Enhanced Variable Rate Codec

About: Enhanced Variable Rate Codec is a research topic. Over the lifetime, 357 publications have been published within this topic receiving 4842 citations. The topic is also known as: EVRC.

A novel transcoding scheme from EVRC to G.729ab

Abstract: In this present day of Internet and wireless it has become increasingly important to have an interoperable communication between these two systems. It is obvious that at the present environment a direct speech transcoding scheme holds the key for the efficient and seamless transmission of speech communication between the two systems. This paper presents a direct speech transcoding scheme from the CDMA (code division multiple access) standard EVRC (enhanced variable rate codec) to the ITU (International Telecommunication Union) standard G.729ab. The EVRC is developed by Lucent and adopted as TIA/IS 127 standard by TIA (Telecommunications Industries Association). It is the most widely used speech codec for the present CDMA mobile systems. Again, it is also a very competent candidate for the 3rd generation mobile system for speech coding for its high quality. The ITU standard G.729ab is the de facto workhorse for low bit rate speech coding over VOP (voice over packet) networks. The motivation behind this transcoding scheme is to transform the EVRC parameters into the G.729ab parameters directly without going through the whole process of decoding the EVRC parameters and then encoding the resultant synthetic speech using the G.729ab encoder so that it improves the delay characteristics and also the quality of the speech. In the same time this approach also results in considerable savings in computations.

Audio bandwidth detection in the EVS codec

Abstract: Speech and audio codecs are usually designed such that they encode all the frequency bands of the input signal spectrum. If the higher bands do not contain any perceptually meaningful content, these codecs often do not work optimally as they assign part of the available bit budget to encode these bands. In this paper we describe a bandwidth detection algorithm that determines the effective audio bandwidth of the input signal. This information is used to set the codec to its optimal configuration and consequently increase the coding efficiency for band-limited signals by allocating bits to encode only the useful bandwidth. The presented algorithm has been used in the new codec for Enhanced Voice Services (EVS), recently standardized by 3GPP, but it can be employed in other codecs as well.

Design of a (127,99) four error correcting BCH codec

Abstract: The design of a codec based on the (127,99) four-error-correcting BCH code is described. This code was chosen as a compromise between overall performance and implementation complexity for a frequency-hopped spread-spectrum system operating under worst-case partial band noise jamming and worst-case multitone jamming. The codec is designed for implementation with application-specific integrated circuits. The approach has a short implementation cycle, requires a very small number of integrated circuit chips, and yields a codec that can operate up to about 5 Mb/s. >

Improving EVRC half rate by the algebraic VQ-CELP

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 subvectors. 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.

Preprocessing for variable rate audio encoding

Abstract: Since music signals are encoded by a voice encoding method optimized to human voice signals such as EVRC (Enhanced Variable Rate Coding) in a cellular communication system, the music signals are often distorted by such encoding method, and listeners experience pauses in music caused by such voice-optimized encoding method. To improve the perceptual sound quality of music, a method for preprocessing digital audio data is provided in order to prevent the problem of pause in music signals in a cellular phone. In particular, AGC (Automatic Gain Control) preprocessing and PHE (Pitch Harmonics Enhancement) is performed to the digital audio data having low dynamic range. By this method, the number of pauses in music signal is reduced, and the perceptual sound quality of the music is improved.