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Merged revisions 59227 via svnmerge from

Browse source 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines

Change this to a single dp function to make oej happy.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher 2007-03-26 21:44:59 +00:00
parent d6943624c8
commit 0e0600a446
1 changed files with 31 additions and 66 deletions
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97
channels/chan_sip.c
View file

@ -14800,11 +14800,16 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
} }
} }
static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname, char *args, char *buf, size_t buflen) static int acf_rtpqos_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{ {
struct ast_rtp_quality qos; struct ast_rtp_quality qos;
struct sip_pvt *p = chan->tech_pvt; struct sip_pvt *p = chan->tech_pvt;
char *all = ""; char *all = "", *parse = ast_strdupa(preparse);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(type);
AST_APP_ARG(field);
);
AST_STANDARD_APP_ARGS(args, parse);
/* Sanity check */ /* Sanity check */
if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) { if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
@ -14814,36 +14819,36 @@ static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname,
memset(buf, 0, buflen); memset(buf, 0, buflen);
memset(&qos, 0, sizeof(qos)); memset(&qos, 0, sizeof(qos));
if (strcmp(funcname, "RTPAUDIOQOS") == 0) { if (strcasecmp(args.type, "AUDIO") == 0) {
all = ast_rtp_get_quality(p->rtp, &qos); all = ast_rtp_get_quality(p->rtp, &qos);
} else if (strcmp(funcname, "RTPVIDEOQOS") == 0) { } else if (strcasecmp(args.type, "VIDEO") == 0) {
all = ast_rtp_get_quality(p->vrtp, &qos); all = ast_rtp_get_quality(p->vrtp, &qos);
} else if (strcmp(funcname, "RTPTEXTQOS") == 0) { } else if (strcasecmp(args.type, "TEXT") == 0) {
all = ast_rtp_get_quality(p->trtp, &qos); all = ast_rtp_get_quality(p->trtp, &qos);
} }
if (strcasecmp(args, "local_ssrc") == 0) if (strcasecmp(args.field, "local_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.local_ssrc); snprintf(buf, buflen, "%u", qos.local_ssrc);
else if (strcasecmp(args, "local_lostpackets") == 0) else if (strcasecmp(args.field, "local_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.local_lostpackets); snprintf(buf, buflen, "%u", qos.local_lostpackets);
else if (strcasecmp(args, "local_jitter") == 0) else if (strcasecmp(args.field, "local_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0); snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
else if (strcasecmp(args, "local_count") == 0) else if (strcasecmp(args.field, "local_count") == 0)
snprintf(buf, buflen, "%u", qos.local_count); snprintf(buf, buflen, "%u", qos.local_count);
else if (strcasecmp(args, "remote_ssrc") == 0) else if (strcasecmp(args.field, "remote_ssrc") == 0)
snprintf(buf, buflen, "%u", qos.remote_ssrc); snprintf(buf, buflen, "%u", qos.remote_ssrc);
else if (strcasecmp(args, "remote_lostpackets") == 0) else if (strcasecmp(args.field, "remote_lostpackets") == 0)
snprintf(buf, buflen, "%u", qos.remote_lostpackets); snprintf(buf, buflen, "%u", qos.remote_lostpackets);
else if (strcasecmp(args, "remote_jitter") == 0) else if (strcasecmp(args.field, "remote_jitter") == 0)
snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0); snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
else if (strcasecmp(args, "remote_count") == 0) else if (strcasecmp(args.field, "remote_count") == 0)
snprintf(buf, buflen, "%u", qos.remote_count); snprintf(buf, buflen, "%u", qos.remote_count);
else if (strcasecmp(args, "rtt") == 0) else if (strcasecmp(args.field, "rtt") == 0)
snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0); snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
else if (strcasecmp(args, "all") == 0) else if (strcasecmp(args.field, "all") == 0)
ast_copy_string(buf, all, buflen); ast_copy_string(buf, all, buflen);
else { else {
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", args, funcname); ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1; return -1;
} }
return 0; return 0;
@ -18130,9 +18135,9 @@ static struct ast_cli_entry cli_sip[] = {
sip_reload_usage }, sip_reload_usage },
}; };
struct ast_custom_function acf_audiortpqos = { struct ast_custom_function acf_rtpqos = {
.name = "RTPAUDIOQOS", .name = "RTPQOS",
.synopsis = "Retrieve statistics about an RTP audio stream", .synopsis = "Retrieve statistics about an RTP stream",
.desc = .desc =
"The following statistics may be retrieved:\n" "The following statistics may be retrieved:\n"
" local_ssrc - Local SSRC (stream ID)\n" " local_ssrc - Local SSRC (stream ID)\n"
@ -18144,47 +18149,11 @@ struct ast_custom_function acf_audiortpqos = {
" remote_jitter - Remote reported jitter\n" " remote_jitter - Remote reported jitter\n"
" remote_count - Number of transmitted packets\n" " remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n" " rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)", " all - All statistics (in a form suited to logging, but not for parsing)\n"
.syntax = "RTPAUDIOQOS(<field>)", "\n"
.read = acf_audiortpqos_read, "Type may be specified as \"audio\", \"video\", or \"text\".\n",
}; .syntax = "RTPQOS(<type>|<field>)",
.read = acf_rtpqos_read,
struct ast_custom_function acf_videortpqos = {
.name = "RTPVIDEOQOS",
.synopsis = "Retrieve statistics about an RTP video stream",
.desc =
"The following statistics may be retrieved:\n"
" local_ssrc - Local SSRC (stream ID)\n"
" local_lostpackets - Local lost packets\n"
" local_jitter - Local calculated jitter\n"
" local_count - Number of received packets\n"
" remote_ssrc - Remote SSRC (stream ID)\n"
" remote_lostpackets - Remote lost packets\n"
" remote_jitter - Remote reported jitter\n"
" remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)",
.syntax = "RTPVIDEOQOS(<field>)",
.read = acf_audiortpqos_read,
};
struct ast_custom_function acf_textrtpqos = {
.name = "RTPTEXTQOS",
.synopsis = "Retrieve statistics about an RTP text stream",
.desc =
"The following statistics may be retrieved:\n"
" local_ssrc - Local SSRC (stream ID)\n"
" local_lostpackets - Local lost packets\n"
" local_jitter - Local calculated jitter\n"
" local_count - Number of received packets\n"
" remote_ssrc - Remote SSRC (stream ID)\n"
" remote_lostpackets - Remote lost packets\n"
" remote_jitter - Remote reported jitter\n"
" remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)",
.syntax = "RTPTEXTQOS(<field>)",
.read = acf_audiortpqos_read,
}; };
/*! \brief PBX load module - initialization */ /*! \brief PBX load module - initialization */
@ -18236,9 +18205,7 @@ static int load_module(void)
ast_custom_function_register(&sippeer_function); ast_custom_function_register(&sippeer_function);
ast_custom_function_register(&sipchaninfo_function); ast_custom_function_register(&sipchaninfo_function);
ast_custom_function_register(&checksipdomain_function); ast_custom_function_register(&checksipdomain_function);
ast_custom_function_register(&acf_audiortpqos); ast_custom_function_register(&acf_rtpqos);
ast_custom_function_register(&acf_videortpqos);
ast_custom_function_register(&acf_textrtpqos);
/* Register manager commands */ /* Register manager commands */
ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers, ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
@ -18268,9 +18235,7 @@ static int unload_module(void)
ast_custom_function_unregister(&sippeer_function); ast_custom_function_unregister(&sippeer_function);
ast_custom_function_unregister(&sip_header_function); ast_custom_function_unregister(&sip_header_function);
ast_custom_function_unregister(&checksipdomain_function); ast_custom_function_unregister(&checksipdomain_function);
ast_custom_function_unregister(&acf_audiortpqos); ast_custom_function_unregister(&acf_rtpqos);
ast_custom_function_unregister(&acf_videortpqos);
ast_custom_function_unregister(&acf_textrtpqos);
/* Unregister dial plan applications */ /* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode); ast_unregister_application(app_dtmfmode);