[asterisk-commits] trunk - r7547 in /trunk: apps/app_meetme.c
channel.c include/asterisk/channel.h
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 20 07:07:06 CST 2005
Author: markster
Date: Tue Dec 20 07:07:02 2005
New Revision: 7547
URL: http://svn.digium.com/view/asterisk?rev=7547&view=rev
Log:
Major peformance improvements to meetme
Modified:
trunk/apps/app_meetme.c
trunk/channel.c
trunk/include/asterisk/channel.h
Modified: trunk/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_meetme.c?rev=7547&r1=7546&r2=7547&view=diff
==============================================================================
--- trunk/apps/app_meetme.c (original)
+++ trunk/apps/app_meetme.c Tue Dec 20 07:07:02 2005
@@ -54,6 +54,8 @@
#include "asterisk/cli.h"
#include "asterisk/say.h"
#include "asterisk/utils.h"
+#include "asterisk/translate.h"
+#include "asterisk/ulaw.h"
static const char *tdesc = "MeetMe conference bridge";
@@ -95,6 +97,10 @@
" 's' -- Present menu (user or admin) when '*' is received ('send' to menu)\n"
" 't' -- set talk only mode. (Talk only, no listening)\n"
" 'T' -- set talker detection (sent to manager interface and meetme list)\n"
+" 'o' -- set talker optimization - treats talkers who aren't speaking as\n"
+" being muted, meaning (a) No encode is done on transmission and\n"
+" (b) Received audio that is not registered as talking is omitted\n"
+" causing no buildup in background noise\n"
" 'v' -- video mode\n"
" 'w' -- wait until the marked user enters the conference\n"
" 'x' -- close the conference when last marked user exits\n"
@@ -129,6 +135,8 @@
LOCAL_USER_DECL;
static struct ast_conference {
+ ast_mutex_t playlock; /* Conference specific lock (players) */
+ ast_mutex_t listenlock; /* Conference specific lock (listeners) */
char confno[AST_MAX_EXTENSION]; /* Conference */
struct ast_channel *chan; /* Announcements channel */
int fd; /* Announcements fd */
@@ -147,6 +155,9 @@
const char *recordingformat; /* Format to record the Conference in */
char pin[AST_MAX_EXTENSION]; /* If protected by a PIN */
char pinadmin[AST_MAX_EXTENSION]; /* If protected by a admin PIN */
+ struct ast_frame *transframe[32];
+ struct ast_frame *origframe;
+ struct ast_trans_pvt *transpath[32];
struct ast_conference *next;
} *confs;
@@ -182,6 +193,8 @@
#define MEETME_DELAYDETECTTALK 300
#define MEETME_DELAYDETECTENDTALK 1000
+#define AST_FRAME_BITS 32
+
enum volume_action {
VOL_UP,
VOL_DOWN,
@@ -190,6 +203,7 @@
AST_MUTEX_DEFINE_STATIC(conflock);
static int admin_exec(struct ast_channel *chan, void *data);
+static struct ast_frame null_frame = { AST_FRAME_NULL, };
static void *recordthread(void *args);
@@ -200,8 +214,9 @@
#define LEAVE 1
#define MEETME_RECORD_OFF 0
-#define MEETME_RECORD_ACTIVE 1
-#define MEETME_RECORD_TERMINATE 2
+#define MEETME_RECORD_STARTED 1
+#define MEETME_RECORD_ACTIVE 2
+#define MEETME_RECORD_TERMINATE 3
#define CONF_SIZE 320
@@ -227,12 +242,14 @@
#define CONFFLAG_EMPTYNOPIN (1 << 20)
#define CONFFLAG_ALWAYSPROMPT (1 << 21)
#define CONFFLAG_ANNOUNCEUSERCOUNT (1 << 22) /* If set, when user joins the conference, they will be told the number of users that are already in */
+#define CONFFLAG_OPTIMIZETALKER (1 << 23) /* If set, treats talking users as muted users */
AST_APP_OPTIONS(meetme_opts, {
AST_APP_OPTION('a', CONFFLAG_ADMIN ),
AST_APP_OPTION('c', CONFFLAG_ANNOUNCEUSERCOUNT ),
AST_APP_OPTION('T', CONFFLAG_MONITORTALKER ),
+ AST_APP_OPTION('o', CONFFLAG_OPTIMIZETALKER ),
AST_APP_OPTION('i', CONFFLAG_INTROUSER ),
AST_APP_OPTION('m', CONFFLAG_MONITOR ),
AST_APP_OPTION('p', CONFFLAG_POUNDEXIT ),
@@ -406,6 +423,8 @@
unsigned char *data;
int len;
int res = -1;
+ short *data2;
+ int x;
if (!chan->_softhangup)
res = ast_autoservice_start(chan);
@@ -425,8 +444,12 @@
data = NULL;
len = 0;
}
- if (data)
- careful_write(conf->fd, data, len, 1);
+ if (data) {
+ data2 = alloca(len * 2);
+ for (x=0;x<len;x++)
+ data2[x] = AST_MULAW(data[x]);
+ careful_write(conf->fd, (unsigned char *)data2, len << 1, 1);
+ }
ast_mutex_unlock(&conflock);
@@ -450,12 +473,16 @@
/* Make a new one */
cnf = calloc(1, sizeof(*cnf));
if (cnf) {
+ ast_mutex_init(&cnf->playlock);
+ ast_mutex_init(&cnf->listenlock);
ast_copy_string(cnf->confno, confno, sizeof(cnf->confno));
ast_copy_string(cnf->pin, pin, sizeof(cnf->pin));
ast_copy_string(cnf->pinadmin, pinadmin, sizeof(cnf->pinadmin));
cnf->markedusers = 0;
- cnf->chan = ast_request("zap", AST_FORMAT_ULAW, "pseudo", NULL);
+ cnf->chan = ast_request("zap", AST_FORMAT_SLINEAR, "pseudo", NULL);
if (cnf->chan) {
+ ast_set_read_format(cnf->chan, AST_FORMAT_SLINEAR);
+ ast_set_write_format(cnf->chan, AST_FORMAT_SLINEAR);
cnf->fd = cnf->chan->fds[0]; /* for use by conf_play() */
} else {
ast_log(LOG_WARNING, "Unable to open pseudo channel - trying device\n");
@@ -482,6 +509,7 @@
cnf = NULL;
goto cnfout;
}
+
/* Fill the conference struct */
cnf->start = time(NULL);
cnf->zapconf = ztc.confno;
@@ -752,7 +780,8 @@
static int conf_free(struct ast_conference *conf)
{
struct ast_conference *prev = NULL, *cur = confs;
-
+ int x;
+
while (cur) {
if (cur == conf) {
if (prev)
@@ -779,6 +808,14 @@
}
}
+ for (x=0;x<AST_FRAME_BITS;x++) {
+ if (conf->transframe[x])
+ ast_frfree(conf->transframe[x]);
+ if (conf->transpath[x])
+ ast_translator_free_path(conf->transpath[x]);
+ if (conf->origframe)
+ ast_frfree(conf->origframe);
+ }
if (conf->chan)
ast_hangup(conf->chan);
else
@@ -831,21 +868,26 @@
return ret;
}
- if (confflags & CONFFLAG_RECORDCONF && conf->recording !=MEETME_RECORD_ACTIVE) {
- conf->recordingfilename = pbx_builtin_getvar_helper(chan, "MEETME_RECORDINGFILE");
+ if (confflags & CONFFLAG_RECORDCONF) {
if (!conf->recordingfilename) {
- snprintf(recordingtmp, sizeof(recordingtmp), "meetme-conf-rec-%s-%s", conf->confno, chan->uniqueid);
- conf->recordingfilename = ast_strdupa(recordingtmp);
- }
- conf->recordingformat = pbx_builtin_getvar_helper(chan, "MEETME_RECORDINGFORMAT");
- if (!conf->recordingformat) {
- snprintf(recordingtmp, sizeof(recordingtmp), "wav");
- conf->recordingformat = ast_strdupa(recordingtmp);
- }
+ conf->recordingfilename = pbx_builtin_getvar_helper(chan, "MEETME_RECORDINGFILE");
+ if (!conf->recordingfilename) {
+ snprintf(recordingtmp, sizeof(recordingtmp), "meetme-conf-rec-%s-%s", conf->confno, chan->uniqueid);
+ conf->recordingfilename = ast_strdupa(recordingtmp);
+ }
+ conf->recordingformat = pbx_builtin_getvar_helper(chan, "MEETME_RECORDINGFORMAT");
+ if (!conf->recordingformat) {
+ snprintf(recordingtmp, sizeof(recordingtmp), "wav");
+ conf->recordingformat = ast_strdupa(recordingtmp);
+ }
+ ast_verbose(VERBOSE_PREFIX_4 "Starting recording of MeetMe Conference %s into file %s.%s.\n",
+ conf->confno, conf->recordingfilename, conf->recordingformat);
+ }
+ }
+
+ if ((conf->recording == MEETME_RECORD_OFF) && ((confflags & CONFFLAG_RECORDCONF) || (conf->chan))) {
pthread_attr_init(&conf->attr);
pthread_attr_setdetachstate(&conf->attr, PTHREAD_CREATE_DETACHED);
- ast_verbose(VERBOSE_PREFIX_4 "Starting recording of MeetMe Conference %s into file %s.%s.\n",
- conf->confno, conf->recordingfilename, conf->recordingformat);
ast_pthread_create(&conf->recordthread, &conf->attr, recordthread, conf);
}
@@ -861,7 +903,7 @@
if (confflags & CONFFLAG_MARKEDUSER)
conf->markedusers++;
- ast_mutex_lock(&conflock);
+ ast_mutex_lock(&conf->playlock);
if (!conf->firstuser) {
/* Fill the first new User struct */
user->user_no = 1;
@@ -873,7 +915,7 @@
user->prevuser = conf->lastuser;
if (conf->lastuser->nextuser) {
ast_log(LOG_WARNING, "Error in User Management!\n");
- ast_mutex_unlock(&conflock);
+ ast_mutex_unlock(&conf->playlock);
goto outrun;
} else {
conf->lastuser->nextuser = user;
@@ -886,7 +928,7 @@
user->adminflags = 0;
user->talking = -1;
conf->users++;
- ast_mutex_unlock(&conflock);
+ ast_mutex_unlock(&conf->playlock);
if (confflags & CONFFLAG_EXIT_CONTEXT) {
if ((agifile = pbx_builtin_getvar_helper(chan, "MEETME_EXIT_CONTEXT")))
@@ -1029,7 +1071,7 @@
ztc.chan = 0;
ztc.confno = conf->zapconf;
- ast_mutex_lock(&conflock);
+ ast_mutex_lock(&conf->playlock);
if (!(confflags & CONFFLAG_QUIET) && (confflags & CONFFLAG_INTROUSER) && conf->users > 1) {
if (conf->chan && ast_fileexists(user->namerecloc, NULL, NULL)) {
@@ -1050,7 +1092,7 @@
if (ioctl(fd, ZT_SETCONF, &ztc)) {
ast_log(LOG_WARNING, "Error setting conference\n");
close(fd);
- ast_mutex_unlock(&conflock);
+ ast_mutex_unlock(&conf->playlock);
goto outrun;
}
ast_log(LOG_DEBUG, "Placed channel %s in ZAP conf %d\n", chan->name, conf->zapconf);
@@ -1069,7 +1111,7 @@
conf_play(chan, conf, ENTER);
}
- ast_mutex_unlock(&conflock);
+ ast_mutex_unlock(&conf->playlock);
conf_flush(fd, chan);
@@ -1106,7 +1148,7 @@
x = 1;
ast_channel_setoption(chan, AST_OPTION_TONE_VERIFY, &x, sizeof(char), 0);
}
- if (confflags & CONFFLAG_MONITORTALKER && !(dsp = ast_dsp_new())) {
+ if (confflags & (CONFFLAG_MONITORTALKER | CONFFLAG_OPTIMIZETALKER) && !(dsp = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
res = -1;
}
@@ -1145,6 +1187,7 @@
}
c = ast_waitfor_nandfds(&chan, 1, &fd, nfds, NULL, &outfd, &ms);
+
/* Update the struct with the actual confflags */
user->userflags = confflags;
@@ -1269,14 +1312,17 @@
user->zapchannel = !retryzap;
goto zapretry;
}
- f = ast_read(c);
+ if (!(confflags & CONFFLAG_MONITOR))
+ f = ast_read(c);
+ else
+ f = ast_read_noaudio(c);
if (!f)
break;
if ((f->frametype == AST_FRAME_VOICE) && (f->subclass == AST_FORMAT_SLINEAR)) {
if (user->talk.actual)
ast_frame_adjust_volume(f, user->talk.actual);
- if (confflags & CONFFLAG_MONITORTALKER) {
+ if (confflags & (CONFFLAG_MONITORTALKER | CONFFLAG_OPTIMIZETALKER)) {
int totalsilence;
if (user->talking == -1)
@@ -1285,7 +1331,8 @@
res = ast_dsp_silence(dsp, f, &totalsilence);
if (!user->talking && totalsilence < MEETME_DELAYDETECTTALK) {
user->talking = 1;
- manager_event(EVENT_FLAG_CALL, "MeetmeTalking",
+ if (confflags & CONFFLAG_MONITORTALKER)
+ manager_event(EVENT_FLAG_CALL, "MeetmeTalking",
"Channel: %s\r\n"
"Uniqueid: %s\r\n"
"Meetme: %s\r\n"
@@ -1294,7 +1341,8 @@
}
if (user->talking && totalsilence > MEETME_DELAYDETECTENDTALK) {
user->talking = 0;
- manager_event(EVENT_FLAG_CALL, "MeetmeStopTalking",
+ if (confflags & CONFFLAG_MONITORTALKER)
+ manager_event(EVENT_FLAG_CALL, "MeetmeStopTalking",
"Channel: %s\r\n"
"Uniqueid: %s\r\n"
"Meetme: %s\r\n"
@@ -1315,7 +1363,8 @@
don't want to block, but we do want to at least *try*
to write out all the samples.
*/
- careful_write(fd, f->data, f->datalen, 0);
+ if (user->talking || !(confflags & CONFFLAG_OPTIMIZETALKER))
+ careful_write(fd, f->data, f->datalen, 0);
}
} else if ((f->frametype == AST_FRAME_DTMF) && (confflags & CONFFLAG_EXIT_CONTEXT)) {
char tmp[2];
@@ -1511,10 +1560,46 @@
fr.samples = res/2;
fr.data = buf;
fr.offset = AST_FRIENDLY_OFFSET;
- if (user->listen.actual)
- ast_frame_adjust_volume(&fr, user->listen.actual);
- if (ast_write(chan, &fr) < 0) {
- ast_log(LOG_WARNING, "Unable to write frame to channel: %s\n", strerror(errno));
+ if (!user->listen.actual &&
+ ((confflags & CONFFLAG_MONITOR) ||
+ (user->adminflags & ADMINFLAG_MUTED) ||
+ (user->talking && (confflags & CONFFLAG_OPTIMIZETALKER))
+ )) {
+ int index;
+ for (index=0;index<AST_FRAME_BITS;index++)
+ if (chan->rawwriteformat & (1 << index))
+ break;
+ if (index >= AST_FRAME_BITS)
+ goto bailoutandtrynormal;
+ ast_mutex_lock(&conf->listenlock);
+ if (!conf->transframe[index]) {
+ if (conf->origframe) {
+ if (!conf->transpath[index])
+ conf->transpath[index] = ast_translator_build_path((1 << index), AST_FORMAT_SLINEAR);
+ if (conf->transpath[index]) {
+ conf->transframe[index] = ast_translate(conf->transpath[index], conf->origframe, 0);
+ if (!conf->transframe[index])
+ conf->transframe[index] = &null_frame;
+ }
+ }
+ }
+ if (conf->transframe[index]) {
+ if (conf->transframe[index]->frametype != AST_FRAME_NULL) {
+ if (ast_write(chan, conf->transframe[index]))
+ ast_log(LOG_WARNING, "Unable to write frame to channel: %s\n", strerror(errno));
+ }
+ } else {
+ ast_mutex_unlock(&conf->listenlock);
+ goto bailoutandtrynormal;
+ }
+ ast_mutex_unlock(&conf->listenlock);
+ } else {
+bailoutandtrynormal:
+ if (user->listen.actual)
+ ast_frame_adjust_volume(&fr, user->listen.actual);
+ if (ast_write(chan, &fr) < 0) {
+ ast_log(LOG_WARNING, "Unable to write frame to channel: %s\n", strerror(errno));
+ }
}
} else
ast_log(LOG_WARNING, "Failed to read frame: %s\n", strerror(errno));
@@ -1556,7 +1641,7 @@
outrun:
ast_mutex_lock(&conflock);
- if (confflags & CONFFLAG_MONITORTALKER && dsp)
+ if (dsp)
ast_dsp_free(dsp);
if (user->user_no) { /* Only cleanup users who really joined! */
@@ -1998,15 +2083,16 @@
return res;
}
-static struct ast_conf_user* find_user(struct ast_conference *conf, char *callerident) {
+static struct ast_conf_user* find_user(struct ast_conference *conf, char *callerident)
+{
struct ast_conf_user *user = NULL;
- char usrno[1024] = "";
-
+ int cid;
+
+ sscanf(callerident, "%i", &cid);
if (conf && callerident) {
user = conf->firstuser;
while (user) {
- snprintf(usrno, sizeof(usrno), "%d", user->user_no);
- if (strcmp(usrno, callerident) == 0)
+ if (cid == user->user_no)
return user;
user = user->nextuser;
}
@@ -2100,7 +2186,7 @@
if (user && (user->adminflags & ADMINFLAG_MUTED)) {
user->adminflags ^= ADMINFLAG_MUTED;
} else {
- ast_log(LOG_NOTICE, "Specified User not found or he muted himself!");
+ ast_log(LOG_NOTICE, "Specified User not found or he muted himself!\n");
}
break;
case 110: /* n: Unmute all users */
@@ -2140,39 +2226,62 @@
struct ast_conference *cnf = args;
struct ast_frame *f=NULL;
int flags;
- struct ast_filestream *s;
+ struct ast_filestream *s=NULL;
int res=0;
+ int x;
+ const char *oldrecordingfilename = NULL;
if (!cnf || !cnf->chan) {
pthread_exit(0);
}
+
ast_stopstream(cnf->chan);
flags = O_CREAT|O_TRUNC|O_WRONLY;
- s = ast_writefile(cnf->recordingfilename, cnf->recordingformat, NULL, flags, 0, 0644);
-
- if (s) {
- cnf->recording = MEETME_RECORD_ACTIVE;
- while (ast_waitfor(cnf->chan, -1) > -1) {
- f = ast_read(cnf->chan);
- if (!f) {
- res = -1;
+
+
+ cnf->recording = MEETME_RECORD_ACTIVE;
+ while (ast_waitfor(cnf->chan, -1) > -1) {
+ if (cnf->recording == MEETME_RECORD_TERMINATE) {
+ ast_mutex_lock(&conflock);
+ ast_mutex_unlock(&conflock);
+ break;
+ }
+ if (!s && cnf->recordingfilename && (cnf->recordingfilename != oldrecordingfilename)) {
+ s = ast_writefile(cnf->recordingfilename, cnf->recordingformat, NULL, flags, 0, 0644);
+ oldrecordingfilename = cnf->recordingfilename;
+ }
+
+ f = ast_read(cnf->chan);
+ if (!f) {
+ res = -1;
+ break;
+ }
+ if (f->frametype == AST_FRAME_VOICE) {
+ ast_mutex_lock(&cnf->listenlock);
+ for (x=0;x<AST_FRAME_BITS;x++) {
+ /* Free any translations that have occured */
+ if (cnf->transframe[x]) {
+ ast_frfree(cnf->transframe[x]);
+ cnf->transframe[x] = NULL;
+ }
+ if (cnf->origframe)
+ ast_frfree(cnf->origframe);
+ cnf->origframe = f;
+ }
+ ast_mutex_unlock(&cnf->listenlock);
+ if (s)
+ res = ast_writestream(s, f);
+ if (res) {
+ ast_frfree(f);
break;
}
- if (f->frametype == AST_FRAME_VOICE) {
- res = ast_writestream(s, f);
- if (res)
- break;
- }
- ast_frfree(f);
- if (cnf->recording == MEETME_RECORD_TERMINATE) {
- ast_mutex_lock(&conflock);
- ast_mutex_unlock(&conflock);
- break;
- }
- }
- cnf->recording = MEETME_RECORD_OFF;
+ }
+ ast_frfree(f);
+ }
+ cnf->recording = MEETME_RECORD_OFF;
+ if (s)
ast_closestream(s);
- }
+
pthread_exit(0);
}
Modified: trunk/channel.c
URL: http://svn.digium.com/view/asterisk/trunk/channel.c?rev=7547&r1=7546&r2=7547&view=diff
==============================================================================
--- trunk/channel.c (original)
+++ trunk/channel.c Tue Dec 20 07:07:02 2005
@@ -1773,7 +1773,7 @@
return 0; /* Time is up */
}
-struct ast_frame *ast_read(struct ast_channel *chan)
+static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
{
struct ast_frame *f = NULL;
int blah;
@@ -1897,7 +1897,10 @@
if (f && (f->frametype == AST_FRAME_VOICE)) {
- if (!(f->subclass & chan->nativeformats)) {
+ if (dropaudio) {
+ ast_frfree(f);
+ f = &null_frame;
+ } else if (!(f->subclass & chan->nativeformats)) {
/* This frame can't be from the current native formats -- drop it on the
floor */
ast_log(LOG_NOTICE, "Dropping incompatible voice frame on %s of format %s since our native format has changed to %s\n", chan->name, ast_getformatname(f->subclass), ast_getformatname(chan->nativeformats));
@@ -1996,6 +1999,16 @@
chan->fin++;
ast_mutex_unlock(&chan->lock);
return f;
+}
+
+struct ast_frame *ast_read(struct ast_channel *chan)
+{
+ return __ast_read(chan, 0);
+}
+
+struct ast_frame *ast_read_noaudio(struct ast_channel *chan)
+{
+ return __ast_read(chan, 1);
}
int ast_indicate(struct ast_channel *chan, int condition)
@@ -2247,7 +2260,12 @@
break;
default:
if (chan->tech->write) {
- f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
+ /* Bypass translator if we're writing format in the raw write format. This
+ allows mixing of native / non-native formats */
+ if (fr->subclass == chan->rawwriteformat)
+ f = fr;
+ else
+ f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
if (f) {
if (f->frametype == AST_FRAME_VOICE && chan->spies)
queue_frame_to_spies(chan, f, SPY_WRITE);
Modified: trunk/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/channel.h?rev=7547&r1=7546&r2=7547&view=diff
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Tue Dec 20 07:07:02 2005
@@ -741,6 +741,15 @@
disconnected. */
struct ast_frame *ast_read(struct ast_channel *chan);
+/*! Reads a frame, returning AST_FRAME_NULL frame if audio. */
+/*!
+ * \param chan channel to read a frame from
+ * Read a frame. Returns a frame, or NULL on error. If it returns NULL, you
+ best just stop reading frames and assume the channel has been
+ disconnected. Audio is replaced with AST_FRAME_NULL to avoid
+ transcode when the resulting audio is not necessary. */
+struct ast_frame *ast_read_noaudio(struct ast_channel *chan);
+
/*! Write a frame to a channel */
/*!
* \param chan destination channel of the frame
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