Version 0.1.11 from FTP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@416 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1113 insertions, 0 deletions

diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
new file mode 100755
index 000000000..56fca5c2d
--- /dev/null
+++ b/channels/chan_alsa.c
@@ -0,0 +1,1113 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Copyright (C) 2002, Linux Support Services
+ *
+ * By Matthew Fredrickson <creslin@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/options.h>
+#include <asterisk/pbx.h>
+#include <asterisk/config.h>
+#include <asterisk/cli.h>
+#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>
+#include <alsa/asoundlib.h>
+#include "busy.h"
+#include "ringtone.h"
+#include "ring10.h"
+#include "answer.h"
+
+#ifdef ALSA_MONITOR
+#include "alsa-monitor.h"
+#endif
+
+#define DEBUG 0
+/* Which device to use */
+#define ALSA_INDEV "default"
+#define ALSA_OUTDEV "default"
+#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
+
+static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
+//static int block = O_NONBLOCK;
+static char indevname[50] = ALSA_INDEV;
+static char outdevname[50] = ALSA_OUTDEV;
+
+static struct timeval lasttime;
+
+static int usecnt;
+static int needanswer = 0;
+static int needringing = 0;
+static int needhangup = 0;
+static int silencesuppression = 0;
+static int silencethreshold = 1000;
+
+static char digits[80] = "";
+static char text2send[80] = "";
+
+static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static char *type = "Console";
+static char *desc = "ALSA Console Channel Driver";
+static char *tdesc = "ALSA Console Channel Driver";
+static char *config = "alsa.conf";
+
+static char context[AST_MAX_EXTENSION] = "default";
+static char language[MAX_LANGUAGE] = "";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+/* Command pipe */
+static int cmd[2];
+
+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_EXTENSION];
+#if 0
+	snd_pcm_t *card;
+#endif
+	snd_pcm_t *icard, *ocard;
+	
+} alsa;
+
+static int time_has_passed()
+{
+	struct timeval tv;
+	int ms;
+	gettimeofday(&tv, NULL);
+	ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
+			(tv.tv_usec - lasttime.tv_usec) / 1000;
+	if (ms > MIN_SWITCH_TIME)
+		return -1;
+	return 0;
+}
+
+/* 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
+//static int buffersize = 3;
+
+//static int full_duplex = 0;
+
+/* Are we reading or writing (simulated full duplex) */
+//static int readmode = 1;
+
+/* File descriptors for sound device */
+static int readdev = -1;
+static int writedev = -1;
+
+static int autoanswer = 1;
+ 
+static int calc_loudness(short *frame)
+{
+	int sum = 0;
+	int x;
+	for (x=0;x<FRAME_SIZE;x++) {
+		if (frame[x] < 0)
+			sum -= frame[x];
+		else
+			sum += frame[x];
+	}
+	sum = sum/FRAME_SIZE;
+	return sum;
+}
+
+static int cursound = -1;
+static int sampsent = 0;
+static int silencelen=0;
+static int offset=0;
+static int nosound=0;
+
+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 = 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;
+	snd_pcm_t *handle = NULL;
+	snd_pcm_hw_params_t *hwparams = NULL;
+	snd_pcm_sw_params_t *swparams = NULL;
+	struct pollfd pfd;
+	int period_size = PERIOD_FRAMES * 4;
+	//int period_bytes = 0;
+	int buffer_size = 0;
+	
+	unsigned int rate = DESIRED_RATE;
+	unsigned int per_min = 1;
+	//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));
+	}
+
+	rate = snd_pcm_hw_params_set_rate_near(handle, hwparams, rate, 0);
+
+	if (rate != DESIRED_RATE) {
+		ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
+	}
+
+	err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, period_size, 0);
+	if (err < 0) {
+		ast_log(LOG_ERROR, "period_size(%d frames) is bad: %s\n", period_size, snd_strerror(err));
+	} else {
+		ast_log(LOG_DEBUG, "Period size is %d\n", err);
+	}
+	period_size = 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 %d: %s\n", buffer_size, snd_strerror(err));
+	} else {
+		ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err);
+	}
+	buffer_size = err;
+
+	err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, 0);
+	if (err < 0) {
+		ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err));
+	}
+
+#if 0
+	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
+
+	err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, buffer_size);
+	if (err < 0) {
+		ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err));
+	}
+
+	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()
+{
+	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_verbose( " << Console Received digit %c >> \n", digit);
+	return 0;
+}
+
+static int alsa_text(struct ast_channel *c, char *text)
+{
+	ast_verbose( " << Console Received text %s >> \n", text);
+	return 0;
+}
+
+static int alsa_call(struct ast_channel *c, char *dest, int timeout)
+{
+	int res = 3;
+	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+	if (autoanswer) {
+		ast_verbose( " << Auto-answered >> \n" );
+		needanswer = 1;
+	} else {
+		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+		needringing = 1;
+		write(sndcmd[1], &res, sizeof(res));
+	}
+	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_verbose( " << Console call has been answered >> \n");
+	answer_sound();
+	c->state = AST_STATE_UP;
+	cursound = -1;
+	return 0;
+}
+
+static int alsa_hangup(struct ast_channel *c)
+{
+	int res;
+	cursound = -1;
+	c->pvt->pvt = NULL;
+	alsa.owner = NULL;
+	ast_verbose( " << Hangup on console >> \n");
+	ast_pthread_mutex_lock(&usecnt_lock);
+	usecnt--;
+	ast_pthread_mutex_unlock(&usecnt_lock);
+	needhangup = 0;
+	needanswer = 0;
+	if (hookstate) {
+		res = 2;
+		write(sndcmd[1], &res, sizeof(res));
+	}
+	return 0;
+}
+
+#if 0
+static int soundcard_writeframe(short *data)
+{	
+	/* Write an exactly FRAME_SIZE sized of frame */
+	static int bufcnt = 0;
+	static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
+	struct audio_buf_info info;
+	int res;
+	int fd = sounddev;
+	static int warned=0;
+	if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
+		if (!warned)
+			ast_log(LOG_WARNING, "Error reading output space\n");
+		bufcnt = buffersize;
+		warned++;
+	}
+	if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
+		/* We've run out of stuff, buffer again */
+		bufcnt = 0;
+	}
+	if (bufcnt == buffersize) {
+		/* Write sample immediately */
+		res = write(fd, ((void *)data), FRAME_SIZE * 2);
+	} else {
+		/* Copy the data into our buffer */
+		res = FRAME_SIZE * 2;
+		memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
+		bufcnt++;
+		if (bufcnt == buffersize) {
+			res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
+		}
+	}
+	return res;
+}
+#endif
+
+static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
+{
+	int res;
+	static char sizbuf[8000];
+	static int sizpos = 0;
+	int len = sizpos;
+	int pos;
+	//size_t frames = 0;
+	snd_pcm_state_t state;
+	/* Immediately return if no sound is enabled */
+	if (nosound)
+		return 0;
+	/* 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");
+		return -1;
+	}
+	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));
+			return -1;
+		} else if (res < 0) {
+			ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
+			return -1;
+		}
+	} else {
+		if (res == -ESTRPIPE) {
+			ast_log(LOG_ERROR, "You've got some big problems\n");
+		}
+	}
+
+	return 0;
+}
+
+
+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;
+	int res;
+	int b;
+	int nonull=0;
+	snd_pcm_state_t state;
+	int r = 0;
+	int off = 0;
+
+	/* Acknowledge any pending cmd */
+	res = read(cmd[0], &b, sizeof(b));
+	if (res > 0)
+		nonull = 1;
+	
+	f.frametype = AST_FRAME_NULL;
+	f.subclass = 0;
+	f.timelen = 0;
+	f.datalen = 0;
+	f.data = NULL;
+	f.offset = 0;
+	f.src = type;
+	f.mallocd = 0;
+	
+	if (needringing) {
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_RINGING;
+		needringing = 0;
+		return &f;
+	}
+	
+	if (needhangup) {
+		needhangup = 0;
+		return NULL;
+	}
+	if (strlen(text2send)) {
+		f.frametype = AST_FRAME_TEXT;
+		f.subclass = 0;
+		f.data = text2send;
+		f.datalen = strlen(text2send);
+		strcpy(text2send,"");
+		return &f;
+	}
+	if (strlen(digits)) {
+		f.frametype = AST_FRAME_DTMF;
+		f.subclass = digits[0];
+		for (res=0;res<strlen(digits);res++)
+			digits[res] = digits[res + 1];
+		return &f;
+	}
+	
+	if (needanswer) {
+		needanswer = 0;
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_ANSWER;
+		chan->state = AST_STATE_UP;
+		return &f;
+	}
+	
+	if (nonull)
+		return &f;
+		
+	
+	state = snd_pcm_state(alsa.ocard);
+	if (state == SND_PCM_STATE_XRUN) {
+		snd_pcm_prepare(alsa.ocard);
+	}
+
+	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));
+		return NULL;
+	} 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 */
+			return &f;
+		}
+		f.frametype = AST_FRAME_VOICE;
+		f.subclass = AST_FORMAT_SLINEAR;
+		f.timelen = FRAME_SIZE / 8;
+		f.datalen = FRAME_SIZE * 2;
+		f.data = buf;
+		f.offset = AST_FRIENDLY_OFFSET;
+		f.src = type;
+		f.mallocd = 0;
+#ifdef ALSA_MONITOR
+		alsa_monitor_read((char *)buf, FRAME_SIZE * 2);
+#endif		
+
+#if 0
+		{ static int fd = -1;
+		  if (fd < 0)
+		  	fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT);
+		  write(fd, f.data, f.datalen);
+		}
+#endif		
+	}
+	return &f;
+}
+
+static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+	struct chan_alsa_pvt *p = newchan->pvt->pvt;
+	p->owner = newchan;
+	return 0;
+}
+
+static int alsa_indicate(struct ast_channel *chan, int cond)
+{
+	int res;
+	switch(cond) {
+	case AST_CONTROL_BUSY:
+		res = 1;
+		break;
+	case AST_CONTROL_CONGESTION:
+		res = 2;
+		break;
+	case AST_CONTROL_RINGING:
+		res = 0;
+		break;
+	default:
+		ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
+		return -1;
+	}
+	if (res > -1) {
+		write(sndcmd[1], &res, sizeof(res));
+	}
+	return 0;	
+}
+
+static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state)
+{
+	struct ast_channel *tmp;
+	tmp = ast_channel_alloc();
+	if (tmp) {
+		snprintf(tmp->name, sizeof(tmp->name), "ALSA/%s", indevname);
+		tmp->type = type;
+		tmp->fds[0] = readdev;
+		tmp->fds[1] = cmd[0];
+		tmp->nativeformats = AST_FORMAT_SLINEAR;
+		tmp->pvt->pvt = p;
+		tmp->pvt->send_digit = alsa_digit;
+		tmp->pvt->send_text = alsa_text;
+		tmp->pvt->hangup = alsa_hangup;
+		tmp->pvt->answer = alsa_answer;
+		tmp->pvt->read = alsa_read;
+		tmp->pvt->call = alsa_call;
+		tmp->pvt->write = alsa_write;
+		tmp->pvt->indicate = alsa_indicate;
+		tmp->pvt->fixup = alsa_fixup;
+		if (strlen(p->context))
+			strncpy(tmp->context, p->context, sizeof(tmp->context)-1);
+		if (strlen(p->exten))
+			strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1);
+		if (strlen(language))
+			strncpy(tmp->language, language, sizeof(tmp->language)-1);
+		p->owner = tmp;
+		tmp->state = state;
+		ast_pthread_mutex_lock(&usecnt_lock);
+		usecnt++;
+		ast_pthread_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;
+			}
+		}
+	}
+	return tmp;
+}
+
+static struct ast_channel *alsa_request(char *type, int format, void *data)
+{
+	int oldformat = format;
+	struct ast_channel *tmp;
+	format &= AST_FORMAT_SLINEAR;
+	if (!format) {
+		ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
+		return NULL;
+	}
+	if (alsa.owner) {
+		ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
+		return NULL;
+	}
+	tmp= alsa_new(&alsa, AST_STATE_DOWN);
+	if (!tmp) {
+		ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
+	}
+	return tmp;
+}
+
+static int console_autoanswer(int fd, int argc, char *argv[])
+{
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (argc == 1) {
+		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+		return RESULT_SUCCESS;
+	} else {
+		if (!strcasecmp(argv[1], "on"))
+			autoanswer = -1;
+		else if (!strcasecmp(argv[1], "off"))
+			autoanswer = 0;
+		else
+			return RESULT_SHOWUSAGE;
+	}
+	return RESULT_SUCCESS;
+}
+
+static char *autoanswer_complete(char *line, char *word, int pos, int state)
+{
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+	switch(state) {
+	case 0:
+		if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+			return strdup("on");
+	case 1:
+		if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+			return strdup("off");
+	default:
+		return NULL;
+	}
+	return NULL;
+}
+
+static 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[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!alsa.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	hookstate = 1;
+	cursound = -1;
+	needanswer++;
+	answer_sound();
+	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 = 1;
+	if (argc < 1)
+		return RESULT_SHOWUSAGE;
+	if (!alsa.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	if (strlen(text2send))
+		ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
+	strcpy(text2send, "");
+	while(tmparg <= argc) {
+		strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send));
+		strncat(text2send, " ", sizeof(text2send) - strlen(text2send));
+	}
+	needanswer++;
+	return RESULT_SUCCESS;
+}
+
+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[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	cursound = -1;
+	if (!alsa.owner && !hookstate) {
+		ast_cli(fd, "No call to hangup up\n");
+		return RESULT_FAILURE;
+	}
+	hookstate = 0;
+	if (alsa.owner)
+		needhangup++;
+	return RESULT_SUCCESS;
+}
+
+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;
+	int b = 0;
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (alsa.owner) {
+		if (argc == 2) {
+			strncat(digits, argv[1], sizeof(digits) - strlen(digits));
+			/* Wake up the polling thread */
+			write(cmd[1], &b, sizeof(b));
+		} else {
+			ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
+			return RESULT_FAILURE;
+		}
+		return RESULT_SUCCESS;
+	}
+	mye = exten;
+	myc = context;
+	if (argc == 2) {
+		strncpy(tmp, argv[1], sizeof(tmp)-1);
+		strtok(tmp, "@");
+		tmp2 = strtok(NULL, "@");
+		if (strlen(tmp))
+			mye = tmp;
+		if (tmp2 && strlen(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_UP);
+	} else
+		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
+	return RESULT_SUCCESS;
+}
+
+static char dial_usage[] =
+"Usage: dial [extension[@context]]\n"
+"       Dials a given extensison (";
+
+
+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 }
+};
+
+int load_module()
+{
+	int res;
+	int x;
+	int flags;
+	struct ast_config *cfg = ast_load(config);
+	struct ast_variable *v;
+	res = pipe(cmd);
+	res = pipe(sndcmd);
+	if (res) {
+		ast_log(LOG_ERROR, "Unable to create pipe\n");
+		return -1;
+	}
+	flags = fcntl(cmd[0], F_GETFL);
+	fcntl(cmd[0], F_SETFL, flags | O_NONBLOCK);
+	flags = fcntl(cmd[1], F_GETFL);
+	fcntl(cmd[1], F_SETFL, flags | O_NONBLOCK);
+	res = soundcard_init();
+	if (res < 0) {
+		close(cmd[1]);
+		close(cmd[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;
+	}
+#if 0
+	if (!full_duplex)
+		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+#endif
+	res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, alsa_request);
+	if (res < 0) {
+		ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
+		return -1;
+	}
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_register(myclis + x);
+	if (cfg) {
+		v = ast_variable_browse(cfg, "general");
+		while(v) {
+			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_destroy(cfg);
+	}
+	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;
+}
+
+
+
+int unload_module()
+{
+	int x;
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_unregister(myclis + x);
+	close(readdev);
+	close(writedev);
+	if (cmd[0] > 0) {
+		close(cmd[0]);
+		close(cmd[1]);
+	}
+	if (sndcmd[0] > 0) {
+		close(sndcmd[0]);
+		close(sndcmd[1]);
+	}
+	if (alsa.owner)
+		ast_softhangup(alsa.owner);
+	if (alsa.owner)
+		return -1;
+	return 0;
+}
+
+char *description()
+{
+	return desc;
+}
+
+int usecount()
+{
+	int res;
+	ast_pthread_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	ast_pthread_mutex_unlock(&usecnt_lock);
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}