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
275 lines
7.7 KiB
C
275 lines
7.7 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@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 Network broadcast sound support channel driver
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>nbs</depend>
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<support_level>extended</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_REGISTER_FILE()
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#include <sys/socket.h>
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#include <sys/time.h>
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#include <arpa/inet.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include <nbs.h>
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#include "asterisk/lock.h"
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#include "asterisk/channel.h"
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#include "asterisk/config.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/format_cache.h"
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static const char tdesc[] = "Network Broadcast Sound Driver";
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static char context[AST_MAX_EXTENSION] = "default";
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static const char type[] = "NBS";
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/* NBS creates private structures on demand */
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struct nbs_pvt {
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NBS *nbs;
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struct ast_channel *owner; /* Channel we belong to, possibly NULL */
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char app[16]; /* Our app */
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char stream[80]; /* Our stream */
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struct ast_module_user *u; /*! for holding a reference to this module */
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};
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static struct ast_channel *nbs_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
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static int nbs_call(struct ast_channel *ast, const char *dest, int timeout);
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static int nbs_hangup(struct ast_channel *ast);
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static struct ast_frame *nbs_xread(struct ast_channel *ast);
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static int nbs_xwrite(struct ast_channel *ast, struct ast_frame *frame);
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static struct ast_channel_tech nbs_tech = {
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.type = type,
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.description = tdesc,
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.requester = nbs_request,
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.call = nbs_call,
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.hangup = nbs_hangup,
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.read = nbs_xread,
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.write = nbs_xwrite,
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};
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static int nbs_call(struct ast_channel *ast, const char *dest, int timeout)
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{
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struct nbs_pvt *p;
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p = ast_channel_tech_pvt(ast);
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if ((ast_channel_state(ast) != AST_STATE_DOWN) && (ast_channel_state(ast) != AST_STATE_RESERVED)) {
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ast_log(LOG_WARNING, "nbs_call called on %s, neither down nor reserved\n", ast_channel_name(ast));
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return -1;
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}
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/* When we call, it just works, really, there's no destination... Just
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ring the phone and wait for someone to answer */
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ast_debug(1, "Calling %s on %s\n", dest, ast_channel_name(ast));
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/* If we can't connect, return congestion */
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if (nbs_connect(p->nbs)) {
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ast_log(LOG_WARNING, "NBS Connection failed on %s\n", ast_channel_name(ast));
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ast_queue_control(ast, AST_CONTROL_CONGESTION);
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} else {
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ast_setstate(ast, AST_STATE_RINGING);
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ast_queue_control(ast, AST_CONTROL_ANSWER);
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}
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return 0;
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}
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static void nbs_destroy(struct nbs_pvt *p)
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{
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if (p->nbs)
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nbs_delstream(p->nbs);
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ast_module_user_remove(p->u);
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ast_free(p);
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}
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static struct nbs_pvt *nbs_alloc(const char *data)
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{
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struct nbs_pvt *p;
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int flags = 0;
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char stream[256];
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char *opts;
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ast_copy_string(stream, data, sizeof(stream));
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if ((opts = strchr(stream, ':'))) {
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*opts = '\0';
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opts++;
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} else
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opts = "";
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p = ast_calloc(1, sizeof(*p));
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if (p) {
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if (!ast_strlen_zero(opts)) {
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if (strchr(opts, 'm'))
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flags |= NBS_FLAG_MUTE;
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if (strchr(opts, 'o'))
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flags |= NBS_FLAG_OVERSPEAK;
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if (strchr(opts, 'e'))
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flags |= NBS_FLAG_EMERGENCY;
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if (strchr(opts, 'O'))
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flags |= NBS_FLAG_OVERRIDE;
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} else
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flags = NBS_FLAG_OVERSPEAK;
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ast_copy_string(p->stream, stream, sizeof(p->stream));
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p->nbs = nbs_newstream("asterisk", stream, flags);
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if (!p->nbs) {
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ast_log(LOG_WARNING, "Unable to allocate new NBS stream '%s' with flags %d\n", stream, flags);
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ast_free(p);
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p = NULL;
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} else {
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/* Set for 8000 hz mono, 640 samples */
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nbs_setbitrate(p->nbs, 8000);
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nbs_setchannels(p->nbs, 1);
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nbs_setblocksize(p->nbs, 640);
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nbs_setblocking(p->nbs, 0);
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}
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}
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return p;
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}
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static int nbs_hangup(struct ast_channel *ast)
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{
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struct nbs_pvt *p;
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p = ast_channel_tech_pvt(ast);
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ast_debug(1, "nbs_hangup(%s)\n", ast_channel_name(ast));
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if (!ast_channel_tech_pvt(ast)) {
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ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
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return 0;
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}
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nbs_destroy(p);
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ast_channel_tech_pvt_set(ast, NULL);
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ast_setstate(ast, AST_STATE_DOWN);
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return 0;
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}
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static struct ast_frame *nbs_xread(struct ast_channel *ast)
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{
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ast_debug(1, "Returning null frame on %s\n", ast_channel_name(ast));
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return &ast_null_frame;
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}
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static int nbs_xwrite(struct ast_channel *ast, struct ast_frame *frame)
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{
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struct nbs_pvt *p = ast_channel_tech_pvt(ast);
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if (ast_channel_state(ast) != AST_STATE_UP) {
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/* Don't try tos end audio on-hook */
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return 0;
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}
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if (nbs_write(p->nbs, frame->data.ptr, frame->datalen / 2) < 0)
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return -1;
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return 0;
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}
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static struct ast_channel *nbs_new(struct nbs_pvt *i, int state, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
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{
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struct ast_channel *tmp;
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tmp = ast_channel_alloc(1, state, 0, 0, "", "s", context, assignedids, requestor, 0, "NBS/%s", i->stream);
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if (tmp) {
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ast_channel_tech_set(tmp, &nbs_tech);
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ast_channel_set_fd(tmp, 0, nbs_fd(i->nbs));
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ast_channel_nativeformats_set(tmp, nbs_tech.capabilities);
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ast_channel_set_rawreadformat(tmp, ast_format_slin);
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ast_channel_set_rawwriteformat(tmp, ast_format_slin);
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ast_channel_set_writeformat(tmp, ast_format_slin);
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ast_channel_set_readformat(tmp, ast_format_slin);
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if (state == AST_STATE_RING)
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ast_channel_rings_set(tmp, 1);
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ast_channel_tech_pvt_set(tmp, i);
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ast_channel_context_set(tmp, context);
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ast_channel_exten_set(tmp, "s");
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ast_channel_language_set(tmp, "");
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i->owner = tmp;
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i->u = ast_module_user_add(tmp);
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ast_channel_unlock(tmp);
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if (state != AST_STATE_DOWN) {
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if (ast_pbx_start(tmp)) {
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ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
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ast_hangup(tmp);
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}
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}
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} else
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ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
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return tmp;
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}
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static struct ast_channel *nbs_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
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{
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struct nbs_pvt *p;
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struct ast_channel *tmp = NULL;
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if (ast_format_cap_iscompatible_format(cap, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
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struct ast_str *cap_buf = ast_str_alloca(64);
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ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%s'\n",
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ast_format_cap_get_names(cap, &cap_buf));
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return NULL;
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}
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p = nbs_alloc(data);
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if (p) {
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tmp = nbs_new(p, AST_STATE_DOWN, assignedids, requestor);
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if (!tmp)
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nbs_destroy(p);
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}
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return tmp;
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}
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static int unload_module(void)
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{
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/* First, take us out of the channel loop */
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ast_channel_unregister(&nbs_tech);
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ao2_ref(nbs_tech.capabilities, -1);
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nbs_tech.capabilities = NULL;
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return 0;
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}
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static int load_module(void)
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{
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if (!(nbs_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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return AST_MODULE_LOAD_FAILURE;
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}
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ast_format_cap_append(nbs_tech.capabilities, ast_format_slin, 0);
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/* Make sure we can register our channel type */
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if (ast_channel_register(&nbs_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Network Broadcast Sound Support");
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