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[core] Add SMBF_PAUSE media bug flag to pause an individual media bug.

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* [core] Add SMBF_PAUSE media bug flag to pause an individual media bug.
[core] Add switch_ivr_record_session_pause() to pause a session recording.
[mod_dptools] Add record_session_pause and record_session_resume dialplan APPs.
[mod_commands] Add pause/resume sub-commands to uuid_record API.
This commit is contained in:
Chris Rienzo
2020-12-11 18:20:33 -05:00
committed by Andrey Volk
parent a162cf86e8
commit 60c4b62254
11 changed files with 232 additions and 14 deletions
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5
tests/unit/.gitignore vendored
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@@ -17,9 +17,12 @@ switch_core
switch_core_codec
switch_core_db
switch_core_file
switch_core_session
switch_core_video
switch_eavesdrop
switch_event
switch_hash
switch_ivr_async
switch_ivr_originate
switch_ivr_play_say
switch_packetizer
@@ -37,7 +40,9 @@ switch_vpx
switch_xml
switch_estimators
switch_jitter_buffer
test_sofia
.deps/
Makefile
conf/*/
conf_playsay/*/
conf_async/*/

2
tests/unit/Makefile.am
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@@ -2,7 +2,7 @@ include $(top_srcdir)/build/modmake.rulesam
noinst_PROGRAMS = switch_event switch_hash switch_ivr_originate switch_utils switch_core switch_console switch_vpx switch_core_file \
switch_ivr_play_say switch_core_codec switch_rtp switch_xml
noinst_PROGRAMS += switch_core_video switch_core_db switch_vad switch_packetizer test_sofia switch_core_asr switch_core_session
noinst_PROGRAMS += switch_core_video switch_core_db switch_vad switch_packetizer test_sofia switch_ivr_async switch_core_asr switch_core_session
AM_LDFLAGS += -avoid-version -no-undefined $(SWITCH_AM_LDFLAGS) $(openssl_LIBS)
AM_LDFLAGS += $(FREESWITCH_LIBS) $(switch_builddir)/libfreeswitch.la $(CORE_LIBS) $(APR_LIBS)

47
tests/unit/conf_async/freeswitch.xml Normal file
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@@ -0,0 +1,47 @@
<?xml version="1.0"?>
<document type="freeswitch/xml">
<X-PRE-PROCESS cmd="exec-set" data="test=echo 1234"/>
<X-PRE-PROCESS cmd="set" data="default_password=$${test}"/>
<section name="configuration" description="Various Configuration">
<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_loopback"/>
<load module="mod_opus"/>
<load module="mod_commands"/>
<load module="mod_sndfile"/>
<load module="mod_dptools"/>
<load module="mod_tone_stream"/>
<load module="mod_test"/>
</modules>
</configuration>
<configuration name="console.conf" description="Console Logger">
<mappings>
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
<settings>
<param name="colorize" value="true"/>
<param name="loglevel" value="debug"/>
</settings>
</configuration>
<configuration name="timezones.conf" description="Timezones">
<timezones>
<zone name="GMT" value="GMT0" />
</timezones>
</configuration>
<X-PRE-PROCESS cmd="include" data="vpx.conf.xml"/>
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<context name="default">
<extension name="sample">
<condition>
<action application="info"/>
</condition>
</extension>
</context>
</section>
</document>

124
tests/unit/switch_ivr_async.c Normal file
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@@ -0,0 +1,124 @@
/*
* FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
* Copyright (C) 2005-2020, Anthony Minessale II <anthm@freeswitch.org>
*
* Version: MPL 1.1
*
* The contents of this file are subject to the Mozilla Public License Version
* 1.1 (the "License"); you may not use this file except in compliance with
* the License. You may obtain a copy of the License at
* http://www.mozilla.org/MPL/
*
* Software distributed under the License is distributed on an "AS IS" basis,
* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
* for the specific language governing rights and limitations under the
* License.
*
* The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
*
* The Initial Developer of the Original Code is
* Anthony Minessale II <anthm@freeswitch.org>
* Portions created by the Initial Developer are Copyright (C)
* the Initial Developer. All Rights Reserved.
*
* Contributor(s):
* Chris Rienzo <chris@signalwire.com>
*
*
* switch_ivr_async.c -- Async IVR tests
*
*/
#include <switch.h>
#include <stdlib.h>
#include <test/switch_test.h>
static switch_status_t partial_play_and_collect_input_callback(switch_core_session_t *session, void *input, switch_input_type_t input_type, void *data, __attribute__((unused))unsigned int len)
{
switch_status_t status = SWITCH_STATUS_SUCCESS;
int *count = (int *)data;
if (input_type == SWITCH_INPUT_TYPE_EVENT) {
switch_event_t *event = (switch_event_t *)input;
if (event->event_id == SWITCH_EVENT_DETECTED_SPEECH) {
const char *speech_type = switch_event_get_header(event, "Speech-Type");
if (zstr(speech_type) || strcmp(speech_type, "detected-partial-speech")) {
return status;
}
(*count)++;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "partial events count: %d\n", *count);
char *body = switch_event_get_body(event);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "body=[%s]\n", body);
}
} else if (input_type == SWITCH_INPUT_TYPE_DTMF) {
// never mind
}
return status;
}
FST_CORE_BEGIN("./conf_async")
{
FST_SUITE_BEGIN(switch_ivr_play_async)
{
FST_SETUP_BEGIN()
{
fst_requires_module("mod_tone_stream");
fst_requires_module("mod_sndfile");
fst_requires_module("mod_dptools");
fst_requires_module("mod_test");
}
FST_SETUP_END()
FST_TEARDOWN_BEGIN()
{
}
FST_TEARDOWN_END()
FST_SESSION_BEGIN(session_record_pause)
{
const char *record_filename = switch_core_session_sprintf(fst_session, "%s%s%s.wav", SWITCH_GLOBAL_dirs.temp_dir, SWITCH_PATH_SEPARATOR, switch_core_session_get_uuid(fst_session));
switch_status_t status;
status = switch_ivr_record_session_event(fst_session, record_filename, 0, NULL, NULL);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_record_session() to return SWITCH_STATUS_SUCCESS");
status = switch_ivr_play_file(fst_session, NULL, "tone_stream://%(400,200,400,450);%(400,2000,400,450)", NULL);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_play_file() to return SWITCH_STATUS_SUCCESS");
status = switch_ivr_record_session_pause(fst_session, record_filename, SWITCH_TRUE);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_record_session_pause(SWITCH_TRUE) to return SWITCH_STATUS_SUCCESS");
switch_ivr_play_file(fst_session, NULL, "silence_stream://1000,0", NULL);
status = switch_ivr_record_session_pause(fst_session, record_filename, SWITCH_FALSE);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_record_session_pause(SWITCH_FALSE) to return SWITCH_STATUS_SUCCESS");
status = switch_ivr_play_file(fst_session, NULL, "tone_stream://%(400,200,400,450)", NULL);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_play_file() to return SWITCH_STATUS_SUCCESS");
status = switch_ivr_stop_record_session(fst_session, record_filename);
fst_xcheck(status == SWITCH_STATUS_SUCCESS, "Expect switch_ivr_stop_record_session() to return SWITCH_STATUS_SUCCESS");
switch_ivr_play_file(fst_session, NULL, "silence_stream://100,0", NULL);
fst_xcheck(switch_file_exists(record_filename, fst_pool) == SWITCH_STATUS_SUCCESS, "Expect recording file to exist");
unlink(record_filename);
const char *duration_ms_str = switch_channel_get_variable(fst_channel, "record_ms");
fst_requires(duration_ms_str != NULL);
int duration_ms = atoi(duration_ms_str);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(fst_session), SWITCH_LOG_NOTICE, "Recording duration is %s ms\n", duration_ms_str);
fst_xcheck(duration_ms > 3500 && duration_ms < 3700, "Expect recording to be between 3500 and 3700 ms");
}
FST_SESSION_END()
}
FST_SUITE_END()
}
FST_CORE_END()