diff --git a/channels/chan_oss.c b/channels/chan_oss.c
index 8b61abf87b012f1cb78fd50f6f4b4722594d0bf5..b8f0fb6efd8a77be4c97d5fe20b2285cf42c1e96 100755
--- a/channels/chan_oss.c
+++ b/channels/chan_oss.c
@@ -1,28 +1,27 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
- * Use /dev/dsp as a channel, and the console to command it :).
- *
- * The full-duplex "simulation" is pretty weak. This is generally a
- * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
- * writing a driver.
- *
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
+ *
+ * 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
*/
+#include <stdio.h>
+#include <ctype.h> /* for isalnum */
+#include <string.h>
#include <unistd.h>
-#include <fcntl.h>
-#include <errno.h>
#include <sys/ioctl.h>
+#include <fcntl.h>
#include <sys/time.h>
-#include <string.h>
#include <stdlib.h>
-#include <stdio.h>
+#include <errno.h>
+
#ifdef __linux
#include <linux/soundcard.h>
@@ -44,16 +43,132 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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"
+/* ringtones we use */
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"
+/*
+ * 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
+
+[general]
+; general config options, default values are shown
+; all but debug can go also in the device-specific sections.
+; debug=0x0 ; misc debug flags, default is 0
+
+[card1]
+; autoanswer = no ; no autoanswer on call
+; autohangup = yes ; hangup when other party closes
+; extension=s ; default extension to call
+; context=default ; default context
+; language="" ; default language
+; overridecontext=no ; the whole dial string is considered an extension.
+ ; if yes, the last @ will start the context
+
+; device=/dev/dsp ; device to open
+; mixer="-f /dev/mixer0 pcm 80 ; mixer command to run on start
+; queuesize=10 ; frames in device driver
+; frags=8 ; argument to SETFRAGMENT
+
+.. 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"
@@ -61,42 +176,29 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define DEV_DSP "/dev/dsp"
#endif
-/* Lets use 160 sample frames, just like GSM. */
-#define FRAME_SIZE 160
-
-/* 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
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+#ifndef MAX
+#define MAX(a,b) ((a) > (b) ? (a) : (b))
+#endif
-static struct timeval lasttime;
static int usecnt;
-static int silencesuppression = 0;
-static int silencethreshold = 1000;
-static int playbackonly = 0;
-
-
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-static const char type[] = "Console";
-static const char desc[] = "OSS Console Channel Driver";
-static const char tdesc[] = "OSS Console Channel Driver";
-static const char config[] = "oss.conf";
-
-static char context[AST_MAX_CONTEXT] = "default";
-static char language[MAX_LANGUAGE] = "";
-static char exten[AST_MAX_EXTENSION] = "s";
+static char *config = "oss.conf"; /* default config file */
-static int hookstate=0;
-
-static short silence[FRAME_SIZE] = {0, };
+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;
@@ -105,25 +207,99 @@ struct sound {
};
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 },
+ { 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 */
};
-/* Sound command pipe */
-static int sndcmd[2];
-static struct chan_oss_pvt {
- /* We only have one OSS structure -- near sighted perhaps, but it
- keeps this driver as simple as possible -- as it should be. */
+/*
+ * 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 *type; /* XXX maybe take the one from oss_tech */
+ 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;
+ char device[64]; /* device to open */
+
+ pthread_t sthread;
+
struct ast_channel *owner;
- char exten[AST_MAX_EXTENSION];
- char context[AST_MAX_CONTEXT];
-} oss;
+ char ext[AST_MAX_EXTENSION];
+ char ctx[AST_MAX_CONTEXT];
+ char language[MAX_LANGUAGE];
+
+ /* 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 = {
+ .type = "Console",
+ .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 },
+};
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
+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(struct ast_channel *c, char digit);
static int oss_text(struct ast_channel *c, const char *text);
static int oss_hangup(struct ast_channel *c);
@@ -135,704 +311,649 @@ static int oss_indicate(struct ast_channel *chan, int cond);
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static const struct ast_channel_tech oss_tech = {
- .type = type,
- .description = tdesc,
- .capabilities = AST_FORMAT_SLINEAR,
- .requester = oss_request,
- .send_digit = oss_digit,
- .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,
+ .type = "Console",
+ .description = "OSS Console Channel Driver",
+ .capabilities = AST_FORMAT_SLINEAR,
+ .requester = oss_request,
+ .send_digit = oss_digit,
+ .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,
};
-static int time_has_passed(void)
+/*
+ * returns a pointer to the descriptor with the given name
+ */
+static struct chan_oss_pvt *find_desc(char *dev)
{
- 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. */
+ struct chan_oss_pvt *o;
-static pthread_t sthread;
+ for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next)
+ ;
+ if (o == NULL)
+ ast_log(LOG_WARNING, "could not find <%s>\n", dev);
+ return o;
+}
-#define MAX_BUFFER_SIZE 100
-static int buffersize = 3;
+/*
+ * split a string in extension-context, returns pointers to malloc'ed
+ * strings.
+ * If we have 'overridecontext' then the last @ is considered as
+ * a context separator, and the context is overridden.
+ * This is usually not very necessary as you can play with the dialplan,
+ * and it is nice not to need it because you have '@' in SIP addresses.
+ * Return value is the buffer address.
+ */
+static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
+{
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
+ if (ext == NULL || ctx == NULL)
+ return NULL; /* error */
+ *ext = *ctx = NULL;
+ if (src && *src != '\0')
+ *ext = strdup(src);
+ if (*ext == NULL)
+ return NULL;
+ if (!o->overridecontext) {
+ /* parse from the right */
+ *ctx = strrchr(*ext, '@');
+ if (*ctx)
+ *(*ctx)++ = '\0';
+ }
+ return *ext;
+}
-static int full_duplex = 0;
+/*
+ * 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;
-/* Are we reading or writing (simulated full duplex) */
-static int readmode = 1;
+ 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;
+}
-/* File descriptor for sound device */
-static int sounddev = -1;
+/* Write an exactly FRAME_SIZE sized frame */
+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
+{
+ int res;
-static int autoanswer = 1;
-
-#if 0
-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];
+ 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;
}
- sum = sum/FRAME_SIZE;
- return sum;
+ o->w_errors = 0;
+ return write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
}
-#endif
-
-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)
+/*
+ * 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 total = FRAME_SIZE;
- short *frame = NULL;
- int amt=0;
- int res;
- int myoff;
- audio_buf_info abi;
- if (cursound > -1) {
- res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
- if (res) {
- ast_log(LOG_WARNING, "Unable to read output space\n");
- return -1;
- }
- /* Calculate how many samples we can send, max */
- if (total > (abi.fragments * abi.fragsize / 2))
- total = abi.fragments * abi.fragsize / 2;
- 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;
+ int ofs, l, start;
+ int l_sampsent = o->sampsent;
+ struct sound *s;
+
+ if (o->cursound < 0) /* no sound to send */
+ return;
+ s = &sounds[o->cursound];
+ for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
+ l = s->samplen - l_sampsent; /* # of available samples */
+ if (l > 0) {
+ start = l_sampsent % s->datalen; /* source offset */
+ if (l > FRAME_SIZE - ofs) /* don't overflow the frame */
+ l = FRAME_SIZE - ofs;
+ if (l > s->datalen - start) /* don't overflow the source */
+ l = s->datalen - start;
+ bcopy(s->data + start, myframe + ofs, l*2);
+ if (0)
+ ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n",
+ l_sampsent, l, s->samplen, ofs);
+ l_sampsent += l;
+ } else { /* end of samples, maybe some silence */
+ static const short silence[FRAME_SIZE] = {0, };
+
+ l += s->silencelen;
+ if (l > 0) {
+ if (l > FRAME_SIZE - ofs)
+ l = FRAME_SIZE - ofs;
+ bcopy(silence, myframe + ofs, l*2);
+ l_sampsent += l;
+ } else { /* silence is over, restart sound if loop */
+ if (s->repeat == 0) { /* last block */
+ o->cursound = -1;
+ o->nosound = 0; /* allow audio data */
+ if (ofs < FRAME_SIZE) /* pad with silence */
+ bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2);
}
+ l_sampsent = 0;
}
}
- if (frame)
- res = write(sounddev, frame, res * 2);
- if (res > 0)
- return 0;
- return res;
}
- return 0;
+ l = soundcard_writeframe(o, myframe);
+ if (l > 0)
+ o->sampsent = l_sampsent; /* update status */
}
-static void *sound_thread(void *unused)
+static void *sound_thread(void *arg)
{
- fd_set rfds;
- fd_set wfds;
- int max;
- int res;
char ign[4096];
- if (read(sounddev, ign, sizeof(sounddev)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
- for(;;) {
+ struct chan_oss_pvt *o = (struct chan_oss_pvt *)arg;
+
+ /*
+ * Just in case, kick the driver by trying to read from it.
+ * Ignore errors - this read is almost guaranteed to fail.
+ */
+ read(o->sounddev, ign, sizeof(ign));
+ for (;;) {
+ fd_set rfds, wfds;
+ int maxfd, res;
+
FD_ZERO(&rfds);
FD_ZERO(&wfds);
- max = sndcmd[0];
- FD_SET(sndcmd[0], &rfds);
- if (!oss.owner) {
- FD_SET(sounddev, &rfds);
- if (sounddev > max)
- max = sounddev;
- }
- if (cursound > -1) {
- FD_SET(sounddev, &wfds);
- if (sounddev > max)
- max = sounddev;
+ FD_SET(o->sndcmd[0], &rfds);
+ maxfd = o->sndcmd[0]; /* pipe from the main process */
+ if (o->cursound > -1 && o->sounddev < 0)
+ setformat(o, O_RDWR); /* need the channel, try to reopen */
+ else if (o->cursound == -1 && o->owner == NULL)
+ setformat(o, O_CLOSE); /* can close */
+ if (o->sounddev > -1) {
+ if (!o->owner) { /* no one owns the audio, so we must drain it */
+ FD_SET(o->sounddev, &rfds);
+ maxfd = MAX(o->sounddev, maxfd);
+ }
+ if (o->cursound > -1) {
+ FD_SET(o->sounddev, &wfds);
+ maxfd = MAX(o->sounddev, maxfd);
+ }
}
- res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+ /* ast_select emulates linux behaviour in terms of timeout handling */
+ res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL);
if (res < 1) {
ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
+ sleep(1);
continue;
}
- if (FD_ISSET(sndcmd[0], &rfds)) {
- read(sndcmd[0], &cursound, sizeof(cursound));
- silencelen = 0;
- offset = 0;
- sampsent = 0;
+ 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 (FD_ISSET(sounddev, &rfds)) {
- /* Ignore read */
- if (read(sounddev, ign, sizeof(ign)) < 0)
- ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+ if (o->sounddev > -1) {
+ if (FD_ISSET(o->sounddev, &rfds)) /* read and ignore errors */
+ read(o->sounddev, ign, sizeof(ign));
+ if (FD_ISSET(o->sounddev, &wfds))
+ send_sound(o);
}
- if (FD_ISSET(sounddev, &wfds))
- if (send_sound())
- ast_log(LOG_WARNING, "Failed to write sound\n");
}
- /* Never reached */
- return NULL;
+ return NULL; /* Never reached */
}
-#if 0
-static int silence_suppress(short *buf)
+/*
+ * 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)
{
-#define SILBUF 3
- int loudness;
- static int silentframes = 0;
- static char silbuf[FRAME_SIZE * 2 * SILBUF];
- static int silbufcnt=0;
- if (!silencesuppression)
+ 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;
- loudness = calc_loudness((short *)(buf));
- if (option_debug)
- ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
- if (loudness < silencethreshold) {
- silentframes++;
- silbufcnt++;
- /* Keep track of the last few bits of silence so we can play
- them as lead-in when the time is right */
- if (silbufcnt >= SILBUF) {
- /* Make way for more buffer */
- memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
- silbufcnt--;
- }
- memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
- if (silentframes > 10) {
- /* We've had plenty of silence, so compress it now */
- return 1;
- }
- } else {
- silentframes=0;
- /* Write any buffered silence we have, it may have something
- important */
- if (silbufcnt) {
- write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
- silbufcnt = 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;
}
- return 0;
-}
-#endif
-
-static int setformat(void)
-{
- int fmt, desired, res, fd = sounddev;
- static int warnedalready = 0;
- static int warnedalready2 = 0;
+ if (o->owner)
+ o->owner->fds[0] = fd;
#if __BYTE_ORDER == __LITTLE_ENDIAN
fmt = AFMT_S16_LE;
#else
fmt = AFMT_S16_BE;
#endif
-
res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
return -1;
}
- res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
-
- /* Check to see if duplex set (FreeBSD Bug)*/
- res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
-
- if ((fmt & DSP_CAP_DUPLEX) && !res) {
- if (option_verbose > 1)
- ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
- full_duplex = -1;
+ switch (mode) {
+ case O_RDWR:
+ res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+ /* Check to see if duplex set (FreeBSD Bug)*/
+ res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
+ if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
+ if (option_verbose > 1)
+ ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+ o->duplex = M_FULL;
+ };
+ break;
+ case O_WRONLY:
+ o->duplex = M_WRITE;
+ break;
+ case O_RDONLY:
+ o->duplex = M_READ;
+ break;
}
+
fmt = 0;
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
return -1;
}
- /* 8000 Hz desired */
- desired = 8000;
- fmt = desired;
+ fmt = desired = 8000; /* 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 (!warnedalready++)
- ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
- }
-#if 1
- fmt = BUFFER_FMT;
- res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
- if (res < 0) {
- if (!warnedalready2++)
- ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
- }
-#endif
- return 0;
-}
-
-static int soundcard_setoutput(int force)
-{
- /* Make sure the soundcard is in output mode. */
- int fd = sounddev;
- if (full_duplex || (!readmode && !force))
- return 0;
- readmode = 0;
- if (force || time_has_passed()) {
- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
- /* Keep the same fd reserved by closing the sound device and copying stdin at the same
- time. */
- /* dup2(0, sound); */
- close(sounddev);
- fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
- if (fd < 0) {
- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
- return -1;
+ 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;
}
- /* dup2 will close the original and make fd be sound */
- if (dup2(fd, sounddev) < 0) {
- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
- return -1;
- }
- if (setformat()) {
- return -1;
- }
- return 0;
}
- return 1;
-}
-
-static int soundcard_setinput(int force)
-{
- int fd = sounddev;
- if (full_duplex || (readmode && !force))
- return 0;
- readmode = -1;
- if (force || time_has_passed()) {
- ioctl(sounddev, SNDCTL_DSP_RESET, 0);
- close(sounddev);
- /* dup2(0, sound); */
- fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
- if (fd < 0) {
- ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
- return -1;
- }
- /* dup2 will close the original and make fd be sound */
- if (dup2(fd, sounddev) < 0) {
- ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
- return -1;
- }
- if (setformat()) {
- return -1;
+ /*
+ * 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;
+ }
}
- return 0;
}
- return 1;
-}
-
-static int soundcard_init(void)
-{
- /* Assume it's full duplex for starters */
- int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
- if (fd < 0) {
- ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
- return fd;
- }
- gettimeofday(&lasttime, NULL);
- sounddev = fd;
- setformat();
- if (!full_duplex)
- soundcard_setinput(1);
- return sounddev;
+ /* 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(struct ast_channel *c, char digit)
{
+ /* no better use for received digits than print them */
ast_verbose( " << Console Received digit %c >> \n", digit);
return 0;
}
static int oss_text(struct ast_channel *c, const char *text)
{
+ /* print received messages */
ast_verbose( " << Console Received text %s >> \n", text);
return 0;
}
+/* Play ringtone 'x' on device 'o' */
+static void ring(struct chan_oss_pvt *o, int x)
+{
+ write(o->sndcmd[1], &x, sizeof(x));
+}
+
+
+/*
+ * handler for incoming calls. Either autoanswer, or start ringing
+ */
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
- int res = 3;
+ struct chan_oss_pvt *o = c->tech_pvt;
struct ast_frame f = { 0, };
- ast_verbose( " << Call placed to '%s' on console >> \n", dest);
- if (autoanswer) {
+
+ ast_verbose(" << Call to '%s' on console from <%s><%s><%s> >>\n",
+ dest, c->cid.cid_dnid, c->cid.cid_num, c->cid.cid_name);
+ if (o->autoanswer) {
ast_verbose( " << Auto-answered >> \n" );
f.frametype = AST_FRAME_CONTROL;
f.subclass = AST_CONTROL_ANSWER;
ast_queue_frame(c, &f);
} else {
- nosound = 1;
- ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+ 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);
- write(sndcmd[1], &res, sizeof(res));
+ ring(o, AST_CONTROL_RING);
}
return 0;
}
-static void answer_sound(void)
-{
- int res;
- nosound = 1;
- res = 4;
- write(sndcmd[1], &res, sizeof(res));
-
-}
-
+/*
+ * 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");
- answer_sound();
+#if 0
+ /* play an answer tone (XXX do we really need it ?) */
+ ring(o, AST_CONTROL_ANSWER);
+#endif
ast_setstate(c, AST_STATE_UP);
- cursound = -1;
- nosound=0;
+ o->cursound = -1;
+ o->nosound=0;
return 0;
}
static int oss_hangup(struct ast_channel *c)
{
- int res = 0;
- cursound = -1;
+ struct chan_oss_pvt *o = c->tech_pvt;
+
+ o->cursound = -1;
+ o->nosound = 0;
c->tech_pvt = NULL;
- oss.owner = NULL;
+ o->owner = NULL;
ast_verbose( " << Hangup on console >> \n");
- ast_mutex_lock(&usecnt_lock);
+ ast_mutex_lock(&usecnt_lock); /* XXX not sure why */
usecnt--;
ast_mutex_unlock(&usecnt_lock);
- if (hookstate) {
- if (autoanswer) {
+ if (o->hookstate) {
+ if (o->autoanswer || o->autohangup) {
/* Assume auto-hangup too */
- hookstate = 0;
+ o->hookstate = 0;
+ setformat(o, O_CLOSE);
} else {
/* Make congestion noise */
- res = 2;
- write(sndcmd[1], &res, sizeof(res));
+ ring(o, AST_CONTROL_CONGESTION);
}
}
return 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;
-}
-
-
-static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+/* used for data coming from the network */
+static int oss_write(struct ast_channel *c, struct ast_frame *f)
{
- int res;
- static char sizbuf[8000];
- static int sizpos = 0;
- int len = sizpos;
- int pos;
+ int src;
+ struct chan_oss_pvt *o = c->tech_pvt;
+
/* Immediately return if no sound is enabled */
- if (nosound)
+ if (o->nosound)
return 0;
/* Stop any currently playing sound */
- cursound = -1;
- if (!full_duplex && !playbackonly) {
- /* If we're half duplex, we have to switch to read mode
- to honor immediate needs if necessary. But if we are in play
- back only mode, then we don't switch because the console
- is only being used one way -- just to playback something. */
- res = soundcard_setinput(1);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set device to input mode\n");
- return -1;
+ 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;
- }
- res = soundcard_setoutput(0);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set output device\n");
- return -1;
- } else if (res > 0) {
- /* The device is still in read mode, and it's too soon to change it,
- so just pretend we wrote it */
- return 0;
- }
- /* 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;
- while(len - pos > FRAME_SIZE * 2) {
- soundcard_writeframe((short *)(sizbuf + pos));
- pos += FRAME_SIZE * 2;
}
- if (len - pos)
- memmove(sizbuf, sizbuf + pos, len - pos);
- sizpos = len - pos;
return 0;
}
-static struct ast_frame *oss_read(struct ast_channel *chan)
+static struct ast_frame *oss_read(struct ast_channel *c)
{
- static struct ast_frame f;
- static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
- static int readpos = 0;
int res;
-
-#if 0
- ast_log(LOG_DEBUG, "oss_read()\n");
-#endif
-
- f.frametype = AST_FRAME_NULL;
- f.subclass = 0;
- f.samples = 0;
- f.datalen = 0;
- f.data = NULL;
- f.offset = 0;
- f.src = type;
- f.mallocd = 0;
- f.delivery.tv_sec = 0;
- f.delivery.tv_usec = 0;
-
- res = soundcard_setinput(0);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set input mode\n");
- return NULL;
- }
- if (res > 0) {
- /* Theoretically shouldn't happen, but anyway, return a NULL frame */
- return &f;
- }
- res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
- if (res < 0) {
- ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
-#if 0
- CRASH;
-#endif
- return NULL;
- }
- readpos += res;
-
- if (readpos >= FRAME_SIZE * 2) {
- /* A real frame */
- readpos = 0;
- 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.samples = FRAME_SIZE;
- f.datalen = FRAME_SIZE * 2;
- f.data = buf + AST_FRIENDLY_OFFSET;
- f.offset = AST_FRIENDLY_OFFSET;
- f.src = type;
- f.mallocd = 0;
- f.delivery.tv_sec = 0;
- f.delivery.tv_usec = 0;
-#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;
+ struct chan_oss_pvt *o = c->tech_pvt;
+ struct ast_frame *f = &o->read_f;
+
+ /* 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 = o->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;
+ f->offset = AST_FRIENDLY_OFFSET;
+ return f;
}
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
- struct chan_oss_pvt *p = newchan->tech_pvt;
- p->owner = newchan;
+ struct chan_oss_pvt *o = newchan->tech_pvt;
+ o->owner = newchan;
return 0;
}
-static int oss_indicate(struct ast_channel *chan, int cond)
+static int oss_indicate(struct ast_channel *c, int cond)
{
+ struct chan_oss_pvt *o = c->tech_pvt;
int res;
+
switch(cond) {
case AST_CONTROL_BUSY:
- res = 1;
- break;
case AST_CONTROL_CONGESTION:
- res = 2;
- break;
case AST_CONTROL_RINGING:
- res = 0;
+ res = cond;
break;
+
case -1:
- cursound = -1;
+ o->cursound = -1;
return 0;
+
default:
- ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
+ ast_log(LOG_WARNING,
+ "Don't know how to display condition %d on %s\n",
+ cond, c->name);
return -1;
}
- if (res > -1) {
- write(sndcmd[1], &res, sizeof(res));
- }
+ if (res > -1)
+ ring(o, res);
return 0;
}
-static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+/*
+ * allocate a new channel.
+ */
+static struct ast_channel *oss_new(struct chan_oss_pvt *o,
+ char *ext, char *ctx, int state)
{
- struct ast_channel *tmp;
- tmp = ast_channel_alloc(1);
- if (tmp) {
- tmp->tech = &oss_tech;
- snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
- tmp->type = type;
- tmp->fds[0] = sounddev;
- tmp->nativeformats = AST_FORMAT_SLINEAR;
- tmp->readformat = AST_FORMAT_SLINEAR;
- tmp->writeformat = AST_FORMAT_SLINEAR;
- tmp->tech_pvt = p;
- 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;
- 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;
- }
+ struct ast_channel *c;
+
+ c = ast_channel_alloc(1);
+ if (c == NULL)
+ return NULL;
+ c->tech = &oss_tech;
+ snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5);
+ c->type = o->type;
+ c->fds[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 (ctx && !ast_strlen_zero(ctx))
+ ast_copy_string(c->context, ctx, sizeof(c->context));
+ if (ext && !ast_strlen_zero(ext))
+ ast_copy_string(c->exten, ext, sizeof(c->exten));
+ if (o->language && !ast_strlen_zero(o->language))
+ ast_copy_string(c->language, o->language, sizeof(c->language));
+
+ o->owner = c;
+ ast_setstate(c, 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(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 ? */
+ /* XXX what about usecnt ? */
}
}
- return tmp;
+ return c;
}
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
+static struct ast_channel *oss_request(const char *type,
+ int format, void *data, int *cause)
{
- 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);
+ struct ast_channel *c;
+ struct chan_oss_pvt *o = find_desc(data);
+
+ 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", (char *)data);
+ /* XXX we could default to 'dsp' perhaps ? */
return NULL;
}
- if (oss.owner) {
- ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+ 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;
}
- tmp= oss_new(&oss, AST_STATE_DOWN);
- if (!tmp) {
+ 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 tmp;
+ return c;
}
static int console_autoanswer(int fd, int argc, char *argv[])
{
- if ((argc != 1) && (argc != 2))
- return RESULT_SHOWUSAGE;
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
if (argc == 1) {
- ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+ ast_cli(fd, "Auto answer is %s.\n", o->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;
}
+ if (argc != 2)
+ return RESULT_SHOWUSAGE;
+ if (o == NULL) {
+ ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
+ oss_active);
+ return RESULT_FAILURE;
+ }
+ if (!strcasecmp(argv[1], "on"))
+ o->autoanswer = -1;
+ else if (!strcasecmp(argv[1], "off"))
+ o->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
+ int l = strlen(word);
+
switch(state) {
case 0:
- if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+ if (l && !strncasecmp(word, "on", MIN(l, 2)))
return strdup("on");
case 1:
- if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+ if (l && !strncasecmp(word, "off", MIN(l, 3)))
return strdup("off");
default:
return NULL;
@@ -846,19 +967,28 @@ static char autoanswer_usage[] =
" argument, displays the current on/off status of autoanswer.\n"
" The default value of autoanswer is in 'oss.conf'.\n";
+/*
+ * answer command from the console
+ */
static int console_answer(int fd, int argc, char *argv[])
{
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
if (argc != 1)
return RESULT_SHOWUSAGE;
- if (!oss.owner) {
+ if (!o->owner) {
ast_cli(fd, "No one is calling us\n");
return RESULT_FAILURE;
}
- hookstate = 1;
- cursound = -1;
- ast_queue_frame(oss.owner, &f);
- answer_sound();
+ o->hookstate = 1;
+ o->cursound = -1;
+ o->nosound = 0;
+ ast_queue_frame(o->owner, &f);
+#if 0
+ /* XXX do we really need it ? considering we shut down immediately... */
+ ring(o, AST_CONTROL_ANSWER);
+#endif
return RESULT_SUCCESS;
}
@@ -866,30 +996,34 @@ static char sendtext_usage[] =
"Usage: send text <message>\n"
" Sends a text message for display on the remote terminal.\n";
+/*
+ * concatenate all arguments into a single string
+ */
static int console_sendtext(int fd, int argc, char *argv[])
{
+ struct chan_oss_pvt *o = find_desc(oss_active);
int tmparg = 2;
- char text2send[256] = "";
+ char text2send[TEXT_SIZE] = "";
struct ast_frame f = { 0, };
+
if (argc < 2)
return RESULT_SHOWUSAGE;
- if (!oss.owner) {
- ast_cli(fd, "No one is calling us\n");
+ if (!o->owner) {
+ ast_cli(fd, "Not in a call\n");
return RESULT_FAILURE;
}
- if (strlen(text2send))
- ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
- text2send[0] = '\0';
- while(tmparg < argc) {
- strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
- strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
+ while (tmparg < argc) {
+ strncat(text2send, argv[tmparg++],
+ sizeof(text2send) - strlen(text2send) - 1);
+ strncat(text2send, " ",
+ sizeof(text2send) - strlen(text2send) - 1);
}
- if (strlen(text2send)) {
+ if (!ast_strlen_zero(text2send)) {
f.frametype = AST_FRAME_TEXT;
f.subclass = 0;
f.data = text2send;
f.datalen = strlen(text2send);
- ast_queue_frame(oss.owner, &f);
+ ast_queue_frame(o->owner, &f);
}
return RESULT_SUCCESS;
}
@@ -900,86 +1034,91 @@ static char answer_usage[] =
static int console_hangup(int fd, int argc, char *argv[])
{
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
if (argc != 1)
return RESULT_SHOWUSAGE;
- cursound = -1;
- if (!oss.owner && !hookstate) {
+ o->cursound = -1;
+ o->nosound = 0;
+ if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
ast_cli(fd, "No call to hangup up\n");
return RESULT_FAILURE;
}
- hookstate = 0;
- if (oss.owner) {
- ast_queue_hangup(oss.owner);
- }
+ o->hookstate = 0;
+ if (o->owner)
+ ast_queue_hangup(o->owner);
+ setformat(o, O_CLOSE);
return RESULT_SUCCESS;
}
+static char hangup_usage[] =
+"Usage: hangup\n"
+" Hangs up any call currently placed on the console.\n";
+
+
static int console_flash(int fd, int argc, char *argv[])
{
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
if (argc != 1)
return RESULT_SHOWUSAGE;
- cursound = -1;
- if (!oss.owner) {
+ o->cursound = -1;
+ if (!o->owner) { /* XXX maybe !o->hookstate too ? */
ast_cli(fd, "No call to flash\n");
return RESULT_FAILURE;
}
- hookstate = 0;
- if (oss.owner) {
- ast_queue_frame(oss.owner, &f);
- }
+ o->hookstate = 0;
+ if (o->owner) /* XXX must be true, right ? */
+ ast_queue_frame(o->owner, &f);
return RESULT_SUCCESS;
}
-static char hangup_usage[] =
-"Usage: hangup\n"
-" Hangs up any call currently placed on the console.\n";
-
static char flash_usage[] =
"Usage: flash\n"
" Flashes the 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 x;
- struct ast_frame f = { AST_FRAME_DTMF, 0 };
- if ((argc != 1) && (argc != 2))
+ char *s = NULL, *mye = NULL, *myc = NULL;
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
+ if (argc != 1 && argc != 2)
return RESULT_SHOWUSAGE;
- if (oss.owner) {
- if (argc == 2) {
- for (x=0;x<strlen(argv[1]);x++) {
- f.subclass = argv[1][x];
- ast_queue_frame(oss.owner, &f);
- }
- } else {
- ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
+ if (o->owner) { /* already in a call */
+ int i;
+ struct ast_frame f = { AST_FRAME_DTMF, 0 };
+
+ if (argc == 1) { /* argument is mandatory here */
+ ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
return RESULT_FAILURE;
}
+ s = argv[1];
+ /* 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 RESULT_SUCCESS;
}
- 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 (strlen(tmp))
- mye = tmp;
- if (tmp2 && strlen(tmp2))
- myc = tmp2;
- }
+ /* if we have an argument split it into extension and context */
+ if (argc == 2)
+ s = ast_ext_ctx(argv[1], &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)) {
- strncpy(oss.exten, mye, sizeof(oss.exten)-1);
- strncpy(oss.context, myc, sizeof(oss.context)-1);
- hookstate = 1;
- oss_new(&oss, AST_STATE_RINGING);
+ o->hookstate = 1;
+ oss_new(o, mye, myc, AST_STATE_RINGING);
} else
ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
+ if (s)
+ free(s);
return RESULT_SUCCESS;
}
@@ -987,31 +1126,60 @@ static char dial_usage[] =
"Usage: dial [extension[@context]]\n"
" Dials a given extensison (and context if specified)\n";
+static char mute_usage[] =
+"Usage: mute\nMutes the microphone\n";
+
+static char unmute_usage[] =
+"Usage: unmute\nUnmutes the microphone\n";
+
+static int console_mute(int fd, int argc, char *argv[])
+{
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
+ if (argc != 1)
+ return RESULT_SHOWUSAGE;
+ o->mute = 1;
+ return RESULT_SUCCESS;
+}
+
+static int console_unmute(int fd, int argc, char *argv[])
+{
+ struct chan_oss_pvt *o = find_desc(oss_active);
+
+ if (argc != 1)
+ return RESULT_SHOWUSAGE;
+ o->mute = 0;
+ return RESULT_SUCCESS;
+}
+
static int console_transfer(int fd, int argc, char *argv[])
{
- char tmp[256];
- char *context;
+ struct chan_oss_pvt *o = find_desc(oss_active);
+ struct ast_channel *b = NULL;
+ char *tmp, *ext, *ctx;
+
if (argc != 2)
return RESULT_SHOWUSAGE;
- if (oss.owner && ast_bridged_channel(oss.owner)) {
- strncpy(tmp, argv[1], sizeof(tmp) - 1);
- context = strchr(tmp, '@');
- if (context) {
- *context = '\0';
- context++;
- } else
- context = oss.owner->context;
- if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
- ast_cli(fd, "Whee, transferring %s to %s@%s.\n",
- ast_bridged_channel(oss.owner)->name, tmp, context);
- if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
- ast_cli(fd, "Failed to transfer :(\n");
- } else {
- ast_cli(fd, "No such extension exists\n");
- }
- } else {
+ 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[1], &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)
+ free(tmp);
return RESULT_SUCCESS;
}
@@ -1020,93 +1188,211 @@ static char transfer_usage[] =
" 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 == 1)
+ ast_cli(fd, "active console is [%s]\n", oss_active);
+ else if (argc != 2)
+ return RESULT_SHOWUSAGE;
+ else {
+ struct chan_oss_pvt *o;
+ if (strcmp(argv[1], "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[1]);
+ if (o == NULL)
+ ast_cli(fd, "No device [%s] exists\n", argv[1]);
+ else
+ oss_active = o->name;
+ }
+ return RESULT_SUCCESS;
+}
+
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 },
{ { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage },
{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
+ { { "mute", NULL }, console_mute, "Disable mic input", mute_usage },
+ { { "unmute", NULL }, console_unmute, "Enable mic input", unmute_usage },
{ { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_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 }
+ { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete },
+ { { "console", NULL }, console_active, "Sets/displays active console",
+ "console foo sets foo as the console"}
};
-int load_module()
+/*
+ * 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 res;
- int x;
- struct ast_config *cfg;
- struct ast_variable *v;
- res = pipe(sndcmd);
- if (res) {
- ast_log(LOG_ERROR, "Unable to create pipe\n");
- return -1;
+ 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;
+ }
}
- 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 OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
+ if (o->mixer_cmd)
+ free(o->mixer_cmd);
+ o->mixer_cmd = strdup(s);
+ ast_log(LOG_WARNING, "setting mixer %s\n", s);
+}
+
+/*
+ * 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 {
+ o = (struct chan_oss_pvt *)malloc(sizeof *o);
+ if (o == NULL) /* fail */
+ return NULL;
+ *o = oss_default;
+ /* "general" is also the default thing */
+ if (strcmp(ctg, "general") == 0) {
+ o->name = strdup("dsp");
+ oss_active = o->name;
+ goto openit;
}
- return 0;
+ o->name = strdup(ctg);
}
- if (!full_duplex)
- ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
- res = ast_channel_register(&oss_tech);
- if (res < 0) {
- ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
- return -1;
+
+ /* fill other fields from configuration */
+ for (v = ast_variable_browse(cfg, ctg);v; v=v->next) {
+ M_START(v->name, v->value);
+
+ 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("extension", o->ext)
+ M_F("mixer", store_mixer(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);
+ free(cmd);
}
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_register(myclis + x);
- if ((cfg = ast_config_load(config))) {
- 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, "playbackonly"))
- playbackonly = ast_true(v->value);
- v=v->next;
+ if (o == &oss_default) /* we are done with the default */
+ return NULL;
+
+openit:
+#if TRYOPEN
+ if (setformat(o, O_RDWR) < 0) { /* open device */
+ if (option_verbose > 0) {
+ ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg);
+ ast_verbose(VERBOSE_PREFIX_2 "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(&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)
+ free(o);
+ return NULL;
+}
+
+int load_module(void)
+{
+ int i;
+ struct ast_config *cfg;
+
+ /* load config file */
+ cfg = ast_config_load(config);
+ if (cfg != NULL) {
+ char *ctg = NULL; /* first pass is 'general' */
+
+ do {
+ store_config(cfg, ctg);
+ } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
ast_config_destroy(cfg);
}
- ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+ 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 -1;
+ }
+ i = ast_channel_register(&oss_tech);
+ if (i < 0) {
+ ast_log(LOG_ERROR, "Unable to register channel class '%s'\n",
+ oss_default.type);
+ /* XXX should cleanup allocated memory etc. */
+ return -1;
+ }
+ ast_cli_register_multiple(myclis, sizeof(myclis)/sizeof(struct ast_cli_entry));
return 0;
}
-
int unload_module()
{
- int x;
+ struct chan_oss_pvt *o;
ast_channel_unregister(&oss_tech);
- for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
- ast_cli_unregister(myclis + x);
- close(sounddev);
- if (sndcmd[0] > 0) {
- close(sndcmd[0]);
- close(sndcmd[1]);
+ ast_cli_unregister_multiple(myclis,
+ sizeof(myclis)/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 ? */
}
- if (oss.owner)
- ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
- if (oss.owner)
- return -1;
return 0;
}
char *description()
{
- return (char *) desc;
+ return (char *)oss_tech.description;
}
int usecount()
diff --git a/channels/chan_oss_old.c b/channels/chan_oss_old.c
new file mode 100755
index 0000000000000000000000000000000000000000..8b61abf87b012f1cb78fd50f6f4b4722594d0bf5
--- /dev/null
+++ b/channels/chan_oss_old.c
@@ -0,0 +1,1120 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as a channel, and the console to command it :).
+ *
+ * The full-duplex "simulation" is pretty weak. This is generally a
+ * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
+ * writing a driver.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#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>
+
+#ifdef __linux
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.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 "busy.h"
+#include "ringtone.h"
+#include "ring10.h"
+#include "answer.h"
+
+/* Which device to use */
+#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
+#define DEV_DSP "/dev/audio"
+#else
+#define DEV_DSP "/dev/dsp"
+#endif
+
+/* Lets use 160 sample frames, just like GSM. */
+#define FRAME_SIZE 160
+
+/* 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 struct timeval lasttime;
+
+static int usecnt;
+static int silencesuppression = 0;
+static int silencethreshold = 1000;
+static int playbackonly = 0;
+
+
+AST_MUTEX_DEFINE_STATIC(usecnt_lock);
+
+static const char type[] = "Console";
+static const char desc[] = "OSS Console Channel Driver";
+static const char tdesc[] = "OSS Console Channel Driver";
+static const char config[] = "oss.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_oss_pvt {
+ /* We only have one OSS 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];
+} oss;
+
+static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
+static int oss_digit(struct ast_channel *c, char digit);
+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);
+static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+
+static const struct ast_channel_tech oss_tech = {
+ .type = type,
+ .description = tdesc,
+ .capabilities = AST_FORMAT_SLINEAR,
+ .requester = oss_request,
+ .send_digit = oss_digit,
+ .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,
+};
+
+static int time_has_passed(void)
+{
+ 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. */
+
+static 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 descriptor for sound device */
+static int sounddev = -1;
+
+static int autoanswer = 1;
+
+#if 0
+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;
+}
+#endif
+
+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;
+ audio_buf_info abi;
+ if (cursound > -1) {
+ res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
+ if (res) {
+ ast_log(LOG_WARNING, "Unable to read output space\n");
+ return -1;
+ }
+ /* Calculate how many samples we can send, max */
+ if (total > (abi.fragments * abi.fragsize / 2))
+ total = abi.fragments * abi.fragsize / 2;
+ 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;
+ }
+ }
+ }
+ if (frame)
+ res = write(sounddev, frame, res * 2);
+ if (res > 0)
+ return 0;
+ return res;
+ }
+ return 0;
+}
+
+static void *sound_thread(void *unused)
+{
+ fd_set rfds;
+ fd_set wfds;
+ int max;
+ int res;
+ char ign[4096];
+ if (read(sounddev, ign, sizeof(sounddev)) < 0)
+ ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+ for(;;) {
+ FD_ZERO(&rfds);
+ FD_ZERO(&wfds);
+ max = sndcmd[0];
+ FD_SET(sndcmd[0], &rfds);
+ if (!oss.owner) {
+ FD_SET(sounddev, &rfds);
+ if (sounddev > max)
+ max = sounddev;
+ }
+ if (cursound > -1) {
+ FD_SET(sounddev, &wfds);
+ if (sounddev > max)
+ max = sounddev;
+ }
+ res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+ if (res < 1) {
+ ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
+ continue;
+ }
+ if (FD_ISSET(sndcmd[0], &rfds)) {
+ read(sndcmd[0], &cursound, sizeof(cursound));
+ silencelen = 0;
+ offset = 0;
+ sampsent = 0;
+ }
+ if (FD_ISSET(sounddev, &rfds)) {
+ /* Ignore read */
+ if (read(sounddev, ign, sizeof(ign)) < 0)
+ ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+ }
+ if (FD_ISSET(sounddev, &wfds))
+ if (send_sound())
+ ast_log(LOG_WARNING, "Failed to write sound\n");
+ }
+ /* Never reached */
+ return NULL;
+}
+
+#if 0
+static int silence_suppress(short *buf)
+{
+#define SILBUF 3
+ int loudness;
+ static int silentframes = 0;
+ static char silbuf[FRAME_SIZE * 2 * SILBUF];
+ static int silbufcnt=0;
+ if (!silencesuppression)
+ return 0;
+ loudness = calc_loudness((short *)(buf));
+ if (option_debug)
+ ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
+ if (loudness < silencethreshold) {
+ silentframes++;
+ silbufcnt++;
+ /* Keep track of the last few bits of silence so we can play
+ them as lead-in when the time is right */
+ if (silbufcnt >= SILBUF) {
+ /* Make way for more buffer */
+ memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
+ silbufcnt--;
+ }
+ memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
+ if (silentframes > 10) {
+ /* We've had plenty of silence, so compress it now */
+ return 1;
+ }
+ } else {
+ silentframes=0;
+ /* Write any buffered silence we have, it may have something
+ important */
+ if (silbufcnt) {
+ write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
+ silbufcnt = 0;
+ }
+ }
+ return 0;
+}
+#endif
+
+static int setformat(void)
+{
+ int fmt, desired, res, fd = sounddev;
+ static int warnedalready = 0;
+ static int warnedalready2 = 0;
+
+#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;
+ }
+ res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+
+ /* Check to see if duplex set (FreeBSD Bug)*/
+ res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
+
+ if ((fmt & DSP_CAP_DUPLEX) && !res) {
+ if (option_verbose > 1)
+ ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+ full_duplex = -1;
+ }
+ 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;
+ }
+ /* 8000 Hz desired */
+ desired = 8000;
+ fmt = 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 (!warnedalready++)
+ ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+ }
+#if 1
+ fmt = BUFFER_FMT;
+ res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+ if (res < 0) {
+ if (!warnedalready2++)
+ ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+ }
+#endif
+ return 0;
+}
+
+static int soundcard_setoutput(int force)
+{
+ /* Make sure the soundcard is in output mode. */
+ int fd = sounddev;
+ if (full_duplex || (!readmode && !force))
+ return 0;
+ readmode = 0;
+ if (force || time_has_passed()) {
+ ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+ /* Keep the same fd reserved by closing the sound device and copying stdin at the same
+ time. */
+ /* dup2(0, sound); */
+ close(sounddev);
+ fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
+ if (fd < 0) {
+ ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+ return -1;
+ }
+ /* dup2 will close the original and make fd be sound */
+ if (dup2(fd, sounddev) < 0) {
+ ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+ return -1;
+ }
+ if (setformat()) {
+ return -1;
+ }
+ return 0;
+ }
+ return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+ int fd = sounddev;
+ if (full_duplex || (readmode && !force))
+ return 0;
+ readmode = -1;
+ if (force || time_has_passed()) {
+ ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+ close(sounddev);
+ /* dup2(0, sound); */
+ fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
+ if (fd < 0) {
+ ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+ return -1;
+ }
+ /* dup2 will close the original and make fd be sound */
+ if (dup2(fd, sounddev) < 0) {
+ ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+ return -1;
+ }
+ if (setformat()) {
+ return -1;
+ }
+ return 0;
+ }
+ return 1;
+}
+
+static int soundcard_init(void)
+{
+ /* Assume it's full duplex for starters */
+ int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
+ if (fd < 0) {
+ ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+ return fd;
+ }
+ gettimeofday(&lasttime, NULL);
+ sounddev = fd;
+ setformat();
+ if (!full_duplex)
+ soundcard_setinput(1);
+ return sounddev;
+}
+
+static int oss_digit(struct ast_channel *c, char digit)
+{
+ ast_verbose( " << Console Received digit %c >> \n", digit);
+ return 0;
+}
+
+static int oss_text(struct ast_channel *c, const char *text)
+{
+ ast_verbose( " << Console Received text %s >> \n", text);
+ return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+ int res = 3;
+ struct ast_frame f = { 0, };
+ ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+ if (autoanswer) {
+ ast_verbose( " << Auto-answered >> \n" );
+ f.frametype = AST_FRAME_CONTROL;
+ f.subclass = AST_CONTROL_ANSWER;
+ ast_queue_frame(c, &f);
+ } else {
+ nosound = 1;
+ 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);
+ 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 oss_answer(struct ast_channel *c)
+{
+ ast_verbose( " << Console call has been answered >> \n");
+ answer_sound();
+ ast_setstate(c, AST_STATE_UP);
+ cursound = -1;
+ nosound=0;
+ return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+ int res = 0;
+ cursound = -1;
+ c->tech_pvt = NULL;
+ oss.owner = NULL;
+ ast_verbose( " << Hangup on console >> \n");
+ ast_mutex_lock(&usecnt_lock);
+ usecnt--;
+ ast_mutex_unlock(&usecnt_lock);
+ if (hookstate) {
+ if (autoanswer) {
+ /* Assume auto-hangup too */
+ hookstate = 0;
+ } else {
+ /* Make congestion noise */
+ res = 2;
+ write(sndcmd[1], &res, sizeof(res));
+ }
+ }
+ return 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;
+}
+
+
+static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+{
+ int res;
+ static char sizbuf[8000];
+ static int sizpos = 0;
+ int len = sizpos;
+ int pos;
+ /* Immediately return if no sound is enabled */
+ if (nosound)
+ return 0;
+ /* Stop any currently playing sound */
+ cursound = -1;
+ if (!full_duplex && !playbackonly) {
+ /* If we're half duplex, we have to switch to read mode
+ to honor immediate needs if necessary. But if we are in play
+ back only mode, then we don't switch because the console
+ is only being used one way -- just to playback something. */
+ res = soundcard_setinput(1);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Unable to set device to input mode\n");
+ return -1;
+ }
+ return 0;
+ }
+ res = soundcard_setoutput(0);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Unable to set output device\n");
+ return -1;
+ } else if (res > 0) {
+ /* The device is still in read mode, and it's too soon to change it,
+ so just pretend we wrote it */
+ return 0;
+ }
+ /* 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;
+ while(len - pos > FRAME_SIZE * 2) {
+ soundcard_writeframe((short *)(sizbuf + pos));
+ pos += FRAME_SIZE * 2;
+ }
+ if (len - pos)
+ memmove(sizbuf, sizbuf + pos, len - pos);
+ sizpos = len - pos;
+ return 0;
+}
+
+static struct ast_frame *oss_read(struct ast_channel *chan)
+{
+ static struct ast_frame f;
+ static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
+ static int readpos = 0;
+ int res;
+
+#if 0
+ ast_log(LOG_DEBUG, "oss_read()\n");
+#endif
+
+ f.frametype = AST_FRAME_NULL;
+ f.subclass = 0;
+ f.samples = 0;
+ f.datalen = 0;
+ f.data = NULL;
+ f.offset = 0;
+ f.src = type;
+ f.mallocd = 0;
+ f.delivery.tv_sec = 0;
+ f.delivery.tv_usec = 0;
+
+ res = soundcard_setinput(0);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Unable to set input mode\n");
+ return NULL;
+ }
+ if (res > 0) {
+ /* Theoretically shouldn't happen, but anyway, return a NULL frame */
+ return &f;
+ }
+ res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
+#if 0
+ CRASH;
+#endif
+ return NULL;
+ }
+ readpos += res;
+
+ if (readpos >= FRAME_SIZE * 2) {
+ /* A real frame */
+ readpos = 0;
+ 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.samples = FRAME_SIZE;
+ f.datalen = FRAME_SIZE * 2;
+ f.data = buf + AST_FRIENDLY_OFFSET;
+ f.offset = AST_FRIENDLY_OFFSET;
+ f.src = type;
+ f.mallocd = 0;
+ f.delivery.tv_sec = 0;
+ f.delivery.tv_usec = 0;
+#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 oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+ struct chan_oss_pvt *p = newchan->tech_pvt;
+ p->owner = newchan;
+ return 0;
+}
+
+static int oss_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;
+ case -1:
+ cursound = -1;
+ return 0;
+ 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 *oss_new(struct chan_oss_pvt *p, int state)
+{
+ struct ast_channel *tmp;
+ tmp = ast_channel_alloc(1);
+ if (tmp) {
+ tmp->tech = &oss_tech;
+ snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+ tmp->type = type;
+ tmp->fds[0] = sounddev;
+ tmp->nativeformats = AST_FORMAT_SLINEAR;
+ tmp->readformat = AST_FORMAT_SLINEAR;
+ tmp->writeformat = AST_FORMAT_SLINEAR;
+ tmp->tech_pvt = p;
+ 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;
+ 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;
+ }
+ }
+ }
+ return tmp;
+}
+
+static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
+{
+ 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 (oss.owner) {
+ ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+ *cause = AST_CAUSE_BUSY;
+ return NULL;
+ }
+ tmp= oss_new(&oss, AST_STATE_DOWN);
+ if (!tmp) {
+ ast_log(LOG_WARNING, "Unable to create new OSS 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 'oss.conf'.\n";
+
+static int console_answer(int fd, int argc, char *argv[])
+{
+ struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+ if (argc != 1)
+ return RESULT_SHOWUSAGE;
+ if (!oss.owner) {
+ ast_cli(fd, "No one is calling us\n");
+ return RESULT_FAILURE;
+ }
+ hookstate = 1;
+ cursound = -1;
+ ast_queue_frame(oss.owner, &f);
+ 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 = 2;
+ char text2send[256] = "";
+ struct ast_frame f = { 0, };
+ if (argc < 2)
+ return RESULT_SHOWUSAGE;
+ if (!oss.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");
+ text2send[0] = '\0';
+ while(tmparg < argc) {
+ strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
+ strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
+ }
+ if (strlen(text2send)) {
+ f.frametype = AST_FRAME_TEXT;
+ f.subclass = 0;
+ f.data = text2send;
+ f.datalen = strlen(text2send);
+ ast_queue_frame(oss.owner, &f);
+ }
+ return RESULT_SUCCESS;
+}
+
+static char answer_usage[] =
+"Usage: answer\n"
+" Answers an incoming call on the console (OSS) channel.\n";
+
+static int console_hangup(int fd, int argc, char *argv[])
+{
+ if (argc != 1)
+ return RESULT_SHOWUSAGE;
+ cursound = -1;
+ if (!oss.owner && !hookstate) {
+ ast_cli(fd, "No call to hangup up\n");
+ return RESULT_FAILURE;
+ }
+ hookstate = 0;
+ if (oss.owner) {
+ ast_queue_hangup(oss.owner);
+ }
+ return RESULT_SUCCESS;
+}
+
+static int console_flash(int fd, int argc, char *argv[])
+{
+ struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+ if (argc != 1)
+ return RESULT_SHOWUSAGE;
+ cursound = -1;
+ if (!oss.owner) {
+ ast_cli(fd, "No call to flash\n");
+ return RESULT_FAILURE;
+ }
+ hookstate = 0;
+ if (oss.owner) {
+ ast_queue_frame(oss.owner, &f);
+ }
+ return RESULT_SUCCESS;
+}
+
+static char hangup_usage[] =
+"Usage: hangup\n"
+" Hangs up any call currently placed on the console.\n";
+
+
+static char flash_usage[] =
+"Usage: flash\n"
+" Flashes the 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 x;
+ struct ast_frame f = { AST_FRAME_DTMF, 0 };
+ if ((argc != 1) && (argc != 2))
+ return RESULT_SHOWUSAGE;
+ if (oss.owner) {
+ if (argc == 2) {
+ for (x=0;x<strlen(argv[1]);x++) {
+ f.subclass = argv[1][x];
+ ast_queue_frame(oss.owner, &f);
+ }
+ } 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) {
+ char *stringp=NULL;
+ strncpy(tmp, argv[1], sizeof(tmp)-1);
+ stringp=tmp;
+ strsep(&stringp, "@");
+ tmp2 = strsep(&stringp, "@");
+ if (strlen(tmp))
+ mye = tmp;
+ if (tmp2 && strlen(tmp2))
+ myc = tmp2;
+ }
+ if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
+ strncpy(oss.exten, mye, sizeof(oss.exten)-1);
+ strncpy(oss.context, myc, sizeof(oss.context)-1);
+ hookstate = 1;
+ oss_new(&oss, AST_STATE_RINGING);
+ } 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 (and context if specified)\n";
+
+static int console_transfer(int fd, int argc, char *argv[])
+{
+ char tmp[256];
+ char *context;
+ if (argc != 2)
+ return RESULT_SHOWUSAGE;
+ if (oss.owner && ast_bridged_channel(oss.owner)) {
+ strncpy(tmp, argv[1], sizeof(tmp) - 1);
+ context = strchr(tmp, '@');
+ if (context) {
+ *context = '\0';
+ context++;
+ } else
+ context = oss.owner->context;
+ if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
+ ast_cli(fd, "Whee, transferring %s to %s@%s.\n",
+ ast_bridged_channel(oss.owner)->name, tmp, context);
+ if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
+ ast_cli(fd, "Failed to transfer :(\n");
+ } else {
+ ast_cli(fd, "No such extension exists\n");
+ }
+ } else {
+ ast_cli(fd, "There is no call to transfer\n");
+ }
+ return RESULT_SUCCESS;
+}
+
+static char transfer_usage[] =
+"Usage: transfer <extension>[@context]\n"
+" Transfers the currently connected call to the given extension (and\n"
+"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 },
+ { { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage },
+ { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
+ { { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_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;
+ struct ast_config *cfg;
+ struct ast_variable *v;
+ 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 OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
+ }
+ return 0;
+ }
+ if (!full_duplex)
+ ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+ res = ast_channel_register(&oss_tech);
+ 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 = ast_config_load(config))) {
+ 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, "playbackonly"))
+ playbackonly = ast_true(v->value);
+ v=v->next;
+ }
+ ast_config_destroy(cfg);
+ }
+ ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+ return 0;
+}
+
+
+
+int unload_module()
+{
+ int x;
+
+ ast_channel_unregister(&oss_tech);
+ for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+ ast_cli_unregister(myclis + x);
+ close(sounddev);
+ if (sndcmd[0] > 0) {
+ close(sndcmd[0]);
+ close(sndcmd[1]);
+ }
+ if (oss.owner)
+ ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
+ if (oss.owner)
+ return -1;
+ return 0;
+}
+
+char *description()
+{
+ return (char *) desc;
+}
+
+int usecount()
+{
+ return usecnt;
+}
+
+char *key()
+{
+ return ASTERISK_GPL_KEY;
+}