Version 0.1.0 from FTP
authorMark Spencer <markster@digium.com>
Fri, 12 Nov 1999 23:51:16 +0000 (23:51 +0000)
committerMark Spencer <markster@digium.com>
Fri, 12 Nov 1999 23:51:16 +0000 (23:51 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_intercom.c [new file with mode: 0755]

diff --git a/apps/app_intercom.c b/apps/app_intercom.c
new file mode 100755 (executable)
index 0000000..cf078b7
--- /dev/null
@@ -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;
+}