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: http://svn.digium.com/svn/asterisk/trunk@175508 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 277 insertions, 0 deletions

diff --git a/formats/format_siren14.c b/formats/format_siren14.c
new file mode 100644
index 000000000..109c586fb
--- /dev/null
+++ b/formats/format_siren14.c
@@ -0,0 +1,139 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Anthony Minessale and Digium, Inc.
+ * Anthony Minessale (anthmct@yahoo.com)
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief ITU G.722.1 Annex C (Siren14, licensed from Polycom) format, 48kbps bitrate only
+ * \arg File name extensions: siren14
+ * \ingroup formats
+ */
+ 
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+#include "asterisk/endian.h"
+
+#define BUF_SIZE	120		/* 20 milliseconds == 120 bytes, 640 samples */
+#define SAMPLES_TO_BYTES(x)	((typeof(x)) x / ((float) 640 / 120))
+#define BYTES_TO_SAMPLES(x)	((typeof(x)) x * ((float) 640 / 120))
+
+static struct ast_frame *siren14read(struct ast_filestream *s, int *whennext)
+{
+	int res;
+	/* Send a frame from the file to the appropriate channel */
+
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_SIREN14;
+	s->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		return NULL;
+	}
+	*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
+	ast_log(LOG_DEBUG, "Read frame of %d bytes and %d samples\n", res, s->fr.samples);
+	return &s->fr;
+}
+
+static int siren14write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_SIREN14) {
+		ast_log(LOG_WARNING, "Asked to write non-Siren14 frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if ((res = fwrite(f->data.ptr, 1, f->datalen, fs->f)) != f->datalen) {
+		ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+		return -1;
+	}
+	return 0;
+}
+
+static int siren14seek(struct ast_filestream *fs, off_t sample_offset, int whence)
+{
+	off_t offset = 0, min = 0, cur, max;
+
+	sample_offset = SAMPLES_TO_BYTES(sample_offset);
+
+	cur = ftello(fs->f);
+
+	fseeko(fs->f, 0, SEEK_END);
+
+	max = ftello(fs->f);
+
+	if (whence == SEEK_SET)
+		offset = sample_offset;
+	else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
+		offset = sample_offset + cur;
+	else if (whence == SEEK_END)
+		offset = max - sample_offset;
+
+	if (whence != SEEK_FORCECUR)
+		offset = (offset > max) ? max : offset;
+
+	/* always protect against seeking past begining. */
+	offset = (offset < min) ? min : offset;
+
+	return fseeko(fs->f, offset, SEEK_SET);
+}
+
+static int siren14trunc(struct ast_filestream *fs)
+{
+	return ftruncate(fileno(fs->f), ftello(fs->f));
+}
+
+static off_t siren14tell(struct ast_filestream *fs)
+{
+	return BYTES_TO_SAMPLES(ftello(fs->f));
+}
+
+static const struct ast_format siren14_f = {
+	.name = "siren14",
+	.exts = "siren14",
+	.format = AST_FORMAT_SIREN14,
+	.write = siren14write,
+	.seek = siren14seek,
+	.trunc = siren14trunc,
+	.tell = siren14tell,
+	.read = siren14read,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+};
+
+static int load_module(void)
+{
+	if (ast_format_register(&siren14_f))
+		return AST_MODULE_LOAD_DECLINE;
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_unregister(siren14_f.name);
+}	
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "ITU G.722.1 Annex C (Siren14, licensed from Polycom)");
diff --git a/formats/format_siren7.c b/formats/format_siren7.c
new file mode 100644
index 000000000..0b61a72b2
--- /dev/null
+++ b/formats/format_siren7.c
@@ -0,0 +1,138 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Anthony Minessale and Digium, Inc.
+ * Anthony Minessale (anthmct@yahoo.com)
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief ITU G.722.1 (Siren7, licensed from Polycom) format, 32kbps bitrate only
+ * \arg File name extensions: siren7
+ * \ingroup formats
+ */
+ 
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+#include "asterisk/endian.h"
+
+#define BUF_SIZE	80		/* 20 milliseconds == 80 bytes, 320 samples */
+#define SAMPLES_TO_BYTES(x)	x / (320 / 80)
+#define BYTES_TO_SAMPLES(x)	x * (320 / 80)
+
+static struct ast_frame *siren7read(struct ast_filestream *s, int *whennext)
+{
+	int res;
+	/* Send a frame from the file to the appropriate channel */
+
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_SIREN7;
+	s->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		return NULL;
+	}
+	*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
+	return &s->fr;
+}
+
+static int siren7write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_SIREN7) {
+		ast_log(LOG_WARNING, "Asked to write non-Siren7 frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if ((res = fwrite(f->data.ptr, 1, f->datalen, fs->f)) != f->datalen) {
+		ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+		return -1;
+	}
+	return 0;
+}
+
+static int siren7seek(struct ast_filestream *fs, off_t sample_offset, int whence)
+{
+	off_t offset = 0, min = 0, cur, max;
+
+	sample_offset = SAMPLES_TO_BYTES(sample_offset);
+
+	cur = ftello(fs->f);
+
+	fseeko(fs->f, 0, SEEK_END);
+
+	max = ftello(fs->f);
+
+	if (whence == SEEK_SET)
+		offset = sample_offset;
+	else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
+		offset = sample_offset + cur;
+	else if (whence == SEEK_END)
+		offset = max - sample_offset;
+
+	if (whence != SEEK_FORCECUR)
+		offset = (offset > max) ? max : offset;
+
+	/* always protect against seeking past begining. */
+	offset = (offset < min) ? min : offset;
+
+	return fseeko(fs->f, offset, SEEK_SET);
+}
+
+static int siren7trunc(struct ast_filestream *fs)
+{
+	return ftruncate(fileno(fs->f), ftello(fs->f));
+}
+
+static off_t siren7tell(struct ast_filestream *fs)
+{
+	return BYTES_TO_SAMPLES(ftello(fs->f));
+}
+
+static const struct ast_format siren7_f = {
+	.name = "siren7",
+	.exts = "siren7",
+	.format = AST_FORMAT_SIREN7,
+	.write = siren7write,
+	.seek = siren7seek,
+	.trunc = siren7trunc,
+	.tell = siren7tell,
+	.read = siren7read,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+};
+
+static int load_module(void)
+{
+	if (ast_format_register(&siren7_f))
+		return AST_MODULE_LOAD_DECLINE;
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_unregister(siren7_f.name);
+}	
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "ITU G.722.1 (Siren7, licensed from Polycom)");