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throw errors when configuration is invalid. check for null point in asr_feed to prevent segfault when testing against certain asr servers. 

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5606 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Traun Leyden 2007-08-17 21:52:42 +00:00
parent 87d0979fd2
commit 1c0c21477d
1 changed files with 22 additions and 17 deletions
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39
src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c
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@ -24,6 +24,7 @@
* Contributor(s):
*
* Traun Leyden <tleyden@branchcut.com>
* Arsen Chaloyan <achaloyan@yahoo.com>
*
* Module which acts as an MRCP client to an MRCP speech recognition
* server. In other words it bridges freeswitch to an external speech
@ -404,19 +405,6 @@ static switch_status_t openmrcp_asr_open(switch_asr_handle_t *ah, char *codec, i
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;
@ -501,6 +489,9 @@ static switch_status_t openmrcp_asr_feed(switch_asr_handle_t *ah, void *data, un
media_frame.codec_frame.size = 160;
media_frame.codec_frame.buffer = data;
while(len >= media_frame.codec_frame.size) {
if (!audio_sink) {
return SWITCH_STATUS_GENERR;
}
audio_sink->method_set->write_frame(audio_sink,&media_frame);
len -= (unsigned int)media_frame.codec_frame.size;
@ -653,7 +644,6 @@ static mrcp_status_t synth_speak(mrcp_client_context_t *context, openmrcp_sessio
mrcp_generic_header_t *generic_header;
mrcp_message_t *mrcp_message;
//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"
@ -899,6 +889,9 @@ static switch_status_t do_config()
}
}
}
else {
goto error;
}
if ((profiles = switch_xml_child(cfg, "profiles"))) {
for (xprofile = switch_xml_child(profiles, "profile"); xprofile; xprofile = xprofile->next) {
const char *profile_name = switch_xml_attr_soft(xprofile, "name");
@ -963,10 +956,18 @@ static switch_status_t do_config()
}
}
}
else {
goto error;
}
switch_xml_free(xml);
return SWITCH_STATUS_SUCCESS;
error:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to load module configuration\n");
switch_xml_free(xml);
return SWITCH_STATUS_TERM;
}
static switch_status_t openmrcp_profile_run(openmrcp_profile_t *profile)
@ -1015,7 +1016,9 @@ static switch_status_t openmrcp_init()
switch_core_hash_init(&openmrcp_module.profile_hash,openmrcp_module.pool);
/* read config */
do_config();
if (do_config() != SWITCH_STATUS_SUCCESS) {
return SWITCH_STATUS_FALSE;
}
/* run default asr/tts profiles */
if(openmrcp_module.asr_profile) {
@ -1036,7 +1039,9 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_openmrcp_load)
mrcp_global_init();
/* initialize openmrcp */
openmrcp_init();
if (openmrcp_init() != SWITCH_STATUS_SUCCESS) {
return SWITCH_STATUS_FALSE;
}
/* indicate that the module should continue to be loaded */
return SWITCH_STATUS_SUCCESS;