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Expand codec bitfield from 32 bits to 64 bits.

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Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2009-11-04 14:05:12 +00:00
parent 6a50e7a031
commit d8e0c58437
124 changed files with 1777 additions and 1366 deletions
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28
include/asterisk/rtp_engine.h
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@@ -221,7 +221,7 @@ struct ast_rtp_payload_type {
/*! Is this an Asterisk value */
int asterisk_format;
/*! Actual internal value of the payload */
int code;
format_t code;
};
/*! Structure that represents statistics from an RTP instance */
@@ -328,7 +328,7 @@ struct ast_rtp_engine {
/*! Callback for setting an RTP property */
void (*prop_set)(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
/*! Callback for setting a payload */
void (*payload_set)(struct ast_rtp_instance *instance, int payload, int astformat, int format);
void (*payload_set)(struct ast_rtp_instance *instance, int payload, int astformat, format_t format);
/*! Callback for setting packetization preferences */
void (*packetization_set)(struct ast_rtp_instance *instance, struct ast_codec_pref *pref);
/*! Callback for setting the remote address that RTP is to be sent to */
@@ -352,9 +352,9 @@ struct ast_rtp_engine {
/*! Callback to locally bridge two RTP instances */
int (*local_bridge)(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
/*! Callback to set the read format */
int (*set_read_format)(struct ast_rtp_instance *instance, int format);
int (*set_read_format)(struct ast_rtp_instance *instance, format_t format);
/*! Callback to set the write format */
int (*set_write_format)(struct ast_rtp_instance *instance, int format);
int (*set_write_format)(struct ast_rtp_instance *instance, format_t format);
/*! Callback to make two instances compatible */
int (*make_compatible)(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
/*! Callback to see if two instances are compatible with DTMF */
@@ -399,9 +399,9 @@ struct ast_rtp_glue {
*/
enum ast_rtp_glue_result (*get_trtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
/*! Callback for updating the destination that the remote side should send RTP to */
int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active);
int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, format_t codecs, int nat_active);
/*! Callback for retrieving codecs that the channel can do */
int (*get_codec)(struct ast_channel *chan);
format_t (*get_codec)(struct ast_channel *chan);
/*! Linked list information */
AST_RWLIST_ENTRY(ast_rtp_glue) entry;
};
@@ -994,7 +994,7 @@ struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs
*
* \since 1.6.3
*/
unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code);
unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, format_t code);
/*!
* \brief Retrieve all formats that were found
@@ -1015,7 +1015,7 @@ unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code);
*
* \since 1.6.3
*/
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats);
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, format_t *astformats, int *nonastformats);
/*!
* \brief Retrieve a payload based on whether it is an Asterisk format and the code
@@ -1036,7 +1036,7 @@ void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astforma
*
* \since 1.6.3
*/
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code);
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const format_t code);
/*!
* \brief Retrieve mime subtype information on a payload
@@ -1058,7 +1058,7 @@ int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asteris
*
* \since 1.6.3
*/
const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int code, enum ast_rtp_options options);
const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const format_t code, enum ast_rtp_options options);
/*!
* \brief Convert formats into a string and put them into a buffer
@@ -1082,7 +1082,7 @@ const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const int co
*
* \since 1.6.3
*/
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options);
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const format_t capability, const int asterisk_format, enum ast_rtp_options options);
/*!
* \brief Set codec packetization preferences
@@ -1464,7 +1464,7 @@ char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_r
*
* \since 1.6.3
*/
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int format);
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, format_t format);
/*!
* \brief Tell underlying RTP engine that audio frames will be provided in a specific format
@@ -1485,7 +1485,7 @@ int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, int form
*
* \since 1.6.3
*/
int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, int format);
int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, format_t format);
/*!
* \brief Request that the underlying RTP engine make two RTP instances compatible with eachother
@@ -1527,7 +1527,7 @@ int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_in
*
* \since 1.6.3
*/
int ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, int to_endpoint, int to_asterisk);
format_t ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, int to_endpoint, int to_asterisk);
/*!
* \brief Indicate to the RTP engine that packets are now expected to be sent/received on the RTP instance