Analog and audio

Hi,I am using FP-AUD-BVLINK2 pack on Bluecoin. I want integrate fp-aud-smartmic1 Beamforming Acoustic Library with BVLink2 Example. Problem occurs when I compile (After including all necessary header and library files).It seems that BVLink Pack uses Floating Point Hardware Implemented. But smartmic1 BF Acoustic Library uses MIX HW/SW FPU option.I am unable to complete this project. Is there any solution to make AcousticBF library to use Hardware FPU?

part # 09395699 delco ic. can you give me your part # It is a st part.

I'm trying to identify this IC, marked N00X828 Googling doesn't get me anything. Part of an audio circuit.

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We want to utilize the diode STTH3003. It would be great to have some information about the dielectric strength of the plastic material.Is this information available?Thanks and best regards!Markus

One of the sample devices required is the STA540 amplifier. It seems to be a good device which will allow students to work with quad, stereo, and bridged outputs.

Hello!I want to ask you some advice in building a class D amplifier using the integrated circuit STA516BEFrom the information provided by the integrated datasheet, I am misled by some features.I want to use figure no. 8 (dual BTL) for speakers with an 8 ohm impedanceI don't understand the configuration of pins 25 and 28 (PWRDN / TH_WAR) - do I have to make other changes to the diagram to make the amplifier work?How many amps does the amplifier consume? (how many amps should the source / transformer be)What is the recommended voltage for the 8 ohm load? (52V or 65V)

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Dear ST support teamAs shown in the diagram, I use a current transformer (PT1) and an amplifier to measure AC voltage:When U1A is LM258DR of TI, the system works perfectly well. After replacing U1A with LM258DT of ST, the waveform measured by the oscilloscope is distorted. The following picture is a screenshot of the oscilloscope, the yellow is the waveform of the circuit using LM258DR, and the blue is the ST chip:The model of T1 is ZMPT107-1, which is a transformer of 2mA/2mA. When I tested the 3-foot waveform of T1, I found abnormal condition: Because of the feature of OP AMP, the current output from T1 goes through R3, and there is no current coming in from 2-foot of OP AMP, so the voltage of T1 3-foot should be equal to 0. In circuit using TI chip, this is true, but in circuit using ST chip,There is a strange waveform. The yellow one in the TI chip circuit and the blue one in the ST chip circuit are shown below:According to the datasheet, the parameters of LM258DR and LM258DT are v

The M41T62 driver provided in STSW-M41T62 appears to have an error related to calibration of positive adjustments. Refer to the code below from line 750 of M41T62.cThe sign parameter is 0 or 1. To program the M41T62 for positive compensation requires programming the control register bit 5 with a logical 1. The code does not do this as it effects only bit 0 of the control register.I suggest adding the following code at line 758:sign = sign <<5;M41T62_Error_et M41T62_Set_SignForCalibration(M41T62_SignCalibrationStatus_et sign){ uint8_t tmp; M41T62_assert_param(IS_M41T62_CALIBRATION_SIGN(sign)); if(M41T62_ReadRegs(M41T62_CALIBRATIONREG, 1, &tmp))  return M41T62_ERROR; tmp &= ~M41T62_SIGN_CALIBRATION_MASK; tmp |= ((uint8_t)sign); if(M41T62_WriteRegs(M41T62_CALIBRATIONREG, 1, &tmp))  return M41T62_ERROR; return M41T62_OK;}

I have just bought the STEVAL-IME009V1. It comes with a factory settings DFU file and DfuSe evaluation tool (STSW-STM32080).It generates and extracts hex files, but how can I edit those files to create impluses and burst pulses with specific frequencies?I had the STEVAL-IME013V1 evaluation board before and it had the ultrasound pulser waveform demo, which has a very simple to interface to create pulses and upload them to board. Is contacting ST the only solution to edit these files in STEVAL-IME009V? This won't be practical at all.

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Hello Support,I have application in wjhich i have to sense current for negative supply. specifications are as below.Negative supply voltage range( at RS- pin) : -5 V to -2 VCurrent range that needs to be sensed: 0.1 uA to 20uAThe output of OPAMP needs to be read by 10 bit ADC of controller. we have one application note AN4451 in which it shows nA range. but in our application the voltage is negative so please guide regarding that or suggest some part number.Thanks

Hello ST experts, I am working on TDA7801 chip and I didn't find an explained way for programming it or even a simple code. To be specific, I can't read or write data to chip registers through I2C.!! Can you help?.best regards Hossam

Dear ST support team,Good morning,I want to find alternative part for TSZ122IST . we check the data sheet and find out the TSZ122IYDT. But we didn't saw the difference between them.Please support to explain this case Thank and Best Regard,Danny,

HiThe Android All-in-One product I'm using has an amplifier chip called TDA7838 built-in. I would like to know the difference between the TDA7850 amplifier chip. I would also like to receive a datasheet for the TDA7838. I look forward to your answer.

Hello.I'm trying to import the SMR module of the X-CUBE-AUDIO package in one of the projects I am working on. I wish to know the sound pressure level in dB on my microphones.My application generates a PCM buffer filled this way: 32 samples in 16-bit format (16 samples for each channel). The sampling frequency is 16 kHz.It seems that everything is set rightly but, when i call the smr_process function, the application goes in Hard Fault.I am following the API call procedure shown in UM2031, pp. 16-17 (file attached).Anyone could please help me understand what I'm doing wrong?Below there's part of the code I am using.The project runs on STM32F767 MCU./* SMR Manager */ void *pSmrPersistentMem = NULL; void *pSmrScratchMem = NULL;   static smr_static_param_t smr_static_param; static smr_dynamic_param_t smr_dynamic_param;   static buffer_t BufferHandler; static buffer_t *pBufferHandler = &BufferHandler;   uint16_t PCM_Data_SMR[32]; /* PCM data buffer */     void

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Hello, I cannot get SoundSL working to detect the source of a sound direction correctly. To check how the library performs itself, I wrote a little test script.I assume a 1kHz sine wave to be sampled by a simulated microphone. At a sampling rate of 16kHz, the buffer for 1ms duration consists of 16 samples (per channel) that describe an entire period. At a sound speed of 350m/s and a distance of 0.1m between the microphones, the phase shift should not exceed approximately +/- 90 degrees (+/- 285us). According to my understanding, a phaseshift of [-90 degrees phase shift; 90 degrees phase shift] should result in about [-90 degree angle; 90 degrees angle] where in-phase would be 0 degrees (same distance to both microphones).In the test script, I compute the sine wave values for a particular offset and feed the values to the library until the result can be calculated. Once one angle is gathered, I increase the phase shift and redo the whole process. The angles are then stored in an array.

Hello, I'm trying to get Acoustic_SL running to detect the sound location with two INMP441 MEMS microphones. Unluckily, Acoustic_SL does only return two angles {-87, 88} independent of the processed PCM values. May anyone have an idea of what might go wrong? Note: I extracted the important code snipped from a larger C++ project to post it on the forum.Angle estimates independent of the actual sound source.Some defines and the compression function. The compression throws away some of the MSB bits to gain sensitivity.DMA callbacks.Main application. Example of compressed samples with alternating L/R channel.

Hello,I am using a TSZ124 chip to build a circuit for a breadboard experiment. I attached the schematic at the end of this message. I saw a very strange DC offset when I use an oscilloscope to look at its output pin (the input signal itself has no DC offset). Because the experiment is a breadboard experiment and the tsz124 chip that I use has a surface-mount package, I used an adapter (PA0033) for it. During the time when I finished soldering everything, the tsz124 chip works normally with no DC offset. However, after a few minutes or a few hours, the DC offset appears at the output pin. If I resolder everything, the DC offset disappears, but it comes back again after a certain time. I do not understand why this strange thing happen.In addition, if I replace the tsz124 with an LM324, there is no DC offset anymore.

Can someone please explain to me what the meaning of the threshold voltage on the VNPxxN07 transistors and if it is different from the threshold voltage mentioned, what input (gate or drain) voltage is necessary for the protection circuits to be active ?Thank you so much!