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ff0523a3d0e706bec9c799946f932477def45453
asterisk/res/res_audiosocket.c
Sven Kube ff0523a3d0 res_audiosocket: add message types for all slin sample rates 
Extend audiosocket messages with types 0x11 - 0x18 to create asterisk
frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 format, enabling the transmission of audio at a higher sample
rates. For audiosocket messages sent by Asterisk, the message kind is
determined by the format of the originating asterisk frame.

UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
slin192 audio. External applications using audiosocket may need to be
updated to support these message types if the audiosocket channel is
created with one of these audio formats.
2025-10-17 13:05:34 +00:00

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2019, CyCore Systems, Inc
*
* Seán C McCord <scm@cycoresys.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 AudioSocket support for Asterisk
*
* \author Seán C McCord <scm@cycoresys.com>
*
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
#include <arpa/inet.h> /* For byte-order conversion. */
#include "asterisk/file.h"
#include "asterisk/res_audiosocket.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/uuid.h"
#include "asterisk/format_cache.h"
#define MODULE_DESCRIPTION "AudioSocket support functions for Asterisk"
#define MAX_CONNECT_TIMEOUT_MSEC 2000
/*!
* \internal
* \brief Attempt to complete the audiosocket connection.
*
* \param server Url that we are trying to connect to.
* \param addr Address that host was resolved to.
* \param netsockfd File descriptor of socket.
*
* \retval 0 when connection is succesful.
* \retval 1 when there is an error.
*/
static int handle_audiosocket_connection(const char *server,
const struct ast_sockaddr addr, const int netsockfd)
{
struct pollfd pfds[1];
int res, conresult;
socklen_t reslen;
reslen = sizeof(conresult);
pfds[0].fd = netsockfd;
pfds[0].events = POLLOUT;
while ((res = ast_poll(pfds, 1, MAX_CONNECT_TIMEOUT_MSEC)) != 1) {
if (errno != EINTR) {
if (!res) {
ast_log(LOG_WARNING, "AudioSocket connection to '%s' timed"
"out after MAX_CONNECT_TIMEOUT_MSEC (%d) milliseconds.\n",
server, MAX_CONNECT_TIMEOUT_MSEC);
} else {
ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", server,
strerror(errno));
}
return -1;
}
}
if (getsockopt(pfds[0].fd, SOL_SOCKET, SO_ERROR, &conresult, &reslen) < 0) {
ast_log(LOG_WARNING, "Connection to '%s' failed with error: %s\n",
ast_sockaddr_stringify(&addr), strerror(errno));
return -1;
}
if (conresult) {
ast_log(LOG_WARNING, "Connecting to '%s' failed for url '%s': %s\n",
ast_sockaddr_stringify(&addr), server, strerror(conresult));
return -1;
}
return 0;
}
const int ast_audiosocket_connect(const char *server, struct ast_channel *chan)
{
int s = -1;
struct ast_sockaddr *addrs = NULL;
int num_addrs = 0, i = 0;
if (chan && ast_autoservice_start(chan) < 0) {
ast_log(LOG_WARNING, "Failed to start autoservice for channel "
"'%s'\n", ast_channel_name(chan));
goto end;
}
if (ast_strlen_zero(server)) {
ast_log(LOG_ERROR, "No AudioSocket server provided\n");
goto end;
}
if (!(num_addrs = ast_sockaddr_resolve(&addrs, server, PARSE_PORT_REQUIRE,
AST_AF_UNSPEC))) {
ast_log(LOG_ERROR, "Failed to resolve AudioSocket service using '%s' - "
"requires a valid hostname and port\n", server);
goto end;
}
/* Connect to AudioSocket service */
for (i = 0; i < num_addrs; i++) {
if (!ast_sockaddr_port(&addrs[i])) {
/* If there's no port, other addresses should have the
* same problem. Stop here.
*/
ast_log(LOG_ERROR, "No port provided for '%s'\n",
ast_sockaddr_stringify(&addrs[i]));
s = -1;
goto end;
}
if ((s = ast_socket_nonblock(addrs[i].ss.ss_family, SOCK_STREAM,
IPPROTO_TCP)) < 0) {
ast_log(LOG_WARNING, "Unable to create socket: '%s'\n", strerror(errno));
continue;
}
/*
* Disable Nagle's algorithm by setting the TCP_NODELAY socket option.
* This reduces latency by preventing delays caused by packet buffering.
*/
if (setsockopt(s, IPPROTO_TCP, TCP_NODELAY, &(int){1}, sizeof(int)) < 0) {
ast_log(LOG_ERROR, "Failed to set TCP_NODELAY on AudioSocket: %s\n", strerror(errno));
}
if (ast_connect(s, &addrs[i]) && errno == EINPROGRESS) {
if (handle_audiosocket_connection(server, addrs[i], s)) {
close(s);
continue;
}
} else {
ast_log(LOG_ERROR, "Connection to '%s' failed with unexpected error: %s\n",
ast_sockaddr_stringify(&addrs[i]), strerror(errno));
close(s);
s = -1;
}
break;
}
end:
if (addrs) {
ast_free(addrs);
}
if (chan && ast_autoservice_stop(chan) < 0) {
ast_log(LOG_WARNING, "Failed to stop autoservice for channel '%s'\n",
ast_channel_name(chan));
close(s);
return -1;
}
if (i == num_addrs) {
ast_log(LOG_ERROR, "Failed to connect to AudioSocket service\n");
close(s);
return -1;
}
return s;
}
const int ast_audiosocket_init(const int svc, const char *id)
{
uuid_t uu;
int ret = 0;
uint8_t buf[3 + 16];
if (ast_strlen_zero(id)) {
ast_log(LOG_ERROR, "No UUID for AudioSocket\n");
return -1;
}
if (uuid_parse(id, uu)) {
ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", id);
return -1;
}
buf[0] = AST_AUDIOSOCKET_KIND_UUID;
buf[1] = 0x00;
buf[2] = 0x10;
memcpy(buf + 3, uu, 16);
if (write(svc, buf, 3 + 16) != 3 + 16) {
ast_log(LOG_WARNING, "Failed to write data to AudioSocket because: %s\n", strerror(errno));
ret = -1;
}
return ret;
}
const int ast_audiosocket_send_frame(const int svc, const struct ast_frame *f)
{
int datalen = f->datalen;
if (f->frametype == AST_FRAME_DTMF) {
datalen = 1;
}
{
uint8_t buf[3 + datalen];
uint16_t *length = (uint16_t *) &buf[1];
/* Audio format is 16-bit, 8kHz signed linear mono for dialplan app,
depends on agreed upon audio codec for channel driver interface. */
switch (f->frametype) {
case AST_FRAME_VOICE:
if (ast_format_cmp(f->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin12) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN12;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin16) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN16;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin24) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN24;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin32) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN32;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin44) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN44;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin48) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN48;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin96) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN96;
} else if (ast_format_cmp(f->subclass.format, ast_format_slin192) == AST_FORMAT_CMP_EQUAL) {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO_SLIN192;
} else {
buf[0] = AST_AUDIOSOCKET_KIND_AUDIO;
}
*length = htons(datalen);
memcpy(&buf[3], f->data.ptr, datalen);
break;
case AST_FRAME_DTMF:
buf[0] = AST_AUDIOSOCKET_KIND_DTMF;
buf[3] = (uint8_t) f->subclass.integer;
*length = htons(1);
break;
default:
ast_log(LOG_ERROR, "Unsupported frame type %d for AudioSocket\n", f->frametype);
return -1;
}
if (write(svc, buf, 3 + datalen) != 3 + datalen) {
ast_log(LOG_WARNING, "Failed to write data to AudioSocket because: %s\n", strerror(errno));
return -1;
}
}
return 0;
}
struct ast_frame *ast_audiosocket_receive_frame(const int svc)
{
return ast_audiosocket_receive_frame_with_hangup(svc, NULL);
}
struct ast_frame *ast_audiosocket_receive_frame_with_hangup(const int svc,
int *const hangup)
{
int i = 0, n = 0, ret = 0;
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
.src = "AudioSocket",
.mallocd = AST_MALLOCD_DATA,
};
uint8_t header[3];
uint8_t *kind = &header[0];
uint16_t *length = (uint16_t *) &header[1];
uint8_t *data;
if (hangup) {
*hangup = 0;
}
n = read(svc, &header, 3);
if (n == -1) {
ast_log(LOG_WARNING, "Failed to read header from AudioSocket because: %s\n", strerror(errno));
return NULL;
}
if (n == 0 || *kind == AST_AUDIOSOCKET_KIND_HANGUP) {
/* Socket closure or requested hangup. */
if (hangup) {
*hangup = 1;
}
return NULL;
}
switch (*kind) {
case AST_AUDIOSOCKET_KIND_AUDIO:
f.subclass.format = ast_format_slin;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN12:
f.subclass.format = ast_format_slin12;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN16:
f.subclass.format = ast_format_slin16;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN24:
f.subclass.format = ast_format_slin24;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN32:
f.subclass.format = ast_format_slin32;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN44:
f.subclass.format = ast_format_slin44;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN48:
f.subclass.format = ast_format_slin48;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN96:
f.subclass.format = ast_format_slin96;
break;
case AST_AUDIOSOCKET_KIND_AUDIO_SLIN192:
f.subclass.format = ast_format_slin192;
break;
default:
ast_log(LOG_ERROR, "Received AudioSocket message other than hangup or audio, refer to protocol specification for valid message types\n");
return NULL;
}
/* Swap endianess of length if needed. */
*length = ntohs(*length);
if (*length < 1) {
ast_log(LOG_ERROR, "Invalid message length received from AudioSocket server. \n");
return NULL;
}
data = ast_malloc(*length);
if (!data) {
ast_log(LOG_ERROR, "Failed to allocate for data from AudioSocket\n");
return NULL;
}
ret = 0;
n = 0;
i = 0;
while (i < *length) {
n = read(svc, data + i, *length - i);
if (n == -1) {
if (errno == EAGAIN || errno == EWOULDBLOCK) {
int poll_result = ast_wait_for_input(svc, 5);
if (poll_result == 1) {
continue;
} else if (poll_result == 0) {
ast_log(LOG_WARNING, "Poll timed out while waiting for data\n");
} else {
ast_log(LOG_WARNING, "Poll error: %s\n", strerror(errno));
}
}
ast_log(LOG_ERROR, "Failed to read payload from AudioSocket: %s\n", strerror(errno));
ret = -1;
break;
}
if (n == 0) {
ast_log(LOG_ERROR, "Insufficient payload read from AudioSocket\n");
ret = -1;
break;
}
i += n;
}
if (ret != 0) {
ast_free(data);
return NULL;
}
f.data.ptr = data;
f.datalen = *length;
f.samples = *length / 2;
/* The frame steals data, so it doesn't need to be freed here */
return ast_frisolate(&f);
}
static int load_module(void)
{
ast_verb(5, "Loading AudioSocket Support module\n");
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_verb(5, "Unloading AudioSocket Support module\n");
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "AudioSocket support",
.support_level = AST_MODULE_SUPPORT_EXTENDED,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);
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