chan_alsa.c

Asterisk中信道部分的源码 。。。。

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * By Matthew Fredrickson <[e-mail]>
 *
 * 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 ALSA sound card channel driver 
 *
 * \author Matthew Fredrickson <[e-mail]>
 *
 * \par See also
 * \arg Config_alsa
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>asound</depend>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 118953 $")

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

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>

#include "asterisk/frame.h"
#include "asterisk/logger.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"

#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"

#ifdef ALSA_MONITOR
#include "alsa-monitor.h"
#endif

/*! 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;

#define DEBUG 0
/* Which device to use */
#define ALSA_INDEV "default"
#define ALSA_OUTDEV "default"
#define DESIRED_RATE 8000

/* Lets use 160 sample frames, just like GSM.  */
#define FRAME_SIZE 160
#define PERIOD_FRAMES 80		/* 80 Frames, at 2 bytes each */

/* When you set the frame size, you have to come up with
   the right buffer format as well. */
/* 5 64-byte frames = one frame */
#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);

/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600

#if __BYTE_ORDER == __LITTLE_ENDIAN
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
#else
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
#endif

static char indevname[50] = ALSA_INDEV;
static char outdevname[50] = ALSA_OUTDEV;

#if 0
static struct timeval lasttime;
#endif

static int silencesuppression = 0;
static int silencethreshold = 1000;

AST_MUTEX_DEFINE_STATIC(alsalock);

static const char tdesc[] = "ALSA Console Channel Driver";
static const char config[] = "alsa.conf";

static char context[AST_MAX_CONTEXT] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";
static char mohinterpret[MAX_MUSICCLASS];

static int hookstate = 0;

static short silence[FRAME_SIZE] = { 0, };

struct sound {
	int ind;
	short *data;
	int datalen;
	int samplen;
	int silencelen;
	int repeat;
};

static struct sound sounds[] = {
	{AST_CONTROL_RINGING, ringtone, sizeof(ringtone) / 2, 16000, 32000, 1},
	{AST_CONTROL_BUSY, busy, sizeof(busy) / 2, 4000, 4000, 1},
	{AST_CONTROL_CONGESTION, busy, sizeof(busy) / 2, 2000, 2000, 1},
	{AST_CONTROL_RING, ring10, sizeof(ring10) / 2, 16000, 32000, 1},
	{AST_CONTROL_ANSWER, answer, sizeof(answer) / 2, 2200, 0, 0},
};

/* Sound command pipe */
static int sndcmd[2];

static struct chan_alsa_pvt {
	/* We only have one ALSA structure -- near sighted perhaps, but it
	   keeps this driver as simple as possible -- as it should be. */
	struct ast_channel *owner;
	char exten[AST_MAX_EXTENSION];
	char context[AST_MAX_CONTEXT];
#if 0
	snd_pcm_t *card;
#endif
	snd_pcm_t *icard, *ocard;

} alsa;

/* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
   with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
   usually plenty. */

pthread_t sthread;

#define MAX_BUFFER_SIZE 100

/* File descriptors for sound device */
static int readdev = -1;
static int writedev = -1;

static int autoanswer = 1;

static int cursound = -1;
static int sampsent = 0;
static int silencelen = 0;
static int offset = 0;
static int nosound = 0;

/* ZZ */
static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause);
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
static int alsa_text(struct ast_channel *c, const char *text);
static int alsa_hangup(struct ast_channel *c);
static int alsa_answer(struct ast_channel *c);
static struct ast_frame *alsa_read(struct ast_channel *chan);
static int alsa_call(struct ast_channel *c, char *dest, int timeout);
static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);

static const struct ast_channel_tech alsa_tech = {
	.type = "Console",
	.description = tdesc,
	.capabilities = AST_FORMAT_SLINEAR,
	.requester = alsa_request,
	.send_digit_end = alsa_digit,
	.send_text = alsa_text,
	.hangup = alsa_hangup,
	.answer = alsa_answer,
	.read = alsa_read,
	.call = alsa_call,
	.write = alsa_write,
	.indicate = alsa_indicate,
	.fixup = alsa_fixup,
};

static int send_sound(void)
{
	short myframe[FRAME_SIZE];
	int total = FRAME_SIZE;
	short *frame = NULL;
	int amt = 0, res, myoff;
	snd_pcm_state_t state;

	if (cursound == -1)
		return 0;
	
	res = total;
	if (sampsent < sounds[cursound].samplen) {
		myoff = 0;
		while (total) {
			amt = total;
			if (amt > (sounds[cursound].datalen - offset))
				amt = sounds[cursound].datalen - offset;
			memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
			total -= amt;
			offset += amt;
			sampsent += amt;
			myoff += amt;
			if (offset >= sounds[cursound].datalen)
				offset = 0;
		}
		/* Set it up for silence */
		if (sampsent >= sounds[cursound].samplen)
			silencelen = sounds[cursound].silencelen;
		frame = myframe;
	} else {
		if (silencelen > 0) {
			frame = silence;
			silencelen -= res;
		} else {
			if (sounds[cursound].repeat) {
				/* Start over */
				sampsent = 0;
				offset = 0;
			} else {
				cursound = -1;
				nosound = 0;
			}
			return 0;
		}
	}
	
	if (res == 0 || !frame)
		return 0;

#ifdef ALSA_MONITOR
	alsa_monitor_write((char *) frame, res * 2);
#endif
	state = snd_pcm_state(alsa.ocard);
	if (state == SND_PCM_STATE_XRUN)
		snd_pcm_prepare(alsa.ocard);
	res = snd_pcm_writei(alsa.ocard, frame, res);
	if (res > 0)
		return 0;
	return 0;
}

static void *sound_thread(void *unused)
{
	fd_set rfds;
	fd_set wfds;
	int max, res;

	for (;;) {
		FD_ZERO(&rfds);
		FD_ZERO(&wfds);
		max = sndcmd[0];
		FD_SET(sndcmd[0], &rfds);
		if (cursound > -1) {
			FD_SET(writedev, &wfds);
			if (writedev > max)
				max = writedev;
		}
#ifdef ALSA_MONITOR
		if (!alsa.owner) {
			FD_SET(readdev, &rfds);
			if (readdev > max)
				max = readdev;
		}
#endif
		res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
		if (res < 1) {
			ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
			continue;
		}
#ifdef ALSA_MONITOR
		if (FD_ISSET(readdev, &rfds)) {
			/* Keep the pipe going with read audio */
			snd_pcm_state_t state;
			short buf[FRAME_SIZE];
			int r;

			state = snd_pcm_state(alsa.ocard);
			if (state == SND_PCM_STATE_XRUN) {
				snd_pcm_prepare(alsa.ocard);
			}
			r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE);
			if (r == -EPIPE) {
#if DEBUG
				ast_log(LOG_ERROR, "XRUN read\n");
#endif
				snd_pcm_prepare(alsa.icard);
			} else if (r == -ESTRPIPE) {
				ast_log(LOG_ERROR, "-ESTRPIPE\n");
				snd_pcm_prepare(alsa.icard);
			} else if (r < 0) {
				ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
			} else
				alsa_monitor_read((char *) buf, r * 2);
		}
#endif
		if (FD_ISSET(sndcmd[0], &rfds)) {
			read(sndcmd[0], &cursound, sizeof(cursound));
			silencelen = 0;
			offset = 0;
			sampsent = 0;
		}
		if (FD_ISSET(writedev, &wfds))
			if (send_sound())
				ast_log(LOG_WARNING, "Failed to write sound\n");
	}
	/* Never reached */
	return NULL;
}

static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
{
	int err;
	int direction;
	snd_pcm_t *handle = NULL;
	snd_pcm_hw_params_t *hwparams = NULL;
	snd_pcm_sw_params_t *swparams = NULL;
	struct pollfd pfd;
	snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
	/* int period_bytes = 0; */
	snd_pcm_uframes_t buffer_size = 0;

	unsigned int rate = DESIRED_RATE;
#if 0
	unsigned int per_min = 1;
#endif
	/* unsigned int per_max = 8; */
	snd_pcm_uframes_t start_threshold, stop_threshold;

	err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
	if (err < 0) {
		ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
		return NULL;
	} else
		ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");

	hwparams = alloca(snd_pcm_hw_params_sizeof());
	memset(hwparams, 0, snd_pcm_hw_params_sizeof());
	snd_pcm_hw_params_any(handle, hwparams);

	err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (err < 0)
		ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));

	err = snd_pcm_hw_params_set_format(handle, hwparams, format);
	if (err < 0)
		ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));

	err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
	if (err < 0)
		ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));

	direction = 0;
	err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
	if (rate != DESIRED_RATE)
		ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);

	direction = 0;
	err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
	if (err < 0)
		ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
	else
		ast_log(LOG_DEBUG, "Period size is %d\n", err);

	buffer_size = 4096 * 2;		/* period_size * 16; */
	err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
	if (err < 0)
		ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
	else
		ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err);

#if 0
	direction = 0;
	err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction);
	if (err < 0)
		ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err));

	err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0);
	if (err < 0)
		ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err));
#endif

	err = snd_pcm_hw_params(handle, hwparams);
	if (err < 0)
		ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));

	swparams = alloca(snd_pcm_sw_params_sizeof());
	memset(swparams, 0, snd_pcm_sw_params_sizeof());
	snd_pcm_sw_params_current(handle, swparams);

#if 1
	if (stream == SND_PCM_STREAM_PLAYBACK)
		start_threshold = period_size;
	else
		start_threshold = 1;

	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
	if (err < 0)
		ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
#endif

#if 1
	if (stream == SND_PCM_STREAM_PLAYBACK)
		stop_threshold = buffer_size;
	else
		stop_threshold = buffer_size;

	err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
	if (err < 0)
		ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
#endif
#if 0
	err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES);
	if (err < 0)
		ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err));
#endif

#if 0
	err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold);
	if (err < 0)
		ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err));
#endif
	err = snd_pcm_sw_params(handle, swparams);
	if (err < 0)
		ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));

	err = snd_pcm_poll_descriptors_count(handle);