asterisk/codecs/codec_dahdi.c
Shaun Ruffell c4d7e7e270 codec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec.
This fixes a Segmentation fault introduced in r419044 "media formats: re-architect
handling of media for performance improvements".

The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the
ast_translator structure when these fields were never set. Now instead of trying to map
the new core codec descriptions to the way DAHDI defines different codecs, we will store
the DAHDI specific formats in 'struct translator' directly so we can refer to them without
mapping.

This also allows us to remove the "global_format_map" structure, since we can now query
the list of translators directly to make sure we do not ever register a DAHDI based
translator for a specific path more than once and eliminate the need to keep the list and
the map in sync.

ASTERISK-24435 #close
Reported by: Marian Koniuszko

Review: https://reviewboard.asterisk.org/r/4105/
........

Merged revisions 426097 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-22 21:41:31 +00:00

881 lines
21 KiB
C

/*
* 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
<support_level>core</support_level>
<depend>dahdi</depend>
***/
#include "asterisk.h"
#include <stdbool.h>
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <fcntl.h>
#include <netinet/in.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <sys/poll.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 flags;
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;
}
flags = fcntl(fd, F_GETFL);
if (flags > - 1) {
if (fcntl(fd, F_SETFL, flags | O_NONBLOCK))
ast_log(LOG_WARNING, "Could not set non-block mode!\n");
}
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,
);