Speech and Audio Technologies

A DSP Powered Solid State Audio System

Authors:

Jason D Kridner,
Mark T Nadeski,
Pedro R Gelabert,

Page (NA) Paper number 2137

Abstract:

New audio compression algorithms and non-volatile flash memory technology have enabled the creation of portable solid-state personal audio players. This paper presents a low-power portable audio system based on the Texas Instruments TMS320C5000 DSP family. This system is designed to play music and other audio media stored on flash memory cards that can hold over an hour of CD quality music. The flash card provides higher audio quality than a cassette tape, yet is smaller and more durable than a CD. The music or audio material downloaded to the flash is obtained from licensed distributors, either through the internet or through kiosks setup in retail outlets. All the audio decoding and watermarking required by this system is handled by the TMS320C5000 DSP. System performance characteristics are also presented.

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New Implementation Techniques Of A Real-Time MPEG-2 Audio Encoding System

Authors:

Hyen-O Oh,
Sung-Youn Kim,
Dae-Hee Youn,
Il-Whan Cha,

Page (NA) Paper number 1594

Abstract:

In this study, new implementation techniques of a real-time MPEG-2 audio encoding system are presented. The system is developed using general-purpose DSP's. It consists of one master unit and five slave units, and its structure is basically based on our early work. Two fast algorithms are developed and applied to the most compute-intensive routines of the encoding process. These algorithms play a key role to improve the entire system performance. The implemented system is designed to encode audio signal into MPEG-2 layer II bitstream with full configurations up to 5.1 channels and 640Kbps, and intended to support state-of-the-art quality. Generated bitstream can be stored in hard disk on PC or sent to integration system to be multiplexed with corresponding video-bitstream.

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An Improved Residual-Domain Phase/Amplitude Model for Sinusoidal Coding of Speech at Very Low Bit Rates: A Variable Rate Scheme

Authors:

Sassan Ahmadi,

Page (NA) Paper number 2193

Abstract:

An improved harmonic sinusoidal model is presented, where the underlying sine wave amplitudes and phases are efficiently represented using a combination of linear prediction, linear phase alignment, all-pass filtering, and spectral sampling in the residual-domain. The analysis and synthesis systems are introduced and the derivation and encoding of each model parameter is discussed. Performance analysis on a large database indicates effective modeling of the sinusoidal parameters. A variable-rate sinusoidal coder operating at an average bit rate of 1.75 kbps, based on the proposed model, has been developed, yielding reproduced speech of good quality, intelligibility, and naturalness. The proposed model may find applications in low bit rate speech coding in high capacity wireless communication systems.

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Implementation of an Enhanced Fixed Point Variable Bit-Rate MELP Vocoder on TMS320C549

Authors:

Ali E Ertan,
Emre B Aksu,
Hakki G Ilk,
Haydar Karci,
Önder Karpat,
Taner Kolçak,
Levent S;endur,
Mubeccel Demirekler,
Ahmet Enis Çetin,

Page (NA) Paper number 1758

Abstract:

In this paper, a fixed point Variable Bit-Rate (VBR) Mixed Excitation Linear Predictive Coding (MELP) vocoder is presented. The VBR-MELP vocoder is also implemented on a TMS320C54x and it achieves virtually indistinguishable federal standard MELP quality at bit-rates between 1.0 to 1.6 kb/s. The backbone of VBR-MELP vocoder is similar to that of federal standard MELP. It utilizes a novel sub-band based voice activity detector in the back-end of encoder to discriminate background noise from speech activity. Since proposed detector uses only parameters extracted in the encoder, its computational complexity is very low.

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Improving EVRC Half Rate by the Algebraic VQ-CELP

Authors:

Fenghua Liu,
Ryan Heidari,

Page (NA) Paper number 3020

Abstract:

This paper presents an algebraic vector quantized codebook excited linear prediction (AVQ-CELP) speech codec. The objective is to enhance the half rate mode of IS-127, the enhanced variable rate codec (EVRC). In AVQ-CELP scheme, only the perceptually important components are encoded, and the selection of the components is done in a way similar to the ACELP. An open-loop procedure is used to select the sub-vectors. The selected sub-vectors are concatenated and vector quantized. An analysis-by-synthesis strategy is used to determine the optimal excitation. The generalized Lloyd algorithm (GLA) is used to optimize the AVQ codebook. In order to improve the synthesis quality of voiced frames, a two-pulse version of ACELP is used in the strong voiced frames. The proposed algorithm was incorporated in the Nokia CDMA handset prototype. Under a joint collaboration effort with SK Telecom, a field-testing was performed in Korea to evaluate the performance of the proposed AVQ algorithm. The results indicate a considerable improvement relative to the standard EVRC operating at the maximum half-rate.

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A 4 kbps Adaptive Fixed Code-Excited Linear Prediction Speech Coder

Authors:

Hong Kook Kim, AT&T Labs Research, Rm. E148, 180 Park Avenue, Florham Park NJ 07932, USA (USA)
Mi Suk Lee, Dept. of Electrical Eng., Korea Advanced Institute of Science and Technology, 373-1 Kusong-Dong, Yusong-Gu, Taejon 305-701, Korea (Korea)
Hwang Soo Lee, Dept. of Electrical Eng., Korea Advanced Institute of Science and Technology, 373-1 Kusong-Dong, Yusong-Gu, Taejon 305-701, Korea (Korea)

Page (NA) Paper number 1329

Abstract:

In this paper, we propose an adaptive fixed code-excited linear prediction (AF-CELP) speech coder operating at 4 kbps. By exploiting the fact that a fixed codebook contribution to speech signal is also periodic as the corresponding adaptive codebook contribution, the adaptive fixed codebook model efficiently represents excitation signals. In order to overcome the quality degradation caused by the coarse quantization of excitation, a paired pulse algebraic codebook structure is also applied to the excitation model. Additionally, a pitch prefiltering, a noise spreading, and a harmonic enhancement technique are adopted in the decoding process. The spectrogram reading and informal listening tests proved that the AF-CELP reproduces high quality speech.

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Multimode Variable Bit Rate Speech Coding: An Efficient Paradigm For High-Quality Low-Rate Representation Of Speech Signal

Authors:

Amitava Das,
Andy DeJaco,
Sharath Manjunath,
Ananth Ananthapadmanabhan,
Jeff Huang,
Eddie Choy,

Page (NA) Paper number 3014

Abstract:

The speech signal consists of a time-varying ensemble of different types of segments with distinct characteristics, which require different degrees of coding resolution in order to retain an overall high voice quality. A fixed-rate coder can capture such time-varying characteristics only if it operates at a high enough bit rate. At low bit rate, a fixed-rate coder will not be able to capture all of these various segments well and will fail to render high voice quality. A multimode variable bit rate (VBR) coder uses an arsenal of modes, operating at different bit rates. These modes are designed to represent these different speech segments optimally with the ight amount of coding resolution. Thus, a multimode VBR codec adapts the coding mechanism to the input speech and delivers high quality at low (average) rates. This paper presents the essential framework and the unique advantages of a multimode VBR codec and suggests algorithms for the different modes.

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Segmental Prototype Interpolation Coding

Authors:

Costas C.S. Xydeas, University of Manchester, UK (U.K.)
Thomas M Chapman, University of Manchester, UK (U.K.)

Page (NA) Paper number 2340

Abstract:

Current parametric speech coding schemes can achieve high communications quality speech at bit rates in the range of 2.4 to 1.5kbits/sec. Most schemes sample and quantise, at regular intervals, the "tracks in time" generated by the parameters of the speech production model. As a result, reconstructed "parameter tracks" do not evolve "smoothly" with time. Furthermore, no advantage is taken of the "linguistic event" nature of speech. In this paper, model parameter "time tracks" are split into non overlapping speech "event" related segments. These segment based evolutions of model parameters are then vector quantised to provide at the receiver a smooth and subjectively meaningful reconstruction. Thus the paper presents an application of this generic segmental speech model quantisation approach to a 1.5kbits/sec Prototype Interpolation Coding (PIC) system. Results indicate that the proposed methodology can almost halve the bit rate of this PIC system while preserving overall recovered speech quality.

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