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Issue 8246 Doxygen updates (kshumard)

THANK YOU!



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46434 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2006-10-30 16:33:02 +00:00
parent 0ed1ad8e8d
commit 0b22be3874
12 changed files with 34 additions and 26 deletions

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@ -936,6 +936,7 @@ static int make_file(char *dest, int len, char *dir, int num)
/*! \brief basically mkdir -p $dest/$context/$ext/$folder
* \param dest String. base directory.
* \param len Length of dest.
* \param context String. Ignored if is null or empty string.
* \param ext String. Ignored if is null or empty string.
* \param folder String. Ignored if is null or empty string.

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@ -678,7 +678,7 @@ struct sip_history {
AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
/*! \brief sip_auth: Creadentials for authentication to other SIP services */
/*! \brief sip_auth: Credentials for authentication to other SIP services */
struct sip_auth {
char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
char username[256]; /*!< Username */

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@ -250,7 +250,8 @@ static int ogg_vorbis_rewrite(struct ast_filestream *s,
/*!
* \brief Write out any pending encoded data.
* \param s A OGG/Vorbis filestream.
* \param s An OGG/Vorbis filestream.
* \param f The file to write to.
*/
static void write_stream(struct vorbis_desc *s, FILE *f)
{
@ -276,9 +277,9 @@ static void write_stream(struct vorbis_desc *s, FILE *f)
/*!
* \brief Write audio data from a frame to an OGG/Vorbis filestream.
* \param fs A OGG/Vorbis filestream.
* \param f An frame containing audio to be written to the filestream.
* \return -1 ifthere was an error, 0 on success.
* \param fs An OGG/Vorbis filestream.
* \param f A frame containing audio to be written to the filestream.
* \return -1 if there was an error, 0 on success.
*/
static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f)
{

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@ -801,6 +801,8 @@ struct ast_channel *ast_waitfor_nandfds(struct ast_channel **chan, int n, int *f
/*! \brief Waits for input on a group of channels
Wait for input on an array of channels for a given # of milliseconds.
\return Return channel with activity, or NULL if none has activity.
\param chan an array of pointers to channels
\param n number of channels that are to be waited upon
\param ms time "ms" is modified in-place, if applicable */
struct ast_channel *ast_waitfor_n(struct ast_channel **chan, int n, int *ms);
@ -928,9 +930,11 @@ struct ast_channel *ast_walk_channel_by_exten_locked(const struct ast_channel *c
int ast_waitfordigit(struct ast_channel *c, int ms);
/*! \brief Wait for a digit
Same as ast_waitfordigit() with audio fd for outputing read audio and ctrlfd to monitor for reading.
Same as ast_waitfordigit() with audio fd for outputting read audio and ctrlfd to monitor for reading.
* \param c channel to wait for a digit on
* \param ms how many milliseconds to wait
* \param audiofd audio file descriptor to write to if audio frames are received
* \param ctrlfd control file descriptor to monitor for reading
* \return Returns 1 if ctrlfd becomes available */
int ast_waitfordigit_full(struct ast_channel *c, int ms, int audiofd, int ctrlfd);
@ -1014,7 +1018,7 @@ int ast_str2cause(const char *name) attribute_pure;
/*! Gives the string form of a given channel state */
/*!
* \param state state to get the name of
* \param ast_channel_state state to get the name of
* Give a name to a state
* Returns the text form of the binary state given
*/
@ -1344,6 +1348,7 @@ int ast_channel_whisper_start(struct ast_channel *chan);
/*!
\brief Feed an audio frame into the whisper buffer on a channel
\param chan The channel to whisper onto
\param f The frame to to whisper onto chan
\return 0 for success, non-zero for failure
*/
int ast_channel_whisper_feed(struct ast_channel *chan, struct ast_frame *f);

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@ -119,7 +119,7 @@ DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it pub
* \arg \ref cdr_drivers
* \arg \ref Config_cdr CDR configuration files
*
* \verbinclude cdr.txt
* \verbinclude cdrdriver.txt
*/
/*! \page AstREADME README - the general administrator introduction
@ -427,14 +427,11 @@ DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it pub
/*! \page SoundFiles Sound files
* \section SecSound Asterisk Sound files
* Asterisk includes a large amount of sound files. Many of these
* Asterisk includes a large number of sound files. Many of these
* are used by applications and demo scripts within asterisk.
*
* Additional sound files are available in the asterisk-addons
* repository on cvs.digium.com
*
* \section SoundList List of included sound files
* \verbinclude sounds.txt
* repository on svn.digium.com
*/
/*! \addtogroup cdr_drivers Module: CDR Drivers

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@ -218,7 +218,7 @@ int ast_filecopy(const char *oldname, const char *newname, const char *fmt);
/*! Waits for a stream to stop or digit to be pressed */
/*!
* \param c channel to waitstram on
* \param c channel to waitstream on
* \param breakon string of DTMF digits to break upon
* Begins playback of a stream...
* Wait for a stream to stop or for any one of a given digit to arrive, Returns 0
@ -228,7 +228,7 @@ int ast_waitstream(struct ast_channel *c, const char *breakon);
/*! Waits for a stream to stop or digit matching a valid one digit exten to be pressed */
/*!
* \param c channel to waitstram on
* \param c channel to waitstream on
* \param context string of context to match digits to break upon
* Begins playback of a stream...
* Wait for a stream to stop or for any one of a valid extension digit to arrive, Returns 0
@ -238,7 +238,7 @@ int ast_waitstream_exten(struct ast_channel *c, const char *context);
/*! Same as waitstream but allows stream to be forwarded or rewound */
/*!
* \param c channel to waitstram on
* \param c channel to waitstream on
* \param breakon string of DTMF digits to break upon
* \param forward DTMF digit to fast forward upon
* \param rewind DTMF digit to rewind upon

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@ -473,7 +473,7 @@ struct ast_frame *ast_smoother_read(struct ast_smoother *s);
void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix);
/*! \par AudioCodecPref Audio Codec Preferences
/*! \page AudioCodecPref Audio Codec Preferences
In order to negotiate audio codecs in the order they are configured
in <channel>.conf for a device, we set up codec preference lists
in addition to the codec capabilities setting. The capabilities

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@ -22,7 +22,7 @@
* - See \ref LockDef
*/
/* \page LockDef Asterisk thread locking models
/*! \page LockDef Asterisk thread locking models
*
* This file provides different implementation of the functions,
* depending on the platform, the use of DEBUG_THREADS, and the way

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@ -77,9 +77,9 @@ enum ast_module_load_result {
enum ast_module_load_result ast_load_resource(const char *resource_name);
/*!
* \brief Unloads a module.
* \brief Unload a module.
* \param resource_name The name of the module to unload.
* \param unload_mode The force flag. This should be set using one of the AST_FORCE flags.
* \param ast_module_unload_mode The force flag. This should be set using one of the AST_FORCE flags.
*
* This function unloads a module. It will only unload modules that are not in
* use (usecount not zero), unless #AST_FORCE_FIRM or #AST_FORCE_HARD is

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@ -99,8 +99,12 @@ int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect,
return res;
}
/*! \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for
"ludicrous time" (essentially never times out) */
/*! \param c The channel to read from
* \param prompt The file to stream to the channel
* \param s The string to read in to. Must be at least the size of your length
* \param maxlen How many digits to read (maximum)
* \param timeout set timeout to 0 for "standard" timeouts. Set timeout to -1 for
* "ludicrous time" (essentially never times out) */
int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout)
{
int res,to,fto;

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@ -2722,8 +2722,8 @@ YY_BUFFER_STATE ast_yy_scan_string (yyconst char * yystr , yyscan_t yyscanner)
/** Setup the input buffer state to scan the given bytes. The next call to ast_yylex() will
* scan from a @e copy of @a bytes.
* @param bytes the byte buffer to scan
* @param len the number of bytes in the buffer pointed to by @a bytes.
* @param yybytes the byte buffer to scan
* @param _yybytes_len the number of bytes in the buffer pointed to by @a bytes.
* @param yyscanner The scanner object.
* @return the newly allocated buffer state object.
*/
@ -2884,7 +2884,7 @@ void ast_yyset_lineno (int line_number , yyscan_t yyscanner)
}
/** Set the current column.
* @param line_number
* @param column_no
* @param yyscanner The scanner object.
*/
void ast_yyset_column (int column_no , yyscan_t yyscanner)

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@ -144,7 +144,7 @@ int daemon(int, int); /* defined in libresolv of all places */
ast_verbose("certain conditions. Type 'show license' for details.\n"); \
ast_verbose("=========================================================================\n")
/*! \defgroup main_options
/*! \defgroup main_options Main Configuration Options
\brief Main configuration options from \ref Config_ast "asterisk.conf" or
the operating system command line when starting Asterisk
Some of them can be changed in the CLI