[asterisk-commits] russell: branch russell/chan_console r49058 - in
/team/russell/chan_console: ...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Dec 29 06:27:20 MST 2006
Author: russell
Date: Fri Dec 29 07:27:19 2006
New Revision: 49058
URL: http://svn.digium.com/view/asterisk?view=rev&rev=49058
Log:
Merged revisions 49047,49053-49054,49056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r49047 | kpfleming | 2006-12-28 19:33:33 -0500 (Thu, 28 Dec 2006) | 18 lines
Merged revisions 49046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r49046 | kpfleming | 2006-12-28 18:32:59 -0600 (Thu, 28 Dec 2006) | 10 lines
Merged revisions 49045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) | 2 lines
location of the bug posting guidelines has changed
........
................
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r49053 | russell | 2006-12-29 01:26:53 -0500 (Fri, 29 Dec 2006) | 2 lines
Fix a spelling mistake in a comment.
................
r49054 | oej | 2006-12-29 06:02:28 -0500 (Fri, 29 Dec 2006) | 2 lines
Removing extra output
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r49056 | russell | 2006-12-29 08:25:07 -0500 (Fri, 29 Dec 2006) | 2 lines
Convert various comments to doxygen format.
................
Modified:
team/russell/chan_console/ (props changed)
team/russell/chan_console/BUGS
team/russell/chan_console/channels/chan_oss.c
team/russell/chan_console/channels/chan_sip.c
team/russell/chan_console/include/asterisk/smdi.h
Propchange: team/russell/chan_console/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/russell/chan_console/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Dec 29 07:27:19 2006
@@ -1,1 +1,1 @@
-/trunk:1-49038
+/trunk:1-49057
Modified: team/russell/chan_console/BUGS
URL: http://svn.digium.com/view/asterisk/team/russell/chan_console/BUGS?view=diff&rev=49058&r1=49057&r2=49058
==============================================================================
--- team/russell/chan_console/BUGS (original)
+++ team/russell/chan_console/BUGS Fri Dec 29 07:27:19 2006
@@ -10,7 +10,7 @@
learn how you can contribute by acting as a bug marshall
please see:
- http://www.digium.com/index.php?menu=bugguidelines
+ http://www.asterisk.org/developers/bug-guidelines
If you would like to submit a feature request, please
resist the temptation to post it to the bug tracker.
Modified: team/russell/chan_console/channels/chan_oss.c
URL: http://svn.digium.com/view/asterisk/team/russell/chan_console/channels/chan_oss.c?view=diff&rev=49058&r1=49057&r2=49058
==============================================================================
--- team/russell/chan_console/channels/chan_oss.c (original)
+++ team/russell/chan_console/channels/chan_oss.c Fri Dec 29 07:27:19 2006
@@ -285,7 +285,7 @@
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.
@@ -310,8 +310,9 @@
};
-/*
- * descriptor for one of our channels.
+/*!
+ * \brief 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
@@ -321,45 +322,45 @@
struct chan_oss_pvt *next;
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 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 */
+ 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 */
+ int w_errors; /*!< overfull in the write path */
struct timeval lastopen;
int overridecontext;
int mute;
- /* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
- * be representable in 16 bits to avoid overflows.
+ /*! boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
+ * be representable in 16 bits to avoid overflows.
*/
#define BOOST_SCALE (1<<9)
-#define BOOST_MAX 40 /* slightly less than 7 bits */
- int boost; /* input boost, scaled by BOOST_SCALE */
- char device[64]; /* device to open */
+#define BOOST_MAX 40 /*!< slightly less than 7 bits */
+ int boost; /*!< input boost, scaled by BOOST_SCALE */
+ char device[64]; /*!< device to open */
pthread_t sthread;
@@ -371,15 +372,15 @@
char cid_num[256]; /*XXX */
char mohinterpret[MAX_MUSICCLASS];
- /* buffers used in oss_write */
+ /*! 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
+ /*! 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 */
+ int readpos; /*!< read position above */
+ struct ast_frame read_f; /*!< returned by oss_read */
};
static struct chan_oss_pvt oss_default = {
@@ -397,7 +398,7 @@
.boost = BOOST_SCALE,
};
-static char *oss_active; /* the active device */
+static char *oss_active; /*!< the active device */
static int setformat(struct chan_oss_pvt *o, int mode);
@@ -432,8 +433,8 @@
.fixup = oss_fixup,
};
-/*
- * returns a pointer to the descriptor with the given name
+/*!
+ * \brief returns a pointer to the descriptor with the given name
*/
static struct chan_oss_pvt *find_desc(char *dev)
{
@@ -450,14 +451,16 @@
return o;
}
-/*
- * split a string in extension-context, returns pointers to malloc'ed
- * strings.
+/* !
+ * \brief split a string in extension-context, returns pointers to malloc'ed
+ * strings.
+ *
* If we do not 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.
+ *
+ * \return the buffer address.
*/
static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
{
@@ -484,8 +487,8 @@
return *ext;
}
-/*
- * Returns the number of blocks used in the audio output channel
+/*!
+ * \brief Returns the number of blocks used in the audio output channel
*/
static int used_blocks(struct chan_oss_pvt *o)
{
@@ -508,7 +511,7 @@
return o->total_blocks - info.fragments;
}
-/* Write an exactly FRAME_SIZE sized frame */
+/*! Write an exactly FRAME_SIZE sized frame */
static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
{
int res;
@@ -533,8 +536,9 @@
return write(o->sounddev, (void *)data, FRAME_SIZE * 2);
}
-/*
- * Handler for 'sound writable' events from the sound thread.
+/*!
+ * \brief 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
@@ -661,7 +665,7 @@
return NULL; /* Never reached */
}
-/*
+/*!
* 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.
@@ -785,15 +789,15 @@
return 0;
}
-/* Play ringtone 'x' on device 'o' */
+/*! \brief 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
+/*!
+ * \brief handler for incoming calls. Either autoanswer, or start ringing
*/
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
@@ -816,8 +820,8 @@
return 0;
}
-/*
- * remote side answered the phone
+/*!
+ * \brief remote side answered the phone
*/
static int oss_answer(struct ast_channel *c)
{
@@ -856,7 +860,7 @@
return 0;
}
-/* used for data coming from the network */
+/*! \brief used for data coming from the network */
static int oss_write(struct ast_channel *c, struct ast_frame *f)
{
int src;
@@ -991,8 +995,8 @@
return 0;
}
-/*
- * allocate a new channel.
+/*!
+ * \brief allocate a new channel.
*/
static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state)
{
@@ -1104,8 +1108,8 @@
return CLI_SUCCESS;
}
-/*
- * answer command from the console
+/*!
+ * \brief answer command from the console
*/
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
@@ -1136,8 +1140,10 @@
return CLI_SUCCESS;
}
-/*
- * concatenate all arguments into a single string. argv is NULL-terminated
+/*!
+ * \brief Console send text CLI command
+ *
+ * \note concatenate all arguments into a single string. argv is NULL-terminated
* so we can use it right away
*/
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
@@ -1369,8 +1375,8 @@
"console. If a device is specified, the console sound device is changed to\n"
"the device specified.\n";
-/*
- * store the boost factor
+/*!
+ * \brief store the boost factor
*/
static void store_boost(struct chan_oss_pvt *o, char *s)
{
@@ -1424,7 +1430,7 @@
active_usage },
};
-/*
+/*!
* store the mixer argument from the config file, filtering possibly
* invalid or dangerous values (the string is used as argument for
* system("mixer %s")
@@ -1445,7 +1451,7 @@
ast_log(LOG_WARNING, "setting mixer %s\n", s);
}
-/*
+/*!
* store the callerid components
*/
static void store_callerid(struct chan_oss_pvt *o, char *s)
@@ -1453,7 +1459,7 @@
ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
}
-/*
+/*!
* 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)
Modified: team/russell/chan_console/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/russell/chan_console/channels/chan_sip.c?view=diff&rev=49058&r1=49057&r2=49058
==============================================================================
--- team/russell/chan_console/channels/chan_sip.c (original)
+++ team/russell/chan_console/channels/chan_sip.c Fri Dec 29 07:27:19 2006
@@ -14668,7 +14668,6 @@
p->method = req->method; /* Find out which SIP method they are using */
if (option_debug > 3)
ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
- ast_verbose("**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
if (p->icseq && (p->icseq > seqno)) {
if (option_debug)
Modified: team/russell/chan_console/include/asterisk/smdi.h
URL: http://svn.digium.com/view/asterisk/team/russell/chan_console/include/asterisk/smdi.h?view=diff&rev=49058&r1=49057&r2=49058
==============================================================================
--- team/russell/chan_console/include/asterisk/smdi.h (original)
+++ team/russell/chan_console/include/asterisk/smdi.h Fri Dec 29 07:27:19 2006
@@ -88,7 +88,7 @@
*
* The ast_smdi_interface structure holds information on a serial port that
* should be monitored for SMDI activity. The structure contains a message
- * queue of messages that have been recieved on the interface.
+ * queue of messages that have been received on the interface.
*/
struct ast_smdi_interface {
ASTOBJ_COMPONENTS_FULL(struct ast_smdi_interface, SMDI_MAX_FILENAME_LEN, 1);
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