main/asterisk: upgrade to 13.0.0

1 files changed, 15 insertions, 34 deletions

diff --git a/main/asterisk/ASTERISK-18995.patch b/main/asterisk/ASTERISK-18995.patch
index cd144847bec..26131763058 100644
--- a/main/asterisk/ASTERISK-18995.patch
+++ b/main/asterisk/ASTERISK-18995.patch
@@ -1,10 +1,10 @@
---- /dev/null	2011-11-29 09:02:40.279581283 +0200
-+++ b/formats/format_ogg_speex.c	2011-12-08 15:57:12.000000000 +0200
-@@ -0,0 +1,355 @@
+--- /dev/null	2014-10-31 08:01:35.193329595 -0200
++++ asterisk-13.0.0/formats/format_ogg_speex.c	2014-10-31 09:19:34.010493106 -0200
+@@ -0,0 +1,336 @@
 +/*
 + * Asterisk -- An open source telephony toolkit.
 + *
-+ * Copyright (C) 2011, Timo Teräs
++ * Copyright (C) 2011-2014, Timo Teräs
 + *
 + * See http://www.asterisk.org for more information about
 + * the Asterisk project. Please do not directly contact
@@ -36,6 +36,7 @@
 +
 +#include "asterisk/mod_format.h"
 +#include "asterisk/module.h"
++#include "asterisk/format_cache.h"
 +
 +#include <speex/speex_header.h>
 +#include <ogg/ogg.h>
@@ -50,7 +51,6 @@
 +	ogg_page og;
 +	ogg_packet op;
 +
-+	int format_id;
 +	int serialno;
 +
 +	/*! \brief Indicates whether an End of Stream condition has been detected. */
@@ -124,18 +124,17 @@
 +/*!
 + * \brief Create a new OGG/Speex filestream and set it up for reading.
 + * \param fs File that points to on disk storage of the OGG/Speex data.
-+ * \param expected_rate The expected Speex format (sampling rate).
 + * \return The new filestream.
 + */
-+static int ogg_speex_open(struct ast_filestream *fs, int format_id, int expected_rate)
++static int ogg_speex_open(struct ast_filestream *fs)
 +{
 +	char *buffer;
 +	size_t bytes;
 +	struct speex_desc *s = (struct speex_desc *)fs->_private;
 +	SpeexHeader *hdr = NULL;
-+	int i, result;
++	int i, result, expected_rate;
 +
-+	s->format_id = format_id;
++	expected_rate = ast_format_get_sample_rate(fs->fmt->format);
 +	s->serialno = -1;
 +	ogg_sync_init(&s->oy);
 +
@@ -232,13 +231,10 @@
 +	if (read_packet(fs) < 0)
 +		return NULL;
 +
-+	fs->fr.frametype = AST_FRAME_VOICE;
-+	ast_format_set(&fs->fr.subclass.format, s->format_id, 0);
-+	fs->fr.mallocd = 0;
 +	AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
 +	memcpy(fs->fr.data.ptr, s->op.packet, s->op.bytes);
 +	fs->fr.datalen = s->op.bytes;
-+	fs->fr.samples = *whennext = ast_codec_get_samples(&fs->fr);
++	fs->fr.samples = *whennext = ast_codec_samples_count(&fs->fr);
 +
 +	return &fs->fr;
 +}
@@ -274,15 +270,10 @@
 +	return -1;
 +}
 +
-+static int ogg_speex_open_nb(struct ast_filestream *fs) 
-+{
-+	return ogg_speex_open(fs, AST_FORMAT_SPEEX, 8000);
-+}
-+
 +static struct ast_format_def speex_f = {
 +	.name = "ogg_speex",
 +	.exts = "spx",
-+	.open = ogg_speex_open_nb,
++	.open = ogg_speex_open,
 +	.seek = ogg_speex_seek,
 +	.trunc = ogg_speex_trunc,
 +	.tell = ogg_speex_tell,
@@ -292,15 +283,10 @@
 +	.desc_size = sizeof(struct speex_desc),
 +};
 +
-+static int ogg_speex_open_wb(struct ast_filestream *fs) 
-+{
-+	return ogg_speex_open(fs, AST_FORMAT_SPEEX16, 16000);
-+}
-+
 +static struct ast_format_def speex16_f = {
 +	.name = "ogg_speex16",
 +	.exts = "spx16",
-+	.open = ogg_speex_open_wb,
++	.open = ogg_speex_open,
 +	.seek = ogg_speex_seek,
 +	.trunc = ogg_speex_trunc,
 +	.tell = ogg_speex_tell,
@@ -310,15 +296,10 @@
 +	.desc_size = sizeof(struct speex_desc),
 +};
 +
-+static int ogg_speex_open_uwb(struct ast_filestream *fs) 
-+{
-+	return ogg_speex_open(fs, AST_FORMAT_SPEEX32, 32000);
-+}
-+
 +static struct ast_format_def speex32_f = {
 +	.name = "ogg_speex32",
 +	.exts = "spx32",
-+	.open = ogg_speex_open_uwb,
++	.open = ogg_speex_open,
 +	.seek = ogg_speex_seek,
 +	.trunc = ogg_speex_trunc,
 +	.tell = ogg_speex_tell,
@@ -330,9 +311,9 @@
 +
 +static int load_module(void)
 +{
-+	ast_format_set(&speex_f.format, AST_FORMAT_SPEEX, 0);
-+	ast_format_set(&speex16_f.format, AST_FORMAT_SPEEX16, 0);
-+	ast_format_set(&speex32_f.format, AST_FORMAT_SPEEX32, 0);
++	speex_f.format = ast_format_speex;
++	speex16_f.format = ast_format_speex16;
++	speex32_f.format = ast_format_speex32;
 +
 +	if (ast_format_def_register(&speex_f) ||
 +	    ast_format_def_register(&speex16_f) ||