This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1026 lines
26 KiB
C
1026 lines
26 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* By Matthew Fredrickson <creslin@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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* \brief ALSA sound card channel driver
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*
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* \author Matthew Fredrickson <creslin@digium.com>
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*
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* \par See also
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* \arg Config_alsa
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>alsa</depend>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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#include <alsa/asoundlib.h>
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#include "asterisk/frame.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/config.h"
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#include "asterisk/cli.h"
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#include "asterisk/utils.h"
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#include "asterisk/causes.h"
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#include "asterisk/endian.h"
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#include "asterisk/stringfields.h"
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#include "asterisk/abstract_jb.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/poll-compat.h"
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/*! Global jitterbuffer configuration - by default, jb is disabled */
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static struct ast_jb_conf default_jbconf = {
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.flags = 0,
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.max_size = -1,
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.resync_threshold = -1,
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.impl = "",
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.target_extra = -1,
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};
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static struct ast_jb_conf global_jbconf;
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#define DEBUG 0
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/* Which device to use */
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#define ALSA_INDEV "default"
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#define ALSA_OUTDEV "default"
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#define DESIRED_RATE 8000
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/* Lets use 160 sample frames, just like GSM. */
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#define FRAME_SIZE 160
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#define PERIOD_FRAMES 80 /* 80 Frames, at 2 bytes each */
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/* When you set the frame size, you have to come up with
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the right buffer format as well. */
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/* 5 64-byte frames = one frame */
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#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
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/* Don't switch between read/write modes faster than every 300 ms */
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#define MIN_SWITCH_TIME 600
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#if __BYTE_ORDER == __LITTLE_ENDIAN
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static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
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#else
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static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
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#endif
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static char indevname[50] = ALSA_INDEV;
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static char outdevname[50] = ALSA_OUTDEV;
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static int silencesuppression = 0;
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static int silencethreshold = 1000;
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AST_MUTEX_DEFINE_STATIC(alsalock);
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static const char tdesc[] = "ALSA Console Channel Driver";
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static const char config[] = "alsa.conf";
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static char context[AST_MAX_CONTEXT] = "default";
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static char language[MAX_LANGUAGE] = "";
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static char exten[AST_MAX_EXTENSION] = "s";
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static char mohinterpret[MAX_MUSICCLASS];
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static int hookstate = 0;
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static struct chan_alsa_pvt {
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/* We only have one ALSA structure -- near sighted perhaps, but it
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keeps this driver as simple as possible -- as it should be. */
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struct ast_channel *owner;
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char exten[AST_MAX_EXTENSION];
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char context[AST_MAX_CONTEXT];
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snd_pcm_t *icard, *ocard;
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} alsa;
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/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
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with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
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usually plenty. */
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#define MAX_BUFFER_SIZE 100
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/* File descriptors for sound device */
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static int readdev = -1;
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static int writedev = -1;
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static int autoanswer = 1;
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static int mute = 0;
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static int noaudiocapture = 0;
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static struct ast_channel *alsa_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause);
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static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
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static int alsa_text(struct ast_channel *c, const char *text);
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static int alsa_hangup(struct ast_channel *c);
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static int alsa_answer(struct ast_channel *c);
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static struct ast_frame *alsa_read(struct ast_channel *chan);
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static int alsa_call(struct ast_channel *c, char *dest, int timeout);
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static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
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static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
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static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
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static struct ast_channel_tech alsa_tech = {
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.type = "Console",
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.description = tdesc,
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.requester = alsa_request,
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.send_digit_end = alsa_digit,
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.send_text = alsa_text,
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.hangup = alsa_hangup,
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.answer = alsa_answer,
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.read = alsa_read,
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.call = alsa_call,
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.write = alsa_write,
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.indicate = alsa_indicate,
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.fixup = alsa_fixup,
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};
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static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
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{
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int err;
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int direction;
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snd_pcm_t *handle = NULL;
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snd_pcm_hw_params_t *hwparams = NULL;
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snd_pcm_sw_params_t *swparams = NULL;
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struct pollfd pfd;
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snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
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snd_pcm_uframes_t buffer_size = 0;
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unsigned int rate = DESIRED_RATE;
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snd_pcm_uframes_t start_threshold, stop_threshold;
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err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
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if (err < 0) {
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ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
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return NULL;
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} else {
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ast_debug(1, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
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}
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hwparams = alloca(snd_pcm_hw_params_sizeof());
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memset(hwparams, 0, snd_pcm_hw_params_sizeof());
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snd_pcm_hw_params_any(handle, hwparams);
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err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0)
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ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
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err = snd_pcm_hw_params_set_format(handle, hwparams, format);
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if (err < 0)
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ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
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err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
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if (err < 0)
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ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
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direction = 0;
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err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
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if (rate != DESIRED_RATE)
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ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
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direction = 0;
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err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
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if (err < 0)
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ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
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else {
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ast_debug(1, "Period size is %d\n", err);
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}
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buffer_size = 4096 * 2; /* period_size * 16; */
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err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
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if (err < 0)
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ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
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else {
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ast_debug(1, "Buffer size is set to %d frames\n", err);
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}
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err = snd_pcm_hw_params(handle, hwparams);
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if (err < 0)
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ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
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swparams = alloca(snd_pcm_sw_params_sizeof());
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memset(swparams, 0, snd_pcm_sw_params_sizeof());
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snd_pcm_sw_params_current(handle, swparams);
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if (stream == SND_PCM_STREAM_PLAYBACK)
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start_threshold = period_size;
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else
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start_threshold = 1;
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err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
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if (err < 0)
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ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
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if (stream == SND_PCM_STREAM_PLAYBACK)
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stop_threshold = buffer_size;
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else
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stop_threshold = buffer_size;
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err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
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if (err < 0)
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ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
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err = snd_pcm_sw_params(handle, swparams);
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if (err < 0)
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ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
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err = snd_pcm_poll_descriptors_count(handle);
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if (err <= 0)
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ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
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if (err != 1) {
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ast_debug(1, "Can't handle more than one device\n");
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}
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snd_pcm_poll_descriptors(handle, &pfd, err);
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ast_debug(1, "Acquired fd %d from the poll descriptor\n", pfd.fd);
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if (stream == SND_PCM_STREAM_CAPTURE)
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readdev = pfd.fd;
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else
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writedev = pfd.fd;
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return handle;
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}
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static int soundcard_init(void)
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{
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if (!noaudiocapture) {
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alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
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if (!alsa.icard) {
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ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
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return -1;
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}
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}
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alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
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if (!alsa.ocard) {
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ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
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return -1;
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}
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return writedev;
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}
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static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
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{
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ast_mutex_lock(&alsalock);
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ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
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digit, duration);
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ast_mutex_unlock(&alsalock);
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return 0;
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}
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static int alsa_text(struct ast_channel *c, const char *text)
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{
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ast_mutex_lock(&alsalock);
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ast_verbose(" << Console Received text %s >> \n", text);
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ast_mutex_unlock(&alsalock);
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return 0;
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}
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static void grab_owner(void)
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{
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while (alsa.owner && ast_channel_trylock(alsa.owner)) {
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DEADLOCK_AVOIDANCE(&alsalock);
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}
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}
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static int alsa_call(struct ast_channel *c, char *dest, int timeout)
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{
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struct ast_frame f = { AST_FRAME_CONTROL };
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ast_mutex_lock(&alsalock);
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ast_verbose(" << Call placed to '%s' on console >> \n", dest);
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if (autoanswer) {
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ast_verbose(" << Auto-answered >> \n");
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if (mute) {
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ast_verbose( " << Muted >> \n" );
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}
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grab_owner();
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if (alsa.owner) {
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f.subclass.integer = AST_CONTROL_ANSWER;
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ast_queue_frame(alsa.owner, &f);
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ast_channel_unlock(alsa.owner);
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}
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} else {
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ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
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grab_owner();
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if (alsa.owner) {
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f.subclass.integer = AST_CONTROL_RINGING;
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ast_queue_frame(alsa.owner, &f);
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ast_channel_unlock(alsa.owner);
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ast_indicate(alsa.owner, AST_CONTROL_RINGING);
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}
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}
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if (!noaudiocapture) {
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snd_pcm_prepare(alsa.icard);
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snd_pcm_start(alsa.icard);
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}
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ast_mutex_unlock(&alsalock);
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return 0;
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}
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static int alsa_answer(struct ast_channel *c)
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{
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ast_mutex_lock(&alsalock);
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ast_verbose(" << Console call has been answered >> \n");
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ast_setstate(c, AST_STATE_UP);
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if (!noaudiocapture) {
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snd_pcm_prepare(alsa.icard);
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snd_pcm_start(alsa.icard);
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}
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ast_mutex_unlock(&alsalock);
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return 0;
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}
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static int alsa_hangup(struct ast_channel *c)
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{
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ast_mutex_lock(&alsalock);
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c->tech_pvt = NULL;
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alsa.owner = NULL;
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ast_verbose(" << Hangup on console >> \n");
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ast_module_unref(ast_module_info->self);
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hookstate = 0;
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if (!noaudiocapture) {
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snd_pcm_drop(alsa.icard);
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}
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ast_mutex_unlock(&alsalock);
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return 0;
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}
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static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
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{
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static char sizbuf[8000];
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static int sizpos = 0;
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int len = sizpos;
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int pos;
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int res = 0;
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/* size_t frames = 0; */
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snd_pcm_state_t state;
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ast_mutex_lock(&alsalock);
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/* We have to digest the frame in 160-byte portions */
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if (f->datalen > sizeof(sizbuf) - sizpos) {
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ast_log(LOG_WARNING, "Frame too large\n");
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res = -1;
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} else {
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memcpy(sizbuf + sizpos, f->data.ptr, f->datalen);
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len += f->datalen;
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pos = 0;
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state = snd_pcm_state(alsa.ocard);
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if (state == SND_PCM_STATE_XRUN)
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snd_pcm_prepare(alsa.ocard);
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while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
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usleep(1);
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}
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if (res == -EPIPE) {
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#if DEBUG
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ast_debug(1, "XRUN write\n");
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#endif
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snd_pcm_prepare(alsa.ocard);
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while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
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usleep(1);
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}
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if (res != len / 2) {
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ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
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res = -1;
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} else if (res < 0) {
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ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
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res = -1;
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}
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} else {
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if (res == -ESTRPIPE)
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ast_log(LOG_ERROR, "You've got some big problems\n");
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else if (res < 0)
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ast_log(LOG_NOTICE, "Error %d on write\n", res);
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}
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}
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ast_mutex_unlock(&alsalock);
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return res >= 0 ? 0 : res;
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}
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static struct ast_frame *alsa_read(struct ast_channel *chan)
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{
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static struct ast_frame f;
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static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET / 2];
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short *buf;
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static int readpos = 0;
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static int left = FRAME_SIZE;
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snd_pcm_state_t state;
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int r = 0;
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int off = 0;
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ast_mutex_lock(&alsalock);
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f.frametype = AST_FRAME_NULL;
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f.subclass.integer = 0;
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f.samples = 0;
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f.datalen = 0;
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f.data.ptr = NULL;
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f.offset = 0;
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f.src = "Console";
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f.mallocd = 0;
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f.delivery.tv_sec = 0;
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f.delivery.tv_usec = 0;
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if (noaudiocapture) {
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/* Return null frame to asterisk*/
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ast_mutex_unlock(&alsalock);
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return &f;
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}
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state = snd_pcm_state(alsa.icard);
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if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) {
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snd_pcm_prepare(alsa.icard);
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}
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buf = __buf + AST_FRIENDLY_OFFSET / 2;
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r = snd_pcm_readi(alsa.icard, buf + readpos, left);
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if (r == -EPIPE) {
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#if DEBUG
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ast_log(LOG_ERROR, "XRUN read\n");
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#endif
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snd_pcm_prepare(alsa.icard);
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} else if (r == -ESTRPIPE) {
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ast_log(LOG_ERROR, "-ESTRPIPE\n");
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snd_pcm_prepare(alsa.icard);
|
|
} else if (r < 0) {
|
|
ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
|
|
} else if (r >= 0) {
|
|
off -= r;
|
|
}
|
|
/* Update positions */
|
|
readpos += r;
|
|
left -= r;
|
|
|
|
if (readpos >= FRAME_SIZE) {
|
|
/* A real frame */
|
|
readpos = 0;
|
|
left = FRAME_SIZE;
|
|
if (chan->_state != AST_STATE_UP) {
|
|
/* Don't transmit unless it's up */
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
if (mute) {
|
|
/* Don't transmit if muted */
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
|
|
f.frametype = AST_FRAME_VOICE;
|
|
ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
|
|
f.samples = FRAME_SIZE;
|
|
f.datalen = FRAME_SIZE * 2;
|
|
f.data.ptr = buf;
|
|
f.offset = AST_FRIENDLY_OFFSET;
|
|
f.src = "Console";
|
|
f.mallocd = 0;
|
|
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return &f;
|
|
}
|
|
|
|
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
|
|
{
|
|
struct chan_alsa_pvt *p = newchan->tech_pvt;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
p->owner = newchan;
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
|
|
{
|
|
int res = 0;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
switch (cond) {
|
|
case AST_CONTROL_BUSY:
|
|
case AST_CONTROL_CONGESTION:
|
|
case AST_CONTROL_RINGING:
|
|
case -1:
|
|
res = -1; /* Ask for inband indications */
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
case AST_CONTROL_PROCEEDING:
|
|
case AST_CONTROL_VIDUPDATE:
|
|
case AST_CONTROL_SRCUPDATE:
|
|
break;
|
|
case AST_CONTROL_HOLD:
|
|
ast_verbose(" << Console Has Been Placed on Hold >> \n");
|
|
ast_moh_start(chan, data, mohinterpret);
|
|
break;
|
|
case AST_CONTROL_UNHOLD:
|
|
ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
|
|
ast_moh_stop(chan);
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
|
|
res = -1;
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const char *linkedid)
|
|
{
|
|
struct ast_channel *tmp = NULL;
|
|
|
|
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
|
|
return NULL;
|
|
|
|
tmp->tech = &alsa_tech;
|
|
ast_channel_set_fd(tmp, 0, readdev);
|
|
ast_format_set(&tmp->readformat, AST_FORMAT_SLINEAR, 0);
|
|
ast_format_set(&tmp->writeformat, AST_FORMAT_SLINEAR, 0);
|
|
ast_format_cap_add(tmp->nativeformats, &tmp->writeformat);
|
|
|
|
tmp->tech_pvt = p;
|
|
if (!ast_strlen_zero(p->context))
|
|
ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
|
|
if (!ast_strlen_zero(p->exten))
|
|
ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
|
|
if (!ast_strlen_zero(language))
|
|
ast_string_field_set(tmp, language, language);
|
|
p->owner = tmp;
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_jb_configure(tmp, &global_jbconf);
|
|
if (state != AST_STATE_DOWN) {
|
|
if (ast_pbx_start(tmp)) {
|
|
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
|
|
ast_hangup(tmp);
|
|
tmp = NULL;
|
|
}
|
|
}
|
|
|
|
return tmp;
|
|
}
|
|
|
|
static struct ast_channel *alsa_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
|
|
{
|
|
struct ast_format tmpfmt;
|
|
char buf[256];
|
|
struct ast_channel *tmp = NULL;
|
|
|
|
ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0);
|
|
|
|
if (!(ast_format_cap_iscompatible(cap, &tmpfmt))) {
|
|
ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
|
|
return NULL;
|
|
}
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (alsa.owner) {
|
|
ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
|
|
*cause = AST_CAUSE_BUSY;
|
|
} else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN, requestor ? requestor->linkedid : NULL))) {
|
|
ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return tmp;
|
|
}
|
|
|
|
static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
|
|
{
|
|
switch (state) {
|
|
case 0:
|
|
if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
|
|
return ast_strdup("on");
|
|
case 1:
|
|
if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
|
|
return ast_strdup("off");
|
|
default:
|
|
return NULL;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console autoanswer";
|
|
e->usage =
|
|
"Usage: console autoanswer [on|off]\n"
|
|
" Enables or disables autoanswer feature. If used without\n"
|
|
" argument, displays the current on/off status of autoanswer.\n"
|
|
" The default value of autoanswer is in 'alsa.conf'.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return autoanswer_complete(a->line, a->word, a->pos, a->n);
|
|
}
|
|
|
|
if ((a->argc != 2) && (a->argc != 3))
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
if (a->argc == 2) {
|
|
ast_cli(a->fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
|
|
} else {
|
|
if (!strcasecmp(a->argv[2], "on"))
|
|
autoanswer = -1;
|
|
else if (!strcasecmp(a->argv[2], "off"))
|
|
autoanswer = 0;
|
|
else
|
|
res = CLI_SHOWUSAGE;
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console answer";
|
|
e->usage =
|
|
"Usage: console answer\n"
|
|
" Answers an incoming call on the console (ALSA) channel.\n";
|
|
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc != 2)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner) {
|
|
ast_cli(a->fd, "No one is calling us\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
if (mute) {
|
|
ast_verbose( " << Muted >> \n" );
|
|
}
|
|
hookstate = 1;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
if (!noaudiocapture) {
|
|
snd_pcm_prepare(alsa.icard);
|
|
snd_pcm_start(alsa.icard);
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
int tmparg = 3;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console send text";
|
|
e->usage =
|
|
"Usage: console send text <message>\n"
|
|
" Sends a text message for display on the remote terminal.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc < 3)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner) {
|
|
ast_cli(a->fd, "No channel active\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
struct ast_frame f = { AST_FRAME_TEXT };
|
|
char text2send[256] = "";
|
|
|
|
while (tmparg < a->argc) {
|
|
strncat(text2send, a->argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
|
|
strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
|
|
}
|
|
|
|
text2send[strlen(text2send) - 1] = '\n';
|
|
f.data.ptr = text2send;
|
|
f.datalen = strlen(text2send) + 1;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_frame(alsa.owner, &f);
|
|
ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console hangup";
|
|
e->usage =
|
|
"Usage: console hangup\n"
|
|
" Hangs up any call currently placed on the console.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
|
|
if (a->argc != 2)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner && !hookstate) {
|
|
ast_cli(a->fd, "No call to hangup\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
hookstate = 0;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_hangup_with_cause(alsa.owner, AST_CAUSE_NORMAL_CLEARING);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char tmp[256], *tmp2;
|
|
char *mye, *myc;
|
|
const char *d;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console dial";
|
|
e->usage =
|
|
"Usage: console dial [extension[@context]]\n"
|
|
" Dials a given extension (and context if specified)\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if ((a->argc != 2) && (a->argc != 3))
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (alsa.owner) {
|
|
if (a->argc == 3) {
|
|
if (alsa.owner) {
|
|
for (d = a->argv[2]; *d; d++) {
|
|
struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = *d };
|
|
|
|
ast_queue_frame(alsa.owner, &f);
|
|
}
|
|
}
|
|
} else {
|
|
ast_cli(a->fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
|
|
res = CLI_FAILURE;
|
|
}
|
|
} else {
|
|
mye = exten;
|
|
myc = context;
|
|
if (a->argc == 3) {
|
|
char *stringp = NULL;
|
|
|
|
ast_copy_string(tmp, a->argv[2], sizeof(tmp));
|
|
stringp = tmp;
|
|
strsep(&stringp, "@");
|
|
tmp2 = strsep(&stringp, "@");
|
|
if (!ast_strlen_zero(tmp))
|
|
mye = tmp;
|
|
if (!ast_strlen_zero(tmp2))
|
|
myc = tmp2;
|
|
}
|
|
if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
|
|
ast_copy_string(alsa.exten, mye, sizeof(alsa.exten));
|
|
ast_copy_string(alsa.context, myc, sizeof(alsa.context));
|
|
hookstate = 1;
|
|
alsa_new(&alsa, AST_STATE_RINGING, NULL);
|
|
} else
|
|
ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
int toggle = 0;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console {mute|unmute} [toggle]";
|
|
e->usage =
|
|
"Usage: console {mute|unmute} [toggle]\n"
|
|
" Mute/unmute the microphone.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
|
|
if (a->argc > 3) {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
if (a->argc == 3) {
|
|
if (strcasecmp(a->argv[2], "toggle"))
|
|
return CLI_SHOWUSAGE;
|
|
toggle = 1;
|
|
}
|
|
|
|
if (a->argc < 2) {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
if (!strcasecmp(a->argv[1], "mute")) {
|
|
mute = toggle ? !mute : 1;
|
|
} else if (!strcasecmp(a->argv[1], "unmute")) {
|
|
mute = toggle ? !mute : 0;
|
|
} else {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");
|
|
|
|
return res;
|
|
}
|
|
|
|
static struct ast_cli_entry cli_alsa[] = {
|
|
AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
|
|
AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
|
|
AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
|
|
AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
|
|
AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
|
|
AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
|
|
};
|
|
|
|
static int load_module(void)
|
|
{
|
|
struct ast_config *cfg;
|
|
struct ast_variable *v;
|
|
struct ast_flags config_flags = { 0 };
|
|
struct ast_format tmpfmt;
|
|
|
|
if (!(alsa_tech.capabilities = ast_format_cap_alloc())) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
ast_format_cap_add(alsa_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
|
|
|
|
/* Copy the default jb config over global_jbconf */
|
|
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
|
|
|
|
strcpy(mohinterpret, "default");
|
|
|
|
if (!(cfg = ast_config_load(config, config_flags))) {
|
|
ast_log(LOG_ERROR, "Unable to read ALSA configuration file %s. Aborting.\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
|
|
ast_log(LOG_ERROR, "%s is in an invalid format. Aborting.\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
v = ast_variable_browse(cfg, "general");
|
|
for (; v; v = v->next) {
|
|
/* handle jb conf */
|
|
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
|
|
continue;
|
|
}
|
|
|
|
if (!strcasecmp(v->name, "autoanswer")) {
|
|
autoanswer = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "mute")) {
|
|
mute = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "noaudiocapture")) {
|
|
noaudiocapture = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "silencesuppression")) {
|
|
silencesuppression = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "silencethreshold")) {
|
|
silencethreshold = atoi(v->value);
|
|
} else if (!strcasecmp(v->name, "context")) {
|
|
ast_copy_string(context, v->value, sizeof(context));
|
|
} else if (!strcasecmp(v->name, "language")) {
|
|
ast_copy_string(language, v->value, sizeof(language));
|
|
} else if (!strcasecmp(v->name, "extension")) {
|
|
ast_copy_string(exten, v->value, sizeof(exten));
|
|
} else if (!strcasecmp(v->name, "input_device")) {
|
|
ast_copy_string(indevname, v->value, sizeof(indevname));
|
|
} else if (!strcasecmp(v->name, "output_device")) {
|
|
ast_copy_string(outdevname, v->value, sizeof(outdevname));
|
|
} else if (!strcasecmp(v->name, "mohinterpret")) {
|
|
ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
|
|
}
|
|
}
|
|
ast_config_destroy(cfg);
|
|
|
|
if (soundcard_init() < 0) {
|
|
ast_verb(2, "No sound card detected -- console channel will be unavailable\n");
|
|
ast_verb(2, "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (ast_channel_register(&alsa_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_channel_unregister(&alsa_tech);
|
|
ast_cli_unregister_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
|
|
|
|
if (alsa.icard)
|
|
snd_pcm_close(alsa.icard);
|
|
if (alsa.ocard)
|
|
snd_pcm_close(alsa.ocard);
|
|
if (alsa.owner)
|
|
ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
|
|
if (alsa.owner)
|
|
return -1;
|
|
|
|
alsa_tech.capabilities = ast_format_cap_destroy(alsa_tech.capabilities);
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
);
|