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Hi Christopher,<br>
<br>
The questions are cleared to me now. Many thanks for your
explanations!<br>
<br>
Best regards,<br>
<br>
Xing<br>
<br>
<br>
On 16/11/11 15:52, Christopher Rienzo wrote:
<blockquote
cite="mid:CAD-qRkhsPQYDHVS7xSq+hGtBKSsWf=_N1g6hcxDvFAZ9pSNJ1w@mail.gmail.com"
type="cite"><br>
Responses inline<br>
<br>
<br>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div text="#000000" bgcolor="#FFFFFF"> Now it works in my ESL
app though I am just able to do one dialogue ( I need to add
the event catching for furthur dialgoues).<br>
<br>
I have a couple of questions here:<br>
<br>
1. In the first try, my Nuance server was able to be
accessed somehow (FS says the MRCP is not responding in
5000ms, <br>
something like that), then FS says: [WARNING]
rtsp_client.c:386 () Failed to Connect to RTSP Server <a
moz-do-not-send="true" href="http://99.185.85.31:554"
target="_blank"><font color="red"><b>MailScanner warning: numerical links are often malicious:</b></font> 99.185.85.31:554</a>,<br>
later FS says: <br>
[ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel
error!<br>
[ERR] switch_ivr_play_say.c:2439 Invalid TTS module! <br>
<br>
The SYNTHESIZER channel error and Invalid TTS module
error are obvious.<br>
<br>
What I don't understand is why it went to this stange
address: <a moz-do-not-send="true"
href="http://99.185.85.31:554" target="_blank"><font color="red"><b>MailScanner warning: numerical links are often malicious:</b></font> 99.185.85.31:554</a>?<br>
</div>
</blockquote>
<div><br>
check your unimrcp configuration. Make sure the default TTS
and ASR profiles are set to actual servers.<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> 2. I specified TTS
engine in play_and_detect_speech as<br>
"say:unimrcp:nuance5-mrcp1-1: the text to speak"<br>
It works though I didn't specify the TTS voice.<br>
<br>
How do I specify the TTS voice? In the mrcp profile
(how?)? or something like:<br>
"say:unimrcp:nuance5-mrcp1-1:Serena: the text to
speak" (this seems not right.)<br>
</div>
</blockquote>
<div><br>
That won't work. Set the tts_engine variable as I explained
previously, or use say:unimrcp:voice:text to speak with the
desired voice and the correct default TTS profile defined in
unimrcp.conf.xml. This is a limitation of the say: notation.
Alternatively, the voice can be defined with the tts_voice
channel variable.<br>
<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> 3. The barge-in works
well, thanks!. Is the barge-in configurable? In some
scenarios, we might not allow barge-in.<br>
</div>
</blockquote>
<div><br>
If you don't want to barge in, just do "playback (or speak)"
first, then "play_and_detect_speech" with a silence prompt.<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
4. How could I get the text which has spoken to the user
when barge-in occurs?<br>
Or Could I get the time when barge-in occurs? If I know
the barge-in time and rough totale time for the whole text<br>
to be spoken I can figure out the spoken text by
manually checking the recorded audio file later, which would
be painful.<br>
</div>
</blockquote>
<div><br>
If this is necessary, you might want to use the lower-level
functions instead to watch for the begin-speaking event.<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
5. when I use "speak" and "detect_speech" apps in ESL, I
can catch event: DETECTED_SPEECH and speech-type:
begin-speaking<br>
and "detected-speech", then I do the recognition
results processing.<br>
<br>
The new app play_and_detect_speech seems not generate
these events any more. The way that I can think of to get
the results<br>
is to catch event:CHANNEL_EXECUTE_COMPLETE then check if
variable_current_application=play_and_detect_speech, then
get<br>
the results from variable_detect_speech_result.<br>
<br>
Is this the proper way to get the results in ESL app? Or
will play_and_detect_speech later on be consistent with
detect_speech <br>
in term of ASR events?<br>
</div>
</blockquote>
<div><br>
play_and_detect_speech is a higher level abstraction to
simplify things. If you want to have more control, go back to
using the ESL events. Reading the code in mod_dptools and
switch_ivr_async will give you hints about how to do it
correctly.<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
6. I'd like to set start-input-timers=false in the initial
request then start the recognition timers
(start-input-timers=true)<br>
after the TTS finishes.<br>
How possibly could I do this?<br>
</div>
</blockquote>
<div><br>
This is automatically done in the
switch_ivr_play_and_detect_speech() function. You just need
to specify start-input-timers=false in the beginning.<br>
</div>
</div>
<br>
<br>
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</pre>
</blockquote>
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