Codec2

Abstract: A vocoder-based speech synthesis system, named WORLD, was developed in an effort to improve the sound quality of realtime applications using speech. Speech analysis, manipulation, and synthesis on the basis of vocoders are used in various kinds of speech research. Although several high-quality speech synthesis systems have been developed, real-time processing has been difficult with them because of their high computational costs. This new speech synthesis system has not only sound quality but also quick processing. It consists of three analysis algorithms and one synthesis algorithm proposed in our previous research. The effectiveness of the system was evaluated by comparing its output with against natural speech including consonants. Its processing speed was also compared with those of conventional systems. The results showed that WORLD was superior to the other systems in terms of both sound quality and processing speed. In particular, it was over ten times faster than the conventional systems, and the real time factor (RTF) indicated that it was fast enough for real-time processing. key words: speech analysis, speech synthesis, vocoder, sound quality, realtime processing

Abstract: Speech communication, C.Wheddon low bit rate speech coding for practical applications, C.B.Southcott et al an improved implementation of adaptive quantizers for speech waveform encoding schemes, L.F.Lind et al development of a speech codec for the skyphone service, I.Boyd et al an efficient coding scheme for the transmission of high quality music signals, S.M.F.Smyth and P.Challener noise reduction using frequency-domain, non-linear processing for the enhancement of speech, E.Munday formant based speech synthesis, P.M.Hughes TEXtalk - the British Telecom text-to-speech system, D.L.Gibson et al Phone-in competitions - a development and evaluation tool for voice-interactive systems, P.C.Millar et al machine translation of speech, F.W.M.Stentiford and M.G.Steer beyond speech recognition - language processing, R.Linggard hidden Markov models for automatic speech recognition - theory and application, S.J.Cox fixed dimension classifiers for speech recognition, P.Woodland and W.Millar neural arrays for speech recognition, G.D.Tattersall et al multi-layer perceptrons applied to speech technology, N.McCulloch et al single-layer look-up perceptrons (SLLUPS), G.D.Tattersalll et al.

Abstract: Predictive coding methods attempt to minimize the rms error in the coded signal. However, the human ear does not perceive signal distortion on the basis of rms error, regardless of its spectral shape relative to the signal spectrum. In designing a coder for speech signals, it is necessary to consider the spectrum of the quantization noise and its relation to the speech spectrum. The theory of auditory masking suggests that noise in the formant regions would be partially or totally masked by the speech signal. Thus, a large part of the perceived noise in a coder comes from frequency regions where the signal level is low. In this paper, methods for reducing the subjective distortion in predictive coders for speech signals are described and evaluated. Improved speech quality is obtained: 1) by efficient removal of formant and pitch-related redundant structure of speech before quantizing, and 2) by effective masking of the quantizer noise by the speech signal.

Abstract: This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.