Wed Jan 27 20:02:05 2016

Asterisk developer's documentation


chan_alsa.c

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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>alsa</depend>
00032    <support_level>extended</support_level>
00033  ***/
00034 
00035 #include "asterisk.h"
00036 
00037 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 413586 $")
00038 
00039 #include <fcntl.h>
00040 #include <sys/ioctl.h>
00041 #include <sys/time.h>
00042 
00043 #define ALSA_PCM_NEW_HW_PARAMS_API
00044 #define ALSA_PCM_NEW_SW_PARAMS_API
00045 #include <alsa/asoundlib.h>
00046 
00047 #include "asterisk/frame.h"
00048 #include "asterisk/channel.h"
00049 #include "asterisk/module.h"
00050 #include "asterisk/pbx.h"
00051 #include "asterisk/config.h"
00052 #include "asterisk/cli.h"
00053 #include "asterisk/utils.h"
00054 #include "asterisk/causes.h"
00055 #include "asterisk/endian.h"
00056 #include "asterisk/stringfields.h"
00057 #include "asterisk/abstract_jb.h"
00058 #include "asterisk/musiconhold.h"
00059 #include "asterisk/poll-compat.h"
00060 
00061 /*! Global jitterbuffer configuration - by default, jb is disabled
00062  *  \note Values shown here match the defaults shown in alsa.conf.sample */
00063 static struct ast_jb_conf default_jbconf = {
00064    .flags = 0,
00065    .max_size = 200,
00066    .resync_threshold = 1000,
00067    .impl = "fixed",
00068    .target_extra = 40,
00069 };
00070 static struct ast_jb_conf global_jbconf;
00071 
00072 #define DEBUG 0
00073 /* Which device to use */
00074 #define ALSA_INDEV "default"
00075 #define ALSA_OUTDEV "default"
00076 #define DESIRED_RATE 8000
00077 
00078 /* Lets use 160 sample frames, just like GSM.  */
00079 #define FRAME_SIZE 160
00080 #define PERIOD_FRAMES 80      /* 80 Frames, at 2 bytes each */
00081 
00082 /* When you set the frame size, you have to come up with
00083    the right buffer format as well. */
00084 /* 5 64-byte frames = one frame */
00085 #define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
00086 
00087 /* Don't switch between read/write modes faster than every 300 ms */
00088 #define MIN_SWITCH_TIME 600
00089 
00090 #if __BYTE_ORDER == __LITTLE_ENDIAN
00091 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
00092 #else
00093 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
00094 #endif
00095 
00096 static char indevname[50] = ALSA_INDEV;
00097 static char outdevname[50] = ALSA_OUTDEV;
00098 
00099 static int silencesuppression = 0;
00100 static int silencethreshold = 1000;
00101 
00102 AST_MUTEX_DEFINE_STATIC(alsalock);
00103 
00104 static const char tdesc[] = "ALSA Console Channel Driver";
00105 static const char config[] = "alsa.conf";
00106 
00107 static char context[AST_MAX_CONTEXT] = "default";
00108 static char language[MAX_LANGUAGE] = "";
00109 static char exten[AST_MAX_EXTENSION] = "s";
00110 static char mohinterpret[MAX_MUSICCLASS];
00111 
00112 static int hookstate = 0;
00113 
00114 static struct chan_alsa_pvt {
00115    /* We only have one ALSA structure -- near sighted perhaps, but it
00116       keeps this driver as simple as possible -- as it should be. */
00117    struct ast_channel *owner;
00118    char exten[AST_MAX_EXTENSION];
00119    char context[AST_MAX_CONTEXT];
00120    snd_pcm_t *icard, *ocard;
00121 
00122 } alsa;
00123 
00124 /* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
00125    with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
00126    usually plenty. */
00127 
00128 #define MAX_BUFFER_SIZE 100
00129 
00130 /* File descriptors for sound device */
00131 static int readdev = -1;
00132 static int writedev = -1;
00133 
00134 static int autoanswer = 1;
00135 static int mute = 0;
00136 static int noaudiocapture = 0;
00137 
00138 static struct ast_channel *alsa_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
00139 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
00140 static int alsa_text(struct ast_channel *c, const char *text);
00141 static int alsa_hangup(struct ast_channel *c);
00142 static int alsa_answer(struct ast_channel *c);
00143 static struct ast_frame *alsa_read(struct ast_channel *chan);
00144 static int alsa_call(struct ast_channel *c, char *dest, int timeout);
00145 static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
00146 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
00147 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
00148 
00149 static const struct ast_channel_tech alsa_tech = {
00150    .type = "Console",
00151    .description = tdesc,
00152    .capabilities = AST_FORMAT_SLINEAR,
00153    .requester = alsa_request,
00154    .send_digit_end = alsa_digit,
00155    .send_text = alsa_text,
00156    .hangup = alsa_hangup,
00157    .answer = alsa_answer,
00158    .read = alsa_read,
00159    .call = alsa_call,
00160    .write = alsa_write,
00161    .indicate = alsa_indicate,
00162    .fixup = alsa_fixup,
00163 };
00164 
00165 static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
00166 {
00167    int err;
00168    int direction;
00169    snd_pcm_t *handle = NULL;
00170    snd_pcm_hw_params_t *hwparams = NULL;
00171    snd_pcm_sw_params_t *swparams = NULL;
00172    struct pollfd pfd;
00173    snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
00174    snd_pcm_uframes_t buffer_size = 0;
00175    unsigned int rate = DESIRED_RATE;
00176    snd_pcm_uframes_t start_threshold, stop_threshold;
00177 
00178    err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
00179    if (err < 0) {
00180       ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
00181       return NULL;
00182    } else {
00183       ast_debug(1, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
00184    }
00185 
00186    hwparams = ast_alloca(snd_pcm_hw_params_sizeof());
00187    memset(hwparams, 0, snd_pcm_hw_params_sizeof());
00188    snd_pcm_hw_params_any(handle, hwparams);
00189 
00190    err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
00191    if (err < 0)
00192       ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
00193 
00194    err = snd_pcm_hw_params_set_format(handle, hwparams, format);
00195    if (err < 0)
00196       ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
00197 
00198    err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
00199    if (err < 0)
00200       ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
00201 
00202    direction = 0;
00203    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
00204    if (rate != DESIRED_RATE)
00205       ast_log(LOG_WARNING, "Rate not correct, requested %d, got %u\n", DESIRED_RATE, rate);
00206 
00207    direction = 0;
00208    err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
00209    if (err < 0)
00210       ast_log(LOG_ERROR, "period_size(%lu frames) is bad: %s\n", period_size, snd_strerror(err));
00211    else {
00212       ast_debug(1, "Period size is %d\n", err);
00213    }
00214 
00215    buffer_size = 4096 * 2;    /* period_size * 16; */
00216    err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
00217    if (err < 0)
00218       ast_log(LOG_WARNING, "Problem setting buffer size of %lu: %s\n", buffer_size, snd_strerror(err));
00219    else {
00220       ast_debug(1, "Buffer size is set to %d frames\n", err);
00221    }
00222 
00223    err = snd_pcm_hw_params(handle, hwparams);
00224    if (err < 0)
00225       ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
00226 
00227    swparams = ast_alloca(snd_pcm_sw_params_sizeof());
00228    memset(swparams, 0, snd_pcm_sw_params_sizeof());
00229    snd_pcm_sw_params_current(handle, swparams);
00230 
00231    if (stream == SND_PCM_STREAM_PLAYBACK)
00232       start_threshold = period_size;
00233    else
00234       start_threshold = 1;
00235 
00236    err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
00237    if (err < 0)
00238       ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
00239 
00240    if (stream == SND_PCM_STREAM_PLAYBACK)
00241       stop_threshold = buffer_size;
00242    else
00243       stop_threshold = buffer_size;
00244 
00245    err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
00246    if (err < 0)
00247       ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
00248 
00249    err = snd_pcm_sw_params(handle, swparams);
00250    if (err < 0)
00251       ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
00252 
00253    err = snd_pcm_poll_descriptors_count(handle);
00254    if (err <= 0)
00255       ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
00256    if (err != 1) {
00257       ast_debug(1, "Can't handle more than one device\n");
00258    }
00259 
00260    snd_pcm_poll_descriptors(handle, &pfd, err);
00261    ast_debug(1, "Acquired fd %d from the poll descriptor\n", pfd.fd);
00262 
00263    if (stream == SND_PCM_STREAM_CAPTURE)
00264       readdev = pfd.fd;
00265    else
00266       writedev = pfd.fd;
00267 
00268    return handle;
00269 }
00270 
00271 static int soundcard_init(void)
00272 {
00273    if (!noaudiocapture) {
00274       alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
00275       if (!alsa.icard) {
00276          ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
00277          return -1;
00278       }
00279    }
00280 
00281    alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
00282 
00283    if (!alsa.ocard) {
00284       ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
00285       return -1;
00286    }
00287 
00288    return writedev;
00289 }
00290 
00291 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
00292 {
00293    ast_mutex_lock(&alsalock);
00294    ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
00295       digit, duration);
00296    ast_mutex_unlock(&alsalock);
00297 
00298    return 0;
00299 }
00300 
00301 static int alsa_text(struct ast_channel *c, const char *text)
00302 {
00303    ast_mutex_lock(&alsalock);
00304    ast_verbose(" << Console Received text %s >> \n", text);
00305    ast_mutex_unlock(&alsalock);
00306 
00307    return 0;
00308 }
00309 
00310 static void grab_owner(void)
00311 {
00312    while (alsa.owner && ast_channel_trylock(alsa.owner)) {
00313       DEADLOCK_AVOIDANCE(&alsalock);
00314    }
00315 }
00316 
00317 static int alsa_call(struct ast_channel *c, char *dest, int timeout)
00318 {
00319    struct ast_frame f = { AST_FRAME_CONTROL };
00320 
00321    ast_mutex_lock(&alsalock);
00322    ast_verbose(" << Call placed to '%s' on console >> \n", dest);
00323    if (autoanswer) {
00324       ast_verbose(" << Auto-answered >> \n");
00325       if (mute) {
00326          ast_verbose( " << Muted >> \n" );
00327       }
00328       grab_owner();
00329       if (alsa.owner) {
00330          f.subclass.integer = AST_CONTROL_ANSWER;
00331          ast_queue_frame(alsa.owner, &f);
00332          ast_channel_unlock(alsa.owner);
00333       }
00334    } else {
00335       ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
00336       grab_owner();
00337       if (alsa.owner) {
00338          f.subclass.integer = AST_CONTROL_RINGING;
00339          ast_queue_frame(alsa.owner, &f);
00340          ast_channel_unlock(alsa.owner);
00341          ast_indicate(alsa.owner, AST_CONTROL_RINGING);
00342       }
00343    }
00344    if (!noaudiocapture) {
00345       snd_pcm_prepare(alsa.icard);
00346       snd_pcm_start(alsa.icard);
00347    }
00348    ast_mutex_unlock(&alsalock);
00349 
00350    return 0;
00351 }
00352 
00353 static int alsa_answer(struct ast_channel *c)
00354 {
00355    ast_mutex_lock(&alsalock);
00356    ast_verbose(" << Console call has been answered >> \n");
00357    ast_setstate(c, AST_STATE_UP);
00358    if (!noaudiocapture) {
00359       snd_pcm_prepare(alsa.icard);
00360       snd_pcm_start(alsa.icard);
00361    }
00362    ast_mutex_unlock(&alsalock);
00363 
00364    return 0;
00365 }
00366 
00367 static int alsa_hangup(struct ast_channel *c)
00368 {
00369    ast_mutex_lock(&alsalock);
00370    c->tech_pvt = NULL;
00371    alsa.owner = NULL;
00372    ast_verbose(" << Hangup on console >> \n");
00373    ast_module_unref(ast_module_info->self);
00374    hookstate = 0;
00375    if (!noaudiocapture) {
00376       snd_pcm_drop(alsa.icard);
00377    }
00378    ast_mutex_unlock(&alsalock);
00379 
00380    return 0;
00381 }
00382 
00383 static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
00384 {
00385    static char sizbuf[8000];
00386    static int sizpos = 0;
00387    int len = sizpos;
00388    int res = 0;
00389    /* size_t frames = 0; */
00390    snd_pcm_state_t state;
00391 
00392    ast_mutex_lock(&alsalock);
00393 
00394    /* We have to digest the frame in 160-byte portions */
00395    if (f->datalen > sizeof(sizbuf) - sizpos) {
00396       ast_log(LOG_WARNING, "Frame too large\n");
00397       res = -1;
00398    } else {
00399       memcpy(sizbuf + sizpos, f->data.ptr, f->datalen);
00400       len += f->datalen;
00401       state = snd_pcm_state(alsa.ocard);
00402       if (state == SND_PCM_STATE_XRUN)
00403          snd_pcm_prepare(alsa.ocard);
00404       while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
00405          usleep(1);
00406       }
00407       if (res == -EPIPE) {
00408 #if DEBUG
00409          ast_debug(1, "XRUN write\n");
00410 #endif
00411          snd_pcm_prepare(alsa.ocard);
00412          while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
00413             usleep(1);
00414          }
00415          if (res != len / 2) {
00416             ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
00417             res = -1;
00418          } else if (res < 0) {
00419             ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
00420             res = -1;
00421          }
00422       } else {
00423          if (res == -ESTRPIPE)
00424             ast_log(LOG_ERROR, "You've got some big problems\n");
00425          else if (res < 0)
00426             ast_log(LOG_NOTICE, "Error %d on write\n", res);
00427       }
00428    }
00429    ast_mutex_unlock(&alsalock);
00430 
00431    return res >= 0 ? 0 : res;
00432 }
00433 
00434 
00435 static struct ast_frame *alsa_read(struct ast_channel *chan)
00436 {
00437    static struct ast_frame f;
00438    static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET / 2];
00439    short *buf;
00440    static int readpos = 0;
00441    static int left = FRAME_SIZE;
00442    snd_pcm_state_t state;
00443    int r = 0;
00444    int off = 0;
00445 
00446    ast_mutex_lock(&alsalock);
00447    f.frametype = AST_FRAME_NULL;
00448    f.subclass.integer = 0;
00449    f.samples = 0;
00450    f.datalen = 0;
00451    f.data.ptr = NULL;
00452    f.offset = 0;
00453    f.src = "Console";
00454    f.mallocd = 0;
00455    f.delivery.tv_sec = 0;
00456    f.delivery.tv_usec = 0;
00457 
00458    if (noaudiocapture) {
00459       /* Return null frame to asterisk*/
00460       ast_mutex_unlock(&alsalock);
00461       return &f;
00462    }
00463 
00464    state = snd_pcm_state(alsa.icard);
00465    if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) {
00466       snd_pcm_prepare(alsa.icard);
00467    }
00468 
00469    buf = __buf + AST_FRIENDLY_OFFSET / 2;
00470 
00471    r = snd_pcm_readi(alsa.icard, buf + readpos, left);
00472    if (r == -EPIPE) {
00473 #if DEBUG
00474       ast_log(LOG_ERROR, "XRUN read\n");
00475 #endif
00476       snd_pcm_prepare(alsa.icard);
00477    } else if (r == -ESTRPIPE) {
00478       ast_log(LOG_ERROR, "-ESTRPIPE\n");
00479       snd_pcm_prepare(alsa.icard);
00480    } else if (r < 0) {
00481       ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
00482    } else if (r >= 0) {
00483       off -= r;
00484    }
00485 
00486    /* Return NULL frame on error */
00487    if (r < 0) {
00488       ast_mutex_unlock(&alsalock);
00489       return &f;
00490    }
00491 
00492    /* Update positions */
00493    readpos += r;
00494    left -= r;
00495 
00496    if (readpos >= FRAME_SIZE) {
00497       /* A real frame */
00498       readpos = 0;
00499       left = FRAME_SIZE;
00500       if (chan->_state != AST_STATE_UP) {
00501          /* Don't transmit unless it's up */
00502          ast_mutex_unlock(&alsalock);
00503          return &f;
00504       }
00505       if (mute) {
00506          /* Don't transmit if muted */
00507          ast_mutex_unlock(&alsalock);
00508          return &f;
00509       }
00510 
00511       f.frametype = AST_FRAME_VOICE;
00512       f.subclass.codec = AST_FORMAT_SLINEAR;
00513       f.samples = FRAME_SIZE;
00514       f.datalen = FRAME_SIZE * 2;
00515       f.data.ptr = buf;
00516       f.offset = AST_FRIENDLY_OFFSET;
00517       f.src = "Console";
00518       f.mallocd = 0;
00519 
00520    }
00521    ast_mutex_unlock(&alsalock);
00522 
00523    return &f;
00524 }
00525 
00526 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
00527 {
00528    struct chan_alsa_pvt *p = newchan->tech_pvt;
00529 
00530    ast_mutex_lock(&alsalock);
00531    p->owner = newchan;
00532    ast_mutex_unlock(&alsalock);
00533 
00534    return 0;
00535 }
00536 
00537 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
00538 {
00539    int res = 0;
00540 
00541    ast_mutex_lock(&alsalock);
00542 
00543    switch (cond) {
00544    case AST_CONTROL_BUSY:
00545    case AST_CONTROL_CONGESTION:
00546    case AST_CONTROL_RINGING:
00547    case AST_CONTROL_INCOMPLETE:
00548    case -1:
00549       res = -1;  /* Ask for inband indications */
00550       break;
00551    case AST_CONTROL_PROGRESS:
00552    case AST_CONTROL_PROCEEDING:
00553    case AST_CONTROL_VIDUPDATE:
00554    case AST_CONTROL_SRCUPDATE:
00555       break;
00556    case AST_CONTROL_HOLD:
00557       ast_verbose(" << Console Has Been Placed on Hold >> \n");
00558       ast_moh_start(chan, data, mohinterpret);
00559       break;
00560    case AST_CONTROL_UNHOLD:
00561       ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
00562       ast_moh_stop(chan);
00563       break;
00564    default:
00565       ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
00566       res = -1;
00567    }
00568 
00569    ast_mutex_unlock(&alsalock);
00570 
00571    return res;
00572 }
00573 
00574 static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const char *linkedid)
00575 {
00576    struct ast_channel *tmp = NULL;
00577 
00578    if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
00579       return NULL;
00580 
00581    tmp->tech = &alsa_tech;
00582    ast_channel_set_fd(tmp, 0, readdev);
00583    tmp->nativeformats = AST_FORMAT_SLINEAR;
00584    tmp->readformat = AST_FORMAT_SLINEAR;
00585    tmp->writeformat = AST_FORMAT_SLINEAR;
00586    tmp->tech_pvt = p;
00587    if (!ast_strlen_zero(p->context))
00588       ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
00589    if (!ast_strlen_zero(p->exten))
00590       ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
00591    if (!ast_strlen_zero(language))
00592       ast_string_field_set(tmp, language, language);
00593    p->owner = tmp;
00594    ast_module_ref(ast_module_info->self);
00595    ast_jb_configure(tmp, &global_jbconf);
00596    if (state != AST_STATE_DOWN) {
00597       if (ast_pbx_start(tmp)) {
00598          ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
00599          ast_hangup(tmp);
00600          tmp = NULL;
00601       }
00602    }
00603 
00604    return tmp;
00605 }
00606 
00607 static struct ast_channel *alsa_request(const char *type, format_t fmt, const struct ast_channel *requestor, void *data, int *cause)
00608 {
00609    format_t oldformat = fmt;
00610    char buf[256];
00611    struct ast_channel *tmp = NULL;
00612 
00613    if (!(fmt &= AST_FORMAT_SLINEAR)) {
00614       ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), oldformat));
00615       return NULL;
00616    }
00617 
00618    ast_mutex_lock(&alsalock);
00619 
00620    if (alsa.owner) {
00621       ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
00622       *cause = AST_CAUSE_BUSY;
00623    } else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN, requestor ? requestor->linkedid : NULL))) {
00624       ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
00625    }
00626 
00627    ast_mutex_unlock(&alsalock);
00628 
00629    return tmp;
00630 }
00631 
00632 static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
00633 {
00634    switch (state) {
00635       case 0:
00636          if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
00637             return ast_strdup("on");
00638       case 1:
00639          if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
00640             return ast_strdup("off");
00641       default:
00642          return NULL;
00643    }
00644 
00645    return NULL;
00646 }
00647 
00648 static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00649 {
00650    char *res = CLI_SUCCESS;
00651 
00652    switch (cmd) {
00653    case CLI_INIT:
00654       e->command = "console autoanswer";
00655       e->usage =
00656          "Usage: console autoanswer [on|off]\n"
00657          "       Enables or disables autoanswer feature.  If used without\n"
00658          "       argument, displays the current on/off status of autoanswer.\n"
00659          "       The default value of autoanswer is in 'alsa.conf'.\n";
00660       return NULL;
00661    case CLI_GENERATE:
00662       return autoanswer_complete(a->line, a->word, a->pos, a->n);
00663    }
00664 
00665    if ((a->argc != 2) && (a->argc != 3))
00666       return CLI_SHOWUSAGE;
00667 
00668    ast_mutex_lock(&alsalock);
00669    if (a->argc == 2) {
00670       ast_cli(a->fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
00671    } else {
00672       if (!strcasecmp(a->argv[2], "on"))
00673          autoanswer = -1;
00674       else if (!strcasecmp(a->argv[2], "off"))
00675          autoanswer = 0;
00676       else
00677          res = CLI_SHOWUSAGE;
00678    }
00679    ast_mutex_unlock(&alsalock);
00680 
00681    return res;
00682 }
00683 
00684 static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00685 {
00686    char *res = CLI_SUCCESS;
00687 
00688    switch (cmd) {
00689    case CLI_INIT:
00690       e->command = "console answer";
00691       e->usage =
00692          "Usage: console answer\n"
00693          "       Answers an incoming call on the console (ALSA) channel.\n";
00694 
00695       return NULL;
00696    case CLI_GENERATE:
00697       return NULL; 
00698    }
00699 
00700    if (a->argc != 2)
00701       return CLI_SHOWUSAGE;
00702 
00703    ast_mutex_lock(&alsalock);
00704 
00705    if (!alsa.owner) {
00706       ast_cli(a->fd, "No one is calling us\n");
00707       res = CLI_FAILURE;
00708    } else {
00709       if (mute) {
00710          ast_verbose( " << Muted >> \n" );
00711       }
00712       hookstate = 1;
00713       grab_owner();
00714       if (alsa.owner) {
00715          ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
00716          ast_channel_unlock(alsa.owner);
00717       }
00718    }
00719 
00720    if (!noaudiocapture) {
00721       snd_pcm_prepare(alsa.icard);
00722       snd_pcm_start(alsa.icard);
00723    }
00724 
00725    ast_mutex_unlock(&alsalock);
00726 
00727    return res;
00728 }
00729 
00730 static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00731 {
00732    int tmparg = 3;
00733    char *res = CLI_SUCCESS;
00734 
00735    switch (cmd) {
00736    case CLI_INIT:
00737       e->command = "console send text";
00738       e->usage =
00739          "Usage: console send text <message>\n"
00740          "       Sends a text message for display on the remote terminal.\n";
00741       return NULL;
00742    case CLI_GENERATE:
00743       return NULL; 
00744    }
00745 
00746    if (a->argc < 3)
00747       return CLI_SHOWUSAGE;
00748 
00749    ast_mutex_lock(&alsalock);
00750 
00751    if (!alsa.owner) {
00752       ast_cli(a->fd, "No channel active\n");
00753       res = CLI_FAILURE;
00754    } else {
00755       struct ast_frame f = { AST_FRAME_TEXT };
00756       char text2send[256] = "";
00757 
00758       while (tmparg < a->argc) {
00759          strncat(text2send, a->argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
00760          strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
00761       }
00762 
00763       text2send[strlen(text2send) - 1] = '\n';
00764       f.data.ptr = text2send;
00765       f.datalen = strlen(text2send) + 1;
00766       grab_owner();
00767       if (alsa.owner) {
00768          ast_queue_frame(alsa.owner, &f);
00769          ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
00770          ast_channel_unlock(alsa.owner);
00771       }
00772    }
00773 
00774    ast_mutex_unlock(&alsalock);
00775 
00776    return res;
00777 }
00778 
00779 static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00780 {
00781    char *res = CLI_SUCCESS;
00782 
00783    switch (cmd) {
00784    case CLI_INIT:
00785       e->command = "console hangup";
00786       e->usage =
00787          "Usage: console hangup\n"
00788          "       Hangs up any call currently placed on the console.\n";
00789       return NULL;
00790    case CLI_GENERATE:
00791       return NULL; 
00792    }
00793  
00794 
00795    if (a->argc != 2)
00796       return CLI_SHOWUSAGE;
00797 
00798    ast_mutex_lock(&alsalock);
00799 
00800    if (!alsa.owner && !hookstate) {
00801       ast_cli(a->fd, "No call to hangup\n");
00802       res = CLI_FAILURE;
00803    } else {
00804       hookstate = 0;
00805       grab_owner();
00806       if (alsa.owner) {
00807          ast_queue_hangup_with_cause(alsa.owner, AST_CAUSE_NORMAL_CLEARING);
00808          ast_channel_unlock(alsa.owner);
00809       }
00810    }
00811 
00812    ast_mutex_unlock(&alsalock);
00813 
00814    return res;
00815 }
00816 
00817 static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00818 {
00819    char tmp[256], *tmp2;
00820    char *mye, *myc;
00821    const char *d;
00822    char *res = CLI_SUCCESS;
00823 
00824    switch (cmd) {
00825    case CLI_INIT:
00826       e->command = "console dial";
00827       e->usage =
00828          "Usage: console dial [extension[@context]]\n"
00829          "       Dials a given extension (and context if specified)\n";
00830       return NULL;
00831    case CLI_GENERATE:
00832       return NULL;
00833    }
00834 
00835    if ((a->argc != 2) && (a->argc != 3))
00836       return CLI_SHOWUSAGE;
00837 
00838    ast_mutex_lock(&alsalock);
00839 
00840    if (alsa.owner) {
00841       if (a->argc == 3) {
00842          if (alsa.owner) {
00843             for (d = a->argv[2]; *d; d++) {
00844                struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = *d };
00845 
00846                ast_queue_frame(alsa.owner, &f);
00847             }
00848          }
00849       } else {
00850          ast_cli(a->fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
00851          res = CLI_FAILURE;
00852       }
00853    } else {
00854       mye = exten;
00855       myc = context;
00856       if (a->argc == 3) {
00857          char *stringp = NULL;
00858 
00859          ast_copy_string(tmp, a->argv[2], sizeof(tmp));
00860          stringp = tmp;
00861          strsep(&stringp, "@");
00862          tmp2 = strsep(&stringp, "@");
00863          if (!ast_strlen_zero(tmp))
00864             mye = tmp;
00865          if (!ast_strlen_zero(tmp2))
00866             myc = tmp2;
00867       }
00868       if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
00869          ast_copy_string(alsa.exten, mye, sizeof(alsa.exten));
00870          ast_copy_string(alsa.context, myc, sizeof(alsa.context));
00871          hookstate = 1;
00872          alsa_new(&alsa, AST_STATE_RINGING, NULL);
00873       } else
00874          ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
00875    }
00876 
00877    ast_mutex_unlock(&alsalock);
00878 
00879    return res;
00880 }
00881 
00882 static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00883 {
00884    int toggle = 0;
00885    char *res = CLI_SUCCESS;
00886 
00887    switch (cmd) {
00888    case CLI_INIT:
00889       e->command = "console {mute|unmute} [toggle]";
00890       e->usage =
00891          "Usage: console {mute|unmute} [toggle]\n"
00892          "       Mute/unmute the microphone.\n";
00893       return NULL;
00894    case CLI_GENERATE:
00895       return NULL;
00896    }
00897 
00898 
00899    if (a->argc > 3) {
00900       return CLI_SHOWUSAGE;
00901    }
00902 
00903    if (a->argc == 3) {
00904       if (strcasecmp(a->argv[2], "toggle"))
00905          return CLI_SHOWUSAGE;
00906       toggle = 1;
00907    }
00908 
00909    if (a->argc < 2) {
00910       return CLI_SHOWUSAGE;
00911    }
00912 
00913    if (!strcasecmp(a->argv[1], "mute")) {
00914       mute = toggle ? !mute : 1;
00915    } else if (!strcasecmp(a->argv[1], "unmute")) {
00916       mute = toggle ? !mute : 0;
00917    } else {
00918       return CLI_SHOWUSAGE;
00919    }
00920 
00921    ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");
00922 
00923    return res;
00924 }
00925 
00926 static struct ast_cli_entry cli_alsa[] = {
00927    AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
00928    AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
00929    AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
00930    AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
00931    AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
00932    AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
00933 };
00934 
00935 static int load_module(void)
00936 {
00937    struct ast_config *cfg;
00938    struct ast_variable *v;
00939    struct ast_flags config_flags = { 0 };
00940 
00941    /* Copy the default jb config over global_jbconf */
00942    memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
00943 
00944    strcpy(mohinterpret, "default");
00945 
00946    if (!(cfg = ast_config_load(config, config_flags))) {
00947       ast_log(LOG_ERROR, "Unable to read ALSA configuration file %s.  Aborting.\n", config);
00948       return AST_MODULE_LOAD_DECLINE;
00949    } else if (cfg == CONFIG_STATUS_FILEINVALID) {
00950       ast_log(LOG_ERROR, "%s is in an invalid format.  Aborting.\n", config);
00951       return AST_MODULE_LOAD_DECLINE;
00952    }
00953 
00954    v = ast_variable_browse(cfg, "general");
00955    for (; v; v = v->next) {
00956       /* handle jb conf */
00957       if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
00958          continue;
00959       }
00960 
00961       if (!strcasecmp(v->name, "autoanswer")) {
00962          autoanswer = ast_true(v->value);
00963       } else if (!strcasecmp(v->name, "mute")) {
00964          mute = ast_true(v->value);
00965       } else if (!strcasecmp(v->name, "noaudiocapture")) {
00966          noaudiocapture = ast_true(v->value);
00967       } else if (!strcasecmp(v->name, "silencesuppression")) {
00968          silencesuppression = ast_true(v->value);
00969       } else if (!strcasecmp(v->name, "silencethreshold")) {
00970          silencethreshold = atoi(v->value);
00971       } else if (!strcasecmp(v->name, "context")) {
00972          ast_copy_string(context, v->value, sizeof(context));
00973       } else if (!strcasecmp(v->name, "language")) {
00974          ast_copy_string(language, v->value, sizeof(language));
00975       } else if (!strcasecmp(v->name, "extension")) {
00976          ast_copy_string(exten, v->value, sizeof(exten));
00977       } else if (!strcasecmp(v->name, "input_device")) {
00978          ast_copy_string(indevname, v->value, sizeof(indevname));
00979       } else if (!strcasecmp(v->name, "output_device")) {
00980          ast_copy_string(outdevname, v->value, sizeof(outdevname));
00981       } else if (!strcasecmp(v->name, "mohinterpret")) {
00982          ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
00983       }
00984    }
00985    ast_config_destroy(cfg);
00986 
00987    if (soundcard_init() < 0) {
00988       ast_verb(2, "No sound card detected -- console channel will be unavailable\n");
00989       ast_verb(2, "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
00990       return AST_MODULE_LOAD_DECLINE;
00991    }
00992 
00993    if (ast_channel_register(&alsa_tech)) {
00994       ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
00995       return AST_MODULE_LOAD_FAILURE;
00996    }
00997 
00998    ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
00999 
01000    return AST_MODULE_LOAD_SUCCESS;
01001 }
01002 
01003 static int unload_module(void)
01004 {
01005    ast_channel_unregister(&alsa_tech);
01006    ast_cli_unregister_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
01007 
01008    if (alsa.icard)
01009       snd_pcm_close(alsa.icard);
01010    if (alsa.ocard)
01011       snd_pcm_close(alsa.ocard);
01012    if (alsa.owner)
01013       ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
01014    if (alsa.owner)
01015       return -1;
01016 
01017    return 0;
01018 }
01019 
01020 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
01021       .load = load_module,
01022       .unload = unload_module,
01023       .load_pri = AST_MODPRI_CHANNEL_DRIVER,
01024    );

Generated on 27 Jan 2016 for Asterisk - The Open Source Telephony Project by  doxygen 1.6.1