Analog and audio

Dear allBecause the STCS2A have separate pins for the FB and Source it is attempting to Think by manipulating the FB pin's voltage feedback to change the actual current through the leds.I tried to connect an operational amplifier between the source pin and the FB pin and changed the gain to "fool" the FB pin to think that therefrom normal current to too much (which would mean the current through drain/source should go down to match the 0.1V requirement) but I do not get a reliable result.Is there an easy solution to this that I have not seen?I choosed the STCS2A because it have a fault pin (DISC) and it is a "LED current driver".Regards

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Hello,​I want to perform AC analysis using LT Spice. I tried to find SPICE model for op-amp OA1ZHA but could not reach it. I would appreciate it if you told me how.Best,H. Sawano

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Could you explain me what is the "input Current" named in the figure 5 compare to the input bias current and input offset current?Thanks,Lucie

There is a leak path between IN+ and VREF, through the feedback resistors when VREF is connected to GND.I need to know whether this path is still possible when the IC has no VCC applied to it.I disable the IC to save power consumption but if this path is still present in the absence of VCC, then I need to take other measures.

I am trying to confirm a series of electrical measurements of the TSC2012H and have been successful in measuring positive and negative input bias current and both high and low voltage output swings, however have been unable to make the Voltage offset measurement within the spec defined in the data sheet (DS13057). I have used the suggested data sheet parameters, with a VCC of 5V, Vref1 = Vref2 = 2.5V and Vicm = 12V. The amplifier is in a unity gain configuration, however the offset voltages measured continue to be greater than that expected. Could you please advise on how to go about effectively making said measurements.

Hello,we plan to design a fully digital Class-D audio amplifier based on STA309A/STA311B and STA516BE, STA516B, STA510A, STA510F.Datasheets of the related parts use the terms "DDX" and "FFX". But I can found nowhere any explanation what thet technology is about.Could you please help on:What is the difference of DDX and FFX? Just different name for the same thing?What is the difference between DDX/FFX and ordinary PWM?Could I combine any DDX or FFX PWM generator with any DDX or FFX power stage? Or does DDX and FFX have to be used at both ends?Thanks a lot for your support.Kind regardsAndreas

How can I change the "htim2.init..period" using programming. Is it possible to change that ?

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I am using stm32g491ret6 for generating sin, triangle, sawtooth and square waveforms using LUTs.The frequency of the waveforms can be changed by changing the timer auto-reload register value. How I can change the auto-reload register) value programmatically?Please guide

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Hi, I have a problem with audio acquisition using PDM microphones. After using PDM_Filter, single peaks of high amplitude appear in the data in addition to the correct readings. I am using 4 microphones MP34DT01-M, STM32H743 and SAI interface. I take 4096 samples from each microphone, convert PDM to PCM and save to SD card using FATFS. My code (excerpt) is very simple:void HAL_SAI_RxCpltCallback (SAI_HandleTypeDef *hsai) //PDM buffer full { rxstate = 2; //buffer full flag real_round++; //next real round of acquisition }   int main(void) { HAL_SAI_Receive_DMA(&hsai_BlockA1, (uint8_t*)PDM_buffer, PDM_buffer_size); //start acquisition   while (1) { if (rxstate == 2)//if PDM buffer is full { rxstate = 0; //clear the flag current_round++; //next processed round of acquisition   if (current_round <= last_round) //processes and saves a predetermined number of cycles (rounds) { PDM_Filter(&(((uint8_t*)PDM_buffer)[0]), &(((uint16_t*)PCM_buffe

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Hey all,I copied this TDA7294 bridge circuit from the manuall:To this schematic: I have checked and double checked the schematic and it seems accurate to the manual's circuit. All parts are correct values to the manual's circuit. The PCB board is correct to the schematic that I made. All solder joints are done well without bridging. I am powering it with an 18V, 300mA power supply.I am using this power filtering circuit in to the power bridge amp's power circuit (substitute 18V for 9V).I am running it in to a 16ohm speaker load. The application is to test guitar pedal overdrive circuits. I am getting no sound and I can not figure out why. Based on this information, does anyone have any theories as to why? Based on the chip's specs, it should be able to be powered with 18V. Any help would be appreciated. Thank you!

1�?An IIS audio signal that feeds a sine wave2�?High power output3�?The audio waveform is abnormalI don't know where the configuration is wrong, please help with some suggestions, thank you.The desired audio waveform

Posted on January 29, 2018 at 05:48I've just managed to blow up 2 TDA7498E D Class amp chips when testing the first run of 25 production boards. This is the first time we have managed to damage any chips in over a year of prototype testing.We are using the chips in mono mode and have the left and right outputs tied together as per the datasheet (Mono mode is selected by connecting the mode pin 31 to VDSS pin 26). The damaged boards have the four VCC (+35V supply pins) blown off the chip (see attached photo). The chip blows when audio is first output during one of our factory tests (you get a small bang and a puff of smoke).I can get the same result by putting the chip into stereo mode by floating the Mode pin and playing audio (this causes the left and right amplifier outputs to short each other out). I also noticed that when the chip is in Standby the VDDS supply is turned off (e.g. at 0V).What I think may be causing the problem is taking the chip out of standby with audio

LM324QT (QFN16 package) was used in my design as an amplifier. The design has a dual-polarity power supply with +12V, GND and -12V.I connected its exposed centre pad to GND, while connecting the VCC+ to +12V and VCC- to -12V. However, the datasheet states that "The exposed pads of the QFN16 3x3 can be connected to VCC- or left floating".I have already got my board made and tested. What I have noticed is that the GND is not pulled to VCC- and VCC- is not pushed to GND, which seems right. However, I can feel the chips are quite hot.Does this design need to be totally abandoned or is this not the root case of the heat? Is the "exposed pad" internally connected to the VCC- via some low-resistance substrate or totally floating?

IRF520 type MOSFETs are starting to be phased out and so in all review, STP16NF06 should be a stable cross reference from what I have read, would anyone know perhaps is there a better cross for an IRF520 than the ST STP16NF06? i have ask OnSemi and their idea on this question is not much help as they point to MOSFETs that actually lower in current and PD

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I'm speaking about the X-Nucle-PLM01A1 Board that uses the ST7580 IC. I would like to use this IC in a DC Bus application, not AC Line, like most of the reference design does. On page 13 of UM2189 is shown how the boards are connected. My question is, why is the powerline wiring standalone? Why isn't the bus "riding" on top of the 12V VCC?In my application, i would like to do exactly that. Using 12V for Supply and having Powerline on the same 12V line, so the the device will only need two wires.Can i just connect PLM Clamp to VCC Clamp? Or is there any other thing needed? Thanks for your help

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Hello!After somet tests on Nucleo and Discovery I wan't to make some more complicated and practice project. I'm trying to make WAV file player from microSD card file.Hardware is based on STM32F103RCT6 connected with microSD card socket and DAC (PCM1780).Firstly - using FatFs I'm reading WAV file from microSD card to buffer, Secondly - I'm sending buffer to DAC using DMA in circular mode. When half of buffer is sended I load it from microSD (while second part of buffer is palyed). When second part of buffer is palayed I load first part, etc.Sound is deformed and too slow.When I connect oscilloscope I can see that audio signal is generated only in 1/3 time.When I checked LRCK i SDATA signals - reason is clear.Below you can see the most important code:main.c/* Private variables ---------------------------------------------------------*/ I2S_HandleTypeDef hi2s2; DMA_HandleTypeDef hdma_spi2_tx;   SPI_HandleTypeDef hspi1;   /* Private function prototypes ---------------------------