[Freeswitch-users] How to implement TTS barge-in using FS ESL
xl127
x.liu at hw.ac.uk
Wed Nov 16 19:51:16 MSK 2011
Hi Christopher,
The questions are cleared to me now. Many thanks for your explanations!
Best regards,
Xing
On 16/11/11 15:52, Christopher Rienzo wrote:
>
> Responses inline
>
>
> 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).
>
> I have a couple of questions here:
>
> 1. In the first try, my Nuance server was able to be accessed
> somehow (FS says the MRCP is not responding in 5000ms,
> something like that), then FS says: [WARNING]
> rtsp_client.c:386 () Failed to Connect to RTSP Server
> 99.185.85.31:554 <http://99.185.85.31:554>,
> later FS says:
> [ERR] mod_unimrcp.c:1860 (TTS-6) SYNTHESIZER channel error!
> [ERR] switch_ivr_play_say.c:2439 Invalid TTS module!
>
> The SYNTHESIZER channel error and Invalid TTS module error are
> obvious.
>
> What I don't understand is why it went to this stange address:
> 99.185.85.31:554 <http://99.185.85.31:554>?
>
>
> check your unimrcp configuration. Make sure the default TTS and ASR
> profiles are set to actual servers.
>
> 2. I specified TTS engine in play_and_detect_speech as
> "say:unimrcp:nuance5-mrcp1-1: the text to speak"
> It works though I didn't specify the TTS voice.
>
> How do I specify the TTS voice? In the mrcp profile (how?)?
> or something like:
> "say:unimrcp:nuance5-mrcp1-1:Serena: the text to speak"
> (this seems not right.)
>
>
> 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.
>
> 3. The barge-in works well, thanks!. Is the barge-in
> configurable? In some scenarios, we might not allow barge-in.
>
>
> If you don't want to barge in, just do "playback (or speak)" first,
> then "play_and_detect_speech" with a silence prompt.
>
>
> 4. How could I get the text which has spoken to the user when
> barge-in occurs?
> 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
> to be spoken I can figure out the spoken text by manually
> checking the recorded audio file later, which would be painful.
>
>
> If this is necessary, you might want to use the lower-level functions
> instead to watch for the begin-speaking event.
>
>
> 5. when I use "speak" and "detect_speech" apps in ESL, I can
> catch event: DETECTED_SPEECH and speech-type: begin-speaking
> and "detected-speech", then I do the recognition results
> processing.
>
> 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
> is to catch event:CHANNEL_EXECUTE_COMPLETE then check if
> variable_current_application=play_and_detect_speech, then get
> the results from variable_detect_speech_result.
>
> 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
> in term of ASR events?
>
>
> 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.
>
>
> 6. I'd like to set start-input-timers=false in the initial
> request then start the recognition timers (start-input-timers=true)
> after the TTS finishes.
> How possibly could I do this?
>
>
> 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.
>
>
>
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