asterisk/channels/chan_alsa.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * By Matthew Fredrickson <creslin@digium.com>
 *
 * 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 ALSA sound card channel driver 
 *
 * \author Matthew Fredrickson <creslin@digium.com>
 *
 * \par See also
 * \arg Config_alsa
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>asound</depend>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

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

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>

#include "asterisk/frame.h"
#include "asterisk/logger.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 "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"

#ifdef ALSA_MONITOR
#include "alsa-monitor.h"
#endif

/*! 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;

#define DEBUG 0
/* Which device to use */
#define ALSA_INDEV "hw:0,0"
#define ALSA_OUTDEV "hw:0,0"
#define DESIRED_RATE 8000

/* Lets use 160 sample frames, just like GSM.  */
#define FRAME_SIZE 160
#define PERIOD_FRAMES 80 /* 80 Frames, at 2 bytes each */

/* When you set the frame size, you have to come up with
   the right buffer format as well. */
/* 5 64-byte frames = one frame */
#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);

/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600

#if __BYTE_ORDER == __LITTLE_ENDIAN
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
#else
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
#endif

/* static int block = O_NONBLOCK; */
static char indevname[50] = ALSA_INDEV;
static char outdevname[50] = ALSA_OUTDEV;

#if 0
static struct timeval lasttime;
#endif

static int usecnt;
static int silencesuppression = 0;
static int silencethreshold = 1000;

AST_MUTEX_DEFINE_STATIC(usecnt_lock);
AST_MUTEX_DEFINE_STATIC(alsalock);

static const char desc[] = "ALSA Console Channel Driver";
static const char tdesc[] = "ALSA Console Channel Driver";
static const char config[] = "alsa.conf";

static char context[AST_MAX_CONTEXT] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";

static int hookstate=0;

static short silence[FRAME_SIZE] = {0, };

struct sound {
	int ind;
	short *data;
	int datalen;
	int samplen;
	int silencelen;
	int repeat;
};

static struct sound sounds[] = {
	{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
	{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
	{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
	{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
	{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
};

/* Sound command pipe */
static int sndcmd[2];

static struct chan_alsa_pvt {
	/* We only have one ALSA structure -- near sighted perhaps, but it
	   keeps this driver as simple as possible -- as it should be. */
	struct ast_channel *owner;
	char exten[AST_MAX_EXTENSION];
	char context[AST_MAX_CONTEXT];
#if 0
	snd_pcm_t *card;
#endif
	snd_pcm_t *icard, *ocard;
	
} alsa;

/* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
   with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
   usually plenty. */

pthread_t sthread;

#define MAX_BUFFER_SIZE 100

/* File descriptors for sound device */
static int readdev = -1;
static int writedev = -1;

static int autoanswer = 1;

static int cursound = -1;
static int sampsent = 0;
static int silencelen=0;
static int offset=0;
static int nosound=0;

/* ZZ */
static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause);
static int alsa_digit(struct ast_channel *c, char digit);
static int alsa_text(struct ast_channel *c, const char *text);
static int alsa_hangup(struct ast_channel *c);
static int alsa_answer(struct ast_channel *c);
static struct ast_frame *alsa_read(struct ast_channel *chan);
static int alsa_call(struct ast_channel *c, char *dest, int timeout);
static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);

static const struct ast_channel_tech alsa_tech = {
	.type = "Console",
	.description = tdesc,
	.capabilities = AST_FORMAT_SLINEAR,
	.requester = alsa_request,
	.send_digit = alsa_digit,
	.send_text = alsa_text,
	.hangup = alsa_hangup,
	.answer = alsa_answer,
	.read = alsa_read,
	.call = alsa_call,
	.write = alsa_write,
	.indicate = alsa_indicate,
	.fixup = alsa_fixup,
};

static int send_sound(void)
{
	short myframe[FRAME_SIZE];
	int total = FRAME_SIZE;
	short *frame = NULL;
	int amt=0;
	int res;
	int myoff;
	snd_pcm_state_t state;

	if (cursound > -1) {
		res = total;
		if (sampsent < sounds[cursound].samplen) {
			myoff=0;
			while(total) {
				amt = total;
				if (amt > (sounds[cursound].datalen - offset)) 
					amt = sounds[cursound].datalen - offset;
				memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
				total -= amt;
				offset += amt;
				sampsent += amt;
				myoff += amt;
				if (offset >= sounds[cursound].datalen)
					offset = 0;
			}
			/* Set it up for silence */
			if (sampsent >= sounds[cursound].samplen) 
				silencelen = sounds[cursound].silencelen;
			frame = myframe;
		} else {
			if (silencelen > 0) {
				frame = silence;
				silencelen -= res;
			} else {
				if (sounds[cursound].repeat) {
					/* Start over */
					sampsent = 0;
					offset = 0;
				} else {
					cursound = -1;
					nosound = 0;
				}
			return 0;
			}
		}
		
		if (res == 0 || !frame) {
			return 0;
		}
#ifdef ALSA_MONITOR
		alsa_monitor_write((char *)frame, res * 2);
#endif		
		state = snd_pcm_state(alsa.ocard);
		if (state == SND_PCM_STATE_XRUN) {
			snd_pcm_prepare(alsa.ocard);
		}
		res = snd_pcm_writei(alsa.ocard, frame, res);
		if (res > 0)
			return 0;
		return 0;
	}
	return 0;
}

static void *sound_thread(void *unused)
{
	fd_set rfds;
	fd_set wfds;
	int max;
	int res;
	for(;;) {
		FD_ZERO(&rfds);
		FD_ZERO(&wfds);
		max = sndcmd[0];
		FD_SET(sndcmd[0], &rfds);
		if (cursound > -1) {
			FD_SET(writedev, &wfds);
			if (writedev > max)
				max = writedev;
		}
#ifdef ALSA_MONITOR
		if (!alsa.owner) {
			FD_SET(readdev, &rfds);
			if (readdev > max)
				max = readdev;
		}
#endif
		res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
		if (res < 1) {
			ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
			continue;
		}
#ifdef ALSA_MONITOR
		if (FD_ISSET(readdev, &rfds)) {
			/* Keep the pipe going with read audio */
			snd_pcm_state_t state;
			short buf[FRAME_SIZE];
			int r;
			
			state = snd_pcm_state(alsa.ocard);
			if (state == SND_PCM_STATE_XRUN) {
				snd_pcm_prepare(alsa.ocard);
			}
			r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE);
			if (r == -EPIPE) {
#if DEBUG
				ast_log(LOG_ERROR, "XRUN read\n");
#endif
				snd_pcm_prepare(alsa.icard);
			} else if (r == -ESTRPIPE) {
				ast_log(LOG_ERROR, "-ESTRPIPE\n");
				snd_pcm_prepare(alsa.icard);
			} else if (r < 0) {
				ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
			} else
				alsa_monitor_read((char *)buf, r * 2);
		}		
#endif		
		if (FD_ISSET(sndcmd[0], &rfds)) {
			read(sndcmd[0], &cursound, sizeof(cursound));
			silencelen = 0;
			offset = 0;
			sampsent = 0;
		}
		if (FD_ISSET(writedev, &wfds))
			if (send_sound())
				ast_log(LOG_WARNING, "Failed to write sound\n");
	}
	/* Never reached */
	return NULL;
}

static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
{
	int err;
	int direction;
	snd_pcm_t *handle = NULL;
	snd_pcm_hw_params_t *hwparams = NULL;
	snd_pcm_sw_params_t *swparams = NULL;
	struct pollfd pfd;
	snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
	/* int period_bytes = 0; */
	snd_pcm_uframes_t buffer_size = 0;

	unsigned int rate = DESIRED_RATE;
#if 0
	unsigned int per_min = 1;
#endif
	/* unsigned int per_max = 8; */
	snd_pcm_uframes_t start_threshold, stop_threshold;

	err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK);
	if (err < 0) {
		ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
		return NULL;
	} else {
		ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
	}

	snd_pcm_hw_params_alloca(&hwparams);
	snd_pcm_hw_params_any(handle, hwparams);

	err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (err < 0) {
		ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
	}

	err = snd_pcm_hw_params_set_format(handle, hwparams, format);
	if (err < 0) {
		ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
	}

	err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
	if (err < 0) {
		ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
	}

	direction = 0;
	err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
	if (rate != DESIRED_RATE) {
		ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
	}

	direction = 0;
	err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
	if (err < 0) {
		ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
	} else {
		ast_log(LOG_DEBUG, "Period size is %d\n", err);
	}

	buffer_size = 4096 * 2; /* period_size * 16; */
	err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
	if (err < 0) {
		ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
	} else {
		ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err);
	}

#if 0
	direction = 0;
	err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction);
	if (err < 0) {
		ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err));
	}

	err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0);
	if (err < 0) {
		ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err));
	}
#endif

	err = snd_pcm_hw_params(handle, hwparams);
	if (err < 0) {
		ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
	}

	snd_pcm_sw_params_alloca(&swparams);
	snd_pcm_sw_params_current(handle, swparams);

#if 1
	if (stream == SND_PCM_STREAM_PLAYBACK) {
		start_threshold = period_size;
	} else {
		start_threshold = 1;
	}

	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
	if (err < 0) {
		ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
	}
#endif

#if 1
	if (stream == SND_PCM_STREAM_PLAYBACK) {
		stop_threshold = buffer_size;
	} else {
		stop_threshold = buffer_size;
	}
	err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
	if (err < 0) {
		ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
	}
#endif
#if 0
	err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES);
	if (err < 0) {
		ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err));
	}
#endif

#if 0
	err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold);
	if (err < 0) {
		ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err));
	}
#endif
	err = snd_pcm_sw_params(handle, swparams);
	if (err < 0) {
		ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
	}

	err = snd_pcm_poll_descriptors_count(handle);
	if (err <= 0) {
		ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
	}

	if (err != 1) {
		ast_log(LOG_DEBUG, "Can't handle more than one device\n");
	}

	snd_pcm_poll_descriptors(handle, &pfd, err);
	ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd);

	if (stream == SND_PCM_STREAM_CAPTURE)
		readdev = pfd.fd;
	else
		writedev = pfd.fd;

	return handle;
}

static int soundcard_init(void)
{
	alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
	alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);

	if (!alsa.icard || !alsa.ocard) {
		ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n");
		return -1;
	}

	return readdev;
}

static int alsa_digit(struct ast_channel *c, char digit)
{
	ast_mutex_lock(&alsalock);
	ast_verbose( " << Console Received digit %c >> \n", digit);
	ast_mutex_unlock(&alsalock);
	return 0;
}

static int alsa_text(struct ast_channel *c, const char *text)
{
	ast_mutex_lock(&alsalock);
	ast_verbose( " << Console Received text %s >> \n", text);
	ast_mutex_unlock(&alsalock);
	return 0;
}

static void grab_owner(void)
{
	while(alsa.owner && ast_mutex_trylock(&alsa.owner->lock)) {
		ast_mutex_unlock(&alsalock);
		usleep(1);
		ast_mutex_lock(&alsalock);
	}
}

static int alsa_call(struct ast_channel *c, char *dest, int timeout)
{
	int res = 3;
	struct ast_frame f = { AST_FRAME_CONTROL };
	ast_mutex_lock(&alsalock);
	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
	if (autoanswer) {
		ast_verbose( " << Auto-answered >> \n" );
		grab_owner();
		if (alsa.owner) {
			f.subclass = AST_CONTROL_ANSWER;
			ast_queue_frame(alsa.owner, &f);
			ast_mutex_unlock(&alsa.owner->lock);
		}
	} else {
		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
		grab_owner();
		if (alsa.owner) {
			f.subclass = AST_CONTROL_RINGING;
			ast_queue_frame(alsa.owner, &f);
			ast_mutex_unlock(&alsa.owner->lock);
		}
		write(sndcmd[1], &res, sizeof(res));
	}
	snd_pcm_prepare(alsa.icard);
	snd_pcm_start(alsa.icard);
	ast_mutex_unlock(&alsalock);
	return 0;
}

static void answer_sound(void)
{
	int res;
	nosound = 1;
	res = 4;
	write(sndcmd[1], &res, sizeof(res));
	
}

static int alsa_answer(struct ast_channel *c)
{
	ast_mutex_lock(&alsalock);
	ast_verbose( " << Console call has been answered >> \n");
	answer_sound();
	ast_setstate(c, AST_STATE_UP);
	cursound = -1;
	snd_pcm_prepare(alsa.icard);
	snd_pcm_start(alsa.icard);
	ast_mutex_unlock(&alsalock);
	return 0;
}

static int alsa_hangup(struct ast_channel *c)
{
	int res;
	ast_mutex_lock(&alsalock);
	cursound = -1;
	c->tech_pvt = NULL;
	alsa.owner = NULL;
	ast_verbose( " << Hangup on console >> \n");
	ast_mutex_lock(&usecnt_lock);
	usecnt--;
	ast_mutex_unlock(&usecnt_lock);
	if (hookstate) {
		if (autoanswer) {
			hookstate = 0;
		} else {
			/* Congestion noise */
			res = 2;
			write(sndcmd[1], &res, sizeof(res));
			hookstate = 0;
		}
	}
	snd_pcm_drop(alsa.icard);
	ast_mutex_unlock(&alsalock);
	return 0;
}

static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
{
	static char sizbuf[8000];
	static int sizpos = 0;
	int len = sizpos;
	int pos;
	int res = 0;
	/* size_t frames = 0; */
	snd_pcm_state_t state;
	/* Immediately return if no sound is enabled */
	if (nosound)
		return 0;
	ast_mutex_lock(&alsalock);
	/* Stop any currently playing sound */
	if (cursound != -1) {
		snd_pcm_drop(alsa.ocard);
		snd_pcm_prepare(alsa.ocard);
		cursound = -1;
	}
	

	/* We have to digest the frame in 160-byte portions */
	if (f->datalen > sizeof(sizbuf) - sizpos) {
		ast_log(LOG_WARNING, "Frame too large\n");
		res = -1;
	} else {
		memcpy(sizbuf + sizpos, f->data, f->datalen);
		len += f->datalen;
		pos = 0;
#ifdef ALSA_MONITOR
		alsa_monitor_write(sizbuf, len);
#endif
		state = snd_pcm_state(alsa.ocard);
		if (state == SND_PCM_STATE_XRUN) {
			snd_pcm_prepare(alsa.ocard);
		}
		res = snd_pcm_writei(alsa.ocard, sizbuf, len/2);
		if (res == -EPIPE) {
#if DEBUG
			ast_log(LOG_DEBUG, "XRUN write\n");
#endif
			snd_pcm_prepare(alsa.ocard);
			res = snd_pcm_writei(alsa.ocard, sizbuf, len/2);
			if (res != len/2) {
				ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
				res = -1;
			} else if (res < 0) {
				ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
				res = -1;
			}
		} else {
			if (res == -ESTRPIPE) {
				ast_log(LOG_ERROR, "You've got some big problems\n");
			} else if (res < 0)
				ast_log(LOG_NOTICE, "Error %d on write\n", res);
		}
	}
	ast_mutex_unlock(&alsalock);
	if (res > 0)
		res = 0;
	return res;
}


static struct ast_frame *alsa_read(struct ast_channel *chan)
{
	static struct ast_frame f;
	static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET/2];
	short *buf;
	static int readpos = 0;
	static int left = FRAME_SIZE;
	snd_pcm_state_t state;
	int r = 0;
	int off = 0;

	ast_mutex_lock(&alsalock);
	/* Acknowledge any pending cmd */	
	f.frametype = AST_FRAME_NULL;
	f.subclass = 0;
	f.samples = 0;
	f.datalen = 0;
	f.data = NULL;
	f.offset = 0;
	f.src = "Console";
	f.mallocd = 0;
	f.delivery.tv_sec = 0;
	f.delivery.tv_usec = 0;

	state = snd_pcm_state(alsa.icard);
	if ((state != SND_PCM_STATE_PREPARED) && 
	    (state != SND_PCM_STATE_RUNNING)) {
		snd_pcm_prepare(alsa.icard);
	}

	buf = __buf + AST_FRIENDLY_OFFSET/2;

	r = snd_pcm_readi(alsa.icard, buf + readpos, left);
	if (r == -EPIPE) {
#if DEBUG
		ast_log(LOG_ERROR, "XRUN read\n");
#endif
		snd_pcm_prepare(alsa.icard);
	} else if (r == -ESTRPIPE) {
		ast_log(LOG_ERROR, "-ESTRPIPE\n");
		snd_pcm_prepare(alsa.icard);
	} else if (r < 0) {
		ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
	} else if (r >= 0) {
		off -= r;
	}
	/* Update positions */
	readpos += r;
	left -= r;

	if (readpos >= FRAME_SIZE) {
		/* A real frame */
		readpos = 0;
		left = FRAME_SIZE;
		if (chan->_state != AST_STATE_UP) {
			/* Don't transmit unless it's up */
			ast_mutex_unlock(&alsalock);
			return &f;
		}
		f.frametype = AST_FRAME_VOICE;
		f.subclass = AST_FORMAT_SLINEAR;
		f.samples = FRAME_SIZE;
		f.datalen = FRAME_SIZE * 2;
		f.data = buf;
		f.offset = AST_FRIENDLY_OFFSET;
		f.src = "Console";
		f.mallocd = 0;
#ifdef ALSA_MONITOR
		alsa_monitor_read((char *)buf, FRAME_SIZE * 2);
#endif		

	}
	ast_mutex_unlock(&alsalock);
	return &f;
}

static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
	struct chan_alsa_pvt *p = newchan->tech_pvt;
	ast_mutex_lock(&alsalock);
	p->owner = newchan;
	ast_mutex_unlock(&alsalock);
	return 0;
}

static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
{
	int res = 0;
	ast_mutex_lock(&alsalock);
	switch(cond) {
	case AST_CONTROL_BUSY:
		res = 1;
		break;
	case AST_CONTROL_CONGESTION:
		res = 2;
		break;
	case AST_CONTROL_RINGING:
		res = 0;
		break;
	case -1:
		res = -1;
		break;
	case AST_CONTROL_VIDUPDATE:
		res = -1;
		break;
	default:
		ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
		res = -1;
	}
	if (res > -1) {
		write(sndcmd[1], &res, sizeof(res));
	}
	ast_mutex_unlock(&alsalock);
	return res;	
}

static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state)
{
	struct ast_channel *tmp;
	tmp = ast_channel_alloc(1);
	if (tmp) {
		tmp->tech = &alsa_tech;
		ast_string_field_build(tmp, name, "ALSA/%s", indevname);
		tmp->fds[0] = readdev;
		tmp->nativeformats = AST_FORMAT_SLINEAR;
		tmp->readformat = AST_FORMAT_SLINEAR;
		tmp->writeformat = AST_FORMAT_SLINEAR;
		tmp->tech_pvt = p;
		if (!ast_strlen_zero(p->context))
			ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
		if (!ast_strlen_zero(p->exten))
			ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
		if (!ast_strlen_zero(language))
			ast_string_field_set(tmp, language, language);
		p->owner = tmp;
		ast_setstate(tmp, state);
		ast_mutex_lock(&usecnt_lock);
		usecnt++;
		ast_mutex_unlock(&usecnt_lock);
		ast_update_use_count();
		if (state != AST_STATE_DOWN) {
			if (ast_pbx_start(tmp)) {
				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
				ast_hangup(tmp);
				tmp = NULL;
			}
		}
		if (tmp)
			ast_jb_configure(tmp, &global_jbconf);
	}
	return tmp;
}

static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause)
{
	int oldformat = format;
	struct ast_channel *tmp=NULL;
	format &= AST_FORMAT_SLINEAR;
	if (!format) {
		ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
		return NULL;
	}
	ast_mutex_lock(&alsalock);
	if (alsa.owner) {
		ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
		*cause = AST_CAUSE_BUSY;
	} else {
		tmp= alsa_new(&alsa, AST_STATE_DOWN);
		if (!tmp) {
			ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
		}
	}
	ast_mutex_unlock(&alsalock);
	return tmp;
}

static int console_autoanswer(int fd, int argc, char *argv[])
{
	int res = RESULT_SUCCESS;;
	if ((argc != 1) && (argc != 2))
		return RESULT_SHOWUSAGE;
	ast_mutex_lock(&alsalock);
	if (argc == 1) {
		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
	} else {
		if (!strcasecmp(argv[1], "on"))
			autoanswer = -1;
		else if (!strcasecmp(argv[1], "off"))
			autoanswer = 0;
		else
			res = RESULT_SHOWUSAGE;
	}
	ast_mutex_unlock(&alsalock);
	return res;
}

static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
{
#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
	switch(state) {
	case 0:
		if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
			return ast_strdup("on");
	case 1:
		if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
			return ast_strdup("off");
	default:
		return NULL;
	}
	return NULL;
}

static const char autoanswer_usage[] =
"Usage: 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 'alsa.conf'.\n";

static int console_answer(int fd, int argc, char *argv[])
{
	int res = RESULT_SUCCESS;
	if (argc != 1)
		return RESULT_SHOWUSAGE;
	ast_mutex_lock(&alsalock);
	if (!alsa.owner) {
		ast_cli(fd, "No one is calling us\n");
		res = RESULT_FAILURE;
	} else {
		hookstate = 1;
		cursound = -1;
		grab_owner();
		if (alsa.owner) {
			struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
			ast_queue_frame(alsa.owner, &f);
			ast_mutex_unlock(&alsa.owner->lock);
		}
		answer_sound();
	}
	snd_pcm_prepare(alsa.icard);
	snd_pcm_start(alsa.icard);
	ast_mutex_unlock(&alsalock);
	return RESULT_SUCCESS;
}

static char sendtext_usage[] =
"Usage: send text <message>\n"
"       Sends a text message for display on the remote terminal.\n";

static int console_sendtext(int fd, int argc, char *argv[])
{
	int tmparg = 2;
	int res = RESULT_SUCCESS;
	if (argc < 2)
		return RESULT_SHOWUSAGE;
	ast_mutex_lock(&alsalock);
	if (!alsa.owner) {
		ast_cli(fd, "No one is calling us\n");
		res = RESULT_FAILURE;
	} else {
		struct ast_frame f = { AST_FRAME_TEXT, 0 };
		char text2send[256] = "";
		text2send[0] = '\0';
		while(tmparg < argc) {
			strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
			strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
		}
		text2send[strlen(text2send) - 1] = '\n';
		f.data = text2send;
		f.datalen = strlen(text2send) + 1;
		grab_owner();
		if (alsa.owner) {
			ast_queue_frame(alsa.owner, &f);
			f.frametype = AST_FRAME_CONTROL;
			f.subclass = AST_CONTROL_ANSWER;
			f.data = NULL;
			f.datalen = 0;
			ast_queue_frame(alsa.owner, &f);
			ast_mutex_unlock(&alsa.owner->lock);
		}
	}
	ast_mutex_unlock(&alsalock);
	return res;
}

static char answer_usage[] =
"Usage: answer\n"
"       Answers an incoming call on the console (ALSA) channel.\n";

static int console_hangup(int fd, int argc, char *argv[])
{
	int res = RESULT_SUCCESS;
	if (argc != 1)
		return RESULT_SHOWUSAGE;
	cursound = -1;
	ast_mutex_lock(&alsalock);
	if (!alsa.owner && !hookstate) {
		ast_cli(fd, "No call to hangup up\n");
		res = RESULT_FAILURE;
	} else {
		hookstate = 0;
		grab_owner();
		if (alsa.owner) {
			ast_queue_hangup(alsa.owner);
			ast_mutex_unlock(&alsa.owner->lock);
		}
	}
	ast_mutex_unlock(&alsalock);
	return res;
}

static char hangup_usage[] =
"Usage: hangup\n"
"       Hangs up any call currently placed on the console.\n";


static int console_dial(int fd, int argc, char *argv[])
{
	char tmp[256], *tmp2;
	char *mye, *myc;
	char *d;
	int res = RESULT_SUCCESS;
	if ((argc != 1) && (argc != 2))
		return RESULT_SHOWUSAGE;
	ast_mutex_lock(&alsalock);
	if (alsa.owner) {
		if (argc == 2) {
			d = argv[1];
			grab_owner();
			if (alsa.owner) {
				struct ast_frame f = { AST_FRAME_DTMF };
				while(*d) {
					f.subclass = *d;
					ast_queue_frame(alsa.owner, &f);
					d++;
				}
				ast_mutex_unlock(&alsa.owner->lock);
			}
		} else {
			ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
			res = RESULT_FAILURE;
		}
	} else {
		mye = exten;
		myc = context;
		if (argc == 2) {
			char *stringp=NULL;
			strncpy(tmp, argv[1], sizeof(tmp)-1);
			stringp=tmp;
			strsep(&stringp, "@");
			tmp2 = strsep(&stringp, "@");
			if (!ast_strlen_zero(tmp))
				mye = tmp;
			if (!ast_strlen_zero(tmp2))
				myc = tmp2;
		}
		if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
			strncpy(alsa.exten, mye, sizeof(alsa.exten)-1);
			strncpy(alsa.context, myc, sizeof(alsa.context)-1);
			hookstate = 1;
			alsa_new(&alsa, AST_STATE_RINGING);
		} else
			ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
	}
	ast_mutex_unlock(&alsalock);
	return res;
}

static char dial_usage[] =
"Usage: dial [extension[@context]]\n"
"       Dials a given extension (and context if specified)\n";


static struct ast_cli_entry myclis[] = {
	{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
	{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
	{ { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
};

static int load_module(void *mod)
{
	int res;
	int x;
	struct ast_config *cfg;
	struct ast_variable *v;

	/* Copy the default jb config over global_jbconf */
	memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));

	if ((cfg = ast_config_load(config))) {
		v = ast_variable_browse(cfg, "general");
		while(v) {
			/* handle jb conf */
			if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
				v = v->next;
				continue;
			}
			if (!strcasecmp(v->name, "autoanswer"))
				autoanswer = ast_true(v->value);
			else if (!strcasecmp(v->name, "silencesuppression"))
				silencesuppression = ast_true(v->value);
			else if (!strcasecmp(v->name, "silencethreshold"))
				silencethreshold = atoi(v->value);
			else if (!strcasecmp(v->name, "context"))
				strncpy(context, v->value, sizeof(context)-1);
			else if (!strcasecmp(v->name, "language"))
				strncpy(language, v->value, sizeof(language)-1);
			else if (!strcasecmp(v->name, "extension"))
				strncpy(exten, v->value, sizeof(exten)-1);
			else if (!strcasecmp(v->name, "input_device"))
				strncpy(indevname, v->value, sizeof(indevname)-1);
			else if (!strcasecmp(v->name, "output_device"))
				strncpy(outdevname, v->value, sizeof(outdevname)-1);
			v=v->next;
		}
		ast_config_destroy(cfg);
	}
	res = pipe(sndcmd);
	if (res) {
		ast_log(LOG_ERROR, "Unable to create pipe\n");
		return -1;
	}
	res = soundcard_init();
	if (res < 0) {
		if (option_verbose > 1) {
			ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
			ast_verbose(VERBOSE_PREFIX_2 "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
		}
		return 0;
	}

	res = ast_channel_register(&alsa_tech);
	if (res < 0) {
		ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
		return -1;
	}
	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
		ast_cli_register(myclis + x);
	ast_pthread_create(&sthread, NULL, sound_thread, NULL);
#ifdef ALSA_MONITOR
	if (alsa_monitor_start()) {
		ast_log(LOG_ERROR, "Problem starting Monitoring\n");
	}
#endif	 
	return 0;
}

static int unload_module(void *mod)
{
	int x;
	
	ast_channel_unregister(&alsa_tech);
	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
		ast_cli_unregister(myclis + x);
	if (alsa.icard)
		snd_pcm_close(alsa.icard);
	if (alsa.ocard)
		snd_pcm_close(alsa.ocard);
	if (sndcmd[0] > 0) {
		close(sndcmd[0]);
		close(sndcmd[1]);
	}
	if (alsa.owner)
		ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
	if (alsa.owner)
		return -1;
	return 0;
}

static const char *description(void)
{
	return (char *) desc;
}

static const char *key(void)
{
	return ASTERISK_GPL_KEY;
}

STD_MOD(MOD_0, NULL, NULL, NULL);