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TMS320WP010PZ Datasheet(PDF) 1 Page - Texas Instruments |
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TMS320WP010PZ Datasheet(HTML) 1 Page - Texas Instruments |
1 / 18 page TMS320WP010 DIGITAL SIGNAL PROCESSOR FOR AEC/LEC/ANS SPRS040–OCTOBER 1995 1 POST OFFICE BOX 655303 • DALLAS, TEXAS 75265 D Cancels the Acoustic Echo Common in Hands-Free Use of Cellular Phones in Automobiles – Eliminates Echo Created by Speaker Output Feeding Back Into the Microphone D Suppresses Wind, Engine, and Tire Noise D Line Echo Canceler D Standard Independent – IS-54B, IS-136, GSM, AMPS, IS-95 D Continuous, Automatic Adaptation to Changing Acoustic Environment – Initial System Training Not Required D Based on TMS320C5x Platforms D Support for Industry Standard Microcontrollers D Interfaces Easily with TI’s Voice Codec, the TCM320AC36 D Acoustic Echo Canceler Performance – Length of Echo Path up to 44 ms – Initial Adaptation Rate >20 dB/s for Speech Signals – Zero Speech Delay – Echo Return Loss up to 30 dB for Speech Echo in Automobiles and 41 dB for Sine Wave Input D Noise Suppressor Performance – Up to 10 dB Noise Suppression for Hands-Free Use of Cellular Phones in Automobiles Without Distorting Speech Signals – 16 ms Processing Delay – Continuous Adaptation D Demonstration Platform – Allows Evaluation of Software Algorithm Performance in Lab and Automobile – Ability to Enable/Disable AEC, ANS, and LEC description For the cellular phone cradle market, the TMS320WP010 provides enhanced quality, full-duplex, hands-free communications. The ’WP010 is a ROM-coded DSP that provides all the adaptive filtering and processing necessary for acoustic echo cancellation (AEC), line echo cancellation (LEC), and adaptive noise suppression (ANS). The ’WP010’s acoustic echo cancellation function is based on a Finite Impulse Response (FIR) filter with an adaptive coefficient updating algorithm. The adaptive algorithm updates continuously so that it can constantly track acoustic path changes from the speaker to the microphone. Based upon FIR filter coefficients, the echo signal can be estimated and subtracted from the observed signal at the microphone. The acoustic echo cancellation function provides continuous adaptation and needs no initial system training. It is capable of canceling an echo path of up to 44 milliseconds and provides up to 30 dB cancellation for speech signals. In addition, the initial convergence rate is faster than 20 dB per second. The line echo cancellation function cancels the near-end speech that has been reflected at the network switch and transmitted back to the cellular phone. The principle of the line echo cancellation function is very similar to that of the acoustic echo cancellation function. The difference is that the line echo cancellation function is designed to cancel the echo signal caused by network feedback due to impedance mismatch in the network circuit. The line echo cancellation function is capable of canceling a line echo path of up to 16 milliseconds. The received signal at the microphone consists of the near-end speech signal, acoustic feedback echo, and ambient noise from the environment. The purpose of the adaptive noise suppression function is to suppress ambient noise components without distorting near-end speech. The initial adaptation rate is about 3 dB per second without initial system training. It tracks the changes of the noise characteristics and suppresses noise accordingly. The adaptive noise suppression provides up to 10 dB of noise cancellation without degrading near-end speech. Please be aware that an important notice concerning availability, standard warranty, and use in critical applications of Texas Instruments semiconductor products and disclaimers thereto appears at the end of this data sheet. PRODUCTION DATA information is current as of publication date. Products conform to specifications per the terms of Texas Instruments standard warranty. Production processing does not necessarily include testing of all parameters. Copyright © 1995, Texas Instruments Incorporated |
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