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1655 lines
46 KiB
1655 lines
46 KiB
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
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* note-this code best seen with ts=8 (8-spaces tabs) in the editor
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Channel driver for OSS sound cards
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*
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* \author Mark Spencer <markster@digium.com>
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* \author Luigi Rizzo
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*
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* \par See also
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* \arg \ref Config_oss
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>ossaudio</depend>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdio.h>
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#include <ctype.h>
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#include <math.h>
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#include <string.h>
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#include <unistd.h>
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#include <sys/ioctl.h>
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#include <fcntl.h>
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#include <sys/time.h>
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#include <stdlib.h>
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#include <errno.h>
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#ifdef __linux
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#include <linux/soundcard.h>
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#elif defined(__FreeBSD__)
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#include <sys/soundcard.h>
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#else
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#include <soundcard.h>
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#endif
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#include "asterisk/lock.h"
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#include "asterisk/frame.h"
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#include "asterisk/logger.h"
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#include "asterisk/callerid.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/options.h"
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#include "asterisk/pbx.h"
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#include "asterisk/config.h"
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#include "asterisk/cli.h"
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#include "asterisk/utils.h"
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#include "asterisk/causes.h"
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#include "asterisk/endian.h"
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#include "asterisk/stringfields.h"
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#include "asterisk/abstract_jb.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/app.h"
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/* ringtones we use */
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#include "busy.h"
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#include "ringtone.h"
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#include "ring10.h"
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#include "answer.h"
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/*! Global jitterbuffer configuration - by default, jb is disabled */
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static struct ast_jb_conf default_jbconf =
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{
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.flags = 0,
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.max_size = -1,
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.resync_threshold = -1,
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.impl = "",
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};
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static struct ast_jb_conf global_jbconf;
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/*
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* Basic mode of operation:
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*
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* we have one keyboard (which receives commands from the keyboard)
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* and multiple headset's connected to audio cards.
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* Cards/Headsets are named as the sections of oss.conf.
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* The section called [general] contains the default parameters.
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*
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* At any time, the keyboard is attached to one card, and you
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* can switch among them using the command 'console foo'
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* where 'foo' is the name of the card you want.
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*
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* oss.conf parameters are
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START_CONFIG
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[general]
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; General config options, with default values shown.
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; You should use one section per device, with [general] being used
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; for the first device and also as a template for other devices.
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;
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; All but 'debug' can go also in the device-specific sections.
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;
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; debug = 0x0 ; misc debug flags, default is 0
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; Set the device to use for I/O
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; device = /dev/dsp
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; Optional mixer command to run upon startup (e.g. to set
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; volume levels, mutes, etc.
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; mixer =
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; Software mic volume booster (or attenuator), useful for sound
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; cards or microphones with poor sensitivity. The volume level
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; is in dB, ranging from -20.0 to +20.0
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; boost = n ; mic volume boost in dB
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; Set the callerid for outgoing calls
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; callerid = John Doe <555-1234>
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; autoanswer = no ; no autoanswer on call
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; autohangup = yes ; hangup when other party closes
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; extension = s ; default extension to call
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; context = default ; default context for outgoing calls
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; language = "" ; default language
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; Default Music on Hold class to use when this channel is placed on hold in
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; the case that the music class is not set on the channel with
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; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
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; putting this one on hold did not suggest a class to use.
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;
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; mohinterpret=default
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; If you set overridecontext to 'yes', then the whole dial string
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; will be interpreted as an extension, which is extremely useful
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; to dial SIP, IAX and other extensions which use the '@' character.
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; The default is 'no' just for backward compatibility, but the
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; suggestion is to change it.
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; overridecontext = no ; if 'no', the last @ will start the context
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; if 'yes' the whole string is an extension.
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; low level device parameters in case you have problems with the
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; device driver on your operating system. You should not touch these
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; unless you know what you are doing.
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; queuesize = 10 ; frames in device driver
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; frags = 8 ; argument to SETFRAGMENT
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; OSS channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The OSS channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive OSS side will always
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; be used if the sending side can create jitter.
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usualy sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
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; channel. Two implementations are currenlty available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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[card1]
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; device = /dev/dsp1 ; alternate device
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END_CONFIG
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.. and so on for the other cards.
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*/
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/*
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* Helper macros to parse config arguments. They will go in a common
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* header file if their usage is globally accepted. In the meantime,
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* we define them here. Typical usage is as below.
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* Remember to open a block right before M_START (as it declares
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* some variables) and use the M_* macros WITHOUT A SEMICOLON:
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*
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* {
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* M_START(v->name, v->value)
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*
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* M_BOOL("dothis", x->flag1)
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* M_STR("name", x->somestring)
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* M_F("bar", some_c_code)
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* M_END(some_final_statement)
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* ... other code in the block
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* }
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*
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* XXX NOTE these macros should NOT be replicated in other parts of asterisk.
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* Likely we will come up with a better way of doing config file parsing.
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*/
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#define M_START(var, val) \
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char *__s = var; char *__val = val;
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#define M_END(x) x;
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#define M_F(tag, f) if (!strcasecmp((__s), tag)) { f; } else
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#define M_BOOL(tag, dst) M_F(tag, (dst) = ast_true(__val) )
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#define M_UINT(tag, dst) M_F(tag, (dst) = strtoul(__val, NULL, 0) )
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#define M_STR(tag, dst) M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))
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/*
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* The following parameters are used in the driver:
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*
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* FRAME_SIZE the size of an audio frame, in samples.
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* 160 is used almost universally, so you should not change it.
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*
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* FRAGS the argument for the SETFRAGMENT ioctl.
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* Overridden by the 'frags' parameter in oss.conf
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*
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* Bits 0-7 are the base-2 log of the device's block size,
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* bits 16-31 are the number of blocks in the driver's queue.
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* There are a lot of differences in the way this parameter
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* is supported by different drivers, so you may need to
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* experiment a bit with the value.
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* A good default for linux is 30 blocks of 64 bytes, which
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* results in 6 frames of 320 bytes (160 samples).
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* FreeBSD works decently with blocks of 256 or 512 bytes,
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* leaving the number unspecified.
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* Note that this only refers to the device buffer size,
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* this module will then try to keep the lenght of audio
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* buffered within small constraints.
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*
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* QUEUE_SIZE The max number of blocks actually allowed in the device
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* driver's buffer, irrespective of the available number.
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* Overridden by the 'queuesize' parameter in oss.conf
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*
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* Should be >=2, and at most as large as the hw queue above
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* (otherwise it will never be full).
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*/
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#define FRAME_SIZE 160
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#define QUEUE_SIZE 10
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#if defined(__FreeBSD__)
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#define FRAGS 0x8
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#else
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#define FRAGS ( ( (6 * 5) << 16 ) | 0x6 )
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#endif
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/*
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* XXX text message sizes are probably 256 chars, but i am
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* not sure if there is a suitable definition anywhere.
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*/
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#define TEXT_SIZE 256
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#if 0
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#define TRYOPEN 1 /* try to open on startup */
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#endif
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#define O_CLOSE 0x444 /* special 'close' mode for device */
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/* Which device to use */
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#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
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#define DEV_DSP "/dev/audio"
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#else
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#define DEV_DSP "/dev/dsp"
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#endif
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#ifndef MIN
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#define MIN(a,b) ((a) < (b) ? (a) : (b))
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#endif
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#ifndef MAX
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#define MAX(a,b) ((a) > (b) ? (a) : (b))
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#endif
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static char *config = "oss.conf"; /* default config file */
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static int oss_debug;
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/*!
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* Each sound is made of 'datalen' samples of sound, repeated as needed to
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* generate 'samplen' samples of data, then followed by 'silencelen' samples
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* of silence. The loop is repeated if 'repeat' is set.
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*/
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struct sound {
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int ind;
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char *desc;
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short *data;
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int datalen;
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int samplen;
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int silencelen;
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int repeat;
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};
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static struct sound sounds[] = {
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{ AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
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{ AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
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{ AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
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{ AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
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{ AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
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{ -1, NULL, 0, 0, 0, 0 }, /* end marker */
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};
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/*!
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* \brief descriptor for one of our channels.
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*
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* There is one used for 'default' values (from the [general] entry in
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* the configuration file), and then one instance for each device
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* (the default is cloned from [general], others are only created
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* if the relevant section exists).
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*/
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struct chan_oss_pvt {
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struct chan_oss_pvt *next;
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char *name;
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/*!
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* cursound indicates which in struct sound we play. -1 means nothing,
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* any other value is a valid sound, in which case sampsent indicates
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* the next sample to send in [0..samplen + silencelen]
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* nosound is set to disable the audio data from the channel
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* (so we can play the tones etc.).
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*/
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int sndcmd[2]; /*!< Sound command pipe */
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int cursound; /*!< index of sound to send */
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int sampsent; /*!< # of sound samples sent */
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int nosound; /*!< set to block audio from the PBX */
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int total_blocks; /*!< total blocks in the output device */
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int sounddev;
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enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
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int autoanswer;
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int autohangup;
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int hookstate;
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char *mixer_cmd; /*!< initial command to issue to the mixer */
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unsigned int queuesize; /*!< max fragments in queue */
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unsigned int frags; /*!< parameter for SETFRAGMENT */
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int warned; /*!< various flags used for warnings */
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#define WARN_used_blocks 1
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#define WARN_speed 2
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#define WARN_frag 4
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int w_errors; /*!< overfull in the write path */
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struct timeval lastopen;
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int overridecontext;
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int mute;
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/*! boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
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* be representable in 16 bits to avoid overflows.
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*/
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#define BOOST_SCALE (1<<9)
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#define BOOST_MAX 40 /*!< slightly less than 7 bits */
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int boost; /*!< input boost, scaled by BOOST_SCALE */
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char device[64]; /*!< device to open */
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pthread_t sthread;
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struct ast_channel *owner;
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char ext[AST_MAX_EXTENSION];
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char ctx[AST_MAX_CONTEXT];
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char language[MAX_LANGUAGE];
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char cid_name[256]; /*XXX */
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char cid_num[256]; /*XXX */
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char mohinterpret[MAX_MUSICCLASS];
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/*! buffers used in oss_write */
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char oss_write_buf[FRAME_SIZE * 2];
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int oss_write_dst;
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/*! buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
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* plus enough room for a full frame
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*/
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char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
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int readpos; /*!< read position above */
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struct ast_frame read_f; /*!< returned by oss_read */
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};
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static struct chan_oss_pvt oss_default = {
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.cursound = -1,
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.sounddev = -1,
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.duplex = M_UNSET, /* XXX check this */
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.autoanswer = 1,
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.autohangup = 1,
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.queuesize = QUEUE_SIZE,
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.frags = FRAGS,
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.ext = "s",
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.ctx = "default",
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.readpos = AST_FRIENDLY_OFFSET, /* start here on reads */
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.lastopen = { 0, 0 },
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.boost = BOOST_SCALE,
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};
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static char *oss_active; /*!< the active device */
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static int setformat(struct chan_oss_pvt *o, int mode);
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static struct ast_channel *oss_request(const char *type, int format, void *data
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, int *cause);
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static int oss_digit_begin(struct ast_channel *c, char digit);
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static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
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static int oss_text(struct ast_channel *c, const char *text);
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static int oss_hangup(struct ast_channel *c);
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static int oss_answer(struct ast_channel *c);
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static struct ast_frame *oss_read(struct ast_channel *chan);
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static int oss_call(struct ast_channel *c, char *dest, int timeout);
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static int oss_write(struct ast_channel *chan, struct ast_frame *f);
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static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
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static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
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static char tdesc[] = "OSS Console Channel Driver";
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|
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static const struct ast_channel_tech oss_tech = {
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.type = "Console",
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.description = tdesc,
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.capabilities = AST_FORMAT_SLINEAR,
|
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.requester = oss_request,
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.send_digit_begin = oss_digit_begin,
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.send_digit_end = oss_digit_end,
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.send_text = oss_text,
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.hangup = oss_hangup,
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.answer = oss_answer,
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.read = oss_read,
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.call = oss_call,
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.write = oss_write,
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.indicate = oss_indicate,
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.fixup = oss_fixup,
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};
|
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|
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/*!
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* \brief returns a pointer to the descriptor with the given name
|
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*/
|
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static struct chan_oss_pvt *find_desc(char *dev)
|
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{
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struct chan_oss_pvt *o = NULL;
|
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|
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if (!dev)
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ast_log(LOG_WARNING, "null dev\n");
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for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);
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|
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if (!o)
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ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");
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|
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return o;
|
|
}
|
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|
|
/* !
|
|
* \brief split a string in extension-context, returns pointers to malloc'ed
|
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* strings.
|
|
*
|
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* If we do not have 'overridecontext' then the last @ is considered as
|
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* a context separator, and the context is overridden.
|
|
* This is usually not very necessary as you can play with the dialplan,
|
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* and it is nice not to need it because you have '@' in SIP addresses.
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*
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* \return the buffer address.
|
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*/
|
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static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
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{
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struct chan_oss_pvt *o = find_desc(oss_active);
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|
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if (ext == NULL || ctx == NULL)
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return NULL; /* error */
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|
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*ext = *ctx = NULL;
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|
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if (src && *src != '\0')
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*ext = ast_strdup(src);
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|
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if (*ext == NULL)
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return NULL;
|
|
|
|
if (!o->overridecontext) {
|
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/* parse from the right */
|
|
*ctx = strrchr(*ext, '@');
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if (*ctx)
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*(*ctx)++ = '\0';
|
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}
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|
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return *ext;
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}
|
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|
|
/*!
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|
* \brief Returns the number of blocks used in the audio output channel
|
|
*/
|
|
static int used_blocks(struct chan_oss_pvt *o)
|
|
{
|
|
struct audio_buf_info info;
|
|
|
|
if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
|
|
if (!(o->warned & WARN_used_blocks)) {
|
|
ast_log(LOG_WARNING, "Error reading output space\n");
|
|
o->warned |= WARN_used_blocks;
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
if (o->total_blocks == 0) {
|
|
if (0) /* debugging */
|
|
ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
|
|
o->total_blocks = info.fragments;
|
|
}
|
|
|
|
return o->total_blocks - info.fragments;
|
|
}
|
|
|
|
/*! Write an exactly FRAME_SIZE sized frame */
|
|
static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
|
|
{
|
|
int res;
|
|
|
|
if (o->sounddev < 0)
|
|
setformat(o, O_RDWR);
|
|
if (o->sounddev < 0)
|
|
return 0; /* not fatal */
|
|
/*
|
|
* Nothing complex to manage the audio device queue.
|
|
* If the buffer is full just drop the extra, otherwise write.
|
|
* XXX in some cases it might be useful to write anyways after
|
|
* a number of failures, to restart the output chain.
|
|
*/
|
|
res = used_blocks(o);
|
|
if (res > o->queuesize) { /* no room to write a block */
|
|
if (o->w_errors++ == 0 && (oss_debug & 0x4))
|
|
ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
|
|
return 0;
|
|
}
|
|
o->w_errors = 0;
|
|
return write(o->sounddev, (void *)data, FRAME_SIZE * 2);
|
|
}
|
|
|
|
/*!
|
|
* \brief Handler for 'sound writable' events from the sound thread.
|
|
*
|
|
* Builds a frame from the high level description of the sounds,
|
|
* and passes it to the audio device.
|
|
* The actual sound is made of 1 or more sequences of sound samples
|
|
* (s->datalen, repeated to make s->samplen samples) followed by
|
|
* s->silencelen samples of silence. The position in the sequence is stored
|
|
* in o->sampsent, which goes between 0 .. s->samplen+s->silencelen.
|
|
* In case we fail to write a frame, don't update o->sampsent.
|
|
*/
|
|
static void send_sound(struct chan_oss_pvt *o)
|
|
{
|
|
short myframe[FRAME_SIZE];
|
|
int ofs, l, start;
|
|
int l_sampsent = o->sampsent;
|
|
struct sound *s;
|
|
|
|
if (o->cursound < 0) /* no sound to send */
|
|
return;
|
|
|
|
s = &sounds[o->cursound];
|
|
|
|
for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
|
|
l = s->samplen - l_sampsent; /* # of available samples */
|
|
if (l > 0) {
|
|
start = l_sampsent % s->datalen; /* source offset */
|
|
l = MIN(l, FRAME_SIZE - ofs); /* don't overflow the frame */
|
|
l = MIN(l, s->datalen - start); /* don't overflow the source */
|
|
bcopy(s->data + start, myframe + ofs, l * 2);
|
|
if (0)
|
|
ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", l_sampsent, l, s->samplen, ofs);
|
|
l_sampsent += l;
|
|
} else { /* end of samples, maybe some silence */
|
|
static const short silence[FRAME_SIZE] = { 0, };
|
|
|
|
l += s->silencelen;
|
|
if (l > 0) {
|
|
l = MIN(l, FRAME_SIZE - ofs);
|
|
bcopy(silence, myframe + ofs, l * 2);
|
|
l_sampsent += l;
|
|
} else { /* silence is over, restart sound if loop */
|
|
if (s->repeat == 0) { /* last block */
|
|
o->cursound = -1;
|
|
o->nosound = 0; /* allow audio data */
|
|
if (ofs < FRAME_SIZE) /* pad with silence */
|
|
bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs) * 2);
|
|
}
|
|
l_sampsent = 0;
|
|
}
|
|
}
|
|
}
|
|
l = soundcard_writeframe(o, myframe);
|
|
if (l > 0)
|
|
o->sampsent = l_sampsent; /* update status */
|
|
}
|
|
|
|
static void *sound_thread(void *arg)
|
|
{
|
|
char ign[4096];
|
|
struct chan_oss_pvt *o = (struct chan_oss_pvt *) arg;
|
|
|
|
/*
|
|
* Just in case, kick the driver by trying to read from it.
|
|
* Ignore errors - this read is almost guaranteed to fail.
|
|
*/
|
|
read(o->sounddev, ign, sizeof(ign));
|
|
for (;;) {
|
|
fd_set rfds, wfds;
|
|
int maxfd, res;
|
|
struct timeval *to = NULL, t;
|
|
|
|
FD_ZERO(&rfds);
|
|
FD_ZERO(&wfds);
|
|
FD_SET(o->sndcmd[0], &rfds);
|
|
maxfd = o->sndcmd[0]; /* pipe from the main process */
|
|
if (o->cursound > -1 && o->sounddev < 0)
|
|
setformat(o, O_RDWR); /* need the channel, try to reopen */
|
|
else if (o->cursound == -1 && o->owner == NULL)
|
|
setformat(o, O_CLOSE); /* can close */
|
|
if (o->sounddev > -1) {
|
|
if (!o->owner) { /* no one owns the audio, so we must drain it */
|
|
FD_SET(o->sounddev, &rfds);
|
|
maxfd = MAX(o->sounddev, maxfd);
|
|
}
|
|
if (o->cursound > -1) {
|
|
/*
|
|
* We would like to use select here, but the device
|
|
* is always writable, so this would become busy wait.
|
|
* So we rather set a timeout to 1/2 of the frame size.
|
|
*/
|
|
t.tv_sec = 0;
|
|
t.tv_usec = (1000000 * FRAME_SIZE) / (5 * DEFAULT_SAMPLE_RATE);
|
|
to = &t;
|
|
}
|
|
}
|
|
/* ast_select emulates linux behaviour in terms of timeout handling */
|
|
res = ast_select(maxfd + 1, &rfds, &wfds, NULL, to);
|
|
if (res < 0) {
|
|
ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
|
|
sleep(1);
|
|
continue;
|
|
}
|
|
if (FD_ISSET(o->sndcmd[0], &rfds)) {
|
|
/* read which sound to play from the pipe */
|
|
int i, what = -1;
|
|
|
|
read(o->sndcmd[0], &what, sizeof(what));
|
|
for (i = 0; sounds[i].ind != -1; i++) {
|
|
if (sounds[i].ind == what) {
|
|
o->cursound = i;
|
|
o->sampsent = 0;
|
|
o->nosound = 1; /* block audio from pbx */
|
|
break;
|
|
}
|
|
}
|
|
if (sounds[i].ind == -1)
|
|
ast_log(LOG_WARNING, "invalid sound index: %d\n", what);
|
|
}
|
|
if (o->sounddev > -1) {
|
|
if (FD_ISSET(o->sounddev, &rfds)) /* read and ignore errors */
|
|
read(o->sounddev, ign, sizeof(ign));
|
|
if (to != NULL) /* maybe it is possible to write */
|
|
send_sound(o);
|
|
}
|
|
}
|
|
return NULL; /* Never reached */
|
|
}
|
|
|
|
/*!
|
|
* reset and close the device if opened,
|
|
* then open and initialize it in the desired mode,
|
|
* trigger reads and writes so we can start using it.
|
|
*/
|
|
static int setformat(struct chan_oss_pvt *o, int mode)
|
|
{
|
|
int fmt, desired, res, fd;
|
|
|
|
if (o->sounddev >= 0) {
|
|
ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
|
|
close(o->sounddev);
|
|
o->duplex = M_UNSET;
|
|
o->sounddev = -1;
|
|
}
|
|
if (mode == O_CLOSE) /* we are done */
|
|
return 0;
|
|
if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
|
|
return -1; /* don't open too often */
|
|
o->lastopen = ast_tvnow();
|
|
fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
|
|
if (fd < 0) {
|
|
ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
|
|
return -1;
|
|
}
|
|
if (o->owner)
|
|
ast_channel_set_fd(o->owner, 0, fd);
|
|
|
|
#if __BYTE_ORDER == __LITTLE_ENDIAN
|
|
fmt = AFMT_S16_LE;
|
|
#else
|
|
fmt = AFMT_S16_BE;
|
|
#endif
|
|
res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
|
|
if (res < 0) {
|
|
ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
|
|
return -1;
|
|
}
|
|
switch (mode) {
|
|
case O_RDWR:
|
|
res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
|
|
/* Check to see if duplex set (FreeBSD Bug) */
|
|
res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
|
|
if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
|
|
ast_verb(2, "Console is full duplex\n");
|
|
o->duplex = M_FULL;
|
|
};
|
|
break;
|
|
|
|
case O_WRONLY:
|
|
o->duplex = M_WRITE;
|
|
break;
|
|
|
|
case O_RDONLY:
|
|
o->duplex = M_READ;
|
|
break;
|
|
}
|
|
|
|
fmt = 0;
|
|
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
|
|
if (res < 0) {
|
|
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
|
return -1;
|
|
}
|
|
fmt = desired = DEFAULT_SAMPLE_RATE; /* 8000 Hz desired */
|
|
res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
|
|
|
|
if (res < 0) {
|
|
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
|
return -1;
|
|
}
|
|
if (fmt != desired) {
|
|
if (!(o->warned & WARN_speed)) {
|
|
ast_log(LOG_WARNING,
|
|
"Requested %d Hz, got %d Hz -- sound may be choppy\n",
|
|
desired, fmt);
|
|
o->warned |= WARN_speed;
|
|
}
|
|
}
|
|
/*
|
|
* on Freebsd, SETFRAGMENT does not work very well on some cards.
|
|
* Default to use 256 bytes, let the user override
|
|
*/
|
|
if (o->frags) {
|
|
fmt = o->frags;
|
|
res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
|
|
if (res < 0) {
|
|
if (!(o->warned & WARN_frag)) {
|
|
ast_log(LOG_WARNING,
|
|
"Unable to set fragment size -- sound may be choppy\n");
|
|
o->warned |= WARN_frag;
|
|
}
|
|
}
|
|
}
|
|
/* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
|
|
res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
|
|
res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
|
|
/* it may fail if we are in half duplex, never mind */
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* some of the standard methods supported by channels.
|
|
*/
|
|
static int oss_digit_begin(struct ast_channel *c, char digit)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
|
|
{
|
|
/* no better use for received digits than print them */
|
|
ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
|
|
digit, duration);
|
|
return 0;
|
|
}
|
|
|
|
static int oss_text(struct ast_channel *c, const char *text)
|
|
{
|
|
/* print received messages */
|
|
ast_verbose(" << Console Received text %s >> \n", text);
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Play ringtone 'x' on device 'o' */
|
|
static void ring(struct chan_oss_pvt *o, int x)
|
|
{
|
|
write(o->sndcmd[1], &x, sizeof(x));
|
|
}
|
|
|
|
|
|
/*!
|
|
* \brief handler for incoming calls. Either autoanswer, or start ringing
|
|
*/
|
|
static int oss_call(struct ast_channel *c, char *dest, int timeout)
|
|
{
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
struct ast_frame f = { 0, };
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(name);
|
|
AST_APP_ARG(flags);
|
|
);
|
|
char *parse = ast_strdupa(dest);
|
|
|
|
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
|
|
|
|
ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n", dest, c->cid.cid_dnid, c->cid.cid_rdnis, c->cid.cid_name, c->cid.cid_num);
|
|
if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "answer") == 0) {
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_ANSWER;
|
|
ast_queue_frame(c, &f);
|
|
} else if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "noanswer") == 0) {
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_RINGING;
|
|
ast_queue_frame(c, &f);
|
|
ring(o, AST_CONTROL_RING);
|
|
} else if (o->autoanswer) {
|
|
ast_verbose(" << Auto-answered >> \n");
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_ANSWER;
|
|
ast_queue_frame(c, &f);
|
|
} else {
|
|
ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
|
|
f.frametype = AST_FRAME_CONTROL;
|
|
f.subclass = AST_CONTROL_RINGING;
|
|
ast_queue_frame(c, &f);
|
|
ring(o, AST_CONTROL_RING);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief remote side answered the phone
|
|
*/
|
|
static int oss_answer(struct ast_channel *c)
|
|
{
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
|
|
ast_verbose(" << Console call has been answered >> \n");
|
|
#if 0
|
|
/* play an answer tone (XXX do we really need it ?) */
|
|
ring(o, AST_CONTROL_ANSWER);
|
|
#endif
|
|
ast_setstate(c, AST_STATE_UP);
|
|
o->cursound = -1;
|
|
o->nosound = 0;
|
|
return 0;
|
|
}
|
|
|
|
static int oss_hangup(struct ast_channel *c)
|
|
{
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
|
|
o->cursound = -1;
|
|
o->nosound = 0;
|
|
c->tech_pvt = NULL;
|
|
o->owner = NULL;
|
|
ast_verbose(" << Hangup on console >> \n");
|
|
ast_module_unref(ast_module_info->self);
|
|
if (o->hookstate) {
|
|
if (o->autoanswer || o->autohangup) {
|
|
/* Assume auto-hangup too */
|
|
o->hookstate = 0;
|
|
setformat(o, O_CLOSE);
|
|
} else {
|
|
/* Make congestion noise */
|
|
ring(o, AST_CONTROL_CONGESTION);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief used for data coming from the network */
|
|
static int oss_write(struct ast_channel *c, struct ast_frame *f)
|
|
{
|
|
int src;
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
|
|
/* Immediately return if no sound is enabled */
|
|
if (o->nosound)
|
|
return 0;
|
|
/* Stop any currently playing sound */
|
|
o->cursound = -1;
|
|
/*
|
|
* we could receive a block which is not a multiple of our
|
|
* FRAME_SIZE, so buffer it locally and write to the device
|
|
* in FRAME_SIZE chunks.
|
|
* Keep the residue stored for future use.
|
|
*/
|
|
src = 0; /* read position into f->data */
|
|
while (src < f->datalen) {
|
|
/* Compute spare room in the buffer */
|
|
int l = sizeof(o->oss_write_buf) - o->oss_write_dst;
|
|
|
|
if (f->datalen - src >= l) { /* enough to fill a frame */
|
|
memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
|
|
soundcard_writeframe(o, (short *) o->oss_write_buf);
|
|
src += l;
|
|
o->oss_write_dst = 0;
|
|
} else { /* copy residue */
|
|
l = f->datalen - src;
|
|
memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
|
|
src += l; /* but really, we are done */
|
|
o->oss_write_dst += l;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_frame *oss_read(struct ast_channel *c)
|
|
{
|
|
int res;
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
struct ast_frame *f = &o->read_f;
|
|
|
|
/* XXX can be simplified returning &ast_null_frame */
|
|
/* prepare a NULL frame in case we don't have enough data to return */
|
|
bzero(f, sizeof(struct ast_frame));
|
|
f->frametype = AST_FRAME_NULL;
|
|
f->src = oss_tech.type;
|
|
|
|
res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
|
|
if (res < 0) /* audio data not ready, return a NULL frame */
|
|
return f;
|
|
|
|
o->readpos += res;
|
|
if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */
|
|
return f;
|
|
|
|
if (o->mute)
|
|
return f;
|
|
|
|
o->readpos = AST_FRIENDLY_OFFSET; /* reset read pointer for next frame */
|
|
if (c->_state != AST_STATE_UP) /* drop data if frame is not up */
|
|
return f;
|
|
/* ok we can build and deliver the frame to the caller */
|
|
f->frametype = AST_FRAME_VOICE;
|
|
f->subclass = AST_FORMAT_SLINEAR;
|
|
f->samples = FRAME_SIZE;
|
|
f->datalen = FRAME_SIZE * 2;
|
|
f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET;
|
|
if (o->boost != BOOST_SCALE) { /* scale and clip values */
|
|
int i, x;
|
|
int16_t *p = (int16_t *) f->data;
|
|
for (i = 0; i < f->samples; i++) {
|
|
x = (p[i] * o->boost) / BOOST_SCALE;
|
|
if (x > 32767)
|
|
x = 32767;
|
|
else if (x < -32768)
|
|
x = -32768;
|
|
p[i] = x;
|
|
}
|
|
}
|
|
|
|
f->offset = AST_FRIENDLY_OFFSET;
|
|
return f;
|
|
}
|
|
|
|
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
|
|
{
|
|
struct chan_oss_pvt *o = newchan->tech_pvt;
|
|
o->owner = newchan;
|
|
return 0;
|
|
}
|
|
|
|
static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
|
|
{
|
|
struct chan_oss_pvt *o = c->tech_pvt;
|
|
int res = -1;
|
|
|
|
switch (cond) {
|
|
case AST_CONTROL_BUSY:
|
|
case AST_CONTROL_CONGESTION:
|
|
case AST_CONTROL_RINGING:
|
|
res = cond;
|
|
break;
|
|
|
|
case -1:
|
|
o->cursound = -1;
|
|
o->nosound = 0; /* when cursound is -1 nosound must be 0 */
|
|
return 0;
|
|
|
|
case AST_CONTROL_VIDUPDATE:
|
|
res = -1;
|
|
break;
|
|
|
|
case AST_CONTROL_HOLD:
|
|
ast_verbose(" << Console Has Been Placed on Hold >> \n");
|
|
ast_moh_start(c, data, o->mohinterpret);
|
|
break;
|
|
|
|
case AST_CONTROL_UNHOLD:
|
|
ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
|
|
ast_moh_stop(c);
|
|
break;
|
|
|
|
default:
|
|
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name);
|
|
return -1;
|
|
}
|
|
|
|
if (res > -1)
|
|
ring(o, res);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief allocate a new channel.
|
|
*/
|
|
static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state)
|
|
{
|
|
struct ast_channel *c;
|
|
|
|
c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, 0, "OSS/%s", o->device + 5);
|
|
if (c == NULL)
|
|
return NULL;
|
|
c->tech = &oss_tech;
|
|
if (o->sounddev < 0)
|
|
setformat(o, O_RDWR);
|
|
ast_channel_set_fd(c, 0, o->sounddev); /* -1 if device closed, override later */
|
|
c->nativeformats = AST_FORMAT_SLINEAR;
|
|
c->readformat = AST_FORMAT_SLINEAR;
|
|
c->writeformat = AST_FORMAT_SLINEAR;
|
|
c->tech_pvt = o;
|
|
|
|
if (!ast_strlen_zero(o->language))
|
|
ast_string_field_set(c, language, o->language);
|
|
/* Don't use ast_set_callerid() here because it will
|
|
* generate a needless NewCallerID event */
|
|
c->cid.cid_ani = ast_strdup(o->cid_num);
|
|
if (!ast_strlen_zero(ext))
|
|
c->cid.cid_dnid = ast_strdup(ext);
|
|
|
|
o->owner = c;
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_jb_configure(c, &global_jbconf);
|
|
if (state != AST_STATE_DOWN) {
|
|
if (ast_pbx_start(c)) {
|
|
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
|
|
ast_hangup(c);
|
|
o->owner = c = NULL;
|
|
/* XXX what about the channel itself ? */
|
|
}
|
|
}
|
|
|
|
return c;
|
|
}
|
|
|
|
static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
|
|
{
|
|
struct ast_channel *c;
|
|
struct chan_oss_pvt *o;
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(name);
|
|
AST_APP_ARG(flags);
|
|
);
|
|
char *parse = ast_strdupa(data);
|
|
|
|
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
|
|
o = find_desc(args.name);
|
|
|
|
ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, (char *) data);
|
|
if (o == NULL) {
|
|
ast_log(LOG_NOTICE, "Device %s not found\n", args.name);
|
|
/* XXX we could default to 'dsp' perhaps ? */
|
|
return NULL;
|
|
}
|
|
if ((format & AST_FORMAT_SLINEAR) == 0) {
|
|
ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format);
|
|
return NULL;
|
|
}
|
|
if (o->owner) {
|
|
ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
|
|
*cause = AST_CAUSE_BUSY;
|
|
return NULL;
|
|
}
|
|
c = oss_new(o, NULL, NULL, AST_STATE_DOWN);
|
|
if (c == NULL) {
|
|
ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
|
|
return NULL;
|
|
}
|
|
return c;
|
|
}
|
|
|
|
static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console autoanswer [on|off]";
|
|
e->usage =
|
|
"Usage: console autoanswer [on|off]\n"
|
|
" Enables or disables autoanswer feature. If used without\n"
|
|
" argument, displays the current on/off status of autoanswer.\n"
|
|
" The default value of autoanswer is in 'oss.conf'.\n";
|
|
return NULL;
|
|
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc == e->args - 1) {
|
|
ast_cli(a->fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
|
|
return CLI_SUCCESS;
|
|
}
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
if (o == NULL) {
|
|
ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
|
|
oss_active);
|
|
return CLI_FAILURE;
|
|
}
|
|
if (!strcasecmp(a->argv[e->args-1], "on"))
|
|
o->autoanswer = 1;
|
|
else if (!strcasecmp(a->argv[e->args - 1], "off"))
|
|
o->autoanswer = 0;
|
|
else
|
|
return CLI_SHOWUSAGE;
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
/*!
|
|
* \brief answer command from the console
|
|
*/
|
|
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console answer";
|
|
e->usage =
|
|
"Usage: console answer\n"
|
|
" Answers an incoming call on the console (OSS) channel.\n";
|
|
return NULL;
|
|
|
|
case CLI_GENERATE:
|
|
return NULL; /* no completion */
|
|
}
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
if (!o->owner) {
|
|
ast_cli(a->fd, "No one is calling us\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
o->hookstate = 1;
|
|
o->cursound = -1;
|
|
o->nosound = 0;
|
|
ast_queue_frame(o->owner, &f);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
/*!
|
|
* \brief Console send text CLI command
|
|
*
|
|
* \note concatenate all arguments into a single string. argv is NULL-terminated
|
|
* so we can use it right away
|
|
*/
|
|
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
char buf[TEXT_SIZE];
|
|
|
|
if (cmd == CLI_INIT) {
|
|
e->command = "console send text";
|
|
e->usage =
|
|
"Usage: console send text <message>\n"
|
|
" Sends a text message for display on the remote terminal.\n";
|
|
return NULL;
|
|
} else if (cmd == CLI_GENERATE)
|
|
return NULL;
|
|
|
|
if (a->argc < e->args + 1)
|
|
return CLI_SHOWUSAGE;
|
|
if (!o->owner) {
|
|
ast_cli(a->fd, "Not in a call\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
|
|
if (!ast_strlen_zero(buf)) {
|
|
struct ast_frame f = { 0, };
|
|
int i = strlen(buf);
|
|
buf[i] = '\n';
|
|
f.frametype = AST_FRAME_TEXT;
|
|
f.subclass = 0;
|
|
f.data = buf;
|
|
f.datalen = i + 1;
|
|
ast_queue_frame(o->owner, &f);
|
|
}
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
if (cmd == CLI_INIT) {
|
|
e->command = "console hangup";
|
|
e->usage =
|
|
"Usage: console hangup\n"
|
|
" Hangs up any call currently placed on the console.\n";
|
|
return NULL;
|
|
} else if (cmd == CLI_GENERATE)
|
|
return NULL;
|
|
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
o->cursound = -1;
|
|
o->nosound = 0;
|
|
if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
|
|
ast_cli(a->fd, "No call to hang up\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
o->hookstate = 0;
|
|
if (o->owner)
|
|
ast_queue_hangup(o->owner);
|
|
setformat(o, O_CLOSE);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
if (cmd == CLI_INIT) {
|
|
e->command = "console flash";
|
|
e->usage =
|
|
"Usage: console flash\n"
|
|
" Flashes the call currently placed on the console.\n";
|
|
return NULL;
|
|
} else if (cmd == CLI_GENERATE)
|
|
return NULL;
|
|
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
o->cursound = -1;
|
|
o->nosound = 0; /* when cursound is -1 nosound must be 0 */
|
|
if (!o->owner) { /* XXX maybe !o->hookstate too ? */
|
|
ast_cli(a->fd, "No call to flash\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
o->hookstate = 0;
|
|
if (o->owner) /* XXX must be true, right ? */
|
|
ast_queue_frame(o->owner, &f);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *s = NULL, *mye = NULL, *myc = NULL;
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
if (cmd == CLI_INIT) {
|
|
e->command = "console dial";
|
|
e->usage =
|
|
"Usage: console dial [extension[@context]]\n"
|
|
" Dials a given extension (and context if specified)\n";
|
|
return NULL;
|
|
} else if (cmd == CLI_GENERATE)
|
|
return NULL;
|
|
|
|
if (a->argc > e->args + 1)
|
|
return CLI_SHOWUSAGE;
|
|
if (o->owner) { /* already in a call */
|
|
int i;
|
|
struct ast_frame f = { AST_FRAME_DTMF, 0 };
|
|
|
|
if (a->argc == e->args) { /* argument is mandatory here */
|
|
ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
s = a->argv[e->args];
|
|
/* send the string one char at a time */
|
|
for (i = 0; i < strlen(s); i++) {
|
|
f.subclass = s[i];
|
|
ast_queue_frame(o->owner, &f);
|
|
}
|
|
return CLI_SUCCESS;
|
|
}
|
|
/* if we have an argument split it into extension and context */
|
|
if (a->argc == e->args + 1)
|
|
s = ast_ext_ctx(a->argv[e->args], &mye, &myc);
|
|
/* supply default values if needed */
|
|
if (mye == NULL)
|
|
mye = o->ext;
|
|
if (myc == NULL)
|
|
myc = o->ctx;
|
|
if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
|
|
o->hookstate = 1;
|
|
oss_new(o, mye, myc, AST_STATE_RINGING);
|
|
} else
|
|
ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
|
|
if (s)
|
|
ast_free(s);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
char *s;
|
|
|
|
if (cmd == CLI_INIT) {
|
|
e->command = "console {mute|unmute}";
|
|
e->usage =
|
|
"Usage: console {mute|unmute}\n"
|
|
" Mute/unmute the microphone.\n";
|
|
return NULL;
|
|
} else if (cmd == CLI_GENERATE)
|
|
return NULL;
|
|
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
s = a->argv[e->args-1];
|
|
if (!strcasecmp(s, "mute"))
|
|
o->mute = 1;
|
|
else if (!strcasecmp(s, "unmute"))
|
|
o->mute = 0;
|
|
else
|
|
return CLI_SHOWUSAGE;
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_transfer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
struct ast_channel *b = NULL;
|
|
char *tmp, *ext, *ctx;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console transfer";
|
|
e->usage =
|
|
"Usage: console transfer <extension>[@context]\n"
|
|
" Transfers the currently connected call to the given extension (and\n"
|
|
" context if specified)\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc != 3)
|
|
return CLI_SHOWUSAGE;
|
|
if (o == NULL)
|
|
return CLI_FAILURE;
|
|
if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
|
|
ast_cli(a->fd, "There is no call to transfer\n");
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
tmp = ast_ext_ctx(a->argv[2], &ext, &ctx);
|
|
if (ctx == NULL) /* supply default context if needed */
|
|
ctx = o->owner->context;
|
|
if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
|
|
ast_cli(a->fd, "No such extension exists\n");
|
|
else {
|
|
ast_cli(a->fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx);
|
|
if (ast_async_goto(b, ctx, ext, 1))
|
|
ast_cli(a->fd, "Failed to transfer :(\n");
|
|
}
|
|
if (tmp)
|
|
ast_free(tmp);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *console_active(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console active";
|
|
e->usage =
|
|
"Usage: console active [device]\n"
|
|
" If used without a parameter, displays which device is the current\n"
|
|
" console. If a device is specified, the console sound device is changed to\n"
|
|
" the device specified.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc == 2)
|
|
ast_cli(a->fd, "active console is [%s]\n", oss_active);
|
|
else if (a->argc != 3)
|
|
return CLI_SHOWUSAGE;
|
|
else {
|
|
struct chan_oss_pvt *o;
|
|
if (strcmp(a->argv[2], "show") == 0) {
|
|
for (o = oss_default.next; o; o = o->next)
|
|
ast_cli(a->fd, "device [%s] exists\n", o->name);
|
|
return CLI_SUCCESS;
|
|
}
|
|
o = find_desc(a->argv[2]);
|
|
if (o == NULL)
|
|
ast_cli(a->fd, "No device [%s] exists\n", a->argv[2]);
|
|
else
|
|
oss_active = o->name;
|
|
}
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
/*!
|
|
* \brief store the boost factor
|
|
*/
|
|
static void store_boost(struct chan_oss_pvt *o, char *s)
|
|
{
|
|
double boost = 0;
|
|
if (sscanf(s, "%lf", &boost) != 1) {
|
|
ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
|
|
return;
|
|
}
|
|
if (boost < -BOOST_MAX) {
|
|
ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
|
|
boost = -BOOST_MAX;
|
|
} else if (boost > BOOST_MAX) {
|
|
ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
|
|
boost = BOOST_MAX;
|
|
}
|
|
boost = exp(log(10) * boost / 20) * BOOST_SCALE;
|
|
o->boost = boost;
|
|
ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
|
|
}
|
|
|
|
static char *console_boost(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct chan_oss_pvt *o = find_desc(oss_active);
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console boost";
|
|
e->usage =
|
|
"Usage: console boost [boost in dB]\n"
|
|
" Sets or display mic boost in dB\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc == 2)
|
|
ast_cli(a->fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
|
|
else if (a->argc == 3)
|
|
store_boost(o, a->argv[2]);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static struct ast_cli_entry cli_oss[] = {
|
|
AST_CLI(console_answer, "Answer an incoming console call"),
|
|
AST_CLI(console_hangup, "Hangup a call on the console"),
|
|
AST_CLI(console_flash, "Flash a call on the console"),
|
|
AST_CLI(console_dial, "Dial an extension on the console"),
|
|
AST_CLI(console_mute, "Disable/Enable mic input"),
|
|
AST_CLI(console_transfer, "Transfer a call to a different extension"),
|
|
AST_CLI(console_sendtext, "Send text to the remote device"),
|
|
AST_CLI(console_autoanswer, "Sets/displays autoanswer"),
|
|
AST_CLI(console_boost, "Sets/displays mic boost in dB"),
|
|
AST_CLI(console_active, "Sets/displays active console"),
|
|
};
|
|
|
|
/*!
|
|
* store the mixer argument from the config file, filtering possibly
|
|
* invalid or dangerous values (the string is used as argument for
|
|
* system("mixer %s")
|
|
*/
|
|
static void store_mixer(struct chan_oss_pvt *o, char *s)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < strlen(s); i++) {
|
|
if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) {
|
|
ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
|
|
return;
|
|
}
|
|
}
|
|
if (o->mixer_cmd)
|
|
ast_free(o->mixer_cmd);
|
|
o->mixer_cmd = ast_strdup(s);
|
|
ast_log(LOG_WARNING, "setting mixer %s\n", s);
|
|
}
|
|
|
|
/*!
|
|
* store the callerid components
|
|
*/
|
|
static void store_callerid(struct chan_oss_pvt *o, char *s)
|
|
{
|
|
ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
|
|
}
|
|
|
|
/*!
|
|
* grab fields from the config file, init the descriptor and open the device.
|
|
*/
|
|
static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
|
|
{
|
|
struct ast_variable *v;
|
|
struct chan_oss_pvt *o;
|
|
|
|
if (ctg == NULL) {
|
|
o = &oss_default;
|
|
ctg = "general";
|
|
} else {
|
|
if (!(o = ast_calloc(1, sizeof(*o))))
|
|
return NULL;
|
|
*o = oss_default;
|
|
/* "general" is also the default thing */
|
|
if (strcmp(ctg, "general") == 0) {
|
|
o->name = ast_strdup("dsp");
|
|
oss_active = o->name;
|
|
goto openit;
|
|
}
|
|
o->name = ast_strdup(ctg);
|
|
}
|
|
|
|
strcpy(o->mohinterpret, "default");
|
|
|
|
o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */
|
|
/* fill other fields from configuration */
|
|
for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
|
|
M_START(v->name, v->value);
|
|
|
|
/* handle jb conf */
|
|
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
|
|
continue;
|
|
|
|
M_BOOL("autoanswer", o->autoanswer)
|
|
M_BOOL("autohangup", o->autohangup)
|
|
M_BOOL("overridecontext", o->overridecontext)
|
|
M_STR("device", o->device)
|
|
M_UINT("frags", o->frags)
|
|
M_UINT("debug", oss_debug)
|
|
M_UINT("queuesize", o->queuesize)
|
|
M_STR("context", o->ctx)
|
|
M_STR("language", o->language)
|
|
M_STR("mohinterpret", o->mohinterpret)
|
|
M_STR("extension", o->ext)
|
|
M_F("mixer", store_mixer(o, v->value))
|
|
M_F("callerid", store_callerid(o, v->value))
|
|
M_F("boost", store_boost(o, v->value))
|
|
|
|
M_END(/* */);
|
|
}
|
|
if (ast_strlen_zero(o->device))
|
|
ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
|
|
if (o->mixer_cmd) {
|
|
char *cmd;
|
|
|
|
asprintf(&cmd, "mixer %s", o->mixer_cmd);
|
|
ast_log(LOG_WARNING, "running [%s]\n", cmd);
|
|
system(cmd);
|
|
ast_free(cmd);
|
|
}
|
|
if (o == &oss_default) /* we are done with the default */
|
|
return NULL;
|
|
|
|
openit:
|
|
#ifdef TRYOPEN
|
|
if (setformat(o, O_RDWR) < 0) { /* open device */
|
|
ast_verb(1, "Device %s not detected\n", ctg);
|
|
ast_verb(1, "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
|
|
goto error;
|
|
}
|
|
if (o->duplex != M_FULL)
|
|
ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
|
|
#endif /* TRYOPEN */
|
|
if (pipe(o->sndcmd) != 0) {
|
|
ast_log(LOG_ERROR, "Unable to create pipe\n");
|
|
goto error;
|
|
}
|
|
ast_pthread_create_background(&o->sthread, NULL, sound_thread, o);
|
|
/* link into list of devices */
|
|
if (o != &oss_default) {
|
|
o->next = oss_default.next;
|
|
oss_default.next = o;
|
|
}
|
|
return o;
|
|
|
|
error:
|
|
if (o != &oss_default)
|
|
ast_free(o);
|
|
return NULL;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
struct ast_config *cfg = NULL;
|
|
char *ctg = NULL;
|
|
struct ast_flags config_flags = { 0 };
|
|
|
|
/* Copy the default jb config over global_jbconf */
|
|
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
|
|
|
|
/* load config file */
|
|
if (!(cfg = ast_config_load(config, config_flags))) {
|
|
ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
do {
|
|
store_config(cfg, ctg);
|
|
} while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
|
|
|
|
ast_config_destroy(cfg);
|
|
|
|
if (find_desc(oss_active) == NULL) {
|
|
ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
|
|
/* XXX we could default to 'dsp' perhaps ? */
|
|
/* XXX should cleanup allocated memory etc. */
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
if (ast_channel_register(&oss_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
ast_cli_register_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
|
|
static int unload_module(void)
|
|
{
|
|
struct chan_oss_pvt *o;
|
|
|
|
ast_channel_unregister(&oss_tech);
|
|
ast_cli_unregister_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));
|
|
|
|
for (o = oss_default.next; o; o = o->next) {
|
|
close(o->sounddev);
|
|
if (o->sndcmd[0] > 0) {
|
|
close(o->sndcmd[0]);
|
|
close(o->sndcmd[1]);
|
|
}
|
|
if (o->owner)
|
|
ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
|
|
if (o->owner) /* XXX how ??? */
|
|
return -1;
|
|
/* XXX what about the thread ? */
|
|
/* XXX what about the memory allocated ? */
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");
|