[asterisk-commits] mjordan: branch 12 r414934 - in /branches/12: ./ funcs/ include/asterisk/ mai...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 30 07:39:43 CDT 2014
Author: mjordan
Date: Fri May 30 07:39:36 2014
New Revision: 414934
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=414934
Log:
TALK_DETECT: A channel function that raises events when talking is detected
This patch adds a new channel function TALK_DETECT that, when set on a
channel, causes events indicating the start/stop of talking on a channel to be
emitted to both AMI and ARI clients.
The function allows setting both the silence threshold (the length of silence
after which we decide no one is talking) as well as the talking threshold (the
amount of energy that counts as talking). Parameters can be updated on a channel
after talk detection has been enabled, and talk detection can be removed at
any time.
The events raised by the function use a nomenclature similar to existing AMI/ARI
events.
For AMI: ChannelTalkingStart/ChannelTalkingStop
For ARI: ChannelTalkingStarted/ChannelTalkingFinished
Review: https://reviewboard.asterisk.org/r/3563/
ASTERISK-23786 #close
Reported by: Matt Jordan
Added:
branches/12/funcs/func_talkdetect.c (with props)
Modified:
branches/12/CHANGES
branches/12/include/asterisk/stasis_channels.h
branches/12/main/audiohook.c
branches/12/main/stasis_channels.c
branches/12/res/ari/ari_model_validators.c
branches/12/res/ari/ari_model_validators.h
branches/12/rest-api/api-docs/events.json
Modified: branches/12/CHANGES
URL: http://svnview.digium.com/svn/asterisk/branches/12/CHANGES?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/CHANGES (original)
+++ branches/12/CHANGES Fri May 30 07:39:36 2014
@@ -18,6 +18,26 @@
connect with an incoming caller after being alerted to the presence
of the incoming caller. The most likely reason this would happen is
the agent did not acknowledge the call in time.
+
+AMI
+------------------
+ * New events have been added for the TALK_DETECT function. When the function
+ is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
+ emitted to connected AMI clients indicating the start/stop of talking on
+ the channel.
+
+ARI
+------------------
+ * New event models have been aded for the TALK_DETECT function. When the
+ function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
+ events will be emitted to connected WebSockets subscribed to the channel,
+ indicating the start/stop of talking on the channel.
+
+Functions
+------------------
+ * A new function, TALK_DETECT, has been added. When set on a channel, this
+ fucntion causes events indicating the starting/stoping of talking on said
+ channel to be emitted to both AMI and ARI clients.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
Added: branches/12/funcs/func_talkdetect.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/funcs/func_talkdetect.c?view=auto&rev=414934
==============================================================================
--- branches/12/funcs/func_talkdetect.c (added)
+++ branches/12/funcs/func_talkdetect.c Fri May 30 07:39:36 2014
@@ -1,0 +1,404 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2014, Digium, Inc.
+ *
+ * Matt Jordan <mjordan at 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 Function that raises events when talking is detected on a channel
+ *
+ * \author Matt Jordan <mjordan at digium.com>
+ *
+ * \ingroup functions
+ */
+
+/*** MODULEINFO
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/app.h"
+#include "asterisk/dsp.h"
+#include "asterisk/audiohook.h"
+#include "asterisk/stasis.h"
+#include "asterisk/stasis_channels.h"
+
+/*** DOCUMENTATION
+ <function name="TALK_DETECT" language="en_US">
+ <synopsis>
+ Raises notifications when Asterisk detects silence or talking on a channel.
+ </synopsis>
+ <syntax>
+ <parameter name="action" required="true">
+ <optionlist>
+ <option name="remove">
+ <para>W/O. Remove talk detection from the channel.</para>
+ </option>
+ <option name="set">
+ <para>W/O. Enable TALK_DETECT and/or configure talk detection
+ parameters. Can be called multiple times to change parameters
+ on a channel with talk detection already enabled.</para>
+ <argument name="dsp_silence_threshold" required="false">
+ <para>The time in milliseconds before which a user is considered silent.</para>
+ </argument>
+ <argument name="dsp_talking_threshold" required="false">
+ <para>The time in milliseconds after which a user is considered talking.</para>
+ </argument>
+ </option>
+ </optionlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>The TALK_DETECT function enables events on the channel
+ it is applied to. These events can be emited over AMI, ARI, and
+ potentially other Asterisk modules that listen for the internal
+ notification.</para>
+ <para>The function has two parameters that can optionally be passed
+ when <literal>set</literal> on a channel: <replaceable>dsp_talking_threshold</replaceable>
+ and <replaceable>dsp_silence_threshold</replaceable>.</para>
+ <para><replaceable>dsp_talking_threshold</replaceable> is the time in milliseconds of sound
+ above what the dsp has established as base line silence for a user
+ before a user is considered to be talking. By default, the value of
+ <replaceable>silencethreshold</replaceable> from <filename>dsp.conf</filename>
+ is used. If this value is set too tight events may be
+ falsely triggered by variants in room noise.</para>
+ <para>Valid values are 1 through 2^31.</para>
+ <para><replaceable>dsp_silence_threshold</replaceable> is the time in milliseconds of sound
+ falling within what the dsp has established as baseline silence before
+ a user is considered be silent. If this value is set too low events
+ indicating the user has stopped talking may get falsely sent out when
+ the user briefly pauses during mid sentence.</para>
+ <para>The best way to approach this option is to set it slightly above
+ the maximum amount of ms of silence a user may generate during
+ natural speech.</para>
+ <para>By default this value is 2500ms. Valid values are 1
+ through 2^31.</para>
+ <para>Example:</para>
+ <para>same => n,Set(TALK_DETECT(set)=) ; Enable talk detection</para>
+ <para>same => n,Set(TALK_DETECT(set)=1200) ; Update existing talk detection's silence threshold to 1200 ms</para>
+ <para>same => n,Set(TALK_DETECT(remove)=) ; Remove talk detection</para>
+ <para>same => n,Set(TALK_DETECT(set)=,128) ; Enable and set talk threshold to 128</para>
+ <para>This function will set the following variables:</para>
+ <note>
+ <para>The TALK_DETECT function uses an audiohook to inspect the
+ voice media frames on a channel. Other functions, such as JITTERBUFFER,
+ DENOISE, and AGC use a similar mechanism. Audiohooks are processed
+ in the order in which they are placed on the channel. As such,
+ it typically makes sense to place functions that modify the voice
+ media data prior to placing the TALK_DETECT function, as this will
+ yield better results.</para>
+ <para>Example:</para>
+ <para>same => n,Set(DENOISE(rx)=on) ; Denoise received audio</para>
+ <para>same => n,Set(TALK_DETECT(set)=) ; Perform talk detection on the denoised received audio</para>
+ </note>
+ </description>
+ </function>
+ ***/
+
+#define DEFAULT_SILENCE_THRESHOLD 2500
+
+/*! \brief Private data structure used with the function's datastore */
+struct talk_detect_params {
+ /*! The audiohook for the function */
+ struct ast_audiohook audiohook;
+ /*! Our threshold above which we consider someone talking */
+ int dsp_talking_threshold;
+ /*! How long we'll wait before we decide someone is silent */
+ int dsp_silence_threshold;
+ /*! Whether or not the user is currently talking */
+ int talking;
+ /*! The time the current burst of talking started */
+ struct timeval talking_start;
+ /*! The DSP used to do the heavy lifting */
+ struct ast_dsp *dsp;
+};
+
+/*! \internal \brief Destroy the datastore */
+static void datastore_destroy_cb(void *data) {
+ struct talk_detect_params *td_params = data;
+
+ ast_audiohook_destroy(&td_params->audiohook);
+
+ if (td_params->dsp) {
+ ast_dsp_free(td_params->dsp);
+ }
+ ast_free(data);
+}
+
+/*! \brief The channel datastore the function uses to store state */
+static const struct ast_datastore_info talk_detect_datastore = {
+ .type = "talk_detect",
+ .destroy = datastore_destroy_cb
+};
+
+/*! \internal \brief An audiohook modification callback
+ *
+ * This processes the read side of a channel's voice data to see if
+ * they are talking
+ *
+ * \note We don't actually modify the audio, so this function always
+ * returns a 'failure' indicating that it didn't modify the data
+ */
+static int talk_detect_audiohook_cb(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
+{
+ int total_silence;
+ int update_talking = 0;
+ struct ast_datastore *datastore;
+ struct talk_detect_params *td_params;
+ struct stasis_message *message;
+
+ if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
+ return 1;
+ }
+
+ if (direction != AST_AUDIOHOOK_DIRECTION_READ) {
+ return 1;
+ }
+
+ if (frame->frametype != AST_FRAME_VOICE) {
+ return 1;
+ }
+
+ if (!(datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL))) {
+ return 1;
+ }
+ td_params = datastore->data;
+
+ ast_dsp_silence(td_params->dsp, frame, &total_silence);
+
+ if (total_silence < td_params->dsp_silence_threshold) {
+ if (!td_params->talking) {
+ update_talking = 1;
+ td_params->talking_start = ast_tvnow();
+ }
+ td_params->talking = 1;
+ } else {
+ if (td_params->talking) {
+ update_talking = 1;
+ }
+ td_params->talking = 0;
+ }
+
+ if (update_talking) {
+ struct ast_json *blob = NULL;
+
+ if (!td_params->talking) {
+ int64_t diff_ms = ast_tvdiff_ms(ast_tvnow(), td_params->talking_start);
+ diff_ms -= td_params->dsp_silence_threshold;
+
+ blob = ast_json_pack("{s: i}", "duration", diff_ms);
+ if (!blob) {
+ return 1;
+ }
+ }
+
+ ast_verb(4, "%s is now %s\n", ast_channel_name(chan),
+ td_params->talking ? "talking" : "silent");
+ message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
+ td_params->talking ? ast_channel_talking_start() : ast_channel_talking_stop(),
+ blob);
+ if (message) {
+ stasis_publish(ast_channel_topic(chan), message);
+ }
+
+ ast_json_unref(blob);
+ }
+
+ return 1;
+}
+
+/*! \internal \brief Disable talk detection on the channel */
+static int remove_talk_detect(struct ast_channel *chan)
+{
+ struct ast_datastore *datastore = NULL;
+ struct talk_detect_params *td_params;
+ SCOPED_CHANNELLOCK(chan_lock, chan);
+
+ datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
+ if (!datastore) {
+ ast_log(AST_LOG_WARNING, "Cannot remove TALK_DETECT from %s: TALK_DETECT not currently enabled\n",
+ ast_channel_name(chan));
+ return -1;
+ }
+ td_params = datastore->data;
+
+ if (ast_audiohook_remove(chan, &td_params->audiohook)) {
+ ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT audiohook from channel %s\n",
+ ast_channel_name(chan));
+ return -1;
+ }
+
+ if (ast_channel_datastore_remove(chan, datastore)) {
+ ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT datastore from channel %s\n",
+ ast_channel_name(chan));
+ return -1;
+ }
+ ast_datastore_free(datastore);
+
+ return 0;
+}
+
+/*! \internal \brief Enable talk detection on the channel */
+static int set_talk_detect(struct ast_channel *chan, int dsp_silence_threshold, int dsp_talking_threshold)
+{
+ struct ast_datastore *datastore = NULL;
+ struct talk_detect_params *td_params;
+ SCOPED_CHANNELLOCK(chan_lock, chan);
+
+ datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
+ if (!datastore) {
+ datastore = ast_datastore_alloc(&talk_detect_datastore, NULL);
+ if (!datastore) {
+ return -1;
+ }
+
+ td_params = ast_calloc(1, sizeof(*td_params));
+ if (!td_params) {
+ ast_datastore_free(datastore);
+ return -1;
+ }
+
+ ast_audiohook_init(&td_params->audiohook,
+ AST_AUDIOHOOK_TYPE_MANIPULATE,
+ "TALK_DETECT",
+ AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
+ td_params->audiohook.manipulate_callback = talk_detect_audiohook_cb;
+ ast_set_flag(&td_params->audiohook, AST_AUDIOHOOK_TRIGGER_READ);
+
+ td_params->dsp = ast_dsp_new_with_rate(ast_format_rate(ast_channel_rawreadformat(chan)));
+ if (!td_params->dsp) {
+ ast_datastore_free(datastore);
+ ast_free(td_params);
+ return -1;
+ }
+ datastore->data = td_params;
+
+ ast_channel_datastore_add(chan, datastore);
+ ast_audiohook_attach(chan, &td_params->audiohook);
+ } else {
+ /* Talk detection already enabled; update existing settings */
+ td_params = datastore->data;
+ }
+
+ td_params->dsp_talking_threshold = dsp_talking_threshold;
+ td_params->dsp_silence_threshold = dsp_silence_threshold;
+
+ ast_dsp_set_threshold(td_params->dsp, td_params->dsp_talking_threshold);
+
+ return 0;
+}
+
+/*! \internal \brief TALK_DETECT write function callback */
+static int talk_detect_fn_write(struct ast_channel *chan, const char *function, char *data, const char *value)
+{
+ int res;
+
+ if (!chan) {
+ return -1;
+ }
+
+ if (ast_strlen_zero(data)) {
+ ast_log(AST_LOG_WARNING, "TALK_DETECT requires an argument\n");
+ return -1;
+ }
+
+ if (!strcasecmp(data, "set")) {
+ int dsp_silence_threshold = DEFAULT_SILENCE_THRESHOLD;
+ int dsp_talking_threshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
+
+ if (!ast_strlen_zero(value)) {
+ char *parse = ast_strdupa(value);
+
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(silence_threshold);
+ AST_APP_ARG(talking_threshold);
+ );
+
+ AST_STANDARD_APP_ARGS(args, parse);
+
+ if (!ast_strlen_zero(args.silence_threshold)) {
+ if (sscanf(args.silence_threshold, "%30d", &dsp_silence_threshold) != 1) {
+ ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_silence_threshold\n",
+ args.silence_threshold);
+ return -1;
+ }
+
+ if (dsp_silence_threshold < 1) {
+ ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_silence_threshold\n",
+ dsp_silence_threshold);
+ return -1;
+ }
+ }
+
+ if (!ast_strlen_zero(args.talking_threshold)) {
+ if (sscanf(args.talking_threshold, "%30d", &dsp_talking_threshold) != 1) {
+ ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_talking_threshold\n",
+ args.talking_threshold);
+ return -1;
+ }
+
+ if (dsp_talking_threshold < 1) {
+ ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_talking_threshold\n",
+ dsp_silence_threshold);
+ return -1;
+ }
+ }
+ }
+
+ res = set_talk_detect(chan, dsp_silence_threshold, dsp_talking_threshold);
+ } else if (!strcasecmp(data, "remove")) {
+ res = remove_talk_detect(chan);
+ } else {
+ ast_log(AST_LOG_WARNING, "TALK_DETECT: unknown option %s\n", data);
+ res = -1;
+ }
+
+ return res;
+}
+
+/*! \brief Definition of the TALK_DETECT function */
+static struct ast_custom_function talk_detect_function = {
+ .name = "TALK_DETECT",
+ .write = talk_detect_fn_write,
+};
+
+/*! \internal \brief Unload the module */
+static int unload_module(void)
+{
+ int res = 0;
+
+ res |= ast_custom_function_unregister(&talk_detect_function);
+
+ return res;
+}
+
+/*! \internal \brief Load the module */
+static int load_module(void)
+{
+ int res = 0;
+
+ res |= ast_custom_function_register(&talk_detect_function);
+
+ return res ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Talk detection dialplan function");
Propchange: branches/12/funcs/func_talkdetect.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: branches/12/funcs/func_talkdetect.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: branches/12/funcs/func_talkdetect.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: branches/12/include/asterisk/stasis_channels.h
URL: http://svnview.digium.com/svn/asterisk/branches/12/include/asterisk/stasis_channels.h?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/include/asterisk/stasis_channels.h (original)
+++ branches/12/include/asterisk/stasis_channels.h Fri May 30 07:39:36 2014
@@ -500,6 +500,22 @@
struct stasis_message_type *ast_channel_moh_stop_type(void);
/*!
+ * \since 12.4.0
+ * \brief Message type for a channel starting talking
+ *
+ * \retval A stasis message type
+ */
+struct stasis_message_type *ast_channel_talking_start(void);
+
+/*!
+ * \since 12.4.0
+ * \brief Message type for a channel stopping talking
+ *
+ * \retval A stasis message type
+ */
+struct stasis_message_type *ast_channel_talking_stop(void);
+
+/*!
* \since 12
* \brief Publish in the \ref ast_channel_topic or \ref ast_channel_topic_all
* topics a stasis message for the channels involved in a dial operation.
Modified: branches/12/main/audiohook.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/audiohook.c?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/main/audiohook.c (original)
+++ branches/12/main/audiohook.c Fri May 30 07:39:36 2014
@@ -857,17 +857,15 @@
}
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
/* Feed in frame to manipulation. */
- if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
- /* XXX IGNORE FAILURE */
-
+ if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
/* If the manipulation fails then the frame will be returned in its original state.
* Since there are potentially more manipulator callbacks in the list, no action should
* be taken here to exit early. */
+ middle_frame_manipulated = 1;
}
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
- middle_frame_manipulated = 1;
}
/* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
Modified: branches/12/main/stasis_channels.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/stasis_channels.c?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/main/stasis_channels.c (original)
+++ branches/12/main/stasis_channels.c Fri May 30 07:39:36 2014
@@ -85,6 +85,34 @@
</see-also>
</managerEventInstance>
</managerEvent>
+ <managerEvent language="en_US" name="ChannelTalkingStart">
+ <managerEventInstance class="EVENT_FLAG_CLASS">
+ <synopsis>Raised when talking is detected on a channel.</synopsis>
+ <syntax>
+ <channel_snapshot/>
+ </syntax>
+ <see-also>
+ <ref type="function">TALK_DETECT</ref>
+ <ref type="managerEvent">ChannelTalkingStop</ref>
+ </see-also>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="ChannelTalkingStop">
+ <managerEventInstance class="EVENT_FLAG_CLASS">
+ <synopsis>Raised when talking is no longer detected on a channel.</synopsis>
+ <syntax>
+ <channel_snapshot/>
+ <parameter name="Duration">
+ <para>The length in time, in milliseconds, that talking was
+ detected on the channel.</para>
+ </parameter>
+ </syntax>
+ <see-also>
+ <ref type="function">TALK_DETECT</ref>
+ <ref type="managerEvent">ChannelTalkingStart</ref>
+ </see-also>
+ </managerEventInstance>
+ </managerEvent>
***/
#define NUM_MULTI_CHANNEL_BLOB_BUCKETS 7
@@ -972,6 +1000,58 @@
}
return json;
+}
+
+static struct ast_manager_event_blob *talking_start_to_ami(struct stasis_message *msg)
+{
+ struct ast_str *channel_string;
+ struct ast_channel_blob *obj = stasis_message_data(msg);
+ struct ast_manager_event_blob *blob;
+
+ channel_string = ast_manager_build_channel_state_string(obj->snapshot);
+ if (!channel_string) {
+ return NULL;
+ }
+
+ blob = ast_manager_event_blob_create(EVENT_FLAG_CALL, "ChannelTalkingStart",
+ "%s", ast_str_buffer(channel_string));
+ ast_free(channel_string);
+
+ return blob;
+}
+
+static struct ast_json *talking_start_to_json(struct stasis_message *message,
+ const struct stasis_message_sanitizer *sanitize)
+{
+ return channel_blob_to_json(message, "ChannelTalkingStarted", sanitize);
+}
+
+static struct ast_manager_event_blob *talking_stop_to_ami(struct stasis_message *msg)
+{
+ struct ast_str *channel_string;
+ struct ast_channel_blob *obj = stasis_message_data(msg);
+ int duration = ast_json_integer_get(ast_json_object_get(obj->blob, "duration"));
+ struct ast_manager_event_blob *blob;
+
+ channel_string = ast_manager_build_channel_state_string(obj->snapshot);
+ if (!channel_string) {
+ return NULL;
+ }
+
+ blob = ast_manager_event_blob_create(EVENT_FLAG_CALL, "ChannelTalkingStop",
+ "%s"
+ "Duration: %d\r\n",
+ ast_str_buffer(channel_string),
+ duration);
+ ast_free(channel_string);
+
+ return blob;
+}
+
+static struct ast_json *talking_stop_to_json(struct stasis_message *message,
+ const struct stasis_message_sanitizer *sanitize)
+{
+ return channel_blob_to_json(message, "ChannelTalkingFinished", sanitize);
}
/*!
@@ -1008,6 +1088,14 @@
STASIS_MESSAGE_TYPE_DEFN(ast_channel_agent_logoff_type,
.to_ami = agent_logoff_to_ami,
);
+STASIS_MESSAGE_TYPE_DEFN(ast_channel_talking_start,
+ .to_ami = talking_start_to_ami,
+ .to_json = talking_start_to_json,
+ );
+STASIS_MESSAGE_TYPE_DEFN(ast_channel_talking_stop,
+ .to_ami = talking_stop_to_ami,
+ .to_json = talking_stop_to_json,
+ );
/*! @} */
@@ -1038,6 +1126,8 @@
STASIS_MESSAGE_TYPE_CLEANUP(ast_channel_monitor_stop_type);
STASIS_MESSAGE_TYPE_CLEANUP(ast_channel_agent_login_type);
STASIS_MESSAGE_TYPE_CLEANUP(ast_channel_agent_logoff_type);
+ STASIS_MESSAGE_TYPE_CLEANUP(ast_channel_talking_start);
+ STASIS_MESSAGE_TYPE_CLEANUP(ast_channel_talking_stop);
}
int ast_stasis_channels_init(void)
@@ -1084,6 +1174,8 @@
res |= STASIS_MESSAGE_TYPE_INIT(ast_channel_moh_stop_type);
res |= STASIS_MESSAGE_TYPE_INIT(ast_channel_monitor_start_type);
res |= STASIS_MESSAGE_TYPE_INIT(ast_channel_monitor_stop_type);
+ res |= STASIS_MESSAGE_TYPE_INIT(ast_channel_talking_start);
+ res |= STASIS_MESSAGE_TYPE_INIT(ast_channel_talking_stop);
return res;
}
Modified: branches/12/res/ari/ari_model_validators.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/res/ari/ari_model_validators.c?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/res/ari/ari_model_validators.c (original)
+++ branches/12/res/ari/ari_model_validators.c Fri May 30 07:39:36 2014
@@ -3070,6 +3070,180 @@
return ast_ari_validate_channel_state_change;
}
+int ast_ari_validate_channel_talking_finished(struct ast_json *json)
+{
+ int res = 1;
+ struct ast_json_iter *iter;
+ int has_type = 0;
+ int has_application = 0;
+ int has_channel = 0;
+ int has_duration = 0;
+
+ for (iter = ast_json_object_iter(json); iter; iter = ast_json_object_iter_next(json, iter)) {
+ if (strcmp("type", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_type = 1;
+ prop_is_valid = ast_ari_validate_string(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished field type failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("application", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_application = 1;
+ prop_is_valid = ast_ari_validate_string(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished field application failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("timestamp", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ prop_is_valid = ast_ari_validate_date(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished field timestamp failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("channel", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_channel = 1;
+ prop_is_valid = ast_ari_validate_channel(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished field channel failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("duration", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_duration = 1;
+ prop_is_valid = ast_ari_validate_int(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished field duration failed validation\n");
+ res = 0;
+ }
+ } else
+ {
+ ast_log(LOG_ERROR,
+ "ARI ChannelTalkingFinished has undocumented field %s\n",
+ ast_json_object_iter_key(iter));
+ res = 0;
+ }
+ }
+
+ if (!has_type) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished missing required field type\n");
+ res = 0;
+ }
+
+ if (!has_application) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished missing required field application\n");
+ res = 0;
+ }
+
+ if (!has_channel) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished missing required field channel\n");
+ res = 0;
+ }
+
+ if (!has_duration) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingFinished missing required field duration\n");
+ res = 0;
+ }
+
+ return res;
+}
+
+ari_validator ast_ari_validate_channel_talking_finished_fn(void)
+{
+ return ast_ari_validate_channel_talking_finished;
+}
+
+int ast_ari_validate_channel_talking_started(struct ast_json *json)
+{
+ int res = 1;
+ struct ast_json_iter *iter;
+ int has_type = 0;
+ int has_application = 0;
+ int has_channel = 0;
+
+ for (iter = ast_json_object_iter(json); iter; iter = ast_json_object_iter_next(json, iter)) {
+ if (strcmp("type", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_type = 1;
+ prop_is_valid = ast_ari_validate_string(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted field type failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("application", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_application = 1;
+ prop_is_valid = ast_ari_validate_string(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted field application failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("timestamp", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ prop_is_valid = ast_ari_validate_date(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted field timestamp failed validation\n");
+ res = 0;
+ }
+ } else
+ if (strcmp("channel", ast_json_object_iter_key(iter)) == 0) {
+ int prop_is_valid;
+ has_channel = 1;
+ prop_is_valid = ast_ari_validate_channel(
+ ast_json_object_iter_value(iter));
+ if (!prop_is_valid) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted field channel failed validation\n");
+ res = 0;
+ }
+ } else
+ {
+ ast_log(LOG_ERROR,
+ "ARI ChannelTalkingStarted has undocumented field %s\n",
+ ast_json_object_iter_key(iter));
+ res = 0;
+ }
+ }
+
+ if (!has_type) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted missing required field type\n");
+ res = 0;
+ }
+
+ if (!has_application) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted missing required field application\n");
+ res = 0;
+ }
+
+ if (!has_channel) {
+ ast_log(LOG_ERROR, "ARI ChannelTalkingStarted missing required field channel\n");
+ res = 0;
+ }
+
+ return res;
+}
+
+ari_validator ast_ari_validate_channel_talking_started_fn(void)
+{
+ return ast_ari_validate_channel_talking_started;
+}
+
int ast_ari_validate_channel_userevent(struct ast_json *json)
{
int res = 1;
@@ -3647,6 +3821,12 @@
if (strcmp("ChannelStateChange", discriminator) == 0) {
return ast_ari_validate_channel_state_change(json);
} else
+ if (strcmp("ChannelTalkingFinished", discriminator) == 0) {
+ return ast_ari_validate_channel_talking_finished(json);
+ } else
+ if (strcmp("ChannelTalkingStarted", discriminator) == 0) {
+ return ast_ari_validate_channel_talking_started(json);
+ } else
if (strcmp("ChannelUserevent", discriminator) == 0) {
return ast_ari_validate_channel_userevent(json);
} else
@@ -3805,6 +3985,12 @@
} else
if (strcmp("ChannelStateChange", discriminator) == 0) {
return ast_ari_validate_channel_state_change(json);
+ } else
+ if (strcmp("ChannelTalkingFinished", discriminator) == 0) {
+ return ast_ari_validate_channel_talking_finished(json);
+ } else
+ if (strcmp("ChannelTalkingStarted", discriminator) == 0) {
+ return ast_ari_validate_channel_talking_started(json);
} else
if (strcmp("ChannelUserevent", discriminator) == 0) {
return ast_ari_validate_channel_userevent(json);
Modified: branches/12/res/ari/ari_model_validators.h
URL: http://svnview.digium.com/svn/asterisk/branches/12/res/ari/ari_model_validators.h?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/res/ari/ari_model_validators.h (original)
+++ branches/12/res/ari/ari_model_validators.h Fri May 30 07:39:36 2014
@@ -789,6 +789,42 @@
* See \ref ast_ari_model_validators.h for more details.
*/
ari_validator ast_ari_validate_channel_state_change_fn(void);
+
+/*!
+ * \brief Validator for ChannelTalkingFinished.
+ *
+ * Talking is no longer detected on the channel.
+ *
+ * \param json JSON object to validate.
+ * \returns True (non-zero) if valid.
+ * \returns False (zero) if invalid.
+ */
+int ast_ari_validate_channel_talking_finished(struct ast_json *json);
+
+/*!
+ * \brief Function pointer to ast_ari_validate_channel_talking_finished().
+ *
+ * See \ref ast_ari_model_validators.h for more details.
+ */
+ari_validator ast_ari_validate_channel_talking_finished_fn(void);
+
+/*!
+ * \brief Validator for ChannelTalkingStarted.
+ *
+ * Talking was detected on the channel.
+ *
+ * \param json JSON object to validate.
+ * \returns True (non-zero) if valid.
+ * \returns False (zero) if invalid.
+ */
+int ast_ari_validate_channel_talking_started(struct ast_json *json);
+
+/*!
+ * \brief Function pointer to ast_ari_validate_channel_talking_started().
+ *
+ * See \ref ast_ari_model_validators.h for more details.
+ */
+ari_validator ast_ari_validate_channel_talking_started_fn(void);
/*!
* \brief Validator for ChannelUserevent.
@@ -1274,6 +1310,17 @@
* - application: string (required)
* - timestamp: Date
* - channel: Channel (required)
+ * ChannelTalkingFinished
+ * - type: string (required)
+ * - application: string (required)
+ * - timestamp: Date
+ * - channel: Channel (required)
+ * - duration: int (required)
+ * ChannelTalkingStarted
+ * - type: string (required)
+ * - application: string (required)
+ * - timestamp: Date
+ * - channel: Channel (required)
* ChannelUserevent
* - type: string (required)
* - application: string (required)
Modified: branches/12/rest-api/api-docs/events.json
URL: http://svnview.digium.com/svn/asterisk/branches/12/rest-api/api-docs/events.json?view=diff&rev=414934&r1=414933&r2=414934
==============================================================================
--- branches/12/rest-api/api-docs/events.json (original)
+++ branches/12/rest-api/api-docs/events.json Fri May 30 07:39:36 2014
@@ -159,6 +159,8 @@
"ChannelUserevent",
"ChannelHangupRequest",
"ChannelVarset",
+ "ChannelTalkingStarted",
+ "ChannelTalkingFinished",
"EndpointStateChange",
"Dial",
"StasisEnd",
@@ -572,6 +574,33 @@
}
}
},
+ "ChannelTalkingStarted": {
+ "id": "ChannelTalkingStarted",
+ "description": "Talking was detected on the channel.",
+ "properties": {
+ "channel": {
+ "required": true,
+ "type": "Channel",
+ "description": "The channel on which talking started."
+ }
+ }
+ },
+ "ChannelTalkingFinished": {
+ "id": "ChannelTalkingFinished",
+ "description": "Talking is no longer detected on the channel.",
+ "properties": {
+ "channel": {
+ "required": true,
+ "type": "Channel",
+ "description": "The channel on which talking completed."
+ },
+ "duration": {
+ "required": true,
+ "type": "int",
+ "description": "The length of time, in milliseconds, that talking was detected on the channel"
+ }
+ }
+ },
"EndpointStateChange": {
"id": "EndpointStateChange",
"description": "Endpoint state changed.",
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