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Directed Text to Speech using Gentner AP800

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    Directed Text to Speech using Gentner AP800

    Ok gang,

    I need some creative help here. I think I have got to compliacated for my own good!

    Here is what I currently have and what I am trying to achieve as the end goal.

    Software:
    MainLobby with the Russound CAV6.6 PLugin and the HomeSeer plugin
    HomeSeer 2.2
    J River Media Center
    Hardware: (Home Automation Computer)
    Each channel of the SoundBlasters map to a source on the Russound
    SoundBlaster X-FI #1
    SoundBlaster #1 Front Channel maps to Russound CAV6.6 Source 1
    SoundBlaster #1 Center Channel maps to Russound CAV6.6 Source 2
    SoundBlaster #1 Rear Channel maps to Russound CAV6.6 Source 3
    SoundBlaster X-FI #2
    SoundBlaster #2 Front Channel maps to Russound CAV6.6 Source 4
    SoundBlaster #2 Center Channel maps to Russound CAV6.6 Source 5
    SoundBlaster #2 Rear Channel (Not used)
    Russound XM Tuner maps to Russound CAV6.6 Source 6

    JDS Stargate used to turn relay on/off connected to Russound Page to more precisely control Russound Page

    Gentner AP800 (Voice Recognition)
    8 Mics Around the house tied into the Gentner
    Text to Speech from Home Automation Computer tied into Gentner AP800
    Mic in to Home Automation Computer from Genter AP800

    So, with this setup, I am able to listen to 5 different music sources coming from the J.River Media Center. I am also able to play XM on source 6 on the Russound as well. When a phone call comes in, I am able to interrupt all six Russound sources by using the Stargate to enable the relay to the Russound Page output. The text to speech output from the Home Automation computer announces the caller (the text to speech from the home automation computer is tied into the Gentner AP800). I have Westminster chimes also interrupting every hour playing the chimes. All of this works well.

    Now here is what I am now trying to add to the system. I am just now getting the voice recognition working using open air mics and the Gentner AP800 to work in my system. What I would like to be able to do is when a user speaks a voice recognition command in a room, I would like for the Text to Speech ouput to ONLY output to that room. Using the page feature of the Russound, all rooms will get the text to speech output, so that will not work. The only thing I can think of is getting rid of source 5 from the J-River and use that as an output from the Gentner AP800 and then issue commands to the CAV6.6 to tie source 5 of the Russound to the Gentner.

    I know I am probably making this more complicated than it needs to be.
    If I have to add more hardware, I am ok with that too. Just not sure what I need to add to make this all work.

    Any advice would be greatly appreciated.

    Thanks!
    Chuck

    #2
    the ap800 has a 25 pin connection, which has a logic of "high and low" . when a mic is gated one of these pins goes high or low (high or low can be programmed in the ap800), so you can use this logic to let you switch speaker thru your cav6 or thru a I/O board. Check your ap800 owners manual. for more info.
    Hector
    ____________________________________
    Win.2003 OS, HS3
    BLDSC,BLstat,BLRadar,BLRamdom,BLOccupied
    BLups,BLrain8,HSTouch,Ultrajones Weatherbug,
    MyTrigger,ACRF2,W800,Zwave
    AP800,Honeywell Stat

    Comment


      #3
      Chuck,
      That sounds like a very cool system. I do agree, however, that if you want to send an audio stream to one or a subset of rooms, it appears that you must use one of the dedicated inputs rather than the paging feature. But, I think you can do that using the AUX input (on the front). The CAV allows you to direct that input by zone as well, so it may provide a way to add another input without sacrificing what you have.
      Mike____________________________________________________________ __________________
      HS3 Pro Edition 3.0.0.548, NUC i3

      HW: Stargate | NX8e | CAV6.6 | Squeezebox | PCS | WGL 800RF | RFXCOM | Vantage Pro | Green-Eye | Edgeport/8 | Way2Call | Ecobee3 | EtherRain | Ubiquiti

      Comment


        #4
        AP800 IsSpeaking Setup

        Thanks Gang!

        That's the only way I could think to do it as well. That a good idea to use the Aux input. I think I will give it a try. The only thing I don't see at this point is how to address the AUX port on the Russound with the Cinemar CAV66 plugin.

        So, here is what I think I need to do,

        When a voice command is sent, I will perform the following steps:

        1) In the IsSpeaking routine, I will search for a match on the "Attention Acknowledge Phrase" set in the Voice Recognition options.
        2) I will look at the device status from dantelope's routines to see which Mic is gated (I could also use the Digital Inputs as I have them hooked up as well)
        3) I will then send an http command to MainLobby's CAV66 software to turn power onto on the proper zone (if not already)
        http://localhost:6246/command?MLServ...r~1~MicRoom~On
        4) I will save the current source selected for the selected zone for restoration later.
        5) I will then send an http command to MainLobby's CAV66 software to select source 5 on the CAV66. (I replace with the Aux zone when I figure out how to do that).
        http://localhost:6246/command?MLServeCmd.MLCAV66|ZoneSource~1~MicRoom~5
        6) After the IsSpeaking routine has completed, I will restore the original source to that zone?

        Question:

        How do I send these http commands in a script in HomeSeer.

        Thanks for all of the help guys!
        Chuck

        Comment


          #5
          I have no experience with MainLobby, so can't help there. One suggestion, if their software supports it, is to preserve the current source selection and switch sources before you turn the zone on.

          Similarly, if the zone was originally off, when you restore the zone to the previously selected source, switch the zone back off first, then restore source selection.
          Mike____________________________________________________________ __________________
          HS3 Pro Edition 3.0.0.548, NUC i3

          HW: Stargate | NX8e | CAV6.6 | Squeezebox | PCS | WGL 800RF | RFXCOM | Vantage Pro | Green-Eye | Edgeport/8 | Way2Call | Ecobee3 | EtherRain | Ubiquiti

          Comment


            #6
            Timing Problem with Gate Command on AP800?

            Ok, I almost have everything working. I just can't seem to get the proper mic status decoded correctly. I was wondering, when I start speaking the VR attention command and the I issue the Gate Command, how long does it take to get the status back and how long does the Mic stay gated?

            Thanks,
            Chuck

            Comment


              #7
              Cluelss in Gentner Land ... Again!

              Gang,

              I think I'm getting closer ... but then again maybe not ...

              I am having a really wierd problem here. Here is my setup.
              Gentner AP800
              4 Crown PZM11 (Ceiling Mics)
              1 Shure RS230 HandHeld Mic
              4 Crown Mics Connected to inputs 1-4
              Shure RS230 Handheld Mic connected to input 8

              Commands are accurately recognized from all 5 mics (different locations). I am trying to determine which Mic gated using the GATE command "#10 GATE". When I use the Shure Microphone the value returned from the GATE command in my code is perfect as follows:

              "#10 GATE 80"

              When I issue the GATE command from my software for the Crown PZM Mics, I always get back "#10 GATE 0", but the Mic is gating and the command is recognized!

              Now, if I move the Shure Mic to another Mic input line such as input #2, I get back
              "#10 GATE 2"

              It almost as if the Crown PZM Mics have enough to gate the Mic on and the command is recognized but for whatever reason not enough to trigger the gate output when I try to read it using the GATE command.

              I also tried to lengthen the hold time on the Mics to 1 sec thinking I was not catching the gated value in time. All settings between the Shure Mic and the Crown Mics are the same including Phantom Power.

              I am at a loss as to what is going on here.

              Any help would be greatly appreciated!

              Thanks,
              Chuck

              Comment


                #8
                Wow, I found the answer - Increase the HOLD time

                Gang,

                I found the answer just in case anyone runs across this.

                I'm not sure why, but the Shure Microphone seemed to keep the gate on longer than the Crown PZM Mics did. Increasing the hold time by another 1/2 sec to 1.5 was enough to get me over the hump for the Crown PZMs and I was able to read the Gate Status before it went away.

                Chuck

                Comment


                  #9
                  Upgraded from AP800 to XAP800

                  I am trying to echo cancel the Text to Speech (TTS) from my computer such that my mics don't pick it up, but I am not sure I have it setup corrently. Can you guys take a look at these screen shots from the XAP800 and let me know if I am doing this correctly.

                  Thanks!
                  Chuck






                  Comment

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