From 36a5806089f9ef165b81c28d1e06e0548f815a36 Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Fri, 12 Nov 1999 23:51:16 +0000
Subject: [PATCH] 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 | 189 ++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 189 insertions(+)
create mode 100755 apps/app_intercom.c
diff --git a/apps/app_intercom.c b/apps/app_intercom.c
new file mode 100755
index 0000000000..cf078b70ac
--- /dev/null
+++ b/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;
+}
--
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