Analog and audio

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)

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

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!

Hello.I would like to do the following but since I don't have the necessary skills I ask you. If possible, I prefer to control everything with the keys.I make a list of the functions that I would like to implement:ALARM SECTION- wake up in the morning with a piece of music chosen at random from a list of music files in memory (each time a different music file is chosen. I had thought of a clock radio but I think it is better this way).- daily or weekly alarm programming (i.e. the right to choose on which days the alarm should sound).- automatic update of summer time - winter time- buffer battery in case of power blackout (in this case I think the alarm with buzzer is better to consume little energy and not risk resetting the buffer battery)- alarm disablingAUDIO SECTION- audio player with output that goes to the external I2s DAC.four keys for playback, but I specify that each key corresponds t

I am currently using the HAL function ```HAL_I2S_Receive (I2S_HandleTypeDef * hi2s, uint16_t * pData, uint16_t Size, uint32_t Timeout)```in order to collect an input signal (currently being tested with a function generator 440 Hz sine wave). However, we noticed a couple of things: 1) The data being stored in every index of our uint16_t * pData buffer is radically different. The values do not follow a trend. We expected to see an increase and a decrease in the values as it reads sine wave, however, we just see random values ranging from 0 - (2^16).2) When we stop the signal from the function generator the buffer continues to get filled up. This behavior continues even after unplugging the function generator line from the input to the codec.MCU  - STM32F446RECODEC - PCM3060Why would the values in the buffer continue to be filled up, and why are they not showing a trend that would re

TDA7577, PIN16 pin is left floating to disable I2C bus mode, the speaker short-circuit fault is detected, PIN14-CD_OUT output is permanently low, and the fault still exists after power-on again, and the speaker is in no sound output protection state. May I ask TDA7577 in the I2C disabled mode, How to clear faults through hardware control and restore normal operation?

I am using a STM32F4 MCU, currently prototyping on a Nucleo-F446RE. I have a PCM3060 audio CODEC to do the analog to digital conversion. I am attempting to use the I2S protocol in order to interface the stm32 chip with the CODEC to process real time audio signals. I have been using stm32CubeIDE to set up the peripherals and their settings.Using I2C, I have configured the CODEC's ADC to act as a master, and the data coming out of it is 24Bit I2S format at 48KHz.On the STM32 side of things, I have the I2S peripheral set up as half-duplex slave receive, data format as 24Bit on a 32Bit data frame, and 48Khz audio frequency. I have been testing with inputting a low-frequency 440Hz sine wave to the codec and try to get some data coming out of the Digital Output pin. I have confirmed that the data coming out is correct by directly sending this ADC data out to the data in of the DAC, and seeing the analog output of the CODEC on an oscilloscope.