Mon Jun 27 16:50:49 2011

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  ***/
00033 
00034 #include "asterisk.h"
00035 
00036 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 278132 $")
00037 
00038 #include <fcntl.h>
00039 #include <sys/ioctl.h>
00040 #include <sys/time.h>
00041 
00042 #define ALSA_PCM_NEW_HW_PARAMS_API
00043 #define ALSA_PCM_NEW_SW_PARAMS_API
00044 #include <alsa/asoundlib.h>
00045 
00046 #include "asterisk/frame.h"
00047 #include "asterisk/channel.h"
00048 #include "asterisk/module.h"
00049 #include "asterisk/pbx.h"
00050 #include "asterisk/config.h"
00051 #include "asterisk/cli.h"
00052 #include "asterisk/utils.h"
00053 #include "asterisk/causes.h"
00054 #include "asterisk/endian.h"
00055 #include "asterisk/stringfields.h"
00056 #include "asterisk/abstract_jb.h"
00057 #include "asterisk/musiconhold.h"
00058 #include "asterisk/poll-compat.h"
00059 
00060 /*! Global jitterbuffer configuration - by default, jb is disabled */
00061 static struct ast_jb_conf default_jbconf = {
00062    .flags = 0,
00063    .max_size = -1,
00064    .resync_threshold = -1,
00065    .impl = "",
00066    .target_extra = -1,
00067 };
00068 static struct ast_jb_conf global_jbconf;
00069 
00070 #define DEBUG 0
00071 /* Which device to use */
00072 #define ALSA_INDEV "default"
00073 #define ALSA_OUTDEV "default"
00074 #define DESIRED_RATE 8000
00075 
00076 /* Lets use 160 sample frames, just like GSM.  */
00077 #define FRAME_SIZE 160
00078 #define PERIOD_FRAMES 80      /* 80 Frames, at 2 bytes each */
00079 
00080 /* When you set the frame size, you have to come up with
00081    the right buffer format as well. */
00082 /* 5 64-byte frames = one frame */
00083 #define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
00084 
00085 /* Don't switch between read/write modes faster than every 300 ms */
00086 #define MIN_SWITCH_TIME 600
00087 
00088 #if __BYTE_ORDER == __LITTLE_ENDIAN
00089 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
00090 #else
00091 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
00092 #endif
00093 
00094 static char indevname[50] = ALSA_INDEV;
00095 static char outdevname[50] = ALSA_OUTDEV;
00096 
00097 static int silencesuppression = 0;
00098 static int silencethreshold = 1000;
00099 
00100 AST_MUTEX_DEFINE_STATIC(alsalock);
00101 
00102 static const char tdesc[] = "ALSA Console Channel Driver";
00103 static const char config[] = "alsa.conf";
00104 
00105 static char context[AST_MAX_CONTEXT] = "default";
00106 static char language[MAX_LANGUAGE] = "";
00107 static char exten[AST_MAX_EXTENSION] = "s";
00108 static char mohinterpret[MAX_MUSICCLASS];
00109 
00110 static int hookstate = 0;
00111 
00112 static struct chan_alsa_pvt {
00113    /* We only have one ALSA structure -- near sighted perhaps, but it
00114       keeps this driver as simple as possible -- as it should be. */
00115    struct ast_channel *owner;
00116    char exten[AST_MAX_EXTENSION];
00117    char context[AST_MAX_CONTEXT];
00118    snd_pcm_t *icard, *ocard;
00119 
00120 } alsa;
00121 
00122 /* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
00123    with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
00124    usually plenty. */
00125 
00126 #define MAX_BUFFER_SIZE 100
00127 
00128 /* File descriptors for sound device */
00129 static int readdev = -1;
00130 static int writedev = -1;
00131 
00132 static int autoanswer = 1;
00133 static int mute = 0;
00134 static int noaudiocapture = 0;
00135 
00136 static struct ast_channel *alsa_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
00137 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
00138 static int alsa_text(struct ast_channel *c, const char *text);
00139 static int alsa_hangup(struct ast_channel *c);
00140 static int alsa_answer(struct ast_channel *c);
00141 static struct ast_frame *alsa_read(struct ast_channel *chan);
00142 static int alsa_call(struct ast_channel *c, char *dest, int timeout);
00143 static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
00144 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
00145 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
00146 
00147 static const struct ast_channel_tech alsa_tech = {
00148    .type = "Console",
00149    .description = tdesc,
00150    .capabilities = AST_FORMAT_SLINEAR,
00151    .requester = alsa_request,
00152    .send_digit_end = alsa_digit,
00153    .send_text = alsa_text,
00154    .hangup = alsa_hangup,
00155    .answer = alsa_answer,
00156    .read = alsa_read,
00157    .call = alsa_call,
00158    .write = alsa_write,
00159    .indicate = alsa_indicate,
00160    .fixup = alsa_fixup,
00161 };
00162 
00163 static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
00164 {
00165    int err;
00166    int direction;
00167    snd_pcm_t *handle = NULL;
00168    snd_pcm_hw_params_t *hwparams = NULL;
00169    snd_pcm_sw_params_t *swparams = NULL;
00170    struct pollfd pfd;
00171    snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
00172    snd_pcm_uframes_t buffer_size = 0;
00173    unsigned int rate = DESIRED_RATE;
00174    snd_pcm_uframes_t start_threshold, stop_threshold;
00175 
00176    err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
00177    if (err < 0) {
00178       ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
00179       return NULL;
00180    } else {
00181       ast_debug(1, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
00182    }
00183 
00184    hwparams = alloca(snd_pcm_hw_params_sizeof());
00185    memset(hwparams, 0, snd_pcm_hw_params_sizeof());
00186    snd_pcm_hw_params_any(handle, hwparams);
00187 
00188    err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
00189    if (err < 0)
00190       ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
00191 
00192    err = snd_pcm_hw_params_set_format(handle, hwparams, format);
00193    if (err < 0)
00194       ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
00195 
00196    err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
00197    if (err < 0)
00198       ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
00199 
00200    direction = 0;
00201    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
00202    if (rate != DESIRED_RATE)
00203       ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
00204 
00205    direction = 0;
00206    err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
00207    if (err < 0)
00208       ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
00209    else {
00210       ast_debug(1, "Period size is %d\n", err);
00211    }
00212 
00213    buffer_size = 4096 * 2;    /* period_size * 16; */
00214    err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
00215    if (err < 0)
00216       ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
00217    else {
00218       ast_debug(1, "Buffer size is set to %d frames\n", err);
00219    }
00220 
00221    err = snd_pcm_hw_params(handle, hwparams);
00222    if (err < 0)
00223       ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
00224 
00225    swparams = alloca(snd_pcm_sw_params_sizeof());
00226    memset(swparams, 0, snd_pcm_sw_params_sizeof());
00227    snd_pcm_sw_params_current(handle, swparams);
00228 
00229    if (stream == SND_PCM_STREAM_PLAYBACK)
00230       start_threshold = period_size;
00231    else
00232       start_threshold = 1;
00233 
00234    err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
00235    if (err < 0)
00236       ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
00237 
00238    if (stream == SND_PCM_STREAM_PLAYBACK)
00239       stop_threshold = buffer_size;
00240    else
00241       stop_threshold = buffer_size;
00242 
00243    err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
00244    if (err < 0)
00245       ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
00246 
00247    err = snd_pcm_sw_params(handle, swparams);
00248    if (err < 0)
00249       ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
00250 
00251    err = snd_pcm_poll_descriptors_count(handle);
00252    if (err <= 0)
00253       ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
00254    if (err != 1) {
00255       ast_debug(1, "Can't handle more than one device\n");
00256    }
00257 
00258    snd_pcm_poll_descriptors(handle, &pfd, err);
00259    ast_debug(1, "Acquired fd %d from the poll descriptor\n", pfd.fd);
00260 
00261    if (stream == SND_PCM_STREAM_CAPTURE)
00262       readdev = pfd.fd;
00263    else
00264       writedev = pfd.fd;
00265 
00266    return handle;
00267 }
00268 
00269 static int soundcard_init(void)
00270 {
00271    if (!noaudiocapture) {
00272       alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
00273       if (!alsa.icard) {
00274          ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
00275          return -1;
00276       }
00277    }
00278 
00279    alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
00280 
00281    if (!alsa.ocard) {
00282       ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
00283       return -1;
00284    }
00285 
00286    return writedev;
00287 }
00288 
00289 static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
00290 {
00291    ast_mutex_lock(&alsalock);
00292    ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
00293       digit, duration);
00294    ast_mutex_unlock(&alsalock);
00295 
00296    return 0;
00297 }
00298 
00299 static int alsa_text(struct ast_channel *c, const char *text)
00300 {
00301    ast_mutex_lock(&alsalock);
00302    ast_verbose(" << Console Received text %s >> \n", text);
00303    ast_mutex_unlock(&alsalock);
00304 
00305    return 0;
00306 }
00307 
00308 static void grab_owner(void)
00309 {
00310    while (alsa.owner && ast_channel_trylock(alsa.owner)) {
00311       DEADLOCK_AVOIDANCE(&alsalock);
00312    }
00313 }
00314 
00315 static int alsa_call(struct ast_channel *c, char *dest, int timeout)
00316 {
00317    struct ast_frame f = { AST_FRAME_CONTROL };
00318 
00319    ast_mutex_lock(&alsalock);
00320    ast_verbose(" << Call placed to '%s' on console >> \n", dest);
00321    if (autoanswer) {
00322       ast_verbose(" << Auto-answered >> \n");
00323       if (mute) {
00324          ast_verbose( " << Muted >> \n" );
00325       }
00326       grab_owner();
00327       if (alsa.owner) {
00328          f.subclass.integer = AST_CONTROL_ANSWER;
00329          ast_queue_frame(alsa.owner, &f);
00330          ast_channel_unlock(alsa.owner);
00331       }
00332    } else {
00333       ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
00334       grab_owner();
00335       if (alsa.owner) {
00336          f.subclass.integer = AST_CONTROL_RINGING;
00337          ast_queue_frame(alsa.owner, &f);
00338          ast_channel_unlock(alsa.owner);
00339          ast_indicate(alsa.owner, AST_CONTROL_RINGING);
00340       }
00341    }
00342    if (!noaudiocapture) {
00343       snd_pcm_prepare(alsa.icard);
00344       snd_pcm_start(alsa.icard);
00345    }
00346    ast_mutex_unlock(&alsalock);
00347 
00348    return 0;
00349 }
00350 
00351 static int alsa_answer(struct ast_channel *c)
00352 {
00353    ast_mutex_lock(&alsalock);
00354    ast_verbose(" << Console call has been answered >> \n");
00355    ast_setstate(c, AST_STATE_UP);
00356    if (!noaudiocapture) {
00357       snd_pcm_prepare(alsa.icard);
00358       snd_pcm_start(alsa.icard);
00359    }
00360    ast_mutex_unlock(&alsalock);
00361 
00362    return 0;
00363 }
00364 
00365 static int alsa_hangup(struct ast_channel *c)
00366 {
00367    ast_mutex_lock(&alsalock);
00368    c->tech_pvt = NULL;
00369    alsa.owner = NULL;
00370    ast_verbose(" << Hangup on console >> \n");
00371    ast_module_unref(ast_module_info->self);
00372    hookstate = 0;
00373    if (!noaudiocapture) {
00374       snd_pcm_drop(alsa.icard);
00375    }
00376    ast_mutex_unlock(&alsalock);
00377 
00378    return 0;
00379 }
00380 
00381 static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
00382 {
00383    static char sizbuf[8000];
00384    static int sizpos = 0;
00385    int len = sizpos;
00386    int pos;
00387    int res = 0;
00388    /* size_t frames = 0; */
00389    snd_pcm_state_t state;
00390 
00391    ast_mutex_lock(&alsalock);
00392 
00393    /* We have to digest the frame in 160-byte portions */
00394    if (f->datalen > sizeof(sizbuf) - sizpos) {
00395       ast_log(LOG_WARNING, "Frame too large\n");
00396       res = -1;
00397    } else {
00398       memcpy(sizbuf + sizpos, f->data.ptr, f->datalen);
00399       len += f->datalen;
00400       pos = 0;
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    /* Update positions */
00486    readpos += r;
00487    left -= r;
00488 
00489    if (readpos >= FRAME_SIZE) {
00490       /* A real frame */
00491       readpos = 0;
00492       left = FRAME_SIZE;
00493       if (chan->_state != AST_STATE_UP) {
00494          /* Don't transmit unless it's up */
00495          ast_mutex_unlock(&alsalock);
00496          return &f;
00497       }
00498       if (mute) {
00499          /* Don't transmit if muted */
00500          ast_mutex_unlock(&alsalock);
00501          return &f;
00502       }
00503 
00504       f.frametype = AST_FRAME_VOICE;
00505       f.subclass.codec = AST_FORMAT_SLINEAR;
00506       f.samples = FRAME_SIZE;
00507       f.datalen = FRAME_SIZE * 2;
00508       f.data.ptr = buf;
00509       f.offset = AST_FRIENDLY_OFFSET;
00510       f.src = "Console";
00511       f.mallocd = 0;
00512 
00513    }
00514    ast_mutex_unlock(&alsalock);
00515 
00516    return &f;
00517 }
00518 
00519 static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
00520 {
00521    struct chan_alsa_pvt *p = newchan->tech_pvt;
00522 
00523    ast_mutex_lock(&alsalock);
00524    p->owner = newchan;
00525    ast_mutex_unlock(&alsalock);
00526 
00527    return 0;
00528 }
00529 
00530 static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
00531 {
00532    int res = 0;
00533 
00534    ast_mutex_lock(&alsalock);
00535 
00536    switch (cond) {
00537    case AST_CONTROL_BUSY:
00538    case AST_CONTROL_CONGESTION:
00539    case AST_CONTROL_RINGING:
00540    case -1:
00541       res = -1;  /* Ask for inband indications */
00542       break;
00543    case AST_CONTROL_PROGRESS:
00544    case AST_CONTROL_PROCEEDING:
00545    case AST_CONTROL_VIDUPDATE:
00546    case AST_CONTROL_SRCUPDATE:
00547       break;
00548    case AST_CONTROL_HOLD:
00549       ast_verbose(" << Console Has Been Placed on Hold >> \n");
00550       ast_moh_start(chan, data, mohinterpret);
00551       break;
00552    case AST_CONTROL_UNHOLD:
00553       ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
00554       ast_moh_stop(chan);
00555       break;
00556    default:
00557       ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
00558       res = -1;
00559    }
00560 
00561    ast_mutex_unlock(&alsalock);
00562 
00563    return res;
00564 }
00565 
00566 static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const char *linkedid)
00567 {
00568    struct ast_channel *tmp = NULL;
00569 
00570    if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
00571       return NULL;
00572 
00573    tmp->tech = &alsa_tech;
00574    ast_channel_set_fd(tmp, 0, readdev);
00575    tmp->nativeformats = AST_FORMAT_SLINEAR;
00576    tmp->readformat = AST_FORMAT_SLINEAR;
00577    tmp->writeformat = AST_FORMAT_SLINEAR;
00578    tmp->tech_pvt = p;
00579    if (!ast_strlen_zero(p->context))
00580       ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
00581    if (!ast_strlen_zero(p->exten))
00582       ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
00583    if (!ast_strlen_zero(language))
00584       ast_string_field_set(tmp, language, language);
00585    p->owner = tmp;
00586    ast_module_ref(ast_module_info->self);
00587    ast_jb_configure(tmp, &global_jbconf);
00588    if (state != AST_STATE_DOWN) {
00589       if (ast_pbx_start(tmp)) {
00590          ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
00591          ast_hangup(tmp);
00592          tmp = NULL;
00593       }
00594    }
00595 
00596    return tmp;
00597 }
00598 
00599 static struct ast_channel *alsa_request(const char *type, format_t fmt, const struct ast_channel *requestor, void *data, int *cause)
00600 {
00601    format_t oldformat = fmt;
00602    char buf[256];
00603    struct ast_channel *tmp = NULL;
00604 
00605    if (!(fmt &= AST_FORMAT_SLINEAR)) {
00606       ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), oldformat));
00607       return NULL;
00608    }
00609 
00610    ast_mutex_lock(&alsalock);
00611 
00612    if (alsa.owner) {
00613       ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
00614       *cause = AST_CAUSE_BUSY;
00615    } else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN, requestor ? requestor->linkedid : NULL))) {
00616       ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
00617    }
00618 
00619    ast_mutex_unlock(&alsalock);
00620 
00621    return tmp;
00622 }
00623 
00624 static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
00625 {
00626    switch (state) {
00627       case 0:
00628          if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
00629             return ast_strdup("on");
00630       case 1:
00631          if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
00632             return ast_strdup("off");
00633       default:
00634          return NULL;
00635    }
00636 
00637    return NULL;
00638 }
00639 
00640 static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00641 {
00642    char *res = CLI_SUCCESS;
00643 
00644    switch (cmd) {
00645    case CLI_INIT:
00646       e->command = "console autoanswer";
00647       e->usage =
00648          "Usage: console autoanswer [on|off]\n"
00649          "       Enables or disables autoanswer feature.  If used without\n"
00650          "       argument, displays the current on/off status of autoanswer.\n"
00651          "       The default value of autoanswer is in 'alsa.conf'.\n";
00652       return NULL;
00653    case CLI_GENERATE:
00654       return autoanswer_complete(a->line, a->word, a->pos, a->n);
00655    }
00656 
00657    if ((a->argc != 2) && (a->argc != 3))
00658       return CLI_SHOWUSAGE;
00659 
00660    ast_mutex_lock(&alsalock);
00661    if (a->argc == 2) {
00662       ast_cli(a->fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
00663    } else {
00664       if (!strcasecmp(a->argv[2], "on"))
00665          autoanswer = -1;
00666       else if (!strcasecmp(a->argv[2], "off"))
00667          autoanswer = 0;
00668       else
00669          res = CLI_SHOWUSAGE;
00670    }
00671    ast_mutex_unlock(&alsalock);
00672 
00673    return res;
00674 }
00675 
00676 static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00677 {
00678    char *res = CLI_SUCCESS;
00679 
00680    switch (cmd) {
00681    case CLI_INIT:
00682       e->command = "console answer";
00683       e->usage =
00684          "Usage: console answer\n"
00685          "       Answers an incoming call on the console (ALSA) channel.\n";
00686 
00687       return NULL;
00688    case CLI_GENERATE:
00689       return NULL; 
00690    }
00691 
00692    if (a->argc != 2)
00693       return CLI_SHOWUSAGE;
00694 
00695    ast_mutex_lock(&alsalock);
00696 
00697    if (!alsa.owner) {
00698       ast_cli(a->fd, "No one is calling us\n");
00699       res = CLI_FAILURE;
00700    } else {
00701       if (mute) {
00702          ast_verbose( " << Muted >> \n" );
00703       }
00704       hookstate = 1;
00705       grab_owner();
00706       if (alsa.owner) {
00707          ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
00708          ast_channel_unlock(alsa.owner);
00709       }
00710    }
00711 
00712    if (!noaudiocapture) {
00713       snd_pcm_prepare(alsa.icard);
00714       snd_pcm_start(alsa.icard);
00715    }
00716 
00717    ast_mutex_unlock(&alsalock);
00718 
00719    return res;
00720 }
00721 
00722 static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00723 {
00724    int tmparg = 3;
00725    char *res = CLI_SUCCESS;
00726 
00727    switch (cmd) {
00728    case CLI_INIT:
00729       e->command = "console send text";
00730       e->usage =
00731          "Usage: console send text <message>\n"
00732          "       Sends a text message for display on the remote terminal.\n";
00733       return NULL;
00734    case CLI_GENERATE:
00735       return NULL; 
00736    }
00737 
00738    if (a->argc < 3)
00739       return CLI_SHOWUSAGE;
00740 
00741    ast_mutex_lock(&alsalock);
00742 
00743    if (!alsa.owner) {
00744       ast_cli(a->fd, "No channel active\n");
00745       res = CLI_FAILURE;
00746    } else {
00747       struct ast_frame f = { AST_FRAME_TEXT };
00748       char text2send[256] = "";
00749 
00750       while (tmparg < a->argc) {
00751          strncat(text2send, a->argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
00752          strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
00753       }
00754 
00755       text2send[strlen(text2send) - 1] = '\n';
00756       f.data.ptr = text2send;
00757       f.datalen = strlen(text2send) + 1;
00758       grab_owner();
00759       if (alsa.owner) {
00760          ast_queue_frame(alsa.owner, &f);
00761          ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
00762          ast_channel_unlock(alsa.owner);
00763       }
00764    }
00765 
00766    ast_mutex_unlock(&alsalock);
00767 
00768    return res;
00769 }
00770 
00771 static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00772 {
00773    char *res = CLI_SUCCESS;
00774 
00775    switch (cmd) {
00776    case CLI_INIT:
00777       e->command = "console hangup";
00778       e->usage =
00779          "Usage: console hangup\n"
00780          "       Hangs up any call currently placed on the console.\n";
00781       return NULL;
00782    case CLI_GENERATE:
00783       return NULL; 
00784    }
00785  
00786 
00787    if (a->argc != 2)
00788       return CLI_SHOWUSAGE;
00789 
00790    ast_mutex_lock(&alsalock);
00791 
00792    if (!alsa.owner && !hookstate) {
00793       ast_cli(a->fd, "No call to hangup\n");
00794       res = CLI_FAILURE;
00795    } else {
00796       hookstate = 0;
00797       grab_owner();
00798       if (alsa.owner) {
00799          ast_queue_hangup_with_cause(alsa.owner, AST_CAUSE_NORMAL_CLEARING);
00800          ast_channel_unlock(alsa.owner);
00801       }
00802    }
00803 
00804    ast_mutex_unlock(&alsalock);
00805 
00806    return res;
00807 }
00808 
00809 static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00810 {
00811    char tmp[256], *tmp2;
00812    char *mye, *myc;
00813    const char *d;
00814    char *res = CLI_SUCCESS;
00815 
00816    switch (cmd) {
00817    case CLI_INIT:
00818       e->command = "console dial";
00819       e->usage =
00820          "Usage: console dial [extension[@context]]\n"
00821          "       Dials a given extension (and context if specified)\n";
00822       return NULL;
00823    case CLI_GENERATE:
00824       return NULL;
00825    }
00826 
00827    if ((a->argc != 2) && (a->argc != 3))
00828       return CLI_SHOWUSAGE;
00829 
00830    ast_mutex_lock(&alsalock);
00831 
00832    if (alsa.owner) {
00833       if (a->argc == 3) {
00834          if (alsa.owner) {
00835             for (d = a->argv[2]; *d; d++) {
00836                struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = *d };
00837 
00838                ast_queue_frame(alsa.owner, &f);
00839             }
00840          }
00841       } else {
00842          ast_cli(a->fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
00843          res = CLI_FAILURE;
00844       }
00845    } else {
00846       mye = exten;
00847       myc = context;
00848       if (a->argc == 3) {
00849          char *stringp = NULL;
00850 
00851          ast_copy_string(tmp, a->argv[2], sizeof(tmp));
00852          stringp = tmp;
00853          strsep(&stringp, "@");
00854          tmp2 = strsep(&stringp, "@");
00855          if (!ast_strlen_zero(tmp))
00856             mye = tmp;
00857          if (!ast_strlen_zero(tmp2))
00858             myc = tmp2;
00859       }
00860       if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
00861          ast_copy_string(alsa.exten, mye, sizeof(alsa.exten));
00862          ast_copy_string(alsa.context, myc, sizeof(alsa.context));
00863          hookstate = 1;
00864          alsa_new(&alsa, AST_STATE_RINGING, NULL);
00865       } else
00866          ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
00867    }
00868 
00869    ast_mutex_unlock(&alsalock);
00870 
00871    return res;
00872 }
00873 
00874 static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
00875 {
00876    int toggle = 0;
00877    char *res = CLI_SUCCESS;
00878 
00879    switch (cmd) {
00880    case CLI_INIT:
00881       e->command = "console {mute|unmute} [toggle]";
00882       e->usage =
00883          "Usage: console {mute|unmute} [toggle]\n"
00884          "       Mute/unmute the microphone.\n";
00885       return NULL;
00886    case CLI_GENERATE:
00887       return NULL;
00888    }
00889 
00890 
00891    if (a->argc > 3) {
00892       return CLI_SHOWUSAGE;
00893    }
00894 
00895    if (a->argc == 3) {
00896       if (strcasecmp(a->argv[2], "toggle"))
00897          return CLI_SHOWUSAGE;
00898       toggle = 1;
00899    }
00900 
00901    if (a->argc < 2) {
00902       return CLI_SHOWUSAGE;
00903    }
00904 
00905    if (!strcasecmp(a->argv[1], "mute")) {
00906       mute = toggle ? !mute : 1;
00907    } else if (!strcasecmp(a->argv[1], "unmute")) {
00908       mute = toggle ? !mute : 0;
00909    } else {
00910       return CLI_SHOWUSAGE;
00911    }
00912 
00913    ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");
00914 
00915    return res;
00916 }
00917 
00918 static struct ast_cli_entry cli_alsa[] = {
00919    AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
00920    AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
00921    AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
00922    AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
00923    AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
00924    AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
00925 };
00926 
00927 static int load_module(void)
00928 {
00929    struct ast_config *cfg;
00930    struct ast_variable *v;
00931    struct ast_flags config_flags = { 0 };
00932 
00933    /* Copy the default jb config over global_jbconf */
00934    memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
00935 
00936    strcpy(mohinterpret, "default");
00937 
00938    if (!(cfg = ast_config_load(config, config_flags))) {
00939       ast_log(LOG_ERROR, "Unable to read ALSA configuration file %s.  Aborting.\n", config);
00940       return AST_MODULE_LOAD_DECLINE;
00941    } else if (cfg == CONFIG_STATUS_FILEINVALID) {
00942       ast_log(LOG_ERROR, "%s is in an invalid format.  Aborting.\n", config);
00943       return AST_MODULE_LOAD_DECLINE;
00944    }
00945 
00946    v = ast_variable_browse(cfg, "general");
00947    for (; v; v = v->next) {
00948       /* handle jb conf */
00949       if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
00950          continue;
00951       }
00952 
00953       if (!strcasecmp(v->name, "autoanswer")) {
00954          autoanswer = ast_true(v->value);
00955       } else if (!strcasecmp(v->name, "mute")) {
00956          mute = ast_true(v->value);
00957       } else if (!strcasecmp(v->name, "noaudiocapture")) {
00958          noaudiocapture = ast_true(v->value);
00959       } else if (!strcasecmp(v->name, "silencesuppression")) {
00960          silencesuppression = ast_true(v->value);
00961       } else if (!strcasecmp(v->name, "silencethreshold")) {
00962          silencethreshold = atoi(v->value);
00963       } else if (!strcasecmp(v->name, "context")) {
00964          ast_copy_string(context, v->value, sizeof(context));
00965       } else if (!strcasecmp(v->name, "language")) {
00966          ast_copy_string(language, v->value, sizeof(language));
00967       } else if (!strcasecmp(v->name, "extension")) {
00968          ast_copy_string(exten, v->value, sizeof(exten));
00969       } else if (!strcasecmp(v->name, "input_device")) {
00970          ast_copy_string(indevname, v->value, sizeof(indevname));
00971       } else if (!strcasecmp(v->name, "output_device")) {
00972          ast_copy_string(outdevname, v->value, sizeof(outdevname));
00973       } else if (!strcasecmp(v->name, "mohinterpret")) {
00974          ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
00975       }
00976    }
00977    ast_config_destroy(cfg);
00978 
00979    if (soundcard_init() < 0) {
00980       ast_verb(2, "No sound card detected -- console channel will be unavailable\n");
00981       ast_verb(2, "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
00982       return AST_MODULE_LOAD_DECLINE;
00983    }
00984 
00985    if (ast_channel_register(&alsa_tech)) {
00986       ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
00987       return AST_MODULE_LOAD_FAILURE;
00988    }
00989 
00990    ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
00991 
00992    return AST_MODULE_LOAD_SUCCESS;
00993 }
00994 
00995 static int unload_module(void)
00996 {
00997    ast_channel_unregister(&alsa_tech);
00998    ast_cli_unregister_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
00999 
01000    if (alsa.icard)
01001       snd_pcm_close(alsa.icard);
01002    if (alsa.ocard)
01003       snd_pcm_close(alsa.ocard);
01004    if (alsa.owner)
01005       ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
01006    if (alsa.owner)
01007       return -1;
01008 
01009    return 0;
01010 }
01011 
01012 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
01013       .load = load_module,
01014       .unload = unload_module,
01015       .load_pri = AST_MODPRI_CHANNEL_DRIVER,
01016    );

Generated on Mon Jun 27 16:50:49 2011 for Asterisk - The Open Source Telephony Project by  doxygen 1.4.7