Audio Solutions for Hearables and Wearables

The ADAU1860/ADAU1860-1 are codecs with three inputs and one output that incorporate two digital signal processors (DSPs). The path from the analog input to the DSP core to the analog output is optimized for low latency and is ideal for noise canceling earphones. With the addition of just a few passive components, the ADAU1860/ADAU1860-1 provide a complete earphone solution.

APPLICATIONS

• Noise canceling handsets, headsets, and headphones
• Bluetooth active noise canceling (ANC) handsets, headsets, and headphones
• Personal navigation devices
• Digital still and video cameras
• Musical instrument effect processors
• Multimedia speaker systems
• Smartphones

The SSM6515 is a fully integrated, high efficiency, mono Class-D audio amplifier with digital input that is ideally suited for use in wireless headphones. The application circuit requires few external components and operates from a 1.8 V (PV_DD) supply and a 1.2 V to 1.8 V (IOV_DD) supply. With 1.8 V signaling, these can be shared. It can deliver 85 mW of continuous output power into a 16 Ω load with <1% total harmonic distortion plus noise (THD + N).

The SSM6515 features a high efficiency, low noise modulation scheme that requires no external inductor/capacitor (LC) output filters. This scheme continues to provide high efficiency even at very low output power resulting in battery life savings compared to conventional headphone amplifiers. The device operates with > 90% efficiency at 55 mW into a 16 Ω load and it has a signal-to-noise ratio (SNR) of 113 dB, A weighted. Spread spectrum, stable common-mode pulse density modulation provides a much lower electromagnetic interference (EMI) radiated emissions compared with other Class-D architectures.

The digital input eliminates the need of an external digital-to-analog converter (DAC). The SSM6515 has a micropower shutdown mode with a typical shutdown current of 6 μA at the 1.8 V PV_DD supply and can automatically power down with no input. The device also includes pop and click suppression circuitry that minimizes voltage glitches at the output during turn on and turn off. Inputs support various serial audio formats, including I²S, time division multiplexing (TDM), and pulse density modulation (PDM).

The SSM6515 is designed to operate with an I²C control interface or in standalone mode and specified over the temperature range of −40°C to +85°C. The device has built-in output short-circuit protection. The SSM6515 is available in a halide free, 11-ball, 0.984 mm × 1.444 mm, 0.35 mm pitch wafer-level chip scale package (WLCSP).

APPLICATIONS

• Wireless headphones
• Active noise canceling headphones
• Smartphone earpiece speakers
• Hearing assistance devices
• Portable electronics

The MAX98050 is a low-power, high-performance audio CODEC with integrated low-latency digital filters for wireless hearables, headsets, and headphones.

The device features a mono playback channel with a 5-band biquad equalizer and a high-efficiency, fully differential hybrid class-AB/class-D headphone amplifier. The playback headphone amplifier is optimized for extremely low output noise levels and minimized quiescent power consumption while maximizing output power efficiency.

The device includes three microphone input channels capable of supporting voice, ambient, and anti-noise record use cases. Each channel can individually record from either an external analog or digital microphone, and then can route audio data to both the record channels (to the digital audio interface) and the internal low-latency digital filter channels. A fourth record channel to the digital audio interface is provided to allow the host to monitor the playback channel output digital data.

Two low-latency, digital filter channels are provided that can support a variety of use cases. These include noise cancellation applications (feed-forward, feedback, and hybrid filter profiles), both voice and ambient enhancement and transparency applications, and sidetone applications. Each channel provides a 12-band digital biquad filter, a DRC or Limiter stage, a digital volume control block, and an ultrasound filter. In addition, a third internal digital filter channel is provided in the playback channel that is structured for playback compensation and/or general feedback compensation filters. This channel includes a 10-band biquad filter and a digital volume control block. To support dynamic or adaptive filter and use case transitions, the biquad filters and volume controls provide banked profiles that can be updated/swapped without disruption to audio playback.

The flexible digital audio interface supports common PCM audio data formats such as I²S, left-justified, and TDM timing with 16-bit, 24-bit, or 32-bit data word and channel lengths. The PCM interface supports playback sample rates from 8kHz to 192kHz and record sample rates from 8kHz to 48kHz. The device requires external bit and frame clocks but eliminates the need for a high-speed external master clock.

The device is available in a small 6.7mm² 36-bump WLP package (0.4mm pitch) with a low 0.5mm height ideal for tight form factors. The device operates over the extended -40°C to +85°C temperature range.

Applications

• True Wireless Hearables and Headsets
• Stereo Bluetooth Headsets and Headphones
• Mono Bluetooth Earpieces
• PSAP Devices and Hearing Assist Devices
• VR/AR Headsets and Head-Mounted Devices

The SigmaStudio^® graphical development tool is the programming, development, and tuning software for the SigmaDSP^® and SHARC^® audio processors and A²B^® transceivers. Familiar audio processing blocks can be wired together as in a schematic, and the compiler generates DSP-ready code and a control surface for setting and tuning parameters. This tool allows engineers with no DSP code writing experience to easily implement a DSP into their design and yet is still powerful enough to satisfy the demands of experienced DSP designers. SigmaStudio links with both Analog Devices evaluation boards and production designs to provide full in-circuit real-time IC control.

SigmaStudio includes an extensive library of algorithms to perform audio processing such as filtering, mixing, and dynamics processing, as well as basic low-level DSP functions and control blocks. Advanced record-side processing algorithms such as Enhanced Stereo Capture and wind noise detection are included in the standard libraries. Plug-in algorithms from Analog Devices and 3rd party partners can be added to SigmaStudio's drag-and-drop library.

Along with its graphical DSP signal flow development, SigmaStudio also includes other features to speed up the design cycle from product concept to release. SigmaStudio includes tools for intuitively setting control registers, calculating tables of filter coefficients, visualizing filter magnitude and phase responses, generating C header files, and sequencing a series of controls to ease your transition from SigmaStudio to system implementation on your microcontroller.

SigmaStudio^®+ is an integrated system level solution for programming, configuring and tuning software for audio processors, A2B^® transceivers and Algorithms/IPs. SigmaStudio+ offers the following advantages:

Intuitive System Design

SigmaStudio+ architecture has been enhanced to support a system consisting of multiple platforms, network transceivers and connecting devices. This architecture provides a systemic approach for a seamless integration of processors, A2B transceivers and codecs in a single schematic.

Faster deployment cycle due to UI programming

SigmaStudio+ allows engineers with no DSP programming experience to easily implement their design and yet is still powerful enough to satisfy the demands of experienced DSP designers. In addition, the product links with both evaluation boards and production designs to provide full in-circuit real-time control.

Drag-and-drop, connect and download features enable customers to realize complex audio signal chain intuitively in real time with minimal effort. Seamless support for tuning interface with customer IP protection enables rapid development from concept to final product release.

Along with its graphical DSP signal flow development, SigmaStudio+ has other features such as intuitively setting control registers, calculating tables of filter coefficients, visualizing filter magnitude and phase responses, generating C header files, and sequencing a series of controls to ease transition from SigmaStudio+ to system implementation on target platform.

Efficient audio processing algorithm library

SigmaStudio+ includes an extensive library of algorithms to perform audio processing such as filtering, mixing, and dynamics processing, as well as basic low-level DSP functions and control blocks. Audio processing blocks can be wired together in a schematic and the compiler generates DSP ready code and a control surface for setting and tuning parameters.

Optimized library modules catering to a wide range of audio applications can be configured for precision, channels, stages and slew type which provides flexibility to OEMs to design and develop distinguished products.

Support to design custom algorithms (Algorithm Designer)

SigmaStudio+ leverages an upgraded algorithm designer, which allows customers to develop custom modules and integrate IPs to distinguish their products without compromising on their proprietary designs

Advanced Features for Design and Debug

Enhanced debug features, intuitive error messages, integrated help content, reconnect to target without reloading the schematic, accurate MIPS and memory profile information for load balancing aid in rapid development of products. In addition, Apache Thrift Scripting is supported for automation and testing.