Digital, PDM Microphone MEMS (Silicon) Omnidirectional (-37dB ±1dB @ 94dB SPL) Solder Pads #CommonPartsLibrary #Audio #MEMS #Microphone

20 Hz ~ 20 kHz Analog Microphone MEMS (Silicon) 1.5 V ~ 3.6 V Omnidirectional (-38db ±1dB) Solder Pads #commonpartslibrary #audio #microphone

20 Hz ~ 19 kHz Digital, PDM Microphone MEMS (Silicon) 1.62 V ~ 3.6 V Omnidirectional (-36dB ±1dB @ 94dB SPL) Solder Pads #commonpartslibrary #microphone #audio

20 Hz ~ 20 kHz Digital, PDM Microphone MEMS (Silicon) 1.62 V ~ 3.6 V Omnidirectional (-36dB ±1dB @ 94dB SPL) Solder Pads The IM69D130V01XTSA1 is a high-performance digital MEMS microphone designed for applications requiring compact size, low power consumption, and reliable acoustic performance. The device integrates a MEMS sensing element with signal conditioning and digital output functionality within a surface-mount package suitable for automated assembly processes. The component is intended for use in voice-enabled systems, consumer electronics, industrial equipment, communication devices, and embedded audio applications where consistent sensitivity, low noise operation, and environmental robustness are required. The microphone supports digital audio interfacing and is optimized for far-field voice capture and speech recognition environments. The device construction is designed to provide stable operation under standard environmental and mechanical conditions. It is compatible with modern PCB manufacturing processes and supports integration into space-constrained electronic assemblies. Electrical and acoustic characteristics are engineered to maintain reliable performance across varying operating conditions. The component complies with standard electronic manufacturing and handling requirements for moisture sensitivity, electrostatic discharge protection, and reflow soldering compatibility. Proper PCB layout, grounding practices, and acoustic port alignment are recommended to achieve optimal system-level performance. #CommonPartsLibrary #Audio #MEMS #Microphone

20 Hz ~ 20 kHz Digital, PDM Microphone MEMS (Silicon) 1.6 V ~ 3.6 V Omnidirectional (-26dB ±1dB @ 94dB SPL) Solder Pads #CommonPartsLibrary #Audio #MEMS #Microphone

The IA611 SmartMic is an "always-on" Audio Processor featuring Voice Wake and Voice ID keyword detector, a two-second buffer', and Knowles' proven high performance acoustic SiSonic™ MEMS microphone technology in a single, miniature, top-port package. The IA611 offers flexibility by supporting the most relevant audio and data interfaces. Its integrated programmable DSP with 248 kBytes of RAM is available for customer and 3rd party algorithms, enabling unlimited creativity. The solution pushes the system performance to ultra-low power with its custom core design, and accelerates time to market with its highly integrated combination of hardware, software, and firmware.

MEMS audio sensor omnidirectional stereo digital microphone #CommonPartsLibrary #IntegratedCircuit #MEMS

ARM® Cortex®-M33 RT-600 Microcontroller IC 32-Bit Dual-Core 300MHz External Program Memory 176-VFBGA (9x9) The RT600 is a family of dual-core microcontrollers for embedded applications featuring an Arm Cortex-M33 CPU combined with a Cadence Xtensa HiFi4 advanced Audio Digital Signal Processor CPU. The Cortex-M33 includes two hardware coprocessors providing enhanced performance for an array of complex algorithms. The family offers a rich set of peripherals and very low power consumption. The Arm Cortex-M33 is a next generation core based on the ARMv8-M architecture that offers system enhancements, such as ARM TrustZone® security, single-cycle digital signal processing, and a tightly-coupled coprocessor interface, combined with low power consumption, enhanced debug features, and a high level of support block integration. The ARM Cortex-M33 CPU employs a 3-stage instruction pipe and includes an internal prefetch unit that supports speculative branching. A hardware floating-point processor is integrated into the core. On the RT600, the Cortex-M33 is augmented with two hardware coprocessors providing accelerated support for additional DSP algorithms and cryptography. The Cadence Xtensa HiFi 4 Audio DSP engine is a highly optimized audio processor designed especially for efficient execution of audio and voice codecs and pre- and post-processing modules. It supports four 32x32-bit MACs, some support for 72-bit accumulators, limited ability to support eight 32x16-bit MACs, and the ability to issue two 64-bit loads per cycle. There is a floating point unit providing up to four single-precision IEEE floating point MACs per cycle. The RT600 provides up to 4.5 MB of on-chip SRAM (plus an additional 128 KB of tightly-coupled HiFi4 ram) and several high-bandwidth interfaces to access off-chip flash. The FlexSPI flash interface supports two channels and includes an 32 KB cache and an on-the-fly decryption engine. The RT600 is designed to allow the CortexM33 to operate at frequencies of up to 300 MHz and the HiFi4 DSP to operate at frequencies of up to 600 MHz. Key Features Dual-core architecture with Arm Cortex-M33 CPU and Cadence Xtensa HiFi4 DSP Cortex-M33 operating at up to 300 MHz HiFi4 DSP operating at up to 600 MHz Up to 4.5 MB on-chip SRAM 128 KB DSP tightly coupled memory (TCM) FlexSPI interface supporting Quad/Octal SPI Flash with execute-in-place (XIP) High-speed USB 2.0 Host/Device with integrated PHY Digital microphone (DMIC) interface supporting up to 8 channels Multiple I2S, SPI, I²C, UART, SDIO, and I3C interfaces 12-bit ADC with up to 1 MSPS sampling rate Hardware cryptography acceleration (AES, SHA, RSA, ECC) Arm TrustZone® security technology Secure boot and physically unclonable function (PUF) key storage Multiple low-power operating modes for battery-powered applications Up to 147 GPIOs depending on package option 176-pin VFBGA package (9 mm × 9 mm) Typical Applications Voice-controlled smart devices Audio processing and DSP systems Smart speakers and soundbars Industrial HMI and control panels Edge AI and machine learning applications IoT gateways and connected devices Wearable and portable electronics #NXP #iMXRT685 #RT600 #CortexM33 #HiFi4DSP #Microcontroller #EmbeddedSystems #EdgeAI #VoiceRecognition #AudioProcessing #IoT #IndustrialAutomation #SecureMCU #MachineLearning #MCUDesign

Audio Amp Microphone 1-CH Mono Class-AB 14-TDFN EP T/R #commonpartslibrary #integratedcircuit #linearamplifier #mono

100 Hz ~ 10 kHz Analog Microphone MEMS (Silicon) 1.6 V ~ 3.6 V Omnidirectional (-43dB ±1dB @ 94dB SPL) Solder Pads #CommonPartsLibrary #Audio #Microphone #CMM-2718

ADC, Audio 32 b 8k ~ 192k I2S 20-WQFN (3x3) The PCM1822 is a high-performance, Burr-Brown™ audio analog-to-digital converter (ADC) that supports simultaneous sampling of up to two analog channels. The device supports single-ended and differential line and microphone inputs with a 1-VRMS full-scale signal. The device integrates a phase-locked loop (PLL), a DC removal high-pass filter (HPF), and supports sample rates up to 192 kHz. The device supports time-division multiplexing (TDM) or I2S audio formats, selectable with the hardware pin level. Additionally, the PCM1822 supports master and slave mode selection for the audio bus interface operation. These integrated high-performance features, along with the ability to be powered from a single supply of 3.3 V, make the device an excellent choice for costsensitive, space-constrained audio systems in far-field microphone recording applications. The PCM1822 is specified from –40°C to +125°C, and is offered in a 20-pin WQFN package. Features • Stereo high-performance ADC: – 2-channel analog microphones or line-in • ADC line and microphone differential and singleended input performance: – PCM1822 dynamic range: • 117-dB, dynamic range enhancer enabled • 111-dB, dynamic range enhancer disabled – THD+N: –95 dB • 1-VRMS full-scale input • ADC sample rate (fS) = 8 kHz to 192 kHz • Hardware pin control configurations • Linear-phase or low-latency filter selection • Flexible audio serial data interface: – Master or slave interface selection – 32-bits, 2-channel TDM – 32-bits, 2-channel I2S • Automatic power-down upon loss of audio clocks • Integrated high-performance audio PLL • Single-supply operation: 3.3 V • I/O-supply operation: 3.3 V or 1.8 V • Power consumption for 3.3-V AVDD supply: – 19.6 mW/channel at 16-kHz sample rate – 21.3 mW/channel at 48-kHz sample rate #CommonPartsLibrary #IntegratedCircuit #PCM1822IRTER #TexasInstruments #AudioADC #StereoADC #BurrBrown #I2S #TDM #AudioElectronics #EmbeddedAudio #SmartSpeaker #AVReceiver #HighResolutionAudio #PCBDesign #ElectronicsDesign #SignalProcessing

ADC, Audio 32 b 7.35k ~ 768k I2C, I2S, TDM 20-WQFN (3x3) The TLV320ADC3120IRTER is a low-power, high-performance stereo audio analog-to-digital converter designed for voice, audio capture, and digital signal processing applications. The device integrates programmable microphone bias generation, low-noise input amplifiers, digital filtering, automatic gain control, and high-resolution audio conversion into a compact package. It supports a wide range of analog microphone and line-level input configurations while delivering excellent audio fidelity and power efficiency. The converter is optimized for portable electronics, smart devices, industrial audio systems, communication equipment, and embedded voice-processing applications requiring high-quality audio digitization. Engineering Specifications Manufacturer: Texas Instruments Part Number: TLV320ADC3120IRTER Device Type: Stereo Audio Analog-to-Digital Converter Audio Channels: Stereo Input Architecture Converter Type: Sigma-Delta Analog-to-Digital Converter Resolution: High-Resolution Audio Conversion Input Configuration: Differential and Single-Ended Inputs Microphone Support: Integrated Microphone Bias Generation Programmable Gain: Integrated Low-Noise Programmable Gain Amplifiers Digital Processing: Integrated Digital Filters Automatic Gain Control: Supported Audio Interface: Digital Audio Serial Interface Power Consumption: Low Power Operation Clocking Support: Flexible Audio Clock Configuration Noise Performance: Low Noise Audio Acquisition Dynamic Range: High Dynamic Range Package Type: WQFN Package Mounting Style: Surface Mount Operating Temperature Range: Industrial Grade RoHS Compliant: Yes Applications: Voice Capture Systems, Smart Speakers, Audio Recording Equipment, Industrial Audio Devices, Communication Systems, Embedded Audio Platforms, Human Machine Interfaces, Portable Electronics Key Features High-performance stereo audio conversion Integrated microphone bias circuitry Low-noise programmable gain amplifiers Automatic gain control functionality Flexible digital audio interface support Optimized for voice and audio capture applications Low power consumption for portable devices Integrated digital signal processing features High audio fidelity and dynamic range Compact surface-mount package for space-constrained designs #TLV320ADC3120IRTER #TexasInstruments #AudioADC #AudioConverter #StereoADC #DigitalAudio #VoiceCapture #AudioProcessing #EmbeddedAudio #MicrophoneInterface #AudioElectronics #DSP #IndustrialElectronics #SmartDevices #PortableElectronics #SignalProcessing #SurfaceMount #ElectronicComponents #PCBDesign #AudioSystemDesign

Audio 16 b, 20 b, 24 b, 32 b 768k I2C, I2S, TDM 20-WQFN (3x3) The TAA3020IRTER is a high-performance 2-channel audio analog-to-digital converter (ADC) from Texas Instruments, designed for professional and embedded audio applications. It supports up to two analog microphone/line inputs or four digital PDM microphone channels with simultaneous sampling. The device delivers excellent audio quality with a 104 dB dynamic range, supports sample rates up to 768 kHz, and integrates advanced audio processing features such as programmable gain, digital filters, automatic gain control (AGC), voice activity detection (VAD), and an integrated PLL. It communicates through an I²C control interface and supports I²S, TDM, and Left-Justified (LJ) digital audio formats, making it ideal for smart speakers, conferencing systems, IP cameras, and other high-performance audio acquisition systems. Key Features High-performance 2-channel audio ADC Supports 2 analog microphone/line inputs or 4 digital PDM microphone inputs 104 dB dynamic range (DR) and –95 dB THD+N Sampling rates from 8 kHz to 768 kHz Programmable microphone gain from 0 dB to 42 dB Digital volume control from –100 dB to +27 dB Integrated programmable HPF, biquad filters, and low-latency filter modes Built-in Automatic Gain Control (AGC) and Voice Activity Detection (VAD) Integrated audio PLL and programmable microphone bias supply Supports I²S, TDM, and Left-Justified (LJ) audio interfaces Configurable for 1.8 V or 3.3 V supply operation Available in a compact 20-pin WQFN (RTE) package Operating temperature range: –40°C to +105°C #AudioADC #TexasInstruments #TAA3020 #HighPerformanceADC #I2S #TDM #PDMMicrophone #AudioProcessing #AGC #VAD #EmbeddedAudio #ProfessionalAudio

SSM2167-1RMZ-R7 Microphone Audio mikroBUS™ Click™ Platform Evaluation Expansion Board

100 Hz ~ 8 kHz Analog Microphone MEMS (Silicon) 1.5 V ~ 3.6 V Noise Cancelling (-31dB) Solder Pads #CommonPartsLibrary #Audio #Microphone

100 Hz ~ 10 kHz Digital, PDM Microphone MEMS (Silicon) Omnidirectional () Solder Pads #CommonPartsLibrary #Audio #MEMS #Microphone

Analog Microphone Electret Condenser 2V ~ 10V Omnidirectional (-35dB ±4dB) PC Pins #Microphone #Audio

20 Hz ~ 10 kHz Digital, PDM Microphone MEMS (Silicon) 1.65 V ~ 3.6 V Omnidirectional (-41dB ±1dB @ 94dB SPL) Solder Pads #CommonPartsLibrary #Audio #MEMS #Microphone

Microphone Preamplifiers High-Perform Digital Pre-Amp Controller #commonpartslibrary #integratedcircuit

100 Hz ~ 10 kHz Digital, PDM Microphone MEMS (Silicon) 1.6 V ~ 3.6 V Omnidirectional (-26dB ±1dB @ 94dB SPL) Solder Pads #commonpartslibrary #audioproduct #microphone

Microphone Amplifier with AGC and Low-Noise Microphone Bias, TDFN-14 audio microphone amplifier *DFN*EP*3x3mm*P0.4mm*

The INMP441 is a high-performance, low power, digital-output, omnidirectional MEMS microphone with a bottom port.