remove obsolete stuff

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9124 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 0 insertions, 236 deletions

diff --git a/apps/app_intercom.c b/apps/app_intercom.c
deleted file mode 100644
index 28d258c28..000000000
--- a/apps/app_intercom.c
+++ /dev/null
@@ -1,236 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*! \file
- *
- * \brief Use /dev/dsp as an intercom.
- *
- * \author Mark Spencer <markster@digium.com>
- * 
- * \ingroup applications
- */
- 
-#include <stdio.h>
-#include <unistd.h>
-#include <errno.h>
-#include <sys/ioctl.h>
-#include <string.h>
-#include <stdlib.h>
-#include <sys/time.h>
-#include <netinet/in.h>
-
-#if defined(__linux__)
-#include <linux/soundcard.h>
-#elif defined(__FreeBSD__)
-#include <sys/soundcard.h>
-#else
-#include <soundcard.h>
-#endif
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#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"
-
-#ifdef __OpenBSD__
-#define DEV_DSP "/dev/audio"
-#else
-#define DEV_DSP "/dev/dsp"
-#endif
-
-/* 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";
-
-static char *synopsis = "(Obsolete) Send to Intercom";
-static char *descrip = 
-"  Intercom(): Sends the user to the intercom (i.e. /dev/dsp).  This program\n"
-"is generally considered  obselete by the chan_oss module.  User can terminate\n"with a DTMF tone, or by hangup.\n";
-
-STANDARD_LOCAL_USER;
-
-LOCAL_USER_DECL;
-
-AST_MUTEX_DEFINE_STATIC(sound_lock);
-static int sound = -1;
-
-static int write_audio(short *data, int len)
-{
-	int res;
-	struct audio_buf_info info;
-	ast_mutex_lock(&sound_lock);
-	if (sound < 0) {
-		ast_log(LOG_WARNING, "Sound device closed?\n");
-		ast_mutex_unlock(&sound_lock);
-		return -1;
-	}
-    if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
-		ast_log(LOG_WARNING, "Unable to read output space\n");
-		ast_mutex_unlock(&sound_lock);
-        return -1;
-    }
-	res = write(sound, data, len);
-	ast_mutex_unlock(&sound_lock);
-	return res;
-}
-
-static int create_audio(void)
-{
-	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", desired, fmt);
-	}
-#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;
-	int oreadformat;
-	LOCAL_USER_ADD(u);
-	/* Remember original read format */
-	oreadformat = chan->readformat;
-	/* Set mode to signed linear */
-	res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
-		LOCAL_USER_REMOVE(u);
-		return -1;
-	}
-	/* Read packets from the channel */
-	while(!res) {
-		res = ast_waitfor(chan, -1);
-		if (res > 0) {
-			res = 0;
-			f = ast_read(chan);
-			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 (!res)
-		ast_set_read_format(chan, oreadformat);
-
-	LOCAL_USER_REMOVE(u);
-
-	return res;
-}
-
-int unload_module(void)
-{
-	int res;
-
-	if (sound > -1)
-		close(sound);
-
-	res = ast_unregister_application(app);
-
-	STANDARD_HANGUP_LOCALUSERS;
-
-	return res;
-}
-
-int load_module(void)
-{
-	if (create_audio())
-		return -1;
-	return ast_register_application(app, intercom_exec, synopsis, descrip);
-}
-
-char *description(void)
-{
-	return tdesc;
-}
-
-int usecount(void)
-{
-	int res;
-	STANDARD_USECOUNT(res);
-	return res;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}