codec_dahdi.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * DAHDI native transcoding support
 *
 * Copyright (C) 1999 - 2008, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 * Kevin P. Fleming <kpfleming@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */
 
/*! \file
 *
 * \brief Translate between various formats natively through DAHDI transcoding
 *
 * \ingroup codecs
 */
 
/*** MODULEINFO
    <depend>dahdi</depend>
    <support_level>core</support_level>
 ***/
 
#include "asterisk.h"
#include <stdbool.h>
 
#include <poll.h>
#include <fcntl.h>
#include <netinet/in.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <dahdi/user.h>
 
#include "asterisk/lock.h"
#include "asterisk/translate.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/cli.h"
#include "asterisk/channel.h"
#include "asterisk/utils.h"
#include "asterisk/linkedlists.h"
#include "asterisk/ulaw.h"
#include "asterisk/format_compatibility.h"
 
#define BUFFER_SIZE 8000
 
#define G723_SAMPLES 240
#define G729_SAMPLES 160
#define ULAW_SAMPLES 160
 
/* Defines from DAHDI. */
#ifndef DAHDI_FORMAT_MAX_AUDIO
/*! G.723.1 compression */
#define DAHDI_FORMAT_G723_1    (1 << 0)
/*! GSM compression */
#define DAHDI_FORMAT_GSM       (1 << 1)
/*! Raw mu-law data (G.711) */
#define DAHDI_FORMAT_ULAW      (1 << 2)
/*! Raw A-law data (G.711) */
#define DAHDI_FORMAT_ALAW      (1 << 3)
/*! ADPCM (G.726, 32kbps) */
#define DAHDI_FORMAT_G726      (1 << 4)
/*! ADPCM (IMA) */
#define DAHDI_FORMAT_ADPCM     (1 << 5)
/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
#define DAHDI_FORMAT_SLINEAR   (1 << 6)
/*! LPC10, 180 samples/frame */
#define DAHDI_FORMAT_LPC10     (1 << 7)
/*! G.729A audio */
#define DAHDI_FORMAT_G729A     (1 << 8)
/*! SpeeX Free Compression */
#define DAHDI_FORMAT_SPEEX     (1 << 9)
/*! iLBC Free Compression */
#define DAHDI_FORMAT_ILBC      (1 << 10)
#endif
 
static struct channel_usage {
    int total;
    int encoders;
    int decoders;
} channels;
 
#if defined(NOT_NEEDED)
/*!
 * \internal
 * \brief Convert DAHDI format bitfield to old Asterisk format bitfield.
 * \since 13.0.0
 *
 * \param dahdi Bitfield from DAHDI to convert.
 *
 * \note They should be the same values but they don't have to be.
 *
 * \return Old Asterisk bitfield equivalent.
 */
static uint64_t bitfield_dahdi2ast(unsigned dahdi)
{
    uint64_t ast;
 
    switch (dahdi) {
    case DAHDI_FORMAT_G723_1:
        ast = AST_FORMAT_G723;
        break;
    case DAHDI_FORMAT_GSM:
        ast = AST_FORMAT_GSM;
        break;
    case DAHDI_FORMAT_ULAW:
        ast = AST_FORMAT_ULAW;
        break;
    case DAHDI_FORMAT_ALAW:
        ast = AST_FORMAT_ALAW;
        break;
    case DAHDI_FORMAT_G726:
        ast = AST_FORMAT_G726_AAL2;
        break;
    case DAHDI_FORMAT_ADPCM:
        ast = AST_FORMAT_ADPCM;
        break;
    case DAHDI_FORMAT_SLINEAR:
        ast = AST_FORMAT_SLIN;
        break;
    case DAHDI_FORMAT_LPC10:
        ast = AST_FORMAT_LPC10;
        break;
    case DAHDI_FORMAT_G729A:
        ast = AST_FORMAT_G729;
        break;
    case DAHDI_FORMAT_SPEEX:
        ast = AST_FORMAT_SPEEX;
        break;
    case DAHDI_FORMAT_ILBC:
        ast = AST_FORMAT_ILBC;
        break;
    default:
        ast = 0;
        break;
    }
 
    return ast;
}
#endif  /* defined(NOT_NEEDED) */
 
/*!
 * \internal
 * \brief Get the ast_codec by DAHDI format.
 * \since 13.0.0
 *
 * \param dahdi_fmt DAHDI specific codec identifier.
 *
 * \return Specified codec if exists otherwise NULL.
 */
static const struct ast_codec *get_dahdi_codec(uint32_t dahdi_fmt)
{
    const struct ast_codec *codec;
 
    static const struct ast_codec dahdi_g723_1 = {
        .name = "g723",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_gsm = {
        .name = "gsm",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_ulaw = {
        .name = "ulaw",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_alaw = {
        .name = "alaw",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_g726 = {
        .name = "g726",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_adpcm = {
        .name = "adpcm",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_slinear = {
        .name = "slin",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_lpc10 = {
        .name = "lpc10",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_g729a = {
        .name = "g729",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_speex = {
        .name = "speex",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
    static const struct ast_codec dahdi_ilbc = {
        .name = "ilbc",
        .type = AST_MEDIA_TYPE_AUDIO,
        .sample_rate = 8000,
    };
 
    switch (dahdi_fmt) {
    case DAHDI_FORMAT_G723_1:
        codec = &dahdi_g723_1;
        break;
    case DAHDI_FORMAT_GSM:
        codec = &dahdi_gsm;
        break;
    case DAHDI_FORMAT_ULAW:
        codec = &dahdi_ulaw;
        break;
    case DAHDI_FORMAT_ALAW:
        codec = &dahdi_alaw;
        break;
    case DAHDI_FORMAT_G726:
        codec = &dahdi_g726;
        break;
    case DAHDI_FORMAT_ADPCM:
        codec = &dahdi_adpcm;
        break;
    case DAHDI_FORMAT_SLINEAR:
        codec = &dahdi_slinear;
        break;
    case DAHDI_FORMAT_LPC10:
        codec = &dahdi_lpc10;
        break;
    case DAHDI_FORMAT_G729A:
        codec = &dahdi_g729a;
        break;
    case DAHDI_FORMAT_SPEEX:
        codec = &dahdi_speex;
        break;
    case DAHDI_FORMAT_ILBC:
        codec = &dahdi_ilbc;
        break;
    default:
        codec = NULL;
        break;
    }
 
    return codec;
}
 
static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 
static struct ast_cli_entry cli[] = {
    AST_CLI_DEFINE(handle_cli_transcoder_show, "Display DAHDI transcoder utilization.")
};
 
struct translator {
    struct ast_translator t;
    uint32_t src_dahdi_fmt;
    uint32_t dst_dahdi_fmt;
    AST_LIST_ENTRY(translator) entry;
};
 
#ifndef container_of
#define container_of(ptr, type, member) \
    ((type *)((char *)(ptr) - offsetof(type, member)))
#endif
 
static AST_LIST_HEAD_STATIC(translators, translator);
 
struct codec_dahdi_pvt {
    int fd;
    struct dahdi_transcoder_formats fmts;
    unsigned int softslin:1;
    unsigned int fake:2;
    uint16_t required_samples;
    uint16_t samples_in_buffer;
    uint16_t samples_written_to_hardware;
    uint8_t ulaw_buffer[1024];
};
 
/* Only used by a decoder */
static int ulawtolin(struct ast_trans_pvt *pvt, int samples)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
    int i = samples;
    uint8_t *src = &dahdip->ulaw_buffer[0];
    int16_t *dst = pvt->outbuf.i16 + pvt->datalen;
 
    /* convert and copy in outbuf */
    while (i--) {
        *dst++ = AST_MULAW(*src++);
    }
 
    return 0;
}
 
/* Only used by an encoder. */
static int lintoulaw(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
    int i = f->samples;
    uint8_t *dst = &dahdip->ulaw_buffer[dahdip->samples_in_buffer];
    int16_t *src = f->data.ptr;
 
    if (dahdip->samples_in_buffer + i > sizeof(dahdip->ulaw_buffer)) {
        ast_log(LOG_ERROR, "Out of buffer space!\n");
        return -i;
    }
 
    while (i--) {
        *dst++ = AST_LIN2MU(*src++);
    }
 
    dahdip->samples_in_buffer += f->samples;
    return 0;
}
 
static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
    struct channel_usage copy;
 
    switch (cmd) {
    case CLI_INIT:
        e->command = "transcoder show";
        e->usage =
            "Usage: transcoder show\n"
            "       Displays channel utilization of DAHDI transcoder(s).\n";
        return NULL;
    case CLI_GENERATE:
        return NULL;
    }
 
    if (a->argc != 2)
        return CLI_SHOWUSAGE;
 
    copy = channels;
 
    if (copy.total == 0)
        ast_cli(a->fd, "No DAHDI transcoders found.\n");
    else
        ast_cli(a->fd, "%d/%d encoders/decoders of %d channels are in use.\n", copy.encoders, copy.decoders, copy.total);
 
    return CLI_SUCCESS;
}
 
static void dahdi_write_frame(struct codec_dahdi_pvt *dahdip, const uint8_t *buffer, const ssize_t count)
{
    int res;
    if (!count) return;
    res = write(dahdip->fd, buffer, count);
    if (-1 == res) {
        ast_log(LOG_ERROR, "Failed to write to transcoder: %s\n", strerror(errno));
    }
    if (count != res) {
        ast_log(LOG_ERROR, "Requested write of %zd bytes, but only wrote %d bytes.\n", count, res);
    }
}
 
static int dahdi_encoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
    if (!f->subclass.format) {
        /* We're just faking a return for calculation purposes. */
        dahdip->fake = 2;
        pvt->samples = f->samples;
        return 0;
    }
 
    /* Buffer up the packets and send them to the hardware if we
     * have enough samples set up. */
    if (dahdip->softslin) {
        if (lintoulaw(pvt, f)) {
             return -1;
        }
    } else {
        /* NOTE:  If softslin support is not needed, and the sample
         * size is equal to the required sample size, we wouldn't
         * need this copy operation.  But at the time this was
         * written, only softslin is supported. */
        if (dahdip->samples_in_buffer + f->samples > sizeof(dahdip->ulaw_buffer)) {
            ast_log(LOG_ERROR, "Out of buffer space.\n");
            return -1;
        }
        memcpy(&dahdip->ulaw_buffer[dahdip->samples_in_buffer], f->data.ptr, f->samples);
        dahdip->samples_in_buffer += f->samples;
    }
 
    while (dahdip->samples_in_buffer >= dahdip->required_samples) {
        dahdi_write_frame(dahdip, dahdip->ulaw_buffer, dahdip->required_samples);
        dahdip->samples_written_to_hardware += dahdip->required_samples;
        dahdip->samples_in_buffer -= dahdip->required_samples;
        if (dahdip->samples_in_buffer) {
            /* Shift any remaining bytes down. */
            memmove(dahdip->ulaw_buffer, &dahdip->ulaw_buffer[dahdip->required_samples],
                dahdip->samples_in_buffer);
        }
    }
    pvt->samples += f->samples;
    pvt->datalen = 0;
    return -1;
}
 
static void dahdi_wait_for_packet(int fd)
{
    struct pollfd p = {0};
    p.fd = fd;
    p.events = POLLIN;
    poll(&p, 1, 10);
}
 
static struct ast_frame *dahdi_encoder_frameout(struct ast_trans_pvt *pvt)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
    int res;
 
    if (2 == dahdip->fake) {
        struct ast_frame frm = {
            .frametype = AST_FRAME_VOICE,
            .samples = dahdip->required_samples,
            .src = pvt->t->name,
        };
 
        dahdip->fake = 1;
        pvt->samples = 0;
 
        return ast_frisolate(&frm);
    } else if (1 == dahdip->fake) {
        dahdip->fake = 0;
        return NULL;
    }
 
    if (dahdip->samples_written_to_hardware >= dahdip->required_samples) {
        dahdi_wait_for_packet(dahdip->fd);
    }
 
    res = read(dahdip->fd, pvt->outbuf.c + pvt->datalen, pvt->t->buf_size - pvt->datalen);
    if (-1 == res) {
        if (EWOULDBLOCK == errno) {
            /* Nothing waiting... */
            return NULL;
        } else {
            ast_log(LOG_ERROR, "Failed to read from transcoder: %s\n", strerror(errno));
            return NULL;
        }
    } else {
        pvt->f.datalen = res;
        pvt->f.samples = ast_codec_samples_count(&pvt->f);
 
        dahdip->samples_written_to_hardware =
          (dahdip->samples_written_to_hardware >= pvt->f.samples) ?
             dahdip->samples_written_to_hardware - pvt->f.samples : 0;
 
        pvt->samples = 0;
        pvt->datalen = 0;
        return ast_frisolate(&pvt->f);
    }
 
    /* Shouldn't get here... */
    return NULL;
}
 
static int dahdi_decoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
    if (!f->subclass.format) {
        /* We're just faking a return for calculation purposes. */
        dahdip->fake = 2;
        pvt->samples = f->samples;
        return 0;
    }
 
    if (!f->datalen) {
        if (f->samples != dahdip->required_samples) {
            ast_log(LOG_ERROR, "%d != %d %d\n", f->samples, dahdip->required_samples, f->datalen);
        }
    }
    dahdi_write_frame(dahdip, f->data.ptr, f->datalen);
    dahdip->samples_written_to_hardware += f->samples;
    pvt->samples += f->samples;
    pvt->datalen = 0;
    return -1;
}
 
static struct ast_frame *dahdi_decoder_frameout(struct ast_trans_pvt *pvt)
{
    int res;
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
    if (2 == dahdip->fake) {
        struct ast_frame frm = {
            .frametype = AST_FRAME_VOICE,
            .samples = dahdip->required_samples,
            .src = pvt->t->name,
        };
 
        dahdip->fake = 1;
        pvt->samples = 0;
 
        return ast_frisolate(&frm);
    } else if (1 == dahdip->fake) {
        pvt->samples = 0;
        dahdip->fake = 0;
        return NULL;
    }
 
    if (dahdip->samples_written_to_hardware >= ULAW_SAMPLES) {
        dahdi_wait_for_packet(dahdip->fd);
    }
 
    /* Let's check to see if there is a new frame for us.... */
    if (dahdip->softslin) {
        res = read(dahdip->fd, dahdip->ulaw_buffer, sizeof(dahdip->ulaw_buffer));
    } else {
        res = read(dahdip->fd, pvt->outbuf.c + pvt->datalen, pvt->t->buf_size - pvt->datalen);
    }
 
    if (-1 == res) {
        if (EWOULDBLOCK == errno) {
            /* Nothing waiting... */
            return NULL;
        } else {
            ast_log(LOG_ERROR, "Failed to read from transcoder: %s\n", strerror(errno));
            return NULL;
        }
    } else {
        if (dahdip->softslin) {
            ulawtolin(pvt, res);
            pvt->f.datalen = res * 2;
        } else {
            pvt->f.datalen = res;
        }
        pvt->datalen = 0;
        pvt->f.samples = res;
        pvt->samples = 0;
        dahdip->samples_written_to_hardware =
            (dahdip->samples_written_to_hardware >= res) ?
                    dahdip->samples_written_to_hardware - res : 0;
 
        return ast_frisolate(&pvt->f);
    }
 
    /* Shouldn't get here... */
    return NULL;
}
 
 
static void dahdi_destroy(struct ast_trans_pvt *pvt)
{
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
 
    switch (dahdip->fmts.dstfmt) {
    case DAHDI_FORMAT_G729A:
    case DAHDI_FORMAT_G723_1:
        ast_atomic_fetchadd_int(&channels.encoders, -1);
        break;
    default:
        ast_atomic_fetchadd_int(&channels.decoders, -1);
        break;
    }
 
    close(dahdip->fd);
}
 
static struct ast_format *dahdi_format_to_cached(int format)
{
    switch (format) {
    case DAHDI_FORMAT_G723_1:
        return ast_format_g723;
    case DAHDI_FORMAT_GSM:
        return ast_format_gsm;
    case DAHDI_FORMAT_ULAW:
        return ast_format_ulaw;
    case DAHDI_FORMAT_ALAW:
        return ast_format_alaw;
    case DAHDI_FORMAT_G726:
        return ast_format_g726;
    case DAHDI_FORMAT_ADPCM:
        return ast_format_adpcm;
    case DAHDI_FORMAT_SLINEAR:
        return ast_format_slin;
    case DAHDI_FORMAT_LPC10:
        return ast_format_lpc10;
    case DAHDI_FORMAT_G729A:
        return ast_format_g729;
    case DAHDI_FORMAT_SPEEX:
        return ast_format_speex;
    case DAHDI_FORMAT_ILBC:
        return ast_format_ilbc;
    }
 
    /* This will never be reached */
    ast_assert(0);
    return NULL;
}
 
static int dahdi_translate(struct ast_trans_pvt *pvt, uint32_t dst_dahdi_fmt, uint32_t src_dahdi_fmt)
{
    /* Request translation through zap if possible */
    int fd;
    struct codec_dahdi_pvt *dahdip = pvt->pvt;
    int tried_once = 0;
    const char *dev_filename = "/dev/dahdi/transcode";
 
    if ((fd = open(dev_filename, O_RDWR)) < 0) {
        ast_log(LOG_ERROR, "Failed to open %s: %s\n", dev_filename, strerror(errno));
        return -1;
    }
 
    dahdip->fmts.srcfmt = src_dahdi_fmt;
    dahdip->fmts.dstfmt = dst_dahdi_fmt;
 
    ast_assert(pvt->f.subclass.format == NULL);
    pvt->f.subclass.format = ao2_bump(dahdi_format_to_cached(dahdip->fmts.dstfmt));
 
    ast_debug(1, "Opening transcoder channel from %s to %s.\n", pvt->t->src_codec.name, pvt->t->dst_codec.name);
 
retry:
    if (ioctl(fd, DAHDI_TC_ALLOCATE, &dahdip->fmts)) {
        if ((ENODEV == errno) && !tried_once) {
            /* We requested to translate to/from an unsupported
             * format.  Most likely this is because signed linear
             * was not supported by any hardware devices even
             * though this module always registers signed linear
             * support. In this case we'll retry, requesting
             * support for ULAW instead of signed linear and then
             * we'll just convert from ulaw to signed linear in
             * software. */
            if (dahdip->fmts.srcfmt == DAHDI_FORMAT_SLINEAR) {
                ast_debug(1, "Using soft_slin support on source\n");
                dahdip->softslin = 1;
                dahdip->fmts.srcfmt = DAHDI_FORMAT_ULAW;
            } else if (dahdip->fmts.dstfmt == DAHDI_FORMAT_SLINEAR) {
                ast_debug(1, "Using soft_slin support on destination\n");
                dahdip->softslin = 1;
                dahdip->fmts.dstfmt = DAHDI_FORMAT_ULAW;
            }
            tried_once = 1;
            goto retry;
        }
        ast_log(LOG_ERROR, "Unable to attach to transcoder: %s\n", strerror(errno));
        close(fd);
 
        return -1;
    }
 
    ast_fd_set_flags(fd, O_NONBLOCK);
 
    dahdip->fd = fd;
 
    dahdip->required_samples = ((dahdip->fmts.dstfmt|dahdip->fmts.srcfmt) & (DAHDI_FORMAT_G723_1)) ? G723_SAMPLES : G729_SAMPLES;
 
    switch (dahdip->fmts.dstfmt) {
    case DAHDI_FORMAT_G729A:
        ast_atomic_fetchadd_int(&channels.encoders, +1);
        break;
    case DAHDI_FORMAT_G723_1:
        ast_atomic_fetchadd_int(&channels.encoders, +1);
        break;
    default:
        ast_atomic_fetchadd_int(&channels.decoders, +1);
        break;
    }
 
    return 0;
}
 
static int dahdi_new(struct ast_trans_pvt *pvt)
{
    struct translator *zt = container_of(pvt->t, struct translator, t);
 
    return dahdi_translate(pvt, zt->dst_dahdi_fmt, zt->src_dahdi_fmt);
}
 
static struct ast_frame *fakesrc_sample(void)
{
    /* Don't bother really trying to test hardware ones. */
    static struct ast_frame f = {
        .frametype = AST_FRAME_VOICE,
        .samples = 160,
        .src = __PRETTY_FUNCTION__
    };
 
    return &f;
}
 
static bool is_encoder(uint32_t src_dahdi_fmt)
{
    return ((src_dahdi_fmt & (DAHDI_FORMAT_ULAW | DAHDI_FORMAT_ALAW | DAHDI_FORMAT_SLINEAR)) > 0);
}
 
/* Must be called with the translators list locked. */
static int register_translator(uint32_t dst_dahdi_fmt, uint32_t src_dahdi_fmt)
{
    const struct ast_codec *dst_codec;
    const struct ast_codec *src_codec;
    struct translator *zt;
    int res;
 
    dst_codec = get_dahdi_codec(dst_dahdi_fmt);
    src_codec = get_dahdi_codec(src_dahdi_fmt);
    if (!dst_codec || !src_codec) {
        return -1;
    }
 
    if (!(zt = ast_calloc(1, sizeof(*zt)))) {
        return -1;
    }
 
    zt->src_dahdi_fmt = src_dahdi_fmt;
    zt->dst_dahdi_fmt = dst_dahdi_fmt;
 
    snprintf(zt->t.name, sizeof(zt->t.name), "dahdi_%s_to_%s",
        src_codec->name, dst_codec->name);
 
    memcpy(&zt->t.src_codec, src_codec, sizeof(*src_codec));
    memcpy(&zt->t.dst_codec, dst_codec, sizeof(*dst_codec));
    zt->t.buf_size = BUFFER_SIZE;
    if (is_encoder(src_dahdi_fmt)) {
        zt->t.framein = dahdi_encoder_framein;
        zt->t.frameout = dahdi_encoder_frameout;
    } else {
        zt->t.framein = dahdi_decoder_framein;
        zt->t.frameout = dahdi_decoder_frameout;
    }
    zt->t.destroy = dahdi_destroy;
    zt->t.buffer_samples = 0;
    zt->t.newpvt = dahdi_new;
    zt->t.sample = fakesrc_sample;
    zt->t.native_plc = 0;
 
    zt->t.desc_size = sizeof(struct codec_dahdi_pvt);
    if ((res = ast_register_translator(&zt->t))) {
        ast_free(zt);
        return -1;
    }
 
    AST_LIST_INSERT_HEAD(&translators, zt, entry);
 
    return res;
}
 
static void unregister_translators(void)
{
    struct translator *cur;
 
    AST_LIST_LOCK(&translators);
    while ((cur = AST_LIST_REMOVE_HEAD(&translators, entry))) {
        ast_unregister_translator(&cur->t);
        ast_free(cur);
    }
    AST_LIST_UNLOCK(&translators);
}
 
/* Must be called with the translators list locked. */
static bool is_already_registered(uint32_t dstfmt, uint32_t srcfmt)
{
    bool res = false;
    const struct translator *zt;
 
    AST_LIST_TRAVERSE(&translators, zt, entry) {
        if ((zt->src_dahdi_fmt == srcfmt) && (zt->dst_dahdi_fmt == dstfmt)) {
            res = true;
            break;
        }
    }
 
    return res;
}
 
static void build_translators(uint32_t dstfmts, uint32_t srcfmts)
{
    uint32_t srcfmt;
    uint32_t dstfmt;
 
    AST_LIST_LOCK(&translators);
 
    for (srcfmt = 1; srcfmt != 0; srcfmt <<= 1) {
        for (dstfmt = 1; dstfmt != 0; dstfmt <<= 1) {
            if (!(dstfmts & dstfmt) || !(srcfmts & srcfmt)) {
                continue;
            }
            if (is_already_registered(dstfmt, srcfmt)) {
                continue;
            }
            register_translator(dstfmt, srcfmt);
        }
    }
 
    AST_LIST_UNLOCK(&translators);
}
 
static int find_transcoders(void)
{
    struct dahdi_transcoder_info info = { 0, };
    int fd;
 
    if ((fd = open("/dev/dahdi/transcode", O_RDWR)) < 0) {
        ast_log(LOG_ERROR, "Failed to open /dev/dahdi/transcode: %s\n", strerror(errno));
        return 0;
    }
 
    for (info.tcnum = 0; !ioctl(fd, DAHDI_TC_GETINFO, &info); info.tcnum++) {
        ast_verb(2, "Found transcoder '%s'.\n", info.name);
 
        /* Complex codecs need to support signed linear.  If the
         * hardware transcoder does not natively support signed linear
         * format, we will emulate it in software directly in this
         * module. Also, do not allow direct ulaw/alaw to complex
         * codec translation, since that will prevent the generic PLC
         * functions from working. */
        if (info.dstfmts & (DAHDI_FORMAT_ULAW | DAHDI_FORMAT_ALAW)) {
            info.dstfmts |= DAHDI_FORMAT_SLINEAR;
            info.dstfmts &= ~(DAHDI_FORMAT_ULAW | DAHDI_FORMAT_ALAW);
        }
        if (info.srcfmts & (DAHDI_FORMAT_ULAW | DAHDI_FORMAT_ALAW)) {
            info.srcfmts |= DAHDI_FORMAT_SLINEAR;
            info.srcfmts &= ~(DAHDI_FORMAT_ULAW | DAHDI_FORMAT_ALAW);
        }
 
        build_translators(info.dstfmts, info.srcfmts);
        ast_atomic_fetchadd_int(&channels.total, info.numchannels / 2);
    }
 
    close(fd);
 
    if (!info.tcnum) {
        ast_verb(2, "No hardware transcoders found.\n");
    }
 
    return 0;
}
 
static int reload(void)
{
    return AST_MODULE_LOAD_SUCCESS;
}
 
static int unload_module(void)
{
    ast_cli_unregister_multiple(cli, ARRAY_LEN(cli));
    unregister_translators();
 
    return 0;
}
 
static int load_module(void)
{
    find_transcoders();
    ast_cli_register_multiple(cli, ARRAY_LEN(cli));
    return AST_MODULE_LOAD_SUCCESS;
}
 
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Generic DAHDI Transcoder Codec Translator",
    .support_level = AST_MODULE_SUPPORT_CORE,
    .load = load_module,
    .unload = unload_module,
    .reload = reload,
);