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Convert various comments to doxygen format.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2006-12-29 13:25:07 +00:00
parent fc86ea2eed
commit 1f8001e816
1 changed files with 60 additions and 54 deletions
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114
channels/chan_oss.c
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@@ -285,7 +285,7 @@ static char *config = "oss.conf"; /* default config file */
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 @@ static struct sound sounds[] = {
};
/*
* 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 {
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 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 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 */
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 */
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 @@ struct chan_oss_pvt {
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 @@ static struct chan_oss_pvt oss_default = {
.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 @@ static const struct ast_channel_tech oss_tech = {
.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 @@ static struct chan_oss_pvt *find_desc(char *dev)
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 @@ static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
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 @@ static int used_blocks(struct chan_oss_pvt *o)
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 @@ static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
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 @@ static void *sound_thread(void *arg)
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 @@ static int oss_text(struct ast_channel *c, const char *text)
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 @@ static int oss_call(struct ast_channel *c, char *dest, int timeout)
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 @@ static int oss_hangup(struct ast_channel *c)
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 @@ static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_
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 @@ static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli
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 @@ static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
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 @@ static const char active_usage[] =
"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 @@ static struct ast_cli_entry cli_oss[] = {
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 @@ static void store_mixer(struct chan_oss_pvt *o, char *s)
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 @@ static void store_callerid(struct chan_oss_pvt *o, char *s)
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)