Analog and audio

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 ---------------------------

Dear fellow Devs,I got a problem using the PDM to PCM library. While it seems to be working fine for a single conversion. There seems to be some kind of gap and noise between the results.The picture below shows a 440Hz sine wave.The class i wrote looks like that:class IM69D130_t { private: IM69D130_settings_t IM69D130_settings; static volatile bool Complete; static volatile bool HalfComplete; static volatile bool Spi_Overflow; void TransformAndTransferToRam(); bool CaptureComplete = false; static uint16_t Dataarray[DataArraySize]; int32_t Datalength = 32; static PDM_Filter_Handler_t PDMFilter; APS6404L_t* SRAM; SPI_HandleTypeDef* MicSPI; uint32_t currentaddress = 0; static PDM_Filter_Config_t PDM_FilterConfig; uint32_t TrueFreq = 48000000; public: IM69D130_t(APS6404L_t* SRamIn, SPI_HandleTypeDef* MicSPI); void ReStartCapture(); int32_t SetSamplesToCapture(uint32_t Samples16Bit); static void HalfCompleteCallback(); static void CpltCallback(); ui

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I need to fix an audio amplifier that uses a TDA7266M. Unfortunately, I can only find TDA7266 (stereo version) in stores in my city. I compared the pins and it seems that I can use one channel without any modifications. What should I do with a disconnected channel? Is it enough to connect the input via a capacitor to ground?

Im trying to build a guitar pedal on a STM32F446RE. 'SoundIN' is assigned to a pointer that is assigned to address value of the adc samples. 'SoundOUT' is half of soundIN (Just a test to see if works). However, output pointer '*outPoint' is showing a different value. When I input a sine wave of say 2Vpp, I get the exact same signal out. Shouldn't I be getting 1Vpp at the output ?Figure 1 shows the debugger output. Shouldn't 'outPoint' be a similar value to 'soundOUT' ? Figure 2 shows my double buffering code.

I'm facing the issue with recording an Audio via STBLE iOS APP. connected with blue tile chip.this one is not reflecting on Android. only getting iOS STBLE App is having chopped audio. this issue is also open on github. Please check this attached audio which I have recorded from STBLE iOS APP. Please anyone help me how to fix this.

We are developing a system which uses a Wide range air fuel sensor control IC "L9780" which is planned to be interfaced to the Bosch LSU4.9 Probe Sensor. We have worked previously with Bosch's CJ125 and planning on going over to use ST's L9780, but we need to fill in the gaps to proceed.In the datasheet ( the only document available ) The application notes is missing a lot of basic information.I've contacted ST Microelectronics, but seems they have nothing to help filling the gaps.I would appreciate if someone would kindly help us out. I'm willing to share more information if desired. My questions so far are as follows :1-   No details on how to calculate the Nernst cell impedance from the RCT2 amplifier readings2-   No information about the how the impedance measurement is used in the heater PID control programMore to come later if we can find an expert to answer or application notes.RegardsSultan

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HI, Project - Battery connected to load/charger with contactor on the positive terminal of the battery and current is monitored, whenever the current is above threshold contactor should be turned off.Objective is to whenever the current sensor output (DHAB 118, unipolar hall effect ) crosses the threshold comparator should give microcontroller an interrupt signal indicating the short circuit current. I have configured TS334 in window comparator configuration for detecting the overcurrent, wiring diagram of how its done is shown in the image attached. Problem being faced is comparator is giving false triggering as shown in the saleae capture. Kindly help me in solving this issue.Regards,Harish

I matched figure 3 in the datasheet with a level3a spice model and I can get a good match. When I used your provided model the crossover is VERY low. Also I don't understant how the datasheet has 80dB of DC gain, but the model and open loop response look like 40 some odd DB.

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I need to talk to someone at ST about the B-LCDAD-HDMI1. My company manufactures the ADV7533. I am available on LinkedIn and at analog.com.-Matt

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Preparing for some audio related project....If I understand correctly, I2S can transmit and receive simultaneously like SPI.Is this mode used practically? Are there codecs that can simultaneously transmit via I2S (generate sound) and receive digitized data (from a mic)?Or is this typically done using two separate I2S or SAI interfaces?

如果通过调节STA350BW输出的LC滤波器，�?�以将频�?延伸至40kHz�?�？输入I2S采样率为192kHz。Is it possible to set the frequency response of STA350BW to up to 40kHz by having a higher cut-off frequency LC filter? The I2S input frequency is set to 192kHz.

Hi There, I purchased an STEVAL-CCA044V1 development board to evaluate the TDA7498E device as an input source for characterising audio filters. I configured it for MONO mode as follows:placed J6placed J3placed J9placed J8connected the WR/L jumpers to connect the outputs.Powered with 28V input.Loaded with a 4R resistor.Input audio coming as a sine wave from signal generator.No matter what I feed into the input (100-2000Hz, 0-3Vpk-pk) all I get on the output is a ~39kHz approximately sinusoidal oscillation. This oscillation does stop when the sytem is muted or in standby (as per the Errata), implying that something is working as expected. But it doesn't appear to be functioning as an amplifier at all.Does anyone have experience using this eval board and have any advice as to what might be wrong here? Thanks, David

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Hi, am having an STM32F303K8 and using its DAC to generate a sinewave of a given frequency. Am using TIM6 update event to effect the results on the DAC periphery. AM running the APB1 bus at 16MHz, and my sinewave table has 256 samples. I set my prescalar to 0 and ARR to 2, and the closest i can produce is about 940Hz on the scope. I want to generate a Sinewave of 40KHz. I dont know what am doing wrong. Here are the relevenat pieces of my code. // DAC variables #define SINE_TABLE_SIZE 256 static uint16_t sinetable[SINE_TABLE_SIZE];   void generateSineTable(void) { for (int i = 0; i < SINE_TABLE_SIZE; i++) { sinetable[i] = (uint16_t)(2047.5 * (sinf(2 * M_PI * i / SINE_TABLE_SIZE) + 1.0)); } }   void configureDAC(void) { // Enable DAC clock RCC->APB1ENR |= RCC_APB1ENR_DAC1EN;   // Enable GPIOA clock RCC->AHBENR |= RCC_AHBENR_GPIOAEN;   // Configure PA4 as analog mode GPIOA->MODER |= GPIO_MODER_MODER4;   // Enable DAC channel 1 DAC->CR |=

I have a problem with the following:I built the single power supply circuit which mentioned in the datasheet (second picture). I use 20.5V Vs and 2x15W 8ohm speaker for output.I measured the Mute pin(5) with multimeter and always showed me 14V.It is enough for play mode. Nothing came out.Then tried to ground the Mute Pin also.The drawings are ok in the circuit.Anybody has an idea what should I try?st.com/resource/en/datasheet/tda7265.pdfPage 2/11 Fig 2: Typical Application Circuit in Single Supplythanks!