ADC/DAC
Produk (3.100+)
PCM1748E - 24-Bit Stereo Audio DAC | Texas Instruments
The Texas Instruments PCM1748E is a CMOS, monolithic, two-channel digital-to-analog converter IC with support circuitry for stereo digital audio applications. The device uses a 24-bit enhanced multilevel delta-sigma DAC architecture with fourth-order noise shaping and 8-level amplitude quantization, providing a compact solution for digital-to-analog conversion in consumer, communications, and embedded audio systems. It is supplied in a 16-pin SSOP package according to the verified Texas Instruments product information. A stereo audio DAC is a specialized data-conversion component that accepts digital audio samples and produces corresponding analog left and right channel signals. In the broader semiconductor hierarchy, the PCM1748E belongs to the audio DAC class, within ADC/DAC and interface IC families. The enhanced multilevel delta-sigma architecture is intended to improve dynamic performance and clock tolerance compared with earlier converter implementations while keeping the external signal path manageable. The verified product information identifies the PCM1748 family as a 24-bit audio DAC with I2S support and a 16-SSOP package. Distributor listings also identify the PCM1748E as a stereo audio DAC and describe the device as a 24-bit converter operating with digital audio input. The available snippets mention sampling-rate values that are not fully consistent across sources, including 96 kHz, 100 kHz, and 192 kHz; therefore, the exact supported sampling-rate range should be confirmed against the manufacturer datasheet before design release. Typical signal-chain use places the PCM1748E after an audio source such as a digital interface, USB audio controller, or embedded audio processor. The digital stream is converted into left and right analog outputs for headphone, line-level, mixer, or amplifier circuitry. Because the converter is designed for stereo audio, it is useful in compact playback systems where two-channel conversion, small package size, and straightforward digital control are more important than a high channel count. Designers should verify the SSOP-16 land pattern, digital interface format, system clocking, analog output implementation, and decoupling requirements against the current Texas Instruments datasheet. Layout should keep digital clock and data routing away from analog output nodes, while supply and ground connections should follow the manufacturer’s recommended application circuit. The 16-pin SSOP package also requires careful solderability and assembly planning when lead pitch and thermal exposure are significant design constraints. This page consolidates verified manufacturer and distributor information, identifies the PCM1748E’s converter and packaging characteristics, and provides application-oriented guidance for engineers evaluating it as a stereo audio DAC. It also distinguishes documented part variants from packaging-only references and avoids treating unverified electrical limits as guaranteed specifications.
PCM1748KE - 24-Bit Stereo Audio DAC | Texas Instruments
The Texas Instruments PCM1748KE is a 24-bit stereo digital-to-analog converter with a 96 kS/s sampling rate, 106 dB SNR, and an SSOP-16 (DBQ) package. It converts two digital audio channels into balanced or single-ended analog outputs and supports 4.5 V to 5.5 V operation. An audio DAC is an interface IC that reconstructs analog voltage signals from digital audio data. Its position in the signal chain is between a digital audio source, such as a transmitter, decoder, or audio processor, and an analog amplifier, mixer, headphone driver, or line output stage. The PCM1748KE belongs to the broader DAC, ADC/DAC, and integrated-circuit hierarchy while specializing in high-resolution stereo audio playback. Key features include two 24-bit DAC channels, enhanced multilevel delta-sigma conversion, fourth-order noise shaping, and 8-level amplitude quantization. These features support the stated 106 dB signal-to-noise ratio and improve tolerance to clock jitter. The converter operates from 4.5 V to 5.5 V and is specified across a -25°C to +85°C temperature range. Internally, the PCM1748 uses an enhanced multilevel delta-sigma architecture designed for excellent dynamic performance. Fourth-order noise shaping moves quantization noise away from the most sensitive part of the audio band, while 8-level amplitude quantization reduces demands on the internal conversion system. Digital audio data is accepted through a serial interface identified in the product listing as I2S-compatible. Typical applications include digital-audio decoding, multichannel audio equipment, powered speakers, soundbars, compact home-audio systems, and industrial monitoring equipment. It is particularly useful when two analog outputs are required and a 24-bit conversion system is preferred over lower-resolution converters. Its 16-pin surface-mount package also supports compact audio boards. Designers should treat the PCM1748KE as a digital-to-analog output stage rather than a complete line amplifier. The analog outputs still require appropriate filtering, grounding, and signal-path design for the intended load. Use the manufacturer pin-description table when assigning PCB tracks, and verify interface timing and output-mode requirements against the manufacturer datasheet before releasing the layout. This page organizes the PCM1748KE technical details, package information, drop-in alternatives, application guidance, and design considerations into an engineering-focused selection reference. It also identifies where direct substitution requires careful comparison of interface, package, and operating requirements.
PCM1750U - Dual 18-Bit Audio ADC | Texas Instruments | SOIC-28
The Texas Instruments PCM1750U is a dual-channel, 18-bit monolithic audio analog-to-digital converter using successive-approximation conversion and a serial interface, housed in a 28-pin SOIC package. The converter is designed for applications requiring simultaneous digital capture of two audio channels while retaining an 18-bit resolution and a compact surface-mount footprint. An audio ADC is a data-converter IC that converts left- and right-channel analog audio signals into digital sample data for transmission, storage, processing, or playback. The PCM1750U belongs to the broader ADC/DAC category, with its application-specific role defined by audio signal acquisition. It is a dual CMOS device, allowing two analog audio inputs to be represented within one package rather than requiring two separate converter packages. The principal verified characteristics are 18-bit resolution, a dual-channel architecture, successive-approximation conversion, and a serial output format. These features make the PCM1750U suitable for digital audio recording, stereo capture, multimedia equipment, and other systems that require two synchronized audio conversion channels. The distributor classification identifies the device as an audio ADC in a 28-pin SOIC package. Successive-approximation conversion implements a binary-search sequence that progressively resolves the sampled input voltage. This architecture is appropriate for audio systems requiring deterministic serial data transfer and practical converter performance. Because no complete manufacturer datasheet was included in the verified dataset, the clock frequency, conversion rate, dynamic range, full-scale range, channel isolation, and analog input impedance are not independently stated here and must be confirmed in the manufacturer datasheet. Typical implementation places the PCM1750U after stereo input conditioning, with anti-alias filtering or other analog front-end stages ahead of its analog inputs. The serial interface then connects to a digital audio processor, audio codec, microcontroller, or logic device capable of receiving the converter's data, clock, and control signals. PCB layout should keep analog and digital return paths separated, minimize digital clock coupling into the analog inputs, and provide stable bypassing near the device. The principal design trade-off is that the PCM1750U integrates only the conversion function shown in the verified information; system-level audio performance also depends on the input amplifier, anti-alias filter, reference implementation, clock source, grounding, and software configuration. Before finalizing a design, verify timing, power-supply requirements, analog input range, and pin functions against the current Texas Instruments datasheet. This page consolidates verified distributor information, source traceability, replacement guidance, and engineering-focused selection notes. It also distinguishes confirmed PCM1750U characteristics from parameters that still require datasheet verification, providing a practical basis for sourcing and early design assessment without inventing unavailable electrical details.
PCM1760P - 20-Bit Dual-Input ADC | Texas Instruments
The Texas Instruments PCM1760P is a 20-bit, dual-input sigma-delta analog-to-digital converter with a 48 kHz sampling rate, housed in a 28-pin through-hole PDIP package. It combines a multi-bit enhanced-noise-shaping modulation stage with a decimating digital-filter function for high-resolution audio digitization. The manufacturer datasheet specifies typical total harmonic distortion plus noise of -92 dB and a maximum value of -90 dB, providing a concrete dynamic-performance benchmark for audio signal-chain design. An ADC converts continuous analog voltage signals into numerical data for processing, storage, or transmission. The PCM1760P belongs to the sigma-delta ADC family, whose architecture uses continuous-time modulation and digital filtering to shape quantization noise away from the signal band. In the broader component hierarchy, it is an ADC within the ADC/DAC category, which forms part of mixed-signal and data-conversion semiconductor systems used in professional audio, communications, instrumentation, and consumer equipment. The principal device features include two analog input channels, 20-bit conversion resolution, and operation at up to 48 kHz. Its multi-bit enhanced-noise-shaping architecture is intended to improve in-band conversion performance, while the associated decimating digital-filter section supports practical audio-rate output. The dual-channel arrangement allows simultaneous or multiplexed capture of stereo audio without requiring two separate converter packages. A 28-pin PDIP package also supports conventional through-hole PCB assembly and prototype rework. PCM1760P combines a multi-bit monolithic modulator with a monolithic decimation digital filter. In operation, each analog input is sampled and converted by the modulator before the digital filter removes high-frequency quantization noise and reduces the conversion rate. This division between analog modulation and digital filtering is central to the performance of the device. The resulting 20-bit, 48 kHz conversion capability is suited to systems that require greater nominal resolution than conventional low-cost audio converters while retaining a straightforward external interface architecture. Typical uses include stereo audio capture, professional recording equipment, broadcast monitoring, consumer audio equipment, and instrumentation requiring two-channel sampling. In each case, the ADC converts the left and right analog channels into digital samples while the internal decimation process controls out-of-band noise. System designers should also consider clock quality, analog input conditioning, grounding, decoupling, and digital-interface timing because these external factors can materially affect the converter's measured performance. For PCB implementation, place bypass capacitors close to the device supply pins and route analog and digital returns carefully to minimize shared-impedance coupling. The through-hole package simplifies inspection and manual rework, but it occupies considerably more board area than most surface-mount audio converters. Confirm clock requirements and interface signal levels against the manufacturer pin description and application guidance before finalizing the schematic. This product page combines the manufacturer-level architecture information, exact package and conversion specifications, application guidance, and drop-in-replacement evaluation for PCM1760P sourcing and engineering review. It is intended to help buyers identify the correct 28-pin audio ADC while providing practical context for signal-chain design and board implementation.
PCM1760U - 20-Bit Dual Audio ADC | Texas Instruments
Texas Instruments PCM1760U is a 20-bit, dual-channel, multibit enhanced noise-shaping analog-to-digital converter system for professional audio acquisition. It converts two analog inputs at up to 48 kHz and is supplied in a 28-pin SOIC package. The converter combines a dual 20-bit monolithic modulator with a monolithic decimation digital filter, providing a compact signal-chain solution for high-precision audio systems. An audio ADC is a data-acquisition IC that samples incoming analog waveforms and converts their voltage amplitudes into high-resolution digital codes. The PCM1760U belongs to the broader ADC category, the ADC/DAC interface hierarchy, and ultimately the semiconductor and signal-conversion market. It is especially relevant when a design needs simultaneous conversion of two analog audio channels with high dynamic performance and manageable digital-filter implementation. Key characteristics include 20-bit resolution, two analog inputs, sigma-delta conversion, and a maximum sampling rate of 48 kHz. The manufacturer describes typical THD+N of -92 dB and a maximum THD+N of -90 dB. Its enhanced noise-shaping architecture reduces quantization noise within the audio band, while the integrated decimation filter prepares the converted stream for downstream digital processing. The PCM1760U integrates both the analog-to-digital modulator and the decimation filtering stage rather than requiring a separate external decimation filter for a conventional implementation. That integration can reduce board complexity and simplify clocking, data routing, and software configuration. The converter architecture remains specifically optimized for audio-rate signals, making it different from general-purpose high-speed ADCs intended for communications, instrumentation, or industrial waveform capture. Typical applications include professional digital audio equipment, multichannel recording interfaces, studio monitor controllers, digital mixing consoles, broadcast consoles, and consumer audio products requiring two-channel conversion. It is also suitable for USB or networked audio subsystems when the downstream interface accepts the PCM1760U's audio-rate serial data and clocking arrangement. Design the clock and digital interface carefully because converter performance depends on clean references, appropriate analog input conditioning, and a stable master clock. Use the manufacturer datasheet's recommended input and output-capacitor configuration, keep analog traces short, and provide return paths that avoid digital switching currents. Confirm interface voltage levels and data-format requirements against the exact system clock architecture. This product summary combines the converter's stated resolution, channel count, sampling rate, distortion performance, packaging, architecture, and application guidance with practical design notes. It also distinguishes the U package from package or ordering variants and provides verified drop-in alternatives with explicit engineering differences.
PCM1772PW - 24-Bit Stereo Audio DAC | Texas Instruments
The Texas Instruments PCM1772PW is a 24-bit, low-power stereo audio digital-to-analog converter with integrated line-out circuitry and software control, housed in a 16-pin TSSOP package. It supports I2S audio data at sample rates up to 50 kSPS and is designed to provide 98 dB signal-to-noise ratio for compact audio systems. The device accepts digital stereo audio, converts it into analog left and right channels, and drives the line outputs through its integrated amplifier circuitry. An audio DAC is a signal-conversion IC that transforms digital audio samples into continuous analog voltages suitable for amplification, headphone stages, mixer inputs, or line-level equipment. It belongs to the digital-to-analog converter category within broader data-conversion and signal-processing integrated circuits. Its digital interface connects to an audio codec or processor, while the line outputs connect to the next analog stage in the signal chain. The PCM1772PW combines a 24-bit conversion resolution, a maximum 50 kSPS sampling rate, and 98 dB SNR in one low-power CMOS device. Software-controlled operation allows the host processor to manage supported device functions without a dedicated control microcontroller. Its stereo line-out architecture reduces the need for external output amplifiers in conventional line-level audio designs, although output loading, coupling, and headroom still require careful evaluation. Texas Instruments describes the converter architecture as an enhanced multilevel design using noise shaping and multilevel amplitude quantization. These techniques are intended to improve dynamic performance and clock-jitter tolerance compared with a basic binary conversion approach. The PCM1772 and related PCM family devices are monolithic CMOS integrated circuits combining stereo DACs, line-out circuitry, and support functions in compact packages. Typical uses include digital active speakers, compact audio appliances, consumer audio equipment, computer peripherals, and embedded audio systems that receive standard I2S streams. In these applications, the PCM1772PW can establish an analog line output close to the digital audio source, reducing system size while retaining a conventional stereo line-output interface. Its 24-bit resolution is also useful in designs where the host digital audio source provides higher-resolution samples. The most important design check is the relationship between the digital audio source, clocking method, supported audio format, analog output loading, and power-supply arrangement. Use the pin-description table and recommended application circuit in the manufacturer datasheet when assigning tracks, selecting output components, and validating signal levels. Confirm whether the required I2S master or slave relationship and clock configuration match the surrounding codec or processor. This product page consolidates the confirmed PCM1772PW capabilities, package information, sourcing workflow, and engineering guidance. It also distinguishes confirmed same-family alternatives from unsupported cross-package substitutions, helping engineers identify candidates while still requiring a final pin, timing, interface, and acoustic review before PCB release.
PCM1792DB - 24-Bit 192kHz Stereo DAC | Texas Instruments
Texas Instruments PCM1792DB is a 24-bit stereo digital-to-analog converter supporting 192 kHz PCM sampling and 200 kHz DSD operation, housed in a 28-lead SSOP package. It uses advanced segment DAC architecture, balanced differential current outputs, and a serial control interface for programmable audio operation. The converter is designed for high-fidelity audio systems requiring low distortion and high dynamic range. An audio DAC converts digital PCM or DSD data into analog voltage or current signals suitable for amplification and loudspeaker drive. Within the signal chain, it sits between a digital audio source and an analog output stage. The PCM1792DB combines two converter channels and associated support circuitry in one monolithic CMOS device, simplifying stereo audio designs while preserving independent balanced outputs. The PCM1792DB provides a dynamic range of 132 dB for mono operation at 9 Vrms and 129 dB for stereo operation at 4.5 Vrms. Its THD+N is 0.0004%, and its differential current output reaches 7.8 mA peak-to-peak. These characteristics make the converter appropriate for premium digital audio equipment, professional monitoring, USB and network audio products, and other systems where low-level analog distortion can affect sound quality. Advanced segment DAC architecture is the core technical feature of this converter. According to the manufacturer datasheet, the architecture improves dynamic performance and tolerance to clock jitter. The device uses an 8-times oversampling digital filter, balanced current outputs that allow output topology optimization, and a serial control interface for mode configuration. These functions support integration into designs requiring both PCM playback and Direct Stream Digital operation. Typical applications include high-resolution stereo DAC equipment, professional audio monitors, digital mixing consoles, USB audio interfaces, and home-theater signal processing. The differential outputs are particularly useful when the downstream amplifier or current-voltage conversion stage can preserve the benefits of balanced operation. System designers must account for the required analog supplies, output load, external conversion network, and clocking arrangement when laying out the PCB. A critical implementation consideration is maintaining clean, low-noise analog and digital supply rails. Place the bypass capacitors specified by the manufacturer close to their corresponding power pins, route high-current outputs away from sensitive digital and analog inputs, and use a symmetric ground strategy around the converter outputs. The 28-lead SSOP footprint also requires careful soldering practices because its lead pitch is relatively narrow. This page consolidates the PCM1792DB electrical characteristics, package details, application guidance, and replacement considerations into an engineering-focused reference. It also distinguishes true drop-in interchangeability from related converters and identifies the design variables that most often determine success in a high-performance audio DAC implementation.
PCM1793DB - 24-Bit 192kHz Stereo DAC | Texas Instruments
Texas Instruments PCM1793DB is a 24-bit, 192 kHz sampling stereo digital-to-analog converter with balanced voltage outputs and hardware control, packaged in a 28-lead SSOP. The monolithic CMOS device combines two converter channels and support circuitry, while its advanced segment DAC architecture provides excellent dynamic performance and improved tolerance to clock jitter. Typical signal-to-noise ratio is 113 dB, typical total harmonic distortion plus noise is 0.001%, and typical full-scale output is 2.1 V rms at the postamplifier. A digital-to-analog converter, or DAC, converts digital audio samples into an analog voltage waveform. The PCM1793 belongs to the broader hierarchy of stereo DAC, audio converter, signal-conversion IC, and semiconductor. Balanced voltage outputs support noise-rejecting analog signal paths, allowing a designer to optimize the output stage for the target amplifier and load. Its I2S-compatible digital interface accepts audio data from a digital audio source, while the DAC performs the final conversion before the analog output stage. The principal features are 24-bit resolution, a maximum sampling rate of 192 kHz, balanced outputs, 113 dB dynamic range, and a typical THD+N of 0.001%. Hardware control provides an external interface for adjusting supported operating functions without relying entirely on software commands. The advanced segment DAC architecture is specifically designed to improve dynamic performance and reduce sensitivity to timing variations in the incoming digital clock. The converter uses a monolithic CMOS implementation containing stereo digital-to-analog converters and associated support circuitry. Balanced voltage outputs distinguish the PCM1793 from converters that provide differential current outputs. At the postamplifier, a typical full-scale output of 2.1 V rms provides a defined analog reference level for subsequent voltage amplification or line-level routing. Its operating supply range is 4.5 V to 5.5 V. Typical applications include high-resolution stereo audio equipment, CD and DVD player digital audio paths, professional audio converters, powered speakers, headphone-amplifier front ends, and digital audio system boards. The balanced output topology is useful when the downstream circuitry supports balanced inputs, while a carefully designed single-ended summing network can serve conventional line outputs. Designers should treat the power supply, digital clock, ground layout, and output network as a complete signal-chain design. Place decoupling and output-network components close to their associated pins, keep analog return paths separate from switching and digital return currents, and provide filtering appropriate to the audio bandwidth. The SSOP package also requires attention to solder-joint quality and thermal conditions during assembly. This product overview combines converter architecture, key electrical characteristics, application guidance, package information, and sourcing-oriented details. It is intended to help engineers determine whether the PCM1793DB's balanced voltage output, dynamic performance, and interface characteristics fit a stereo DAC design before beginning a detailed schematic and layout review.
PCM1794DB - 24-Bit 192kHz Stereo DAC | Texas Instruments
The Texas Instruments PCM1794DB is a 24-bit stereo digital-to-analog converter supporting sampling rates up to 192 kHz. It uses advanced segment DAC architecture, includes support circuitry, and is packaged in a 28-lead SSOP. The converter provides balanced current outputs, allowing the analog output stage to be optimized for either differential or single-ended signal-chain architectures. A digital-to-analog converter, or DAC, converts digital audio samples into an analog voltage or current representation. The PCM1794DB belongs to the stereo audio DAC category within the broader ADC/DAC and interface-converter hierarchy. It is designed for high-resolution audio equipment, where digital source quality must be translated into a low-distortion analog output for headphone, line, or preamplifier stages. Key features include 24-bit data resolution, 192 kHz maximum sampling capability, advanced segment DAC architecture, balanced current outputs, and support for common digital audio interfaces including I2S, PCM, and left-justified formats. The manufacturer describes the converter as a monolithic CMOS integrated circuit with improved tolerance to clock jitter. Its balanced outputs support flexible analog signal routing while preserving channel symmetry. The PCM1794DB is normally used with external current-to-voltage conversion, filtering, and output-buffer circuitry. The balanced output architecture can reduce even-order distortion and common-mode effects when paired with a suitably matched analog stage. The converter also includes digital control functions for reset and operating-mode configuration, allowing it to be coordinated with an audio system clock and source device. Typical applications include high-resolution stereo DAC modules, USB and network audio converters, CD-quality playback equipment, digital mixing consoles, headphone amplifiers, and balanced line-output stages. The 192 kHz sampling capability makes it appropriate for studio monitoring and archival audio systems, although system clocking and analog implementation remain important to achieving the intended performance. When laying out a PCM1794DB circuit, keep digital clock and data traces away from the sensitive analog output network, use short and properly terminated clock routes, and place filtering and I/V components close to their associated pins. Verify power sequencing, reset timing, and the required 1024 system-clock initialization period before enabling normal audio operation. This product page consolidates the manufacturer device information, distributor identification, package details, and practical implementation guidance for engineers evaluating the PCM1794DB in a new audio-converter design.
PCM1796DB - 24-Bit 192 kHz Stereo DAC | Texas Instruments
Texas Instruments PCM1796DB is a 24-bit stereo audio digital-to-analog converter with a sampling rate of 192 kHz, 123 dB SNR, balanced current outputs, and a 28-lead SSOP package. It uses TI's advanced segment DAC architecture to provide strong dynamic performance and improved tolerance to clock jitter for high-fidelity audio systems. An audio DAC is a data-conversion IC that transforms digital PCM samples into analog current signals suitable for a downstream operational amplifier, reconstruction filter, or speaker system. Within the semiconductor hierarchy, a DAC belongs to the ADC/DAC class of data-converter ICs. The PCM1796DB is designed for stereo audio conversion where signal fidelity, channel balance, and compatibility with digital audio interfaces are primary design considerations. The converter supports 24-bit resolution and operation up to 192 kHz, while the listed DSD capability reaches 200 kHz. Its 123 dB signal-to-noise ratio supports low-noise conversion in demanding listening equipment. Balanced current outputs provide a flexible analog architecture, allowing the designer to optimize the external amplifier and filtering stages. Software control supports configurable system behavior when the DAC is managed by an audio controller or digital audio interface. Internally, the PCM1796DB combines stereo digital-to-analog converters and support circuitry in one monolithic CMOS device. The segment DAC architecture separates the conversion function into smaller segments, improving dynamic behavior and reducing sensitivity to timing errors in the digital audio stream. This architecture is particularly useful in designs where incoming clock jitter could otherwise affect the analog output. Typical applications include high-resolution CD and DVD audio players, network audio receivers, digital music systems, professional audio equipment, home-theater processors, and multichannel audio platforms. The balanced current-output topology can support transformer-balanced or line-level single-ended outputs when paired with suitable external circuitry. The converter is especially appropriate when a design requires 24-bit resolution and high sampling rates without using an external discrete DAC architecture. The principal design consideration is the analog output stage. Because the device produces current rather than a finished voltage, component selection, current-to-voltage conversion, ground treatment, and reconstruction filtering directly influence sound quality. Place the DAC, its digital interface components, and the analog output circuitry according to the pin descriptions and application information in the manufacturer datasheet. Maintain the stated supply ranges of 3 V to 3.6 V and 4.75 V to 5.25 V. This product page combines manufacturer specifications, application guidance, package information, and compatibility considerations for engineers evaluating the PCM1796DB. It also highlights the significance of its segment DAC architecture, 123 dB SNR, 192 kHz sampling capability, and 28-pin SSOP implementation when planning a high-fidelity audio converter stage.
PCM1800E - 20-Bit 48 kSPS Stereo ADC | Texas Instruments
The Texas Instruments PCM1800E is a 20-bit stereo analog-to-digital converter (ADC) with single-ended analog inputs, a maximum sampling rate of 48 kSPS, a 4.5 V to 5.5 V supply range, and a 24-pin SSOP package. It converts two analog input channels into digital data and is intended for cost-sensitive consumer audio equipment. According to the Texas Instruments PCM1800 product page, the converter uses a delta-sigma modulator, 64-times oversampling, a digital decimation filter, and a serial interface supporting master and slave modes. An ADC is a mixed-signal semiconductor that converts continuously varying analog voltages into numerical values suitable for digital processing. The PCM1800 belongs to the delta-sigma ADC family and the broader data-converter and signal-processing IC hierarchy. Its single-ended inputs simplify source connections when the signal source can be referenced adequately against the converter's analog ground. Digital audio equipment commonly uses this architecture to support CD-compatible 44.1 kSPS and 48 kSPS playback or recording rates. Key features of the PCM1800E include dual 20-bit monolithic conversion, single-ended voltage inputs, 64-times oversampling, and four supported data formats. The decimation filter has a stated passband ripple of ±0.05 dB and stopband attenuation of -65 dB. The reported signal-to-noise ratio is 95 dB, making the device relevant to audio preamplifiers, consumer receivers, digital mixers, and other systems requiring stereo audio capture at 48 kSPS. The delta-sigma architecture converts the analog signal through oversampling and digital filtering before producing the final serial output. PCM1800 documentation identifies both master and slave operating modes, allowing the converter either to generate the serial clock or synchronize it with an external audio processor. Four data formats provide design flexibility when connecting the ADC to codecs, digital audio processors, microcontrollers, or programmable logic. The serial interface and decimation filter therefore form the boundary between the analog audio inputs and the digital signal path. Typical PCM1800E applications include stereo consumer audio capture, digital recording equipment, audio mixer inputs, consumer media systems, and digital signal processing front ends. In a typical signal chain, a stereo source is buffered, level-shifted into the ADC input range, sampled by the delta-sigma modulator, decimated, and transferred to a digital audio processor through the serial interface. The 48 kSPS rate supports professional audio-standard sampling rates but does not itself provide the analog front-end gain, anti-alias filtering, clock generation, or output audio processing required by a complete system. A key design consideration is preserving signal integrity at the single-ended analog inputs and the digital serial interface. Input routing should minimize noise pickup, keep the analog and digital return paths controlled, and account for the converter's input voltage range. Clock and data traces require suitable timing margins, while bypass capacitors should be placed close to the supply pins. The SSOP-24 package is suitable for compact designs, but its lead pitch and exposed analog, digital, and supply connections demand careful layout and assembly. This page consolidates the verified PCM1800 family specifications, source links, application guidance, and replacement-status analysis into one engineering-focused reference. It also separates published facts from unavailable details so designers can identify what must still be confirmed in the manufacturer datasheet before schematic release, PCB layout, or production qualification.
PCM1801U - 16-Bit Stereo Audio ADC | Texas Instruments
The Texas Instruments PCM1801U is a 16-bit, stereo, single-ended audio analog-to-digital converter with a 48 kSPS conversion rate, 64-times oversampling, and a serial interface, housed in a 14-pin SOIC package. It is designed for cost-sensitive consumer audio equipment where a compact stereo converter with typical 93 dB SNR and -88 dB THD+N performance is required. An audio ADC is a mixed-signal semiconductor that converts left- and right-channel analog voltages into digital sample data for processing, storage, or transmission. The PCM1801 belongs to the ADC and DAC hierarchy, uses a delta-sigma modulator and digital decimation filter, and supports slave-mode operation with two serial data formats. This architecture provides a practical bridge between analog audio sources and digital audio systems. Key features include dual 16-bit monolithic conversion channels, single-ended analog voltage inputs, 64-times oversampling, a 48 kSPS maximum sampling rate, and a serial output interface. The decimation filter has a specified passband ripple of ±0.05 dB and stopband attenuation of -65 dB. Typical signal-chain performance includes THD+N of -88 dB, SNR of 93 dB, and dynamic range of 93 dB. Internally, the PCM1801 uses a delta-sigma modulator followed by a digital decimation filter to reduce aliasing and shape quantization noise. Its serial interface supports slave operation and two data formats, giving the system designer flexibility in clocking and data framing. The single-ended inputs simplify source interfacing, although input filtering, grounding, and source impedance remain critical to achieving the stated audio performance. Typical uses include consumer digital audio equipment, stereo recording products, USB audio interfaces, set-top boxes, and other compact audio systems requiring two analog inputs and a serial digital interface. The 48 kSPS rate aligns with standard CD-rate audio, while the 14-pin SOIC package simplifies assembly in cost-sensitive designs. When laying out the circuit, place input and decoupling capacitors close to their associated pins, maintain a low-impedance ground return for the analog section, and route serial clock and data traces away from sensitive analog inputs. Verify the external interface voltage levels and serial timing before connecting the converter to a host processor. This product page consolidates verified manufacturer and distributor information, price context, package details, application guidance, and substitution limitations. It distinguishes documented specifications from missing data and avoids presenting electrically different converters as guaranteed drop-in replacements.
PCM1802DB - 24-Bit 96 kHz Stereo Audio ADC | Texas Instruments
Texas Instruments PCM1802DB is a high-performance stereo audio analog-to-digital converter with single-ended analog inputs, 24-bit resolution, and a maximum sampling rate of 96 kHz. It is packaged in a 20-SSOP package and is designed for digital audio capture systems requiring separate analog and digital supply rails. An audio ADC converts left and right analog audio signals into digital sample data for transmission, recording, processing, or playback. The PCM1802DB uses a delta-sigma modulator with 64-times or 128-times oversampling, followed by a digital decimation filter and a high-pass filter that removes the DC component from the input signal. This architecture is part of the broader analog-to-digital converter, data-converter, signal-processing, and semiconductor hierarchy. The device provides two single-ended analog input channels, a 24-bit conversion resolution, and a serial digital interface. Its analog supply operating range is 2.7 V to 3.6 V, while the digital supply range begins at 4.5 V according to the manufacturer product information. The 96 kHz sampling rate supports professional and consumer audio acquisition, including line-level stereo capture and digital audio recording. The PCM1802DB is intended for audio interfaces, USB audio bridges, digital mixers, sound-card designs, broadcast equipment, and other systems that convert analog stereo audio into digital data. Its single-ended inputs simplify source connections compared with differential-input converters, although circuit design must still address analog signal conditioning, filtering, grounding, and digital-interface compatibility. The 20-SSOP package requires careful PCB layout around the analog and digital sections. The manufacturer datasheet should be used as the authority for decoupling values, clock requirements, supported master and slave operating modes, pin assignments, and digital timing. This page combines the manufacturer product information with practical application guidance, engineering comparisons, and sourcing-oriented details.
PCM1803DB - 24-Bit 96-kHz Stereo ADC | Texas Instruments
The Texas Instruments PCM1803DB is a 24-bit, 96-kHz stereo delta-sigma analog-to-digital converter with single-ended analog voltage inputs, housed in a 20-pin SSOP package. It accepts 3 Vp-p input signals, uses 64-times or 128-times oversampling, and supports both master and slave operating modes. The converter is designed for cost-sensitive consumer audio systems operating from a 5 V analog supply and a 3.3 V digital supply. An audio ADC is an integrated circuit that converts left- and right-channel analog audio voltages into digital data for processing, storage, transmission, or playback. PCM1803DB belongs to the ADC/DAC class of data-conversion ICs and uses a delta-sigma modulator followed by a digital decimation filter. An integrated high-pass filter removes the DC component of the input signal, while master and slave mode support allows the converter to operate within different system clocking arrangements. The principal features are 24-bit resolution, a maximum sampling rate of 96 kHz, stereo single-ended input, and selectable 64-times or 128-times oversampling. Its serial interface supports audio-system integration, and the single-ended input architecture is useful when the source has a relatively simple single-ended output stage. The converter is fabricated using a CMOS process and is offered in a compact 20-pin SSOP package suited to cost-sensitive consumer audio equipment. PCM1803DB is positioned between basic codec functions and more complex audio-conversion solutions. The delta-sigma architecture moves much of the signal-shaping and filtering work into the digital domain, while the decimation and high-pass filters reduce unwanted DC content and aliasing components before the final digital data is presented. System designers still need to provide correct analog and digital supplies, suitable input conditioning, a supported serial clock configuration, and decoupling close to the device pins. Typical uses include consumer stereo audio equipment, digital recording systems, multimedia appliances, and compact audio interfaces requiring two single-ended analog channels. Its 96-kHz conversion rate supports high-resolution digital-audio workflows, while the 5 V analog and 3.3 V digital supply arrangement can simplify mixed analog and digital system design. Its 24-bit output resolution also provides more numerical headroom than lower-resolution converters when downstream gain staging is carefully controlled. For a robust implementation, treat the analog input network as a precision signal path and verify the clock, format, and master or slave configuration against the manufacturer datasheet. Place bypass capacitors close to the supply pins, keep digital return currents away from sensitive analog input routing, and confirm that the external interface device supports the converter's serial output protocol and timing. This page consolidates the PCM1803DB's key conversion specifications, sourcing information, application guidance, and same-family drop-in considerations. It also highlights practical checks involving supply domains, clocking, filtering, PCB layout, and input conditioning that engineers should complete before releasing a new audio design.
PCM1804DB - 24-Bit 192kHz Stereo ADC | Texas Instruments
The Texas Instruments PCM1804DB is a 24-bit, 192 kHz stereo audio analog-to-digital converter with fully differential analog inputs and an I2S-compatible digital interface, housed in a 28-pin SSOP package. Typical performance includes 112 dB dynamic range, 111 dB signal-to-noise ratio, and -102 dB total harmonic distortion plus noise. These headline specifications place it firmly in the professional and high-grade consumer audio conversion class. A delta-sigma audio ADC is a semiconductor that converts left- and right-channel analog voltages into high-resolution digital samples. It belongs to the ADC and DAC hierarchy, where an ADC performs the analog-to-digital conversion stage needed by digital audio equipment. The PCM1804 combines a precision delta-sigma modulator, a linear-phase antialias digital filter, and a high-pass filter that removes DC offset from the input signal. Key features include support for 24-bit resolution, a maximum sampling rate of 192 kHz, fully differential stereo inputs, and PCM audio operation. The converter can also produce DSD-format data because its delta-sigma modulator operates from the same signal chain. The antialias filter limits out-of-band energy before the digitized signal reaches the audio processor, while the high-pass filter helps prevent DC offset from consuming digital headroom. The PCM1804 uses an advanced CMOS process and integrates both converter channels in one package. Its 28-pin SSOP arrangement supports analog input, digital data, clock, control, and power connections. Linear-phase digital filtering is particularly useful in music-production equipment because it provides predictable passband behavior and helps reduce aliasing without introducing a steep magnitude distortion across the audio band. Typical applications include professional audio interfaces, digital mixing consoles, multichannel audio recorders, studio monitors, broadcast equipment, and high-end consumer audio. Fully differential inputs are useful when the source can drive balanced analog signals, while the 192 kHz capability supports high-resolution recording workflows. The digital output can feed a suitable audio processor, DAC, or transport through the appropriate clock and data interface. PCB design should keep the analog input and digital return paths separated, provide clean reference and supply connections, and follow the manufacturer’s recommended decoupling and layout arrangement. Because audio converters can resolve small signal details, thermal design and grounding are as important as nominal resolution. Confirm clock polarity, master-versus-slave operation, and format settings in the datasheet before finalizing the digital interface. This product record combines the Texas Instruments device specifications, application guidance, verified package information, and drop-in comparison details in one engineering-focused reference. It also highlights practical design considerations for clocking, filtering, signal integrity, and PCB implementation that are not apparent from the MPN alone.
PCM1807PW - 24-Bit, 96kHz Stereo ADC | Texas Instruments
The Texas Instruments PCM1807PW is a 24-bit, 96kHz stereo delta-sigma analog-to-digital converter with single-ended analog inputs, 64-times oversampling, and a 14-pin TSSOP package. It accepts two analog channels and supports both master and slave operating modes for flexible audio-clock synchronization. An audio ADC is a signal-conversion IC that converts continuous analog voltage waveforms into digital audio samples. The PCM1807 sits within the ADC-DAC interface category, combining a delta-sigma modulator, digital decimation filter, and DC-blocking high-pass filter. This integrated filtering allows a stereo analog source to connect directly to a digital audio system while limiting out-of-band noise and removing DC offset from the input signal. The principal specifications are 24-bit resolution, a maximum sampling rate of 96kHz, and two single-ended analog inputs. The analog input range is 3 Vp-p, while the converter uses 64-times oversampling and includes a digital decimation filter. These features make the device appropriate for consumer audio equipment, digital recording systems, sound-card designs, and embedded audio products requiring stereo PCM output. The PCM1807 uses a delta-sigma modulation architecture to encode each analog input as a high-frequency bit stream. A digital decimation filter then reduces that stream to the required audio sample rate, while the high-pass filter removes the DC component of the input. Master and slave modes give the system designer control over whether the converter uses an externally supplied master clock and serial clock or operates according to its own clock relationships. Typical applications include stereo audio capture, USB or microcontroller-based sound-card front ends, consumer playback equipment, and measurement products that accept line-level single-ended signals. The 96kHz sampling capability supports high-resolution digital audio workflows, while the 24-bit conversion format preserves fine amplitude detail for subsequent mixing, recording, or signal processing. PCB implementation should preserve short, separated analog and digital return paths and place input filtering components close to the corresponding pins. The analog input network must also be designed for the converter's single-ended 3 Vp-p input range. Designers should use the manufacturer datasheet pin-description and typical-application diagrams when assigning tracks, grounding, and power-supply connections. This page organizes the converter's verified electrical characteristics, package details, commercial information, application guidance, and verified replacement information into an engineering-focused resource. It also identifies the documented same-footprint option while avoiding unsupported cross-brand substitutions.
PCM1808PW - 24-Bit 96-kHz Stereo Audio ADC | TI (Tape & Reel)
The Texas Instruments PCM1808PW is a 24-bit, 96-kHz stereo audio ADC with single-ended analog inputs, 99-dB SNR, and 64-times delta-sigma oversampling, housed in a 14-pin TSSOP package. It converts two analog audio channels into digital data while integrating a decimation filter and a high-pass filter that removes input DC offset. An audio ADC is a data-acquisition IC that transforms continuous analog voltage signals into digital codes for storage, processing, transmission, or playback. The PCM1808 belongs to the broader delta-sigma ADC family, within ADCs, data converters, and signal-chain semiconductors. Its stereo architecture supports two single-ended inputs rather than balanced differential channels, making it suitable for compact consumer and embedded audio equipment. Key verified features include 24-bit resolution, a maximum sampling rate of 96 kHz, 99-dB signal-to-noise ratio, 64-times oversampling, and an integrated digital decimation filter. The manufacturer describes support for both master and slave operating modes. These features allow the converter to capture audio-band signals while reducing analog front-end requirements and limiting out-of-band energy before digital processing. The PCM1808 uses a delta-sigma modulator architecture, with the decimation filter and high-pass filter implemented on-chip. The 99-dB SNR specification indicates a favorable noise floor for high-resolution audio capture, while the 24-bit word length preserves fine amplitude detail. The supplied distributor record identifies the device as a serial, single-ended-input, stereo audio ADC in a 14-TSSOP package. Typical applications include digital audio recording, multimedia input stages, USB or embedded audio interfaces, consumer audio equipment, and systems requiring stereo analog-to-digital conversion. In a signal chain, an analog preamplifier drives the PCM1808 inputs, and the resulting digital stream is routed to a processor, codec interface, or storage subsystem. The converter is especially appropriate where two single-ended channels and a compact surface-mount package are preferred. Designers should verify the complete electrical interface, pin names, clocking requirements, and supported operating conditions against the official PCM1808 datasheet before layout. The TSSOP footprint requires careful analog trace routing and decoupling. Because the verified source provides no exact pinout table or detailed application-specific companion parts, those values are intentionally not invented here. This page consolidates verified manufacturer and distributor information while clearly separating unavailable parameters.
PCM1808PWR - 24-Bit 96-kHz Stereo Audio ADC | TI
The Texas Instruments PCM1808PWR is a 24-bit, 96-kHz stereo audio ADC with single-ended analog inputs, 99-dB SNR, and 64-times delta-sigma oversampling, housed in a 14-pin TSSOP package. It converts two analog audio channels into digital data while integrating a decimation filter and a high-pass filter that removes input DC offset. An audio ADC is a data-acquisition IC that transforms continuous analog voltage signals into digital codes for storage, processing, transmission, or playback. The PCM1808 belongs to the broader delta-sigma ADC family, within ADCs, data converters, and signal-chain semiconductors. Its stereo architecture supports two single-ended inputs rather than balanced differential channels, making it suitable for compact consumer and embedded audio equipment. Key verified features include 24-bit resolution, a maximum sampling rate of 96 kHz, 99-dB signal-to-noise ratio, 64-times oversampling, and an integrated digital decimation filter. The manufacturer describes support for both master and slave operating modes. These features allow the converter to capture audio-band signals while reducing analog front-end requirements and limiting out-of-band energy before digital processing. The PCM1808 uses a delta-sigma modulator architecture, with the decimation filter and high-pass filter implemented on-chip. The 99-dB SNR specification indicates a favorable noise floor for high-resolution audio capture, while the 24-bit word length preserves fine amplitude detail. The supplied distributor record identifies the device as a serial, single-ended-input, stereo audio ADC in a 14-TSSOP package. Typical applications include digital audio recording, multimedia input stages, USB or embedded audio interfaces, consumer audio equipment, and systems requiring stereo analog-to-digital conversion. In a signal chain, an analog preamplifier drives the PCM1808 inputs, and the resulting digital stream is routed to a processor, codec interface, or storage subsystem. The converter is especially appropriate where two single-ended channels and a compact surface-mount package are preferred. Designers should verify the complete electrical interface, pin names, clocking requirements, and supported operating conditions against the official PCM1808 datasheet before layout. The TSSOP footprint requires careful analog trace routing and decoupling. Because the verified source provides no exact pinout table or detailed application-specific companion parts, those values are intentionally not invented here. This page consolidates verified manufacturer and distributor information while clearly separating unavailable parameters.
PCM3501E - 16-Bit Mono Voice/Modem Codec | Texas Instruments
The Texas Instruments PCM3501E is a low-voltage, low-power, 16-bit mono SoundPlus voice and modem codec in a 24-pin SSOP (DB) package. The device is designed for voice-band signal conversion, with support for operation rates up to 56 kbps and sampling frequencies up to 26 kHz. Its analog performance includes a typical dynamic range of 88 dB when measured at 8 kHz sampling with a 1 kHz input frequency. The PCM3501E is a single-channel codec intended for embedded voice, modem, telephony, communications, and audio acquisition systems. A voice codec is a mixed-signal semiconductor that converts between analog audio or voice signals and a digital serial interface. The PCM3501E belongs to the broader codec and audio interface hierarchy, which includes analog-to-digital converters, digital-to-analog converters, voice processors, and communication interface ICs. Its 16-bit resolution supports voice-band digital representation, while the mono architecture reduces system complexity compared with stereo codecs. Differential input and output capability is specifically identified for modem analog front-end designs. The principal device characteristics are its 16-bit delta-sigma ADC and DAC sections, differential input and output support, operation to 56 kbps, and maximum sampling frequency of 26 kHz. The 88 dB typical dynamic range provides the stated signal-performance target for voice-band applications. The low-voltage and low-power positioning makes the codec suitable for compact embedded equipment where the system must convert voice signals without introducing unnecessary circuit complexity. The PCM3501E is appropriate for modem analog front ends, telephone interfaces, voice recording systems, industrial communication terminals, and other mono voice-processing equipment. Its serial digital interface allows the analog front end to connect to a host processor or communications controller. Designers should select the digital interface mode and clocking arrangement according to the manufacturer datasheet, and should confirm the system sampling rate and serial framing before committing the PCB layout. A key design consideration is the distinction between the codec signal path and the surrounding analog implementation. Input and output filtering, microphone or line-source conditioning, clocking, digital interface routing, and supply decoupling all affect final voice quality. Use the manufacturer application circuit and pin-description table when assigning tracks, grounding the analog return path, and placing bypass capacitors. This product page consolidates the manufacturer device identity, package information, verified technical characteristics, practical application guidance, and design considerations. It also identifies the limitation that the cross-reference results do not establish five distinct pin-compatible drop-in replacements, avoiding unsupported substitution claims.
PCM4202IDBREP - 24-Bit 216kHz Stereo Audio ADC, 28-SSOP | Texas Instruments
The Texas Instruments PCM4202IDBREP is a high-performance, 24-bit, 216 kHz sampling, stereo audio analog-to-digital converter (ADC) housed in a 28-pin SSOP (DB) package. As an Enhanced Product (EP) variant, it is rated for industrial temperature operation from -40C to +85C, and the device supports 24-bit linear PCM output data at sampling frequencies up to 216 kHz through a standard I2S, left-justified, or right-justified interface. The PCM4202IDBREP delivers dynamic performance of 118 dB SNR and 105 dB THD+N (typical), making it suitable for professional and broadcast audio recording. A stereo audio ADC is a high-resolution converter that digitizes two channels of analog audio into a bitstream suitable for storage, mixing, or further DSP processing. In the broader signal chain hierarchy, the PCM4202IDBREP sits between microphone preamplifiers / line inputs and downstream digital processing, occupying the analog-to-digital conversion stage of the audio chain. The PCM4202IDBREP architecture uses a 1-bit delta-sigma modulator per channel with a novel density-modulated dither scheme for improved dynamic performance at low signal levels. Key features include 118 dB dynamic range (A-weighted), 105 dB THD+N (-1 dBFS), 216 kHz maximum sampling rate, support for 64x or 128x oversampled 1-bit Direct Stream Digital (DSD) output, a flexible PCM/DSD mode-selection pin, and a power-down mode for energy-aware applications. The device operates from a 3.3 V or 5 V analog supply and a 3.3 V digital supply, with internal low-dropout regulators that simplify system power design. Technical depth: the delta-sigma architecture oversamples the input at a multiple of the target PCM rate, then shapes quantization noise away from the audio band so that a simple decimation filter yields a wide-dynamic-range, low-distortion conversion. The built-in dither reduces idle-channel tones and eliminates correlated pattern noise at low signal levels. Differential analog inputs and a dedicated SCK/BCK/LRCK PCM interface, plus a separate DSD clock/data output, allow direct connection to DSP processors, DACs, or asynchronous sample-rate converters. Typical applications include professional studio recording interfaces, broadcast digital mixing consoles, audiophile outboard A/D converters, high-end USB audio interfaces, and DSD-capable recording chains. The PCM4202IDBREP is particularly valued in designs where the host DSP or FPGA already handles PCM/DSD routing and needs a converter whose interface timing is straightforward to integrate. Design consideration: power-supply decoupling is critical for achieving the headline dynamic-range number. Place a 10 uF bulk capacitor plus 0.1 uF and 1 nF ceramics directly at each AVDD/DVDD pin, and keep the analog ground return path isolated from the digital return at the device. For best performance, configure MCLK at the appropriate multiple of the target sampling rate and verify that BCK, LRCK, and SCK meet the timing windows specified in the manufacturer datasheet. This page synthesizes distributor pricing, drop-in alternatives, and practical design notes not found in the manufacturer datasheet, including verified cross-reference data from the TI cross-reference search and major franchised distributors.
PCM5100PW - 2VRMS Stereo DAC | Texas Instruments
Texas Instruments PCM5100PW is a stereo digital-to-analog converter with a 100 dB dynamic range, 2 VRMS output, and an integrated audio PLL. The device accepts 16-bit, 24-bit, and 32-bit PCM data at sampling rates up to 384 kHz and is packaged in a 20-pin TSSOP. Its 2 VRMS DirectPath output architecture is intended for compact, high-performance consumer audio equipment. An audio DAC is a signal-conversion integrated circuit that translates digital audio samples into an analog voltage suitable for amplification or line-level output. Within the broader semiconductor hierarchy, the PCM5100PW is a DAC, an audio data converter, and a mixed-signal integrated circuit. It sits between a digital audio source, such as a USB interface or application processor, and an analog headphone, line, or power-amplifier stage. Key features include a 100 dB stereo DAC dynamic range, 2 VRMS output capability, selectable digital-filter latency, and an integrated audio PLL. The PCM5100PW also uses DirectPath technology, requires no DC-blocking output capacitors, and includes an internal negative charge pump. These features support a compact analog output network while preserving the digital PCM input architecture. The 32-bit PCM interface accepts 16-bit, 24-bit, or 32-bit word lengths at up to 384 kHz sampling rate. Its integrated PLL provides a convenient internal clock for the stereo DAC section. According to the Texas Instruments PCM5100 family datasheet, the device also supports internal pop-free operation, which helps reduce startup transients in audio equipment. Typical applications include digital active speakers, compact stereo audio systems, USB audio adapters, Bluetooth audio receivers, game-console accessories, and portable consumer-audio products. The 3.0 V to 3.6 V supply range and 20-pin TSSOP package make the device suitable for designs where a small PCB footprint and integrated digital-to-analog conversion are priorities. When laying out the PCB, place the digital interface and analog output paths in their respective functional regions, keep the analog output route short, and follow the manufacturer’s recommended decoupling and grounding arrangement. Confirm the output configuration, digital-filter latency, clocking method, and analog load requirements before finalizing the board design. This product overview combines the manufacturer’s key device characteristics with application guidance, package information, and drop-in replacement considerations. It also distinguishes the PCM5100PW from related PCM510x devices so engineers can select the appropriate variant without relying on packaging-only ordering differences.
PCM5101PW - 106dB Stereo DAC, 2Vrms | Texas Instruments
Texas Instruments PCM5101PW is a 106 dB stereo digital-to-analog converter with 2 Vrms output, an integrated audio PLL, and support for 16-, 24-, and 32-bit PCM data at sampling rates up to 384 kHz. The device is packaged in a 20-pin TSSOP (PW) package and operates from a 3.3 V supply according to the manufacturer product listing. An audio DAC is a signal-conversion integrated circuit that reconstructs a continuous analog output from digital audio samples. The PCM5101 belongs to the DAC hierarchy within audio interface and data-converter ICs. Its integrated digital filtering, low-noise architecture, and DirectPath-style output stage are intended to simplify the analog output stage while preserving low out-of-band noise. The PCM5101 supports 16-, 24-, and 32-bit PCM data and a maximum sample rate of 384 kHz. It offers selectable digital-filter latency, allowing system designers to balance responsiveness against out-of-band noise reduction. The manufacturer describes the device as a 106 dB stereo DAC with 2 Vrms output and an integrated audio PLL, which supports clock generation for digital-to-analog conversion from a serial PCM interface. The 2 Vrms output level can drive line-level audio paths without requiring an external output-voltage amplifier when the downstream load and impedance are appropriate. Its internal negative charge pump supports pop-free operation without external DC-blocking capacitors. The integrated PLL also reduces dependence on a separate audio clock source when the host interface supplies a suitable reference clock. Typical uses include digital audio players, USB and network audio equipment, home-theater audio paths, active speakers, soundbars, and embedded multimedia systems. The converter can also serve as the output stage of a mixed-signal audio product where a digital audio stream must become a line-level analog signal. Care is needed to preserve output quality through correct grounding, filtering, and routing. For layout, place input, output, and decoupling connections directly adjacent to their associated pins, and maintain a quiet analog return path that is not shared with high-current digital return currents. Verify the external clock topology, digital interface levels, and output load against the manufacturer datasheet. The 20-pin TSSOP package also makes thermal checking and assembly planning important during PCB layout. This product overview combines the exact ordering information, converter capabilities, package details, lifecycle status, and design trade-offs needed to evaluate PCM5101PW. It also organizes drop-in family alternatives, application guidance, comparison parameters, and practical implementation notes so engineers can move from part selection to board-level implementation with fewer manual cross-checks.
PCM5102APWR - 2VRMS 112dB Stereo DAC w/ PLL | TI
The Texas Instruments PCM5102APWR is a 32-bit stereo audio DAC with integrated PLL and DirectPath ground-centered 2.1 VRMS line-level output, housed in a 20-pin TSSOP (PW) package rated -25C to +85C. It delivers 112 dB dynamic range, 100 dB THD+N, and supports PCM sample rates from 8 kHz to 384 kHz in I2S, left-justified, right-justified, and DSP-compatible formats. A stereo DAC (Digital-to-Analog Converter) is a mixed-signal IC that reconstructs an analog audio waveform from a digital bitstream. In the broader hierarchy, the PCM5102A falls under: DAC > audio DAC > data converter > mixed-signal IC > semiconductor. DirectPath architecture eliminates the need for large DC-blocking output capacitors, simplifying board layout and extending low-frequency response to DC. Key differentiating specifications include: integrated PLL eliminates external MCLK on most systems (BCLK-only operation); 112 dB dynamic range places it among the highest-performance integrated stereo DACs in the TSSOP-20 class; software-controllable soft-mute and de-emphasis filters via I2C/HW pins enable design reuse across product lines. The PCM5102A uses TI's advanced segment-DAC architecture with a precision current-source array and on-chip PLL clock multiplier, achieving excellent dynamic performance and improved tolerance to clock jitter. The integrated DirectPath charge pump generates a negative rail that allows true ground-centered outputs without coupling capacitors, eliminating low-frequency roll-off and click-pop artifacts on power-up. Typical applications include consumer audio equipment (networked streamers, USB audio interfaces, Bluetooth/wireless speakers), home-theater soundbars and AVRs, automotive infotainment main units, musical instrument effect pedals requiring 2 VRMS line outputs, and DIY audiophile DACs feeding power amplifiers or headphones. The 32-bit/384 kHz PCM interface also makes it suited for studio-grade playback systems. When designing with PCM5102APWR, consider that VCC must be 3.3 V (digital) with separate AVDD typically derived from an LDO; expose the BSTOUT/PLL filter pins; follow TI's recommended PCB layout to keep PLL and analog grounds separated. This page synthesizes distributor pricing, drop-in alternatives (PCM5100A, PCM5101A, PCM5102A-Q1), and PCB layout notes not bundled in the manufacturer datasheet.
PCM5102PW - 112dB Stereo DAC, 2VRMS, TSSOP-20 | TI
The Texas Instruments PCM5102PW is a stereo digital-to-analog converter with 112 dB dynamic range, 2 VRMS DirectPath output, and a 32-bit PCM interface supporting operation up to 384 kHz. The device is supplied in a 20-pin TSSOP package for surface-mount audio and digital signal-processing systems. Its architecture supports 16-bit, 24-bit, and 32-bit word lengths, providing compatibility with common high-resolution audio sources while maintaining a serial PCM data interface. An audio DAC is a digital-to-analog converter, a semiconductor in the broader signal-conversion and audio-processing hierarchy. It accepts encoded digital audio samples and reconstructs an analog output waveform for headphone amplifiers, line-level inputs, powered speakers, or other downstream analog stages. The PCM5102 integrates an audio PLL, allowing the digital interface and internal clocking to be coordinated for multiple standard sample rates. The principal performance characteristics are 112 dB dynamic range, 2 VRMS output capability, operation up to 384 kHz, and support for 16-, 24-, and 32-bit PCM data. The manufacturer describes the output stage as 2 VRMS DirectPath, which removes the requirement for conventional DC-blocking output capacitors in the intended application topology. A selectable digital-filter latency function provides another configuration option for systems that must balance conversion performance with timing requirements. The PCM5102 family also incorporates an internal negative charge pump and supports pop-free output behavior. These features address common implementation concerns in audio conversion systems: reducing low-frequency output offset, controlling startup transients, and simplifying the analog output network. The 3 V to 3.6 V supply range supports integration with low-voltage digital audio equipment, although the exact internal voltage rails and operating limits must be taken from the manufacturer datasheet. Typical uses include hi-fi audio playback, USB and network-audio interface outputs, digital mixing consoles, headphone-driver stages, portable media equipment, and embedded audio systems. The integrated audio PLL is particularly useful when the converter must derive a usable clock from an incoming PCM stream. Its 384 kHz capability accommodates high-resolution source formats and studio-oriented playback equipment. When laying out a design, treat the analog output path as a precision signal path and place the device close to the appropriate output-network components. Confirm the manufacturer-recommended decoupling, exposed-pad or thermal-land requirements, and analog grounding arrangement for the selected TSSOP footprint before finalizing the PCB. The PCM5102PW is packaged for a 20-pin TSSOP, so software and hardware teams should verify the exact lead assignment in the pin-description table before routing signals. This product page consolidates the confirmed PCM5102PW electrical, package, interface, and sourcing details with practical design guidance, same-family drop-in candidates, and application-oriented explanations. It adds engineering context beyond a basic catalog listing while keeping unavailable pin-level, lifecycle, and compliance conclusions separate from confirmed manufacturer information.