Analog and audio

I need to design a comparator with Vih=1.3V and Vil=1.2V. Therefore hysteresis is 100mV. The circuit will be working on a 3.3V supply. During high state, the o/p of the comparator will be 3.3V and at the low state, it will be 0V. I will be using a comparator with a push-pull output configuration. Please see my design calculations in the file attached. I would like to know two things1)May I know my design calculation is correct or not.2)For finding Vref where I need to plug the value of R1 and R2.In equation 1 or 2RegardsHARI

现在有一个项目上用了LM358�?放，�?�电压跟�?器用，用在电机上的，现在出现的问题如下：358是15V供电�?�了一百多�?��?��?，有差�?多五六个出现了下述情况，�?良的芯片输入电压V+为0V时输出电压有�??几�?，�?良的芯片正常工作时，输出电压�?�是正常的，�?�是输入电压是多少输出电压就是多少。项目比较急，麻烦尽快帮分�?下原因​原�?�图如下:​

Dears, Good morning.Regarding TDA7419's spectrum analyzer, the datasheet says:"A reset of the sample capacitors is induced whenever SAclk remains high for the time tintres" From the tables, Tintres is 3ms.Question: Is there a way to partially discharge the sample capacitor? I would like, for example, do 10 reads and for each one obtain decay of 10%. Maybe if I do each SAclk high for 300us (1/10 of Tintres) I can achieve this behavior for each reading cycle.Thank you very much,Leonardo.

Is commercial product part # supposed to match exactly with the finished goods part# in its basic part (USBLC6-2P6), would you explain the finished product code with “/J�? means in this case?

would you check and explain what is different in between USBLC6-2P6 and USBLC6-2P6/J

Hi all!I've been strugling with this TS4909 for 2 weeks now, and cannot get it work properly...I've designed a custom application with TS4909 as audio amp.The problem is is has high white noise on the output.My power supply is clean (from LDO), no noise on it, also tried to directly connect a 3,6V lithium bat on it, with nothing else connected from the circuit, also still the same.This White noise is present even if the inputs are shunted.My design follows the AN and datsheet design.Any help solving this?thanks

..

Hello everyone,I'm working on nucleo-f446re and x-nucleo-cca01m1 demoboard, using last version of FP-AUD-SMARTMIC1 software.The STM32F4 sends audio data to two speakers through the STA350BW via I2S interface.The output sampling frequency is 32kHz. I2S MCLK frequency is 1MHz and each sample has 16 bits./* Prepare hAudioOut handle */ mx_i2s_config.AudioFreq = AudioOutCtx[Instance].SampleRate; mx_i2s_config.CPOL = I2S_CPOL_LOW; mx_i2s_config.DataFormat = I2S_DATAFORMAT_16B; mx_i2s_config.MCLKOutput = I2S_MCLKOUTPUT_ENABLE; mx_i2s_config.Mode = I2S_MODE_MASTER_TX; mx_i2s_config.Standard = I2S_STANDARD_PHILIPS; #if (!defined(USE_STM32F0XX_NUCLEO) && !defined(USE_STM32L0XX_NUCLEO)) mx_i2s_config.ClockSource = I2S_CLOCK_PLL; #endifInitially the I2S buffer is empty. After a while, the buffer is filled with data and speakers reproduce an audio signal different than zero. In the image below, the blue line represents the I2S SD, and the purple one represents the output analog sig

Hello,I am using +/-40V to power TDA7293 and have no issues with its application. I want to power an audio processor from the +40 supply rail. The audio processor needs 3.3V.Kindly suggest a scheme to achieve this.I am concerned about the power dissipation in the 3.3V regulator and power sequencing.I would like to retain using the mute/stby ckt of the amplifier.Thanks in advance!Ravindra.

would you let me know MTBF or FIT value of LMV321ILT??thank you.

When I googled this, I only found documentation for SAI peripheral slave interface which indicates SAI slave works without MCLK in I2S mode.Nothing about I2S peripheral slave interface, though.I am working with the STM32F411CEU6, which has I2S but not SAI.

I am looking for a audio processor that will provide 8 channel automatic mixing. That is, when someone is speaking at a certain threshold on a given mic, the other mics are muted or gain is significantly reduced.

Hi. I am planning to use TDA7498E in one of my projects and was going through the datasheet of the same and got confused at one point. In the absolute maximum ratings, it is written that the INPA, INNA, INPB, INNB have the maximum input rating of -0.3V to +4V.The thing which got me confused is that since the analog audio signal is more or less like a sine wave with equal peaks both in positive and negative, then how are we supposed to provide the input to the TDA7498E?

Hello all,I am trying to draw the tda7491lp footprint (powersso-36 package), but am unable to find any documents/information for it. In the datasheet there is a package outline drawing, where I am able get most of the information needed, but there are no pastemask/soldermask specifications. Also I am under the assumption that those values are the pin dimensions and not the size of the landing pads. Am I supposed to go off the sdk board gerbers for the footprint?Thanks,Alex

..

In the datasheet for the TSV711 is sated (p20): 4.8 Macromodel Accurate macromodels of the TSV71x devices are available on the STMicroelectronics’ website at www.st.com.So, where are they??? I looked everywhere, Digikey has a deadlink to the macromodel for the TSV71x. Why that document was removed?

Normally i use MMBT222A. So even if use that part does it make any changes?.Note that am using a single tda ic for driving two speakers with 24-27v toroidal transformer. Am building a active powered speaker for studio monitor with full wood and 2xcomponent speaker(both speaker with 4 component , namely low,mid,vocal,high). Am using single ended input, L+R+GND. Also am using a dspic microcontroller for display and control.So all controls are made digitalaly. Am using micochip 100K digital pot(mcp series). The gain setting pins need to be controlled by microcontroller. So what i can do for that. If i just pull it up or down using a mmbt222a transistor, will there be any problems?. Same as for gain pins. Not that my digital level of microcontroller is 5v. So please guide me as per my requirements. 

..

..

I try to generate pulses using STEVAL-IME013V1.The power supply information is as follows:VDD=+5V DCVSS=-5V DCHVM=+10V DCHVM_CW=-10V DCActually the board can generate the preset waveforms by the power supply.Show as the figure.But I am curious at the sharp burrs phonemon.The sharp burrs amplitude even exceeds HVP(10V).How can I solve the problem?

Hi,According to the AcousticSL Software Library documentation, the AcousticSL_Handler_t Struct contains initialization parameters called ptr_M1_channels described as "Number of channels in the stream of Microphone 1. Deafult value is 1."I don't understand what this means.The example project initializes each of them to 4 for a 4 microphone array but isn't there only one stream per microphone? I would appreciate clarification,Thanks,Michael

..

..

Hi,I'm trying to attach a RaspberryPi-like SBC to the STA350BW dev board. I have I2C working (bit-banged), and have also connected the I2S LRCLK, BCLK and DATA signals.However, looking at the documentation, it appears that a master clock (MCLK) is also required. Is this the case? I'm not sure I can generate a suitably fast MCLK from the SBC. If I'm able to obtain a suitable crystal oscillator, is this likely to work even if the BCLK and MCLK may not be *exactly* accurate?Reading the status register returns 0xFF, which suggests that the PLL is not locked. I'm guessing this is why I'm not getting any audio at all (despite disabling the IDE bit).Grateful for any advice.Edit:I've hooked up a clock source, and the PLL is now showing as locked. But I'm still not getting a peep out of the amp. The status register shows 0x7f (127) - suggesting that there are no other faults.I've played with master volume, channel volume, as well as disabling IDE, BCLE, and bypassing most of the DSP functionali

Hi all, Can somebody support in getting datasheet of "324 ST MZCS903" which is a 14 - pin chip.Am unable to get from internet .Thanks in advance.Rohitrohit.kadecha@battrixx.com