Analog and audio

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I want to converter 0-500uA to Respective voltage.

I can get the frequencies close but not exact. I currently have a circular DMA buffer that I put the data in as I receive it from the DFSDM which feeds the SAI. I have this working but since they aren't exactly the same frequency there is underrun or overrun issues. Is there a way to do this without an elastic buffer I'm missing? I'm using the STM32L476RG and currently developing with the Nucleo L476RG and the X-Nucleo-CCA02M development boards.

Hi,I am currently working on developping a guitar digital looping pedal. I following my last post, I have starting looking into using the 32F746G DISCOVERY for my app, as it includes enough RAM to store audio data and has some interesting audio features as well. Since so far my project has been developped on Nucleo-F446RE, I'm wondering how I will be able to transfer what's already made. Precision on what my prototype would do:Sample the electric guitar signal with ADC, and reproduce it in real time with DAC On the 1st press of a button, sampled data is saved in memory, while still being reproduced through DACOn 2nd press of the same button, the saved audio would start playing in a loop, while being combined with what is being played in real time.Can erase/overwrite a loop on the press of another button.Here are my principal concerns:The F746G interfaces to an audio codec via I2C,

Hi guys, I am using an STM32F446RE nucleo board with an electret microphone(Waveshare Sound Sensor v2). Currently I am trying to sample audio, store it in a buffer and output through the DAC to an amplifier to a headphone jack. I am sampling the ADC at a rate of 44.1 kHz and playing back at the same rate, the DAC and ADC are triggered using a timer and DMA is used to store and send audio using a buffer. if I connect headphones directly to the output of the microphone, there is no noise on the signal, however when listening to the audio that gets sampled by the ADC and played back through the DAC, there is a lot of noise on the signal. I am using 8 bit resolution. I measured the voltage on the ADC pin while the ADC is running and it measures to be around 0.55 V. However if i decrease the sampling rate the voltage sits at roughly zero. How can I reduce this noise? I have tried changing the sampling time cycles but that didn't help.TIA

I need to measure a signal with very low power, the signal can range anywhere between -1 and +10 Volts but i only need the 0 to 3volts for my ADC input(my battery is also 3V). is there an opamp that fits my requirements that is less than 50uA?

I use an opamp to amplify my signal for the internal interrupt of my MCU, but after i this i also need to measure the input signal of the opamp(V+) with an ADC to measure weight. I need to know if there will be any significant changes made to the signal because this can effect my results.

Diode ST ………..

Hello everyone,I'm trying to build a system where I could read an analog signal and save it into an SD CARD. Besides this, I need to process the signal to activate, or not, the ACC reader for a few seconds.To save the values, I decided to use two buffers for each peripheral. My idea is to save a mini batch of the raw signals, in fixed intervals, into the SD Card. So, when one buffer is being uploaded to the SD card, the other is being used to recall the values from adc or accAs beginner in MCU world, I'm looking for some advisers before start the hands-on. I drew the system based on a lot of projects on the internet, but still not sure about the architecture.The draft of the system is bellow...Thank you in advance!

Please tell me the RthJA of TS27L2CD (SO-8). missing in spec​It's very urgent so please reply quickly

I'm trying to run Audio_GrEq Audio Graphical Equalizer on the stm32f407vet6 microcontroller. The source of the input sound is i2s2, the output sound is i2s3. The SGTL5000 codec is connected to i2s2 and is configured to operate at a frequency of 48 kHz, 32 Bits. The PCM5102 DAC is connected to i2s3. i2s2 receives data in ring DMA mode. By interruption to fill half of the buffer, its transfer via i2s3 starts. The same thing happens with the interrupt on the full filling of the buffer. And it works, the data goes directly through the microcontroller.Now I want to use the Audio_GrEq Audio Graphical Equalizer. He took initialization with an example. And it goes without errors. I placed the greq_process function just before the transfer on i2s3. However, it does not work (GREQ_UNSUPPORTED_NUMBER_OF_BYTEPERSAMPLE error). I can not understand what is the reason?

Hey guys, I am using a STM32 nucleo f446re board and I have a waveshare sound sensor v2 that I want to use. So basically just want to record audio and store it on an SD card. I want to sample audio at 44100 Hz using 8 bits, should I use a timer to generate an interrupt every 1/44100 seconds to do the conversion or is there a better way to implement an exact sample rate. Thank you

would you please help me identify this smd component? it is used in a smps circuit.

for reading micro sd audio data (SPI), I need to choose a Audio IC with Vol +, Vol-, next and pre funtion. Anyone How has expereince with that?Thank you

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On a existing Electronic board, the following schematic is implemented:May I use it in single end mode with a serial capacitor between VS+ (or VS-) and the load connected to GND?May the other output VS- (or VS+) stay unconnected?What is the bias voltage on the output VS- or VS+? (VDD/2?)Thanks. Best regards.Sylvain

Good morning,I am working on a basic test bench application and considering the TS3022, unfortunately the inverting input voltage may go beyond the VCC rail (which I cannot avoid due to pre-existing architecture).I am wondering:is there integrated ESD diodes in the TS3022?if yes, can I use them as overvoltage protection?if yes, what is the max steady state current?Thank you for the feeback.Best Regards,Gauthier Lemey

I did this a fair while ago, so even I have struggled to remember how it works exactly (I wish I added more comments back then). The attached files test/demonstrate an FIR routine that takes less than one cycle per tap. But it is only an 8 tap filter (with 8 taps I can fit all the delays and coefficients in registers). I'm not sure how useful it is. If someone knows the maths of how to string them together to create a more powerful filter then it would be of use. After all, simple analog filters are cascaded to create more powerful filters - why can't it be done for simple FIR filters? DSP guru(s) required. The filter has low start and end overhead and can be easily cascaded.Please ensure you verify that it works properly before using with anything critical...

Real-time clock M41T62 per day leave for 6 minutes henceforth. I found out that the quartz frequency is about 32.9 kHz. Program calibration does not reduce this inaccuracy. Do I have a defective chip or I don’t understand something?

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Hello,i want to use GM68020H to research ,but i can't find out its detail data sheet ,can you send one to me?My email :2664822931@qq.comThank you very much!

I have used STPM34 In my design. I am measuring voltage and current of single phase supply. Line is connected Pin number 9 of STPM34 and neutral is connected with circuit ground via ferrite bead. check R23 in schematic. I have attached schematicMy question is that what will be happen if i connected Line and neutral reverse?