chan_oss.c

Asterisk中信道部分的源码 。。。。

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <[e-mail]>
 *
 * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
 * note-this code best seen with ts=8 (8-spaces tabs) in the editor
 *
 * 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 Channel driver for OSS sound cards
 *
 * \author Mark Spencer <[e-mail]>
 * \author Luigi Rizzo
 *
 * \par See also
 * \arg \ref Config_oss
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>ossaudio</depend>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 108796 $")

#include <stdio.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <sys/time.h>
#include <stdlib.h>
#include <errno.h>

#ifdef __linux
#include <linux/soundcard.h>
#elif defined(__FreeBSD__)
#include <sys/soundcard.h>
#else
#include <soundcard.h>
#endif

#include "asterisk/lock.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/callerid.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/cli.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/endian.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"

/* ringtones we use */
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"

/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
	.flags = 0,
	.max_size = -1,
	.resync_threshold = -1,
	.impl = "",
};
static struct ast_jb_conf global_jbconf;

/*
 * Basic mode of operation:
 *
 * we have one keyboard (which receives commands from the keyboard)
 * and multiple headset's connected to audio cards.
 * Cards/Headsets are named as the sections of oss.conf.
 * The section called [general] contains the default parameters.
 *
 * At any time, the keyboard is attached to one card, and you
 * can switch among them using the command 'console foo'
 * where 'foo' is the name of the card you want.
 *
 * oss.conf parameters are
START_CONFIG

[general]
    ; General config options, with default values shown.
    ; You should use one section per device, with [general] being used
    ; for the first device and also as a template for other devices.
    ;
    ; All but 'debug' can go also in the device-specific sections.
    ;
    ; debug = 0x0		; misc debug flags, default is 0

    ; Set the device to use for I/O
    ; device = /dev/dsp

    ; Optional mixer command to run upon startup (e.g. to set
    ; volume levels, mutes, etc.
    ; mixer =

    ; Software mic volume booster (or attenuator), useful for sound
    ; cards or microphones with poor sensitivity. The volume level
    ; is in dB, ranging from -20.0 to +20.0
    ; boost = n			; mic volume boost in dB

    ; Set the callerid for outgoing calls
    ; callerid = John Doe <555-1234>

    ; autoanswer = no		; no autoanswer on call
    ; autohangup = yes		; hangup when other party closes
    ; extension = s		; default extension to call
    ; context = default		; default context for outgoing calls
    ; language = ""		; default language

    ; Default Music on Hold class to use when this channel is placed on hold in
    ; the case that the music class is not set on the channel with
    ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
    ; putting this one on hold did not suggest a class to use.
    ;
    ; mohinterpret=default

    ; If you set overridecontext to 'yes', then the whole dial string
    ; will be interpreted as an extension, which is extremely useful
    ; to dial SIP, IAX and other extensions which use the '@' character.
    ; The default is 'no' just for backward compatibility, but the
    ; suggestion is to change it.
    ; overridecontext = no	; if 'no', the last @ will start the context
				; if 'yes' the whole string is an extension.

    ; low level device parameters in case you have problems with the
    ; device driver on your operating system. You should not touch these
    ; unless you know what you are doing.
    ; queuesize = 10		; frames in device driver
    ; frags = 8			; argument to SETFRAGMENT

    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
                                  ; OSS channel. Defaults to "no". An enabled jitterbuffer will
                                  ; be used only if the sending side can create and the receiving
                                  ; side can not accept jitter. The OSS channel can't accept jitter,
                                  ; thus an enabled jitterbuffer on the receive OSS side will always
                                  ; be used if the sending side can create jitter.

    ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

    ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                                  ; resynchronized. Useful to improve the quality of the voice, with
                                  ; big jumps in/broken timestamps, usualy sent from exotic devices
                                  ; and programs. Defaults to 1000.

    ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of an OSS
                                  ; channel. Two implementations are currenlty available - "fixed"
                                  ; (with size always equals to jbmax-size) and "adaptive" (with
                                  ; variable size, actually the new jb of IAX2). Defaults to fixed.

    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
    ;-----------------------------------------------------------------------------------

[card1]
    ; device = /dev/dsp1	; alternate device

END_CONFIG

.. and so on for the other cards.

 */

/*
 * Helper macros to parse config arguments. They will go in a common
 * header file if their usage is globally accepted. In the meantime,
 * we define them here. Typical usage is as below.
 * Remember to open a block right before M_START (as it declares
 * some variables) and use the M_* macros WITHOUT A SEMICOLON:
 *
 *	{
 *		M_START(v->name, v->value) 
 *
 *		M_BOOL("dothis", x->flag1)
 *		M_STR("name", x->somestring)
 *		M_F("bar", some_c_code)
 *		M_END(some_final_statement)
 *		... other code in the block
 *	}
 *
 * XXX NOTE these macros should NOT be replicated in other parts of asterisk. 
 * Likely we will come up with a better way of doing config file parsing.
 */
#define M_START(var, val) \
        char *__s = var; char *__val = val;
#define M_END(x)   x;
#define M_F(tag, f)			if (!strcasecmp((__s), tag)) { f; } else
#define M_BOOL(tag, dst)	M_F(tag, (dst) = ast_true(__val) )
#define M_UINT(tag, dst)	M_F(tag, (dst) = strtoul(__val, NULL, 0) )
#define M_STR(tag, dst)		M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))

/*
 * The following parameters are used in the driver:
 *
 *  FRAME_SIZE	the size of an audio frame, in samples.
 *		160 is used almost universally, so you should not change it.
 *
 *  FRAGS	the argument for the SETFRAGMENT ioctl.
 *		Overridden by the 'frags' parameter in oss.conf
 *
 *		Bits 0-7 are the base-2 log of the device's block size,
 *		bits 16-31 are the number of blocks in the driver's queue.
 *		There are a lot of differences in the way this parameter
 *		is supported by different drivers, so you may need to
 *		experiment a bit with the value.
 *		A good default for linux is 30 blocks of 64 bytes, which
 *		results in 6 frames of 320 bytes (160 samples).
 *		FreeBSD works decently with blocks of 256 or 512 bytes,
 *		leaving the number unspecified.
 *		Note that this only refers to the device buffer size,
 *		this module will then try to keep the lenght of audio
 *		buffered within small constraints.
 *
 *  QUEUE_SIZE	The max number of blocks actually allowed in the device
 *		driver's buffer, irrespective of the available number.
 *		Overridden by the 'queuesize' parameter in oss.conf
 *
 *		Should be >=2, and at most as large as the hw queue above
 *		(otherwise it will never be full).
 */

#define FRAME_SIZE	160
#define	QUEUE_SIZE	10

#if defined(__FreeBSD__)
#define	FRAGS	0x8
#else
#define	FRAGS	( ( (6 * 5) << 16 ) | 0x6 )
#endif

/*
 * XXX text message sizes are probably 256 chars, but i am
 * not sure if there is a suitable definition anywhere.
 */
#define TEXT_SIZE	256

#if 0
#define	TRYOPEN	1				/* try to open on startup */
#endif
#define	O_CLOSE	0x444			/* special 'close' mode for device */
/* Which device to use */
#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
#define DEV_DSP "/dev/audio"
#else
#define DEV_DSP "/dev/dsp"
#endif

#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif

static char *config = "oss.conf";	/* default config file */

static int oss_debug;

/*
 * Each sound is made of 'datalen' samples of sound, repeated as needed to
 * generate 'samplen' samples of data, then followed by 'silencelen' samples
 * of silence. The loop is repeated if 'repeat' is set.
 */
struct sound {
	int ind;
	char *desc;
	short *data;
	int datalen;
	int samplen;
	int silencelen;
	int repeat;
};

static struct sound sounds[] = {
	{ AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
	{ AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
	{ AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
	{ AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
	{ AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
	{ -1, NULL, 0, 0, 0, 0 },	/* end marker */
};


/*
 * descriptor for one of our channels.
 * There is one used for 'default' values (from the [general] entry in
 * the configuration file), and then one instance for each device
 * (the default is cloned from [general], others are only created
 * if the relevant section exists).
 */
struct chan_oss_pvt {
	struct chan_oss_pvt *next;

	char *name;
	/*
	 * cursound indicates which in struct sound we play. -1 means nothing,
	 * any other value is a valid sound, in which case sampsent indicates
	 * the next sample to send in [0..samplen + silencelen]
	 * nosound is set to disable the audio data from the channel
	 * (so we can play the tones etc.).
	 */
	int sndcmd[2];				/* Sound command pipe */
	int cursound;				/* index of sound to send */
	int sampsent;				/* # of sound samples sent  */
	int nosound;				/* set to block audio from the PBX */

	int total_blocks;			/* total blocks in the output device */
	int sounddev;
	enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
	int autoanswer;
	int autohangup;
	int hookstate;
	char *mixer_cmd;			/* initial command to issue to the mixer */
	unsigned int queuesize;		/* max fragments in queue */
	unsigned int frags;			/* parameter for SETFRAGMENT */

	int warned;					/* various flags used for warnings */
#define WARN_used_blocks	1
#define WARN_speed		2
#define WARN_frag		4
	int w_errors;				/* overfull in the write path */
	struct timeval lastopen;

	int overridecontext;
	int mute;

	/* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
	 * be representable in 16 bits to avoid overflows.
	 */
#define	BOOST_SCALE	(1<<9)
#define	BOOST_MAX	40			/* slightly less than 7 bits */
	int boost;					/* input boost, scaled by BOOST_SCALE */
	char device[64];			/* device to open */

	pthread_t sthread;

	struct ast_channel *owner;
	char ext[AST_MAX_EXTENSION];
	char ctx[AST_MAX_CONTEXT];
	char language[MAX_LANGUAGE];
	char cid_name[256];			/*XXX */
	char cid_num[256];			/*XXX */
	char mohinterpret[MAX_MUSICCLASS];

	/* buffers used in oss_write */
	char oss_write_buf[FRAME_SIZE * 2];
	int oss_write_dst;
	/* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
	 * plus enough room for a full frame
	 */
	char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
	int readpos;				/* read position above */
	struct ast_frame read_f;	/* returned by oss_read */
};

static struct chan_oss_pvt oss_default = {
	.cursound = -1,
	.sounddev = -1,
	.duplex = M_UNSET,			/* XXX check this */
	.autoanswer = 1,
	.autohangup = 1,
	.queuesize = QUEUE_SIZE,
	.frags = FRAGS,
	.ext = "s",
	.ctx = "default",
	.readpos = AST_FRIENDLY_OFFSET,	/* start here on reads */
	.lastopen = { 0, 0 },
	.boost = BOOST_SCALE,
};

static char *oss_active;	 /* the active device */

static int setformat(struct chan_oss_pvt *o, int mode);

static struct ast_channel *oss_request(const char *type, int format, void *data
, int *cause);
static int oss_digit_begin(struct ast_channel *c, char digit);
static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
static int oss_text(struct ast_channel *c, const char *text);
static int oss_hangup(struct ast_channel *c);
static int oss_answer(struct ast_channel *c);
static struct ast_frame *oss_read(struct ast_channel *chan);
static int oss_call(struct ast_channel *c, char *dest, int timeout);
static int oss_write(struct ast_channel *chan, struct ast_frame *f);
static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static char tdesc[] = "OSS Console Channel Driver";

static const struct ast_channel_tech oss_tech = {
	.type = "Console",
	.description = tdesc,
	.capabilities = AST_FORMAT_SLINEAR,
	.requester = oss_request,
	.send_digit_begin = oss_digit_begin,
	.send_digit_end = oss_digit_end,
	.send_text = oss_text,
	.hangup = oss_hangup,
	.answer = oss_answer,
	.read = oss_read,
	.call = oss_call,
	.write = oss_write,
	.indicate = oss_indicate,
	.fixup = oss_fixup,
};

/*
 * returns a pointer to the descriptor with the given name
 */
static struct chan_oss_pvt *find_desc(char *dev)
{
	struct chan_oss_pvt *o = NULL;

	if (!dev)
		ast_log(LOG_WARNING, "null dev\n");

	for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);

	if (!o)
		ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");

	return o;
}

/*
 * split a string in extension-context, returns pointers to malloc'ed
 * strings.
 * If we do not have 'overridecontext' then the last @ is considered as
 * a context separator, and the context is overridden.
 * This is usually not very necessary as you can play with the dialplan,
 * and it is nice not to need it because you have '@' in SIP addresses.
 * Return value is the buffer address.
 */
static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
{
	struct chan_oss_pvt *o = find_desc(oss_active);

	if (ext == NULL || ctx == NULL)
		return NULL;			/* error */

	*ext = *ctx = NULL;

	if (src && *src != '\0')
		*ext = ast_strdup(src);