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TLV320ADC3101EVM-K: TLV320ADC3101:

Part Number: TLV320ADC3101EVM-K
Other Parts Discussed in Thread: TLV320ADC3101

Dear TI expert,

I am a new guy for ADC3101, and I have buy an evaluation borad of ADC3101, and I have several question needing support

1. what is the function difference of PPS and ADC3101 control software? 

2. I want to use PPS to develope a simple flow: low pass filter of the imcoming data with 1/2  Fs ----> decimation 2X of data after low pass---> absolute calculate of the data after decimation -----RMS value calculation, can you help draw a demo?

I found the decimation function seems like not supported by this chip.and it seems like the supported function of this miniDSP is very limited

3. how can I embedded my own algorithm into the miniDSP if PPS not support.

  • agong, 

    1. There are two difference use cases

    A. For users that want to make use of the MiniDSP. The control software is intended the help users with the input/output configuration and programming the PLL for clocks dissimilar from the EVM.  PPS does not provide a Graphical interface for these. 

    B. For users that are happy using the built in process flows (ie they do not need to program the MiniDSP) they can use the Control software exclusively.

    2.  Let me loop in one of our algorithms experts to comment on the capabilities here.

    3. Unfortunately the MiniDSP can only be programmed with PPS.  New components could theoretically be developed, but it is unlikely that this will happen. 

    best regards,

    -Steve Wilson

  • thanks for your reply.

    I want to use the miniDSP to to a absolut calculation, is that  possible to realise? looking forward to the reply of the algorithm expert

    in addition, I want to know whether the  ADC3100EVM can save the recording file into the computer when after press the "Recording " button in the control software, and how should I do the configuration?

  • Hi,

    Yes, the miniDSP library has a component to support absolute value calculation.

    The EVM can be used to capture ADC data using the USB Audio interface. Recorded audio can be captured on the PC using Audio editing software (e.g. Audacity). The software has menu items to save the file to disk in different formats like wav, mp3 etc.

    Best Regards.

  • Dear Dilgith,

    1. is the absolute component named  with "ScaleAbs" which is  with red cycle in the attached picture?

    2.  for the process flow, the first component is  framework, then followed by a decimator, in the picture, I set a 4x decimator, do the decimator have an anti-alias filter inside this component? because normally,  anti-alias filter should be put before the decimator.

    3. for the FIR inside the picture, I want to import the coefficient from a file by Matlab, could you pls tell what is the format of the file? or do you have an example?

    4. for the real application, I want to take a 10 second data-acquisition from the sensor and then do the calculation, how should I achieve this? for the EVM board, is there any memory where I can save the data buffer? if not, how can I visualise the data result of the time wave? for example, as the data process flow in the attached file, how can  I check the result after the FIR and ScaleAbs?

    5. for the real application , I want to switch the sampling frequency  between two different value, e.g, from 48K, to 8K and then go back to 48K. if so, should I power off the ADC power supply and re-write the registor value with the controll chip like MCU for each time of the switch?

    6 can  the ADC support such application of frequent power on and off?

  • Hi,

    1. Yes. ScaleAbs is the component to be used for absolute value.

    2. Yes. The device provides sufficient anti-aliasing filtering by a combination of analog and digital filtering. The TLV320ADC3101 integrates a second-order analog anti-aliasing filter with 20-dB attenuation at 1 MHz. The process flow is a representation of the digital processing and so the analog anti-aliasing filter is not shown. The decimator component is a digital filter that decimates and attenuates high frequencies. The pass band is from 0 to 0.39*Fs, where Fs is the sampling frequency. For other filter specifications, please refer to the ADC Digital Decimation Filter characteristics in the Electrical Char Table on page 7 of the datasheet.

    3. Please find attached an example FIR coefficient file with 64 taps. Valid values are within the range: [-1.0 1.0). If you have filter taps which are greater than 1.0 or less than -1.0, you should normalize them by dividing by the maximum absolute value.

    4. Please install Audacity for recording the I2S samples from the EVM. Audacity will record the audio samples and then display the waveform. You can also save the recorded samples into a *.wav file. The signal connected to the input port of MonoI2S_Out can be captured. If you would like just the output of FIR and not including ScaleAbs, then connect the output of FIR to MonoI2S_Out.

    5. The device supports switching between sampling rates. The ADC would have to be powered down and powered back up by the MCU. If you need different process flows at 8 kHz and 48 kHz, then the miniDSP registers would also have to be updated. In master mode, the device generates the clocks and so the sample rate can be changed by updating the register configuration. For slave mode, the clocks are input to the device and will have to be handled appropriately.

    6. Yes, frequent powering down and powering back up is supported by the device.

    Best Regards.

    b0 -1.000708456168 
    b1 -0.000680515110 
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  • Dear Diljith,

    thanks very much for your support. very helpful.

    In addition, several more questions :

    1. I am trying to play the sound on my PC which is recorded from the EVM, but can not hear, do you have any suggestion for the configuration for the PC and EVM board. the status is that the EVM can be recognised by the computer but just without sound out from the computer

    2. could you kindly explain more about the three Audio-Recording method? Line-in, external microphone, on-board microphone, what is the meaning and difference of each?

    3. in the componnents , there is a "splitter" componnet, can you explain more about this? is this components just a copy of the ADC data in to different process flow? 

    4. in the application, if I want to apply different algorithm for one data block from ADC, for examplem , 1 second of data acquisition, can I just use the splitter to do this?

    5. is there any third-party componnent which can do the FFT inside the DSP?

  • 1/2. I hope you are using the Control Software for generating your configuration. If not please install it. Please also go through the EVM user's guide to set up the EVM for recording. Refer to the QuickStart panel documentation (4.1.1.2) to begin with. It describes the different recording options and how to set them up. On-board microphones are on the EVM itself and can be used readily. External microphones have to interfaced to the ADC through the connector. Line inputs are obtained from other sound cards and have higher impedance than microphones.

    3. Splitter component creates multiple copies of the input. For e.g. a two-output Splitter creates two copies of the input.

    4/5. The miniDSP is designed for sample-based processing algorithms, i.e. algorithms that process one sample at a time. Algorithms that process a block of samples cannot be implemented as is and, wherever possible, have to be converted to sample-based processing for implementation on this device. One second of storage at 16 kHz will need 16000 memory words and is beyond the available data memory. For the same reason it is difficult to do FFT, which require grouping data into blocks of samples.

    Best Regards.

  • Dear Diljith,

    thanks for your reply.

    now the Audacity  can successfully recording the timewave form, but it seems like the timewave form is not from the DSP calculated result. but the raw data of the incoming.

    because I have already apply a Abs function in the DSP, so I suppose the timewave should only have the upper half, but the result is not like that but with the raw data displaying

    1. shoud I do any configuration if I want to see the result after the DSP calculation using the USB cable & Audacity software?

    2. as you said, the miniDSP in ADC3101 is very limited with the calculation power, do TI have the similar chip but which has more calculation ability and the same feature of low-power comsuption?

    3. I add a Probe-point algorithm in the process flw for data visulisation as instructed by the manual, but can not work, the data can not be displayed in the plot, is there any trick for the setting to make it display correctly?

    4. I have already installed control software, could you advise how can I generate and export the confiration script?

  • Dear  Expert,

    Looking forward to your reply. thanks

  • Hi,

    1. Please give us more information on how you got the basic recording working. Would you mind sharing the screen shot of the settings in control software?

    2. AIC3254 and AIC3262 are codecs with more processing capability. What kind of processing/algorithms is required by your application?

    3. Please share the process flow for me to reproduce the issue. You may share the process flow privately if you so desire.

    4. The configuration script can be recorded and then saved into a text file. In the Command Line Interface, you will see a record button on the bottom left. If you enable it all the device writes get recorded, which can then be saved using File-->Save Script File... option.

    Best Regards,

  • Hi,

    Let us know if you need further help on this. I shall close this for now. Please reply to reopen or you may start a new thread.

    Best Regards.

  • Dear Diljith,

    Many thanks for your reply.

    1. pls find the attached picture of  the recording setting

    2. As I said previously, FFT is our very important algorithm which we want to implement, because we are developing an edge-computing device. the application is not a real-time case, but we want the chip is low power consumption because we use the battery for power. 

    3. I have already sent you the flow privately

    4. I try to use the way you suggested but seems not to work when I change some settings, e.g. changing the bit of ADC. the script can not be updating accordingly.

  • Unfortunately, we do not have devices in our codec family that support FFT libraries. We would recommend a low-power DSP like C55 for such applications.

    Best Regards.

  • Dear Expert,

    1.for the process flow which I sent you in private, could you kindly help to check? thanks.It seems like the algorithm do not work the original raw waveform, can you pls use your EVM to verify?

    2. in addition, can you pls advise the way that I can combine the CS and PPS software? is it possible to record a script which both recording the configuration of the CS software and PPS software?

  • We shall respond on your newer thread which had queries on PPS and CS.

    Best Regards.