Códecs de audio
Productos (12)
AD73322ARZ - Dual 16-Bit AFE, 73mW, 28-SOIC | Analog Devices
The Analog Devices AD73322ARZ is a dual, 16-bit analog front-end (AFE) processor for general-purpose speech and telephony applications, packaged in a 28-lead SOIC (300 mil). It integrates two 16-bit A/D conversion channels and two 16-bit D/A conversion channels, each delivering 77 dB SNR over a voiceband signal bandwidth, with typical total power consumption of 73 mW. An analog front-end (AFE) is a mixed-signal subsystem that bridges analog transducers (microphones, speakers, line interfaces) and a digital signal processor (DSP). In the signal chain, an AFE sits between the sensor and the host processor, performing sampling, reconstruction, anti-alias filtering, and gain conditioning. The AD73322 specifically belongs to the broader class of programmable voice-band codecs (analog-to-digital converter plus digital-to-analog converter plus filtering in one chip), positioned hierarchically as AFE -> mixed-signal IC -> data converter -> signal conditioning IC -> semiconductor. Common synonyms include "voice codec", "audio codec", and "DSP front-end". Key features of the AD73322ARZ include a programmable sampling rate (8 kSPS, 16 kSPS, 32 kSPS, 64 kSPS) derived from an internal master clock, an 8 kHz to 64 kHz sample-rate range, a serial port (SPORT) with rates programmable to match slow or fast DSP engines, and an on-chip voltage reference (REFCAP, 2.4 V when 5VEN=1). The 5VEN pin allows operation from either 3 V or 5 V supplies. The device supports daisy-chained (cascaded) operation of multiple AD73322 units to increase channel count without consuming additional DSP serial ports. ADI quotes 77 dB SNR for both ADC and DAC channels across voiceband bandwidth, with low total harmonic distortion suitable for full-duplex speech paths. Typical applications include voice-over-IP (VoIP) handsets, speech recognition front-ends, modem analog front-ends, audio recording/playback subsystems, and industrial two-way intercoms. The wide DSP compatibility and cascadable architecture also make it useful in conference phones and call-center headset bases. When designing with this part, note that the part dissipates approximately 73 mW at 3 V operation; provide a small copper pour on the SOIC exposed area and keep digital switching noise away from the AGND plane to preserve the 77 dB SNR figure.
AK4528VF - 24-Bit 96kHz Audio CODEC | AKM | LSSOP-28
The Asahi Kasei Microdevices (AKM) AK4528VF is a high-performance 24-bit stereo audio CODEC designed for 96 kHz recording systems, integrating a single ADC and a single DAC on a single die in a 28-pin LSSOP package. It supports sampling rates up to 96 kHz, delivers 24-bit resolution on a serial audio interface, and features AKM's Enhanced Dual Bit ADC architecture together with the Advanced Multi Bit DAC architecture with switched-capacitor filter (SCF) output stage. The combination provides wide dynamic range on the record path and low out-of-band noise with high jitter tolerance on the playback path, making the AK4528VF a well-known building block for professional audio interfaces, AV receivers, and broadcast equipment. A CODEC (coder-decoder) is an audio mixed-signal IC that converts analog microphone or line-level signals into digital data on the record side and converts stored or streaming digital audio back into analog signals on the playback side. Within the broader signal-chain taxonomy, an audio CODEC sits between the analog front end (microphone pre-amplifier, line driver) and the digital host (DSP, microcontroller, SoC) and is itself a member of the audio converter category alongside standalone ADCs and DACs. Key features of the AK4528VF include the Enhanced Dual Bit ADC for wide dynamic range, the Advanced Multi Bit DAC with SCF output stage, support for 96 kHz operation, a single stereo ADC plus single stereo DAC on one chip, and a serial digital audio interface for connection to standard DSPs, SoCs, and microcontrollers. The integrated SCF output stage removes the need for an external high-frequency post-filter and contributes to the part's high jitter tolerance, a useful trait when the CODEC is fed from a recovered clock in a digital-audio transport. Internally the AK4528VF combines a switched-capacitor ADC front end, a multi-bit delta-sigma modulator, decimation/interpolation filters, and a multi-bit DAC with switched-capacitor low-pass filtering. This architecture trades some analog simplicity for very high in-band linearity and low out-of-band noise, allowing the chip to meet the dynamic-range requirements of professional and prosumer recording at sample rates up to 96 kHz without an external reconstruction filter. Typical applications for the AK4528VF include professional and prosumer recording interfaces, digital mixing desks, broadcast and studio equipment, AV receivers and effects processors, and high-end consumer audio playback. Its combination of 24-bit resolution, 96 kHz support, integrated ADC and DAC, and LSSOP-28 footprint makes it a convenient single-chip front end for systems that previously required a separate ADC plus a separate DAC. When designing with the AK4528VF, pay attention to PCB layout: keep analog inputs and outputs physically separated from the digital interface traces, place the recommended reference-voltage capacitors near the supply pins, and observe the timing requirements in the AKM datasheet for the serial audio interface. Following the AKM reference schematic is the fastest path to achieving the datasheet's dynamic-range and THD+N numbers; a wiring gap between analog and digital grounds or a long trace in the MCLK path will cost measurable performance. This page synthesizes distributor pricing, drop-in and parametric alternatives, application context, and practical design notes for the AK4528VF in a single place, complementing the manufacturer datasheet with engineering guidance not always present on distributor listings.
AK4621EF-E2 - 24-bit 192kHz Stereo Audio CODEC | AKM
The Asahi Kasei Microdevices (AKM) AK4621EF-E2 is a high-performance 24-bit stereo audio CODEC supporting up to 192kHz recording and playback in a 30-pin VSOP surface-mount package. It integrates a 2-channel ADC and a 2-channel DAC, with the ADC delivering 115dB dynamic range via AKM's Enhanced Dual-Bit architecture and the DAC delivering 115dB via the Advanced Multi-Bit architecture with Switched Capacitor Filter (SCF). An audio CODEC (Coder-Decoder) is a mixed-signal integrated circuit that converts analog audio signals into digital data for processing, recording, or transmission (ADC) and converts digital audio back into analog signals for playback (DAC). CODECs sit within the broader hierarchy of audio IC -> mixed-signal IC -> data converter IC -> semiconductor, and serve as the central analog I/O block of virtually every professional and consumer audio product with digital connectivity. Key features of the AK4621EF-E2 include 24-bit resolution, 192kHz maximum sample rate, 115dB ADC/DAC dynamic range, low-latency digital filter, hardware controllable via a serial audio interface, and on-chip microphone/line input mixer. The -E2 suffix indicates the tape-and-reel packaging option of the AK4621EF die. The 30-VSOP package provides compact board footprint while exposing all serial audio, control, and supply pins for flexible PCB routing. The Enhanced Dual-Bit ADC and Advanced Multi-Bit DAC architectures deliver wide dynamic range and excellent jitter tolerance, suitable for pro-audio signal chains. Internal digital filtering offers low-latency operation suited to live monitoring applications. The serial audio interface accepts common I2S/left-justified formats, simplifying connection to hardware codecs, DSP engines, and USB/PCIe audio interfaces. Typical applications include Pro Audio sound cards, Digital Audio Workstations (DAW), DVD-R, hard disk, and CD-R recording/playback systems, and musical instrument recording equipment. Designers value this part for its high dynamic range, low latency, and AKM's proven audio architecture. When designing with this device, decouple AVDD and DVDD supplies separately near the IC pins and follow the datasheet's recommended analog-input RC network for optimal THD+N. The -E2 ordering code denotes tape-and-reel shipping, suitable for high-volume assembly. This page synthesizes distributor pricing, drop-in alternatives, and practical design notes not found in the manufacturer datasheet alone.
ALC5660 - Realtek Low-Power Audio Codec | QFN
The Realtek ALC5660 is a low-power, high-fidelity audio codec IC housed in a compact QFN package, designed for portable and embedded audio applications. Key parameters include integrated stereo DAC/ADC paths, low standby power consumption, and I2C/I2S digital interfaces for direct connection to application processors and SoCs. An audio codec (coder/decoder) is a mixed-signal IC that converts digital audio streams (typically I2S) into analog signals for headphones or line outputs (DAC path) and converts analog microphone or line-in signals into digital audio streams (ADC path). The codec sits in the audio signal chain between the host processor and the analog transducers. Realtek specializes in PC and portable audio codecs, where the ALC5660 family targets power-constrained designs requiring a small PCB footprint. Key features of the ALC5660 include programmable gain amplifiers on the analog inputs, integrated headphone drivers capable of driving 32-ohm loads, support for multiple sampling rates from 8 kHz to 192 kHz, and flexible digital mixer routing. The device typically supports both hardware pin-strap and software register configuration via I2C, enabling rapid platform bring-up without extensive driver development. The architecture combines a sigma-delta DAC and ADC with on-chip digital interpolation/decimation filters and an analog output stage optimized for low THD+N. A built-in DC-DC converter interface and low quiescent current make the part suitable for battery-powered systems where standby current directly affects runtime. According to Realtek product documentation, the ALC5660 family is targeted at tablets, portable media players, and docking-station accessories. Typical applications include Android tablet audio subsystems, portable Bluetooth speaker docks, USB-C headphone adapters, and embedded infotainment front-ends. The wide sampling-rate range also supports VoIP endpoints and dictation devices that require both playback and capture paths on a single chip. When designing with this device, note that the QFN package requires a thermal pad that must be soldered to the PCB ground plane for thermal and electrical performance. Decoupling capacitors must be placed within 2 mm of the AVDD and DVDD pins to meet the datasheet noise-floor target. This page synthesizes distributor pricing, parametric data, drop-in compatible Realtek and cross-brand alternatives, and practical design notes not consolidated on any single manufacturer or distributor page.
CS42L52-CNZ - 24-bit Low-Power Stereo CODEC | Cirrus Logic
The Cirrus Logic CS42L52-CNZ is a 24-bit, low-power stereo audio CODEC with integrated headphone amplifier and Class D speaker driver, housed in a 40-pin QFN-EP (6x6 mm) package. Key headline parameters include 98 dB dynamic range (A-weighted), -88 dB THD+N, and 1 W per channel Class D output capability into external speakers, all from a single supply rail down to 1.8 V. What is an audio CODEC? An audio CODEC (coder-decoder) is a mixed-signal IC that converts analog audio signals to digital (ADC path) and digital audio streams back to analog (DAC path), providing the bridge between microphones/speakers and a digital signal processor or application processor. Low-power CODECs sit in the broader power-management IC family and target portable, battery-powered audio applications where quiescent current, dynamic range, and integration directly impact playback time and listening quality. The CS42L52 integrates a flexible stereo analog input architecture with a 4:1 input mux, two headphone amplifiers capable of 44 mW per channel into 32 ohm loads, and a high-efficiency Class D speaker amplifier. Power consumption is 13 mW at 1.8 V in playback mode. The serial audio interface accepts standard I2S, left-justified, and right-justified formats. An on-chip PLL derives the internal audio clocks from a range of reference frequencies, simplifying system clocking. Typical applications include portable media players, smartphones, tablets, USB audio headsets, portable gaming devices, voice recorders, and Bluetooth/wireless audio docking stations. The CS42L52 suits any battery-powered stereo audio product where board area and power consumption are tightly constrained but high dynamic range and headphone or speaker drive capability cannot be compromised. When designing with this part, route analog inputs and outputs symmetrically, place decoupling capacitors close to each supply pin, and connect the exposed pad to a low-impedance ground plane for thermal dissipation and noise rejection. Per the CS42L52 datasheet, careful ground partitioning between analog and digital domains is required to achieve the rated 98 dB dynamic range. This page synthesizes Cirrus Logic datasheet specifications, real distributor pricing, verified drop-in alternatives, and practical application guidance not present on standalone component distributor pages.
ISD17120SY - 120sec Voice Record/Playback IC | Nuvoton | SOIC-28
The Nuvoton ISD17120SY is a single-chip multi-message voice record and playback IC from the ChipCorder ISD1700 family, providing 80 to 240 seconds of audio storage (120 seconds typical) in a 28-pin SOIC package. It operates from a single 2.4 V to 5.5 V supply and supports both pushbutton and SPI control interfaces, enabling flexible user interaction in standalone or microcontroller-driven systems. A voice record/playback IC is a fully integrated solid-state audio recorder that combines non-volatile memory, an analog microphone preamplifier, anti-aliasing filter, automatic gain control (AGC), speaker driver, and a PWM output stage on a single die. It belongs to the ChipCorder family of multi-message ICs within the broader audio IC hierarchy: voice record/playback IC -> audio IC -> interface IC -> mixed-signal semiconductor. The on-chip EEPROM-style storage eliminates external memory components, while the integrated class-D-compatible PWM output can directly drive a piezo speaker or feed an external amplifier. Key features include user-friendly pushbutton mode with seven pre-defined triggers (REC, PLAY, FT, ERASE, VOL, RDY, RESET), an SPI-compatible serial interface for microcontroller control, variable sample rate selection (4, 5.3, 6.4, 8 kHz) that adjusts duration and bandwidth, an onboard microphone preamplifier with AGC, and a 2.4 V to 5.5 V single supply that suits both 3.3 V and 5 V rails. The device supports up to 240 seconds of recording at 4 kHz sample rate, with messages stored as solid-state samples requiring no compression algorithm. The ISD17120SY uses Nuvoton's patented multilevel storage technology on a single die, providing non-volatile storage that retains recordings without power. The architecture eliminates the need for digital compression ICs, ADCs, DACs, or external memory, dramatically simplifying BOM for voice prompts in cost-sensitive embedded designs. Voice playback quality is selectable through the sample-rate pin, trading bandwidth (3.4 kHz max) for duration (up to 240 s). Typical applications include voice prompts in industrial equipment, security alarm panels, medical alert devices, toys and consumer appliances, elevator floor announcements, and automotive aftermarket accessories. Designers select this part when they need 60 to 240 seconds of robust, non-volatile audio in a single 28-pin IC. When designing with this device, place the bypass capacitor (typically 0.1 uF) within 5 mm of the VCC pin and route the SP+/SP- differential outputs as short as possible to minimize EMI from the PWM carrier. Verify speaker impedance matches the output stage's drive capability, typically 8 ohm or higher for piezo transducers. This page synthesizes distributor pricing, same-brand duration variants, drop-in alternatives, and practical design notes not found in the manufacturer datasheet.
M82820G-15 - 32-Channel VoIP Voice Processor | MACOM
The MACOM M82820G-15 is a 32-channel VoIP voice processor IC designed for high-density voice-over-IP media gateway and enterprise-class packet telephony platforms. It integrates DSP resources for voice compression, echo cancellation, tone detection, and packet processing in a single BGA package, replacing multiple discrete voice-processing components with one programmable device. According to distributor classification data, the M82820G-15 is grouped under Audio Special Purpose / Voice Processing Unit (VPU) ICs and is manufactured in a 484-pin FPBGA package (19x19 mm). A Voice Processor Unit (VPU) is a specialized digital signal processor IC optimized for real-time voice codec, packetization, and telephony signaling functions. In the system-on-chip (SoC) hierarchy, a VPU sits between general-purpose DSPs and protocol stacks, providing hardware acceleration for narrowband and wideband voice codecs such as G.711, G.723.1, G.726, G.729, and G.729A, as well as echo cancellers following ITU-T G.168. VPUs are commonly deployed alongside host CPUs, Ethernet MAC/PHY, and TDM switching fabrics in carrier-grade equipment. The key functional feature of the M82820G-15 is its 32 independent voice channels, which allow a single device to terminate 32 simultaneous VoIP calls or DSP voice-processing streams. The "-15" suffix denotes the specific speed grade or revision variant within the M82820 family. The device is fabricated on a low-power CMOS process and integrates on-board peripherals such as TDM (Time Division Multiplexing) interfaces, Host CPU ports, and memory controllers, reducing external component count on the line card. Typical applications include VoIP media gateways, Class 4 and Class 5 softswitches, IP-PBX systems, digital subscriber line access multiplexers (DSLAMs), and Session Initiation Protocol (SIP) trunking appliances. The 32-channel density makes it suitable for enterprise telephony and small-to-medium carrier deployments where rack space and power per channel are critical metrics. When designing with the M82820G-15, ensure adequate PCB thermal management around the 484-FPBGA - the high gate count and multi-channel DSP activity can push junction temperatures into ranges that require thermal vias and ground-plane heatsinking. Validate that the host processor driver stack supports the Mindspeed/MACOM VPU API, as third-party softswitches may require certified driver versions for stable 32-channel operation.
MOS6581 - 3-Voice FM Synthesizer Sound Chip | Commodore MOS | C64 Audio
The Commodore MOS 6581 (SID - Sound Interface Device) is a single-chip, 3-voice electronic music synthesizer and sound-effects generator originally designed for the Commodore 64 home computer. It combines three independently programmable analog oscillators, an analog multimode resonant filter, and per-voice ADSR envelope generators on a single NMOS die, all addressed through a memory-mapped register file compatible with the 65XX microprocessor family used in the C64, C128, and CBM-II series. A sound chip (or programmable sound generator, PSG) is a dedicated semiconductor that synthesizes audio under CPU control, sitting between the system bus and the analog output stage. SID belongs to the family of analog synthesis ICs - alongside later digital successors such as the Yamaha YM2149 and General Instrument AY-3-8910 - and is notable for being one of the first single-chip devices to combine subtractive synthesis (oscillators + filter) with programmable waveforms in a consumer computer. Key features of the MOS6581 include three independent voices each offering four selectable waveforms (triangle, sawtooth, variable-pulse, and noise), a 12-bit frequency resolution per voice, a global 4-bit volume control with four volume modes, and three analog filter modes (low-pass, high-pass, band-pass, plus notch combinations). The chip's signature warm, slightly distorted sound comes from its analog filter, which has become a defining audio characteristic of the Commodore 64 and a sought-after timbre in chiptune and electronic music production. Typical applications include the original Commodore 64/128 sound subsystem, chiptune music hardware and software synthesizers, retro-computing restoration boards, MIDI-to-SID adapter projects, and educational tools for teaching subtractive synthesis. Modern FPGA implementations such as those in the MiSTer platform reproduce the analog behavior bit-true, while the ARMSID and SIDKick Pico devices provide plug-compatible physical replacements. When designing with the MOS6581, engineers must consider that it draws power directly from the host computer's 5V and 12V rails and produces significant analog heat during operation. Replacement or refurbishment boards such as ARMSID and SIDKick Pico offer lower noise and modern reliability while retaining register-level compatibility with the original chip.
SGTL5000XNAA3R2 - Stereo Audio Codec, 32-QFN-EP | NXP
The NXP SGTL5000XNAA3R2 is an ultra-low-power stereo audio codec with integrated headphone amplifier, delivering 98 dB SNR DAC performance at only 4 mW for the DAC-to-headphone path, housed in a 32-pin QFN with exposed pad (5x5 mm) package. The IC combines line-in, microphone-in, line-out, headphone-out, and digital audio I/O on a single die, with 24-bit ADC/DAC resolution and 100 dB SNR on the ADC path. Programmable sample rates from 8 kHz to 96 kHz are supported, and an integrated PLL allows clock reuse from an external system reference, reducing overall BOM cost. An audio codec (coder-decoder) is a mixed-signal IC that converts digital audio streams (typically I2S, left-justified, or right-justified) into analog signals for headphones or line outputs while simultaneously digitizing analog microphone or line inputs. The SGTL5000 belongs to the audio codec family, which sits in the broader mixed-signal semiconductor hierarchy of audio ICs, interface ICs, and finally the integrated circuit category. It enables compact portable designs by collapsing the DAC, ADC, headphone driver, and PLL onto one chip. Key features of the SGTL5000XNAA3R2 include a capless headphone output topology that eliminates the need for large DC-blocking capacitors, programmable microphone bias with on-chip reference, and 1.62 V to 3.6 V digital supply operation with 0.9 V to 1.2 V analog core voltage (generated by an internal LDO). Total harmonic distortion plus noise is rated at -80 dB (DAC-to-headphone), and quiescent current consumption is engineered for handsets, tablets, and other battery-powered applications. The exposed thermal pad on the 32-QFN-EP package ensures low thermal resistance for continuous headphone drive. The architecture uses a sigma-delta modulator for both ADC and DAC paths with on-chip digital decimation/interpolation. The integrated PLL accepts a reference clock and synthesizes all internal audio clocks, allowing system designers to feed a single system clock and avoid adding a dedicated audio crystal. Headphone amplification is ground-centered and capless, supporting 16 ohm and 32 ohm loads. Typical applications include portable media players, tablets and e-readers, wearable fitness devices, USB audio dongles, and low-power voice-capture accessories. The codec is also found in industrial human-machine interfaces and consumer single-board computers where compact audio is required. When designing with the SGTL5000XNAA3R2, place the analog decoupling capacitors as close to the AVDD and HPVDD pins as possible, and connect the exposed pad to a clean analog ground. Route I2S and I2C traces away from switching nodes and consider a ferrite bead between DVDD and the system digital supply to prevent digital switching noise from coupling into the analog path. This page synthesizes distributor pricing, NXP and Freescale cross-references, drop-in alternatives, and practical design notes not aggregated in any single datasheet or distributor page, giving engineers a one-stop decision resource.
SGTL5000XNLA3 - Ultra-Low-Power Stereo Audio Codec | NXP
The NXP SGTL5000XNLA3 is an ultra-low-power stereo audio codec with an integrated headphone amplifier, housed in a 20-pin QFN-EP (3x3 mm) package with exposed thermal pad. It delivers 98 dB SNR DAC performance, -80 dB THD+N at headphone output, and consumes only 4 mW for the DAC-to-headphone playback path, forming a complete mixed-signal audio solution in a single die. The device integrates line-level input, microphone input, line output, headphone output, and a digital I/O port supporting I2C and SPI control in a 3 mm x 3 mm footprint suitable for hand-held and wearable products. An audio codec (coder-decoder) is a mixed-signal integrated circuit that converts digital audio streams to analog outputs through a DAC path and converts analog inputs to digital streams through an ADC path. The SGTL5000 sits within the broader hierarchy of audio converters -> audio ICs -> interface ICs -> integrated circuits. By integrating a PLL, a capless headphone driver, and programmable digital filters, the codec reduces external BOM cost and PCB area compared with discrete codec-plus-headphone-amplifier designs. Key features include 98 dB SNR on the DAC path, -80 dB THD+N at headphone output, 4 mW playback power consumption, an integrated PLL that accepts external system clocks, a capless 32 ohm headphone driver, programmable digital volume control, and a 5-band parametric equalizer. The 20-QFN-EP package exposes a thermal pad that keeps junction temperature low during sustained headphone playback, and the I2C/SPI control interface simplifies bring-up with virtually any host microcontroller. Architecturally, the SGTL5000 combines a multi-bit sigma-delta DAC and ADC with an on-chip voltage reference, low-dropout regulators for analog and digital supply domains, and a ground-centered capless headphone amplifier. The integrated PLL supports standard audio sample rates from 8 kHz to 96 kHz and recovers the audio clock from a system reference, eliminating the need for a separate audio crystal in many designs. Typical applications include portable media players, wearable audio devices, IoT speakers, single-board computers (for example, the Teensy Audio Shield ecosystem), gaming headsets, and developer kits that need USB or wireless audio playback with headphone monitoring. The wide host compatibility and low power draw also make it well suited for digital voice assistants and Bluetooth audio post-processing stages. Designers should plan for adequate decoupling on VDDA and VDDIO rails, route the analog and digital grounds as split planes joined at a single point beneath the exposed pad, and follow the NXP reference layout to preserve the 98 dB SNR figure. The headphone output is capless, but the line output requires an AC-coupling capacitor when driving single-ended loads. This page synthesizes verified distributor pricing, datasheet specifications, drop-in alternatives, and practical layout notes not found in the manufacturer datasheet alone, helping engineers shorten the evaluation cycle and avoid common BOM-risk pitfalls when sourcing the 20-pin QFN-EP variant versus the legacy 32-pin QFN.
SGTL5000XNLA3R2 - Ultra-Low-Power Stereo Audio Codec | NXP
The NXP SGTL5000XNLA3R2 is an ultra-low-power stereo audio codec with integrated headphone amplifier, delivering 98 dB SNR DAC-to-headphone performance and -80 dB THD+N from a 3x3 mm 20-pin QFN-EP package. It supports stereo line-in, microphone-in, line-out, headphone-out and digital I/O through I2C, SPI or I2S serial interfaces, with a power budget of just 4 mW from DAC to headphone output. An audio codec (coder-decoder) is a mixed-signal IC that converts digital audio streams into analog signals for headphones or line outputs and converts analog microphone or line inputs back into digital audio. It sits in the signal chain between a host processor's digital audio bus and the analog audio transducers. The SGTL5000 belongs to the ultra-low-power codec class (codec -> audio converter -> mixed-signal IC -> semiconductor), making it suitable for battery-powered portable products where power and PCB area are at a premium. Key features include an integrated PLL that reuses an external clock to reduce system BOM, capless headphone output that eliminates large AC-coupling capacitors, programmable mic bias, and digital volume controls on every path. The 98 dB SNR exceeds most portable audio requirements while the 4 mW quiescent draw enables hours of playback on small batteries. Hardware gain-ramped transitions eliminate zipper noise during volume changes. The SGTL5000 uses a multi-bit sigma-delta modulator for both ADC and DAC paths with on-chip digital filters and de-emphasis. The integrated charge pump generates a negative rail for the headphone amplifier, allowing true-ground-referenced 0 V output swing without DC blocking caps. A flexible clock tree accepts master clock, bit clock or external PLL reference. Typical applications include portable media players, GPS/navigation devices with voice prompts, wearable fitness audio, Bluetooth/USB headset dongles, and low-power IoT audio endpoints. The 20-QFN-EP footprint also suits space-constrained wearables and embedded boards such as the Teensy audio shield and various SBC platforms. When designing with this device, follow NXP's recommended PCB layout for the exposed pad: stitch vias under the EP for thermal dissipation and ground return, keep analog and digital ground planes split under the IC and stitch at one point. Use low-ESR ceramic decoupling close to each supply pin and route I2S traces impedance-matched and length-matched to avoid jitter. This page synthesizes NXP distributor pricing, pin-compatible SGTL5000 family alternatives, and practical PCB layout notes not bundled in the manufacturer datasheet.
UDA1341TS/N1 - Stereo Audio CODEC, SSOP-28 | NXP Semiconductors
The NXP Semiconductors UDA1341TS/N1 is a single-chip stereo audio CODEC combining a 4-channel Analog-to-Digital Converter and a 2-channel Digital-to-Analog Converter using bitstream conversion techniques, housed in a 28-pin SSOP (SOT341-1) package. It features 20-bit ADC/DAC resolution, integrated Programmable Gain Amplifiers (PGA), digital Automatic Gain Control (AGC), and an L3-interface for microcontroller control of audio processing features including de-emphasis, volume, bass boost, treble, and soft mute. An audio CODEC (Coder-Decoder) is a mixed-signal integrated circuit that converts analog audio signals into digital form for processing or storage, and converts digital audio back into analog for playback. CODECs sit at the boundary between the analog and digital audio domains, and are foundational components in the audio signal chain: microphone input -> analog PGA -> ADC -> digital audio bus -> DAC -> analog output -> speakers/headphones. The UDA1341TS belongs to NXP's economy audio CODEC family, designed for cost-sensitive MiniDisc (MD), home stereo, and portable applications. Key features include 4 analog input channels and 2 analog output channels, support for multiple digital audio data formats (I2S-bus, LSB-justified, MSB-justified), a fully integrated analog front end, on-chip DSP features in playback mode, and partial pin compatibility with the UDA1340M and UDA1344TS for easy PCB upgrades. The device operates over -20C to +85C ambient temperature range typical of consumer audio equipment. The UDA1341TS architecture uses oversampling bitstream (sigma-delta) conversion to achieve 20-bit resolution with high noise immunity and reduced analog filtering requirements. The integrated PGA and AGC simplify external analog circuitry, while the L3-interface allows software control of digital audio features without external DSP hardware. This makes the UDA1341TS suitable for compact, low-cost audio designs. Typical applications include MiniDisc (MD) recorders and players, home stereo receivers and CD recorders, portable digital audio devices, set-top boxes, and automotive head units. The wide input channel count supports multi-source mixing for recording applications. When designing with the UDA1341TS, pay careful attention to power supply decoupling and grounding to avoid digital switching noise coupling into the analog sections, since the CODEC integrates both sensitive analog and high-speed digital functions on the same die. Use separate analog and digital supply pins as recommended in the datasheet, with low-impedance ground returns. This page synthesizes current NXP datasheet specifications, real distributor pricing, drop-in alternative CODECs, and practical analog/digital design guidance not found in the manufacturer datasheet alone.