Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.

High quality coding of wideband audio signals using transform coded excitation (TCX)

Abstract: This paper describes the application of transform coded excitation (TCX) coding to encoding wideband speech and audio signals in the bit rate range of 16 kbits/s to 32 kbits/s. The approach uses a combination of time domain (linear prediction; pitch prediction) and frequency domain (transform coding; dynamic bit allocation) techniques, and utilizes a synthesis model similar to that of linear prediction coders such as CELP. However, at the encoder, the high complexity analysis-by-synthesis technique is bypassed by directly quantizing the so-called target signal in the frequency domain. The innovative excitation is derived at the decoder by inverse filtering the quantized target signal. The algorithm is intended for applications whereby a large number of bits is available for the innovative excitation. The TCX algorithm is utilized to encode wideband speech and audio signals with a 50-7000 Hz bandwidth. Novel quantization procedures including inter-frame prediction in the frequency domain are proposed to encode the target signal. The proposed algorithm achieves very high quality for speech at 16 kbits/s, and for music at 24 kbits/s. >

A speech coder based on decomposition of characteristic waveforms

Abstract: For low-rate speech coding it is advantageous to represent the speech signal as an evolving characteristic waveform (CW). The CW evolves slowly when the speech signal is clearly voiced and rapidly when the speech signal is clearly unvoiced. The voiced (periodic) and unvoiced (nonperiodic) components of the speech signal can be separated by a simple nonadaptive filter in the CW domain. Because of perceptual effects, a significant increase in coding efficiency is obtained by coding these two components separately. A 2.4 kb/s coder using these principles was developed. In an independent evaluation, the performance of the 2.4 kb/s waveform interpolation (WI) coder was found to be at least equivalent to the 4.8 kb/s FS1016 standard for all of the many tests.

Modeling of speech signals using fractional calculus

Abstract: In this paper, we present a novel approach for speech signal modeling using fractional calculus. This approach is contrasted with the celebrated Linear Predictive Coding (LPC) approach which is based on integer order models. It is demonstrated via numerical simulations that by using a few integrals of fractional orders as basis functions, the speech signal can be modeled accurately. The new approach has the merit of requiring a smaller number of model parameters, and is demonstrated to be superior to the LPC approach in capturing the details of the modeled signal.

System and method for scaleable streamed audio transmission over a network

Abstract: An audio data transmission system encodes audio files into individual audio data blocks which contain a variable number bits of digital audio data that were sampled at a selectable sample rate. The number of bits of digital data and the input sampling rate are scaleable to produce an encoded bit stream bit rate that is less than or equal to an effective operational bit rate of a recipient's modem. The audio data transmission system uses computing units which are designed to select an appropriate combination of block size and input sampling rate to maximize the available bandwidth of the receiving modem. For example, if the modem connection speed for one modem is 14.4 kbps, a version of the audio data compressed at 13000 bits/s might be sent to the recipient; if the modem connection speed for another modem is 28.8 kbps, a version of the audio data compressed at 24255 bits/s might be sent to the receiver. The audio data blocks are then transmitted at the encoded bit stream bit rate to the intended recipient's modem. The audio data blocks are decoded at the recipient to reconstruct the audio file and immediately play the audio file as it is received. The audio data transmission system can be implemented in online service systems, ITV systems, computer data network systems, and communication systems.