diff options
| -rw-r--r-- | testing/asterisk/APKBUILD | 14 | ||||
| -rw-r--r-- | testing/asterisk/ASTERISK-18976.patch | 142 | ||||
| -rw-r--r-- | testing/asterisk/ASTERISK-18977.patch | 63 | ||||
| -rw-r--r-- | testing/asterisk/ASTERISK-18994.patch | 28 | ||||
| -rw-r--r-- | testing/asterisk/ASTERISK-18995.patch | 358 |
5 files changed, 602 insertions, 3 deletions
diff --git a/testing/asterisk/APKBUILD b/testing/asterisk/APKBUILD index d237a45f0b..0a6c924133 100644 --- a/testing/asterisk/APKBUILD +++ b/testing/asterisk/APKBUILD @@ -2,7 +2,7 @@ # Maintainer: Timo Teras <timo.teras@iki.fi> pkgname=asterisk pkgver=10.0.0_rc2 -pkgrel=0 +pkgrel=1 pkgdesc="Asterisk: A Module Open Source PBX System" url="http://www.asterisk.org/" arch="all" @@ -11,7 +11,7 @@ depends= makedepends="autoconf automake libtool ncurses-dev popt-dev newt-dev zlib-dev postgresql-dev unixodbc-dev dahdi-tools-dev libpri-dev tar freetds-dev openssl-dev lua-dev alsa-lib-dev spandsp-dev tiff-dev - libresample sqlite-dev wget speex-dev" + libresample sqlite-dev wget speex-dev libogg-dev" install="$pkgname.pre-install $pkgname.pre-upgrade $pkgname.post-install" subpackages="$pkgname-dev $pkgname-doc $pkgname-pgsql $pkgname-odbc $pkgname-tds $pkgname-fax $pkgname-sample-config:sample $pkgname-sqlite @@ -20,6 +20,10 @@ source="http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-$ 100-uclibc-daemon.patch 101-caps-uclibc.patch 400-bug-227.patch + ASTERISK-18976.patch + ASTERISK-18977.patch + ASTERISK-18994.patch + ASTERISK-18995.patch asterisk.initd asterisk.confd asterisk.logrotate" @@ -28,7 +32,7 @@ _builddir="$srcdir/$pkgname-${pkgver/_/-}" prepare() { cd "$_builddir" - for i in ../[1-9]*.patch; do + for i in ../[1-9A]*.patch; do msg "Apply $i" patch -p1 < $i || return 1 done @@ -163,6 +167,10 @@ md5sums="72e0df43d5bcf44400dcf0014fd40a3f asterisk-10.0.0-rc2.tar.gz b00c9d98ce2ad445501248a197c6e436 100-uclibc-daemon.patch 6e1129e30c4fd2c25c86c81685a485a9 101-caps-uclibc.patch 79e9634b5054bceb3b8dc246654bb243 400-bug-227.patch +da5a3c500192dee4275aae5235d25f97 ASTERISK-18976.patch +1ddadef41aa7120e168738b6f3ed8917 ASTERISK-18977.patch +0af5e797f0a99d0f81f95e3710baf5b6 ASTERISK-18994.patch +bc6713f5434e07b79d3afdd155461d72 ASTERISK-18995.patch 0ce3219e05f55af884a7dc6b99c2b276 asterisk.initd ed31d7ba37bcf8b0346dcf8593c395f0 asterisk.confd 3e65172275684373e1a25c8a11224411 asterisk.logrotate" diff --git a/testing/asterisk/ASTERISK-18976.patch b/testing/asterisk/ASTERISK-18976.patch new file mode 100644 index 0000000000..36c613ff4f --- /dev/null +++ b/testing/asterisk/ASTERISK-18976.patch @@ -0,0 +1,142 @@ +--- a/pbx/pbx_lua.c.orig 2011-12-08 08:28:31.000000000 +0200 ++++ b/pbx/pbx_lua.c 2011-12-08 08:54:22.000000000 +0200 +@@ -94,7 +94,6 @@ + static void lua_create_application_metatable(lua_State *L); + static void lua_create_autoservice_functions(lua_State *L); + static void lua_create_hangup_function(lua_State *L); +-static void lua_detect_goto(lua_State *L); + static void lua_concat_args(lua_State *L, int start, int nargs); + + static void lua_state_destroy(void *data); +@@ -213,19 +212,10 @@ + chan = lua_touserdata(L, -1); + lua_pop(L, 1); + ++ context = ast_strdupa(chan->context); ++ exten = ast_strdupa(chan->exten); ++ priority = chan->priority; + +- lua_getfield(L, LUA_REGISTRYINDEX, "context"); +- context = ast_strdupa(lua_tostring(L, -1)); +- lua_pop(L, 1); +- +- lua_getfield(L, LUA_REGISTRYINDEX, "exten"); +- exten = ast_strdupa(lua_tostring(L, -1)); +- lua_pop(L, 1); +- +- lua_getfield(L, LUA_REGISTRYINDEX, "priority"); +- priority = lua_tointeger(L, -1); +- lua_pop(L, 1); +- + lua_concat_args(L, 2, nargs); + data = lua_tostring(L, -1); + +@@ -256,75 +246,51 @@ + return lua_error(L); + } + +- lua_detect_goto(L); +- +- return 0; +-} +- +-/*! +- * \brief Detect if a Goto or other dialplan jump has been executed and return +- * control to the pbx engine. +- */ +-static void lua_detect_goto(lua_State *L) +-{ +- struct ast_channel *chan; +- +- lua_getfield(L, LUA_REGISTRYINDEX, "channel"); +- chan = lua_touserdata(L, -1); +- lua_pop(L, 1); +- +- /* check context */ +- lua_getfield(L, LUA_REGISTRYINDEX, "context"); +- lua_pushstring(L, chan->context); +- if (!lua_equal(L, -1, -2)) { ++ if (strcmp(context, chan->context)) { ++ lua_pushstring(L, context); ++ lua_pushstring(L, chan->context); + lua_pushliteral(L, "context"); +- goto e_goto_detected; +- } +- lua_pop(L, 2); +- +- /* check exten */ +- lua_getfield(L, LUA_REGISTRYINDEX, "exten"); +- lua_pushstring(L, chan->exten); +- if (!lua_equal(L, -1, -2)) { ++ } else if (strcmp(exten, chan->exten)) { ++ lua_pushstring(L, exten); ++ lua_pushstring(L, chan->exten); + lua_pushliteral(L, "exten"); +- goto e_goto_detected; +- } +- lua_pop(L, 2); +- +- /* check priority */ +- lua_getfield(L, LUA_REGISTRYINDEX, "priority"); +- lua_pushinteger(L, chan->priority); +- if (!lua_equal(L, -1, -2)) { ++ } else if (priority != chan->priority) { ++ lua_pushinteger(L, priority); ++ lua_pushinteger(L, chan->priority); + lua_pushliteral(L, "priority"); +- goto e_goto_detected; ++ } else { ++ /* no goto - restore the original position back ++ * to lua state, in case this was a recursive dialplan ++ * call (a dialplan application re-entering dialplan) */ ++ lua_update_registry(L, context, exten, priority); ++ return 0; + } +- lua_pop(L, 2); +- return; +- +-e_goto_detected: +- /* format our debug message */ +- lua_insert(L, -3); + +- lua_pushliteral(L, " changed from "); ++ /* goto detected - construct error message */ + lua_insert(L, -3); +- +- lua_pushliteral(L, " to "); +- lua_insert(L, -2); +- +- lua_concat(L, 5); +- +- ast_debug(2, "Goto detected: %s\n", lua_tostring(L, -1)); +- lua_pop(L, 1); +- +- /* let the lua engine know it needs to return control to the pbx */ +- lua_pushinteger(L, LUA_GOTO_DETECTED); ++ ++ lua_pushliteral(L, " changed from "); ++ lua_insert(L, -3); ++ ++ lua_pushliteral(L, " to "); ++ lua_insert(L, -2); ++ ++ lua_concat(L, 5); ++ ++ ast_debug(2, "Goto detected: %s\n", lua_tostring(L, -1)); ++ lua_pop(L, 1); ++ ++ /* let the lua engine know it needs to return control to the pbx */ ++ lua_pushinteger(L, LUA_GOTO_DETECTED); + lua_error(L); ++ ++ return 0; + } + + /*! + * \brief [lua_CFunction] Used to get the value of a variable or dialplan + * function (for access from lua, don't call directly) +- * ++ * + * The value of the variable or function is returned. This function is the + * 'get()' function in the following example as would be seen in + * extensions.lua. diff --git a/testing/asterisk/ASTERISK-18977.patch b/testing/asterisk/ASTERISK-18977.patch new file mode 100644 index 0000000000..fffce032cd --- /dev/null +++ b/testing/asterisk/ASTERISK-18977.patch @@ -0,0 +1,63 @@ +--- a/res/res_musiconhold.c.orig ++++ b/res/res_musiconhold.c +@@ -158,6 +158,7 @@ + char name[MAX_MUSICCLASS]; + struct ast_format origwfmt; + struct ast_format mohwfmt; ++ int announcement; + int samples; + int sample_queue; + int pos; +@@ -173,6 +174,7 @@ + #define MOH_SORTALPHA (1 << 4) + + #define MOH_CACHERTCLASSES (1 << 5) /*!< Should we use a separate instance of MOH for each user or not */ ++#define MOH_ANNOUNCEMENT (1 << 6) + + /* Custom astobj2 flag */ + #define MOH_NOTDELETED (1 << 30) /*!< Find only records that aren't deleted? */ +@@ -183,6 +185,7 @@ + char name[MAX_MUSICCLASS]; + char dir[256]; + char args[256]; ++ char announcement[80]; + char mode[80]; + char digit; + /*! A dynamically sized array to hold the list of filenames in "files" mode */ +@@ -278,6 +281,7 @@ + } + + state->save_pos = state->pos; ++ state->announcement = 0; + + state->class = mohclass_unref(state->class, "Unreffing channel's music class upon deactivation of generator"); + } +@@ -293,6 +297,16 @@ + chan->stream = NULL; + } + ++ if (ast_test_flag(state->class, MOH_ANNOUNCEMENT) && state->announcement == 0) { ++ state->announcement = 1; ++ if (ast_openstream_full(chan, state->class->announcement, chan->language, 1)) { ++ ast_debug(1, "%s Opened announcement '%s'\n", chan->name, state->class->announcement); ++ return 0; ++ } ++ } else { ++ state->announcement = 0; ++ } ++ + if (!state->class->total_files) { + ast_log(LOG_WARNING, "No files available for class '%s'\n", state->class->name); + return -1; +@@ -1731,7 +1745,10 @@ + ast_copy_string(class->dir, var->value, sizeof(class->dir)); + else if (!strcasecmp(var->name, "application")) + ast_copy_string(class->args, var->value, sizeof(class->args)); +- else if (!strcasecmp(var->name, "digit") && (isdigit(*var->value) || strchr("*#", *var->value))) ++ else if (!strcasecmp(var->name, "announcement")) { ++ ast_copy_string(class->announcement, var->value, sizeof(class->announcement)); ++ ast_set_flag(class, MOH_ANNOUNCEMENT); ++ } else if (!strcasecmp(var->name, "digit") && (isdigit(*var->value) || strchr("*#", *var->value))) + class->digit = *var->value; + else if (!strcasecmp(var->name, "random")) + ast_set2_flag(class, ast_true(var->value), MOH_RANDOMIZE); diff --git a/testing/asterisk/ASTERISK-18994.patch b/testing/asterisk/ASTERISK-18994.patch new file mode 100644 index 0000000000..702e91375d --- /dev/null +++ b/testing/asterisk/ASTERISK-18994.patch @@ -0,0 +1,28 @@ +--- a/main/file.c.orig 2011-12-09 11:32:05.000000000 +0200 ++++ b/main/file.c 2011-12-09 11:35:22.000000000 +0200 +@@ -1005,6 +1005,7 @@ + struct ast_filestream *fs; + struct ast_filestream *vfs=NULL; + char fmt[256]; ++ off_t pos; + int seekattempt; + int res; + +@@ -1017,12 +1018,14 @@ + /* check to see if there is any data present (not a zero length file), + * done this way because there is no where for ast_openstream_full to + * return the file had no data. */ +- seekattempt = fseek(fs->f, -1, SEEK_END); ++ pos = ftello(fs->f); ++ seekattempt = fseeko(fs->f, -1, SEEK_END); + if (seekattempt && errno == EINVAL) { + /* Zero-length file, as opposed to a pipe */ + return 0; +- } else { +- ast_seekstream(fs, 0, SEEK_SET); ++ } ++ if (seekattempt == 0) { ++ fseeko(fs->f, pos, SEEK_SET); + } + + vfs = ast_openvstream(chan, filename, preflang); diff --git a/testing/asterisk/ASTERISK-18995.patch b/testing/asterisk/ASTERISK-18995.patch new file mode 100644 index 0000000000..cd144847be --- /dev/null +++ b/testing/asterisk/ASTERISK-18995.patch @@ -0,0 +1,358 @@ +--- /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 @@ ++/* ++ * Asterisk -- An open source telephony toolkit. ++ * ++ * Copyright (C) 2011, Timo Teräs ++ * ++ * 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 OGG/Speex streams. ++ * \arg File name extension: spx ++ * \ingroup formats ++ */ ++ ++/*** MODULEINFO ++ <depend>speex</depend> ++ <depend>ogg</depend> ++ <support_level>extended</support_level> ++ ***/ ++ ++#include "asterisk.h" ++ ++ASTERISK_FILE_VERSION(__FILE__, "$Revision$") ++ ++#include "asterisk/mod_format.h" ++#include "asterisk/module.h" ++ ++#include <speex/speex_header.h> ++#include <ogg/ogg.h> ++ ++#define BLOCK_SIZE 4096 /* buffer size for feeding OGG routines */ ++#define BUF_SIZE 200 ++ ++struct speex_desc { /* format specific parameters */ ++ /* structures for handling the Ogg container */ ++ ogg_sync_state oy; ++ ogg_stream_state os; ++ ogg_page og; ++ ogg_packet op; ++ ++ int format_id; ++ int serialno; ++ ++ /*! \brief Indicates whether an End of Stream condition has been detected. */ ++ int eos; ++}; ++ ++static int read_packet(struct ast_filestream *fs) ++{ ++ struct speex_desc *s = (struct speex_desc *)fs->_private; ++ char *buffer; ++ int result; ++ size_t bytes; ++ ++ while (1) { ++ /* Get one packet */ ++ result = ogg_stream_packetout(&s->os, &s->op); ++ if (result > 0) { ++ if (s->op.bytes>=5 && !memcmp(s->op.packet, "Speex", 5)) ++ s->serialno = s->os.serialno; ++ if (s->serialno == -1 || s->os.serialno != s->serialno) ++ continue; ++ return 0; ++ } ++ ++ if (result < 0) ++ ast_log(LOG_WARNING, ++ "Corrupt or missing data at this page position; continuing...\n"); ++ ++ /* No more packets left in the current page... */ ++ if (s->eos) { ++ /* No more pages left in the stream */ ++ return -1; ++ } ++ ++ while (!s->eos) { ++ /* See if OGG has any pages in it's internal buffers */ ++ result = ogg_sync_pageout(&s->oy, &s->og); ++ if (result > 0) { ++ /* Read all streams. */ ++ if (ogg_page_serialno(&s->og) != s->os.serialno) ++ ogg_stream_reset_serialno(&s->os, ogg_page_serialno(&s->og)); ++ /* Yes, OGG has more pages in it's internal buffers, ++ add the page to the stream state */ ++ result = ogg_stream_pagein(&s->os, &s->og); ++ if (result == 0) { ++ /* Yes, got a new,valid page */ ++ if (ogg_page_eos(&s->og) && ++ ogg_page_serialno(&s->og) == s->serialno) ++ s->eos = 1; ++ break; ++ } ++ ast_log(LOG_WARNING, ++ "Invalid page in the bitstream; continuing...\n"); ++ } ++ ++ if (result < 0) ++ ast_log(LOG_WARNING, ++ "Corrupt or missing data in bitstream; continuing...\n"); ++ ++ /* No, we need to read more data from the file descrptor */ ++ /* get a buffer from OGG to read the data into */ ++ buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE); ++ bytes = fread(buffer, 1, BLOCK_SIZE, fs->f); ++ ogg_sync_wrote(&s->oy, bytes); ++ if (bytes == 0) ++ s->eos = 1; ++ } ++ } ++} ++ ++/*! ++ * \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) ++{ ++ char *buffer; ++ size_t bytes; ++ struct speex_desc *s = (struct speex_desc *)fs->_private; ++ SpeexHeader *hdr = NULL; ++ int i, result; ++ ++ s->format_id = format_id; ++ s->serialno = -1; ++ ogg_sync_init(&s->oy); ++ ++ buffer = ogg_sync_buffer(&s->oy, BLOCK_SIZE); ++ bytes = fread(buffer, 1, BLOCK_SIZE, fs->f); ++ ogg_sync_wrote(&s->oy, bytes); ++ ++ result = ogg_sync_pageout(&s->oy, &s->og); ++ if (result != 1) { ++ if(bytes < BLOCK_SIZE) { ++ ast_log(LOG_ERROR, "Run out of data...\n"); ++ } else { ++ ast_log(LOG_ERROR, "Input does not appear to be an Ogg bitstream.\n"); ++ } ++ ogg_sync_clear(&s->oy); ++ return -1; ++ } ++ ++ ogg_stream_init(&s->os, ogg_page_serialno(&s->og)); ++ if (ogg_stream_pagein(&s->os, &s->og) < 0) { ++ ast_log(LOG_ERROR, "Error reading first page of Ogg bitstream data.\n"); ++ goto error; ++ } ++ ++ if (read_packet(fs) < 0) { ++ ast_log(LOG_ERROR, "Error reading initial header packet.\n"); ++ goto error; ++ } ++ ++ hdr = speex_packet_to_header((char*)s->op.packet, s->op.bytes); ++ if (memcmp(hdr->speex_string, "Speex ", 8)) { ++ ast_log(LOG_ERROR, "OGG container does not contain Speex audio!\n"); ++ goto error; ++ } ++ if (hdr->frames_per_packet != 1) { ++ ast_log(LOG_ERROR, "Only one frame-per-packet OGG/Speex files are currently supported!\n"); ++ goto error; ++ } ++ if (hdr->nb_channels != 1) { ++ ast_log(LOG_ERROR, "Only monophonic OGG/Speex files are currently supported!\n"); ++ goto error; ++ } ++ if (hdr->rate != expected_rate) { ++ ast_log(LOG_ERROR, "Unexpected sampling rate (%d != %d)!\n", ++ hdr->rate, expected_rate); ++ goto error; ++ } ++ ++ /* this packet is the comment */ ++ if (read_packet(fs) < 0) { ++ ast_log(LOG_ERROR, "Error reading comment packet.\n"); ++ goto error; ++ } ++ for (i = 0; i < hdr->extra_headers; i++) { ++ if (read_packet(fs) < 0) { ++ ast_log(LOG_ERROR, "Error reading extra header packet %d.\n", i+1); ++ goto error; ++ } ++ } ++ free(hdr); ++ ++ return 0; ++error: ++ if (hdr) ++ free(hdr); ++ ogg_stream_clear(&s->os); ++ ogg_sync_clear(&s->oy); ++ return -1; ++} ++ ++/*! ++ * \brief Close a OGG/Speex filestream. ++ * \param fs A OGG/Speex filestream. ++ */ ++static void ogg_speex_close(struct ast_filestream *fs) ++{ ++ struct speex_desc *s = (struct speex_desc *)fs->_private; ++ ++ ogg_stream_clear(&s->os); ++ ogg_sync_clear(&s->oy); ++} ++ ++/*! ++ * \brief Read a frame full of audio data from the filestream. ++ * \param fs The filestream. ++ * \param whennext Number of sample times to schedule the next call. ++ * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data. ++ */ ++static struct ast_frame *ogg_speex_read(struct ast_filestream *fs, ++ int *whennext) ++{ ++ struct speex_desc *s = (struct speex_desc *)fs->_private; ++ ++ 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); ++ ++ return &fs->fr; ++} ++ ++/*! ++ * \brief Trucate an OGG/Speex filestream. ++ * \param s The filestream to truncate. ++ * \return 0 on success, -1 on failure. ++ */ ++ ++static int ogg_speex_trunc(struct ast_filestream *s) ++{ ++ ast_log(LOG_WARNING, "Truncation is not supported on OGG/Speex streams!\n"); ++ return -1; ++} ++ ++/*! ++ * \brief Seek to a specific position in an OGG/Speex filestream. ++ * \param s The filestream to truncate. ++ * \param sample_offset New position for the filestream, measured in 8KHz samples. ++ * \param whence Location to measure ++ * \return 0 on success, -1 on failure. ++ */ ++static int ogg_speex_seek(struct ast_filestream *s, off_t sample_offset, int whence) ++{ ++ ast_log(LOG_WARNING, "Seeking is not supported on OGG/Speex streams!\n"); ++ return -1; ++} ++ ++static off_t ogg_speex_tell(struct ast_filestream *s) ++{ ++ ast_log(LOG_WARNING, "Telling is not supported on OGG/Speex streams!\n"); ++ 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, ++ .seek = ogg_speex_seek, ++ .trunc = ogg_speex_trunc, ++ .tell = ogg_speex_tell, ++ .read = ogg_speex_read, ++ .close = ogg_speex_close, ++ .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET, ++ .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, ++ .seek = ogg_speex_seek, ++ .trunc = ogg_speex_trunc, ++ .tell = ogg_speex_tell, ++ .read = ogg_speex_read, ++ .close = ogg_speex_close, ++ .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET, ++ .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, ++ .seek = ogg_speex_seek, ++ .trunc = ogg_speex_trunc, ++ .tell = ogg_speex_tell, ++ .read = ogg_speex_read, ++ .close = ogg_speex_close, ++ .buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET, ++ .desc_size = sizeof(struct speex_desc), ++}; ++ ++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); ++ ++ if (ast_format_def_register(&speex_f) || ++ ast_format_def_register(&speex16_f) || ++ ast_format_def_register(&speex32_f)) ++ return AST_MODULE_LOAD_FAILURE; ++ ++ return AST_MODULE_LOAD_SUCCESS; ++} ++ ++static int unload_module(void) ++{ ++ int res = 0; ++ res |= ast_format_def_unregister(speex_f.name); ++ res |= ast_format_def_unregister(speex16_f.name); ++ res |= ast_format_def_unregister(speex32_f.name); ++ return res; ++} ++ ++AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Speex audio", ++ .load = load_module, ++ .unload = unload_module, ++ .load_pri = AST_MODPRI_APP_DEPEND ++); |