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Audio and Speech Solutions for Communications Equipment

Communications equipment such as phones, walkie-talkies, etc. need to process speech or audio signals for primarily two purposes:

  • Encoding or decoding the signal for efficient or fast communication
  • Remove or suppress acoustic echoes, noise from telecom switching networks or simply background noise

Eliminating unwanted noise or echoes is a processor-intensive task. Microchip provides the following libraries to aid noise removal in communication systems:

These libraries may be used in a variety of applications such as Mobile Hands Free Kits, Speakerphones, Intercoms, Emergency Alarm Units & Teleconferencing units.

Noise Suppression (NS) Library is used to reduce unwanted noise mixed with speech. The NS library is suitable for applications without an isolated ‘noise reference’ source. The Noise Suppression algorithm is a frequency-domain algorithm based on Spectral Subtraction. A Voice Activity Detector differentiates noise and speech. Noise reduction filters self-adjust every 10ms during periods of speech inactivity (using an FFT). The speech is continuously filtered, reducing its noise content
Acoustic Echo Cancellation library is used to cancel unwanted echo caused by acoustic path from speaker to microphone. The AEC library is ITU G.167 compliant and suitable for systems where speaker and microphone are close to each other. The library supports full-duplex operation and is based on Least Mean Square (LMS) Adaptive Filtering. A Voice Activity Detector and Double Talk Detector determine when the Adaptive Filter should be adjusted. An LMS Filter iteratively models echo propagation path during periods of far-end speech. The filtered output is subtracted from microphone input every 10 ms, thus removing echo. The AEC library supports cancellation of echo “tail lengths” that are 16, 32 or 64 ms long and can be used with audio signals with bandwidth, 0-4 kHz, at 8 kHz sampling rate.
The Line Echo Cancellation library is used to cancel unwanted echo generated by various telephone circuits (e.g. hybrids) or digital network components. The LEC algorithm is a time-domain algorithm based on Least Mean Square (LMS) Adaptive Filtering. It is an ITU G.168 compliant solution and supports echo "tail lengths" that are 16, 32 or 64 ms long.
The dsPIC DSC Speech and Audio Fast Forward (SAFF) tool provides users with real-time control of Microchip’s Speech and Audio Processing Algorithms as they are running in an embedded system application. This PC GUI tool facilitates easy tuning of Microchip’s Noise Suppression, Acoustic Echo Cancellation, Line Echo Cancellation and Equalizer algorithms in speech and audio applications by simplifying parametric tuning of algorithms.
The dsPIC DSC Automatic Gain Control Library automatically adjusts the amplitude of a speech signal to match a set level. This is useful in speech applications where the distance between the speech source and the microphone is not fixed. The Automatic Gain Control Library can be used readily with Microchip's Speech and Audio Solutions for speech signal pre-processing.