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siren: Add format attribute modules for Siren7 and Siren14.

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This change removes hardcoded SDP parsing and generation for
Siren7 and Siren14 from chan_sip and moves it to format attribute
modules so it can also be used by chan_pjsip.

With this the fmtp lines for both are added with the bitrate
information.

ASTERISK-26021

Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037
This commit is contained in:
Joshua Colp
2016-06-22 13:19:32 -03:00
parent 46cc7f114d
commit 75818b4084
3 changed files with 189 additions and 25 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
channels/chan_sip.c
View File

@@ -11332,25 +11332,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
}
if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 32000) {
ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
} else {
found = TRUE;
}
}
} else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 48000) {
ast_log(LOG_WARNING, "Got Siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
} else {
found = TRUE;
}
}
} else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 64000) {
ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate);
@@ -13009,12 +12991,6 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
} else if (ast_format_cmp(format, ast_format_g723) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
} else if (ast_format_cmp(format, ast_format_siren7) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we only expect 32Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
} else if (ast_format_cmp(format, ast_format_siren14) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we only expect 48Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
} else if (ast_format_cmp(format, ast_format_g719) == AST_FORMAT_CMP_EQUAL) {
/* Indicate that we only expect 64Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);

94
res/res_format_attr_siren14.c Normal file
View File

@@ -0,0 +1,94 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2016, Digium, Inc.
*
* Joshua Colp <jcolp@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 Siren14 format attribute interface
*
* \author Joshua Colp <jcolp@digium.com>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#include "asterisk/module.h"
#include "asterisk/format.h"
/* Destroy is a required callback and must exist */
static void siren14_destroy(struct ast_format *format)
{
}
/* Clone is a required callback and must exist */
static int siren14_clone(const struct ast_format *src, struct ast_format *dst)
{
return 0;
}
static struct ast_format *siren14_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
{
unsigned int val;
if (sscanf(attributes, "bitrate=%30u", &val) == 1) {
if (val != 48000) {
ast_log(LOG_WARNING, "Got siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", val);
return NULL;
}
}
/* We aren't modifying the format and once passed back it won't be touched, so use what we were given */
return ao2_bump((struct ast_format *)format);
}
static void siren14_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
{
ast_str_append(str, 0, "a=fmtp:%u bitrate=48000\r\n", payload);
}
static struct ast_format_interface siren14_interface = {
.format_destroy = siren14_destroy,
.format_clone = siren14_clone,
.format_parse_sdp_fmtp = siren14_parse_sdp_fmtp,
.format_generate_sdp_fmtp = siren14_generate_sdp_fmtp,
};
static int load_module(void)
{
if (ast_format_interface_register("siren14", &siren14_interface)) {
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Siren14 Format Attribute Module",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);

94
res/res_format_attr_siren7.c Normal file
View File

@@ -0,0 +1,94 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2016, Digium, Inc.
*
* Joshua Colp <jcolp@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 Siren7 format attribute interface
*
* \author Joshua Colp <jcolp@digium.com>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#include "asterisk/module.h"
#include "asterisk/format.h"
/* Destroy is a required callback and must exist */
static void siren7_destroy(struct ast_format *format)
{
}
/* Clone is a required callback and must exist */
static int siren7_clone(const struct ast_format *src, struct ast_format *dst)
{
return 0;
}
static struct ast_format *siren7_parse_sdp_fmtp(const struct ast_format *format, const char *attributes)
{
unsigned int val;
if (sscanf(attributes, "bitrate=%30u", &val) == 1) {
if (val != 32000) {
ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", val);
return NULL;
}
}
/* We aren't modifying the format and once passed back it won't be touched, so use what we were given */
return ao2_bump((struct ast_format *)format);
}
static void siren7_generate_sdp_fmtp(const struct ast_format *format, unsigned int payload, struct ast_str **str)
{
ast_str_append(str, 0, "a=fmtp:%u bitrate=32000\r\n", payload);
}
static struct ast_format_interface siren7_interface = {
.format_destroy = siren7_destroy,
.format_clone = siren7_clone,
.format_parse_sdp_fmtp = siren7_parse_sdp_fmtp,
.format_generate_sdp_fmtp = siren7_generate_sdp_fmtp,
};
static int load_module(void)
{
if (ast_format_interface_register("siren7", &siren7_interface)) {
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Siren7 Format Attribute Module",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
);