asterisk/res/ari/resource_sounds.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00

227 lines
6.2 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2012 - 2013, Digium, Inc.
*
* David M. Lee, II <dlee@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 /api-docs/sounds.{format} implementation- Sound resources
*
* \author David M. Lee, II <dlee@digium.com>
*/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#include "resource_sounds.h"
#include "asterisk/media_index.h"
#include "asterisk/sounds_index.h"
#include "asterisk/format.h"
#include "asterisk/format_cap.h"
#include "asterisk/json.h"
/*! \brief arguments that are necessary for adding format/lang pairs */
struct lang_format_info {
struct ast_json *format_list; /*!< The embedded array to which format/lang pairs should be added */
const char *filename; /*!< Name of the file for which to add format/lang pairs */
const char *format_filter; /*!< Format filter provided in the request */
};
/*! \brief Add format/lang pairs to the array embedded in the sound object */
static int add_format_information_cb(void *obj, void *arg, int flags)
{
char *language = obj;
struct lang_format_info *args = arg;
int idx;
RAII_VAR(struct ast_format_cap *, cap, NULL, ao2_cleanup);
RAII_VAR(struct ast_media_index *, sounds_index, ast_sounds_get_index(), ao2_cleanup);
if (!sounds_index) {
return CMP_STOP;
}
cap = ast_media_get_format_cap(sounds_index, args->filename, language);
if (!cap) {
return CMP_STOP;
}
for (idx = 0; idx < ast_format_cap_count(cap); idx++) {
struct ast_format *format = ast_format_cap_get_format(cap, idx);
struct ast_json *lang_format_pair;
if (!ast_strlen_zero(args->format_filter)
&& strcmp(args->format_filter, ast_format_get_name(format))) {
ao2_ref(format, -1);
continue;
}
lang_format_pair = ast_json_pack("{s: s, s: s}",
"language", language,
"format", ast_format_get_name(format));
if (!lang_format_pair) {
ao2_ref(format, -1);
return CMP_STOP;
}
ast_json_array_append(args->format_list, lang_format_pair);
ao2_ref(format, -1);
}
return 0;
}
/*! \brief Filter out all languages not matching the specified language */
static int filter_langs_cb(void *obj, void *arg, int flags)
{
char *lang_filter = arg;
char *lang = obj;
if (strcmp(lang, lang_filter)) {
return CMP_MATCH;
}
return 0;
}
/*! \brief Generate a Sound structure as documented in sounds.json for the specified filename */
static struct ast_json *create_sound_blob(const char *filename,
struct ast_ari_sounds_list_args *args)
{
RAII_VAR(struct ast_json *, sound, NULL, ast_json_unref);
RAII_VAR(struct ao2_container *, languages, NULL, ao2_cleanup);
const char *description;
struct ast_json *format_lang_list;
struct lang_format_info info;
RAII_VAR(struct ast_media_index *, sounds_index, ast_sounds_get_index(), ao2_cleanup);
if (!sounds_index) {
return NULL;
}
description = ast_media_get_description(sounds_index, filename, "en");
if (ast_strlen_zero(description)) {
sound = ast_json_pack("{s: s, s: []}",
"id", filename,
"formats");
} else {
sound = ast_json_pack("{s: s, s: s, s: []}",
"id", filename,
"text", description,
"formats");
}
if (!sound) {
return NULL;
}
format_lang_list = ast_json_object_get(sound, "formats");
if (!format_lang_list) {
return NULL;
}
languages = ast_media_get_variants(sounds_index, filename);
if (!languages || !ao2_container_count(languages)) {
return NULL;
}
/* filter requested languages */
if (args && !ast_strlen_zero(args->lang)) {
char *lang_filter = ast_strdupa(args->lang);
ao2_callback(languages, OBJ_NODATA | OBJ_MULTIPLE | OBJ_UNLINK, filter_langs_cb, lang_filter);
if (!languages || !ao2_container_count(languages)) {
return NULL;
}
}
info.filename = filename;
info.format_list = format_lang_list;
info.format_filter = NULL;
if (args) {
info.format_filter = args->format;
}
ao2_callback(languages, OBJ_NODATA, add_format_information_cb, &info);
/* no format/lang pairs for this sound so nothing to return */
if (!ast_json_array_size(format_lang_list)) {
return NULL;
}
return ast_json_ref(sound);
}
/*! \brief Generate a Sound structure and append it to the output blob */
static int append_sound_cb(void *obj, void *arg, void *data, int flags)
{
struct ast_json *sounds_array = arg;
char *filename = obj;
struct ast_ari_sounds_list_args *args = data;
struct ast_json *sound_blob = create_sound_blob(filename, args);
if (!sound_blob) {
return 0;
}
ast_json_array_append(sounds_array, sound_blob);
return 0;
}
void ast_ari_sounds_list(struct ast_variable *headers,
struct ast_ari_sounds_list_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct ao2_container *, sound_files, NULL, ao2_cleanup);
struct ast_json *sounds_blob;
RAII_VAR(struct ast_media_index *, sounds_index, ast_sounds_get_index(), ao2_cleanup);
if (!sounds_index) {
ast_ari_response_error(response, 500, "Internal Error", "Sounds index not available");
return;
}
sound_files = ast_media_get_media(sounds_index);
if (!sound_files) {
ast_ari_response_error(response, 500, "Internal Error", "Allocation Error");
return;
}
sounds_blob = ast_json_array_create();
if (!sounds_blob) {
ast_ari_response_error(response, 500, "Internal Error", "Allocation Error");
return;
}
ao2_callback_data(sound_files, OBJ_NODATA, append_sound_cb, sounds_blob, args);
if (!ast_json_array_size(sounds_blob)) {
ast_ari_response_error(response, 404, "Not Found", "No sounds found that matched the query");
return;
}
ast_ari_response_ok(response, sounds_blob);
}
void ast_ari_sounds_get(struct ast_variable *headers,
struct ast_ari_sounds_get_args *args,
struct ast_ari_response *response)
{
struct ast_json *sound_blob;
sound_blob = create_sound_blob(args->sound_id, NULL);
if (!sound_blob) {
ast_ari_response_error(response, 404, "Not Found", "Sound not found");
return;
}
ast_ari_response_ok(response, sound_blob);
}