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return success from asr_feed when audio_sink hasnt yet been created. prevents a sporadic seg fault

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git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5650 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Traun Leyden 2007-08-21 16:48:56 +00:00
parent 1de25f3514
commit 31ae26d668
1 changed files with 13 additions and 4 deletions
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17
src/mod/asr_tts/mod_openmrcp/mod_openmrcp.c
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@ -179,7 +179,7 @@ static mrcp_status_t openmrcp_on_session_terminate(mrcp_client_context_t *contex
openmrcp_session_destroy(openmrcp_session); openmrcp_session_destroy(openmrcp_session);
} }
else { else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "abnormal session terminate\n"); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "abnormal session terminate\n");
} }
return MRCP_STATUS_SUCCESS; return MRCP_STATUS_SUCCESS;
} }
@ -289,6 +289,7 @@ static mrcp_status_t openmrcp_recog_start(mrcp_client_context_t *context, openmr
generic_header = mrcp_generic_header_prepare(mrcp_message); generic_header = mrcp_generic_header_prepare(mrcp_message);
if(!generic_header) { if(!generic_header) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Could not prepare generic_header\n");
return MRCP_STATUS_FAILURE; return MRCP_STATUS_FAILURE;
} }
@ -390,7 +391,15 @@ static switch_status_t openmrcp_asr_feed(switch_asr_handle_t *ah, void *data, un
media_frame.codec_frame.buffer = data; media_frame.codec_frame.buffer = data;
while(len >= media_frame.codec_frame.size) { while(len >= media_frame.codec_frame.size) {
if (!audio_sink) { if (!audio_sink) {
return SWITCH_STATUS_GENERR; /*!
If there is no audio_sink established yet, discard the audio and
return SUCCESS. returning anything other than success causes freeswitch
to abort the call and tear down the channel. One reason there might
not be an audio sink is that the openmrcp client lib is still in the startup
process.
*/
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "!audio_sink\n");
return SWITCH_STATUS_SUCCESS;
} }
audio_sink->method_set->write_frame(audio_sink,&media_frame); audio_sink->method_set->write_frame(audio_sink,&media_frame);
@ -445,7 +454,7 @@ static switch_status_t openmrcp_asr_close(switch_asr_handle_t *ah, switch_asr_fl
// terminate client session // terminate client session
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Going to TERMINATE SESSION\n"); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Going to TERMINATE SESSION\n");
switch_set_flag_locked(asr_session, FLAG_TERMINATING); switch_set_flag_locked(asr_session, FLAG_TERMINATING);
mrcp_client_context_session_terminate(context, asr_session->client_session); mrcp_client_context_session_terminate(context, asr_session->client_session);
switch_set_flag(ah, SWITCH_ASR_FLAG_CLOSED); switch_set_flag(ah, SWITCH_ASR_FLAG_CLOSED);
@ -661,7 +670,7 @@ static switch_status_t openmrcp_read_tts(switch_speech_handle_t *sh, void *data,
} }
if (!switch_test_flag(tts_session, FLAG_FEED_STARTED)) { if (!switch_test_flag(tts_session, FLAG_FEED_STARTED)) {
switch_set_flag(tts_session, FLAG_FEED_STARTED); switch_set_flag(tts_session, FLAG_FEED_STARTED); // shouldn't this use set_flag_locked? -tl
if(audio_source->method_set->open) { if(audio_source->method_set->open) {
audio_source->method_set->open(audio_source); audio_source->method_set->open(audio_source);
} }