[svn-commits] tilghman: branch tilghman/codec_bits3 r225761 - in /team/tilghman/codec_bits3...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Oct 25 12:50:25 CDT 2009
Author: tilghman
Date: Sun Oct 25 12:50:19 2009
New Revision: 225761
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=225761
Log:
Quite a few more architectural changes
Modified:
team/tilghman/codec_bits3/channels/chan_iax2.c
team/tilghman/codec_bits3/channels/chan_skinny.c
team/tilghman/codec_bits3/channels/chan_unistim.c
team/tilghman/codec_bits3/include/asterisk/channel.h
team/tilghman/codec_bits3/include/asterisk/pbx.h
team/tilghman/codec_bits3/include/asterisk/translate.h
team/tilghman/codec_bits3/main/channel.c
team/tilghman/codec_bits3/main/pbx.c
team/tilghman/codec_bits3/main/translate.c
team/tilghman/codec_bits3/pbx/pbx_spool.c
Modified: team/tilghman/codec_bits3/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/channels/chan_iax2.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/channels/chan_iax2.c (original)
+++ team/tilghman/codec_bits3/channels/chan_iax2.c Sun Oct 25 12:50:19 2009
@@ -11718,7 +11718,7 @@
{
int callno;
int res;
- int fmt, native;
+ format_t fmt, native;
struct sockaddr_in sin;
struct ast_channel *c;
struct parsed_dial_string pds;
Modified: team/tilghman/codec_bits3/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/channels/chan_skinny.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/channels/chan_skinny.c (original)
+++ team/tilghman/codec_bits3/channels/chan_skinny.c Sun Oct 25 12:50:19 2009
@@ -141,7 +141,7 @@
static const char tdesc[] = "Skinny Client Control Protocol (Skinny)";
static const char config[] = "skinny.conf";
-static int default_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW;
+static format_t default_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW;
static struct ast_codec_pref default_prefs;
enum skinny_codecs {
@@ -1229,9 +1229,9 @@
int instance; \
int group; \
int needdestroy; \
- int confcapability; \
+ format_t confcapability; \
struct ast_codec_pref confprefs; \
- int capability; \
+ format_t capability; \
struct ast_codec_pref prefs; \
int nonCodecCapability; \
int onhooktime; \
@@ -1309,9 +1309,9 @@
int registered; \
int lastlineinstance; \
int lastcallreference; \
- int confcapability; \
+ format_t confcapability; \
struct ast_codec_pref confprefs; \
- int capability; \
+ format_t capability; \
int earlyrtp; \
int transfer; \
int callwaiting; \
@@ -1371,7 +1371,7 @@
AST_LIST_ENTRY(skinnysession) list;
};
-static struct ast_channel *skinny_request(const char *type, int format, const struct ast_channel *requestor, void *data, int *cause);
+static struct ast_channel *skinny_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
static AST_LIST_HEAD_STATIC(sessions, skinnysession);
static int skinny_devicestate(void *data);
@@ -2704,7 +2704,7 @@
}
-static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
+static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, format_t codecs, int nat_active)
{
struct skinny_subchannel *sub;
struct skinny_line *l;
@@ -2748,7 +2748,7 @@
fmt = ast_codec_pref_getsize(&l->prefs, ast_best_codec(l->capability));
if (skinnydebug)
- ast_verb(1, "Setting payloadType to '%d' (%d ms)\n", fmt.bits, fmt.cur_ms);
+ ast_verb(1, "Setting payloadType to '%s' (%d ms)\n", ast_getformatname(fmt.bits), fmt.cur_ms);
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
@@ -4021,9 +4021,9 @@
if (ast) {
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
- if (f->subclass != ast->nativeformats) {
- ast_debug(1, "Oooh, format changed to %d\n", f->subclass);
- ast->nativeformats = f->subclass;
+ if (f->subclass.codec != ast->nativeformats) {
+ ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(f->subclass.codec));
+ ast->nativeformats = f->subclass.codec;
ast_set_read_format(ast, ast->readformat);
ast_set_write_format(ast, ast->writeformat);
}
@@ -4054,9 +4054,13 @@
return 0;
}
} else {
- if (!(frame->subclass & ast->nativeformats)) {
- ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n",
- frame->subclass, ast->nativeformats, ast->readformat, ast->writeformat);
+ if (!(frame->subclass.codec & ast->nativeformats)) {
+ char buf[256];
+ ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
+ ast_getformatname(frame->subclass.codec),
+ ast_getformatname_multiple(buf, sizeof(buf), ast->nativeformats),
+ ast_getformatname(ast->readformat),
+ ast_getformatname(ast->writeformat));
return -1;
}
}
@@ -4400,8 +4404,12 @@
// Should throw an error
tmp->nativeformats = default_capability;
fmt = ast_best_codec(tmp->nativeformats);
- if (skinnydebug)
- ast_verb(1, "skinny_new: tmp->nativeformats=%d fmt=%d\n", tmp->nativeformats, fmt);
+ if (skinnydebug) {
+ char buf[256];
+ ast_verb(1, "skinny_new: tmp->nativeformats=%s fmt=%s\n",
+ ast_getformatname_multiple(buf, sizeof(buf), tmp->nativeformats),
+ ast_getformatname(fmt));
+ }
if (sub->rtp) {
ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
}
@@ -4746,7 +4754,7 @@
ast_log(LOG_WARNING, "Unsupported digit %d\n", digit);
}
- f.subclass = dgt;
+ f.subclass.integer = dgt;
f.src = "skinny";
@@ -5301,6 +5309,7 @@
uint32_t count = 0;
int codecs = 0;
int i;
+ char buf[256];
count = letohl(req->data.caps.count);
if (count > SKINNY_MAX_CAPABILITIES) {
@@ -5319,7 +5328,7 @@
}
d->capability = d->confcapability & codecs;
- ast_verb(0, "Device capability set to '%d'\n", d->capability);
+ ast_verb(0, "Device capability set to '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), d->capability));
AST_LIST_TRAVERSE(&d->lines, l, list) {
ast_mutex_lock(&l->lock);
l->capability = l->confcapability & d->capability;
@@ -5646,7 +5655,7 @@
fmt = ast_codec_pref_getsize(&l->prefs, ast_best_codec(l->capability));
if (skinnydebug)
- ast_verb(1, "Setting payloadType to '%d' (%d ms)\n", fmt.bits, fmt.cur_ms);
+ ast_verb(1, "Setting payloadType to '%s' (%d ms)\n", ast_getformatname(fmt.bits), fmt.cur_ms);
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
@@ -6576,9 +6585,9 @@
return get_devicestate(l);
}
-static struct ast_channel *skinny_request(const char *type, int format, const struct ast_channel *requestor, void *data, int *cause)
-{
- int oldformat;
+static struct ast_channel *skinny_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
+{
+ format_t oldformat;
struct skinny_line *l;
struct ast_channel *tmpc = NULL;
@@ -6588,7 +6597,7 @@
oldformat = format;
if (!(format &= AST_FORMAT_AUDIO_MASK)) {
- ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
+ ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%s'\n", ast_getformatname_multiple(tmp, sizeof(tmp), format));
return NULL;
}
Modified: team/tilghman/codec_bits3/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/channels/chan_unistim.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/channels/chan_unistim.c (original)
+++ team/tilghman/codec_bits3/channels/chan_unistim.c Sun Oct 25 12:50:19 2009
@@ -676,7 +676,7 @@
static int unload_module(void);
static int reload_config(void);
static void show_main_page(struct unistimsession *pte);
-static struct ast_channel *unistim_request(const char *type, int format, const struct ast_channel *requestor,
+static struct ast_channel *unistim_request(const char *type, format_t format, const struct ast_channel *requestor,
void *data, int *cause);
static int unistim_call(struct ast_channel *ast, char *dest, int timeout);
static int unistim_hangup(struct ast_channel *ast);
@@ -2035,7 +2035,7 @@
struct sockaddr_in us = { 0, };
struct sockaddr_in public = { 0, };
struct sockaddr_in sin = { 0, };
- int codec;
+ format_t codec;
struct sockaddr_in sout = { 0, };
/* Sanity checks */
@@ -2085,13 +2085,14 @@
sin.sin_port = htons(sub->parent->parent->rtp_port);
ast_rtp_instance_set_remote_address(sub->rtp, &sin);
if (!(sub->owner->nativeformats & sub->owner->readformat)) {
- int fmt;
+ format_t fmt;
+ char tmp[256];
fmt = ast_best_codec(sub->owner->nativeformats);
ast_log(LOG_WARNING,
- "Our read/writeformat has been changed to something incompatible : %s (%d), using %s (%d) best codec from %d\n",
+ "Our read/writeformat has been changed to something incompatible: %s, using %s best codec from %s\n",
ast_getformatname(sub->owner->readformat),
- sub->owner->readformat, ast_getformatname(fmt), fmt,
- sub->owner->nativeformats);
+ ast_getformatname(fmt),
+ ast_getformatname_multiple(tmp, sizeof(tmp), sub->owner->nativeformats));
sub->owner->readformat = fmt;
sub->owner->writeformat = fmt;
}
@@ -2102,20 +2103,19 @@
else
memcpy(&public, &public_ip, sizeof(public)); /* override */
if (unistimdebug) {
- ast_verb(0, "RTP started : Our IP/port is : %s:%hd with codec %s (%d)\n",
+ ast_verb(0, "RTP started : Our IP/port is : %s:%hd with codec %s\n",
ast_inet_ntoa(us.sin_addr),
- htons(us.sin_port), ast_getformatname(sub->owner->readformat),
- sub->owner->readformat);
+ htons(us.sin_port), ast_getformatname(sub->owner->readformat));
ast_verb(0, "Starting phone RTP stack. Our public IP is %s\n",
ast_inet_ntoa(public.sin_addr));
}
if ((sub->owner->readformat == AST_FORMAT_ULAW) ||
(sub->owner->readformat == AST_FORMAT_ALAW)) {
if (unistimdebug)
- ast_verb(0, "Sending packet_send_rtp_packet_size for codec %d\n", codec);
+ ast_verb(0, "Sending packet_send_rtp_packet_size for codec %s\n", ast_getformatname(codec));
memcpy(buffsend + SIZE_HEADER, packet_send_rtp_packet_size,
sizeof(packet_send_rtp_packet_size));
- buffsend[10] = codec;
+ buffsend[10] = (int) codec & 0xffffffffLL;
send_client(SIZE_HEADER + sizeof(packet_send_rtp_packet_size), buffsend,
sub->parent->parent->session);
}
@@ -2214,8 +2214,8 @@
else if (sub->owner->readformat == AST_FORMAT_G729A)
buffsend[42] = 2; /* 1 = 10ms (10 bytes), 2 = 20ms (20 bytes) */
else
- ast_log(LOG_WARNING, "Unsupported codec %s (%d) !\n",
- ast_getformatname(sub->owner->readformat), sub->owner->readformat);
+ ast_log(LOG_WARNING, "Unsupported codec %s!\n",
+ ast_getformatname(sub->owner->readformat));
/* Source port for transmit RTP and Destination port for receiving RTP */
buffsend[45] = (htons(sin.sin_port) & 0xff00) >> 8;
buffsend[46] = (htons(sin.sin_port) & 0x00ff);
@@ -2474,7 +2474,7 @@
static int unistim_do_senddigit(struct unistimsession *pte, char digit)
{
- struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass = digit, .src = "unistim" };
+ struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = digit, .src = "unistim" };
struct unistim_subchannel *sub;
sub = pte->device->lines->subs[SUB_REAL];
if (!sub->owner || sub->alreadygone) {
@@ -3936,14 +3936,13 @@
if (sub->owner) {
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
- if (f->subclass != sub->owner->nativeformats) {
+ if (f->subclass.codec != sub->owner->nativeformats) {
ast_debug(1,
- "Oooh, format changed from %s (%d) to %s (%d)\n",
+ "Oooh, format changed from %s to %s\n",
ast_getformatname(sub->owner->nativeformats),
- sub->owner->nativeformats, ast_getformatname(f->subclass),
- f->subclass);
-
- sub->owner->nativeformats = f->subclass;
+ ast_getformatname(f->subclass.codec));
+
+ sub->owner->nativeformats = f->subclass.codec;
ast_set_read_format(sub->owner, sub->owner->readformat);
ast_set_write_format(sub->owner, sub->owner->writeformat);
}
@@ -3979,13 +3978,14 @@
return 0;
}
} else {
- if (!(frame->subclass & ast->nativeformats)) {
+ if (!(frame->subclass.codec & ast->nativeformats)) {
+ char tmp[256];
ast_log(LOG_WARNING,
- "Asked to transmit frame type %s (%d), while native formats is %s (%d) (read/write = %s (%d)/%d)\n",
- ast_getformatname(frame->subclass), frame->subclass,
- ast_getformatname(ast->nativeformats), ast->nativeformats,
- ast_getformatname(ast->readformat), ast->readformat,
- ast->writeformat);
+ "Asked to transmit frame type %s, while native formats is %s (read/write = (%s/%s)\n",
+ ast_getformatname(frame->subclass.codec),
+ ast_getformatname_multiple(tmp, sizeof(tmp), ast->nativeformats),
+ ast_getformatname(ast->readformat),
+ ast_getformatname(ast->writeformat));
return -1;
}
}
@@ -4240,7 +4240,7 @@
send_tone(pte, 0, 0);
f.frametype = AST_FRAME_DTMF;
- f.subclass = digit;
+ f.subclass.integer = digit;
f.src = "unistim";
ast_queue_frame(sub->owner, &f);
@@ -4450,9 +4450,14 @@
if (!tmp->nativeformats)
tmp->nativeformats = CAPABILITY;
fmt = ast_best_codec(tmp->nativeformats);
- if (unistimdebug)
- ast_verb(0, "Best codec = %d from nativeformats %d (line cap=%d global=%d)\n", fmt,
- tmp->nativeformats, l->capability, CAPABILITY);
+ if (unistimdebug) {
+ char tmp1[256], tmp2[256], tmp3[256];
+ ast_verb(0, "Best codec = %s from nativeformats %s (line cap=%s global=%s)\n",
+ ast_getformatname(fmt),
+ ast_getformatname_multiple(tmp1, sizeof(tmp1), tmp->nativeformats),
+ ast_getformatname_multiple(tmp2, sizeof(tmp2), l->capability),
+ ast_getformatname_multiple(tmp3, sizeof(tmp3), CAPABILITY));
+ }
if ((sub->rtp) && (sub->subtype == 0)) {
if (unistimdebug)
ast_verb(0, "New unistim channel with a previous rtp handle ?\n");
@@ -4617,10 +4622,10 @@
/*--- unistim_request: PBX interface function ---*/
/* UNISTIM calls initiated by the PBX arrive here */
-static struct ast_channel *unistim_request(const char *type, int format, const struct ast_channel *requestor, void *data,
+static struct ast_channel *unistim_request(const char *type, format_t format, const struct ast_channel *requestor, void *data,
int *cause)
{
- int oldformat;
+ format_t oldformat;
struct unistim_subchannel *sub;
struct ast_channel *tmpc = NULL;
char tmp[256];
@@ -4629,12 +4634,14 @@
oldformat = format;
format &= CAPABILITY;
ast_log(LOG_NOTICE,
- "Asked to get a channel of format %s while capability is %d result : %s (%d) \n",
- ast_getformatname(oldformat), CAPABILITY, ast_getformatname(format), format);
+ "Asked to get a channel of format %s while capability is %s result : %s\n",
+ ast_getformatname(oldformat),
+ ast_getformatname_multiple(tmp, sizeof(tmp), CAPABILITY),
+ ast_getformatname(format));
if (!format) {
ast_log(LOG_NOTICE,
"Asked to get a channel of unsupported format %s while capability is %s\n",
- ast_getformatname(oldformat), ast_getformatname(CAPABILITY));
+ ast_getformatname(oldformat), ast_getformatname_multiple(tmp, sizeof(tmp), CAPABILITY));
return NULL;
}
Modified: team/tilghman/codec_bits3/include/asterisk/channel.h
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/include/asterisk/channel.h?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/include/asterisk/channel.h (original)
+++ team/tilghman/codec_bits3/include/asterisk/channel.h Sun Oct 25 12:50:19 2009
@@ -1156,7 +1156,7 @@
* \return Returns an ast_channel on success or no answer, NULL on failure. Check the value of chan->_state
* to know if the call was answered or not.
*/
-struct ast_channel *ast_request_and_dial(const char *type, int format, const struct ast_channel *requestor, void *data,
+struct ast_channel *ast_request_and_dial(const char *type, format_t format, const struct ast_channel *requestor, void *data,
int timeout, int *reason, const char *cid_num, const char *cid_name);
/*!
@@ -1173,7 +1173,7 @@
* \return Returns an ast_channel on success or no answer, NULL on failure. Check the value of chan->_state
* to know if the call was answered or not.
*/
-struct ast_channel *__ast_request_and_dial(const char *type, int format, const struct ast_channel *requestor, void *data,
+struct ast_channel *__ast_request_and_dial(const char *type, format_t format, const struct ast_channel *requestor, void *data,
int timeout, int *reason, const char *cid_num, const char *cid_name, struct outgoing_helper *oh);
/*!
Modified: team/tilghman/codec_bits3/include/asterisk/pbx.h
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/include/asterisk/pbx.h?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/include/asterisk/pbx.h (original)
+++ team/tilghman/codec_bits3/include/asterisk/pbx.h Sun Oct 25 12:50:19 2009
@@ -879,11 +879,11 @@
/*! Synchronously or asynchronously make an outbound call and send it to a
particular extension */
-int ast_pbx_outgoing_exten(const char *type, int format, void *data, int timeout, const char *context, const char *exten, int priority, int *reason, int sync, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel);
+int ast_pbx_outgoing_exten(const char *type, format_t format, void *data, int timeout, const char *context, const char *exten, int priority, int *reason, int sync, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel);
/*! Synchronously or asynchronously make an outbound call and send it to a
particular application with given extension */
-int ast_pbx_outgoing_app(const char *type, int format, void *data, int timeout, const char *app, const char *appdata, int *reason, int sync, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel);
+int ast_pbx_outgoing_app(const char *type, format_t format, void *data, int timeout, const char *app, const char *appdata, int *reason, int sync, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel);
/*!
* \brief Evaluate a condition
Modified: team/tilghman/codec_bits3/include/asterisk/translate.h
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/include/asterisk/translate.h?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/include/asterisk/translate.h (original)
+++ team/tilghman/codec_bits3/include/asterisk/translate.h Sun Oct 25 12:50:19 2009
@@ -25,7 +25,7 @@
#define _ASTERISK_TRANSLATE_H
#define MAX_AUDIO_FORMAT 15 /* Do not include video here */
-#define MAX_FORMAT 32 /* Do include video here */
+#define MAX_FORMAT 64 /* Do include video here */
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
@@ -69,51 +69,51 @@
* Generic plc is only available for dstfmt = SLINEAR
*/
struct ast_translator {
- const char name[80]; /*!< Name of translator */
- int srcfmt; /*!< Source format (note: bit position,
- converted to index during registration) */
- int dstfmt; /*!< Destination format (note: bit position,
- converted to index during registration) */
+ const char name[80]; /*!< Name of translator */
+ format_t srcfmt; /*!< Source format (note: bit position,
+ * converted to index during registration) */
+ format_t dstfmt; /*!< Destination format (note: bit position,
+ * converted to index during registration) */
int (*newpvt)(struct ast_trans_pvt *); /*!< initialize private data
- associated with the translator */
+ * associated with the translator */
int (*framein)(struct ast_trans_pvt *pvt, struct ast_frame *in);
- /*!< Input frame callback. Store
- (and possibly convert) input frame. */
+ /*!< Input frame callback. Store
+ * (and possibly convert) input frame. */
struct ast_frame * (*frameout)(struct ast_trans_pvt *pvt);
- /*!< Output frame callback. Generate a frame
- with outbuf content. */
+ /*!< Output frame callback. Generate a frame
+ * with outbuf content. */
void (*destroy)(struct ast_trans_pvt *pvt);
- /*!< cleanup private data, if needed
- (often unnecessary). */
-
- struct ast_frame * (*sample)(void); /*!< Generate an example frame */
-
- /*! \brief size of outbuf, in samples. Leave it 0 if you want the framein
+ /*!< cleanup private data, if needed
+ * (often unnecessary). */
+
+ struct ast_frame * (*sample)(void); /*!< Generate an example frame */
+
+ /*!\brief size of outbuf, in samples. Leave it 0 if you want the framein
* callback deal with the frame. Set it appropriately if you
* want the code to checks if the incoming frame fits the
* outbuf (this is e.g. required for plc).
*/
- int buffer_samples; /*< size of outbuf, in samples */
+ int buffer_samples; /*< size of outbuf, in samples */
/*! \brief size of outbuf, in bytes. Mandatory. The wrapper code will also
* allocate an AST_FRIENDLY_OFFSET space before.
*/
int buf_size;
- int desc_size; /*!< size of private descriptor in pvt->pvt, if any */
- int plc_samples; /*!< set to the plc block size if used, 0 otherwise */
- int useplc; /*!< current status of plc, changed at runtime */
- int native_plc; /*!< true if the translator can do native plc */
-
- struct ast_module *module; /* opaque reference to the parent module */
-
- int cost; /*!< Cost in milliseconds for encoding/decoding 1 second of sound */
- int active; /*!< Whether this translator should be used or not */
- AST_LIST_ENTRY(ast_translator) list; /*!< link field */
+ int desc_size; /*!< size of private descriptor in pvt->pvt, if any */
+ int plc_samples; /*!< set to the plc block size if used, 0 otherwise */
+ int useplc; /*!< current status of plc, changed at runtime */
+ int native_plc; /*!< true if the translator can do native plc */
+
+ struct ast_module *module; /*!< opaque reference to the parent module */
+
+ int cost; /*!< Cost in milliseconds for encoding/decoding 1 second of sound */
+ int active; /*!< Whether this translator should be used or not */
+ AST_LIST_ENTRY(ast_translator) list; /*!< link field */
};
/*! \brief
@@ -205,7 +205,7 @@
* \return Returns 0 on success, -1 if no path could be found.
* \note Modifies dests and srcs in place
*/
-int ast_translator_best_choice(int *dsts, int *srcs);
+format_t ast_translator_best_choice(format_t *dsts, format_t *srcs);
/*!
* \brief Builds a translator path
@@ -214,7 +214,7 @@
* \param source source format
* \return ast_trans_pvt on success, NULL on failure
* */
-struct ast_trans_pvt *ast_translator_build_path(int dest, int source);
+struct ast_trans_pvt *ast_translator_build_path(format_t dest, format_t source);
/*!
* \brief Frees a translator path
@@ -240,7 +240,7 @@
* \param src source format
* \return the number of translation steps required, or -1 if no path is available
*/
-unsigned int ast_translate_path_steps(unsigned int dest, unsigned int src);
+unsigned int ast_translate_path_steps(format_t dest, format_t src);
/*!
* \brief Mask off unavailable formats from a format bitmask
@@ -254,7 +254,7 @@
* \note Only a single audio format and a single video format can be
* present in 'src', or the function will produce unexpected results.
*/
-unsigned int ast_translate_available_formats(unsigned int dest, unsigned int src);
+format_t ast_translate_available_formats(format_t dest, format_t src);
#if defined(__cplusplus) || defined(c_plusplus)
}
Modified: team/tilghman/codec_bits3/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/main/channel.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/main/channel.c (original)
+++ team/tilghman/codec_bits3/main/channel.c Sun Oct 25 12:50:19 2009
@@ -4263,7 +4263,7 @@
}
}
-struct ast_channel *ast_call_forward(struct ast_channel *caller, struct ast_channel *orig, int *timeout, int format, struct outgoing_helper *oh, int *outstate)
+struct ast_channel *ast_call_forward(struct ast_channel *caller, struct ast_channel *orig, int *timeout, format_t format, struct outgoing_helper *oh, int *outstate)
{
char tmpchan[256];
struct ast_channel *new = NULL;
@@ -4339,7 +4339,7 @@
return new;
}
-struct ast_channel *__ast_request_and_dial(const char *type, int format, const struct ast_channel *requestor, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, struct outgoing_helper *oh)
+struct ast_channel *__ast_request_and_dial(const char *type, format_t format, const struct ast_channel *requestor, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name, struct outgoing_helper *oh)
{
int dummy_outstate;
int cause = 0;
@@ -4478,21 +4478,21 @@
return chan;
}
-struct ast_channel *ast_request_and_dial(const char *type, int format, const struct ast_channel *requestor, void *data, int timeout, int *outstate, const char *cidnum, const char *cidname)
+struct ast_channel *ast_request_and_dial(const char *type, format_t format, const struct ast_channel *requestor, void *data, int timeout, int *outstate, const char *cidnum, const char *cidname)
{
return __ast_request_and_dial(type, format, requestor, data, timeout, outstate, cidnum, cidname, NULL);
}
-struct ast_channel *ast_request(const char *type, int format, const struct ast_channel *requestor, void *data, int *cause)
+struct ast_channel *ast_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
{
struct chanlist *chan;
struct ast_channel *c;
- int capabilities;
- int fmt;
+ format_t capabilities;
+ format_t fmt;
int res;
int foo;
- int videoformat = format & AST_FORMAT_VIDEO_MASK;
- int textformat = format & AST_FORMAT_TEXT_MASK;
+ format_t videoformat = format & AST_FORMAT_VIDEO_MASK;
+ format_t textformat = format & AST_FORMAT_TEXT_MASK;
if (!cause)
cause = &foo;
Modified: team/tilghman/codec_bits3/main/pbx.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/main/pbx.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/main/pbx.c (original)
+++ team/tilghman/codec_bits3/main/pbx.c Sun Oct 25 12:50:19 2009
@@ -8224,7 +8224,7 @@
return 0; /* success */
}
-int ast_pbx_outgoing_exten(const char *type, int format, void *data, int timeout, const char *context, const char *exten, int priority, int *reason, int synchronous, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **channel)
+int ast_pbx_outgoing_exten(const char *type, format_t format, void *data, int timeout, const char *context, const char *exten, int priority, int *reason, int synchronous, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **channel)
{
struct ast_channel *chan;
struct async_stat *as;
@@ -8390,7 +8390,7 @@
return NULL;
}
-int ast_pbx_outgoing_app(const char *type, int format, void *data, int timeout, const char *app, const char *appdata, int *reason, int synchronous, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel)
+int ast_pbx_outgoing_app(const char *type, format_t format, void *data, int timeout, const char *app, const char *appdata, int *reason, int synchronous, const char *cid_num, const char *cid_name, struct ast_variable *vars, const char *account, struct ast_channel **locked_channel)
{
struct ast_channel *chan;
struct app_tmp *tmp;
Modified: team/tilghman/codec_bits3/main/translate.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/main/translate.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/main/translate.c (original)
+++ team/tilghman/codec_bits3/main/translate.c Sun Oct 25 12:50:19 2009
@@ -745,15 +745,15 @@
}
/*! \brief Calculate our best translator source format, given costs, and a desired destination */
-int ast_translator_best_choice(int *dst, int *srcs)
+format_t ast_translator_best_choice(format_t *dst, format_t *srcs)
{
int x,y;
- int best = -1;
- int bestdst = 0;
- int cur, cursrc;
+ format_t best = -1;
+ format_t bestdst = 0;
+ format_t cur, cursrc;
int besttime = INT_MAX;
int beststeps = INT_MAX;
- int common = ((*dst) & (*srcs)) & AST_FORMAT_AUDIO_MASK; /* are there common formats ? */
+ format_t common = ((*dst) & (*srcs)) & AST_FORMAT_AUDIO_MASK; /* are there common formats ? */
if (common) { /* yes, pick one and return */
for (cur = 1, y = 0; y <= MAX_AUDIO_FORMAT; cur <<= 1, y++) {
Modified: team/tilghman/codec_bits3/pbx/pbx_spool.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/pbx/pbx_spool.c?view=diff&rev=225761&r1=225760&r2=225761
==============================================================================
--- team/tilghman/codec_bits3/pbx/pbx_spool.c (original)
+++ team/tilghman/codec_bits3/pbx/pbx_spool.c Sun Oct 25 12:50:19 2009
@@ -66,7 +66,7 @@
int retrytime; /*!< How long to wait between retries (in seconds) */
int waittime; /*!< How long to wait for an answer */
long callingpid; /*!< PID which is currently calling */
- int format; /*!< Formats (codecs) for this call */
+ format_t format; /*!< Formats (codecs) for this call */
AST_DECLARE_STRING_FIELDS (
AST_STRING_FIELD(fn); /*!< File name of call file */
AST_STRING_FIELD(tech); /*!< Which channel technology to use for outgoing call */
More information about the svn-commits
mailing list