Issue #8246 - Doxygen fixes from kshumard.

An extra big thankyou is given to everyone that contributes to doxygen!

		THANK YOU!

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 7ec04e4806fbfeb36f185b3acfe22fb8bb54d2f0..25802c8b7215244bc678bf46d0adc72eeccb9e62 100644 (file)
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -929,6 +929,7 @@ 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. 

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5e0399ab0815d47c58f2628617a063c91b155b94..a3611e913b41776373a3ae39f63985db326842c0 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -677,7 +677,7 @@ 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 */

diff --git a/formats/format_ogg_vorbis.c b/formats/format_ogg_vorbis.c
index 5fc17c338631caa9e8ef72e35f2c104b1144c00a..9d80b63d44395557c89b53b06ccffb36d2900198 100644 (file)
--- a/formats/format_ogg_vorbis.c
+++ b/formats/format_ogg_vorbis.c
@@ -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)
 {

diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 0d06ef61306c378c7e8618d36eb2471147a72531..a08a013a4116abde7c25578b07b04f8d5769092b 100644 (file)
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -798,6 +798,8 @@ 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);
 
@@ -925,9 +927,11 @@ 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);
 
@@ -1004,7 +1008,7 @@ 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
  */
@@ -1334,6 +1338,7 @@ 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);

diff --git a/include/asterisk/doxyref.h b/include/asterisk/doxyref.h
index e9fb631107e8f343b21dafba169577da82d272aa..49e2828160cb114c753cfe614f0747a53976d406 100644 (file)
--- a/include/asterisk/doxyref.h
+++ b/include/asterisk/doxyref.h
@@ -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

diff --git a/include/asterisk/file.h b/include/asterisk/file.h
index bf7efef8979bfb6f09c61cab67852c5455445a21..029bcf25d96362f5049c014514f90ce251e39aeb 100644 (file)
--- a/include/asterisk/file.h
+++ b/include/asterisk/file.h
@@ -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

diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index fa1c1289128b926d767419ae1233498576664403..cc3aa5d832a70f2933e864b66eca092e6e79c96c 100644 (file)
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -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

diff --git a/include/asterisk/lock.h b/include/asterisk/lock.h
index 2f63ee183e5964cabf7081cbf3a4eadcb5d65c71..bcd2fdabef93fcd27b58df868e7256927c6ce4b6 100644 (file)
--- a/include/asterisk/lock.h
+++ b/include/asterisk/lock.h
@@ -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

diff --git a/include/asterisk/module.h b/include/asterisk/module.h
index 4ce6e9460b692e57e59bde1909a1bbc174a60a4d..1d00f45049c213d1dfad1dc0fe62982e9d12d919 100644 (file)
--- a/include/asterisk/module.h
+++ b/include/asterisk/module.h
@@ -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 

diff --git a/main/app.c b/main/app.c
index e640a762e1bf769914ccd6ac1229d8474119aac9..f99a33226d2a94ff7d1eb4df183bc570e940c01d 100644 (file)
--- a/main/app.c
+++ b/main/app.c
@@ -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;

diff --git a/main/ast_expr2f.c b/main/ast_expr2f.c
index 7d790924d8e4901732e0db5d064f653f241acd76..3693e4842153b8d1a5195c3087a32c77c1e5f329 100644 (file)
--- a/main/ast_expr2f.c
+++ b/main/ast_expr2f.c
@@ -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)

diff --git a/main/asterisk.c b/main/asterisk.c
index 7b7b11ae77b0c1193471e89ff6ba1eb6dc0e5659..a59b48689ded73921499e6e70a0da53370bd3b2e 100644 (file)
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -145,7 +145,7 @@ 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