The design helps you capture voice in noise, reduce background sounds, and run voice recognition locally or through the cloud for devices and apps.
Capturing clear voice in noisy or reverberant environments remains a key challenge for engineers designing voice-activated and speech recognition systems. Background noise, overlapping voices, and acoustic echo can all degrade recognition accuracy, making devices less responsive or reliable. Smart home appliances, digital assistants, and video conferencing solutions require precise audio capture, yet conventional microphones often fail to separate the speaker’s voice from surrounding interference. Systems with limited computational resources face further challenges, needing to process multiple audio channels in real time without sacrificing performance.
To address these issues, Texas Instruments offers the TIDEP-0088 reference design, which combines a circular microphone array, advanced beamforming, and additional audio-processing algorithms to isolate and enhance speech. The design uses eight microphones, with seven placed around the perimeter and one in the center as a reference, capturing 16-bit samples at 16 kHz. This data is streamed through an analog-to-digital converter to a high-performance DSP that runs beamforming, noise suppression, dynamic-range compression, and acoustic echo cancellation. The result is a clean, focused audio stream that can be processed locally or sent to the cloud for further recognition.
For engineers, this reference platform provides flexibility across different application scenarios. In cloud-connected devices like digital assistants, processed audio can be sent over Ethernet, ensuring voice commands are captured accurately even in noisy rooms. In appliances or systems with limited commands, local voice recognition can run directly on the DSP, reducing latency and removing the dependency on cloud services. Advanced speech-processing applications, such as video conferencing or multi-user voice interfaces, also benefit from the platform’s ability to separate desired speech from background interference.
The beamforming algorithm creates virtual directional microphones that focus on the speaker while suppressing unwanted sounds. Additional audio-processing features, including dynamic-range compression and acoustic echo cancellation, improve sound quality and recognition accuracy. The DSP’s high-performance floating-point processor, with over 2 MB internal memory and support for up to 8 GB external memory, provides enough capacity to handle both limited-dictionary and more complex audio-processing tasks.
The modular nature of the platform allows engineers to integrate both hardware and software components into larger systems. The circular microphone board offers 360-degree coverage, making it easier to capture sound in dynamic environments. Demonstration setups allow direct comparison between the raw microphone output and the enhanced processed audio, helping engineers validate improvements in clarity and recognition performance.
The reference design directly addresses the practical challenges of voice capture in real-world environments. By isolating speech from noise, compensating for reverberation, and providing a flexible, modular platform for both local and cloud-based processing, it helps engineers accelerate development of responsive, voice-enabled devices. Whether for smart home systems, digital assistants, or advanced conferencing solutions, the design provides a reliable foundation for achieving accurate, high-quality audio capture in challenging acoustic conditions.
TI has tested this reference design. It comes with a bill of materials (BOM), schematics, assembly drawing, printed circuit board (PCB) layout, and more. The company’s website has additional data about the reference design. To read more about this reference design, click here.
