874 lines
21 KiB
C
874 lines
21 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* DAHDI native transcoding support
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*
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* Copyright (C) 1999 - 2008, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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* Kevin P. Fleming <kpfleming@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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*
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* \brief Translate between various formats natively through DAHDI transcoding
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*
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* \ingroup codecs
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*/
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/*** MODULEINFO
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<depend>dahdi</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <stdbool.h>
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#include <poll.h>
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#include <fcntl.h>
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#include <netinet/in.h>
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#include <sys/ioctl.h>
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#include <sys/mman.h>
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#include <dahdi/user.h>
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#include "asterisk/lock.h"
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#include "asterisk/translate.h"
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#include "asterisk/config.h"
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#include "asterisk/module.h"
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#include "asterisk/cli.h"
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#include "asterisk/channel.h"
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#include "asterisk/utils.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/ulaw.h"
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#include "asterisk/format_compatibility.h"
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#define BUFFER_SIZE 8000
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#define G723_SAMPLES 240
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#define G729_SAMPLES 160
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#define ULAW_SAMPLES 160
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/* Defines from DAHDI. */
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#ifndef DAHDI_FORMAT_MAX_AUDIO
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/*! G.723.1 compression */
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#define DAHDI_FORMAT_G723_1 (1 << 0)
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/*! GSM compression */
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#define DAHDI_FORMAT_GSM (1 << 1)
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/*! Raw mu-law data (G.711) */
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#define DAHDI_FORMAT_ULAW (1 << 2)
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/*! Raw A-law data (G.711) */
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#define DAHDI_FORMAT_ALAW (1 << 3)
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/*! ADPCM (G.726, 32kbps) */
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#define DAHDI_FORMAT_G726 (1 << 4)
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/*! ADPCM (IMA) */
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#define DAHDI_FORMAT_ADPCM (1 << 5)
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/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
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#define DAHDI_FORMAT_SLINEAR (1 << 6)
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/*! LPC10, 180 samples/frame */
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#define DAHDI_FORMAT_LPC10 (1 << 7)
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/*! G.729A audio */
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#define DAHDI_FORMAT_G729A (1 << 8)
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/*! SpeeX Free Compression */
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#define DAHDI_FORMAT_SPEEX (1 << 9)
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/*! iLBC Free Compression */
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#define DAHDI_FORMAT_ILBC (1 << 10)
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#endif
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static struct channel_usage {
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int total;
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int encoders;
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int decoders;
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} channels;
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#if defined(NOT_NEEDED)
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/*!
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* \internal
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* \brief Convert DAHDI format bitfield to old Asterisk format bitfield.
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* \since 13.0.0
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*
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* \param dahdi Bitfield from DAHDI to convert.
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*
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* \note They should be the same values but they don't have to be.
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*
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* \return Old Asterisk bitfield equivalent.
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*/
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static uint64_t bitfield_dahdi2ast(unsigned dahdi)
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{
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uint64_t ast;
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switch (dahdi) {
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case DAHDI_FORMAT_G723_1:
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ast = AST_FORMAT_G723;
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break;
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case DAHDI_FORMAT_GSM:
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ast = AST_FORMAT_GSM;
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break;
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case DAHDI_FORMAT_ULAW:
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ast = AST_FORMAT_ULAW;
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break;
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case DAHDI_FORMAT_ALAW:
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ast = AST_FORMAT_ALAW;
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break;
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case DAHDI_FORMAT_G726:
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ast = AST_FORMAT_G726_AAL2;
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break;
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case DAHDI_FORMAT_ADPCM:
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ast = AST_FORMAT_ADPCM;
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break;
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case DAHDI_FORMAT_SLINEAR:
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ast = AST_FORMAT_SLIN;
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break;
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case DAHDI_FORMAT_LPC10:
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ast = AST_FORMAT_LPC10;
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break;
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case DAHDI_FORMAT_G729A:
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ast = AST_FORMAT_G729;
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break;
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case DAHDI_FORMAT_SPEEX:
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ast = AST_FORMAT_SPEEX;
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break;
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case DAHDI_FORMAT_ILBC:
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ast = AST_FORMAT_ILBC;
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break;
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default:
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ast = 0;
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break;
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}
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return ast;
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}
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#endif /* defined(NOT_NEEDED) */
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/*!
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* \internal
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* \brief Get the ast_codec by DAHDI format.
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* \since 13.0.0
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*
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* \param dahdi_fmt DAHDI specific codec identifier.
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*
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* \return Specified codec if exists otherwise NULL.
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*/
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static const struct ast_codec *get_dahdi_codec(uint32_t dahdi_fmt)
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{
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const struct ast_codec *codec;
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static const struct ast_codec dahdi_g723_1 = {
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.name = "g723",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_gsm = {
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.name = "gsm",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_ulaw = {
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.name = "ulaw",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_alaw = {
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.name = "alaw",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_g726 = {
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.name = "g726",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_adpcm = {
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.name = "adpcm",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_slinear = {
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_lpc10 = {
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.name = "lpc10",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_g729a = {
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.name = "g729",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_speex = {
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.name = "speex",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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static const struct ast_codec dahdi_ilbc = {
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.name = "ilbc",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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};
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switch (dahdi_fmt) {
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case DAHDI_FORMAT_G723_1:
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codec = &dahdi_g723_1;
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break;
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case DAHDI_FORMAT_GSM:
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codec = &dahdi_gsm;
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break;
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case DAHDI_FORMAT_ULAW:
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codec = &dahdi_ulaw;
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break;
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case DAHDI_FORMAT_ALAW:
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codec = &dahdi_alaw;
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break;
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case DAHDI_FORMAT_G726:
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codec = &dahdi_g726;
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break;
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case DAHDI_FORMAT_ADPCM:
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codec = &dahdi_adpcm;
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break;
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case DAHDI_FORMAT_SLINEAR:
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codec = &dahdi_slinear;
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break;
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case DAHDI_FORMAT_LPC10:
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codec = &dahdi_lpc10;
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break;
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case DAHDI_FORMAT_G729A:
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codec = &dahdi_g729a;
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break;
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case DAHDI_FORMAT_SPEEX:
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codec = &dahdi_speex;
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break;
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case DAHDI_FORMAT_ILBC:
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codec = &dahdi_ilbc;
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break;
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default:
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codec = NULL;
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break;
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}
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return codec;
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}
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static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
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static struct ast_cli_entry cli[] = {
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AST_CLI_DEFINE(handle_cli_transcoder_show, "Display DAHDI transcoder utilization.")
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};
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struct translator {
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struct ast_translator t;
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uint32_t src_dahdi_fmt;
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uint32_t dst_dahdi_fmt;
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AST_LIST_ENTRY(translator) entry;
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};
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#ifndef container_of
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#define container_of(ptr, type, member) \
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((type *)((char *)(ptr) - offsetof(type, member)))
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#endif
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static AST_LIST_HEAD_STATIC(translators, translator);
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struct codec_dahdi_pvt {
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int fd;
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struct dahdi_transcoder_formats fmts;
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unsigned int softslin:1;
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unsigned int fake:2;
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uint16_t required_samples;
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uint16_t samples_in_buffer;
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uint16_t samples_written_to_hardware;
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uint8_t ulaw_buffer[1024];
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};
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/* Only used by a decoder */
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static int ulawtolin(struct ast_trans_pvt *pvt, int samples)
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{
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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int i = samples;
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uint8_t *src = &dahdip->ulaw_buffer[0];
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int16_t *dst = pvt->outbuf.i16 + pvt->datalen;
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/* convert and copy in outbuf */
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while (i--) {
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*dst++ = AST_MULAW(*src++);
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}
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return 0;
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}
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/* Only used by an encoder. */
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static int lintoulaw(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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int i = f->samples;
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uint8_t *dst = &dahdip->ulaw_buffer[dahdip->samples_in_buffer];
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int16_t *src = f->data.ptr;
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if (dahdip->samples_in_buffer + i > sizeof(dahdip->ulaw_buffer)) {
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ast_log(LOG_ERROR, "Out of buffer space!\n");
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return -i;
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}
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while (i--) {
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*dst++ = AST_LIN2MU(*src++);
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}
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dahdip->samples_in_buffer += f->samples;
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return 0;
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}
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static char *handle_cli_transcoder_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
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{
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struct channel_usage copy;
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switch (cmd) {
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case CLI_INIT:
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e->command = "transcoder show";
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e->usage =
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"Usage: transcoder show\n"
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" Displays channel utilization of DAHDI transcoder(s).\n";
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return NULL;
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case CLI_GENERATE:
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return NULL;
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}
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if (a->argc != 2)
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return CLI_SHOWUSAGE;
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copy = channels;
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if (copy.total == 0)
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ast_cli(a->fd, "No DAHDI transcoders found.\n");
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else
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ast_cli(a->fd, "%d/%d encoders/decoders of %d channels are in use.\n", copy.encoders, copy.decoders, copy.total);
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return CLI_SUCCESS;
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}
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static void dahdi_write_frame(struct codec_dahdi_pvt *dahdip, const uint8_t *buffer, const ssize_t count)
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{
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int res;
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if (!count) return;
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res = write(dahdip->fd, buffer, count);
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if (-1 == res) {
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ast_log(LOG_ERROR, "Failed to write to transcoder: %s\n", strerror(errno));
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}
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if (count != res) {
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ast_log(LOG_ERROR, "Requested write of %zd bytes, but only wrote %d bytes.\n", count, res);
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}
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}
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static int dahdi_encoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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if (!f->subclass.format) {
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/* We're just faking a return for calculation purposes. */
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dahdip->fake = 2;
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pvt->samples = f->samples;
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return 0;
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}
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/* Buffer up the packets and send them to the hardware if we
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* have enough samples set up. */
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if (dahdip->softslin) {
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if (lintoulaw(pvt, f)) {
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return -1;
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}
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} else {
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/* NOTE: If softslin support is not needed, and the sample
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* size is equal to the required sample size, we wouldn't
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* need this copy operation. But at the time this was
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* written, only softslin is supported. */
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if (dahdip->samples_in_buffer + f->samples > sizeof(dahdip->ulaw_buffer)) {
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ast_log(LOG_ERROR, "Out of buffer space.\n");
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return -1;
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}
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memcpy(&dahdip->ulaw_buffer[dahdip->samples_in_buffer], f->data.ptr, f->samples);
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dahdip->samples_in_buffer += f->samples;
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}
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while (dahdip->samples_in_buffer >= dahdip->required_samples) {
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dahdi_write_frame(dahdip, dahdip->ulaw_buffer, dahdip->required_samples);
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dahdip->samples_written_to_hardware += dahdip->required_samples;
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dahdip->samples_in_buffer -= dahdip->required_samples;
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if (dahdip->samples_in_buffer) {
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/* Shift any remaining bytes down. */
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memmove(dahdip->ulaw_buffer, &dahdip->ulaw_buffer[dahdip->required_samples],
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dahdip->samples_in_buffer);
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}
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}
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pvt->samples += f->samples;
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pvt->datalen = 0;
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return -1;
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}
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static void dahdi_wait_for_packet(int fd)
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{
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struct pollfd p = {0};
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p.fd = fd;
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p.events = POLLIN;
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poll(&p, 1, 10);
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}
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static struct ast_frame *dahdi_encoder_frameout(struct ast_trans_pvt *pvt)
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{
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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int res;
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if (2 == dahdip->fake) {
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struct ast_frame frm = {
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.frametype = AST_FRAME_VOICE,
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.samples = dahdip->required_samples,
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.src = pvt->t->name,
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};
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dahdip->fake = 1;
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pvt->samples = 0;
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return ast_frisolate(&frm);
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} else if (1 == dahdip->fake) {
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dahdip->fake = 0;
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return NULL;
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}
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if (dahdip->samples_written_to_hardware >= dahdip->required_samples) {
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dahdi_wait_for_packet(dahdip->fd);
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}
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res = read(dahdip->fd, pvt->outbuf.c + pvt->datalen, pvt->t->buf_size - pvt->datalen);
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if (-1 == res) {
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if (EWOULDBLOCK == errno) {
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/* Nothing waiting... */
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return NULL;
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} else {
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ast_log(LOG_ERROR, "Failed to read from transcoder: %s\n", strerror(errno));
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return NULL;
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}
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} else {
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pvt->f.datalen = res;
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pvt->f.samples = ast_codec_samples_count(&pvt->f);
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dahdip->samples_written_to_hardware =
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(dahdip->samples_written_to_hardware >= pvt->f.samples) ?
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dahdip->samples_written_to_hardware - pvt->f.samples : 0;
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pvt->samples = 0;
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pvt->datalen = 0;
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return ast_frisolate(&pvt->f);
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}
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/* Shouldn't get here... */
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return NULL;
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}
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static int dahdi_decoder_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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if (!f->subclass.format) {
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/* We're just faking a return for calculation purposes. */
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dahdip->fake = 2;
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pvt->samples = f->samples;
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return 0;
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}
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if (!f->datalen) {
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if (f->samples != dahdip->required_samples) {
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ast_log(LOG_ERROR, "%d != %d %d\n", f->samples, dahdip->required_samples, f->datalen);
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}
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}
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dahdi_write_frame(dahdip, f->data.ptr, f->datalen);
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dahdip->samples_written_to_hardware += f->samples;
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pvt->samples += f->samples;
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pvt->datalen = 0;
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return -1;
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}
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static struct ast_frame *dahdi_decoder_frameout(struct ast_trans_pvt *pvt)
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{
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int res;
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struct codec_dahdi_pvt *dahdip = pvt->pvt;
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if (2 == dahdip->fake) {
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struct ast_frame frm = {
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.frametype = AST_FRAME_VOICE,
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.samples = dahdip->required_samples,
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.src = pvt->t->name,
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};
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dahdip->fake = 1;
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pvt->samples = 0;
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return ast_frisolate(&frm);
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} 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,
|
|
);
|