remove API function that was added and never used

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app.c

@@ -144,86 +144,6 @@ int ast_app_getdata_full(struct ast_channel *c, char *prompt, char *s, int maxle
 	return res;
 }
 
-int ast_app_getvoice(struct ast_channel *c, char *dest, char *dstfmt, char *prompt, int silence, int maxsec)
-{
-	int res;
-	struct ast_filestream *writer;
-	int rfmt;
-	int totalms=0, total;
-	
-	struct ast_frame *f;
-	struct ast_dsp *sildet;
-	/* Play prompt if requested */
-	if (prompt) {
-		res = ast_stream_and_wait(c, prompt, c->language, "");
-		if (res < 0)
-			return res;
-	}
-	rfmt = c->readformat;
-	res = ast_set_read_format(c, AST_FORMAT_SLINEAR);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
-		return -1;
-	}
-	sildet = ast_dsp_new();
-	if (!sildet) {
-		ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
-		return -1;
-	}
-	writer = ast_writefile(dest, dstfmt, "Voice file", 0, 0, 0666);
-	if (!writer) {
-		ast_log(LOG_WARNING, "Unable to open file '%s' in format '%s' for writing\n", dest, dstfmt);
-		ast_dsp_free(sildet);
-		return -1;
-	}
-	for(;;) {
-		if ((res = ast_waitfor(c, 2000)) < 0) {
-			ast_log(LOG_NOTICE, "Waitfor failed while recording file '%s' format '%s'\n", dest, dstfmt);
-			break;
-		}
-		if (res) {
-			f = ast_read(c);
-			if (!f) {
-				ast_log(LOG_NOTICE, "Hungup while recording file '%s' format '%s'\n", dest, dstfmt);
-				break;
-			}
-			if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
-				/* Ended happily with DTMF */
-				ast_frfree(f);
-				break;
-			} else if (f->frametype == AST_FRAME_VOICE) {
-				ast_dsp_silence(sildet, f, &total); 
-				if (total > silence) {
-					/* Ended happily with silence */
-					ast_frfree(f);
-					break;
-				}
-				totalms += f->samples / 8;
-				if (totalms > maxsec * 1000) {
-					/* Ended happily with too much stuff */
-					ast_log(LOG_NOTICE, "Constraining voice on '%s' to %d seconds\n", c->name, maxsec);
-					ast_frfree(f);
-					break;
-				}
-				res = ast_writestream(writer, f);
-				if (res < 0) {
-					ast_log(LOG_WARNING, "Failed to write to stream at %s!\n", dest);
-					ast_frfree(f);
-					break;
-				}
-					
-			}
-			ast_frfree(f);
-		}
-	}
-	res = ast_set_read_format(c, rfmt);
-	if (res)
-		ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", c->name);
-	ast_dsp_free(sildet);
-	ast_closestream(writer);
-	return 0;
-}
-
 static int (*ast_has_voicemail_func)(const char *mailbox, const char *folder) = NULL;
 static int (*ast_messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs) = NULL;
 

include/asterisk/app.h

@@ -99,9 +99,6 @@ int ast_app_getdata(struct ast_channel *c, char *prompt, char *s, int maxlen, in
 /*! \brief Full version with audiofd and controlfd.  NOTE: returns '2' on ctrlfd available, not '1' like other full functions */
 int ast_app_getdata_full(struct ast_channel *c, char *prompt, char *s, int maxlen, int timeout, int audiofd, int ctrlfd);
 
-/*! Record voice (after playing prompt if specified), waiting for silence (in ms) up to a given timeout (in s) or '#' */
-int ast_app_getvoice(struct ast_channel *c, char *dest, char *dstfmt, char *prompt, int silence, int maxsec);
-
 void ast_install_vm_functions(int (*has_voicemail_func)(const char *mailbox, const char *folder),
 			      int (*messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs));