chan_alsa.c

00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2005, Digium, Inc.
00005  *
00006  * By Matthew Fredrickson <creslin@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*! \file 
00020  * \brief ALSA sound card channel driver 
00021  *
00022  * \author Matthew Fredrickson <creslin@digium.com>
00023  *
00024  * \par See also
00025  * \arg Config_alsa
00026  *
00027  * \ingroup channel_drivers
00028  */
00029 
00030 /*** MODULEINFO
00031    <depend>asound</depend>
00032  ***/
00033 
00034 #include "asterisk.h"
00035 
00036 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 51806 $")
00037 
00038 #include <unistd.h>
00039 #include <fcntl.h>
00040 #include <errno.h>
00041 #include <sys/ioctl.h>
00042 #include <sys/time.h>
00043 #include <string.h>
00044 #include <stdlib.h>
00045 #include <stdio.h>
00046 
00047 #define ALSA_PCM_NEW_HW_PARAMS_API
00048 #define ALSA_PCM_NEW_SW_PARAMS_API
00049 #include <alsa/asoundlib.h>
00050 
00051 #include "asterisk/frame.h"
00052 #include "asterisk/logger.h"
00053 #include "asterisk/channel.h"
00054 #include "asterisk/module.h"
00055 #include "asterisk/options.h"
00056 #include "asterisk/pbx.h"
00057 #include "asterisk/config.h"
00058 #include "asterisk/cli.h"
00059 #include "asterisk/utils.h"
00060 #include "asterisk/causes.h"
00061 #include "asterisk/endian.h"
00062 #include "asterisk/stringfields.h"
00063 #include "asterisk/abstract_jb.h"
00064 #include "asterisk/musiconhold.h"
00065 
00066 #include "busy.h"
00067 #include "ringtone.h"
00068 #include "ring10.h"
00069 #include "answer.h"
00070 
00071 #ifdef ALSA_MONITOR
00072 #include "alsa-monitor.h"
00073 #endif
00074 
00075 /*! Global jitterbuffer configuration - by default, jb is disabled */
00076 static struct ast_jb_conf default_jbconf = {
00077    .flags = 0,
00078    .max_size = -1,
00079    .resync_threshold = -1,
00080    .impl = ""
00081 };
00082 static struct ast_jb_conf global_jbconf;
00083 
00084 #define DEBUG 0
00085 /* Which device to use */
00086 #define ALSA_INDEV "hw:0,0"
00087 #define ALSA_OUTDEV "hw:0,0"
00088 #define DESIRED_RATE 8000
00089 
00090 /* Lets use 160 sample frames, just like GSM.  */
00091 #define FRAME_SIZE 160
00092 #define PERIOD_FRAMES 80      /* 80 Frames, at 2 bytes each */
00093 
00094 /* When you set the frame size, you have to come up with
00095    the right buffer format as well. */
00096 /* 5 64-byte frames = one frame */
00097 #define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
00098 
00099 /* Don't switch between read/write modes faster than every 300 ms */
00100 #define MIN_SWITCH_TIME 600
00101 
00102 #if __BYTE_ORDER == __LITTLE_ENDIAN
00103 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
00104 #else
00105 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
00106 #endif
00107 
00108 /* static int block = O_NONBLOCK; */
00109 static char indevname[50] = ALSA_INDEV;
00110 static char outdevname[50] = ALSA_OUTDEV;
00111 
00112 #if 0
00113 static struct timeval lasttime;
00114 #endif
00115 
00116 static int silencesuppression = 0;
00117 static int silencethreshold = 1000;
00118 
00119 AST_MUTEX_DEFINE_STATIC(alsalock);
00120 
00121 static const char tdesc[] = "ALSA Console Channel Driver";
00122 static const char config[] = "alsa.conf";
00123 
00124 static char context[AST_MAX_CONTEXT] = "default";
00125 static char language[MAX_LANGUAGE] = "";
00126 static char exten[AST_MAX_EXTENSION] = "s";
00127 static char mohinterpret[MAX_MUSICCLASS];
00128 
00129 static int hookstate = 0;
00130 
00131 static short silence[FRAME_SIZE] = { 0, };
00132 
00133 struct sound {
00134    int ind;
00135    short *data;
00136    int datalen;
00137    int samplen;
00138    int silencelen;
00139    int repeat;
00140 };
00141 
00142 static struct sound sounds[] = {
00143    {AST_CONTROL_RINGING, ringtone, sizeof(ringtone) / 2, 16000, 32000, 1},
00144    {AST_CONTROL_BUSY, busy, sizeof(busy) / 2, 4000, 4000, 1},
00145    {AST_CONTROL_CONGESTION, busy, sizeof(busy) / 2, 2000, 2000, 1},
00146    {AST_CONTROL_RING, ring10, sizeof(ring10) / 2, 16000, 32000, 1},
00147    {AST_CONTROL_ANSWER, answer, sizeof(answer) / 2, 2200, 0, 0},
00148 };
00149 
00150 /* Sound command pipe */
00151 static int sndcmd[2];
00152 
00153 static struct chan_alsa_pvt {
00154    /* We only have one ALSA structure -- near sighted perhaps, but it
00155       keeps this driver as simple as possible -- as it should be. */
00156    struct ast_channel *owner;
00157    char exten[AST_MAX_EXTENSION];
00158    char context[AST_MAX_CONTEXT];
00159 #if 0
00160    snd_pcm_t *card;
00161 #endif
00162    snd_pcm_t *icard, *ocard;
00163 
00164 } alsa;
00165 
00166 /* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
00167    with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
00168    usually plenty. */
00169 
00170 pthread_t sthread;
00171 
00172 #define MAX_BUFFER_SIZE 100
00173 
00174 /* File descriptors for sound device */
00175 static int readdev = -1;
00176 static int writedev = -1;
00177 
00178 static int autoanswer = 1;
00179 
00180 static int cursound = -1;
00181 static int sampsent = 0;
00182 static int silencelen = 0;
00183 static int offset = 0;
00184 static int nosound = 0;
00185 
00186 /* ZZ */
00187 static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause);
00188 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
00189 static int alsa_text(struct ast_channel *c, const char *text);
00190 static int alsa_hangup(struct ast_channel *c);
00191 static int alsa_answer(struct ast_channel *c);
00192 static struct ast_frame *alsa_read(struct ast_channel *chan);
00193 static int alsa_call(struct ast_channel *c, char *dest, int timeout);
00194 static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
00195 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
00196 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
00197 
00198 static const struct ast_channel_tech alsa_tech = {
00199    .type = "Console",
00200    .description = tdesc,
00201    .capabilities = AST_FORMAT_SLINEAR,
00202    .requester = alsa_request,
00203    .send_digit_end = alsa_digit,
00204    .send_text = alsa_text,
00205    .hangup = alsa_hangup,
00206    .answer = alsa_answer,
00207    .read = alsa_read,
00208    .call = alsa_call,
00209    .write = alsa_write,
00210    .indicate = alsa_indicate,
00211    .fixup = alsa_fixup,
00212 };
00213 
00214 static int send_sound(void)
00215 {
00216    short myframe[FRAME_SIZE];
00217    int total = FRAME_SIZE;
00218    short *frame = NULL;
00219    int amt = 0, res, myoff;
00220    snd_pcm_state_t state;
00221 
00222    if (cursound == -1)
00223       return 0;
00224 
00225    res = total;
00226    if (sampsent < sounds[cursound].samplen) {
00227       myoff = 0;
00228       while (total) {
00229          amt = total;
00230          if (amt > (sounds[cursound].datalen - offset))
00231             amt = sounds[cursound].datalen - offset;
00232          memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
00233          total -= amt;
00234          offset += amt;
00235          sampsent += amt;
00236          myoff += amt;
00237          if (offset >= sounds[cursound].datalen)
00238             offset = 0;
00239       }
00240       /* Set it up for silence */
00241       if (sampsent >= sounds[cursound].samplen)
00242          silencelen = sounds[cursound].silencelen;
00243       frame = myframe;
00244    } else {
00245       if (silencelen > 0) {
00246          frame = silence;
00247          silencelen -= res;
00248       } else {
00249          if (sounds[cursound].repeat) {
00250             /* Start over */
00251             sampsent = 0;
00252             offset = 0;
00253          } else {
00254             cursound = -1;
00255             nosound = 0;
00256          }
00257          return 0;
00258       }
00259    }
00260 
00261    if (res == 0 || !frame)
00262       return 0;
00263 
00264 #ifdef ALSA_MONITOR
00265    alsa_monitor_write((char *) frame, res * 2);
00266 #endif
00267    state = snd_pcm_state(alsa.ocard);
00268    if (state == SND_PCM_STATE_XRUN)
00269       snd_pcm_prepare(alsa.ocard);
00270    res = snd_pcm_writei(alsa.ocard, frame, res);
00271    if (res > 0)
00272       return 0;
00273    return 0;
00274 }
00275 
00276 static void *sound_thread(void *unused)
00277 {
00278    fd_set rfds;
00279    fd_set wfds;
00280    int max, res;
00281 
00282    for (;;) {
00283       FD_ZERO(&rfds);
00284       FD_ZERO(&wfds);
00285       max = sndcmd[0];
00286       FD_SET(sndcmd[0], &rfds);
00287       if (cursound > -1) {
00288          FD_SET(writedev, &wfds);
00289          if (writedev > max)
00290             max = writedev;
00291       }
00292 #ifdef ALSA_MONITOR
00293       if (!alsa.owner) {
00294          FD_SET(readdev, &rfds);
00295          if (readdev > max)
00296             max = readdev;
00297       }
00298 #endif
00299       res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
00300       if (res < 1) {
00301          ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
00302          continue;
00303       }
00304 #ifdef ALSA_MONITOR
00305       if (FD_ISSET(readdev, &rfds)) {
00306          /* Keep the pipe going with read audio */
00307          snd_pcm_state_t state;
00308          short buf[FRAME_SIZE];
00309          int r;
00310 
00311          state = snd_pcm_state(alsa.ocard);
00312          if (state == SND_PCM_STATE_XRUN) {
00313             snd_pcm_prepare(alsa.ocard);
00314          }
00315          r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE);
00316          if (r == -EPIPE) {
00317 #if DEBUG
00318             ast_log(LOG_ERROR, "XRUN read\n");
00319 #endif
00320             snd_pcm_prepare(alsa.icard);
00321          } else if (r == -ESTRPIPE) {
00322             ast_log(LOG_ERROR, "-ESTRPIPE\n");
00323             snd_pcm_prepare(alsa.icard);
00324          } else if (r < 0) {
00325             ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
00326          } else
00327             alsa_monitor_read((char *) buf, r * 2);
00328       }
00329 #endif
00330       if (FD_ISSET(sndcmd[0], &rfds)) {
00331          read(sndcmd[0], &cursound, sizeof(cursound));
00332          silencelen = 0;
00333          offset = 0;
00334          sampsent = 0;
00335       }
00336       if (FD_ISSET(writedev, &wfds))
00337          if (send_sound())
00338             ast_log(LOG_WARNING, "Failed to write sound\n");
00339    }
00340    /* Never reached */
00341    return NULL;
00342 }
00343 
00344 static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
00345 {
00346    int err;
00347    int direction;
00348    snd_pcm_t *handle = NULL;
00349    snd_pcm_hw_params_t *hwparams = NULL;
00350    snd_pcm_sw_params_t *swparams = NULL;
00351    struct pollfd pfd;
00352    snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
00353    /* int period_bytes = 0; */
00354    snd_pcm_uframes_t buffer_size = 0;
00355 
00356    unsigned int rate = DESIRED_RATE;
00357 #if 0
00358    unsigned int per_min = 1;
00359 #endif
00360    /* unsigned int per_max = 8; */
00361    snd_pcm_uframes_t start_threshold, stop_threshold;
00362 
00363    err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK);
00364    if (err < 0) {
00365       ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
00366       return NULL;
00367    } else {
00368       if (option_debug)
00369          ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
00370    }
00371 
00372    snd_pcm_hw_params_alloca(&hwparams);
00373    snd_pcm_hw_params_any(handle, hwparams);
00374 
00375    err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
00376    if (err < 0)
00377       ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
00378 
00379    err = snd_pcm_hw_params_set_format(handle, hwparams, format);
00380    if (err < 0)
00381       ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
00382 
00383    err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
00384    if (err < 0)
00385       ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
00386 
00387    direction = 0;
00388    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
00389    if (rate != DESIRED_RATE)
00390       ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
00391 
00392    direction = 0;
00393    err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
00394    if (err < 0)
00395       ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
00396    else {
00397       if (option_debug)
00398          ast_log(LOG_DEBUG, "Period size is %d\n", err);
00399    }
00400 
00401    buffer_size = 4096 * 2;    /* period_size * 16; */
00402    err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
00403    if (err < 0)
00404       ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
00405    else {
00406       if (option_debug)
00407          ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err);
00408    }
00409 
00410 #if 0
00411    direction = 0;
00412    err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction);
00413    if (err < 0)
00414       ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err));
00415 
00416    err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0);
00417    if (err < 0)
00418       ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err));
00419 #endif
00420 
00421    err = snd_pcm_hw_params(handle, hwparams);
00422    if (err < 0)
00423       ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
00424 
00425    snd_pcm_sw_params_alloca(&swparams);
00426    snd_pcm_sw_params_current(handle, swparams);
00427 
00428 #if 1
00429    if (stream == SND_PCM_STREAM_PLAYBACK)
00430       start_threshold = period_size;
00431    else
00432       start_threshold = 1;
00433 
00434    err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
00435    if (err < 0)
00436       ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
00437 #endif
00438 
00439 #if 1
00440    if (stream == SND_PCM_STREAM_PLAYBACK)
00441       stop_threshold = buffer_size;
00442    else
00443       stop_threshold = buffer_size;
00444 
00445    err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
00446    if (err < 0)
00447       ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
00448 #endif
00449 #if 0
00450    err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES);
00451    if (err < 0)
00452       ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err));
00453 #endif
00454 
00455 #if 0
00456    err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold);
00457    if (err < 0)
00458       ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err));
00459 #endif
00460    err = snd_pcm_sw_params(handle, swparams);
00461    if (err < 0)
00462       ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
00463 
00464    err = snd_pcm_poll_descriptors_count(handle);
00465    if (err <= 0)
00466       ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
00467    if (err != 1) {
00468       if (option_debug)
00469          ast_log(LOG_DEBUG, "Can't handle more than one device\n");
00470    }
00471 
00472    snd_pcm_poll_descriptors(handle, &pfd, err);
00473    if (option_debug)
00474       ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd);
00475 
00476    if (stream == SND_PCM_STREAM_CAPTURE)
00477       readdev = pfd.fd;
00478    else
00479       writedev = pfd.fd;
00480 
00481    return handle;
00482 }
00483 
00484 static int soundcard_init(void)
00485 {
00486    alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
00487    alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
00488 
00489    if (!alsa.icard || !alsa.ocard) {
00490       ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n");
00491       return -1;
00492    }
00493 
00494    return readdev;
00495 }
00496 
00497 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
00498 {
00499    ast_mutex_lock(&alsalock);
00500    ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
00501       digit, duration);
00502    ast_mutex_unlock(&alsalock);
00503    return 0;
00504 }
00505 
00506 static int alsa_text(struct ast_channel *c, const char *text)
00507 {
00508    ast_mutex_lock(&alsalock);
00509    ast_verbose(" << Console Received text %s >> \n", text);
00510    ast_mutex_unlock(&alsalock);
00511    return 0;
00512 }
00513 
00514 static void grab_owner(void)
00515 {
00516    while (alsa.owner && ast_mutex_trylock(&alsa.owner->lock)) {
00517       ast_mutex_unlock(&alsalock);
00518       usleep(1);
00519       ast_mutex_lock(&alsalock);
00520    }
00521 }
00522 
00523 static int alsa_call(struct ast_channel *c, char *dest, int timeout)
00524 {
00525    int res = 3;
00526    struct ast_frame f = { AST_FRAME_CONTROL };
00527    ast_mutex_lock(&alsalock);
00528    ast_verbose(" << Call placed to '%s' on console >> \n", dest);
00529    if (autoanswer) {
00530       ast_verbose(" << Auto-answered >> \n");
00531       grab_owner();
00532       if (alsa.owner) {
00533          f.subclass = AST_CONTROL_ANSWER;
00534          ast_queue_frame(alsa.owner, &f);
00535          ast_mutex_unlock(&alsa.owner->lock);
00536       }
00537    } else {
00538       ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
00539       grab_owner();
00540       if (alsa.owner) {
00541          f.subclass = AST_CONTROL_RINGING;
00542          ast_queue_frame(alsa.owner, &f);
00543          ast_mutex_unlock(&alsa.owner->lock);
00544       }
00545       write(sndcmd[1], &res, sizeof(res));
00546    }
00547    snd_pcm_prepare(alsa.icard);
00548    snd_pcm_start(alsa.icard);
00549    ast_mutex_unlock(&alsalock);
00550    return 0;
00551 }
00552 
00553 static void answer_sound(void)
00554 {
00555    int res;
00556    nosound = 1;
00557    res = 4;
00558    write(sndcmd[1], &res, sizeof(res));
00559 
00560 }
00561 
00562 static int alsa_answer(struct ast_channel *c)
00563 {
00564    ast_mutex_lock(&alsalock);
00565    ast_verbose(" << Console call has been answered >> \n");
00566    answer_sound();
00567    ast_setstate(c, AST_STATE_UP);
00568    cursound = -1;
00569    snd_pcm_prepare(alsa.icard);
00570    snd_pcm_start(alsa.icard);
00571    ast_mutex_unlock(&alsalock);
00572    return 0;
00573 }
00574 
00575 static int alsa_hangup(struct ast_channel *c)
00576 {
00577    int res;
00578    ast_mutex_lock(&alsalock);
00579    cursound = -1;
00580    c->tech_pvt = NULL;
00581    alsa.owner = NULL;
00582    ast_verbose(" << Hangup on console >> \n");
00583    ast_module_unref(ast_module_info->self);
00584    if (hookstate) {
00585       hookstate = 0;
00586       if (!autoanswer) {
00587          /* Congestion noise */
00588          res = 2;
00589          write(sndcmd[1], &res, sizeof(res));
00590       }
00591    }
00592    snd_pcm_drop(alsa.icard);
00593    ast_mutex_unlock(&alsalock);
00594    return 0;
00595 }
00596 
00597 static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
00598 {
00599    static char sizbuf[8000];
00600    static int sizpos = 0;
00601    int len = sizpos;
00602    int pos;
00603    int res = 0;
00604    /* size_t frames = 0; */
00605    snd_pcm_state_t state;
00606 
00607    /* Immediately return if no sound is enabled */
00608    if (nosound)
00609       return 0;
00610 
00611    ast_mutex_lock(&alsalock);
00612    /* Stop any currently playing sound */
00613    if (cursound != -1) {
00614       snd_pcm_drop(alsa.ocard);
00615       snd_pcm_prepare(alsa.ocard);
00616       cursound = -1;
00617    }
00618 
00619 
00620    /* We have to digest the frame in 160-byte portions */
00621    if (f->datalen > sizeof(sizbuf) - sizpos) {
00622       ast_log(LOG_WARNING, "Frame too large\n");
00623       res = -1;
00624    } else {
00625       memcpy(sizbuf + sizpos, f->data, f->datalen);
00626       len += f->datalen;
00627       pos = 0;
00628 #ifdef ALSA_MONITOR
00629       alsa_monitor_write(sizbuf, len);
00630 #endif
00631       state = snd_pcm_state(alsa.ocard);
00632       if (state == SND_PCM_STATE_XRUN)
00633          snd_pcm_prepare(alsa.ocard);
00634       res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2);
00635       if (res == -EPIPE) {
00636 #if DEBUG
00637          if (option_debug)
00638             ast_log(LOG_DEBUG, "XRUN write\n");
00639 #endif
00640          snd_pcm_prepare(alsa.ocard);
00641          res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2);
00642          if (res != len / 2) {
00643             ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
00644             res = -1;
00645          } else if (res < 0) {
00646             ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
00647             res = -1;
00648          }
00649       } else {
00650          if (res == -ESTRPIPE)
00651             ast_log(LOG_ERROR, "You've got some big problems\n");
00652          else if (res < 0)
00653             ast_log(LOG_NOTICE, "Error %d on write\n", res);
00654       }
00655    }
00656    ast_mutex_unlock(&alsalock);
00657    if (res > 0)
00658       res = 0;
00659    return res;
00660 }
00661 
00662 
00663 static struct ast_frame *alsa_read(struct ast_channel *chan)
00664 {
00665    static struct ast_frame f;
00666    static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET / 2];
00667    short *buf;
00668    static int readpos = 0;
00669    static int left = FRAME_SIZE;
00670    snd_pcm_state_t state;
00671    int r = 0;
00672    int off = 0;
00673 
00674    ast_mutex_lock(&alsalock);
00675    /* Acknowledge any pending cmd */
00676    f.frametype = AST_FRAME_NULL;
00677    f.subclass = 0;
00678    f.samples = 0;
00679    f.datalen = 0;
00680    f.data = NULL;
00681    f.offset = 0;
00682    f.src = "Console";
00683    f.mallocd = 0;
00684    f.delivery.tv_sec = 0;
00685    f.delivery.tv_usec = 0;
00686 
00687    state = snd_pcm_state(alsa.icard);
00688    if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) {
00689       snd_pcm_prepare(alsa.icard);
00690    }
00691 
00692    buf = __buf + AST_FRIENDLY_OFFSET / 2;
00693 
00694    r = snd_pcm_readi(alsa.icard, buf + readpos, left);
00695    if (r == -EPIPE) {
00696 #if DEBUG
00697       ast_log(LOG_ERROR, "XRUN read\n");
00698 #endif
00699       snd_pcm_prepare(alsa.icard);
00700    } else if (r == -ESTRPIPE) {
00701       ast_log(LOG_ERROR, "-ESTRPIPE\n");
00702       snd_pcm_prepare(alsa.icard);
00703    } else if (r < 0) {
00704       ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
00705    } else if (r >= 0) {
00706       off -= r;
00707    }
00708    /* Update positions */
00709    readpos += r;
00710    left -= r;
00711 
00712    if (readpos >= FRAME_SIZE) {
00713       /* A real frame */
00714       readpos = 0;
00715       left = FRAME_SIZE;
00716       if (chan->_state != AST_STATE_UP) {
00717          /* Don't transmit unless it's up */
00718          ast_mutex_unlock(&alsalock);
00719          return &f;
00720       }
00721       f.frametype = AST_FRAME_VOICE;
00722       f.subclass = AST_FORMAT_SLINEAR;
00723       f.samples = FRAME_SIZE;
00724       f.datalen = FRAME_SIZE * 2;
00725       f.data = buf;
00726       f.offset = AST_FRIENDLY_OFFSET;
00727       f.src = "Console";
00728       f.mallocd = 0;
00729 #ifdef ALSA_MONITOR
00730       alsa_monitor_read((char *) buf, FRAME_SIZE * 2);
00731 #endif
00732 
00733    }
00734    ast_mutex_unlock(&alsalock);
00735    return &f;
00736 }
00737 
00738 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
00739 {
00740    struct chan_alsa_pvt *p = newchan->tech_pvt;
00741    ast_mutex_lock(&alsalock);
00742    p->owner = newchan;
00743    ast_mutex_unlock(&alsalock);
00744    return 0;
00745 }
00746 
00747 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
00748 {
00749    int res = 0;
00750 
00751    ast_mutex_lock(&alsalock);
00752 
00753    switch (cond) {
00754       case AST_CONTROL_BUSY:
00755          res = 1;
00756          break;
00757       case AST_CONTROL_CONGESTION:
00758          res = 2;
00759          break;
00760       case AST_CONTROL_RINGING:
00761          break;
00762       case -1:
00763          res = -1;
00764          break;
00765       case AST_CONTROL_VIDUPDATE:
00766          res = -1;
00767          break;
00768       case AST_CONTROL_HOLD:
00769          ast_verbose(" << Console Has Been Placed on Hold >> \n");
00770          ast_moh_start(chan, data, mohinterpret);
00771          break;
00772       case AST_CONTROL_UNHOLD:
00773          ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
00774          ast_moh_stop(chan);
00775          break;
00776       default:
00777          ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
00778          res = -1;
00779    }
00780 
00781    if (res > -1)
00782       write(sndcmd[1], &res, sizeof(res));
00783 
00784    ast_mutex_unlock(&alsalock);
00785 
00786    return res;
00787 }
00788 
00789 static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state)
00790 {
00791    struct ast_channel *tmp = NULL;
00792 
00793    if (!(tmp = ast_channel_alloc(1, state, 0, 0, "ALSA/%s", indevname)))
00794       return NULL;
00795 
00796    tmp->tech = &alsa_tech;
00797    tmp->fds[0] = readdev;
00798    tmp->nativeformats = AST_FORMAT_SLINEAR;
00799    tmp->readformat = AST_FORMAT_SLINEAR;
00800    tmp->writeformat = AST_FORMAT_SLINEAR;
00801    tmp->tech_pvt = p;
00802    if (!ast_strlen_zero(p->context))
00803       ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
00804    if (!ast_strlen_zero(p->exten))
00805       ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
00806    if (!ast_strlen_zero(language))
00807       ast_string_field_set(tmp, language, language);
00808    p->owner = tmp;
00809    ast_module_ref(ast_module_info->self);
00810    ast_jb_configure(tmp, &global_jbconf);
00811    if (state != AST_STATE_DOWN) {
00812       if (ast_pbx_start(tmp)) {
00813          ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
00814          ast_hangup(tmp);
00815          tmp = NULL;
00816       }
00817    }
00818 
00819    return tmp;
00820 }
00821 
00822 static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause)
00823 {
00824    int oldformat = format;
00825    struct ast_channel *tmp = NULL;
00826 
00827    format &= AST_FORMAT_SLINEAR;
00828    if (!format) {
00829       ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
00830       return NULL;
00831    }
00832 
00833    ast_mutex_lock(&alsalock);
00834 
00835    if (alsa.owner) {
00836       ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
00837       *cause = AST_CAUSE_BUSY;
00838    } else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN)))
00839       ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
00840 
00841    ast_mutex_unlock(&alsalock);
00842 
00843    return tmp;
00844 }
00845 
00846 static int console_autoanswer(int fd, int argc, char *argv[])
00847 {
00848    int res = RESULT_SUCCESS;;
00849    if ((argc != 2) && (argc != 3))
00850       return RESULT_SHOWUSAGE;
00851    ast_mutex_lock(&alsalock);
00852    if (argc == 2) {
00853       ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
00854    } else {
00855       if (!strcasecmp(argv[2], "on"))
00856          autoanswer = -1;
00857       else if (!strcasecmp(argv[2], "off"))
00858          autoanswer = 0;
00859       else
00860          res = RESULT_SHOWUSAGE;
00861    }
00862    ast_mutex_unlock(&alsalock);
00863    return res;
00864 }
00865 
00866 static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
00867 {
00868 #ifndef MIN
00869 #define MIN(a,b) ((a) < (b) ? (a) : (b))
00870 #endif
00871    switch (state) {
00872       case 0:
00873          if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
00874             return ast_strdup("on");
00875       case 1:
00876          if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
00877             return ast_strdup("off");
00878       default:
00879          return NULL;
00880    }
00881    return NULL;
00882 }
00883 
00884 static const char autoanswer_usage[] =
00885    "Usage: console autoanswer [on|off]\n"
00886    "       Enables or disables autoanswer feature.  If used without\n"
00887    "       argument, displays the current on/off status of autoanswer.\n"
00888    "       The default value of autoanswer is in 'alsa.conf'.\n";
00889 
00890 static int console_answer(int fd, int argc, char *argv[])
00891 {
00892    int res = RESULT_SUCCESS;
00893 
00894    if (argc != 2)
00895       return RESULT_SHOWUSAGE;
00896 
00897    ast_mutex_lock(&alsalock);
00898 
00899    if (!alsa.owner) {
00900       ast_cli(fd, "No one is calling us\n");
00901       res = RESULT_FAILURE;
00902    } else {
00903       hookstate = 1;
00904       cursound = -1;
00905       grab_owner();
00906       if (alsa.owner) {
00907          struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
00908          ast_queue_frame(alsa.owner, &f);
00909          ast_mutex_unlock(&alsa.owner->lock);
00910       }
00911       answer_sound();
00912    }
00913 
00914    snd_pcm_prepare(alsa.icard);
00915    snd_pcm_start(alsa.icard);
00916 
00917    ast_mutex_unlock(&alsalock);
00918 
00919    return RESULT_SUCCESS;
00920 }
00921 
00922 static const char sendtext_usage[] =
00923    "Usage: console send text <message>\n"
00924    "       Sends a text message for display on the remote terminal.\n";
00925 
00926 static int console_sendtext(int fd, int argc, char *argv[])
00927 {
00928    int tmparg = 3;
00929    int res = RESULT_SUCCESS;
00930 
00931    if (argc < 3)
00932       return RESULT_SHOWUSAGE;
00933 
00934    ast_mutex_lock(&alsalock);
00935 
00936    if (!alsa.owner) {
00937       ast_cli(fd, "No one is calling us\n");
00938       res = RESULT_FAILURE;
00939    } else {
00940       struct ast_frame f = { AST_FRAME_TEXT, 0 };
00941       char text2send[256] = "";
00942       text2send[0] = '\0';
00943       while (tmparg < argc) {
00944          strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
00945          strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
00946       }
00947       text2send[strlen(text2send) - 1] = '\n';
00948       f.data = text2send;
00949       f.datalen = strlen(text2send) + 1;
00950       grab_owner();
00951       if (alsa.owner) {
00952          ast_queue_frame(alsa.owner, &f);
00953          f.frametype = AST_FRAME_CONTROL;
00954          f.subclass = AST_CONTROL_ANSWER;
00955          f.data = NULL;
00956          f.datalen = 0;
00957          ast_queue_frame(alsa.owner, &f);
00958          ast_mutex_unlock(&alsa.owner->lock);
00959       }
00960    }
00961 
00962    ast_mutex_unlock(&alsalock);
00963 
00964    return res;
00965 }
00966 
00967 static const char answer_usage[] =
00968    "Usage: console answer\n"
00969    "       Answers an incoming call on the console (ALSA) channel.\n";
00970 
00971 static int console_hangup(int fd, int argc, char *argv[])
<

Asterisk developer's documentation :: Codename Pineapple

chan_alsa.c