT_SUCCESS;
}
static const char active_usage[] =
"Usage: console active [device]\n"
" If used without a parameter, displays which device is the current\n"
"console. If a device is specified, the console sound device is changed to\n"
"the device specified.\n";
/*!
* \brief store the boost factor
*/
static void store_boost(struct chan_oss_pvt *o, char *s)
{
double boost = 0;
if (sscanf(s, "%lf", &boost) != 1) {
ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
return;
}
if (boost < -BOOST_MAX) {
ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
boost = -BOOST_MAX;
} else if (boost > BOOST_MAX) {
ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
boost = BOOST_MAX;
}
boost = exp(log(10) * boost / 20) * BOOST_SCALE;
o->boost = boost;
ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
}
static int do_boost(int fd, int argc, char *argv[])
{
struct chan_oss_pvt *o = find_desc(oss_active);
if (argc == 2)
ast_cli(fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
else if (argc == 3)
store_boost(o, argv[2]);
return RESULT_SUCCESS;
}
static struct ast_cli_entry cli_oss[] = {
NEW_CLI(console_answer, "Answer an incoming console call"),
NEW_CLI(console_hangup, "Hangup a call on the console"),
NEW_CLI(console_flash, "Flash a call on the console"),
NEW_CLI(console_dial, "Dial an extension on the console"),
NEW_CLI(console_mute, "Disable/Enable mic input"),
{ { "console", "transfer", NULL },
console_transfer, "Transfer a call to a different extension",
transfer_usage },
NEW_CLI(console_sendtext, "Send text to the remote device"),
NEW_CLI(console_autoanswer, "Sets/displays autoanswer"),
{ { "console", "boost", NULL },
do_boost, "Sets/displays mic boost in dB",
NULL },
{ { "console", "active", NULL },
console_active, "Sets/displays active console",
active_usage },
};
/*!
* store the mixer argument from the config file, filtering possibly
* invalid or dangerous values (the string is used as argument for
* system("mixer %s")
*/
static void store_mixer(struct chan_oss_pvt *o, char *s)
{
int i;
for (i = 0; i < strlen(s); i++) {
if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) {
ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
return;
}
}
if (o->mixer_cmd)
ast_free(o->mixer_cmd);
o->mixer_cmd = ast_strdup(s);
ast_log(LOG_WARNING, "setting mixer %s\n", s);
}
/*!
* store the callerid components
*/
static void store_callerid(struct chan_oss_pvt *o, char *s)
{
ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
}
/*!
* grab fields from the config file, init the descriptor and open the device.
*/
static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
{
struct ast_variable *v;
struct chan_oss_pvt *o;
if (ctg == NULL) {
o = &oss_default;
ctg = "general";
} else {
if (!(o = ast_calloc(1, sizeof(*o))))
return NULL;
*o = oss_default;
/* "general" is also the default thing */
if (strcmp(ctg, "general") == 0) {
o->name = ast_strdup("dsp");
oss_active = o->name;
goto openit;
}
o->name = ast_strdup(ctg);
}
strcpy(o->mohinterpret, "default");
o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */
/* fill other fields from configuration */
for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
M_START(v->name, v->value);
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
M_BOOL("autoanswer", o->autoanswer)
M_BOOL("autohangup", o->autohangup)
M_BOOL("overridecontext", o->overridecontext)
M_STR("device", o->device)
M_UINT("frags", o->frags)
M_UINT("debug", oss_debug)
M_UINT("queuesize", o->queuesize)
M_STR("context", o->ctx)
M_STR("language", o->language)
M_STR("mohinterpret", o->mohinterpret)
M_STR("extension", o->ext)
M_F("mixer", store_mixer(o, v->value))
M_F("callerid", store_callerid(o, v->value))
M_F("boost", store_boost(o, v->value))
M_END(/* */);
}
if (ast_strlen_zero(o->device))
ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
if (o->mixer_cmd) {
char *cmd;
asprintf(&cmd, "mixer %s", o->mixer_cmd);
ast_log(LOG_WARNING, "running [%s]\n", cmd);
system(cmd);
ast_free(cmd);
}
if (o == &oss_default) /* we are done with the default */
return NULL;
openit:
#ifdef TRYOPEN
if (setformat(o, O_RDWR) < 0) { /* open device */
ast_verb(1, "Device %s not detected\n", ctg);
ast_verb(1, "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
goto error;
}
if (o->duplex != M_FULL)
ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
#endif /* TRYOPEN */
if (pipe(o->sndcmd) != 0) {
ast_log(LOG_ERROR, "Unable to create pipe\n");
goto error;
}
ast_pthread_create_background(&o->sthread, NULL, sound_thread, o);
/* link into list of devices */
if (o != &oss_default) {
o->next = oss_default.next;
oss_default.next = o;
}
return o;
error:
if (o != &oss_default)
ast_free(o);
return NULL;
}
static int load_module(void)
{
struct ast_config *cfg = NULL;
char *ctg = NULL;
struct ast_flags config_flags = { 0 };
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
/* load config file */
if (!(cfg = ast_config_load(config, config_flags))) {
ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
return AST_MODULE_LOAD_DECLINE;
}
do {
store_config(cfg, ctg);
} while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
ast_config_destroy(cfg);
if (find_desc(oss_active) == NULL) {
ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
/* XXX we could default to 'dsp' perhaps ? */
/* XXX should cleanup allocated memory etc. */
return AST_MODULE_LOAD_FAILURE;
}
if (ast_channel_register(&oss_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
struct chan_oss_pvt *o;
ast_channel_unregister(&oss_tech);
ast_cli_unregister_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
for (o = oss_default.next; o; o = o->next) {
close(o->sounddev);
if (o->sndcmd[0] > 0) {
close(o->sndcmd[0]);
close(o->sndcmd[1]);
}
if (o->owner)
ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
if (o->owner) /* XXX how ??? */
return -1;
/* XXX what about the thread ? */
/* XXX what about the memory allocated ? */
}
return 0;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");
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