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[commit] r5598 - freeswitch/trunk/src/mod/xml_int/mod_xml_curl
by Freeswitch SVN 13 Aug '07

13 Aug '07

Author: anthm Date: Mon Aug 13 19:50:01 2007 New Revision: 5598 Modified: freeswitch/trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c Log: fix small unlikely but still possible bugs Modified: freeswitch/trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c ============================================================================== --- freeswitch/trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c (original) +++ freeswitch/trunk/src/mod/xml_int/mod_xml_curl/mod_xml_curl.c Mon Aug 13 19:50:01 2007 @@ -164,7 +164,7 @@ if (!(bindings_tag = switch_xml_child(cfg, "bindings"))) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Missing <bindings> tag!\n"); - return SWITCH_STATUS_FALSE; + goto done; } for (binding_tag = switch_xml_child(bindings_tag, "binding"); binding_tag; binding_tag = binding_tag->next) { @@ -195,7 +195,7 @@ } if (!(binding = malloc(sizeof(*binding)))) { - return SWITCH_STATUS_FALSE; + goto done; } memset(binding, 0, sizeof(*binding)); @@ -218,6 +218,7 @@ binding = NULL; } + done: switch_xml_free(xml); return x ? SWITCH_STATUS_SUCCESS : SWITCH_STATUS_FALSE;

1 0

[commit] r5597 - freeswitch/trunk/src/mod/asr_tts/mod_openmrcp
by Freeswitch SVN 13 Aug '07

13 Aug '07

Author: greenlizard Date: Mon Aug 13 19:49:37 2007 New Revision: 5597 Modified: freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c Log: speak-finished event now being handled, passed up to freeswitch to let it know speech has stopped. audio sounds fine Modified: freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c ============================================================================== --- freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c (original) +++ freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c Mon Aug 13 19:49:37 2007 @@ -87,7 +87,7 @@ #include <switch.h> #define OPENMRCP_WAIT_TIMEOUT 5000 -#define MY_BUF_LEN 1024 * 32 +#define MY_BUF_LEN 1024 * 128 #define MY_BLOCK_SIZE MY_BUF_LEN SWITCH_MODULE_LOAD_FUNCTION(mod_openmrcp_load); @@ -176,7 +176,8 @@ typedef enum { FLAG_HAS_TEXT = (1 << 0), FLAG_BARGE = (1 << 1), - FLAG_READY = (1 << 2) + FLAG_READY = (1 << 2), + FLAG_SPEAK_COMPLETE = (1 << 3) } mrcp_flag_t; typedef struct { @@ -261,6 +262,8 @@ return MRCP_STATUS_FAILURE; } + switch_buffer_destroy(&tts_session->audio_buffer); + if(tts_session->pool) { apr_pool_destroy(tts_session->pool); tts_session->pool = NULL; @@ -499,8 +502,21 @@ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "!tts_session\n"); return MRCP_STATUS_FAILURE; } + + if (mrcp_message->start_line.method_name) { + if (!strcmp(mrcp_message->start_line.method_name,"SPEAK-COMPLETE")) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "setting FLAG_SPEAK_COMPLETE\n"); + switch_set_flag_locked(tts_session, FLAG_SPEAK_COMPLETE); + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "ignoring method: %s\n", mrcp_message->start_line.method_name); + } + } + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mrcp msg body: %s\n", mrcp_message->body); + + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_CHANNEL_MODIFY); } @@ -581,60 +597,6 @@ } -/** - * Called back by openmrcp client thread every time it receives audio - * from the TTS server we are connected to. Puts audio in a queueu - * and it will be pulled out from read_tts - -static apr_status_t openmrcp_tts_write_frame_NEW(audio_sink_t *sink, media_frame_t *frame) -{ - tts_session_t *tts_session = sink->object; - media_frame_t *media_frame; - switch_byte_t *buffer; - size_t len; - - len = frame->codec_frame.size; - - // create new media frame - media_frame = (media_frame_t *) switch_core_alloc(tts_session->pool, sizeof(media_frame_t)); - if (!media_frame) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "media_frame creation failed\n"); - return SWITCH_STATUS_MEMERR; - } - - // since *frame might get freed by caller (true or false?), allocate a - // new buffer and copy *data into it. - buffer = (switch_byte_t *) switch_core_alloc(tts_session->pool, sizeof(switch_byte_t)*len); - if (!buffer) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate buffer\n"); - return SWITCH_STATUS_MEMERR; - } - buffer = memcpy(buffer, frame->codec_frame.buffer, len); - media_frame->codec_frame.buffer = buffer; - media_frame->codec_frame.size = len; - media_frame->type = MEDIA_FRAME_TYPE_AUDIO; - - - // add audio to buffer - switch_mutex_lock(tts_session->audio_lock); - if (switch_buffer_write(tts_session->audio_buffer, buffer, len) == 0) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Could not write to buffer\n"); - } - switch_mutex_unlock(tts_session->audio_lock); - - switch_mutex_unlock(tts_session->audio_lock); - - if (switch_queue_trypush(tts_session->audio_queue, (void *) media_frame)) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push audio to queue\n"); - return MRCP_STATUS_FAILURE; - } - - return MRCP_STATUS_SUCCESS; - -} -*/ - - /** * Called back by openmcp client thread every time its ready for more audio to send * the recognition server we are connected to. Reads data that was put into a @@ -1140,6 +1102,8 @@ synth_stop(tts_client_context,tts_session); // TODO wait_for_event(tts_session->event_queue,OPENMRCP_EVENT_CHANNEL_MODIFY,5000); + + /* terminate tts session */ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Terminate tts_session\n"); mrcp_client_context_session_terminate(tts_client_context,tts_session->client_session); @@ -1191,18 +1155,62 @@ { media_frame_t *queue_frame = NULL; tts_session_t *tts_session = (tts_session_t *) sh->private_info; + size_t return_len=0; + size_t amt2copy=0; + size_t desired = *datalen; + switch_byte_t *audiodata = (switch_byte_t *) data; + + while(return_len < desired) { + + if (switch_test_flag(tts_session, FLAG_SPEAK_COMPLETE)) { + // tell fs we are done + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "FLAG_SPEAK_COMPLETE\n"); + return SWITCH_STATUS_BREAK; + } + + if (switch_queue_pop(tts_session->audio_queue, (void *) &queue_frame)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "could not pop from queue\n"); + if (switch_test_flag(tts_session, FLAG_SPEAK_COMPLETE)) { + // tell fs we are done + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "FLAG_SPEAK_COMPLETE\n"); + return SWITCH_STATUS_BREAK; + } + break; + } + else { + if (switch_test_flag(tts_session, FLAG_SPEAK_COMPLETE)) { + // tell fs we are done + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "FLAG_SPEAK_COMPLETE\n"); + return SWITCH_STATUS_BREAK; + } + + if (queue_frame->codec_frame.size >= desired) { + amt2copy = desired; + } + else { + // limit the amt we copy to audiodata to be LTE datalen + // if the queue frame has _more_, just ignore it (TODO: fix this!) + amt2copy = queue_frame->codec_frame.size; + } + memcpy(audiodata, queue_frame->codec_frame.buffer, amt2copy); + return_len += amt2copy; + *datalen = return_len; + audiodata += amt2copy; // move pointer forward + *rate = 8000; + } - if (switch_queue_trypop(tts_session->audio_queue, (void *) &queue_frame)) { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "could not pop from queue\n"); - return SWITCH_STATUS_SUCCESS; } - else { - memcpy(data, queue_frame->codec_frame.buffer, queue_frame->codec_frame.size); - *datalen = queue_frame->codec_frame.size; - *rate = 8000; - return SWITCH_STATUS_SUCCESS; + + // double check we actually read something + if (*datalen == 0) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "no data read from buffer\n"); + return SWITCH_STATUS_FALSE; + } + else if (*datalen < desired) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "return_len: (%d) < desired: (%d)\n", return_len, desired); } + return SWITCH_STATUS_SUCCESS; } @@ -1315,6 +1323,8 @@ } } + switch_xml_free(xml); + return SWITCH_STATUS_SUCCESS; }

1 0

[commit] r5595 - freeswitch/trunk/src/mod/xml_int/mod_xml_cdr
by Freeswitch SVN 13 Aug '07

13 Aug '07

Author: anthm Date: Mon Aug 13 19:37:22 2007 New Revision: 5595 Modified: freeswitch/trunk/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c Log: fix missing deref of root xml registry in xml_cdr that causes a hang Modified: freeswitch/trunk/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c ============================================================================== --- freeswitch/trunk/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c (original) +++ freeswitch/trunk/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c Mon Aug 13 19:37:22 2007 @@ -198,7 +198,7 @@ { char *cf = "xml_cdr.conf"; switch_xml_t cfg, xml, settings, param; - + switch_status_t status = SWITCH_STATUS_SUCCESS; /* test global state handlers */ switch_core_add_state_handler(&state_handlers); @@ -212,7 +212,7 @@ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "open of %s failed\n", cf); return SWITCH_STATUS_TERM; } - + if ((settings = switch_xml_child(cfg, "settings"))) { for (param = switch_xml_child(settings, "param"); param; param = param->next) { char *var = (char *) switch_xml_attr_soft(param, "name"); @@ -252,13 +252,15 @@ if(!switch_strlen_zero(globals.url) && switch_strlen_zero(globals.errLogDir)) { if ((globals.errLogDir = switch_mprintf("%s/xml_cdr", SWITCH_GLOBAL_dirs.log_dir))) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n"); - return SWITCH_STATUS_FALSE; + status = SWITCH_STATUS_FALSE; + goto done; } } - /* indicate that the module should continue to be loaded */ - return SWITCH_STATUS_SUCCESS; + done: + switch_xml_free(xml); + return status; }

1 0

[commit] r5593 - freeswitch/trunk/src/mod/asr_tts/mod_openmrcp
by Freeswitch SVN 13 Aug '07

13 Aug '07

Author: greenlizard Date: Mon Aug 13 14:19:55 2007 New Revision: 5593 Added: freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/ freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/Makefile freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c Log: initial dev version of mod_openmrcp, not ready for use yet Added: freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/Makefile ============================================================================== --- (empty file) +++ freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/Makefile Mon Aug 13 14:19:55 2007 @@ -0,0 +1,33 @@ +# define these targets in your makefile if you wish +# local_all local_depend local_clean depend_install local_install local_distclean local_extraclean: + +# and define these variables to impact your build + +# TO GET THIS TO COMPILE +# hack ../../../../build/modmake.rules and remove the +# -Werror from the ALL_CFLAGS variable. + +# TODO +# - re-enable the -Werror flag after openmrcp compilation warnings are fixed +# - needs to compile/link against the _freeswitch_ sofia libs (see FIXME) +# - Should this download and build openmrcp? What's the plan? (see FIXME) +# - somehow consolidate libs and includes into less files + +# FIXME!!! +OPENMRCP_DIR=/usr/src/openmrcp_trunk + +OPENMRCP_INCLUDE=-I$(OPENMRCP_DIR)/mrcpcore/client/include/ -I$(OPENMRCP_DIR)/platform/openmrcpclient/include/ -I$(OPENMRCP_DIR)/mediaframe/include/ -I$(OPENMRCP_DIR)/mrcpcore/engine/include/ -I$(OPENMRCP_DIR)/mrcpcore/include/ -I$(OPENMRCP_DIR)/mrcpcore/parser/include/ -I$(OPENMRCP_DIR)/mrcpcore/server/include/ -I$(OPENMRCP_DIR)/mrcpcore/media/include/ -I$(OPENMRCP_DIR)/mrcpcore/util/include -I$(OPENMRCP_DIR)/mrcpcore/resource/include/ + +OPENMRCP_LIBS=$(OPENMRCP_DIR)/platform/openmrcpclient/.libs/libopenmrcpclient.a $(OPENMRCP_DIR)/platform/openmrcpserver/.libs/libopenmrcpserver.a $(OPENMRCP_DIR)/mrcpcore/client/.libs/libmrcpclient.a $(OPENMRCP_DIR)/mediaframe/.libs/libmediaframe.a $(OPENMRCP_DIR)/mrcpcore/engine/.libs/libmrcpengine.a $(OPENMRCP_DIR)/mrcpcore/parser/.libs/libmrcpparser.a $(OPENMRCP_DIR)/mrcpcore/server/.libs/libmrcpserver.a $(OPENMRCP_DIR)/mrcpcore/media/.libs/libmrcpmedia.a $(OPENMRCP_DIR)/mrcpcore/util/.libs/libmrcputil.a $(OPENMRCP_DIR)/mrcpcore/util/.libs/libmrcputil.a $(OPENMRCP_DIR)/mrcpcore/resource/.libs/libmrcpresource.a $(OPENMRCP_DIR)/module/plugin/demo/.libs/libdemoplugin.a $(OPENMRCP_DIR)/module/mrcpv2agent/.libs/libmrcpv2agent.a $(OPENMRCP_DIR)/module/mrcpmediaframe/.libs/libmrcpmediaframe.a $(OPENMRCP_DIR)/module/mrcpsofiasip/.libs/libmrcpsofiasip.a $(OPENMRCP_DIR)/mrcpcore/.libs/libmrcpcore.a + +MRCP_APR_INCLUDES = -I../../../../libs/apr/include +MRCP_APU_INCLUDES = -I../../../../libs/apr-util/include +MRCP_SOFIA_INCLUDES = -I/usr/local/include/sofia-sip-1.12 # FIXME!! + +LOCAL_CFLAGS= -Wno-unused +LOCAL_CFLAGS+= -Wno-comment -Wno-sign-compare $(OPENMRCP_INCLUDE) $(MRCP_APR_INCLUDES) $(MRCP_APU_INCLUDES) $(MRCP_SOFIA_INCLUDES) + +LOCAL_LDFLAGS=$(OPENMRCP_LIBS) +VERBOSE=TRUE +include ../../../../build/modmake.rules + Added: freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c ============================================================================== --- (empty file) +++ freeswitch/trunk/src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c Mon Aug 13 14:19:55 2007 @@ -0,0 +1,1429 @@ +/* + * FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application + * Copyright (C) 2005/2006, Anthony Minessale II <anthmct(a)yahoo.com> + * + * 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 <anthmct(a)yahoo.com> + * Portions created by the Initial Developer are Copyright (C) + * the Initial Developer. All Rights Reserved. + * + * Contributor(s): + * + * Traun Leyden <tleyden(a)branchcut.com> + * + * Module which acts as an MRCP client to an MRCP speech recognition + * server. In other words it bridges freeswitch to an external speech + * recognition system. Documentation on how to install and configure + * the module is here: http://wiki.freeswitch.org/wiki/Mod_openmrcp + * + * Uses OpenMrcp (http://wiki.freeswitch.org/wiki/OpenMRCP) as the + * the client library. + * + * TODO + * ======= + * + * - MAJOR DESIGN ISSUE!! It is way too expensive to be calling malloc on every audio frame, + * this needs to send the packet directly. OpenMrcp will need a way to disable their own + * timer so that the fs timer is the only one driving the media. + * + * - There are two memory pools in use. One in asr_session which is managed + * by this module, and one in the switch_asr_handle_t, which is managed by freeswitch. + * These need to be consolidated into one. (basically throw away the one in asr_session) + * + * - fs status codes (eg, SWITCH_STATUS_GENERR) and mrcp status codes (MRCP_STATUS_FAILURE) + * are intermixed badly. this needs cleanup + * + * - openmrcp_flush_tts, openmrcp_text_param_tts, openmrcp_numeric_param_tts, + * openmrcp_float_param_tts need to have functionality added + * + * - fix audio problem with TTS, convert from using queue to using a switch_buffer + * (in progress) + * + * - use a regex for extracting xml from raw result received from mrcp recognition + * server + * + */ + +#ifdef __ICC +#pragma warning (disable:188) +#endif + + +#include "openmrcp_client.h" +#include "mrcp_client_context.h" +#include "mrcp_recognizer.h" +#include "mrcp_synthesizer.h" +#include "mrcp_generic_header.h" +#include "rtp_session.h" +#include "mrcp_client.h" +#include "mrcp_client_context.h" +#include "mrcp_client_defs.h" +#include "mrcp_client_session.h" +#include "mrcp_client_resource.h" +#include "mrcp_client_signaling_agent.h" +#include "mrcp_media_agent.h" +#include "mrcp_resource.h" +#include "mrcp_consumer_task.h" +#include <apr_general.h> +#include <apr_file_io.h> +#include <apr_thread_proc.h> +#include <apr_thread_cond.h> +#include <apr_strings.h> + +#include <switch.h> + +#define OPENMRCP_WAIT_TIMEOUT 5000 +#define MY_BUF_LEN 1024 * 32 +#define MY_BLOCK_SIZE MY_BUF_LEN + +SWITCH_MODULE_LOAD_FUNCTION(mod_openmrcp_load); +SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_openmrcp_shutdown); +SWITCH_MODULE_DEFINITION(mod_openmrcp, mod_openmrcp_load, + mod_openmrcp_shutdown, NULL); + +static struct { + char *asr_client_ip; + char *asr_server_ip; + uint32_t asr_proto_version; + uint32_t asr_client_port; + uint32_t asr_server_port; + char *tts_client_ip; + char *tts_server_ip; + uint32_t tts_proto_version; + uint32_t tts_client_port; + uint32_t tts_server_port; + uint32_t rtp_port_min; + uint32_t rtp_port_max; +} globals; + + +typedef enum { + OPENMRCP_EVENT_NONE, + OPENMRCP_EVENT_SESSION_INITIATE, + OPENMRCP_EVENT_SESSION_TERMINATE, + OPENMRCP_EVENT_CHANNEL_CREATE, + OPENMRCP_EVENT_CHANNEL_DESTROY, + OPENMRCP_EVENT_CHANNEL_MODIFY +} openmrcp_event_t; + +static const char *openmrcp_event_names[] = { + "NONE", + "SESSION_INITIATE", + "SESSION_TERMINATE", + "CHANNEL_CREATE", + "CHANNEL_DESTROY", + "CHANNEL_MODIFY", +}; + +typedef struct asr_session_t asr_session_t; +struct asr_session_t { + mrcp_session_t *client_session; + mrcp_client_channel_t *channel; + switch_queue_t *audio_queue; + switch_queue_t *event_queue; + mrcp_message_t *mrcp_message_last_rcvd; + audio_source_t *source; + apr_pool_t *pool; + uint32_t flags; + switch_mutex_t *flag_mutex; +}; + +typedef struct tts_session_t tts_session_t; +struct tts_session_t { + mrcp_session_t *client_session; + mrcp_client_channel_t *channel; + switch_queue_t *audio_queue; // TO BE REMOVED + switch_queue_t *event_queue; + switch_mutex_t *audio_lock; + switch_buffer_t *audio_buffer; + audio_sink_t *sink; + apr_pool_t *pool; + switch_speech_flag_t flags; + switch_mutex_t *flag_mutex; +}; + +static apr_status_t openmrcp_recognizer_read_frame(audio_source_t *source, media_frame_t *frame); +static apr_status_t openmrcp_tts_write_frame(audio_sink_t *sink, media_frame_t *frame); + +static const audio_source_method_set_t audio_source_method_set = { + NULL, + NULL, + NULL, + openmrcp_recognizer_read_frame +}; + +static const audio_sink_method_set_t audio_sink_method_set = { + NULL, + NULL, + NULL, + openmrcp_tts_write_frame +}; + +typedef enum { + FLAG_HAS_TEXT = (1 << 0), + FLAG_BARGE = (1 << 1), + FLAG_READY = (1 << 2) +} mrcp_flag_t; + +typedef struct { + mrcp_client_t *asr_client; + mrcp_client_context_t *asr_client_context; + mrcp_client_t *tts_client; + mrcp_client_context_t *tts_client_context; +} openmrcp_module_t; + +static openmrcp_module_t openmrcp_module; + +static asr_session_t* asr_session_create() +{ + asr_session_t *asr_session; + apr_pool_t *session_pool; + + if(apr_pool_create(&session_pool,NULL) != APR_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed to create session_pool\n"); + return NULL; + } + + asr_session = apr_palloc(session_pool,sizeof(asr_session_t)); + asr_session->pool = session_pool; + asr_session->client_session = NULL; + asr_session->channel = NULL; + asr_session->audio_queue = NULL; + asr_session->event_queue = NULL; + + + /* create an event queue */ + if (switch_queue_create(&asr_session->event_queue, 1000, asr_session->pool)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "event queue creation failed\n"); + } + + return asr_session; +} + + +static tts_session_t* tts_session_create() +{ + tts_session_t *tts_session; + apr_pool_t *session_pool; + + if(apr_pool_create(&session_pool,NULL) != APR_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed to create session_pool\n"); + return NULL; + } + + tts_session = apr_palloc(session_pool,sizeof(tts_session_t)); + tts_session->pool = session_pool; + tts_session->client_session = NULL; + tts_session->channel = NULL; + tts_session->audio_queue = NULL; + tts_session->event_queue = NULL; + + /* create an event queue */ + if (switch_queue_create(&tts_session->event_queue, 1000, tts_session->pool)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "event queue creation failed\n"); + } + + return tts_session; +} + + +static mrcp_status_t asr_session_destroy(asr_session_t *asr_session) +{ + if(!asr_session) { + return MRCP_STATUS_FAILURE; + } + + if(asr_session->pool) { + apr_pool_destroy(asr_session->pool); + asr_session->pool = NULL; + } + return MRCP_STATUS_SUCCESS; +} + + +static mrcp_status_t tts_session_destroy(tts_session_t *tts_session) +{ + if(!tts_session) { + return MRCP_STATUS_FAILURE; + } + + if(tts_session->pool) { + apr_pool_destroy(tts_session->pool); + tts_session->pool = NULL; + } + return MRCP_STATUS_SUCCESS; +} + + +static mrcp_status_t wait_for_event(switch_queue_t *event_queue, openmrcp_event_t openmrcp_event, size_t timeout) +{ + + openmrcp_event_t *popped_event = NULL; + size_t sleep_ms = 100; + + if(event_queue) { + if (switch_queue_trypop(event_queue, (void *) &popped_event)) { + // most likely this failed because queue is empty. sleep for timeout seconds and try again? + if (timeout > 0) { + if (timeout < sleep_ms) { + switch_sleep(timeout * 1000); // ms->us + timeout = 0; + } + else { + switch_sleep(sleep_ms * 1000); // ms->us + timeout -= sleep_ms; + } + + // call recursively + // TODO: This is going to end up in a lot of recursion and + // maybe blow the stack .. rethink this approach + return wait_for_event(event_queue, openmrcp_event, timeout); + + } + else { + // nothing in queue, no timeout left, return failure + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "nothing in queue, no timeout left, return failure\n"); + return MRCP_STATUS_FAILURE; + } + + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped event\n"); + // check if popped event matches the event we are looking for + if (*popped_event == openmrcp_event) { + // just what we were waiting for! + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "just what we were waiting for! rturn success\n"); + return MRCP_STATUS_SUCCESS; + } + else { + // nothing popped, but maybe there's other things in queue, or something will arrive + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped unexpected\n"); + if (!popped_event) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped NULL!!\n"); + } + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped: %d\n", *popped_event); + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped name: %s\n", openmrcp_event_names[*popped_event]); + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "popped [%s] but was expecting [%s], but maybe there's other things in queue, or something will arrive\n", openmrcp_event_names[*popped_event], openmrcp_event_names[openmrcp_event]); + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recursively\n"); + return wait_for_event(event_queue, openmrcp_event, timeout); + } + + } + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "event queue is null\n"); + return MRCP_STATUS_FAILURE; + } + +} + + +static mrcp_status_t asr_session_signal_event(asr_session_t *asr_session, openmrcp_event_t openmrcp_event) +{ + mrcp_status_t status = MRCP_STATUS_SUCCESS; + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Got event: %s\n", openmrcp_event_names[openmrcp_event]); + + + // allocate memory for event + openmrcp_event_t *event2queue = (openmrcp_event_t *) switch_core_alloc(asr_session->pool, sizeof(openmrcp_event_t)); + *event2queue = openmrcp_event; + + // add it to queue + if (switch_queue_trypush(asr_session->event_queue, (void *) event2queue)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push event to queue\n"); + status = SWITCH_STATUS_GENERR; + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "pushed event to queue: %s\n", openmrcp_event_names[*event2queue]); + } + + return status; +} + + +static mrcp_status_t asr_on_session_initiate(mrcp_client_context_t *context, mrcp_session_t *session) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "asr_on_session_initiate called\n"); + asr_session_t *asr_session = mrcp_client_context_session_object_get(session); + if(!asr_session) { + return MRCP_STATUS_FAILURE; + } + return asr_session_signal_event(asr_session,OPENMRCP_EVENT_SESSION_INITIATE); +} + +static mrcp_status_t asr_on_session_terminate(mrcp_client_context_t *context, mrcp_session_t *session) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "asr_on_session_terminate called\n"); + asr_session_t *asr_session = mrcp_client_context_session_object_get(session); + if(!asr_session) { + return MRCP_STATUS_FAILURE; + } + return asr_session_signal_event(asr_session,OPENMRCP_EVENT_SESSION_TERMINATE); +} + +static mrcp_status_t asr_on_channel_add(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_client_channel_t *channel) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "asr_on_channel_add called\n"); + asr_session_t *asr_session = mrcp_client_context_session_object_get(session); + if(!asr_session) { + return MRCP_STATUS_FAILURE; + } + return asr_session_signal_event(asr_session,OPENMRCP_EVENT_CHANNEL_CREATE); +} + +static mrcp_status_t asr_on_channel_remove(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_client_channel_t *channel) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "asr_on_channel_remove called\n"); + asr_session_t *asr_session = mrcp_client_context_session_object_get(session); + if(!asr_session) { + return MRCP_STATUS_FAILURE; + } + return asr_session_signal_event(asr_session,OPENMRCP_EVENT_CHANNEL_DESTROY); +} + +/** this is called by the mrcp core whenever an mrcp message is received from + the other side. */ +static mrcp_status_t asr_on_channel_modify(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_message_t *mrcp_message) +{ + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "asr_on_channel_modify called\n"); + asr_session_t *asr_session = mrcp_client_context_session_object_get(session); + if(!asr_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "!asr_session\n"); + return MRCP_STATUS_FAILURE; + } + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mrcp msg body: %s\n", mrcp_message->body); + + if (!strcmp(mrcp_message->start_line.method_name,"RECOGNITION-COMPLETE")) { + asr_session->mrcp_message_last_rcvd = mrcp_message; + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "setting FLAG_HAS_TEXT\n"); + switch_set_flag_locked(asr_session, FLAG_HAS_TEXT); + } + else if (!strcmp(mrcp_message->start_line.method_name,"START-OF-SPEECH")) { + asr_session->mrcp_message_last_rcvd = mrcp_message; + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "setting FLAG_BARGE\n"); + switch_set_flag_locked(asr_session, FLAG_BARGE); + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "ignoring method: %s\n", mrcp_message->start_line.method_name); + } + + return asr_session_signal_event(asr_session,OPENMRCP_EVENT_CHANNEL_MODIFY); +} + + + +static mrcp_status_t tts_session_signal_event(tts_session_t *tts_session, openmrcp_event_t openmrcp_event) +{ + mrcp_status_t status = MRCP_STATUS_SUCCESS; + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Got event: %s\n", openmrcp_event_names[openmrcp_event]); + + // allocate memory for event + openmrcp_event_t *event2queue = (openmrcp_event_t *) switch_core_alloc(tts_session->pool, sizeof(openmrcp_event_t)); + *event2queue = openmrcp_event; + + // add it to queue + if (switch_queue_trypush(tts_session->event_queue, (void *) event2queue)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push event to queue\n"); + status = SWITCH_STATUS_GENERR; + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "pushed event to queue: %s\n", openmrcp_event_names[*event2queue]); + } + + return status; +} + +static mrcp_status_t tts_on_session_initiate(mrcp_client_context_t *context, mrcp_session_t *session) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "tts_on_session_initiate called\n"); + tts_session_t *tts_session = mrcp_client_context_session_object_get(session); + if(!tts_session) { + return MRCP_STATUS_FAILURE; + } + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_SESSION_INITIATE); +} + +static mrcp_status_t tts_on_session_terminate(mrcp_client_context_t *context, mrcp_session_t *session) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "tts_on_session_terminate called\n"); + tts_session_t *tts_session = mrcp_client_context_session_object_get(session); + if(!tts_session) { + return MRCP_STATUS_FAILURE; + } + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_SESSION_TERMINATE); +} + +static mrcp_status_t tts_on_channel_add(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_client_channel_t *channel) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "tts_on_channel_add called\n"); + tts_session_t *tts_session = mrcp_client_context_session_object_get(session); + if(!tts_session) { + return MRCP_STATUS_FAILURE; + } + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_CHANNEL_CREATE); +} + +static mrcp_status_t tts_on_channel_remove(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_client_channel_t *channel) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "tts_on_channel_remove called\n"); + tts_session_t *tts_session = mrcp_client_context_session_object_get(session); + if(!tts_session) { + return MRCP_STATUS_FAILURE; + } + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_CHANNEL_DESTROY); +} + +/** this is called by the mrcp core whenever an mrcp message is received from + the other side. */ +static mrcp_status_t tts_on_channel_modify(mrcp_client_context_t *context, mrcp_session_t *session, mrcp_message_t *mrcp_message) +{ + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "tts_on_channel_modify called\n"); + tts_session_t *tts_session = mrcp_client_context_session_object_get(session); + if(!tts_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "!tts_session\n"); + return MRCP_STATUS_FAILURE; + } + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mrcp msg body: %s\n", mrcp_message->body); + + return tts_session_signal_event(tts_session,OPENMRCP_EVENT_CHANNEL_MODIFY); +} + + + +static apr_status_t set_default_asr_options(openmrcp_client_options_t *options) +{ + mrcp_logger.priority = MRCP_PRIO_INFO; + options->proto_version = globals.asr_proto_version; + options->client_ip = globals.asr_client_ip; + options->server_ip = globals.asr_server_ip; + options->client_port = globals.asr_client_port; + options->server_port = globals.asr_server_port; + options->rtp_port_min = globals.rtp_port_min; + options->rtp_port_max = globals.rtp_port_max; + return APR_SUCCESS; +} + + +static apr_status_t set_default_tts_options(openmrcp_client_options_t *options) +{ + mrcp_logger.priority = MRCP_PRIO_INFO; + options->proto_version = globals.tts_proto_version; + options->client_ip = globals.tts_client_ip; + options->server_ip = globals.tts_server_ip; + options->client_port = globals.tts_client_port; + options->server_port = globals.tts_server_port; + options->rtp_port_min = globals.rtp_port_min; + options->rtp_port_max = globals.rtp_port_max; + return APR_SUCCESS; +} + + +/** + * Called back by openmrcp client thread every time it receives audio + * from the TTS server we are connected to. Puts audio in a queueu + * and it will be pulled out from read_tts + */ +static apr_status_t openmrcp_tts_write_frame(audio_sink_t *sink, media_frame_t *frame) +{ + tts_session_t *tts_session = sink->object; + media_frame_t *media_frame; + switch_byte_t *buffer; + size_t len; + + len = frame->codec_frame.size; + + /* create new media frame */ + media_frame = (media_frame_t *) switch_core_alloc(tts_session->pool, sizeof(media_frame_t)); + if (!media_frame) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "media_frame creation failed\n"); + return SWITCH_STATUS_MEMERR; + } + + /** + * since *frame might get freed by caller (true or false?), allocate a + * new buffer and copy *data into it. + **/ + buffer = (switch_byte_t *) switch_core_alloc(tts_session->pool, sizeof(switch_byte_t)*len); + if (!buffer) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate buffer\n"); + return SWITCH_STATUS_MEMERR; + } + buffer = memcpy(buffer, frame->codec_frame.buffer, len); + media_frame->codec_frame.buffer = buffer; + media_frame->codec_frame.size = len; + media_frame->type = MEDIA_FRAME_TYPE_AUDIO; + + + /* push audio to queue */ + if (switch_queue_trypush(tts_session->audio_queue, (void *) media_frame)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push audio to queue\n"); + return MRCP_STATUS_FAILURE; + } + + return MRCP_STATUS_SUCCESS; + +} + + +/** + * Called back by openmrcp client thread every time it receives audio + * from the TTS server we are connected to. Puts audio in a queueu + * and it will be pulled out from read_tts + +static apr_status_t openmrcp_tts_write_frame_NEW(audio_sink_t *sink, media_frame_t *frame) +{ + tts_session_t *tts_session = sink->object; + media_frame_t *media_frame; + switch_byte_t *buffer; + size_t len; + + len = frame->codec_frame.size; + + // create new media frame + media_frame = (media_frame_t *) switch_core_alloc(tts_session->pool, sizeof(media_frame_t)); + if (!media_frame) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "media_frame creation failed\n"); + return SWITCH_STATUS_MEMERR; + } + + // since *frame might get freed by caller (true or false?), allocate a + // new buffer and copy *data into it. + buffer = (switch_byte_t *) switch_core_alloc(tts_session->pool, sizeof(switch_byte_t)*len); + if (!buffer) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate buffer\n"); + return SWITCH_STATUS_MEMERR; + } + buffer = memcpy(buffer, frame->codec_frame.buffer, len); + media_frame->codec_frame.buffer = buffer; + media_frame->codec_frame.size = len; + media_frame->type = MEDIA_FRAME_TYPE_AUDIO; + + + // add audio to buffer + switch_mutex_lock(tts_session->audio_lock); + if (switch_buffer_write(tts_session->audio_buffer, buffer, len) == 0) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Could not write to buffer\n"); + } + switch_mutex_unlock(tts_session->audio_lock); + + switch_mutex_unlock(tts_session->audio_lock); + + if (switch_queue_trypush(tts_session->audio_queue, (void *) media_frame)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push audio to queue\n"); + return MRCP_STATUS_FAILURE; + } + + return MRCP_STATUS_SUCCESS; + +} +*/ + + +/** + * Called back by openmcp client thread every time its ready for more audio to send + * the recognition server we are connected to. Reads data that was put into a + * shared fifo queue upon receiving audio frames from asr_feed() + */ +static apr_status_t openmrcp_recognizer_read_frame(audio_source_t *source, media_frame_t *frame) +{ + asr_session_t *asr_session = source->object; + frame->type = MEDIA_FRAME_TYPE_NONE; + apr_status_t result; + media_frame_t *queue_frame = NULL; + + /* pop next media frame data from incoming queue into frame */ + if(asr_session->audio_queue) { + if (switch_queue_trypop(asr_session->audio_queue, (void *) &queue_frame)) { + // switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "could not pop from queue\n"); + result = MRCP_STATUS_FAILURE; + } + else { + frame->codec_frame.size = queue_frame->codec_frame.size; + frame->codec_frame.buffer = queue_frame->codec_frame.buffer; + frame->type = MEDIA_FRAME_TYPE_AUDIO; + result = APR_SUCCESS; + } + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "no audio queue\n"); + result = MRCP_STATUS_FAILURE; + } + + return result; +} + +/** Read in the grammar and construct an MRCP Recognize message that has + The grammar attached as the payload */ +static mrcp_status_t openmrcp_recog_start(mrcp_client_context_t *context, asr_session_t *asr_session, char *path) +{ + + mrcp_generic_header_t *generic_header; + apr_status_t rv; + apr_file_t *fp; + apr_pool_t *mp; + apr_finfo_t finfo; + char *buf1; + apr_size_t bytes2read = 0; + + mrcp_message_t *mrcp_message = mrcp_client_context_message_get(context, asr_session->client_session, asr_session->channel, RECOGNIZER_RECOGNIZE); + + if(!mrcp_message) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not create mrcp msg\n"); + return MRCP_STATUS_FAILURE; + } + + /* open the file with the grammar and read into char* buffer */ + mp = mrcp_message->pool; + if ((rv = apr_file_open(&fp, path, APR_READ, APR_OS_DEFAULT, mp)) != APR_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not read grammar\n"); + return -1; + } + rv = apr_file_info_get(&finfo, APR_FINFO_NORM, fp); + + buf1 = apr_palloc(mp, finfo.size); + bytes2read = finfo.size; + rv = apr_file_read(fp, buf1, &bytes2read); + generic_header = mrcp_generic_header_prepare(mrcp_message); + if(!generic_header) { + return MRCP_STATUS_FAILURE; + } + + generic_header->content_type = mrcp_str_pdup(mrcp_message->pool,"application/srgs+xml"); + mrcp_generic_header_property_add(mrcp_message,GENERIC_HEADER_CONTENT_TYPE); + mrcp_message->body = mrcp_str_pdup(mrcp_message->pool,buf1); + + /* send the MRCP RECOGNIZE message to MRCP server */ + return mrcp_client_context_channel_modify(context, asr_session->client_session, mrcp_message); + +} + + +/** + * Freeswitch calls this from switch_ivr_detect_speech() and then adds a media + * bug to tap into the channel's audio, which will result in all data getting + * passed to asr_feed() and calls to asr_check_results() on each recevied frame. + * + * This code expects certain one-time initialization of the openmrcp client + * engine/systeme to have already taken place.function to open the asr interface + */ +static switch_status_t openmrcp_asr_open(switch_asr_handle_t *ah, char *codec, int rate, char *dest, switch_asr_flag_t *flags) +{ + mrcp_client_context_t *asr_client_context = openmrcp_module.asr_client_context ; + asr_session_t *asr_session; + audio_source_t *source; + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "asr_open called, codec: %s, rate: %d\n", codec, rate); + + /*! + NOTE: According to the current FS media bugs design, the media bug can only feed audio + data in SLIN (L16) format. So we dont need to worry about other codecs. + + NOTE: forcing MRCP to use 20 as the CODEC_FRAME_TIME_BASE effectively ensures + that it matches with 16-bit audio at 8kz with 320 byte frames. in testing, leaving + CODEC_FRAME_TIME_BASE at 10 and using pop (instead of trypop) in + openmrcp_recognizer_read_frame() actually produces clean audio, however it causes + other problems as the full channel/session cleanup never completes in openmrcp, most + likely due to a thread being blocked on a pop call from audio queue. but with trypop + (to avoid the cleanup problem), it only produces clean audio when CODEC_FRAME_TIME_BASE + is set to 20. + */ + if (strcmp(codec,"L16")) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Sorry, only L16 codec supported\n"); + return SWITCH_STATUS_GENERR; + } + if (rate != 8000) { + // TODO: look into supporting other sample rates + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Sorry, only 8kz supported\n"); + return SWITCH_STATUS_GENERR; + } + if (CODEC_FRAME_TIME_BASE != 20) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "You must recompile openmrcp with #define CODEC_FRAME_TIME_BASE 20\n"); + return SWITCH_STATUS_GENERR; + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "CODEC_FRAME_TIME_BASE: %d\n", CODEC_FRAME_TIME_BASE); + } + + /* create session */ + asr_session = asr_session_create(); + if (!asr_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "asr_session creation FAILED\n"); + return SWITCH_STATUS_GENERR; + } + asr_session->client_session = mrcp_client_context_session_create(asr_client_context,asr_session); + if (!asr_session->client_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "asr_session creation FAILED\n"); + return SWITCH_STATUS_GENERR; + } + + /* create audio source */ + source = mrcp_palloc(asr_session->pool,sizeof(audio_source_t)); + source->method_set = &audio_source_method_set; + source->object = asr_session; + asr_session->source = source; + + /** + * create a new fifo queue. incoming audio received from freeswitch + * will be put into this queue, and it will be later pulled out by + * the openmrcp client thread. + */ + if (switch_queue_create(&asr_session->audio_queue, 10000, asr_session->pool)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "audio queue creation failed\n"); + return SWITCH_STATUS_MEMERR; + } + + asr_session->flags = *flags; + switch_mutex_init(&asr_session->flag_mutex, SWITCH_MUTEX_NESTED, asr_session->pool); + + ah->private_info = asr_session; + + return SWITCH_STATUS_SUCCESS; + +} + +/* function to load a grammar to the asr interface */ +static switch_status_t openmrcp_asr_load_grammar(switch_asr_handle_t *ah, char *grammar, char *path) +{ + /** Read grammar from path and create and send and MRCP RECOGNIZE msg + that has the grammar attached to body. + + TODO: - how does DEFINE-GRAMMAR fit into the picture here? (if at all) + */ + + asr_session_t *asr_session = (asr_session_t *) ah->private_info; + mrcp_client_context_t *asr_client_context = openmrcp_module.asr_client_context; + audio_source_t *source = asr_session->source; + + /* create recognizer channel, also starts outgoing rtp media */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Loading grammar\n"); + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Create Recognizer Channel\n"); + asr_session->channel = mrcp_client_recognizer_channel_create(asr_client_context, asr_session->client_session, source); + + if (!asr_session->channel) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to create recognizer channel\n"); + return SWITCH_STATUS_FALSE; + } + + /* wait for recognizer channel creation */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "WAITING FOR CHAN CREATE\n"); + if(wait_for_event(asr_session->event_queue, OPENMRCP_EVENT_CHANNEL_CREATE, OPENMRCP_WAIT_TIMEOUT) == MRCP_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Got channel creation event\n"); + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "WAITING FOR NONE\n"); + if (wait_for_event(asr_session->event_queue,OPENMRCP_EVENT_NONE,200) == MRCP_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "GOT NONE EVENT\n"); + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "NEVER GOT NONE EVENT\n"); + } + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Start Recognizer\n"); + openmrcp_recog_start(asr_client_context, asr_session, path); + + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "never got channel creation event\n"); + } + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Finished loading grammar\n"); + return SWITCH_STATUS_SUCCESS; +} + + +/*! function to feed audio to the ASR*/ +static switch_status_t openmrcp_asr_feed(switch_asr_handle_t *ah, void *data, unsigned int len, switch_asr_flag_t *flags) +{ + + asr_session_t *asr_session = (asr_session_t *) ah->private_info; + media_frame_t *media_frame; + switch_byte_t *buffer; + + /* create new media frame */ + media_frame = (media_frame_t *) switch_core_alloc(asr_session->pool, sizeof(media_frame_t)); + if (!media_frame) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "media_frame creation failed\n"); + return SWITCH_STATUS_MEMERR; + } + + /** + * since *data buffer might get freed by caller (true or false?), allocate a + * new buffer and copy *data into it. + * + * MAJOR DESIGN ISSUE!! It is way too expensive to be calling malloc on every audio frame, + * this needs to send the packet directly. OpenMrcp will need a way to disable their own + * timer so that the fs timer is the only one driving the media. + **/ + buffer = (switch_byte_t *) switch_core_alloc(asr_session->pool, sizeof(switch_byte_t)*len); + if (!buffer) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not allocate buffer\n"); + return SWITCH_STATUS_MEMERR; + } + buffer = memcpy(buffer, data, len); + + media_frame->codec_frame.buffer = buffer; + media_frame->codec_frame.size = len; + media_frame->type = MEDIA_FRAME_TYPE_AUDIO; + + /* push audio to queue */ + if (switch_queue_trypush(asr_session->audio_queue, (void *) media_frame)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "could not push audio to queue\n"); + return SWITCH_STATUS_GENERR; + } + + return SWITCH_STATUS_SUCCESS; +} + + +static switch_status_t openmrcp_asr_pause(switch_asr_handle_t *ah) +{ + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "openmrcp_asr_pause called\n"); + + return SWITCH_STATUS_SUCCESS; + +} + +static switch_status_t openmrcp_asr_resume(switch_asr_handle_t *ah) +{ + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "openmrcp_asr_resume called\n"); + + return SWITCH_STATUS_SUCCESS; +} + + +/*! function to unload a grammar to the asr interface */ +static switch_status_t openmrcp_asr_unload_grammar(switch_asr_handle_t *ah, char *grammar) +{ + + return SWITCH_STATUS_SUCCESS; +} + +/** + * Freeswitch calls this whenever the channel is hungup or the + * speech detection is stopped via a call to switch_ivr_stop_detect_speech() + */ +static switch_status_t openmrcp_asr_close(switch_asr_handle_t *ah, switch_asr_flag_t *flags) +{ + asr_session_t *asr_session = (asr_session_t *) ah->private_info; + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "openmrcp_asr_close()\n"); + + mrcp_client_context_t *context = openmrcp_module.asr_client_context; + + // TODO!! should we do a switch_pool_clear(switch_memory_pool_t *p) on the pool held + // by asr_session? + + // destroy channel + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Going to DESTROY CHANNEL\n"); + mrcp_client_context_channel_destroy(context, asr_session->client_session, asr_session->channel); + if (wait_for_event(asr_session->event_queue,OPENMRCP_EVENT_CHANNEL_DESTROY,10000) == MRCP_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "OPENMRCP_EVENT_CHANNEL_DESTROY received\n"); + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "timed out waiting for OPENMRCP_EVENT_CHANNEL_DESTROY\n"); + } + + // terminate client session + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Going to TERMINATE SESSION\n"); + mrcp_client_context_session_terminate(context, asr_session->client_session); + if (wait_for_event(asr_session->event_queue,OPENMRCP_EVENT_SESSION_TERMINATE,10000) == MRCP_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "OPENMRCP_EVENT_SESSION_TERMINATE recevied\n"); + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "timed out waiting for OPENMRCP_EVENT_SESSION_TERMINATE\n"); + } + + // destroy client session (NOTE: this sends a BYE to the other side) + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Going to DESTROY SESSION\n"); + mrcp_client_context_session_destroy(context, asr_session->client_session); + + // destroys the asr_session struct + asr_session_destroy(asr_session); + + switch_set_flag(ah, SWITCH_ASR_FLAG_CLOSED); + + return SWITCH_STATUS_SUCCESS; +} + + +/** + * Freeswitch calls this method from the speech_thread() thread body method + * in switch_ivr_async.c every time a new frame is received by the media bug + * attached to the audio channel. If this method returns SWITCH_STATUS_SUCCESS, + * then Freeswitch will call openmrcp_asr_get_results() to get the result value. + */ +static switch_status_t openmrcp_asr_check_results(switch_asr_handle_t *ah, switch_asr_flag_t *flags) +{ + asr_session_t *asr_session = (asr_session_t *) ah->private_info; + + switch_status_t rv = (switch_test_flag(asr_session, FLAG_HAS_TEXT) || switch_test_flag(asr_session, FLAG_BARGE)) ? SWITCH_STATUS_SUCCESS : SWITCH_STATUS_FALSE; + + return rv; +} + +/*! This will be called after asr_check_results returns SUCCESS */ +static switch_status_t openmrcp_asr_get_results(switch_asr_handle_t *ah, char **xmlstr, switch_asr_flag_t *flags) +{ + asr_session_t *asr_session = (asr_session_t *) ah->private_info; + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "openmrcp_asr_get_results called\n"); + switch_status_t ret = SWITCH_STATUS_SUCCESS; + + if (switch_test_flag(asr_session, FLAG_BARGE)) { + switch_clear_flag_locked(asr_session, FLAG_BARGE); + ret = SWITCH_STATUS_BREAK; + } + + if (switch_test_flag(asr_session, FLAG_HAS_TEXT)) { + /*! + we have to extract the XML but stripping off the <?xml version="1.0"?> + header. the body looks like: + + Completion-Cause:001 no-match + Content-Type: application/nlsml+xml + Content-Length: 260 + + <?xml version="1.0"?> + <result xmlns="http://www.ietf.org/xml/ns/mrcpv2" xmlns:ex="http://www.example.com/example" score="100" grammar="session:request1@form-level.store"> + <interpretation> <input mode="speech">open a</input> + </interpretation> + </result> + */ + + char *marker = "?>"; // FIXME -- lame and brittle way of doing this. use regex or better. + char *position = strstr(asr_session->mrcp_message_last_rcvd->body, marker); + if (!position) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Bad result received from mrcp server: %s", asr_session->mrcp_message_last_rcvd->body); + ret = SWITCH_STATUS_FALSE; + } + else { + position += strlen(marker); + *xmlstr = strdup(position); + } + + // since we are returning our result here, future calls to check_results + // should return False + switch_clear_flag_locked(asr_session, FLAG_HAS_TEXT); + ret = SWITCH_STATUS_SUCCESS; + } + return ret; +} + + +static mrcp_status_t synth_speak(mrcp_client_context_t *context, tts_session_t *tts_session, char *text) +{ + + //buffer = (switch_byte_t *) switch_core_alloc(asr_session->pool, sizeof(switch_byte_t)*len); + char *text2speak; + const char xml_head[] = + "<?xml version=\"1.0\"?>\r\n" + "<speak>\r\n" + "<paragraph>\r\n" + " <sentence>"; + + const char xml_tail[] = "</sentence>\r\n" + "</paragraph>\r\n" + "</speak>\r\n"; + + size_t len = sizeof(xml_head) + sizeof(text) + sizeof(xml_tail); + text2speak = (char *) switch_core_alloc(tts_session->pool, len); + strcat(text2speak, xml_head); + strcat(text2speak, text); + strcat(text2speak, xml_tail); + + mrcp_generic_header_t *generic_header; + mrcp_message_t *mrcp_message = mrcp_client_context_message_get(context,tts_session->client_session,tts_session->channel,SYNTHESIZER_SPEAK); + if(!mrcp_message) { + return MRCP_STATUS_FAILURE; + } + + generic_header = mrcp_generic_header_prepare(mrcp_message); + if(!generic_header) { + return MRCP_STATUS_FAILURE; + } + + generic_header->content_type = mrcp_str_pdup(mrcp_message->pool,"application/synthesis+ssml"); + mrcp_generic_header_property_add(mrcp_message,GENERIC_HEADER_CONTENT_TYPE); + mrcp_message->body = mrcp_str_pdup(mrcp_message->pool,text2speak); + + return mrcp_client_context_channel_modify(context,tts_session->client_session,mrcp_message); +} + +static mrcp_status_t synth_stop(mrcp_client_context_t *context, tts_session_t *tts_session) +{ + mrcp_message_t *mrcp_message = mrcp_client_context_message_get(context,tts_session->client_session,tts_session->channel,SYNTHESIZER_STOP); + if(!mrcp_message) { + return MRCP_STATUS_FAILURE; + } + + return mrcp_client_context_channel_modify(context,tts_session->client_session,mrcp_message); +} + + +static switch_status_t openmrcp_tts_open(switch_speech_handle_t *sh, char *voice_name, int rate, switch_speech_flag_t *flags) +{ + + tts_session_t *tts_session; + audio_sink_t *sink; + mrcp_client_context_t *tts_client_context = openmrcp_module.tts_client_context ; + + /* create session */ + tts_session = tts_session_create(); + if (!tts_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "tts_session creation FAILED\n"); + return SWITCH_STATUS_GENERR; + } + tts_session->client_session = mrcp_client_context_session_create(tts_client_context,tts_session); + if (!tts_session->client_session) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "tts_session creation FAILED\n"); + return SWITCH_STATUS_GENERR; + } + + + /* create audio sink */ + sink = mrcp_palloc(tts_session->pool,sizeof(audio_sink_t)); + sink->method_set = &audio_sink_method_set; + sink->object = tts_session; + tts_session->sink = sink; + + + /** + * create a new fifo queue. + */ + if (switch_queue_create(&tts_session->audio_queue, 10000, tts_session->pool)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "audio queue creation failed\n"); + return SWITCH_STATUS_MEMERR; + } + + /* create mutex that will be used to lock audio buffer */ + switch_mutex_init(&tts_session->audio_lock, SWITCH_MUTEX_NESTED, sh->memory_pool); + + /* create audio buffer */ + if (switch_buffer_create_dynamic(&tts_session->audio_buffer, MY_BLOCK_SIZE, MY_BUF_LEN, 0) != SWITCH_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Write Buffer Creation Failed!\n"); + return SWITCH_STATUS_MEMERR; + } + + + tts_session->flags = *flags; + switch_mutex_init(&tts_session->flag_mutex, SWITCH_MUTEX_NESTED, tts_session->pool); + + sh->private_info = tts_session; + + return SWITCH_STATUS_SUCCESS; + +} + +static switch_status_t openmrcp_tts_close(switch_speech_handle_t *sh, switch_speech_flag_t *flags) +{ + tts_session_t *tts_session = (tts_session_t *) sh->private_info; + mrcp_client_context_t *tts_client_context = openmrcp_module.tts_client_context ; + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "synth_stop\n"); + synth_stop(tts_client_context,tts_session); // TODO + wait_for_event(tts_session->event_queue,OPENMRCP_EVENT_CHANNEL_MODIFY,5000); + + /* terminate tts session */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Terminate tts_session\n"); + mrcp_client_context_session_terminate(tts_client_context,tts_session->client_session); + /* wait for tts session termination */ + wait_for_event(tts_session->event_queue,OPENMRCP_EVENT_SESSION_TERMINATE,10000); + + /* destroy demo session */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "destroy tts_session\n"); + mrcp_client_context_session_destroy(tts_client_context,tts_session->client_session); + tts_session_destroy(tts_session); + + return SWITCH_STATUS_SUCCESS; +} + +static switch_status_t openmrcp_feed_tts(switch_speech_handle_t *sh, char *text, switch_speech_flag_t *flags) +{ + + tts_session_t *tts_session = (tts_session_t *) sh->private_info; + mrcp_client_context_t *tts_client_context = openmrcp_module.tts_client_context ; + audio_sink_t *sink = tts_session->sink; + + /* create synthesizer channel */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Create Synthesizer Channel\n"); + tts_session->channel = mrcp_client_synthesizer_channel_create(tts_client_context,tts_session->client_session,sink); + /* wait for synthesizer channel creation */ + if(wait_for_event(tts_session->event_queue,OPENMRCP_EVENT_CHANNEL_CREATE,5000) == MRCP_STATUS_SUCCESS) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Got channel create event\n"); + wait_for_event(tts_session->event_queue,OPENMRCP_EVENT_NONE,1000); // XXX: what are we waiting for?? + /* speak */ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Going to speak\n"); + synth_speak(tts_client_context,tts_session, text); + + } + else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Never got channel create event.\n"); + return SWITCH_STATUS_FALSE; + } + + return SWITCH_STATUS_SUCCESS; +} + +/** + * Freeswitch calls this when its ready to read datalen bytes of data. + * + * TODO: check the blocking flag passed in flags and act accordingly + * (see mod_cepstral.c) + */ +static switch_status_t openmrcp_read_tts(switch_speech_handle_t *sh, void *data, size_t *datalen, uint32_t *rate, switch_speech_flag_t *flags) +{ + media_frame_t *queue_frame = NULL; + tts_session_t *tts_session = (tts_session_t *) sh->private_info; + + if (switch_queue_trypop(tts_session->audio_queue, (void *) &queue_frame)) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "could not pop from queue\n"); + return SWITCH_STATUS_SUCCESS; + } + else { + memcpy(data, queue_frame->codec_frame.buffer, queue_frame->codec_frame.size); + *datalen = queue_frame->codec_frame.size; + *rate = 8000; + return SWITCH_STATUS_SUCCESS; + } + +} + + +static void openmrcp_flush_tts(switch_speech_handle_t *sh) +{ + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "flush_tts called\n"); +} + +static void openmrcp_text_param_tts(switch_speech_handle_t *sh, char *param, char *val) +{ + +} + +static void openmrcp_numeric_param_tts(switch_speech_handle_t *sh, char *param, int val) +{ + +} + +static void openmrcp_float_param_tts(switch_speech_handle_t *sh, char *param, double val) +{ + +} + +static switch_asr_interface_t openmrcp_asr_interface = { + /*.interface_name*/ "openmrcp", + /*.asr_open*/ openmrcp_asr_open, + /*.asr_load_grammar*/ openmrcp_asr_load_grammar, + /*.asr_unload_grammar*/ openmrcp_asr_unload_grammar, + /*.asr_close*/ openmrcp_asr_close, + /*.asr_feed*/ openmrcp_asr_feed, + /*.asr_resume*/ openmrcp_asr_resume, + /*.asr_pause*/ openmrcp_asr_pause, + /*.asr_check_results*/ openmrcp_asr_check_results, + /*.asr_get_results*/ openmrcp_asr_get_results +}; + +static switch_speech_interface_t openmrcp_tts_interface = { + /*.interface_name*/ "openmrcp", + /*.speech_open*/ openmrcp_tts_open, + /*.speech_close*/ openmrcp_tts_close, + /*.speech_feed_tts*/ openmrcp_feed_tts, + /*.speech_read_tts*/ openmrcp_read_tts, + /*.speech_flush_tts*/ openmrcp_flush_tts, + /*.speech_text_param_tts*/ openmrcp_text_param_tts, + /*.speech_numeric_param_tts*/ openmrcp_numeric_param_tts, + /*.speech_float_param_tts*/ openmrcp_float_param_tts, + +}; + +static switch_loadable_module_interface_t openmrcp_module_interface = { + /*.module_name */ modname, + /*.endpoint_interface */ NULL, + /*.timer_interface */ NULL, + /*.dialplan_interface */ NULL, + /*.codec_interface */ NULL, + /*.application_interface */ NULL, + /*.api_interface */ NULL, + /*.file_interface */ NULL, + /*.speech_interface */ &openmrcp_tts_interface, + /*.directory_interface */ NULL, + /*.chat_interface */ NULL, + /*.say_interface */ NULL, + /*.asr_interface */ &openmrcp_asr_interface +}; + + +static switch_status_t do_config(void) +{ + char *cf = "mod_openmrcp.conf"; + switch_xml_t cfg, xml, settings, param; + + if (!(xml = switch_xml_open_cfg(cf, &cfg, NULL))) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "open of %s failed\n", cf); + return SWITCH_STATUS_TERM; + } + + memset(&globals,0,sizeof(globals)); + + if ((settings = switch_xml_child(cfg, "settings"))) { + for (param = switch_xml_child(settings, "param"); param; param = param->next) { + char *var = (char *) switch_xml_attr_soft(param, "name"); + char *val = (char *) switch_xml_attr_soft(param, "value"); + + if (!strcasecmp(var, "asr_client_ip")) { + globals.asr_client_ip = val; + } else if (!strcasecmp(var, "asr_server_ip")) { + globals.asr_server_ip = val; + } else if (!strcasecmp(var, "asr_proto_version")) { + globals.asr_proto_version =(uint32_t) atoi(val); + } else if (!strcasecmp(var, "asr_client_port")) { + globals.asr_client_port = (uint32_t) atoi(val); + } else if (!strcasecmp(var, "asr_server_port")) { + globals.asr_server_port = (uint32_t) atoi(val); + } else if (!strcasecmp(var, "tts_client_ip")) { + globals.tts_client_ip = val; + } else if (!strcasecmp(var, "tts_server_ip")) { + globals.tts_server_ip = val; + } else if (!strcasecmp(var, "tts_proto_version")) { + globals.tts_proto_version =(uint32_t) atoi(val); + } else if (!strcasecmp(var, "tts_client_port")) { + globals.tts_client_port = (uint32_t) atoi(val); + } else if (!strcasecmp(var, "tts_server_port")) { + globals.tts_server_port = (uint32_t) atoi(val); + } else if (!strcasecmp(var, "rtp_port_min")) { + globals.rtp_port_min = (uint32_t) atoi(val); + } else if (!strcasecmp(var, "rtp_port_max")) { + globals.rtp_port_max = (uint32_t) atoi(val); + } + + } + } + + return SWITCH_STATUS_SUCCESS; +} + + +static switch_status_t mrcp_init() +{ + /*! + Perform one-time initialization of asr client library + */ + + mrcp_mem_pool_t *pool; + mrcp_client_event_handler_t *asr_event_handler; + mrcp_client_t *asr_client; + mrcp_client_context_t *asr_client_context; + openmrcp_client_options_t *asr_options; + + pool = mrcp_global_pool_get(); + asr_options = mrcp_palloc(pool,sizeof(openmrcp_client_options_t)); + asr_event_handler = mrcp_palloc(pool,sizeof(mrcp_client_event_handler_t)); + set_default_asr_options(asr_options); + + asr_event_handler->on_session_initiate = asr_on_session_initiate; + asr_event_handler->on_session_terminate = asr_on_session_terminate; + asr_event_handler->on_channel_add = asr_on_channel_add; + asr_event_handler->on_channel_remove = asr_on_channel_remove; + asr_event_handler->on_channel_modify = asr_on_channel_modify; + + // create asr client context, which to must be passed to client engine + asr_client_context = mrcp_client_context_create(&openmrcp_module,asr_event_handler); + if(!asr_client_context) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "asr_client_context creation failed\n"); + return SWITCH_STATUS_GENERR; + } + openmrcp_module.asr_client_context = asr_client_context; + + // this basically starts a thread that pulls events from the event queue + // and handles them + asr_client = openmrcp_client_start(asr_options,asr_client_context); + if(!asr_client) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "openasr_client_start FAILED\n"); + mrcp_client_context_destroy(asr_client_context); + return SWITCH_STATUS_GENERR; + } + + openmrcp_module.asr_client = asr_client; + + + /*! + Perform one-time initialization of tts client library + */ + + mrcp_client_event_handler_t *tts_event_handler; + mrcp_client_t *tts_client; + mrcp_client_context_t *tts_client_context; + openmrcp_client_options_t *tts_options; + + pool = mrcp_global_pool_get(); + tts_options = mrcp_palloc(pool,sizeof(openmrcp_client_options_t)); + tts_event_handler = mrcp_palloc(pool,sizeof(mrcp_client_event_handler_t)); + set_default_tts_options(tts_options); + + tts_event_handler->on_session_initiate = tts_on_session_initiate; + tts_event_handler->on_session_terminate = tts_on_session_terminate; + tts_event_handler->on_channel_add = tts_on_channel_add; + tts_event_handler->on_channel_remove = tts_on_channel_remove; + tts_event_handler->on_channel_modify = tts_on_channel_modify; + + // create tts client context, which to must be passed to client engine + tts_client_context = mrcp_client_context_create(&openmrcp_module,tts_event_handler); + if(!tts_client_context) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "tts_client_context creation failed\n"); + return SWITCH_STATUS_GENERR; + } + openmrcp_module.tts_client_context = tts_client_context; + + // this basically starts a thread that pulls events from the event queue + // and handles them + tts_client = openmrcp_client_start(tts_options,tts_client_context); + if(!tts_client) { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "opentts_client_start FAILED\n"); + mrcp_client_context_destroy(tts_client_context); + return SWITCH_STATUS_GENERR; + } + + openmrcp_module.tts_client = tts_client; + + + + return SWITCH_STATUS_SUCCESS; +} + +SWITCH_MODULE_LOAD_FUNCTION(mod_openmrcp_load) +{ + + /* connect my internal structure to the blank pointer passed to me */ + *module_interface = &openmrcp_module_interface; + + /* read config */ + do_config(); + + /* initialize openmrcp */ + mrcp_global_init(); + mrcp_init(); + + /* indicate that the module should continue to be loaded */ + return SWITCH_STATUS_SUCCESS; +} + +SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_openmrcp_shutdown) +{ + return SWITCH_STATUS_UNLOAD; +}

1 0

[commit] r5583 - freeswitch/trunk/src/mod/languages/mod_spidermonkey_core_db
by Freeswitch SVN 10 Aug '07

10 Aug '07

Author: anthm Date: Fri Aug 10 10:27:25 2007 New Revision: 5583 Modified: freeswitch/trunk/src/mod/languages/mod_spidermonkey_core_db/mod_spidermonkey_core_db.c Log: add close method needs testing Modified: freeswitch/trunk/src/mod/languages/mod_spidermonkey_core_db/mod_spidermonkey_core_db.c ============================================================================== --- freeswitch/trunk/src/mod/languages/mod_spidermonkey_core_db/mod_spidermonkey_core_db.c (original) +++ freeswitch/trunk/src/mod/languages/mod_spidermonkey_core_db/mod_spidermonkey_core_db.c Fri Aug 10 10:27:25 2007 @@ -73,17 +73,33 @@ return JS_FALSE; } -static void db_destroy(JSContext * cx, JSObject * obj) +static JSBool db_close(JSContext *cx, JSObject *obj, uintN argc, jsval *argv, jsval *rval) { struct db_obj *dbo = JS_GetPrivate(cx, obj); - + if (dbo) { - switch_memory_pool_t *pool = dbo->pool; if (dbo->stmt) { switch_core_db_finalize(dbo->stmt); dbo->stmt = NULL; } - switch_core_db_close(dbo->db); + if (dbo->db) { + switch_core_db_close(dbo->db); + dbo->db = NULL; + } + } + + return JS_TRUE; +} + +static void db_destroy(JSContext *cx, JSObject *obj) +{ + struct db_obj *dbo = JS_GetPrivate(cx, obj); + + if (dbo) { + switch_memory_pool_t *pool = dbo->pool; + jsval rval = JS_TRUE; + + db_close(cx, obj, 0, NULL, &rval); switch_core_destroy_memory_pool(&pool); pool = NULL; } @@ -119,6 +135,10 @@ struct db_obj *dbo = JS_GetPrivate(cx, obj); *rval = INT_TO_JSVAL(0); + if (!dbo->db) { + return JS_FALSE; + } + if (argc > 0) { char *sql = JS_GetStringBytes(JS_ValueToString(cx, argv[0])); char *err = NULL; @@ -153,6 +173,10 @@ struct db_obj *dbo = JS_GetPrivate(cx, obj); *rval = BOOLEAN_TO_JSVAL(JS_FALSE); + if (!dbo->db) { + return JS_FALSE; + } + if (dbo->stmt) { int running = 1; while (running < 5000) { @@ -180,6 +204,10 @@ char code[1024]; int x; + if (!dbo->db) { + return JS_FALSE; + } + snprintf(code, sizeof(code), "~var _dB_RoW_DaTa_ = {}"); eval_some_js(code, dbo->cx, dbo->obj, rval); if (*rval == JS_FALSE) { @@ -207,6 +235,10 @@ *rval = BOOLEAN_TO_JSVAL(JS_FALSE); + if (!dbo->db) { + return JS_FALSE; + } + if (dbo->stmt) { switch_core_db_finalize(dbo->stmt); dbo->stmt = NULL; @@ -229,6 +261,7 @@ static JSFunctionSpec db_methods[] = { {"exec", db_exec, 1}, + {"close", db_close, 0}, {"next", db_next, 0}, {"fetch", db_fetch, 1}, {"prepare", db_prepare, 0},

1 0

[commit] r5581 - freeswitch/trunk/src/mod/endpoints/mod_iax
by Freeswitch SVN 09 Aug '07

09 Aug '07

Author: anthm Date: Thu Aug 9 14:20:58 2007 New Revision: 5581 Modified: freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c Log: update Modified: freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c (original) +++ freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c Thu Aug 9 14:20:58 2007 @@ -767,6 +767,7 @@ { if (tech_pvt->iax_session) { iax_reject(tech_pvt->iax_session, msg->string_arg ? msg->string_arg : "Call Rejected"); + switch_set_flag(tech_pvt, TFLAG_HANGUP); } } break;

1 0

[commit] r5580 - in freeswitch/trunk/src/mod: endpoints/mod_iax languages/mod_spidermonkey
by Freeswitch SVN 09 Aug '07

09 Aug '07

Author: anthm Date: Thu Aug 9 13:57:14 2007 New Revision: 5580 Modified: freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c freeswitch/trunk/src/mod/languages/mod_spidermonkey/mod_spidermonkey.c Log: small tweak Modified: freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c (original) +++ freeswitch/trunk/src/mod/endpoints/mod_iax/mod_iax.c Thu Aug 9 13:57:14 2007 @@ -763,8 +763,17 @@ assert(tech_pvt != NULL); switch (msg->message_id) { + case SWITCH_MESSAGE_INDICATE_REJECT: + { + if (tech_pvt->iax_session) { + iax_reject(tech_pvt->iax_session, msg->string_arg ? msg->string_arg : "Call Rejected"); + } + } + break; case SWITCH_MESSAGE_INDICATE_ANSWER: - channel_answer_channel(session); + { + channel_answer_channel(session); + } break; default: break; Modified: freeswitch/trunk/src/mod/languages/mod_spidermonkey/mod_spidermonkey.c ============================================================================== --- freeswitch/trunk/src/mod/languages/mod_spidermonkey/mod_spidermonkey.c (original) +++ freeswitch/trunk/src/mod/languages/mod_spidermonkey/mod_spidermonkey.c Thu Aug 9 13:57:14 2007 @@ -1788,9 +1788,9 @@ channel = switch_core_session_get_channel(jss->session); assert(channel != NULL); - - CHANNEL_SANITY_CHECK(); - + /* you can execute some apps before you answer */ + /* CHANNEL_SANITY_CHECK(); */ + if (argc > 0) { const switch_application_interface_t *application_interface;

1 0

[commit] r5579 - freeswitch/trunk/src/mod/applications/mod_conference
by Freeswitch SVN 09 Aug '07

09 Aug '07

Author: mikej Date: Wed Aug 8 23:19:07 2007 New Revision: 5579 Modified: freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c Log: fix for MODAPP-25. Modified: freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c ============================================================================== --- freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c (original) +++ freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c Wed Aug 8 23:19:07 2007 @@ -4451,7 +4451,10 @@ switch_xml_t xml_kvp; assert(conference != NULL); - assert(xml_controls != NULL); + + if (!xml_controls) { + return status; + } /* parse the controls tree for caller control digit strings */ for (xml_kvp = switch_xml_child(xml_controls, "control"); xml_kvp; xml_kvp = xml_kvp->next) {

1 0

[commit] r5578 - freeswitch/trunk/src/mod/endpoints/mod_wanpipe
by Freeswitch SVN 08 Aug '07

08 Aug '07

Author: anthm Date: Wed Aug 8 10:37:19 2007 New Revision: 5578 Modified: freeswitch/trunk/src/mod/endpoints/mod_wanpipe/mod_wanpipe.c Log: add context option Modified: freeswitch/trunk/src/mod/endpoints/mod_wanpipe/mod_wanpipe.c ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_wanpipe/mod_wanpipe.c (original) +++ freeswitch/trunk/src/mod/endpoints/mod_wanpipe/mod_wanpipe.c Wed Aug 8 10:37:19 2007 @@ -116,6 +116,7 @@ int configured_spans; int configured_boost_spans; char *dialplan; + char *context; switch_hash_t *call_hash; switch_mutex_t *hash_mutex; switch_mutex_t *channel_mutex; @@ -387,6 +388,16 @@ globals.dialplan = strdup(dialplan); } +static void set_global_context(char *context) +{ + if (globals.context) { + free(globals.context); + globals.context = NULL; + } + + globals.context = strdup(context); +} + static int str2node(char *node) { @@ -446,6 +457,7 @@ static void set_global_dialplan(char *dialplan); +static void set_global_context(char *context); static int str2node(char *node); static int str2switch(char *swtype); static switch_status_t wanpipe_on_init(switch_core_session_t *session); @@ -1546,7 +1558,7 @@ NULL, NULL, (char *)modname, - NULL, + globals.context, pevent->ring.callednum))) { switch_channel_set_caller_profile(channel, tech_pvt->caller_profile); } @@ -2225,6 +2237,8 @@ globals.dtmf_off = atoi(val); } else if (!strcmp(var, "dialplan")) { set_global_dialplan(val); + } else if (!strcmp(var, "context")) { + set_global_context(val); } else if (!strcmp(var, "suppress-dtmf-tone")) { globals.suppress_dtmf_tone = switch_true(val); } else if (!strcmp(var, "ignore-dtmf-tone")) {

1 0

[commit] r5577 - freeswitch/trunk/src/mod/applications/mod_conference
by Freeswitch SVN 07 Aug '07

07 Aug '07

Author: mikej Date: Mon Aug 6 22:46:55 2007 New Revision: 5577 Modified: freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c Log: fix for MODAPP-17, using the +pin on the conference app args, with nothing following the + will now override and not prompt for a pin Modified: freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c ============================================================================== --- freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c (original) +++ freeswitch/trunk/src/mod/applications/mod_conference/mod_conference.c Mon Aug 6 22:46:55 2007 @@ -4047,7 +4047,7 @@ rl++; /* if this is not an outbound call, deal with conference pins */ - if (!switch_channel_test_flag(channel, CF_OUTBOUND) && conference->pin) { + if (!switch_channel_test_flag(channel, CF_OUTBOUND) && conference->pin && *(conference->pin)) { char pin_buf[80] = ""; int pin_retries = 3; /* XXX - this should be configurable - i'm too lazy to do it right now... */ int pin_valid = 0;

1 0