From b3a0ef24ed5cd824de6e6e4ff3910befc3aa444c Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Sat, 18 Dec 2004 22:04:07 +0000
Subject: [PATCH] Fix comment issues (bug #3089)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channel.c | 2 +-
channels/chan_vpb.c | 201 ++++++++++++++++++++------------------
formats/format_pcm.c | 2 +-
formats/format_pcm_alaw.c | 21 ++--
formats/format_wav.c | 6 +-
logger.c | 4 +-
pbx.c | 24 ++---
res/res_agi.c | 4 +-
res/res_config_odbc.c | 4 +-
res/res_indications.c | 16 +--
res/res_monitor.c | 14 +--
11 files changed, 155 insertions(+), 143 deletions(-)
diff --git a/channel.c b/channel.c
index dca52974a4..c26774f8da 100755
--- a/channel.c
+++ b/channel.c
@@ -643,7 +643,7 @@ void ast_channel_free(struct ast_channel *chan)
while (!AST_LIST_EMPTY(headp)) { /* List Deletion. */
vardata = AST_LIST_FIRST(headp);
AST_LIST_REMOVE_HEAD(headp, entries);
-// printf("deleting var %s=%s\n",ast_var_name(vardata),ast_var_value(vardata));
+/* printf("deleting var %s=%s\n",ast_var_name(vardata),ast_var_value(vardata)); */
ast_var_delete(vardata);
}
diff --git a/channels/chan_vpb.c b/channels/chan_vpb.c
index 29397331ae..cbcd46c5db 100755
--- a/channels/chan_vpb.c
+++ b/channels/chan_vpb.c
@@ -162,8 +162,8 @@ static int use_ast_ind=0;
|VPB_MRING_OFF|VPB_MSTATION_FLASH)
-// Dialing parameters for Australia
-//#define DIAL_WITH_CALL_PROGRESS
+/* Dialing parameters for Australia */
+/* #define DIAL_WITH_CALL_PROGRESS */
VPB_TONE_MAP DialToneMap[] = { { VPB_BUSY_AUST, VPB_CALL_DISCONNECT, 0 },
{ VPB_DIAL, VPB_CALL_DIALTONE, 0 },
{ VPB_RINGBACK_308, VPB_CALL_RINGBACK, 0 },
@@ -277,14 +277,16 @@ static struct vpb_pvt {
static struct ast_channel *vpb_new(struct vpb_pvt *i, int state, char *context);
static void *do_chanreads(void *pvt);
-// Can't get vpb_bridge() working on v4pci without either a horrible
-// high pitched feedback noise or bad hiss noise depending on gain settings
-// Get asterisk to do the bridging
+/* Can't get vpb_bridge() working on v4pci without either a horrible
+* high pitched feedback noise or bad hiss noise depending on gain settings
+* Get asterisk to do the bridging
+*/
#define BAD_V4PCI_BRIDGE
-// This one enables a half duplex bridge which may be required to prevent high pitched
-// feedback when getting asterisk to do the bridging and when using certain gain settings.
-//#define HALF_DUPLEX_BRIDGE
+/* This one enables a half duplex bridge which may be required to prevent high pitched
+ * feedback when getting asterisk to do the bridging and when using certain gain settings.
+ */
+/* #define HALF_DUPLEX_BRIDGE */
/* This is the Native bridge code, which Asterisk will try before using its own bridging code */
static int vpb_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
@@ -394,9 +396,9 @@ static int vpb_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags,
res = vpb_bridge(p0->handle, p1->handle, VPB_BRIDGE_ON, i+1 /* resource 1 & 2 only for V4PCI*/ );
if (res == VPB_OK) {
- //pthread_cond_wait(&bridges[i].cond, &bridges[i].lock); /* Wait for condition signal. */
+ /* pthread_cond_wait(&bridges[i].cond, &bridges[i].lock);*/ /* Wait for condition signal. */
while( !bridges[i].endbridge ) {
- // Are we really ment to be doing nothing ?!?!
+ /* Are we really ment to be doing nothing ?!?! */
who = ast_waitfor_n(cs, 2, &to);
if (!who) {
ast_log(LOG_DEBUG, "vpb_bridge: Empty frame read...\n");
@@ -423,7 +425,7 @@ static int vpb_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags,
}
*/
/* That's all we needed */
- //return 0;
+ /*return 0; */
break;
} else {
if ((f->frametype == AST_FRAME_DTMF) ||
@@ -476,16 +478,17 @@ static double get_time_in_ms()
return ((double)tv.tv_sec*1000)+((double)tv.tv_usec/1000);
}
-// Caller ID can be located in different positions between the rings depending on your Telco
-// Australian (Telstra) callerid starts 700ms after 1st ring and finishes 1.5s after first ring
-// Use ANALYSE_CID to record rings and determine location of callerid
-//#define ANALYSE_CID
+/* Caller ID can be located in different positions between the rings depending on your Telco
+ * Australian (Telstra) callerid starts 700ms after 1st ring and finishes 1.5s after first ring
+ * Use ANALYSE_CID to record rings and determine location of callerid
+ */
+/* #define ANALYSE_CID */
#define RING_SKIP 600
#define CID_MSECS 1700
static void get_callerid(struct vpb_pvt *p)
{
- short buf[CID_MSECS*8]; // 8kHz sampling rate
+ short buf[CID_MSECS*8]; /* 8kHz sampling rate */
double cid_record_time;
int rc;
struct ast_channel *owner = p->owner;
@@ -501,7 +504,7 @@ static void get_callerid(struct vpb_pvt *p)
if (option_verbose>3)
ast_verbose(VERBOSE_PREFIX_4 "CID record - start\n");
- // Skip any trailing ringtone
+ /* Skip any trailing ringtone */
vpb_sleep(RING_SKIP);
if (option_verbose>3)
@@ -509,7 +512,7 @@ static void get_callerid(struct vpb_pvt *p)
get_time_in_ms() - cid_record_time);
cid_record_time = get_time_in_ms();
- // Record bit between the rings which contains the callerid
+ /* Record bit between the rings which contains the callerid */
vpb_record_buf_start(p->handle, VPB_LINEAR);
rc = vpb_record_buf_sync(p->handle, (char*)buf, sizeof(buf));
vpb_record_buf_finish(p->handle);
@@ -533,7 +536,7 @@ static void get_callerid(struct vpb_pvt *p)
VPB_CID *cli_struct = new VPB_CID;
cli_struct->ra_cldn[0]=0;
cli_struct->ra_cn[0]=0;
- // This decodes FSK 1200baud type callerid
+ /* This decodes FSK 1200baud type callerid */
if ((rc=vpb_cid_decode2(cli_struct, buf, CID_MSECS*8)) == VPB_OK ) {
if (owner->cid.cid_num)
free(owner->cid.cid_num);
@@ -593,13 +596,13 @@ static void get_callerid_ast(struct vpb_pvt *p)
return;
}
if (option_verbose>3) ast_verbose(VERBOSE_PREFIX_4 "Collecting Caller ID type[%s/%d]...\n",p->callerid,which_cid);
-// vpb_sleep(RING_SKIP);
-// vpb_record_get_gain(p->handle, &old_gain);
+/* vpb_sleep(RING_SKIP); */
+/* vpb_record_get_gain(p->handle, &old_gain); */
cs = callerid_new(which_cid);
if (cs){
-// vpb_wave_open_write(&ws, file, VPB_MULAW);
-// vpb_record_set_gain(p->handle, 3.0);
-// vpb_record_set_hw_gain(p->handle,12.0);
+/* vpb_wave_open_write(&ws, file, VPB_MULAW); */
+/* vpb_record_set_gain(p->handle, 3.0); */
+/* vpb_record_set_hw_gain(p->handle,12.0); */
vpb_record_buf_start(p->handle, VPB_MULAW);
while((rc == 0)&&(sam_count<8000*3)){
vrc = vpb_record_buf_sync(p->handle, (char*)buf, sizeof(buf));
@@ -611,7 +614,7 @@ static void get_callerid_ast(struct vpb_pvt *p)
if (option_verbose>3) ast_verbose(VERBOSE_PREFIX_4 "Collecting Caller ID samples [%d][%d]...\n",sam_count,rc);
}
vpb_record_buf_finish(p->handle);
-// vpb_wave_close_write(ws);
+/* vpb_wave_close_write(ws); */
if (rc == 1){
callerid_get(cs, &name, &number, &flags);
if (option_verbose>0)
@@ -620,8 +623,8 @@ static void get_callerid_ast(struct vpb_pvt *p)
else {
ast_log(LOG_ERROR, "%s: Failed to decode Caller ID \n", p->dev );
}
-// vpb_record_set_gain(p->handle, old_gain);
-// vpb_record_set_hw_gain(p->handle,6.0);
+/* vpb_record_set_gain(p->handle, old_gain); */
+/* vpb_record_set_hw_gain(p->handle,6.0); */
}
else {
ast_log(LOG_ERROR, "%s: Failed to create Caller ID struct\n", p->dev );
@@ -647,7 +650,7 @@ static void get_callerid_ast(struct vpb_pvt *p)
callerid_free(cs);
}
-// Terminate any tones we are presently playing
+/* Terminate any tones we are presently playing */
static void stoptone( int handle)
{
int ret;
@@ -659,7 +662,7 @@ static void stoptone( int handle)
if (option_verbose > 3){
ast_verbose(VERBOSE_PREFIX_4 "Stop tone collected a wrong event!![%d]\n",je.type);
}
-// vpb_put_event(&je);
+/* vpb_put_event(&je); */
}
vpb_sleep(10);
}
@@ -736,8 +739,8 @@ static inline int monitor_handle_owned(struct vpb_pvt *p, VPB_EVENT *e)
}
} else if (e->data == VPB_GRUNT) {
if( ( get_time_in_ms() - p->lastgrunt ) > gruntdetect_timeout ) {
- // Nothing heard on line for a very long time
- // Timeout connection
+ /* Nothing heard on line for a very long time
+ * Timeout connection */
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "grunt timeout\n");
ast_log(LOG_NOTICE,"%s: Line hangup due of lack of conversation\n",p->dev);
@@ -786,8 +789,9 @@ static inline int monitor_handle_owned(struct vpb_pvt *p, VPB_EVENT *e)
f.subclass = AST_CONTROL_FLASH;
break;
- // Called when dialing has finished and ringing starts
- // No indication that call has really been answered when using blind dialing
+ /* Called when dialing has finished and ringing starts
+ * No indication that call has really been answered when using blind dialing
+ */
case VPB_DIALEND:
if (p->state < 5){
f.subclass = AST_CONTROL_ANSWER;
@@ -870,9 +874,10 @@ static inline int monitor_handle_owned(struct vpb_pvt *p, VPB_EVENT *e)
ast_verbose(VERBOSE_PREFIX_4 "%s: handle_owned: Prepared frame type[%d]subclass[%d], bridge=%p owner=[%s]\n",
p->dev, f.frametype, f.subclass, (void *)p->bridge, p->owner->name);
- // Trylock used here to avoid deadlock that can occur if we
- // happen to be in here handling an event when hangup is called
- // Problem is that hangup holds p->owner->lock
+ /* Trylock used here to avoid deadlock that can occur if we
+ * happen to be in here handling an event when hangup is called
+ * Problem is that hangup holds p->owner->lock
+ */
if ((f.frametype >= 0)&& (f.frametype != AST_FRAME_NULL)&&(p->owner)) {
if (ast_mutex_trylock(&p->owner->lock)==0) {
ast_queue_frame(p->owner, &f);
@@ -911,9 +916,9 @@ static inline int monitor_handle_notowned(struct vpb_pvt *p, VPB_EVENT *e)
if(!strcasecmp(p->callerid, "on")) {
if (option_verbose>3)
ast_verbose(VERBOSE_PREFIX_4 "Using VPB Caller ID\n");
- get_callerid(p); // Australian Caller ID only between 1st and 2nd ring
+ get_callerid(p); /* Australian Caller ID only between 1st and 2nd ring */
}
- get_callerid_ast(p); // Caller ID using the ast functions
+ get_callerid_ast(p); /* Caller ID using the ast functions */
}
break;
@@ -937,18 +942,18 @@ static inline int monitor_handle_notowned(struct vpb_pvt *p, VPB_EVENT *e)
if (p->state == VPB_STATE_PLAYDIAL) {
playtone(p->handle, &Dialtone);
p->wantdtmf = 1;
- p->ext[0] = 0; // Just to be sure & paranoid.
+ p->ext[0] = 0; /* Just to be sure & paranoid. */
}
/* These are not needed as they have timers to restart them
else if (p->state == VPB_STATE_PLAYBUSY) {
playtone(p->handle, &Busytone);
p->wantdtmf = 1;
- p->ext[0] = 0; // Just to be sure & paranoid.
+ p->ext[0] = 0; /* Just to be sure & paranoid. */
}
else if (p->state == VPB_STATE_PLAYRING) {
playtone(p->handle, &Ringbacktone);
p->wantdtmf = 1;
- p->ext[0] = 0; // Just to be sure & paranoid.
+ p->ext[0] = 0; /* Just to be sure & paranoid. */
}
*/
} else {
@@ -1203,7 +1208,7 @@ static int restart_monitor(void)
return error;
}
-// Per board config that must be called after vpb_open()
+/* Per board config that must be called after vpb_open() */
static void mkbrd(vpb_model_t model, int echo_cancel)
{
if(!bridges) {
@@ -1513,18 +1518,18 @@ static int vpb_call(struct ast_channel *ast, char *dest, int timeout)
else {
VPB_CALL call;
- // Dial must timeout or it can leave channels unuseable
+ /* Dial must timeout or it can leave channels unuseable */
if( timeout == 0 )
timeout = TIMER_PERIOD_NOANSWER;
else
- timeout = timeout * 1000; //convert from secs to ms.
-
- // These timeouts are only used with call progress dialing
- call.dialtones = 1; // Number of dialtones to get outside line
- call.dialtone_timeout = VPB_DIALTONE_WAIT; // Wait this long for dialtone (ms)
- call.ringback_timeout = VPB_RINGWAIT; // Wait this long for ringing after dialing (ms)
- call.inter_ringback_timeout = VPB_CONNECTED_WAIT; // If ringing stops for this long consider it connected (ms)
- call.answer_timeout = timeout; // Time to wait for answer after ringing starts (ms)
+ timeout = timeout * 1000; /* convert from secs to ms. */
+
+ /* These timeouts are only used with call progress dialing */
+ call.dialtones = 1; /* Number of dialtones to get outside line */
+ call.dialtone_timeout = VPB_DIALTONE_WAIT; /* Wait this long for dialtone (ms) */
+ call.ringback_timeout = VPB_RINGWAIT; /* Wait this long for ringing after dialing (ms) */
+ call.inter_ringback_timeout = VPB_CONNECTED_WAIT; /* If ringing stops for this long consider it connected (ms) */
+ call.answer_timeout = timeout; /* Time to wait for answer after ringing starts (ms) */
memcpy( &call.tone_map, DialToneMap, sizeof(DialToneMap) );
vpb_set_call(p->handle, &call);
@@ -1659,7 +1664,7 @@ static int vpb_hangup(struct ast_channel *ast)
stoptone(p->handle);
}
} else {
- stoptone(p->handle); // Terminates any dialing
+ stoptone(p->handle); /* Terminates any dialing */
vpb_sethook_sync(p->handle, VPB_ONHOOK);
p->state=VPB_STATE_ONHOOK;
}
@@ -1738,15 +1743,17 @@ static int vpb_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
if(option_verbose>1)
-// ast_verbose( VERBOSE_PREFIX_2 "%s: Answered call from %s on %s [%s]\n", p->dev,
-// p->owner->callerid, ast->name,(p->mode == MODE_FXO)?"FXO":"FXS");
+/*
+ ast_verbose( VERBOSE_PREFIX_2 "%s: Answered call from %s on %s [%s]\n", p->dev,
+ p->owner->callerid, ast->name,(p->mode == MODE_FXO)?"FXO":"FXS");
+*/
ast_verbose( VERBOSE_PREFIX_2 "%s: Answered call on %s [%s]\n", p->dev,
ast->name,(p->mode == MODE_FXO)?"FXO":"FXS");
ast->rings = 0;
if( !p->readthread ){
- // res = ast_mutex_unlock(&p->lock);
- // ast_verbose("%s: unLOCKING in answer [%d]\n", p->dev,res);
+ /* res = ast_mutex_unlock(&p->lock); */
+ /* ast_verbose("%s: unLOCKING in answer [%d]\n", p->dev,res); */
ast_pthread_create(&p->readthread, NULL, do_chanreads, (void *)p);
} else {
if(option_verbose>3)
@@ -1756,8 +1763,8 @@ static int vpb_answer(struct ast_channel *ast)
if(option_verbose>3) {
ast_verbose(VERBOSE_PREFIX_4 "%s: Answered state is up\n",p->dev);
}
- // res = ast_mutex_unlock(&p->lock);
- // ast_verbose("%s: unLOCKING in answer [%d]\n", p->dev,res);
+ /* res = ast_mutex_unlock(&p->lock); */
+ /* ast_verbose("%s: unLOCKING in answer [%d]\n", p->dev,res); */
}
vpb_sleep(500);
if (p->mode == MODE_FXO){
@@ -1852,34 +1859,34 @@ static int vpb_write(struct ast_channel *ast, struct ast_frame *frame)
struct vpb_pvt *p = (struct vpb_pvt *)ast->pvt->pvt;
int res = 0, fmt = 0;
struct timeval play_buf_time_start,play_buf_time_finish;
-// ast_mutex_lock(&p->lock);
+/* ast_mutex_lock(&p->lock); */
if(option_verbose>5)
ast_verbose("%s: vpb_write: Writing to channel\n", p->dev);
if (frame->frametype != AST_FRAME_VOICE) {
if(option_verbose>3)
ast_verbose("%s: vpb_write: Don't know how to handle from type %d\n", ast->name, frame->frametype);
-// ast_mutex_unlock(&p->lock);
+/* ast_mutex_unlock(&p->lock); */
return 0;
} else if (ast->_state != AST_STATE_UP) {
if(option_verbose>3)
ast_verbose("%s: vpb_write: Attempt to Write frame type[%d]subclass[%d] on not up chan\n",ast->name, frame->frametype, frame->subclass);
p->lastoutput = -1;
-// ast_mutex_unlock(&p->lock);
+/* ast_mutex_unlock(&p->lock); */
return 0;
}
-// ast_log(LOG_DEBUG, "%s: vpb_write: Checked frame type..\n", p->dev);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Checked frame type..\n", p->dev); */
fmt = ast2vpbformat(frame->subclass);
if (fmt < 0) {
ast_log(LOG_WARNING, "%s: vpb_write: Cannot handle frames of %d format!\n",ast->name, frame->subclass);
return -1;
}
-// ast_log(LOG_DEBUG, "%s: vpb_write: Checked frame format..\n", p->dev);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Checked frame format..\n", p->dev); */
ast_mutex_lock(&p->play_lock);
-// ast_log(LOG_DEBUG, "%s: vpb_write: Got play lock..\n", p->dev);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Got play lock..\n", p->dev); */
/* Check if we have set up the play_buf */
if (p->lastoutput == -1) {
@@ -1897,14 +1904,14 @@ static int vpb_write(struct ast_channel *ast, struct ast_frame *frame)
- // Apply extra gain !
+ /* Apply extra gain ! */
if( p->txswgain > MAX_VPB_GAIN )
a_gain_vector(p->txswgain - MAX_VPB_GAIN , (short*)frame->data, frame->datalen/sizeof(short));
-// ast_log(LOG_DEBUG, "%s: vpb_write: Applied gain..\n", p->dev);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Applied gain..\n", p->dev); */
-// gettimeofday(&tv, NULL);
-// return ((double)tv.tv_sec*1000)+((double)tv.tv_usec/1000);
+/* gettimeofday(&tv, NULL); */
+/* return ((double)tv.tv_sec*1000)+((double)tv.tv_usec/1000); */
if ((p->read_state == 1)&&(p->play_buf_time<5)){
gettimeofday(&play_buf_time_start,NULL);
@@ -1916,12 +1923,12 @@ static int vpb_write(struct ast_channel *ast, struct ast_frame *frame)
gettimeofday(&play_buf_time_finish,NULL);
if (play_buf_time_finish.tv_sec == play_buf_time_start.tv_sec){
p->play_buf_time=(int)((play_buf_time_finish.tv_usec-play_buf_time_start.tv_usec)/1000);
-// ast_log(LOG_DEBUG, "%s: vpb_write: Timing start(%d) finish(%d)\n", p->dev,play_buf_time_start.tv_usec,play_buf_time_finish.tv_usec);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Timing start(%d) finish(%d)\n", p->dev,play_buf_time_start.tv_usec,play_buf_time_finish.tv_usec); */
}
else {
p->play_buf_time=(int)((play_buf_time_finish.tv_sec - play_buf_time_start.tv_sec)*100)+(int)((play_buf_time_finish.tv_usec-play_buf_time_start.tv_usec)/1000);
}
-// ast_log(LOG_DEBUG, "%s: vpb_write: Wrote data [%d](%d=>%s) to play_buf in [%d]ms..\n", p->dev,frame->datalen,fmt,ast2vpbformatname(frame->subclass),p->play_buf_time);
+/* ast_log(LOG_DEBUG, "%s: vpb_write: Wrote data [%d](%d=>%s) to play_buf in [%d]ms..\n", p->dev,frame->datalen,fmt,ast2vpbformatname(frame->subclass),p->play_buf_time); */
}
else {
p->chuck_count++;
@@ -1930,7 +1937,7 @@ static int vpb_write(struct ast_channel *ast, struct ast_frame *frame)
}
ast_mutex_unlock(&p->play_lock);
-// ast_mutex_unlock(&p->lock);
+/* ast_mutex_unlock(&p->lock); */
if(option_verbose>5)
ast_verbose("%s: vpb_write: Done Writing to channel\n", p->dev);
return 0;
@@ -1990,7 +1997,7 @@ static void *do_chanreads(void *pvt)
}
}
-// if ( (p->owner->_state != AST_STATE_UP) || !bridgerec) {
+/* if ( (p->owner->_state != AST_STATE_UP) || !bridgerec) { */
if ( (p->owner->_state != AST_STATE_UP) ) {
if (option_verbose > 4) {
if (p->owner->_state != AST_STATE_UP)
@@ -2002,10 +2009,11 @@ static void *do_chanreads(void *pvt)
continue;
}
- // Voicetronix DTMF detection can be triggered off ordinary speech
- // This leads to annoying beeps during the conversation
- // Avoid this problem by just setting VPB_GETDTMF when you want to listen for DTMF
- //ignore_dtmf = 1;
+ /* Voicetronix DTMF detection can be triggered off ordinary speech
+ * This leads to annoying beeps during the conversation
+ * Avoid this problem by just setting VPB_GETDTMF when you want to listen for DTMF
+ */
+ /* ignore_dtmf = 1; */
ignore_dtmf = 0; /* set this to 1 to turn this feature on */
getdtmf_var = pbx_builtin_getvar_helper(p->owner,"VPB_GETDTMF");
if( getdtmf_var && ( strcasecmp( getdtmf_var, "yes" ) == 0 ) )
@@ -2019,15 +2027,16 @@ static void *do_chanreads(void *pvt)
}
p->last_ignore_dtmf = ignore_dtmf;
- // Play DTMF digits here to avoid problem you get if playing a digit during
- // a record operation
+ /* Play DTMF digits here to avoid problem you get if playing a digit during
+ * a record operation
+ */
if (option_verbose > 5) {
ast_verbose("%s: chanreads: Checking dtmf's \n", p->dev);
}
ast_mutex_lock(&p->play_dtmf_lock);
if( p->play_dtmf[0] ) {
- // Try to ignore DTMF event we get after playing digit
- // This DTMF is played by asterisk and leads to an annoying trailing beep on CISCO phones
+ /* Try to ignore DTMF event we get after playing digit */
+ /* This DTMF is played by asterisk and leads to an annoying trailing beep on CISCO phones */
if( !ignore_dtmf)
vpb_set_event_mask(p->handle, VPB_EVENTS_NODTMF );
if (strcmp(p->owner->type,"vpb")==0){
@@ -2041,21 +2050,21 @@ static void *do_chanreads(void *pvt)
}
p->play_dtmf[0] = '\0';
ast_mutex_unlock(&p->play_dtmf_lock);
- vpb_sleep(700); // Long enough to miss echo and DTMF event
+ vpb_sleep(700); /* Long enough to miss echo and DTMF event */
if( !ignore_dtmf)
vpb_set_event_mask(p->handle, VPB_EVENTS_ALL );
continue;
}
ast_mutex_unlock(&p->play_dtmf_lock);
-// afmt = (p->owner) ? p->owner->pvt->rawreadformat : AST_FORMAT_SLINEAR;
+/* afmt = (p->owner) ? p->owner->pvt->rawreadformat : AST_FORMAT_SLINEAR; */
if (p->owner){
afmt = p->owner->pvt->rawreadformat;
-// ast_log(LOG_DEBUG,"%s: Record using owner format [%s]\n", p->dev, ast2vpbformatname(afmt));
+/* ast_log(LOG_DEBUG,"%s: Record using owner format [%s]\n", p->dev, ast2vpbformatname(afmt)); */
}
else {
afmt = AST_FORMAT_SLINEAR;
-// ast_log(LOG_DEBUG,"%s: Record using default format [%s]\n", p->dev, ast2vpbformatname(afmt));
+/* ast_log(LOG_DEBUG,"%s: Record using default format [%s]\n", p->dev, ast2vpbformatname(afmt)); */
}
fmt = ast2vpbformat(afmt);
if (fmt < 0) {
@@ -2085,7 +2094,7 @@ static void *do_chanreads(void *pvt)
if (option_verbose > 5) {
ast_verbose("%s: chanreads: got buffer!\n", p->dev);
}
- // Apply extra gain !
+ /* Apply extra gain ! */
if( p->rxswgain > MAX_VPB_GAIN )
a_gain_vector(p->rxswgain - MAX_VPB_GAIN , (short*)readbuf, readlen/sizeof(short));
if (option_verbose > 5) {
@@ -2096,8 +2105,9 @@ static void *do_chanreads(void *pvt)
fr->data = readbuf;
fr->datalen = readlen;
- // Using trylock here to prevent deadlock when channel is hungup
- // (ast_hangup() immediately gets lock)
+ /* Using trylock here to prevent deadlock when channel is hungup
+ * (ast_hangup() immediately gets lock)
+ */
if (p->owner && !p->stopreads ) {
if (option_verbose > 5) {
ast_verbose("%s: chanreads: queueing buffer on read frame q (state[%d])\n", p->dev,p->owner->_state);
@@ -2117,7 +2127,7 @@ static void *do_chanreads(void *pvt)
/*
res = ast_mutex_trylock(&p->owner->lock);
-// res=0;
+/* res=0; */
if (res==0) {
ast_queue_frame(p->owner, fr);
ast_mutex_unlock(&p->owner->lock);
@@ -2139,7 +2149,7 @@ static void *do_chanreads(void *pvt)
else if (res == EBUSY )
if (option_verbose > 4) ast_verbose("%s: chanreads: try owner->lock gave me EBUSY[%d]\n", p->dev,res);
if (option_verbose > 4) ast_verbose("%s: chanreads: Couldnt get lock on owner[%s][%d][%d] channel to send frame!\n", p->dev,p->owner->name,(int)p->owner->lock.__m_owner,(int)p->owner->lock.__m_count);
- //assert(p->dev!=p->dev);
+ /*assert(p->dev!=p->dev); */
}
}
*/
@@ -2190,9 +2200,10 @@ static struct ast_channel *vpb_new(struct vpb_pvt *me, int state, char *context)
strncpy(tmp->name, me->dev, sizeof(tmp->name) - 1);
tmp->type = type;
- // Linear is the preferred format. Although Voicetronix supports other formats
- // they are all converted to/from linear in the vpb code. Best for us to use
- // linear since we can then adjust volume in this modules.
+ /* Linear is the preferred format. Although Voicetronix supports other formats
+ * they are all converted to/from linear in the vpb code. Best for us to use
+ * linear since we can then adjust volume in this modules.
+ */
tmp->nativeformats = prefformat;
tmp->pvt->rawreadformat = AST_FORMAT_SLINEAR;
tmp->pvt->rawwriteformat = AST_FORMAT_SLINEAR;
@@ -2317,8 +2328,8 @@ static float parse_gain_value(char *gain_type, char *value)
/* percentage? */
- //if (value[strlen(value) - 1] == '%')
- // return gain / (float)100;
+ /*if (value[strlen(value) - 1] == '%') */
+ /* return gain / (float)100; */
return gain;
}
@@ -2335,7 +2346,7 @@ int load_module()
int first_channel = 1;
int echo_cancel = DEFAULT_ECHO_CANCEL;
int error = 0; /* Error flag */
- int bal1 = -1; // Special value - means do not set
+ int bal1 = -1; /* Special value - means do not set */
int bal2 = -1;
int bal3 = -1;
char * callerid = NULL;
diff --git a/formats/format_pcm.c b/formats/format_pcm.c
index 6d01271772..14edd0bb6a 100755
--- a/formats/format_pcm.c
+++ b/formats/format_pcm.c
@@ -177,7 +177,7 @@ static int pcm_seek(struct ast_filestream *fs, long sample_offset, int whence)
if (whence != SEEK_FORCECUR) {
offset = (offset > max)?max:offset;
}
- // always protect against seeking past begining.
+ /* always protect against seeking past begining. */
offset = (offset < min)?min:offset;
return lseek(fs->fd, offset, SEEK_SET);
}
diff --git a/formats/format_pcm_alaw.c b/formats/format_pcm_alaw.c
index 866cd23607..b677bb6fef 100755
--- a/formats/format_pcm_alaw.c
+++ b/formats/format_pcm_alaw.c
@@ -3,9 +3,9 @@
*
* Flat, binary, alaw PCM file format.
*
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999, Digium, inc
*
- * Mark Spencer <markster@linux-support.net>
+ * Mark Spencer <markster@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
@@ -39,7 +39,7 @@
#define BUF_SIZE 160 /* 160 samples */
-// #define REALTIME_WRITE
+/* #define REALTIME_WRITE */
struct ast_filestream {
void *reserved[AST_RESERVED_POINTERS];
@@ -192,18 +192,19 @@ static int pcm_write(struct ast_filestream *fs, struct ast_frame *f)
#ifdef REALTIME_WRITE
cur_time = get_time();
- fpos = ( cur_time - fs->start_time ) * 8; // 8 bytes per msec
- // Check if we have written to this position yet. If we have, then increment pos by one frame
- // for some degree of protection against receiving packets in the same clock tick.
+ fpos = ( cur_time - fs->start_time ) * 8; /* 8 bytes per msec */
+ /* Check if we have written to this position yet. If we have, then increment pos by one frame
+ * for some degree of protection against receiving packets in the same clock tick.
+ */
fstat( fs->fd, &stat_buf );
if( stat_buf.st_size > fpos )
{
- fpos += f->datalen; // Incrementing with the size of this current frame
+ fpos += f->datalen; /* Incrementing with the size of this current frame */
}
if( stat_buf.st_size < fpos )
{
- // fill the gap with 0x55 rather than 0.
+ /* fill the gap with 0x55 rather than 0. */
char buf[ 512 ];
unsigned long cur, to_write;
@@ -232,7 +233,7 @@ static int pcm_write(struct ast_filestream *fs, struct ast_frame *f)
ast_log( LOG_WARNING, "Cannot seek in file: %s\n", strerror(errno) );
return -1;
}
-#endif // REALTIME_WRITE
+#endif /* REALTIME_WRITE */
if ((res = write(fs->fd, f->data, f->datalen)) != f->datalen) {
ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
@@ -257,7 +258,7 @@ static int pcm_seek(struct ast_filestream *fs, long sample_offset, int whence)
if (whence != SEEK_FORCECUR) {
offset = (offset > max)?max:offset;
}
- // Always protect against seeking past begining
+ /* Always protect against seeking past begining */
offset = (offset < min)?min:offset;
return lseek(fs->fd, offset, SEEK_SET);
}
diff --git a/formats/format_wav.c b/formats/format_wav.c
index be60534473..c0781f6f03 100755
--- a/formats/format_wav.c
+++ b/formats/format_wav.c
@@ -177,12 +177,12 @@ static int check_header(int fd)
ast_log(LOG_WARNING, "Read failed (Bits Per Sample): %d\n", ltohs(bisam));
return -1;
}
- // Skip any additional header
+ /* Skip any additional header */
if ( lseek(fd,ltohl(hsize)-16,SEEK_CUR) == -1 ) {
ast_log(LOG_WARNING, "Failed to skip remaining header bytes: %d\n", ltohl(hsize)-16 );
return -1;
}
- // Skip any facts and get the first data block
+ /* Skip any facts and get the first data block */
for(;;)
{
char buf[4];
@@ -529,7 +529,7 @@ static int wav_seek(struct ast_filestream *fs, long sample_offset, int whence)
if (whence != SEEK_FORCECUR) {
offset = (offset > max)?max:offset;
}
- // always protect the header space.
+ /* always protect the header space. */
offset = (offset < min)?min:offset;
return lseek(fs->fd,offset,SEEK_SET);
}
diff --git a/logger.c b/logger.c
index 5fb0348afd..84783f798d 100755
--- a/logger.c
+++ b/logger.c
@@ -424,7 +424,7 @@ static int handle_logger_reload(int fd, int argc, char *argv[])
{
if(reload_logger(0))
{
- ast_cli(fd, "Failed to reloadthe logger\n");
+ ast_cli(fd, "Failed to reload the logger\n");
return RESULT_FAILURE;
}
else
@@ -435,7 +435,7 @@ static int handle_logger_rotate(int fd, int argc, char *argv[])
{
if(reload_logger(1))
{
- ast_cli(fd, "Failed to reloadthe logger\n");
+ ast_cli(fd, "Failed to reload the logger and rotate log files\n");
return RESULT_FAILURE;
}
else
diff --git a/pbx.c b/pbx.c
index 44a57d3a18..09a87d2d00 100755
--- a/pbx.c
+++ b/pbx.c
@@ -64,18 +64,18 @@ struct ast_context;
/* ast_exten: An extension */
struct ast_exten {
- char *exten; /* Extension name */
- int matchcid; /* Match caller id ? */
- char *cidmatch; /* Caller id to match for this extension */
- int priority; /* Priority */
- char *label; /* Label */
- struct ast_context *parent; /* An extension */
- char *app; /* Application to execute */
- void *data; /* Data to use */
- void (*datad)(void *); /* Data destructor */
- struct ast_exten *peer; /* Next higher priority with our extension */
- const char *registrar; /* Registrar */
- struct ast_exten *next; /* Extension with a greater ID */
+ char *exten; /* Extension name */
+ int matchcid; /* Match caller id ? */
+ char *cidmatch; /* Caller id to match for this extension */
+ int priority; /* Priority */
+ char *label; /* Label */
+ struct ast_context *parent; /* The context this extension belongs to */
+ char *app; /* Application to execute */
+ void *data; /* Data to use (arguments) */
+ void (*datad)(void *); /* Data destructor */
+ struct ast_exten *peer; /* Next higher priority with our extension */
+ const char *registrar; /* Registrar */
+ struct ast_exten *next; /* Extension with a greater ID */
char stuff[0];
};
diff --git a/res/res_agi.c b/res/res_agi.c
index d7582ba447..e7bced29a7 100755
--- a/res/res_agi.c
+++ b/res/res_agi.c
@@ -276,7 +276,7 @@ static int launch_script(char *script, char *argv[], int *fds, int *efd, int *op
close(fromast[0]);
if (efd) {
- // [PHM 12/18/03]
+ /* [PHM 12/18/03] */
close(audio[0]);
}
@@ -484,7 +484,7 @@ static int handle_getoption(struct ast_channel *chan, AGI *agi, int argc, char *
timeout = atoi(argv[4]);
else if (chan->pbx->dtimeout) {
/* by default dtimeout is set to 5sec */
- timeout = chan->pbx->dtimeout * 1000; //in msec
+ timeout = chan->pbx->dtimeout * 1000; /* in msec */
}
fs = ast_openstream(chan, argv[2], chan->language);
diff --git a/res/res_config_odbc.c b/res/res_config_odbc.c
index 144ca550b5..485a792e7f 100755
--- a/res/res_config_odbc.c
+++ b/res/res_config_odbc.c
@@ -425,7 +425,7 @@ static struct ast_config *config_odbc (const char *database, const char *table,
if (!file || !strcmp (file, "res_config_odbc.conf"))
- return NULL; // cant configure myself with myself !
+ return NULL; /* cant configure myself with myself ! */
obj = fetch_odbc_obj(database);
if (!obj)
@@ -518,7 +518,7 @@ static struct ast_config *config_odbc (const char *database, const char *table,
}
}
- // next row
+ /* next row */
res = SQLFetch (stmt);
}
diff --git a/res/res_indications.c b/res/res_indications.c
index b289612d5d..c89426328f 100755
--- a/res/res_indications.c
+++ b/res/res_indications.c
@@ -33,7 +33,7 @@
#include <asterisk/indications.h>
-// Globals
+/* Globals */
static const char dtext[] = "Indications Configuration";
static const char config[] = "indications.conf";
@@ -41,11 +41,11 @@ static const char config[] = "indications.conf";
* Help for commands provided by this module ...
*/
static char help_add_indication[] =
-"Usage: add indication <country> <indication> \"<tonelist>\"\n"
+"Usage: indication add <country> <indication> \"<tonelist>\"\n"
" Add the given indication to the country.\n";
static char help_remove_indication[] =
-"Usage: remove indication <country> <indication>\n"
+"Usage: indication remove <country> <indication>\n"
" Remove the given indication from the country.\n";
static char help_show_indications[] =
@@ -292,7 +292,7 @@ static int ind_load_module(void)
country = strsep(&c,",");
}
} else {
- // add tone to country
+ /* add tone to country */
struct tone_zone_sound *ps,*ts;
for (ps=NULL,ts=tones->tones; ts; ps=ts, ts=ts->next) {
if (strcasecmp(v->name,ts->name)==0) {
@@ -341,12 +341,12 @@ out: v = v->next;
* CLI entries for commands provided by this module
*/
static struct ast_cli_entry add_indication_cli =
- { { "add", "indication", NULL }, handle_add_indication,
+ { { "indication", "add", NULL }, handle_add_indication,
"Add the given indication to the country", help_add_indication,
NULL };
static struct ast_cli_entry remove_indication_cli =
- { { "remove", "indication", NULL }, handle_remove_indication,
+ { { "indication", "remove", NULL }, handle_remove_indication,
"Remove the given indication from the country", help_remove_indication,
NULL };
@@ -367,8 +367,8 @@ int unload_module(void)
ast_cli_unregister(&add_indication_cli);
ast_cli_unregister(&remove_indication_cli);
ast_cli_unregister(&show_indications_cli);
- ast_unregister_application("Playlist");
- ast_unregister_application("StopPlaylist");
+ ast_unregister_application("Playtones");
+ ast_unregister_application("StopPlaytones");
return 0;
}
diff --git a/res/res_monitor.c b/res/res_monitor.c
index e2cec0dacb..ea14aafa1d 100755
--- a/res/res_monitor.c
+++ b/res/res_monitor.c
@@ -4,7 +4,7 @@
#include <string.h>
#include <sys/types.h>
#include <sys/stat.h>
-#include <libgen.h> //dirname()
+#include <libgen.h> /* dirname() */
#include <asterisk/lock.h>
#include <asterisk/channel.h>
@@ -65,7 +65,7 @@ static char *changemonitor_descrip = "ChangeMonitor\n"
/* Start monitoring a channel */
int ast_monitor_start( struct ast_channel *chan, const char *format_spec,
- const char *fname_base, int need_lock)
+ const char *fname_base, int need_lock)
{
int res = 0;
char tmp[256];
@@ -85,7 +85,7 @@ int ast_monitor_start( struct ast_channel *chan, const char *format_spec,
if (mkdir(AST_MONITOR_DIR, 0770) < 0) {
if (errno != EEXIST) {
ast_log(LOG_WARNING, "Unable to create audio monitor directory: %s\n",
- strerror(errno));
+ strerror(errno));
}
}
@@ -133,14 +133,14 @@ int ast_monitor_start( struct ast_channel *chan, const char *format_spec,
monitor->stop = ast_monitor_stop;
- // Determine file format
+ /* Determine file format */
if (format_spec && !ast_strlen_zero(format_spec)) {
monitor->format = strdup(format_spec);
} else {
monitor->format = strdup("wav");
}
- // open files
+ /* open files */
if (ast_fileexists(monitor->read_filename, NULL, NULL) > 0) {
ast_filedelete(monitor->read_filename, NULL);
}
@@ -410,7 +410,7 @@ static int start_monitor_action(struct mansession *s, struct message *m)
}
if ((!fname) || (ast_strlen_zero(fname))) {
- // No filename base specified, default to channel name as per CLI
+ /* No filename base specified, default to channel name as per CLI */
fname = malloc (FILENAME_MAX);
if (!fname) {
astman_send_error(s, m, "Could not start monitoring channel");
@@ -419,7 +419,7 @@ static int start_monitor_action(struct mansession *s, struct message *m)
}
memset(fname, 0, FILENAME_MAX);
strncpy(fname, c->name, FILENAME_MAX-1);
- // Channels have the format technology/channel_name - have to replace that /
+ /* Channels have the format technology/channel_name - have to replace that / */
if ((d=strchr(fname, '/'))) *d='-';
}
--
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