Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2008, Anthony Minessale and Digium, Inc.
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* Anthony Minessale (anthmct@yahoo.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 ITU G.722.1 (Siren7, licensed from Polycom) format, 32kbps bitrate only
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* \arg File name extensions: siren7
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* \ingroup formats
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*/
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2011-07-14 20:28:54 +00:00
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/mod_format.h"
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#include "asterisk/module.h"
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#include "asterisk/endian.h"
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#define BUF_SIZE 80 /* 20 milliseconds == 80 bytes, 320 samples */
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#define SAMPLES_TO_BYTES(x) x / (320 / 80)
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#define BYTES_TO_SAMPLES(x) x * (320 / 80)
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static struct ast_frame *siren7read(struct ast_filestream *s, int *whennext)
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{
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int res;
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/* Send a frame from the file to the appropriate channel */
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s->fr.frametype = AST_FRAME_VOICE;
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2011-02-03 16:22:10 +00:00
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ast_format_set(&s->fr.subclass.format, AST_FORMAT_SIREN7, 0);
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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s->fr.mallocd = 0;
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AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
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if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
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if (res)
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ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
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return NULL;
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}
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*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
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return &s->fr;
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}
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static int siren7write(struct ast_filestream *fs, struct ast_frame *f)
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{
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int res;
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if (f->frametype != AST_FRAME_VOICE) {
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ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
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return -1;
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}
|
2011-02-03 16:22:10 +00:00
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if (f->subclass.format.id != AST_FORMAT_SIREN7) {
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ast_log(LOG_WARNING, "Asked to write non-Siren7 frame (%s)!\n", ast_getformatname(&f->subclass.format));
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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return -1;
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}
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if ((res = fwrite(f->data.ptr, 1, f->datalen, fs->f)) != f->datalen) {
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ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
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return -1;
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}
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return 0;
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}
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static int siren7seek(struct ast_filestream *fs, off_t sample_offset, int whence)
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{
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off_t offset = 0, min = 0, cur, max;
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sample_offset = SAMPLES_TO_BYTES(sample_offset);
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2012-04-16 20:17:03 +00:00
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if ((cur = ftello(fs->f)) < 0) {
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ast_log(AST_LOG_WARNING, "Unable to determine current position in siren7 filestream %p: %s\n", fs, strerror(errno));
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return -1;
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}
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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2012-04-16 20:17:03 +00:00
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if (fseeko(fs->f, 0, SEEK_END) < 0) {
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ast_log(AST_LOG_WARNING, "Unable to seek to end of siren7 filestream %p: %s\n", fs, strerror(errno));
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return -1;
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}
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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2012-04-17 18:29:51 +00:00
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if ((max = ftello(fs->f)) < 0) {
|
2012-04-16 20:17:03 +00:00
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ast_log(AST_LOG_WARNING, "Unable to determine max position in siren7 filestream %p: %s\n", fs, strerror(errno));
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return -1;
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}
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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if (whence == SEEK_SET)
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offset = sample_offset;
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else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
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offset = sample_offset + cur;
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else if (whence == SEEK_END)
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offset = max - sample_offset;
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if (whence != SEEK_FORCECUR)
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offset = (offset > max) ? max : offset;
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/* always protect against seeking past begining. */
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offset = (offset < min) ? min : offset;
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return fseeko(fs->f, offset, SEEK_SET);
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}
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static int siren7trunc(struct ast_filestream *fs)
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{
|
2012-04-16 20:17:03 +00:00
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int fd;
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off_t cur;
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if ((fd = fileno(fs->f)) < 0) {
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ast_log(AST_LOG_WARNING, "Unable to determine file descriptor for siren7 filestream %p: %s\n", fs, strerror(errno));
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return -1;
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}
|
2012-04-29 19:50:57 +00:00
|
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|
if ((cur = ftello(fs->f)) < 0) {
|
2012-04-16 20:17:03 +00:00
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ast_log(AST_LOG_WARNING, "Unable to determine current position in siren7 filestream %p: %s\n", fs, strerror(errno));
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return -1;
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}
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/* Truncate file to current length */
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return ftruncate(fd, cur);
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
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}
|
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static off_t siren7tell(struct ast_filestream *fs)
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{
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return BYTES_TO_SAMPLES(ftello(fs->f));
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}
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2011-02-03 16:22:10 +00:00
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|
static struct ast_format_def siren7_f = {
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
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.name = "siren7",
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.exts = "siren7",
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.write = siren7write,
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.seek = siren7seek,
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.trunc = siren7trunc,
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.tell = siren7tell,
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.read = siren7read,
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|
|
.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
|
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|
};
|
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|
static int load_module(void)
|
|
|
|
{
|
2011-02-03 16:22:10 +00:00
|
|
|
ast_format_set(&siren7_f.format, AST_FORMAT_SIREN7, 0);
|
|
|
|
if (ast_format_def_register(&siren7_f))
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
|
|
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int unload_module(void)
|
|
|
|
{
|
2011-02-03 16:22:10 +00:00
|
|
|
return ast_format_def_unregister(siren7_f.name);
|
2010-07-20 19:35:02 +00:00
|
|
|
}
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Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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2010-07-26 03:28:02 +00:00
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ITU G.722.1 (Siren7, licensed from Polycom)",
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_APP_DEPEND
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);
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