Analog and audio

I am using a STM32G0C1E-EV board to try to get data from my microphone through the ADC and hear it in real time on my headphones via DAC. I have used a timer in order to get data in 16000 Hz format and I have used DMA circular buffers to get and transmit the data from the ADC to the DAC. The problem is that I only listen noise being outputted even though the data is getting through properly. I am wondering if the configuration I made for the ADC and DMA/Timer is properly done. I am posting the code in case somebody sees the problem. Thank you in advance.

I am unable to "select" one of these 2 radial buttons. This did work before, made a few modifications, now it gives this error. Anyone know how to get it to use my selection?

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I have designed a TDA7575B based audio amplifier, see attached screenshot image. The tone control as configured, will the input signal not be below standard 2v line input voltage that the TDA would typically be used with? I dont know how to add a preamp. Can someone please clarify or help me to get it designed. I apologies and I am new to electronics.

The Evaluation Board for STUW81300 Loop Filter was designed for 50MHz detector frequency. We will use 10MHz detector frequency. Can we still use the same Loop Filter as in EVB?If we can not, how can we design the Loop Filter (with same 60KHz loop bandwidth, ~55degree loop phase-margin, 3mA charge-pump) with 10MHz detector frequency?We want to generate stable 16GHz RF signal from STUW81300 chip with 10MHz reference clock.Thank you in advance.Regards,Tim

I would like to power STuW81300 forever, should I connect Pin4 PW-PD to ground?Thank youRegards,Tim

Here is the datasheet :TS881 - Rail-to-rail 0.9V nanopower, push-pull single comparator - STMicroelectronicsDoes it mean I can use it in single power supply with VS- connected to GND?

Hello,I'm working on a data acquisition project and i'm looking to use the AD7768 ADC.The datasheet says it has a 8 channel data interface, but i cant really find out what data protocol it is. Is it just plain data bits? if yes, can i just read it with I2S, uart, or reading it bit for bit???Does anyone have experience with the ad7768 data interface?Greetings,Angus Schmaloer

The RHF350A has nice properities. Is the a "normal" companion?

Hello, I'm searching for a space grade/rad hard, low noise (<5nV/rtHz), high bandwidth opamp (>100 MHz) for cryogenic operation. SiGe opamps are generally known to perform favorably at cryogenic temperatures. At first glance the RHF350A has good specs for my application but it's not clear to me if it's a SiGe device or not.

As on the title. The data sheet does not inform this parameter so it is hard to define what is the maximum resistance on the feedback network that will produce an acceptable error.The data sheet informs that resistance R2 (bottom resistance) of the feedback network must be >= 5kOhms to minimize leakage current, but not the maximum value.Thanks in advance.

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Hi, I have an amplifier circuit build using TS9222 opamp, with gain = 40 and external EMI filter on the inputs. I want to understand the affect of filter capacitors at the input of opamp on stability of circuit.I have simulated the circuit using LTSPICE and file is attached.Observations:Pole in the feedback path is at ~366kHzphase margin is about 67 deg, And to counter the pole due to CL, output resistor is added outside the loop to add a pole-zero pair at ~56kHzBW of circuit is ~100KHzEven though the pole in the feedback path is at 366KHz but the phase change starts as usual decade before. And as per bode plot or ROC method I see that phase change started before 100KHz (which is bandwidth of circuit). Even though phase margin is 67deg, do I need to add resistors on inv and non inv terminals. I referred one app note which says that I should but I am not getting the math behind it. (below is the link)https://international.electronica-azi.ro/2017/05/09/using-op-amps-to-reduce-near-field-

Hello there, I am newbie to PCB design and I am going to use TDA7575B. Here I have an input from a mobile which have two wires. I have few question regarding its design:Can I joint In1+ to In2+, In1- to In2- ? then those joints will be connected to signal from phoneSecondly, CD_OUT, V, MUTE, ST-BY/HE pins connected where as its just a simple design.On Pin 17, 17ohm setting is written, what's its purpose?  

comparison between old location vs new location (same tooling? same process?)Your PCN CRP/21/11964 states dates from 2020 (see enclosed) ..... is it a mistake ?Please provide us information regarding the dates LTB and LTS.Please advise regarding data qualification and breaking-point dates because we are needing our customer approval to be able to purchase to you.Thanks and regards,Ligia MontenegroQuality LeaderCMAC Microcircuits[deleted email address]

Is it possible to run e.g. 2 x TDA7293 in bridge configuration on a 24 VDC single supply voltage as it can be done on many other ICs by feeding the non invertig input with 1/2 supply voltage ?

If it is a scaled fraction then how does it get filtered from transient changes of the Vdc supply?If it is a fixed constant voltage then the outputs will be biased at that same voltage and peak output audio will be clipped non-symmetrical. For example, if Vdc supply is 18Vdc and Vref is 5Vdc then the output can swing from 5 to 18V ( a change of 13V before clipping) and the output can swing from 5V to ground (a change of only 5V before clipping). This would severely limit the low distortion power rating.

I need to capture as soon as Audio is played. In real time without user intervention how shall we start the ADc conversion and how to validate the data is converted properly. Audio is w,r,t to sampling rate , if so I am using 8 KHz with DMA . If I play audio for 30 sec (8000 *30 *16 bit) = 480000 bytes and 468 Kb. If it is more time how shall we transmit without data loss. Is there is any max minutes based controller architecture design.What is the best sampling rate for normal voice or speech.@Will there is any specific sampling rate for different human acoustic generation like snezing, snoring ?

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Hi,I am currently implementing a guitar effects pedal using the STM32F446RE and the PCM3060 as my CODEC.I have gotten everything to work, except when I try to adjust the following parameters cf0_lim = (int)(time*l_CB0); cf1_lim = (int)(time*l_CB1); cf2_lim = (int)(time*l_CB2); cf3_lim = (int)(time*l_CB3); ap0_lim = (int)(time*l_AP0); ap1_lim = (int)(time*l_AP1); ap2_lim = (int)(time*l_AP2);inside my DMA. For some reason, I cannot update this parameters anywhere in my code. I tried updating them outside of the DMA callbacks for my CODEC and also on the while loop. I am pretty sure these are the parameters that are triggering the HardFault error because whenever I comment them out the code runs fine. I am able to update the "time" and "wet" variables whilst the DMA is running. This is fine, however, without updating the cfx_lim and apx_lim variables, I can't do much to apply this changes to the current effects that are being used for the reverb effect. Is there a reason why this is happe

In some training session they are using ADC calibration in the code , whether it is required ?Any sample reference code for Analog microphone step by step guide or Video. I have seen many Youtube Videos.

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I want to play an audio clip over i2s from Flash memory. My application is mono and a mono source will fit in flash, but not stereo, it is too big.The HAL i2s interface does not appear to offer a mechanism to duplicate mono source bytes (16b) onto the stereo stream. Am I missing something? (STM32F411)

I have my I2C bus connected to an STM6524, and I see that my rising edges are slowed down a bit more than expected. What is the input capacitance of these inputs? Thank you!