asterisk/codecs/codec_dahdi.c

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2023-05-25 18:45:57 +00:00
/*
* 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,
);