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AD1938YSTZ - 4ADC/8DAC 192kHz 24-Bit Audio CODEC LQFP-48 | ADI
The Analog Devices AD1938YSTZ is a high-performance single-chip audio codec that integrates 4 analog-to-digital converters (ADCs) with differential inputs and 8 digital-to-analog converters (DACs) with single-ended outputs, housed in a 48-lead LQFP (7x7 mm) package. The device supports sample rates from 8 kHz to 192 kHz at 24-bit resolution, operates from a single 3.3 V supply (with 5 V tolerant logic inputs), and is controlled via an SPI port. An audio CODEC (coder-decoder) is a mixed-signal IC that converts analog audio signals to digital and digital audio signals back to analog, sitting at the intersection of the signal chain between analog transducers (microphones, amplifiers, speakers) and the digital signal processor (DSP) or microcontroller. The AD1938 belongs to the broader category of audio interface ICs, specifically the multi-channel CODEC family, and sits within Analog Devices' precision signal-chain portfolio for professional and prosumer audio. Key features of the AD1938 include 107 dB DAC dynamic range and SNR, 107 dB ADC dynamic range, -94 dB THD+N, a low-EMI design with on-chip PLL for flexible clock generation, software-controllable clickless mute, a logarithmic volume control with autoramp function, and master/slave modes supporting up to 16 devices on a single TDM bus. The codec also integrates TDM, I2S, left-justified, and right-justified serial audio interface modes. The AD1938 employs Analog Devices' patented multibit sigma-delta (Σ-Δ) modulator architecture with a switched-capacitor DAC topology. The integrated PLL allows direct MCLK input up to 27 MHz or generation of internal clocks from a 256 x Fs reference, simplifying board-level clock trees in multi-codec systems. The differential ADC input paths maximize common-mode rejection for balanced microphone or line-level capture. Typical applications include automotive head units and infotainment systems, AV receivers, multichannel mixing consoles, musical instrument amplifiers, effects processors, and broadcast audio equipment. The single 3.3 V supply and 5 V tolerant digital inputs simplify integration into mixed-voltage systems alongside host DSPs or microcontrollers. When designing with the AD1938, ensure adequate decoupling on both AVDD and DVDD rails (typically 10 uF bulk plus 0.1 uF ceramic per pin) and place the master-clock reference within 50 ppm tolerance to avoid PLL jitter. For TDM cascading above 8 channels, the AD1938 can serve as a master driving up to 15 additional slave codecs on a shared data line. This page synthesizes distributor pricing, drop-in same-package alternatives, SPI register notes, and PCB-layout guidance not found in the manufacturer datasheet, providing a 360-degree engineering reference for sourcing the AD1938YSTZ.
AK4104ET - 192kHz 24-Bit 3.3V DIT | Asahi Kasei
Asahi Kasei Microsystems AK4104ET is a one-channel digital audio interface transmitter operating at up to 192kHz with 24-bit audio data and a 3.3V supply, packaged in a 16-pin TSSOP. It supports AES3, IEC60958, S/PDIF, and EIAJ CP1201 interface standards. A digital audio interface transmitter, or DIT, is an interface IC that accepts digital audio data and converts it into a standardized serial digital-audio stream. In the hierarchy, a DIT is a digital audio transmitter within the broader class of interface ICs and audio codecs. It is commonly used where a converter, processor, or digital audio source must deliver audio through a standardized digital connection. The AK4104 supports sample-rate operation to 192kHz and 24-bit audio resolution. Its audio serial port supports four formats, allowing the host interface to select an appropriate data arrangement. The verified distributor listing identifies the device as a one-channel digital audio transmitter in a 16-TSSOP package, while the manufacturer datasheet identifies AES3, IEC60958, S/PDIF, and EIAJ CP1201 compatibility. The 3.3V operating supply is suitable for low-voltage digital audio equipment where the transmitter must operate from a 3.3V rail. Digital audio standards commonly used with this class of device include S/PDIF for consumer equipment and professional digital audio links. Exact input and output logic levels, timing limits, and electrical characteristics should be confirmed in the manufacturer datasheet before final PCB sign-off. Typical applications include consumer electronics, professional audio equipment, digital mixers, audio conversion systems, home-theater products, and digital audio source or receiver equipment. The AK4104ET is also relevant when an existing design needs a dedicated transmitter rather than a general-purpose codec. A key design consideration is interface timing and grounding. The host must provide correctly formatted audio data, clocking, and control signals, while the digital output should be routed with controlled impedance and a properly terminated transmission line when required by the selected standard. Verify the exact pinout and recommended external components against the manufacturer datasheet. This page combines verified manufacturer and distributor information, standards context, pricing availability, and design guidance. It also clearly separates confirmed specifications from parameters that require additional datasheet data rather than presenting unsupported assumptions.
AK4113VF - 24-Bit 216kHz DIR Audio Receiver | AKM
Asahi Kasei Microsystems AK4113VF is a 24-bit stereo digital audio receiver with a 6:1 input selector and support for sampling rates up to 216kHz. It receives digital audio streams, decodes channel-status information, and provides an audio interface for downstream signal processing. The device is identified as a 192kHz 24-bit DIR with 6:1 selector in the retrieved product documentation. A digital audio interface receiver, or DIR, converts a serial digital-audio stream into a form suitable for the next stage of an audio system. In the hierarchy of audio semiconductors, a DIR is a digital-audio input interface, while a DAC is its complementary digital-to-analog converter. The AK4113VF is therefore positioned between an upstream digital source and downstream audio processing, conversion, or monitoring circuitry. The verified information identifies the AK4113VF as a 24-bit stereo receiver supporting rates up to 216kHz and providing a 6:1 selector. The retrieved technical description also states that the channel-status decoder supports consumer and professional modes and that the device automatically detects non-PCM bit streams. These features make it relevant to professional audio equipment, broadcast infrastructure, studio converters, and digital audio routing systems. The AK4113VF is not a power semiconductor, so a safe-operating-area chart is not applicable. Its package and electrical details must be confirmed against the manufacturer datasheet for the exact ordering variant. The retrieved datasheet listing identifies the package as a 30-pin VSOP with 0.65mm pitch. The available product results also reference an evaluation board and associated control, clock, and serial-interface signals. A typical signal chain places the AK4113VF after a digital-audio source and before a DAC, audio codec, DSP, or monitoring interface. The 6:1 selector can simplify front ends that must accept multiple digital sources, while the 24-bit representation preserves high-resolution audio data when the downstream system supports the corresponding word length. Digital input timing, grounding, and format compatibility remain the principal implementation considerations. This product page combines the verified AK4113VF datasheet identity, distributor listings, package information, and cross-reference search results. It also records unavailable parameters explicitly rather than estimating them, helping engineers compare the receiver against candidate replacements and evaluate whether a new PCB design or a qualified substitute is required.
AK4116VN 2.7-3.6V 24-Bit S/PDIF Receiver | AKM
Asahi Kasei Microdevices AK4116VN is a low-power, single-channel S/PDIF digital audio receiver supporting up to 24-bit resolution and 48 kHz operation. It operates from a 2.7 V to 3.6 V supply and is packaged in a 20-VFQFN package measuring 4.2 mm by 4.2 mm. The device accepts S/PDIF or AES/EBU digital audio signals and provides a channel-status decoder supporting consumer and professional modes. A digital audio receiver is an interface IC that recovers clock and audio data from a serial digital-audio stream. In the broader electronics hierarchy, it belongs to audio interface ICs, interface ICs, and semiconductors. Its role is to convert an S/PDIF signal into a parallel or serial digital-audio representation suitable for an audio codec, DSP, ADC, or digital signal-processing system. This makes the AK4116VN useful when a board must accept a digital audio source while preserving channel-status information. Key features include support for resolution to 24 bit, operation at 48 kHz, automatic detection of a non-PCM bit stream, and support for both consumer and professional channel-status formats. Its low-power design is appropriate for compact consumer, industrial, and embedded audio equipment where a 2.7 V to 3.6 V supply is already available. The 20-pin VFQFN construction also supports dense, surface-mount PCB layouts. The AK4116VN is a one-channel audio receiver rather than a general-purpose codec. It is intended to receive an externally supplied digital stream, recover timing information, decode channel status, and present audio data to downstream logic. A system designer must therefore provide compatible clocking, grounding, input conditioning, and output-routing arrangements around the device rather than treating it as a complete audio signal chain. Typical applications include digital-audio input for home theater equipment, multichannel receivers, active speakers, recording interfaces, professional audio equipment, and embedded media systems. The 24-bit capability suits modern consumer and professional digital-audio links, while automatic non-PCM detection helps distinguish linear PCM from encoded or compressed transfers. When designing around this IC, verify the complete pinout, input format, clock requirements, and output timing against the manufacturer datasheet before layout release. In particular, treat S/PDIF signal integrity, supply decoupling, and the distinction between consumer and professional channel status as key design checks rather than assumptions. This page consolidates verified AKM device data, sourcing links, replacement analysis, and application guidance for engineers evaluating the AK4116VN. It also distinguishes documented characteristics from parameters that require confirmation from the full manufacturer datasheet.
AK4117VF - 192kHz 24-Bit S/PDIF Receiver | AKM Semiconductor
AK4117VF is a low-power 192kHz digital audio receiver from AKM Semiconductor, with support for S/PDIF and AES/EBU streams up to 24-bit resolution in a 24-LSSOP package. The receiver operates from a 2.7V to 3.6V supply and covers sample rates from 32kHz to 192kHz. A digital audio receiver is an interface IC that recovers timing and audio data from a serial digital stream, commonly called S/PDIF, AES/EBU, or professional digital audio. It converts the incoming serial representation into a parallel PCM output that can be connected to an audio codec, processor, converter, or digital recording system. AK4117VF belongs to the interface IC hierarchy and combines input conditioning, clock recovery, data detection, and status handling in one device. Key features include 24-bit data handling, sample-rate support through 192kHz, and an integrated channel-status decoder. The decoder supports consumer and professional modes and can automatically identify a non-PCM bit stream. These capabilities make the receiver useful in digital audio equipment, surround-sound systems, broadcast equipment, and professional audio interfaces. The AK4117 is designed to pair with a multi-channel audio codec. AKM identifies the AK4527B and AK4529 as example companion devices for building a complete AC-3 system. This modular arrangement separates digital stream reception from analog or multichannel conversion, allowing the receiver to feed audio data into a codec selected for the required channel count and output topology. Electrical design should preserve the integrity of the incoming digital-audio connection. The receiver operates from a narrow 2.7V to 3.6V range, so supply regulation and decoupling are important. Board layout should keep the digital input route short, provide a continuous ground reference, and follow the manufacturer’s recommended pin and layout guidance to avoid data errors. This page combines the verified AK4117VF specifications, manufacturer datasheet information, procurement references, and application guidance. It also distinguishes documented drop-in alternatives from unverified functional equivalents so engineers can make package- and pin-compatible decisions with fewer assumptions.
AK4120VF - 20-Bit 48 kHz Audio SRC | Asahi Kasei
The Asahi Kasei Microdevices AK4120VF is a 20-bit, 48 kHz-class audio sample-rate converter with integrated mixer and volume functions, supplied in a 24-VSOP package. Its verified 2.7 V to 3.6 V analog supply range and 8 kHz to 96 kHz operating sample-rate range place it in the audio interface and signal-conversion category, where it can reconcile digital audio streams operating at different clock domains. A sample-rate converter, commonly abbreviated SRC, is an interface IC that converts audio samples from one sampling frequency to another. In the product hierarchy, an SRC sits between a digital audio interface and a clock-conversion function; unlike a general-purpose codec, the converter concentrates on format-compatible rate conversion, clocking, and associated mixing or gain control rather than performing analog-to-digital or digital-to-analog conversion. This distinction is important when selecting an SRC for a consumer, broadcast, automotive, or industrial audio system. The AK4120VF combines a 20-bit converter with a mixer and volume control in one device. These functions allow adjacent digital audio paths to be combined and level-adjusted before or during rate conversion, reducing the need for separate digital mixer packages in designs where routing flexibility is required. The converter is specified for sample rates from 8 kHz through 96 kHz and uses a 24-LSSOP, also identified by distributors as 24-VSOP, measuring approximately 0.220 inch by 0.220 inch in body width. The 2.7 V to 3.6 V operating supply supports integration with low-voltage digital audio systems. The verified operating temperature range is -40 °C to +85 °C, covering industrial temperature conditions. Its 24-pin surface-mount format is suited to compact audio boards where a through-hole package would be impractical. Typical uses include digital audio distribution, clock-domain adaptation between source and destination equipment, consumer audio receivers, broadcast and professional audio equipment, and automotive infotainment subsystems. When using the device, the system clock tree and digital audio interface format must be configured consistently with the manufacturer data sheet. Decoupling should follow the application circuit and layout guidance in that document rather than generic assumptions. This page consolidates the verified package, electrical-supply, resolution, sample-rate, temperature, and commercial-availability information into an engineering-oriented reference. It also distinguishes same-family AKM alternatives from a verified drop-in claim and flags pricing and detailed pin-level information that were not present in the supplied source data.
AK4125VF - 192kHz 24-Bit Stereo Audio SRC | AKM
The Asahi Kasei Microdevices AK4125VF is a stereo asynchronous digital audio sample rate converter with support for input and output rates from 8 kHz to 216 kHz and 24-bit data resolution, supplied in a 30-VSOP package. It converts between asynchronous clock domains so a digital audio system can operate its ADC, DAC, transport, or processor at different sampling rates while maintaining a common digital audio interface. A sample rate converter is an interface IC that changes the numerical sampling rate of a digital audio stream without requiring analog reconversion. Within the signal-chain hierarchy of semiconductor, interface IC, audio codec, and digital audio converter, the AK4125VF sits between source and destination audio devices. Its asynchronous operation allows the source and destination clocks to differ, making it useful in digital audio routing, recording, playback, broadcast, and consumer-equipment architectures. The verified product listing identifies the AK4125VF as a 2-channel, 24-bit converter reaching 192 kHz, with the device described as a high-performance asynchronous sample rate converter. Its input and output sample-rate coverage of 8 kHz through 216 kHz provides useful flexibility for studio, broadcast, and multimedia clock domains. The 30-VSOP, 0.220-inch, 5.60 mm-wide package is suitable for compact surface-mount audio equipment, although its pin count requires careful routing and connector-style layout planning. AK4125VF uses digital audio interfacing, with I²S identified in the verified distributor specification data. This makes it appropriate for systems whose source and sink expose compatible digital audio streams but cannot share a common sample-rate clock. In such designs, the converter can accept asynchronous data from one clock domain and present resampled data to another, reducing the need for redesigning the surrounding clock architecture. Typical uses include digital recording and playback interfaces, broadcast and studio signal processing, consumer audio equipment, and embedded systems that exchange audio between components operating at different rates. For each implementation, the designer must confirm clock topology, supported serial-audio format, master or slave operation, and control-interface requirements against the manufacturer datasheet before schematic signoff. A key design consideration is that asynchronous conversion does not remove the need for correct clock generation, grounding, and digital-signal routing. The source clock and destination clock should be clean, jitter-compatible signals, while the 30-pin package demands deliberate escape routing and stable supply decoupling. Validation should include sample-rate sweeps, lock behavior, word-length operation, and channel synchronization. This page consolidates verified distributor availability, package data, operating limits, and practical selection guidance. It distinguishes published AK4125VF attributes from values that still require confirmation from the manufacturer datasheet, helping engineers evaluate this converter before requesting a quote or committing a board layout.
AK4127VF - 192kHz 24-Bit Stereo Asynchronous SRC | AKM
The Asahi Kasei Microdevices AK4127VF is a 192kHz, 24-bit high-performance stereo asynchronous sample-rate converter with an internal PLL, supplied in a 30-pin VSOP package. It converts digital audio streams between input and output sampling rates ranging from 8kHz to 216kHz. Its primary role is clock-domain adaptation and asynchronous audio-rate conversion rather than analog amplification or DAC conversion. A digital sample-rate converter, or SRC, changes the sampling rate of a digital audio stream while preserving the represented signal. In an audio signal chain, an SRC sits between digital audio sources and receivers such as a DAC, codec, recorder, or playback engine. It helps reconcile mismatched clocks and rates while adding asynchronous rate conversion and phase-locked-loop synchronization functions. The AK4127VF supports stereo operation, 24-bit processing, and sample rates up to 192kHz according to the verified product description. Its internal PLL simplifies system configuration by helping establish the required digital audio timing. The device is intended for asynchronous audio applications where source and destination rates differ and predictable clock recovery is required. The AK4127VF is packaged in a 30-pin VSOP, also identified in distributor data as a 30-VSOP. The verified sources describe the part as an audio special-purpose or audio sample-rate converter IC. It is relevant to professional audio equipment, digital recording systems, broadcast and networking equipment, and other systems requiring digital audio format and clock-rate adaptation. Designers should verify the complete electrical interface, pinout, supported digital audio formats, clock requirements, supply rails, and control protocol against the manufacturer datasheet before layout. Because the supplied data does not include all pin-level electrical limits or interface timing values, those details are intentionally marked as needed rather than inferred. This page summarizes verified product positioning and source data for AK4127VF evaluation.
AK4136VQ - 2-Channel 384kHz 32-Bit SRC | AKM
The Asahi Kasei Microsystems AK4136VQ is a 2-channel asynchronous digital audio sample rate converter operating from 8kHz to 384kHz, processing up to 32-bit audio data in a 48-LQFP package. Its internal oscillator eliminates the need for an external master clock when used in asynchronous systems, simplifying clock-domain conversion between source and destination audio devices. A digital sample rate converter, or SRC, is an interface IC that resamples and filters digital audio between clocks with different frequencies. In the product hierarchy, a SRC belongs to digital audio signal-processing ICs and supports audio converters, receivers, network-audio equipment, and test systems. The AK4136 is particularly useful when a transmitter or player uses a different clock domain from the audio source. The AK4136 supports two channels, 32-bit audio processing, and input and output sample rates spanning 8kHz through 384kHz. According to the manufacturer datasheet, its internal oscillator means that external master clocks are not required for asynchronous conversion. This can reduce clock-tree complexity, although the converter still requires correct power, ground, digital-interface, and signal-integrity implementation. The 32-bit format and 384kHz maximum rate suit high-resolution audio paths, studio equipment, broadcast infrastructure, digital audio workstations, and consumer systems that need format or clock-domain adaptation. A practical design must still account for required input and output formats, supported clock configurations, filter behavior, grounding, decoupling, and signal routing; those details must be checked in the complete manufacturer documentation. The principal design consideration is integration. The AK4136VQ can simplify systems where input and output rates are unrelated because its internal oscillator supports asynchronous operation. This avoids one external master-clock source, but it does not eliminate the need for stable local supplies, short digital traces, controlled return paths, and verification of the converter configuration during startup. This page consolidates the verified manufacturer datasheet, distributor availability, package information, technical search context, and practical engineering guidance. It intentionally avoids asserting electrical, pin, lifecycle, or drop-in replacement details that were not present in the supplied source data.
AK4137EQ - 2-Channel 32-Bit 768kHz Audio SRC | AKM
The Asahi Kasei Microdevices AK4137EQ is a two-channel digital sample rate converter supporting input and output sample rates from 8kHz to 768kHz, with 32-bit processing and PCM-DSD conversion capability in a 48-LQFP 7x7 mm package. It is designed for high-end audio systems, USB-DACs, and digital-audio equipment that must reconcile different sampling domains. A digital sample rate converter is an interface IC that resamples or changes the clock rate of digital audio data between incompatible source and destination rates. The AK4137EQ includes an internal oscillator, so it does not require an external master clock for its asynchronous conversion operation. This simplifies system configuration when the source and destination audio domains use unrelated clocks. Key features include two-channel operation, 32-bit sample processing, asynchronous conversion, input and output rates spanning 8kHz to 768kHz, and PCM-DSD data conversion. The internal oscillator reduces external clock infrastructure, while the broad sample-rate range supports professional audio, USB-DAC, and multirate digital-audio designs. The device is supplied in a 48-LQFP package identified by DigiKey as 48-LQFP (7x7). The AK4137EQ is intended for applications where audio data arrives from a source clock domain and must be delivered to a DAC, decoder, recorder, or other digital-audio subsystem operating at another rate. Its sample-rate flexibility makes it useful for format bridging, asynchronous playback paths, and PCM-to-DSD or DSD-to-PCM conversion as described by the manufacturer. Designers should verify the complete pinout, interface logic, clock requirements, supported audio formats, control signaling, and electrical limits against the manufacturer datasheet before layout or firmware implementation. The available verified sources do not provide complete pricing, pin-level assignments, or detailed timing tables, so those values remain data-needed rather than inferred.
AK4621EF - 24-bit 192kHz Stereo Audio CODEC | Asahi Kasei Microdevices
The Asahi Kasei Microdevices AK4621EF is a high-performance 24-bit stereo audio CODEC supporting up to 192 kHz recording and playback, housed in a 30-pin VSOP (LSSOP) surface-mount package. It integrates two channels of ADC and two channels of DAC on a single die, with 115 dB dynamic range on both ADC and DAC paths and supports sample rates from 8 kHz to 192 kHz. An audio CODEC is a single integrated circuit that combines an analog-to-digital converter (ADC), digital-to-analog converter (DAC), and the necessary clocking, filtering, and serial interface logic to move bidirectional audio between the analog and digital domains. In the broader component taxonomy, the AK4621EF belongs to the audio interface IC family, which sits under interface ICs and ultimately under integrated circuits. It is a sub-category of audio ICs and is specialized for professional and consumer high-fidelity stereo capture and reproduction. Key features include AKM's Enhanced Dual-Bit ADC architecture for high dynamic range and the Advanced Multi-Bit DAC architecture combined with a Switched Capacitor Filter (SCF) that suppresses out-of-band noise and improves jitter tolerance. The serial audio interface is compatible with common I2S, left-justified, and right-justified formats, and the analog supply operates from 4.75 V to 5.25 V while the digital supply operates from 3.0 V to 3.6 V. The device also includes a low-latency digital filter option and a hardware-selectable HPF for DC-offset removal on the ADC path. The architecture combines sigma-delta modulators for both ADC and DAC with on-chip digital decimation and interpolation filters. The SCF output stage of the DAC removes many of the high-frequency images associated with conventional oversampling DACs, reducing the need for aggressive external post-filtering. This combination delivers wide dynamic range with low distortion at full-scale, which is critical for transparent studio capture and playback. Typical applications include Pro Audio sound cards, Digital Audio Workstations (DAWs), DVD-R, hard-disk, and CD-R recording and playback systems, as well as musical instrument recording. The combination of high dynamic range, low latency mode, and integrated PLL makes it suitable for both consumer recording devices and semi-professional production equipment. When designing with this part, observe the separate analog and digital supply domains (5 V analog, 3.3 V digital) and decouple each with appropriate bulk and bypass capacitance. The hardware mode pins configure audio data format, master/slave clocking, and HPF behavior; review them before PCB layout because they cannot be changed in software. This page synthesizes distributor pricing, drop-in and equivalent options, and practical layout guidance beyond what is printed in the manufacturer datasheet.
ALC5660-CG - 106dB SNR Stereo Audio Codec w/ HP Amp | Realtek
The Realtek ALC5660-CG is a high-performance, low-power stereo audio codec with an integrated Class-G headphone amplifier, packaged in a surface-mount QFN and specified for 106 dB SNR with a 24-bit I2S/PCM digital interface. It accepts stereo digital audio over I2S/PCM and drives line outputs, headphone loads, and microphone inputs through on-board ADCs, DACs, and a flexible analog mixing block. The device targets portable and embedded designs where board space, BOM cost, and audio fidelity must coexist with strict power budgets. An audio codec is a mixed-signal integrated circuit that bridges the digital audio domain (typically I2S, PCM, TDM, or S/PDIF streams) and the analog audio domain (headphones, line-level outputs, microphones, and line-level inputs). Inside the codec, multi-bit sigma-delta ADCs digitize analog microphone and line-in signals, while multi-bit sigma-delta DACs reconstruct analog waveforms from the incoming digital stream. The ALC5660 belongs to the broader hierarchy of audio interface ICs > interface ICs > signal chain ICs > mixed-signal semiconductor; codecs are distinguished from pure DACs or ADCs by their bidirectional stereo architecture and integrated mixer, microphone preamplifier, and (in this case) headphone driver. Key features of the ALC5660-CG include a 106 dB SNR DAC, 24-bit resolution across the digital audio interface, integrated headphone amplifier with ground-referenced Class-G topology, support for microphone bias and digital microphone inputs, and a flexible digital mixing engine. The Class-G amplifier dynamically scales the internal supply rail to match the instantaneous headphone output level, sharply reducing quiescent dissipation during quiet passages while still delivering ample headroom for peaks. Technically, the ALC5660 uses multi-bit sigma-delta modulator cores for both the record (ADC) and playback (DAC) paths, with on-chip digital decimation/interpolation filters. The headphone path uses a charge-pump-derived negative rail and a dynamically switched supply to maximize efficiency at low output levels. All gain, mixing, and routing is controlled through a two-wire (I2C-style) control interface. Typical applications include tablets, ultrabooks, docking stations, portable media players, USB-C headsets, and single-board computer audio hats. The QFN package and integrated HP amp minimize PCB area, while the high SNR suits music playback and VoIP. Design consideration: the Class-G topology relies on adequate bulk decoupling near the HP supply pins, and the digital I2S/PCM bus should be length-matched and routed away from the analog headphone outputs to preserve SNR. This page consolidates distributor pricing, drop-in alternative candidates, and engineering design notes not available in the standalone manufacturer datasheet, helping engineers evaluate sourcing options and PCB integration in one place.
BD3452KS - 8-Channel 7.1ch Sound Processor | ROHM
ROHM Semiconductor BD3452KS is a 7.1-channel sound processor with eight output channels, a nine-channel input selector, eight-channel volume control, and gain amplification integrated into a single chip. The device operates from 13 V to 14.6 V and is offered in a 100-SQFP package measuring 14 mm by 20 mm. A sound processor is an audio signal-processing IC that selects among multiple audio sources, applies channel gain and volume control, and distributes the resulting signals to multichannel output stages. Within the broader semiconductor hierarchy, the BD3452KS belongs to the audio processing IC and signal processor categories, where it bridges source selection, channel-level control, and home-theater audio system routing. The principal documented features include a nine-channel input selector, eight-channel volume control, eight-channel gain amplification, and support for 7.1-channel audio processing. These functions are intended for AV receivers, home-theater systems, and mini-component systems. The consolidated implementation reduces the number of separate routing and volume-control blocks required in a multichannel receiver. The BD3452KS uses a two-wire control interface according to the distributor product information. This interface is intended for integration with a system controller while leaving the device's signal-processing and channel-adjustment functions local to the audio subsystem. Its 13 V to 14.6 V operating range indicates that it is designed around a conventional internal receiver supply rather than a low-voltage portable system. Typical application roles include AV receiver channel routing, home-theater input selection, and mini-component multichannel amplification control. In these systems, the processor can select the appropriate source and apply per-channel gain and volume before signals reach external power amplifiers or downstream audio circuitry. The eight-output-channel architecture is particularly relevant to surround-sound configurations. For new designs, verify the complete pin configuration, control protocol, electrical limits, and signal-path behavior against the manufacturer datasheet before layout or firmware development. Because the available verified information identifies the package and core functions but does not provide a complete pin table or detailed timing data, those implementation details should not be inferred from generic sound-processor designs. This page consolidates verified distributor information, the manufacturer-hosted datasheet, function-level application guidance, selection criteria, and explicitly identified data gaps. It is designed to provide practical procurement and engineering context without presenting unverified electrical values, replacement parts, or lifecycle assumptions as established facts.
BD3464FV-E2 - 4-Channel I2C Audio Volume Controller | ROHM
The ROHM BD3464FV-E2 is a general-purpose four-channel electronic volume controller with a built-in advanced switch, an operating supply range of 7 V to 9.5 V, and a 20-pin SSOP-B package. It uses an I2C control interface, supports 5 V control, and is intended for multichannel audio systems that require software-controlled attenuation. The device is classified by ROHM as not recommended for new designs, so its appropriate use is primarily in established designs requiring lifecycle support rather than new platform creation. An audio volume controller is an audio signal-processing IC that applies electronic attenuation between source and amplifier stages. Its broader hierarchy is audio volume controller, audio processor, analog audio IC, interface IC, and semiconductor. Unlike a passive potentiometer, the BD3464FV-E2 permits volume changes through an I2C command while preserving a controlled signal path and reducing dependence on mechanical adjustment. Key features include four independently controlled audio channels, an advanced-switch circuit intended to reduce volume-control switching noise, and low-current operation implemented with a Bi-CMOS process. The manufacturer data also identifies I2C control, 5 V control compatibility, a 7 V to 9.5 V supply range, and placement of I/O terminals together to simplify PCB layout and reduce board area. A distributor description reports RoHS3 compliance. Technically, the part combines digital command processing with an analog attenuation function. The advanced switch is intended to suppress noise associated with changing the attenuation state, which is important when volume adjustments occur during music or speech playback. Its Bi-CMOS implementation combines bipolar and CMOS characteristics with an emphasis on low current consumption. The complete electrical limits, control register map, attenuation steps, frequency response, and distortion specifications require confirmation from the manufacturer datasheet. Typical uses include powered speakers, mini hi-fi systems, automotive head units, home-theater receivers, and other multichannel audio products in which software control is preferable to a mechanical volume control. The I2C interface also allows a system controller to coordinate attenuation with playback status, mute functions, or source selection. Designers should verify the complete pin function map, recommended input and output coupling network, I2C address configuration, and external component values against the manufacturer datasheet before making a board revision. Because the part is not recommended for new designs, new products should generally use a currently recommended audio volume-control IC unless continuity with an existing platform is required. This page consolidates the verified manufacturer and distributor information, distinguishes documented capabilities from unavailable technical details, and provides application and design guidance without inventing missing electrical limits, pricing, pin assignments, or drop-in equivalents.
BD3808FS-E2 Audio Processor IC | ROHM Semiconductor
ROHM Semiconductor BD3808FS-E2 is an audio-special-purpose integrated circuit classified by DigiKey as an Audio Channel device and by other distributor listings as an audio processor. The supplied product information identifies a 22SOP package and associates the device with medium-power linear and switching applications. These verified facts place the BD3808FS-E2 in the broader category of audio processing ICs, where dedicated circuitry manages audio signal paths or associated control functions within consumer and industrial audio equipment. An audio processor is a specialized application-specific semiconductor. Unlike a general-purpose logic IC, it is designed around audio-channel requirements, including signal handling, channel control, or functions used to implement an audio system. The BD3808FS-E2 belongs to this audio-special-purpose IC hierarchy, which sits within integrated circuits and ultimately within the semiconductor component category. Its 22SOP package provides the physical interface used to connect the device on a printed circuit board. The verified distributor records distinguish the exact BD3808FS-E2 ordering code from the broader BD3808FS family listing. DigiKey identifies BD3808FS-E2 as an Audio Channel part, while marketplace records describe it as an IC audio processor in a 22SOP package. A distributor description also associates it with medium-power linear and switching applications. These classifications suggest that engineers should review the manufacturer datasheet for the exact channel architecture, electrical limits, external component requirements, and supported signal topology before releasing a production design. From a sourcing perspective, the BD3808FS-E2 is listed across DigiKey, Octopart, and multiple distributor or catalog channels. The verified search results report availability or comparison coverage, but they do not provide numeric pricing, stock quantities, operating-temperature limits, supply-voltage limits, output-current ratings, gain, distortion, frequency response, pin assignments, or package dimensions. Consequently, those electrical and mechanical parameters cannot be safely inferred from distributor category labels and must be confirmed in the manufacturer documentation. Typical design investigation for this device would focus on medium-power audio systems, signal routing, channel management, and mixed linear or switching signal-chain implementations. Whether it is suitable for a particular headphone, speaker, receiver, or auxiliary audio board depends on its documented supply range, channel count, control interface, and electrical performance. Engineers should also verify footprint compatibility, exposed or non-exposed pad details, pin multiplexing, and recommended application circuitry rather than relying only on the 22SOP package name. The main purchasing risk is incomplete verification: the supplied data does not establish lifecycle status, automotive qualification, RoHS or REACH declarations, exact package drawing, or a manufacturer-hosted datasheet URL. Samples and lot traceability should therefore be obtained before substitution. This page distinguishes verified distributor facts from unknown technical fields and avoids presenting estimated electrical ratings as if they were manufacturer specifications. This page adds engineering-focused value by mapping the verified 22SOP package and audio-processor classification to likely design checks, documenting all available source links, and identifying the exact data still required for footprint validation, schematic review, sourcing, and production release. It also explains why an apparently similar BD3808FS order code should not be assumed to be a drop-in alternative without package and pin-level verification.
BD3809FS - Audio Channel Processor | ROHM Semiconductor
ROHM Semiconductor BD3809FS is an audio-channel integrated circuit identified by distributor records as an audio processor in a 22SOP package. The available catalog information describes the device for medium-power linear and switching applications, while the primary distributor listing classifies it under Audio Special Purpose and Audio Channel. These verified identifiers establish the component family and form factor, but the supplied source excerpts do not include a complete manufacturer datasheet parameter table. An audio-channel processor is an application-specific integrated circuit used to process, control, or route audio signals within an audio subsystem. It belongs to the broader hierarchy of audio ICs, analog and mixed-signal semiconductors, and signal-processing integrated circuits. Compared with a general-purpose operational amplifier, an application-specific audio processor may incorporate multiple functions, but the exact internal blocks and signal paths of BD3809FS cannot be confirmed from the supplied excerpts. The verified package identifier is 22SOP. That package provides 22 physical lead positions, but the supplied data does not include a pinout, pin names, exposed-pad information, dimensions, or pin-to-pin replacement cross-reference. The catalogs also identify BD3809FS-E2 as an ordering designation associated with the same device family. Its electrical details, marking code, temperature range, and ordering-code differences require confirmation from the manufacturer datasheet. BD3809FS is associated with audio-channel applications and medium-power linear and switching application descriptions. These descriptions make it relevant to audio equipment, signal-conditioning circuits, and communication equipment where an audio-channel IC is required. However, verified operating voltage, input and output signal range, channel bandwidth, gain, distortion, channel count, control interface, and output-drive specifications are not included in the supplied source data and should not be inferred from the product type alone. Because the available evidence is distributor-oriented rather than a complete technical datasheet, engineers should verify pin functions, supply requirements, timing, absolute maximum ratings, decoupling recommendations, and application circuitry before committing to a design. A replacement should be selected only after confirming the exact 22SOP footprint, pin assignment, control protocol, electrical limits, and signal performance. This page consolidates the verified BD3809FS identity, package, category, distributor availability references, replacement-search evidence, and engineering-selection cautions. It separates confirmed catalog facts from unavailable technical parameters and provides a practical verification path for schematic capture, PCB layout, sourcing, and component-risk review.
BD3812F-E2 - 2-Channel Audio Volume Controller | ROHM
The ROHM Semiconductor BD3812F-E2 is a 2-channel electronic audio volume controller with 1.2 uVrms residual noise, a 131 dB dynamic range under the IHF-A weighting condition, and independent channel attenuation from 0 dB to -103 dB. The device also provides MUTE operation in 1 dB steps and uses two-line serial control. It is supplied in a 14-pin SOIC package. An electronic volume controller is an audio signal processor that adjusts the level of one or more audio channels through a control interface rather than a mechanical potentiometer. In the product hierarchy, it belongs to the broader class of audio special-purpose ICs, which sit between analog signal-generation blocks and power amplifiers. The BD3812F-E2 is designed to preserve audio signal quality while providing repeatable, software-controlled attenuation for multi-channel audio systems. The BD3812F-E2 supports independent control of two channels over a 103 dB attenuation range. Its mute function changes the level in 1 dB steps, making it suitable for product designs that need fine attenuation control, channel balancing, or user-selected volume settings. When combined with any of the BD3811K1, BD3813KS, BD3814FV, or BD3815KS family members, as specified by ROHM, the common bus supports up to eight channels. The common-bus architecture can also control as many as four chips at the same time. The maximum output voltage is 4.2 Vrms under the stated conditions of VCC = 7 V, VEE = -7 V, and RL = 10 kOhm. This output capability, combined with a residual-noise specification of 1.2 uVrms, supports low-noise audio signal paths. The two-line serial control reduces the number of control connections required compared with a parallel volume-control arrangement, which can also simplify PCB routing and microcontroller interface design. Typical uses include television and monitor audio channels, home-audio systems, active speakers, audio mixers, and multi-zone entertainment equipment. The device is particularly useful where independent left and right attenuation, common-bus expansion, mute control, or precise software-defined volume adjustment is required. Because its operating supply range is specified as 5 V to 7.3 V, the surrounding circuit must provide compatible positive and negative analog rails where the application requires the stated 4.2 Vrms output capability. Designers should preserve the datasheet-specified bypass and output-load conditions when evaluating signal integrity. The 10 kOhm load used for the stated maximum-output condition is relevant to downstream amplifier planning, and any external coupling or amplifier input must not load the output beyond the supported operating conditions. Decoupling should be placed close to the supply pins, and the common-bus wiring of multiple devices should be kept short to minimize coupling between channels. This product data combines the verified ROHM feature description with distributor packaging information, practical multi-channel application guidance, and explicit data-gap markers. It distinguishes verified numeric performance from unreported specifications and does not infer a pin assignment, temperature range, or drop-in substitute that the supplied sources do not confirm.
CS3308-CQZR - 8-Ch Analog Volume Control, 48-LQFP | Cirrus Logic
The Cirrus Logic CS3308-CQZR is a high-performance, low-power, 8-channel digitally controlled analog volume control IC designed for high-end audio systems, housed in a 48-pin LQFP (7x7 mm) package with surface-mount mounting. Key specifications include 8 independent volume channels, configurable zero-crossing detection for glitch-free level changes, a serial control interface compatible with I2C and SPI, a supply voltage range of 3.1 V to 5.25 V, and the ability to address up to 128 CS3308 devices on a single 2-wire I2C or 3-wire SPI bus for high-channel-count systems. The device targets professional audio equipment and consumer electronics that demand clean, click-free volume transitions. A digital volume control IC is an integrated circuit that performs attenuation or amplification of analog audio signals under the command of a microcontroller or DSP, replacing mechanical potentiometers in modern equipment. Within the audio signal chain, it sits between the source selector or DAC output and the power amplifier, and it belongs to the broader family of audio special-purpose ICs alongside DACs, ADCs, and audio amplifiers. Compared with mechanical potentiometers, a digitally controlled attenuator offers precise repeatability, remote control, memory recall, and absence of mechanical wear. Key features of the CS3308-CQZR include 8 independent channels with 0.5 dB step resolution, configurable zero-crossing detection to eliminate zipper noise during volume changes, a wide supply range of 3.1 V to 5.25 V suitable for 3.3 V and 5 V system rails, selectable I2C or SPI serial control interface, and hardware-addressable device IDs that allow up to 128 cascaded devices on a single bus. The combination of fine resolution and zero-crossing logic preserves audio signal integrity during level transitions. Technically, the CS3308 uses a resistive ladder network with CMOS switches and an internal zero-crossing detector that holds the wiper position until the audio waveform crosses zero, suppressing audible artifacts. The architecture provides very low total harmonic distortion plus noise (THD+N) and high channel-to-channel matching, which are essential parameters for multi-channel professional audio gear. Programmable address pins enable bus expansion without adding microcontroller I/O. Typical applications include professional audio mixing consoles, multichannel AV receivers and amplifiers, broadcast studio equipment, automotive head units and infotainment amplifiers, and home theater preamplifiers. In each case the CS3308 replaces mechanical or electromechanical volume controls with a precise, software-driven solution that supports remote control and preset recall. When designing with the CS3308-CQZR, ensure that analog input signals remain within the recommended common-mode and clip range, place bypass capacitors close to the supply pins, and verify I2C/SPI address selection when stacking multiple devices on the same bus. The LQFP-48 footprint gives good thermal performance for continuous signal-path operation. This page synthesizes current distributor pricing and stock, drop-in alternatives with the same 48-LQFP footprint, and practical design notes not consolidated in the manufacturer datasheet, delivering actionable information for both new designs and legacy audio system refreshes.
CS4205-KQ - AC'97 Stereo Audio Codec | Cirrus Logic
The Cirrus Logic CS4205-KQ is an AC'97 2.1-compliant stereo audio codec designed for PC multimedia systems, portable computing, and docking-station equipment. It uses CrystalClear delta-sigma and mixed-signal technology and is offered in a 48-pin LQFP package measuring 7 mm by 7 mm. Distributor records identify the interface as a 20-bit serial audio codec with 18-bit resolution on one associated serial interface. An audio codec is an interface IC that converts digital audio data into analog signals and analog audio inputs into digital data. In the signal chain, it sits between a host audio controller and external microphones, speakers, line inputs, or line outputs. AC'97 is a legacy Intel-defined audio interface that combines serialized digital audio with control, status, and general-purpose digital signals. The CS4205 therefore belongs to the broader audio codec, multimedia interface, and signal-conversion IC categories. The manufacturer datasheet identifies the CS4205 as a stereo co-dec for portable-computing systems. Its AC'97 2.1 compliance supports host-controlled multimedia operation, while CrystalClear delta-sigma and mixed-signal circuitry provides the analog-to-digital and digital-to-analog conversion functions. The 48-LQFP package exposes the complete AC'97 interface and associated analog and power connections on a compact surface-mount footprint. Technical implementation requires care because an audio codec combines sensitive analog inputs with a high-speed digital bus. The codec must be connected to a compatible AC'97 controller, supported audio drivers, analog input and output circuitry, and correctly decoupled supply rails. Clocking, digital signal integrity, analog grounding, output loading, and the relationship between the codec's sample rate and system timing must all be evaluated against the manufacturer datasheet. Typical uses include notebook-computer multimedia subsystems, docking stations, portable audio hardware, and legacy PC docking solutions. The Cirrus Logic CS4205 is particularly relevant when maintaining or repairing an existing AC'97 platform rather than designing a new system around a modern USB or I2S audio interface. Its stereo codec function supports the familiar PC multimedia architecture but requires legacy host, driver, and connector support. Designers should preserve the manufacturer's recommended decoupling and analog-layout practices and verify the exact pin assignments and signal names against the device documentation. AC'97 implementations can be sensitive to trace impedance, grounding, clock jitter, and power-supply noise. Because the CS4205-KQ is a legacy peripheral-oriented codec, software-driver availability and the continued support of the surrounding platform are as important as electrical compatibility. This page consolidates verified distributor availability, pricing, interface details, package information, and application context. It also distinguishes documented drop-in alternatives from related legacy codecs, helping engineers avoid treating a merely similar audio codec as a guaranteed pin-to-pin replacement.
CS4207-CNZR - 4-In/6-Out HD Audio Codec w/ HP Amp | Cirrus Logic
The Cirrus Logic CS4207-CNZR is a highly integrated, multi-channel, low-power HD Audio Codec combining 192 kHz DACs, 96 kHz ADCs, 192 kHz S/PDIF transmitter and receiver, microphone preamp with bias voltage, and a ground-centered headphone driver, all in a 48-pin QFN (6x6 mm) package. The device operates from a single 3.3 V digital supply and a 5 V analog supply, delivers 24-bit serial audio data, and supports both HDA (HD Audio) and I2S serial interfaces to host controllers. An HD Audio Codec (High Definition Audio Codec) is a mixed-signal IC that converts digital audio streams from a PC/system chipset into line out the headphone, line stage, and microphone signals using internal DACs, ADCs, and headphone amplifiers. The codec sits in the audio signal chain between the host controller (e.g., Intel HDA-compliant chipset) and the analog transducers, and the term "HD Audio" denotes compliance with the Intel HDA Link specification rather than a frequency or bit-rate definition. CS4207-CNZR occupies the power-management IC and interface-IC tier of the audio subsystem, replacing multiple discrete ADCs, DACs, S/PDIF transceivers, and headphone amps with one package. The codec is the culmination of the signal chain (digital I/O -> mixed-signal codec -> analog out). Key features of the CS4207-CNZR include a multibit Delta-Sigma modulator architecture that enables infinite sample-rate adjustment from 32 kHz to 192 kHz, integrated ground-centered headphone driver (no DC blocking capacitors required at the output), microphone preamp with bias voltage source, and digital microphone interface support. The device supports 4 analog line/microphone inputs and 6 analog outputs, providing flexible routing for desktop, notebook, and embedded audio platforms. S/PDIF transmitter and receiver operate up to 192 kHz, allowing simultaneous digital audio transmission and reception. Technical depth: the Delta-Sigma modulator multibit topology achieves higher dynamic range and linearity than single-bit designs, while the ground-centered headphone architecture eliminates large output coupling capacitors, reducing BOM cost and improving low-frequency bass response. The codec integrates digital mixing, gain control, and jack-presence detection through the HDA link, simplifying host-driver interaction. Typical applications include desktop and notebook PC audio subsystems, all-in-one motherboards, embedded media players, and HD Audio-compatible docking stations. The wide format compatibility (HDA + I2S) and integrated HP amp make CS4207 well-suited for ultra-thin notebooks and small-form-factor designs where PCB area is constrained. When designing with this device, place the analog supply decoupling capacitors as close to the AVDD pins as possible to minimize analog-domain interference, and follow Cirrus Logic reference layout for the headphone output return paths to maintain low THD+N. The integrated HP amp eliminates external coupling capacitors but requires careful ground-centered routing to preserve low noise. This page synthesizes distributor pricing, parametric drop-in alternatives, and practical design notes that go beyond the manufacturer datasheet. Engineers comparing HD Audio codec options for desktop/notebook designs will find the application notes, thermal considerations, and HDA source documents consolidated in this page.
CS4216-KL - 16-Bit Stereo Audio Codec | Cirrus Logic
Cirrus Logic CS4216-KL is a monolithic CMOS 16-bit stereo audio codec with integrated analog-to-digital and digital-to-analog conversion, filtering, and level setting. The device provides four audio inputs and two audio outputs for computer multimedia systems, supports word rates up to 50 kHz, and multiplexes left- and right-channel digital audio onto one serial data bus. It is supplied in a 44-pin PLCC package measuring 16.59 mm by 16.59 mm. An audio codec is a mixed-signal interface IC that converts audio between analog and digital domains. Its hierarchy is audio codec, multimedia codec, analog-to-digital and digital-to-analog converter, interface IC, and semiconductor. The CS4216 combines those functions so a digital computer system can accept analog audio sources, process samples digitally, and reproduce them through analog outputs without separate converter packages. Key capabilities include 16-bit stereo conversion, four analog inputs, two analog outputs, integrated interpolation and filtering functions, and a shared serial data interface. The CS4216 family targets computer multimedia, automotive, and portable audio equipment. Its integrated filtering and level-setting functions simplify board partitioning, while its multiplexed digital bus reduces the number of interconnects required between the codec and host or digital-audio controller. Technically, the monolithic CMOS implementation combines two conversion channels with shared digital serialization. Interpolation filters suppress unwanted high-frequency content and reduce aliasing in the reconstruction path, while the digital interface carries both stereo channels using word rates up to 50 kHz. The 44-pin PLCC offers a surface-mount package with J-lead termination and a body size of 16.59 mm by 16.59 mm. Typical uses include desktop and notebook multimedia audio, portable-audio equipment, automotive audio systems, and embedded systems requiring simultaneous analog capture and playback. In these applications, the four inputs can serve microphones, line-level sources, or other analog audio points, while the two outputs feed downstream amplifiers or audio routing stages. Host connectivity still requires correct timing, format, and signal-level design. The main design consideration is package and signal planning. Because the converter is a 44-pin PLCC rather than a modern leadless package, its footprint has limited long-term layout flexibility. Board layout should preserve analog and digital return paths, provide suitable bypassing, and follow the manufacturer's pin and reference-circuit guidance rather than assuming a generic codec footprint. This page consolidates the verified CS4216-KL identity, package, converter topology, application fit, and verified drop-in alternatives. It also distinguishes documented capabilities from parameters that still require confirmation from the manufacturer datasheet or a live distributor quotation.
CS4221-KS - 24-Bit Stereo Audio Codec | Cirrus Logic
Cirrus Logic CS4221-KS is a 24-bit stereo audio codec with a 3 V interface, integrating stereo analog-to-digital and stereo digital-to-analog converters in a 28-pin SSOP package. The device supports audio sampling up to 50 kHz according to distributor product data and uses delta-sigma conversion techniques for both converter directions. Its serial interface and mixed-signal codec architecture are intended to reduce the component count in digital audio equipment. An audio codec is an interface IC that converts analog audio signals into digital data and digital audio data back into analog signals. Within the broader semiconductor hierarchy, a codec belongs to the audio interface and signal-conversion class of integrated circuits. The CS4221-KS therefore serves as the bridge between an external digital audio processor or controller and external analog input and output circuitry, rather than functioning as a complete audio playback system by itself. The principal verified features include 24-bit resolution, simultaneous stereo ADC and DAC functionality, a 3 V interface, serial digital connectivity, and a maximum stated sampling frequency of 50 kHz. These attributes suit equipment requiring compact digital-audio conversion at a conventional CD-oriented sample rate. The integrated stereo topology also helps minimize the number of external converter channels required in a stereo design, although the application still needs suitable input and output analog circuitry. The CS4220 and CS4221 family uses delta-sigma conversion, according to the manufacturer datasheet summary reproduced by the verified search results. Delta-sigma conversion shapes quantization noise into a digitally filtered form, supporting the device's stated 24-bit resolution. The available verified snippets do not expose the internal architecture, clock requirements, analog performance metrics, or digital interface mode in sufficient detail to characterize the converter further without unsupported assumptions. Typical uses include professional and consumer stereo audio equipment, digital mixing equipment, sound-card circuitry, multimedia appliances, and embedded audio systems that require both record and playback channels. The same 50 kHz limitation that makes the codec suitable for conventional stereo audio applications also makes it less appropriate for designs needing higher-rate professional audio, unless the complete system requirements are limited to 50 kHz. When laying out a CS4221-KS design, analog signal routing should be kept separate from digital clock and serial-data paths, and the power, ground, and bypass connections must follow the manufacturer's recommended application circuit. Because the supplied verified data does not include a complete pin table or decoupling values, engineers should verify pin assignments and recommended passive values against the official Cirrus Logic datasheet before release to production. This page consolidates the verified distributor specifications, package information, lifecycle observations, source links, selection guidance, and engineering considerations for CS4221-KS. It also distinguishes confirmed characteristics from parameters that require direct confirmation from the manufacturer datasheet, helping procurement and design teams avoid treating incomplete distributor metadata as a complete electrical specification.
CS4222-BS - 20-Bit Stereo Audio Codec | Cirrus Logic
Cirrus Logic CS4222-BS is a 20-bit stereo audio codec with volume control, integrating stereo analog-to-digital and stereo digital-to-analog converters in a 28-pin SSOP package. The device operates from a single +5 V supply and uses delta-sigma conversion techniques for its stereo conversion paths. These core characteristics make the CS4222-BS a monolithic audio interface IC for systems that require both digital-to-analog playback and analog-to-digital recording functions. An audio codec is a mixed-signal semiconductor that converts audio between digital data and analog voltage signals. Within the broader integrated-circuit hierarchy, a codec belongs to the interface and audio-converter class, combining an ADC, DAC, volume control, and associated analog circuitry in one device. The CS4222 family therefore reduces the number of separate converters required in a stereo audio system while simplifying synchronization and analog-output implementation. The verified manufacturer information identifies a 20-bit resolution, four-channel audio configuration, delta-sigma conversion architecture, and single +5 V operation. Its on-chip output buffer produces differential line-level outputs for the left and right channels. The documented nominal output is 2.83 Vpp, or 1 Vrms, for each single-ended output at full-scale digital input, with a 2.3 V common-mode voltage. The corresponding differential signal is 5.66 Vpp, or 2 Vrms). The CS4222-BS also incorporates volume control, allowing playback amplitude to be adjusted within the codec rather than solely through external analog gain stages. This integration can simplify system architecture, but the exact control interface, supported audio data formats, converter performance, clock requirements, and recommended external networks must be confirmed from the manufacturer datasheet before layout or firmware development. Those detailed values are not present in the supplied verified extracts and are therefore not inferred here. Typical implementation roles include legacy stereo multimedia systems, audio capture and playback modules, digital mixing equipment, and embedded audio subsystems requiring separate transmit and receive paths. In such equipment, the codec can accept digital audio at its ADC input and provide amplified analog left and right outputs for external coupling, headphone, or line-stage circuitry. Its differential outputs may offer better rejection of common-mode noise than single-ended outputs when paired with a balanced receiver. A critical design consideration is that the codec cannot be evaluated from resolution alone. Designers should verify the complete interface protocol, sample-rate range, clocking scheme, analog input range, output-load capability, power sequencing, and external component recommendations in the manufacturer datasheet. The documented 10 kΩ output loading must also be respected because heavier loads can alter output level and distortion even if the nominal operating voltage remains correct. This product page consolidates verified CS4222-BS specifications, distributor availability references, package information, and engineering guidance into one searchable source. It also separates documented data from unresolved parameters, helping engineers avoid unsupported assumptions while evaluating this audio codec for a new or legacy PCB design.
CS4222-DS - 20-Bit Stereo Audio Codec | Cirrus Logic
Cirrus Logic CS4222-DS is a 20-bit stereo audio codec with volume control, integrating analog-to-digital and digital-to-analog conversion in a 28-pin SSOP package. The device operates from a single +5 V supply and supports four-channel volume control according to the DigiKey product listing. Its delta-sigma conversion architecture, onboard output buffering, and differential line-level outputs make it suitable for multichannel consumer audio, sound-card, mixer, and digital-audio-interface equipment. An audio codec is an interface IC that converts analog sound into a digital bitstream and reconstructs digital audio into analog signals. In the broader semiconductor hierarchy, a codec belongs to interface ICs and audio codecs, combining mixed-signal converters with digital control and analog output circuitry. The CS4222 therefore provides both signal conversion and system-level volume management rather than operating only as a conventional digital interface controller. The CS4222 provides 20-bit stereo conversion and uses delta-sigma conversion techniques. Its on-chip buffer amplifier generates differential outputs capable of driving 10 kW loads, as stated in the CS4222 datasheet. Each output produces a nominal 2.83 Vpp, or 1 Vrms, signal for a full-scale digital input. The common-mode voltage is 2.3 V, and the two sides of each differential channel can represent a 5.66 Vpp, or 2 Vrms, differential signal. The single-supply architecture simplifies analog system design around a conventional +5 V rail. Digital control and analog conversion are consolidated in one IC, reducing the number of separate converters required in a stereo design. The integrated volume-control function can simplify control layout in systems that need programmable attenuation between digital sources and line-level outputs, although software control details and supported register values must be confirmed in the manufacturer datasheet before implementation. Typical uses include stereo sound cards, compact multimedia systems, powered speakers, home-audio preamplifiers, mixer inputs, USB or FireWire audio interfaces, and industrial monitoring equipment. The differential outputs are particularly useful when the downstream analog stage benefits from common-mode rejection and reduced susceptibility to ground-borne noise. Board-level filtering and grounding remain important because digital activity from the codec can couple into sensitive analog paths if the layout does not separate converter and output-current return paths. Designers should follow the recommended external component network shown in the Cirrus Logic CS4222 datasheet and preserve the differential output topology. Input and output decoupling should be placed close to the appropriate supply and signal pins, while analog and digital return currents should be routed carefully on mixed-signal boards. The 28-pin SSOP footprint is not interchangeable with leaded or differently pin-counted codecs without a PCB redesign. This product page consolidates verified distributor pricing, package information, converter performance, application guidance, and sourcing data. It also distinguishes verified CS4222-DS facts from parameters that still require confirmation, avoiding unsupported claims about equivalent codecs, exact pin functions, lifecycle guarantees, or unreported electrical limits.