Texas Instruments TLV320AIC11CPFB

Overview

TLV320AIC11CPFB from Texas Instruments is a 16-bit sigma-delta audio codec IC designed for voiceband signal conversion in telephony and embedded voice systems. It integrates independent ADC and DAC paths with on-chip antialiasing and interpolation FIR filters, programmable gain amplifiers (PGA), and dual analog input multiplexing (INP/INM or AURXFP/AURXM). It operates at 8 ksps with 39 mW typical power dissipation and supports differential or single-ended analog I/O.

For engineers reviewing the TLV320AIC11CPFB datasheet, TLV320AIC11CPFB pinout, TLV320AIC11CPFB application, or TLV320AIC11CPFB equivalent, key selection considerations include its 48-pin TQFP package, master/slave frame-sync interface, continuous data transfer mode for DSP autobuffering, event-monitor capability for RING/OFF-HOOK detection, and low-voltage digital I/O (1.1 V–3.6 V) compatibility with TI TMS320UC54x and similar DSPs.

Technical Context

The TLV320AIC11CPFB implements a synchronous serial audio interface with configurable master/slave operation via M/S pin and frame-sync (FS/FSD) timing. Its ADC and DAC paths each use oversampling sigma-delta modulation followed by decimation/interpolation filtering, with optional FIR bypass for higher throughput (up to 88 ksps) at reduced SNR.

It supports three distinct communication modes: primary (data conversion), secondary (register configuration via DIN/DOUT), and direct-configuration (DCSI single-wire host interface). Automatic cascading detection (ACD) enables up to eight devices in cascade, with programmable N-divider (control register 2) setting sampling rate via MCLK/(256×N).

Key Specifications

Pinout & Package

The TLV320AIC11CPFB is housed in a 48-pin TQFP (PFB) package with exposed thermal pad, rated for commercial (0°C to 70°C) and industrial (–40°C to 85°C) temperature ranges. Pin functions are validated per SLWS100A datasheet Section 1.3 and 1.5.

Key Features

Applications

Equivalent & Alternatives

The following parts are listed as comparable options for similar audio codec applications.

Compared with TLV320AIC11CPFB, TLV320AIC23BPW offers higher resolution but lacks cascade automation and event monitoring, while PCM3002E reduces integration to lower BOM cost but increases design complexity and board area.

Availability

TLV320AIC11CPFB is available at Aetrix Electronics and suitable for VOIP handsets, automotive hands-free kits, and cable modem voice ports requiring stable component supply across commercial and industrial temperature grades.

Supply support for TLV320AIC11CPFB includes scheduled delivery planning, volume procurement assistance, BOM continuity management, traceable sourcing, and lifecycle availability coordination for OEM customers, industrial embedded developers, connected-device designers, and electronics production programs.

Manufacturer

Texas Instruments is a global semiconductor leader specializing in analog, embedded processing, and connectivity technologies, with decades of leadership in audio signal processing and DSP interface solutions.

The TLV320AIC11CPFB belongs to TI's legacy audio codec product line, engineered specifically for low-power, single-channel voiceband applications in telephony infrastructure, hands-free systems, and embedded voice interfaces.

FAQ

What is the primary function of the TLV320AIC11CPFB in a voice system?

The TLV320AIC11CPFB serves as a complete single-channel audio codec, performing synchronized 16-bit analog-to-digital and digital-to-analog conversion for voiceband signals. It integrates PGA, antialiasing and interpolation filters, and flexible analog I/O to replace multiple discrete components in telephony and VOIP designs. Its architecture directly supports TI DSP autobuffering and event-driven monitoring, making it ideal for real-time voice applications where latency and power efficiency matter.

Does the TLV320AIC11CPFB support master-slave cascading, and how is it configured?

Yes, the TLV320AIC11CPFB supports automatic cascading detection (ACD) for up to eight devices. Cascading is enabled by connecting the FSD output of the master to the FS input of each slave, with M/S pins set accordingly. The device auto-detects cascade depth and adjusts internal clock division (via N-register) to maintain correct sampling rate - no manual address assignment or firmware configuration is required for basic operation.

What are the power supply requirements for the TLV320AIC11CPFB?

The TLV320AIC11CPFB requires three independent supplies: AVDD1/AVDD2 (3 V–5.5 V) for analog core and op-amps, DVDD2 (3 V–5.5 V) for digital core, and DVDD1 (1.1 V–3.6 V) for low-voltage I/O. This separation allows interfacing with mixed-voltage systems - for example, DVDD1 can connect directly to a 1.8 V or 3.3 V DSP I/O rail without level shifters, while AVDD supports electret MIC bias generation.

Can the TLV320AIC11CPFB operate without external analog filters?

Yes - the TLV320AIC11CPFB includes an on-chip antialiasing filter for the ADC path and interpolation filter for the DAC path, both programmable via control register 1. When enabled, these filters eliminate the need for external RC networks. However, if FIR bypass is selected (bit D2 = 1), external antialiasing must be added before INP/INM to prevent aliasing at higher sampling rates like 88 ksps.

How does the event-monitor mode work on the TLV320AIC11CPFB?

The event-monitor mode uses the ALTIN pin to detect external analog voltage transitions (e.g., telephone RING or OFF-HOOK) without engaging the ADC path. When activated via control register 3, the device monitors ALTIN and asserts a flag or triggers an interrupt-capable output (e.g., FLAG pin), enabling the host DSP to respond to telephony events with minimal latency and zero audio conversion overhead - preserving CPU cycles and power.