Version 0.1.0 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_mp3.c

+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Silly application to play an MP3 file -- uses mpg123
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/frame.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <string.h>
+#include <stdio.h>
+#include <signal.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <pthread.h>
+#include <sys/time.h>
+
+#define MPG_123 "/usr/bin/mpg123"
+
+static char *tdesc = "Silly MP3 Application";
+
+static char *app = "MP3Player";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int mp3play(char *filename, int fd)
+{
+	int res;
+	res = fork();
+	if (res < 0) 
+		ast_log(LOG_WARNING, "Fork failed\n");
+	if (res)
+		return res;
+	dup2(fd, STDOUT_FILENO);
+	/* Execute mpg123, but buffer if it's a net connection */
+	if (strncmp(filename, "http://", 7)) 
+	    execl(MPG_123, MPG_123, "-q", "-s", "-b", "1024", "--mono", "-r", "8000", filename, NULL);
+	else
+	    execl(MPG_123, MPG_123, "-q", "-s", "--mono", "-r", "8000", filename, NULL);
+	ast_log(LOG_WARNING, "Execute of mpg123 failed\n");
+	return -1;
+}
+
+static int mp3_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	struct ast_channel *trans;
+	int fds[2];
+	int rfds[2];
+	int ms = -1;
+	int pid;
+	int us;
+	struct timeval tv;
+	struct timeval last;
+	struct ast_frame *f;
+	struct myframe {
+		struct ast_frame f;
+		char offset[AST_FRIENDLY_OFFSET];
+		char frdata[160];
+	} myf;
+	last.tv_usec = 0;
+	last.tv_sec = 0;
+	if (!data) {
+		ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
+		return -1;
+	}
+	if (pipe(fds)) {
+		ast_log(LOG_WARNING, "Unable to create pipe\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	ast_stopstream(chan);
+	if (chan->format & AST_FORMAT_SLINEAR)
+		trans = chan;
+	else
+		trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_OUT);
+	if (trans) {
+		res = mp3play((char *)data, fds[1]);
+		if (res >= 0) {
+			pid = res;
+			/* Order is important -- there's almost always going to be mp3...  we want to prioritize the
+			   user */
+			rfds[0] = trans->fd;
+			rfds[1] = fds[0];
+			for (;;) {
+				CHECK_BLOCKING(trans);
+				res = ast_waitfor_n_fd(rfds, 2, &ms);
+				trans->blocking = 0;
+				if (res < 1) {
+					ast_log(LOG_DEBUG, "Hangup detected\n");
+					res = -1;
+					break;
+				} else if (res == trans->fd) {
+					f = ast_read(trans);
+					if (!f) {
+						ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
+						res = -1;
+						break;
+					}
+					if (f->frametype == AST_FRAME_DTMF) {
+						ast_log(LOG_DEBUG, "User pressed a key\n");
+						ast_frfree(f);
+						res = 0;
+						break;
+					}
+					ast_frfree(f);
+				} else if (res == fds[0]) {
+					gettimeofday(&tv, NULL);
+					if (last.tv_sec || last.tv_usec) {
+						/* We should wait at least a frame length */
+						us = sizeof(myf.frdata) / 16 * 1000;
+						/* Subtract 1,000,000 us for each second late we've passed */
+						us -= (tv.tv_sec - last.tv_sec) * 1000000;
+						/* And one for each us late we've passed */
+						us -= (tv.tv_usec - last.tv_usec);
+						/* Sleep that long if needed */
+						if (us > 0)
+							usleep(us);
+					}
+					last = tv;
+					res = read(fds[0], myf.frdata, sizeof(myf.frdata));
+					if (res > 0) {
+						myf.f.frametype = AST_FRAME_VOICE;
+						myf.f.subclass = AST_FORMAT_SLINEAR;
+						myf.f.datalen = res;
+						myf.f.timelen = res / 16;
+						myf.f.mallocd = 0;
+						myf.f.offset = AST_FRIENDLY_OFFSET;
+						myf.f.src = __PRETTY_FUNCTION__;
+						myf.f.data = myf.frdata;
+						if (ast_write(trans, &myf.f) < 0) {
+							res = -1;
+							break;
+						}
+					} else {
+						ast_log(LOG_DEBUG, "No more mp3\n");
+						res = 0;
+					}
+				} else {
+					ast_log(LOG_DEBUG, "HuhHHH?\n");
+					res = -1;
+					break;
+				}
+			}
+			kill(pid, SIGTERM);
+		}
+		if (trans != chan) 
+			ast_translator_destroy(trans);
+	} else 
+		ast_log(LOG_WARNING, "No translator channel available\n");
+	close(fds[0]);
+	close(fds[1]);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, mp3_exec);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}