Version 0.1.7 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

codecs/codec_adpcm.c

@@ -0,0 +1,545 @@
+/* codec_adpcm.c - translate between signed linear and Dialogic ADPCM
+ * 
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Based on frompcm.c and topcm.c from the Emiliano MIPL browser/
+ * interpreter.  See http://www.bsdtelephony.com.mx
+ *
+ * Copyright (c) 2001 Linux Support Services, Inc.  All rights reserved.
+ *
+ * Karl Sackett <krs@linux-support.net>, 2001-3-21
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/logger.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <fcntl.h>
+#include <netinet/in.h>
+#include <pthread.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+
+#define BUFFER_SIZE   8096	/* size for the translation buffers */
+
+static pthread_mutex_t localuser_lock = PTHREAD_MUTEX_INITIALIZER;
+static int localusecnt = 0;
+
+static char *tdesc = "Adaptive Differential PCM Coder/Decoder";
+
+/* Sample frame data */
+
+#include "slin_adpcm_ex.h"
+#include "adpcm_slin_ex.h"
+
+/*
+ * Step size index shift table 
+ */
+
+static short indsft[8] = { -1, -1, -1, -1, 2, 4, 6, 8 };
+
+/*
+ * Step size table, where stpsz[i]=floor[16*(11/10)^i]
+ */
+
+static short stpsz[49] = {
+  16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73,
+  80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279,
+  307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
+  1060, 1166, 1282, 1411, 1552
+};
+
+/* 
+ * Nibble to bit map
+ */
+
+static short nbl2bit[16][4] = {
+  {1, 0, 0, 0}, {1, 0, 0, 1}, {1, 0, 1, 0}, {1, 0, 1, 1},
+  {1, 1, 0, 0}, {1, 1, 0, 1}, {1, 1, 1, 0}, {1, 1, 1, 1},
+  {-1, 0, 0, 0}, {-1, 0, 0, 1}, {-1, 0, 1, 0}, {-1, 0, 1, 1},
+  {-1, 1, 0, 0}, {-1, 1, 0, 1}, {-1, 1, 1, 0}, {-1, 1, 1, 1}
+};
+
+/*
+ * Decode(encoded)
+ *  Decodes the encoded nibble from the adpcm file.
+ *
+ * Results:
+ *  Returns the encoded difference.
+ *
+ * Side effects:
+ *  Sets the index to the step size table for the next encode.
+ */
+
+static inline short
+decode (unsigned char encoded, short *ssindex)
+{
+  short diff, step;
+  step = stpsz[*ssindex];
+  diff = nbl2bit[encoded][0] * (step * nbl2bit[encoded][1] +
+				(step >> 1) * nbl2bit[encoded][2] +
+				(step >> 2) * nbl2bit[encoded][3] +
+				(step >> 3));
+  *ssindex = *ssindex + indsft[(encoded & 7)];
+  if (*ssindex < 0)
+    *ssindex = 0;
+  else if (*ssindex > 48)
+    *ssindex = 48;
+  return (diff);
+}
+
+/*
+ * Adpcm
+ *  Takes a signed linear signal and encodes it as ADPCM
+ *  For more information see http://support.dialogic.com/appnotes/adpcm.pdf
+ *
+ * Results:
+ *  Foo.
+ *
+ * Side effects:
+ *  signal gets updated with each pass.
+ */
+
+static inline unsigned char
+adpcm (short csig, short *ssindex, short *signal)
+{
+  short diff, step;
+  unsigned char encoded;
+  step = stpsz[*ssindex];
+  /* 
+   * Clip csig if too large or too small
+   */
+   
+  csig >>= 4;
+
+  diff = csig - *signal;
+  
+  if (diff < 0)
+    {
+      encoded = 8;
+      diff = -diff;
+    }
+  else
+    encoded = 0;
+  if (diff >= step)
+    {
+      encoded |= 4;
+      diff -= step;
+    }
+  step >>= 1;
+  if (diff >= step)
+    {
+      encoded |= 2;
+      diff -= step;
+    }
+  step >>= 1;
+  if (diff >= step)
+    encoded |= 1;
+    
+  *signal += decode (encoded, ssindex);
+  return (encoded);
+}
+
+/*
+ * Private workspace for translating signed linear signals to ADPCM.
+ */
+
+struct adpcm_encoder_pvt
+{
+  struct ast_frame f;
+  char offset[AST_FRIENDLY_OFFSET];   /* Space to build offset */
+  short inbuf[BUFFER_SIZE];           /* Unencoded signed linear values */
+  unsigned char outbuf[BUFFER_SIZE];  /* Encoded ADPCM, two nibbles to a word */
+  short ssindex;
+  short signal;
+  int tail;
+};
+
+/*
+ * Private workspace for translating ADPCM signals to signed linear.
+ */
+
+struct adpcm_decoder_pvt
+{
+  struct ast_frame f;
+  char offset[AST_FRIENDLY_OFFSET];	/* Space to build offset */
+  short outbuf[BUFFER_SIZE];	/* Decoded signed linear values */
+  int tail;
+};
+
+/*
+ * AdpcmToLin_New
+ *  Create a new instance of adpcm_decoder_pvt.
+ *
+ * Results:
+ *  Returns a pointer to the new instance.
+ *
+ * Side effects:
+ *  None.
+ */
+
+static struct ast_translator_pvt *
+adpcmtolin_new ()
+{
+  struct adpcm_decoder_pvt *tmp;
+  tmp = malloc (sizeof (struct adpcm_decoder_pvt));
+  if (tmp)
+    {
+	  memset(tmp, 0, sizeof(*tmp));
+      tmp->tail = 0;
+      localusecnt++;
+      ast_update_use_count ();
+    }
+  return (struct ast_translator_pvt *) tmp;
+}
+
+/*
+ * LinToAdpcm_New
+ *  Create a new instance of adpcm_encoder_pvt.
+ *
+ * Results:
+ *  Returns a pointer to the new instance.
+ *
+ * Side effects:
+ *  None.
+ */
+
+static struct ast_translator_pvt *
+lintoadpcm_new ()
+{
+  struct adpcm_encoder_pvt *tmp;
+  tmp = malloc (sizeof (struct adpcm_encoder_pvt));
+  if (tmp)
+    {
+	  memset(tmp, 0, sizeof(*tmp));
+      localusecnt++;
+      ast_update_use_count ();
+      tmp->tail = 0;
+    }
+  return (struct ast_translator_pvt *) tmp;
+}
+
+/*
+ * AdpcmToLin_FrameIn
+ *  Fill an input buffer with packed 4-bit ADPCM values if there is room
+ *  left.
+ *
+ * Results:
+ *  Foo
+ *
+ * Side effects:
+ *  tmp->tail is the number of packed values in the buffer.
+ */
+
+static int
+adpcmtolin_framein (struct ast_translator_pvt *pvt, struct ast_frame *f)
+{
+  struct adpcm_decoder_pvt *tmp = (struct adpcm_decoder_pvt *) pvt;
+  int x;
+  short signal;
+  short ssindex;
+  unsigned char *b;
+
+  if (f->datalen < 3) {
+  	ast_log(LOG_WARNING, "Didn't have at least three bytes of input\n");
+	return -1;
+  }
+
+  if ((f->datalen - 3) * 4 > sizeof(tmp->outbuf)) {
+  	ast_log(LOG_WARNING, "Out of buffer space\n");
+	return -1;
+  }
+
+  /* Reset ssindex and signal to frame's specified values */
+  b = f->data;
+  ssindex = b[0];
+  if (ssindex < 0)
+  	ssindex = 0;
+  if (ssindex > 48)
+    ssindex = 48;
+
+  signal = (b[1] << 8) | b[2]; 
+  
+  for (x=3;x<f->datalen;x++) {
+  	signal += decode(b[x] >> 4, &ssindex);
+	if (signal > 2047)
+		signal = 2047;
+	if (signal < -2048)
+		signal = -2048;
+    tmp->outbuf[tmp->tail++] = signal << 4;
+  	signal +=  decode(b[x] & 0x0f, &ssindex);
+	if (signal > 2047)
+		signal = 2047;
+	if (signal < -2048)
+		signal = -2048;
+    tmp->outbuf[tmp->tail++] = signal << 4;
+  }
+
+  return 0;
+}
+
+/*
+ * AdpcmToLin_FrameOut
+ *  Convert 4-bit ADPCM encoded signals to 16-bit signed linear.
+ *
+ * Results:
+ *  Converted signals are placed in tmp->f.data, tmp->f.datalen
+ *  and tmp->f.timelen are calculated.
+ *
+ * Side effects:
+ *  None.
+ */
+
+static struct ast_frame *
+adpcmtolin_frameout (struct ast_translator_pvt *pvt)
+{
+  struct adpcm_decoder_pvt *tmp = (struct adpcm_decoder_pvt *) pvt;
+
+  if (!tmp->tail)
+    return NULL;
+
+  tmp->f.frametype = AST_FRAME_VOICE;
+  tmp->f.subclass = AST_FORMAT_SLINEAR;
+  tmp->f.datalen = tmp->tail *2;
+  tmp->f.timelen = tmp->tail / 8;
+  tmp->f.mallocd = 0;
+  tmp->f.offset = AST_FRIENDLY_OFFSET;
+  tmp->f.src = __PRETTY_FUNCTION__;
+  tmp->f.data = tmp->outbuf;
+  tmp->tail = 0;
+  return &tmp->f;
+}
+
+/*
+ * LinToAdpcm_FrameIn
+ *  Fill an input buffer with 16-bit signed linear PCM values.
+ *
+ * Results:
+ *  None.
+ *
+ * Side effects:
+ *  tmp->tail is number of signal values in the input buffer.
+ */
+
+static int
+lintoadpcm_framein (struct ast_translator_pvt *pvt, struct ast_frame *f)
+{
+  struct adpcm_encoder_pvt *tmp = (struct adpcm_encoder_pvt *) pvt;
+
+  if ((tmp->tail + f->datalen / 2) < (sizeof (tmp->inbuf) / 2))
+    {
+      memcpy (&tmp->inbuf[tmp->tail], f->data, f->datalen);
+      tmp->tail += f->datalen / 2;
+    }
+  else
+    {
+      ast_log (LOG_WARNING, "Out of buffer space\n");
+      return -1;
+    }
+  return 0;
+}
+
+/*
+ * LinToAdpcm_FrameOut
+ *  Convert a buffer of raw 16-bit signed linear PCM to a buffer
+ *  of 4-bit ADPCM packed two to a byte (Big Endian).
+ *
+ * Results:
+ *  Foo
+ *
+ * Side effects:
+ *  Leftover inbuf data gets packed, tail gets updated.
+ */
+
+static struct ast_frame *
+lintoadpcm_frameout (struct ast_translator_pvt *pvt)
+{
+  struct adpcm_encoder_pvt *tmp = (struct adpcm_encoder_pvt *) pvt;
+  unsigned char adpcm0, adpcm1;
+  int i_max, i;
+  
+  if (tmp->tail < 2) return NULL;
+
+
+  i_max = (tmp->tail / 2) * 2;
+
+  tmp->outbuf[0] = tmp->ssindex & 0xff;
+  tmp->outbuf[1] = (tmp->signal >> 8) & 0xff;
+  tmp->outbuf[2] = (tmp->signal & 0xff);
+
+  for (i = 0; i < i_max; i+=2)
+  {
+    adpcm0 = adpcm(tmp->inbuf[i], &tmp->ssindex, &tmp->signal);
+    adpcm1 = adpcm(tmp->inbuf[i+1], &tmp->ssindex, &tmp->signal);
+    tmp->outbuf[i/2 + 3] = (adpcm0 << 4) | adpcm1;
+  };
+
+
+  tmp->f.frametype = AST_FRAME_VOICE;
+  tmp->f.subclass = AST_FORMAT_ADPCM;
+  tmp->f.timelen = i_max / 8;
+  tmp->f.mallocd = 0;
+  tmp->f.offset = AST_FRIENDLY_OFFSET;
+  tmp->f.src = __PRETTY_FUNCTION__;
+  tmp->f.data = tmp->outbuf;
+  tmp->f.datalen = i_max / 2 + 3;
+
+  /*
+   * If there is a signal left over (there should be no more than
+   * one) move it to the beginning of the input buffer.
+   */
+
+  if (tmp->tail == i_max)
+    tmp->tail = 0;
+  else
+    {
+      tmp->inbuf[0] = tmp->inbuf[tmp->tail];
+      tmp->tail = 1;
+    }
+  return &tmp->f;
+}
+
+
+/*
+ * AdpcmToLin_Sample
+ */
+
+static struct ast_frame *
+adpcmtolin_sample ()
+{
+  static struct ast_frame f;
+  f.frametype = AST_FRAME_VOICE;
+  f.subclass = AST_FORMAT_ADPCM;
+  f.datalen = sizeof (adpcm_slin_ex);
+  f.timelen = sizeof(adpcm_slin_ex) / 4;
+  f.mallocd = 0;
+  f.offset = 0;
+  f.src = __PRETTY_FUNCTION__;
+  f.data = adpcm_slin_ex;
+  return &f;
+}
+
+/*
+ * LinToAdpcm_Sample
+ */
+
+static struct ast_frame *
+lintoadpcm_sample ()
+{
+  static struct ast_frame f;
+  f.frametype = AST_FRAME_VOICE;
+  f.subclass = AST_FORMAT_SLINEAR;
+  f.datalen = sizeof (slin_adpcm_ex);
+  /* Assume 8000 Hz */
+  f.timelen = sizeof (slin_adpcm_ex) / 16;
+  f.mallocd = 0;
+  f.offset = 0;
+  f.src = __PRETTY_FUNCTION__;
+  f.data = slin_adpcm_ex;
+  return &f;
+}
+
+/*
+ * Adpcm_Destroy
+ *  Destroys a private workspace.
+ *
+ * Results:
+ *  It's gone!
+ *
+ * Side effects:
+ *  None.
+ */
+
+static void
+adpcm_destroy (struct ast_translator_pvt *pvt)
+{
+  free (pvt);
+  localusecnt--;
+  ast_update_use_count ();
+}
+
+/*
+ * The complete translator for ADPCMToLin.
+ */
+
+static struct ast_translator adpcmtolin = {
+  "adpcmtolin",
+  AST_FORMAT_ADPCM,
+  AST_FORMAT_SLINEAR,
+  adpcmtolin_new,
+  adpcmtolin_framein,
+  adpcmtolin_frameout,
+  adpcm_destroy,
+  /* NULL */
+  adpcmtolin_sample
+};
+
+/*
+ * The complete translator for LinToADPCM.
+ */
+
+static struct ast_translator lintoadpcm = {
+  "lintoadpcm",
+  AST_FORMAT_SLINEAR,
+  AST_FORMAT_ADPCM,
+  lintoadpcm_new,
+  lintoadpcm_framein,
+  lintoadpcm_frameout,
+  adpcm_destroy,
+  /* NULL */
+  lintoadpcm_sample
+};
+
+int
+unload_module (void)
+{
+  int res;
+  pthread_mutex_lock (&localuser_lock);
+  res = ast_unregister_translator (&lintoadpcm);
+  if (!res)
+    res = ast_unregister_translator (&adpcmtolin);
+  if (localusecnt)
+    res = -1;
+  pthread_mutex_unlock (&localuser_lock);
+  return res;
+}
+
+int
+load_module (void)
+{
+  int res;
+  res = ast_register_translator (&adpcmtolin);
+  if (!res)
+    res = ast_register_translator (&lintoadpcm);
+  else
+    ast_unregister_translator (&adpcmtolin);
+  return res;
+}
+
+/*
+ * Return a description of this module.
+ */
+
+char *
+description (void)
+{
+  return tdesc;
+}
+
+int
+usecount (void)
+{
+  int res;
+  STANDARD_USECOUNT (res);
+  return res;
+}
+
+char *
+key ()
+{
+  return ASTERISK_GPL_KEY;
+}

formats/format_vox.c

@@ -0,0 +1,422 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Flat, binary, ADPCM vox file format.
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/channel.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/sched.h>
+#include <asterisk/module.h>
+#include <arpa/inet.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <pthread.h>
+#include <endian.h>
+
+#define BUF_SIZE 80		/* 160 samples */
+
+struct ast_filestream {
+	void *reserved[AST_RESERVED_POINTERS];
+	/* Believe it or not, we must decode/recode to account for the
+	   weird MS format */
+	/* This is what a filestream means to us */
+	int fd; /* Descriptor */
+	struct ast_channel *owner;
+	struct ast_frame fr;				/* Frame information */
+	char waste[AST_FRIENDLY_OFFSET];	/* Buffer for sending frames, etc */
+	char empty;							/* Empty character */
+	unsigned char buf[BUF_SIZE + 3];				/* Output Buffer */
+	int lasttimeout;
+	struct timeval last;
+	int adj;
+	short signal;						/* Signal level (file side) */
+	short ssindex;						/* Signal ssindex (file side) */
+	struct ast_filestream *next;
+};
+
+
+static struct ast_filestream *glist = NULL;
+static pthread_mutex_t vox_lock = PTHREAD_MUTEX_INITIALIZER;
+static int glistcnt = 0;
+
+static char *name = "vox";
+static char *desc = "Dialogic VOX (ADPCM) File Format";
+static char *exts = "vox";
+
+/*
+ * Step size index shift table 
+ */
+
+static short indsft[8] = { -1, -1, -1, -1, 2, 4, 6, 8 };
+
+/*
+ * Step size table, where stpsz[i]=floor[16*(11/10)^i]
+ */
+
+static short stpsz[49] = {
+  16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73,
+  80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279,
+  307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
+  1060, 1166, 1282, 1411, 1552
+};
+
+/* 
+ * Nibble to bit map
+ */
+
+static short nbl2bit[16][4] = {
+  {1, 0, 0, 0}, {1, 0, 0, 1}, {1, 0, 1, 0}, {1, 0, 1, 1},
+  {1, 1, 0, 0}, {1, 1, 0, 1}, {1, 1, 1, 0}, {1, 1, 1, 1},
+  {-1, 0, 0, 0}, {-1, 0, 0, 1}, {-1, 0, 1, 0}, {-1, 0, 1, 1},
+  {-1, 1, 0, 0}, {-1, 1, 0, 1}, {-1, 1, 1, 0}, {-1, 1, 1, 1}
+};
+
+/*
+ * Decode(encoded)
+ *  Decodes the encoded nibble from the adpcm file.
+ *
+ * Results:
+ *  Returns the encoded difference.
+ *
+ * Side effects:
+ *  Sets the index to the step size table for the next encode.
+ */
+
+static inline short
+decode (unsigned char encoded, short *ssindex)
+{
+  short diff, step;
+  step = stpsz[*ssindex];
+  diff = nbl2bit[encoded][0] * (step * nbl2bit[encoded][1] +
+				(step >> 1) * nbl2bit[encoded][2] +
+				(step >> 2) * nbl2bit[encoded][3] +
+				(step >> 3));
+  *ssindex = *ssindex + indsft[(encoded & 7)];
+  if (*ssindex < 0)
+    *ssindex = 0;
+  else if (*ssindex > 48)
+    *ssindex = 48;
+  return (diff);
+}
+
+/*
+ * Adpcm
+ *  Takes a signed linear signal and encodes it as ADPCM
+ *  For more information see http://support.dialogic.com/appnotes/adpcm.pdf
+ *
+ * Results:
+ *  Foo.
+ *
+ * Side effects:
+ *  signal gets updated with each pass.
+ */
+
+static inline unsigned char
+adpcm (short csig, short *ssindex, short *signal)
+{
+  short diff, step;
+  unsigned char encoded;
+  step = stpsz[*ssindex];
+  /* 
+   * Clip csig if too large or too small
+   */
+   
+  csig >>= 4;
+
+  diff = csig - *signal;
+  
+  if (diff < 0)
+    {
+      encoded = 8;
+      diff = -diff;
+    }
+  else
+    encoded = 0;
+  if (diff >= step)
+    {
+      encoded |= 4;
+      diff -= step;
+    }
+  step >>= 1;
+  if (diff >= step)
+    {
+      encoded |= 2;
+      diff -= step;
+    }
+  step >>= 1;
+  if (diff >= step)
+    encoded |= 1;
+    
+  *signal += decode (encoded, ssindex);
+  return (encoded);
+}
+
+static struct ast_filestream *vox_open(int fd)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		memset(tmp, 0, sizeof(struct ast_filestream));
+		if (pthread_mutex_lock(&vox_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock vox list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->fr.data = tmp->buf;
+		tmp->fr.frametype = AST_FRAME_VOICE;
+		tmp->fr.subclass = AST_FORMAT_ADPCM;
+		/* datalen will vary for each frame */
+		tmp->fr.src = name;
+		tmp->fr.mallocd = 0;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&vox_lock);
+		ast_update_use_count();
+	}
+	return tmp;
+}
+
+static struct ast_filestream *vox_rewrite(int fd, char *comment)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		memset(tmp, 0, sizeof(struct ast_filestream));
+		if (pthread_mutex_lock(&vox_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock vox list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&vox_lock);
+		ast_update_use_count();
+	} else
+		ast_log(LOG_WARNING, "Out of memory\n");
+	return tmp;
+}
+
+static struct ast_frame *vox_read(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+static void vox_close(struct ast_filestream *s)
+{
+	struct ast_filestream *tmp, *tmpl = NULL;
+	if (pthread_mutex_lock(&vox_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock vox list\n");
+		return;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp == s) {
+			if (tmpl)
+				tmpl->next = tmp->next;
+			else
+				glist = tmp->next;
+			break;
+		}
+		tmpl = tmp;
+		tmp = tmp->next;
+	}
+	glistcnt--;
+	if (s->owner) {
+		s->owner->stream = NULL;
+		if (s->owner->streamid > -1)
+			ast_sched_del(s->owner->sched, s->owner->streamid);
+		s->owner->streamid = -1;
+	}
+	pthread_mutex_unlock(&vox_lock);
+	ast_update_use_count();
+	if (!tmp) 
+		ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
+	close(s->fd);
+	free(s);
+}
+
+static int ast_read_callback(void *data)
+{
+	int retval = 0;
+	int res;
+	int delay;
+	struct ast_filestream *s = data;
+	struct timeval tv;
+	int x;
+	/* Send a frame from the file to the appropriate channel */
+
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_ADPCM;
+	s->fr.offset = AST_FRIENDLY_OFFSET;
+	s->fr.mallocd = 0;
+	s->fr.data = s->buf;
+	if ((res = read(s->fd, s->buf + 3, BUF_SIZE)) < 1) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		s->owner->streamid = -1;
+		return 0;
+	}
+	/* Store index, then signal */
+	s->buf[0] = s->ssindex & 0xff;
+	s->buf[1] = (s->signal >> 8) & 0xff;
+	s->buf[2] = s->signal & 0xff;
+	/* Do the decoder to be sure we get the right stuff in the signal and index fields. */
+	for (x=3;x<res+3;x++) {
+		s->signal += decode(s->buf[x] >> 4, &s->ssindex);
+		s->signal += decode(s->buf[x] & 0xf, &s->ssindex);
+	}
+	s->fr.timelen = res / 4;
+	s->fr.datalen = res + 3;
+	delay = s->fr.timelen;
+	/* Lastly, process the frame */
+	if (ast_write(s->owner, &s->fr)) {
+		ast_log(LOG_WARNING, "Failed to write frame\n");
+		s->owner->streamid = -1;
+		return 0;
+	}
+	if (s->last.tv_usec || s->last.tv_usec) {
+		int ms;
+		gettimeofday(&tv, NULL);
+		ms = 1000 * (tv.tv_sec - s->last.tv_sec) + 
+			(tv.tv_usec - s->last.tv_usec) / 1000;
+		s->last.tv_sec = tv.tv_sec;
+		s->last.tv_usec = tv.tv_usec;
+		if ((ms - delay) * (ms - delay) > 4) {
+			/* Compensate if we're more than 2 ms off */
+			s->adj -= (ms - delay);
+		}
+#if 0
+		fprintf(stdout, "Delay is %d, adjustment is %d, last was %d\n", delay, s->adj, ms);
+#endif
+		delay += s->adj;
+		if (delay < 1)
+			delay = 1;
+	} else
+		gettimeofday(&s->last, NULL);
+	if (s->lasttimeout != delay) {
+		/* We'll install the next timeout now. */
+		s->owner->streamid = ast_sched_add(s->owner->sched,
+				delay, ast_read_callback, s); 
+		s->lasttimeout = delay;
+	} else {
+		/* Just come back again at the same time */
+		retval = -1;
+	}
+	return retval;
+}
+
+static int vox_apply(struct ast_channel *c, struct ast_filestream *s)
+{
+	/* Select our owner for this stream, and get the ball rolling. */
+	s->owner = c;
+	ast_read_callback(s);
+	return 0;
+}
+
+static int vox_write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_ADPCM) {
+		ast_log(LOG_WARNING, "Asked to write non-ADPCM frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if (f->datalen < 3) {
+		ast_log(LOG_WARNING, "Invalid frame of data (< 3 bytes long) from %s\n", f->src);
+		return -1;
+	}
+	if ((res = write(fs->fd, f->data + 3, f->datalen)) != f->datalen) {
+			ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+			return -1;
+	}
+	return 0;
+}
+
+static char *vox_getcomment(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+int load_module()
+{
+	return ast_format_register(name, exts, AST_FORMAT_ADPCM,
+								vox_open,
+								vox_rewrite,
+								vox_apply,
+								vox_write,
+								vox_read,
+								vox_close,
+								vox_getcomment);
+								
+								
+}
+
+int unload_module()
+{
+	struct ast_filestream *tmp, *tmpl;
+	if (pthread_mutex_lock(&vox_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock vox list\n");
+		return -1;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp->owner)
+			ast_softhangup(tmp->owner);
+		tmpl = tmp;
+		tmp = tmp->next;
+		free(tmpl);
+	}
+	pthread_mutex_unlock(&vox_lock);
+	return ast_format_unregister(name);
+}	
+
+int usecount()
+{
+	int res;
+	if (pthread_mutex_lock(&vox_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock vox list\n");
+		return -1;
+	}
+	res = glistcnt;
+	pthread_mutex_unlock(&vox_lock);
+	return res;
+}
+
+char *description()
+{
+	return desc;
+}
+
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}