Version 0.1.0 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_intercom.c

@@ -0,0 +1,189 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as an intercom.
+ * 
+ * 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/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <unistd.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <string.h>
+#include <stdlib.h>
+#include <pthread.h>
+#include <sys/time.h>
+#include <linux/soundcard.h>
+#include <netinet/in.h>
+
+#define DEV_DSP "/dev/dsp"
+
+/* Number of 32 byte buffers -- each buffer is 2 ms */
+#define BUFFER_SIZE 32
+
+static char *tdesc = "Intercom using /dev/dsp for output";
+
+static char *app = "Intercom";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
+static int sound = -1;
+
+static int write_audio(short *data, int len)
+{
+	int res;
+	struct audio_buf_info info;
+	pthread_mutex_lock(&sound_lock);
+	if (sound < 0) {
+		ast_log(LOG_WARNING, "Sound device closed?\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+    if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
+		ast_log(LOG_WARNING, "Unable to read output space\n");
+		pthread_mutex_unlock(&sound_lock);
+        return -1;
+    }
+		res = write(sound, data, len);
+	pthread_mutex_unlock(&sound_lock);
+	return res;
+}
+
+static int create_audio()
+{
+	int fmt, desired, res, fd;
+	fd = open(DEV_DSP, O_WRONLY);
+	if (fd < 0) {
+		ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+		close(fd);
+		return -1;
+	}
+	fmt = AFMT_S16_LE;
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		close(fd);
+		return -1;
+	}
+	fmt = 0;
+	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	/* 8000 Hz desired */
+	desired = 8000;
+	fmt = desired;
+	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		close(fd);
+		return -1;
+	}
+	if (fmt != desired) {
+		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n");
+	}
+#if 1
+	/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
+	fmt = ((BUFFER_SIZE) << 16) | (0x0005);
+	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+	}
+#endif
+	sound = fd;
+	return 0;
+}
+
+static int intercom_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct localuser *u;
+	struct ast_frame *f;
+	struct ast_channel *trans;
+	if (!data) {
+		ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	/* See if we need a translator */
+	if (!(chan->format & AST_FORMAT_SLINEAR)) 
+		trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_IN);
+	else
+		trans = chan;
+	if (trans) {
+		/* Read packets from the channel */
+		while(!res) {
+			res = ast_waitfor(trans, -1);
+			if (res > 0) {
+				res = 0;
+				f = ast_read(trans);
+				if (f) {
+					if (f->frametype == AST_FRAME_DTMF) {
+						ast_frfree(f);
+						break;
+					} else {
+						if (f->frametype == AST_FRAME_VOICE) {
+							if (f->subclass == AST_FORMAT_SLINEAR) {
+								res = write_audio(f->data, f->datalen);
+								if (res > 0)
+									res = 0;
+							} else
+								ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
+						}
+					}
+					ast_frfree(f);
+				} else
+					res = -1;
+			}
+		}
+		if (trans != chan)
+			ast_translator_destroy(trans);
+	} else
+		ast_log(LOG_WARNING, "Unable to build translator to signed linear format on '%s'\n", chan->name);
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	if (sound > -1)
+		close(sound);
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	if (create_audio())
+		return -1;
+	return ast_register_application(app, intercom_exec);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}