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Question regarding potential ADAU1761 SigmaDSP Application...

Team,

We have a customer here in town that wants to use the ADAU1761 SigmaDSP to meet the following application requirements.

The customer has a unit that handles long range, 2-way radio (LRR) and supplies an interface that has an audio in and audio out port for the end user. 

They need to build an interface card that will interface to the two audio ports so that they can do the following:

1. Receive audio from the LRR and output it to a set of speakers or a headset for the user
2. Receive audio from the user via a mic/headset and output the audio to the LRR
3. Take the audio from the mic and also output it (mix it) to the user’s headphones/speaker (will feedback/distoration be an issue?).  I think they want the end user to be able to hear his/her own voice.

They believe that several of these requirements will require mixing with the mic when outputting the audio, but also that the mic input alone will have to be routed out to the LRR (non-mixed, of course) to achieve #2.

This looks like the perfect application for the ADAU1761 SigmaDSP, but I wanted to confirm with you to make sure there is not something I am overlooking.

Thanks in advance for your help.

  • Team,

    Is there anyone that can help on this one? It has been several days and the customer asks for an update 1-2 times daily.

    For their customer demonstration, they are thinking that they could even use a devkit for the interim.  What's really important is that are able to provision the necessary audio paths.

    Audio In Channel 1 (mono) ---> Audio Out Channel 1 (to backpack, long range radio transmitter)
    Audio In Channel 2 + Mic In (mono) ---> Audio Out Channel 2 (handset earpiece)

    They feel they can wire in a I2C bus to setup the audio provisioning.  It also may be the case that if they can achieve the above paths, they may not require the I2S/SPI digital audio interface at this time (they may need it later for recording, however).  All the audio is voice quality.

    Mike

  • Based on my understanding of the customer’s request they’re looking to add a sidetone. During a voice call it feels more natural to have some attenuated version of your own voice fed to your ear. Is that a correct understanding?

    If so, then it would be possible to implement this functionality simply using the ADAU1361, which is a codec that features some programmable routing options. Take a look at the playback and record signal paths, shown below. The input signal can be fed to the ADC to be converted to a digital output and sent to the far-side talker. The input signal can simultaneously be routed to the output mixer (referred to as left or right input mixer in the playback signal path diagram). These can also be gained down appropriately before mixed with the DAC signals.

    If the DSP is necessary for other audio processing, then yes, the ADAU1761 is a viable option. In that case, the signal routing/mixing/conditioning can all be handled in the DSP.

  • Thanks Brett.  I will share this and Melissa's response  with the customer and let you know if they have any follow on questions.

  • Team,

    As it turns out, there is an additional function the customer will most likely incorporate that they wanted me to run past you.  Based on the previous dialog, they want to know if a Push-To-Talk provision could be incorporated into the CODEC as well?  That is, the side tone is only present when a Push-To-Talk button is pressed.

    Let me know if this is possible.

    Mike

  • Yes, you can use a GPIO to receive the signal from the button and you can route it to a multiplexer in order to activate/deactivate the signal path.

  • This question has been assumed as answered either offline via email or with a multi-part answer. This question has now been closed out. If you have an inquiry related to this topic please post a new question in the applicable product forum.

    Thank you,
    EZ Admin

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