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
261 lines
7.6 KiB
C
261 lines
7.6 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, 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 Playback a file with audio detect
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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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 "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/utils.h"
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#include "asterisk/dsp.h"
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#include "asterisk/app.h"
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#include "asterisk/format.h"
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#include "asterisk/format_cache.h"
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/*** DOCUMENTATION
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<application name="BackgroundDetect" language="en_US">
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<synopsis>
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Background a file with talk detect.
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</synopsis>
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<syntax>
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<parameter name="filename" required="true" />
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<parameter name="sil">
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<para>If not specified, defaults to <literal>1000</literal>.</para>
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</parameter>
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<parameter name="min">
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<para>If not specified, defaults to <literal>100</literal>.</para>
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</parameter>
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<parameter name="max">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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<parameter name="analysistime">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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</syntax>
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<description>
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<para>Plays back <replaceable>filename</replaceable>, waiting for interruption from a given digit (the digit
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must start the beginning of a valid extension, or it will be ignored). During
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the playback of the file, audio is monitored in the receive direction, and if
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a period of non-silence which is greater than <replaceable>min</replaceable> ms yet less than
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<replaceable>max</replaceable> ms is followed by silence for at least <replaceable>sil</replaceable> ms,
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which occurs during the first <replaceable>analysistime</replaceable> ms, then the audio playback is
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aborted and processing jumps to the <replaceable>talk</replaceable> extension, if available.</para>
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</description>
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</application>
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***/
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static char *app = "BackgroundDetect";
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static int background_detect_exec(struct ast_channel *chan, const char *data)
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{
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int res = 0;
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char *tmp;
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struct ast_frame *fr;
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int notsilent = 0;
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struct timeval start = { 0, 0 };
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struct timeval detection_start = { 0, 0 };
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int sil = 1000;
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int min = 100;
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int max = -1;
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int analysistime = -1;
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int continue_analysis = 1;
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int x;
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RAII_VAR(struct ast_format *, origrformat, NULL, ao2_cleanup);
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struct ast_dsp *dsp = NULL;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(filename);
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AST_APP_ARG(silence);
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AST_APP_ARG(min);
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AST_APP_ARG(max);
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AST_APP_ARG(analysistime);
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);
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if (ast_strlen_zero(data)) {
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ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
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return -1;
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}
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tmp = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, tmp);
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if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%30d", &x) == 1) && (x > 0)) {
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sil = x;
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}
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if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%30d", &x) == 1) && (x > 0)) {
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min = x;
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}
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if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%30d", &x) == 1) && (x > 0)) {
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max = x;
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}
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if (!ast_strlen_zero(args.analysistime) && (sscanf(args.analysistime, "%30d", &x) == 1) && (x > 0)) {
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analysistime = x;
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}
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ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d, analysistime=%d\n", args.filename, sil, min, max, analysistime);
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do {
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if (ast_channel_state(chan) != AST_STATE_UP) {
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if ((res = ast_answer(chan))) {
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break;
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}
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}
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origrformat = ao2_bump(ast_channel_readformat(chan));
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if ((ast_set_read_format(chan, ast_format_slin))) {
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ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
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res = -1;
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break;
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}
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if (!(dsp = ast_dsp_new())) {
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ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
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res = -1;
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break;
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}
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ast_stopstream(chan);
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if (ast_streamfile(chan, tmp, ast_channel_language(chan))) {
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ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", ast_channel_name(chan), (char *)data);
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break;
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}
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detection_start = ast_tvnow();
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while (ast_channel_stream(chan)) {
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res = ast_sched_wait(ast_channel_sched(chan));
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if ((res < 0) && !ast_channel_timingfunc(chan)) {
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res = 0;
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break;
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}
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if (res < 0) {
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res = 1000;
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}
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res = ast_waitfor(chan, res);
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if (res < 0) {
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ast_log(LOG_WARNING, "Waitfor failed on %s\n", ast_channel_name(chan));
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break;
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} else if (res > 0) {
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fr = ast_read(chan);
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if (continue_analysis && analysistime >= 0) {
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/* If we have a limit for the time to analyze voice
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* frames and the time has not expired */
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if (ast_tvdiff_ms(ast_tvnow(), detection_start) >= analysistime) {
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continue_analysis = 0;
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ast_verb(3, "BackgroundDetect: Talk analysis time complete on %s.\n", ast_channel_name(chan));
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}
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}
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if (!fr) {
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res = -1;
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break;
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} else if (fr->frametype == AST_FRAME_DTMF) {
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char t[2];
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t[0] = fr->subclass.integer;
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t[1] = '\0';
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if (ast_canmatch_extension(chan, ast_channel_context(chan), t, 1,
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S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
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/* They entered a valid extension, or might be anyhow */
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res = fr->subclass.integer;
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ast_frfree(fr);
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break;
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}
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} else if ((fr->frametype == AST_FRAME_VOICE) &&
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(ast_format_cmp(fr->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) && continue_analysis) {
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int totalsilence;
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int ms;
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res = ast_dsp_silence(dsp, fr, &totalsilence);
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if (res && (totalsilence > sil)) {
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/* We've been quiet a little while */
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if (notsilent) {
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/* We had heard some talking */
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ms = ast_tvdiff_ms(ast_tvnow(), start);
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ms -= sil;
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if (ms < 0)
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ms = 0;
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if ((ms > min) && ((max < 0) || (ms < max))) {
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char ms_str[12];
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ast_debug(1, "Found qualified token of %d ms\n", ms);
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/* Save detected talk time (in milliseconds) */
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snprintf(ms_str, sizeof(ms_str), "%d", ms);
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pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
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ast_goto_if_exists(chan, ast_channel_context(chan), "talk", 1);
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res = 0;
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ast_frfree(fr);
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break;
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} else {
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ast_debug(1, "Found unqualified token of %d ms\n", ms);
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}
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notsilent = 0;
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}
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} else {
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if (!notsilent) {
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/* Heard some audio, mark the begining of the token */
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start = ast_tvnow();
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ast_debug(1, "Start of voice token!\n");
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notsilent = 1;
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}
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}
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}
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ast_frfree(fr);
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}
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ast_sched_runq(ast_channel_sched(chan));
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}
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ast_stopstream(chan);
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} while (0);
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if (res > -1) {
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if (origrformat && ast_set_read_format(chan, origrformat)) {
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ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
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ast_channel_name(chan), ast_format_get_name(origrformat));
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}
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}
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if (dsp) {
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ast_dsp_free(dsp);
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}
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return res;
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}
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static int unload_module(void)
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{
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return ast_unregister_application(app);
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}
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static int load_module(void)
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{
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return ast_register_application_xml(app, background_detect_exec);
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}
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AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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