Michael J. McLaughlin

Background noise suppression for speech enhancement and coding

A background noise suppression system developed by Motorola is included as a feature in IS-127, the TIA/EIA standard for the enhanced variable rate codec (EVRC) to be used in CDMA based telephone systems. We describe the algorithm used in this system and its implementation. We then present subjective listening test results showing the advantages of using such a system as a prefilter to a speech coder.

The ETSI extended distributed speech recognition (DSR) standards: client side processing and tonal language recognition evaluation

We present work that has been carried out in developing the ETSI extended DSR standards ES 202 211 and ES 202 212 (2003). These standards extend the previous ETSI DSR standards: basic front-end ES 201 108 and advanced (noise robust) front-end ES 202 050 respectively. The extensions enable enhanced tonal language recognition as well as server-side speech reconstruction capability. The paper discusses the client-side estimation of pitch and voicing class parameters whereas a companion paper discusses the server-side speech reconstruction. Experimental results show enhancement of tonal language recognition rates of proprietary recognition engines, when the standard extensions are used.

The ETSI extended distributed speech recognition (DSR) standards: server-side speech reconstruction

In this paper we present work that has been carried out in developing the ETSI Extended DSR standards ES 202 211 and ES 202 212. These standards extend the previous ETSI DSR standards: basic front-end ES 201 108 and advanced (noise robust) front-end ES 202 050 respectively. The extensions enable enhanced tonal language recognition as well as server-side speech reconstruction capability. This paper discusses the server-side speech reconstruction whereas a companion paper discusses the front-end extension and tonal language recognition. Experimental results show that the reconstructed speech produced by the standards is highly intelligible under clean and noisy background conditions with the DRT (diagnostic rhyme test) and TT (transcription test) scores meeting or exceeding the objective values corresponding to the USA DoD (Department of Defence) federal standard MELP (mixed-excitation linear predictive) coder operating at 2400 bit/s.

Speech and channel coding for digital land-mobile radio

The joint development of a medium bit-rate speech coder along with an effective channel coding technique to provide a robust, spectrally efficient, high-quality mobile communication system is described. A subband coder operating at 12 kb/s is used; in the absence of channel errors, it provides speech quality comparable to current analog land-mobile radio systems. The coder design incorporates a unique coding of the side information to facilitate the use of forward-error-correction coding without the need to code the entire bit stream. The use of excessive overhead for redundancy is avoided while the harsh effects of frequent channels are mitigated. These techniques have been used in an experimental FDMA (frequency-division multiple access) digital land-mobile radio system. The combined speech and channel coder operates at 15 kb/s and provides intelligible speech at fading channel error rates up to 8%. >

High performance processor for real-time speech applications

A High Performance Processor design is described which is employed in the study of digital signal processing techniques. The processor is seen to enhance algorithm development as a high speed peripheral to a general purpose minicomputer facility. In addition, it can operate in a stand-alone mode and has been used to implement a real-time Linear Predictive vocoder system.

Design and Test of a Spectrally Efficient Land Mobile Communications System Using LPC Speech

A communication system was built and tested to operate in the land mobile VHF band (150-174 MHz) at a channel separation of only 6 kHz. The audio source was digitally encoded at 2.4 kbits/s using linear predictive coding (LPC). The speech data stream was transmitted by frequency shift keying (FSK) which allowed the use of class-C transmitters and discriminator detection in the receiver. Baseband filtering of the NRZ data resulted in a narrow transmitter spectrum. The receiver had a 3 dB bandwidth of 2.4 kHz which allowed data transmission with minimal intersymbol interference and frequency offset degradation. A 58 percent eye opening was found. Bit error rate (BER) performance was measured with simulated Rayleigh fading at typical 150 MHz rates. Additional tests included capture, ignition noise susceptibility, adjacent channel protection, degradation from frequency offset, and bit error effects upon speech quality. A field test was conducted to compare the speech quality of the digital radio to that of a conventional 5 kHz deviation FM mobile radio.

Modeling and quantization of speech magnitude spectra at low data rates-evaluating design trade-offs

Several designs for modeling and quantizing the discrete harmonic amplitudes in voiced speech spectra are investigated. A cumulative distortion measure is computed for each design. This measure is shown to provide useful insight for evaluating design trade-offs.

Design and test of a spectrally efficient land mobile communications system using LPC speech

A communication system was built and tested to operate in the land mobile VHF band (150-174 MHz) at a channel separation of only 6 kHz. The audio source was digitally encoded at 2.4 kbits/s using linear predictive coding (LPC). The speech data stream was transmitted by frequency shift keying (FSK) which allowed the use of class-C transmitters and discriminator detection in the receiver. Baseband filtering of the NRZ data resulted in a narrow transmitter spectrum. The receiver had a 3 dB bandwidth of 2.4 kHz which allowed data transmission with minimal intersymbol interference and frequency offset degradation. A 58 percent eye opening was found. Bit error rate (BER) performance was measured with simulated Rayleigh fading at typical 150 MHz rates. Additional tests included capture, ignition noise susceptibility, adjacent channel protection, degradation from frequency offset, and bit error effects upon speech quality. A field test was conducted to compare the speech quality of the digital radio to that of a conventional 5 kHz deviation FM mobile radio.

A technique of multi-tap long term predictor (LTP) filter using sub-sample resolution delay [speech coding applications]

The method of a 1/sup st/ order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, equivalently, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Defining the delay with sub-sample resolution enables this novel multi-tap LTP filter to explicitly model delay values that have a fractional component. The filter coefficients, largely freed from implicitly modeling the effect of delays that have a fractional component, seek to maximize the prediction gain of the LTP filter by modeling the frequency dependent gain. This is in contrast to a conventional multitap LTP filter, which applies a single model to tackle the dual tasks of representing the non-integer valued delays and the frequency dependent gain. Experimental results are presented for narrowband and wideband speech. This technique is part of the 3GPP2 source-controlled variable-rate multimode wideband speech codec (VMR-WB) rate set 1 standard.