Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

Spatial speech coding for multi-teleconferencing

Abstract: This paper describes a structural model for the implementation of multichannel speech coding for teleconferencing with spatial audio reproduction. Multiple nonaural speech sources are synthesized into binaural sound to produce a more realistic videoconferencing environment. The activity information of the individual binaural speech, which is determined by the voice activity detection algorithm, is used to calculate two weighting factors prior to mixing. Furthermore, a third level of weight adjustment can be carried out by adjusting these weighting factors before applying to the individual voice source. A scheme to remove undesirable noise spikes is also introduced. Both channels are then coded using the G.723.1 speech codec individually.

Improved audio quality at 48 Kbits/s for MPEG-4 AAC

Abstract: A novel coding scheme for improving reconstructed audio quality at 48 Kbits/s for MPEG-4 AAC based codec is introduced. In contrast to the fixed subband widths in an MPEG-4 codec, a pre-defined configuration is selected and signaled to the decoder in each frame. The selected configuration groups together appropriate number of neighboring samples to form subbands dynamically and improves the efficiency of the two loop search procedure within the encoder. This results in reducing the Noise to Mask ratio and improves the output quality. Performance analysis of the proposed codec through Subjective listening tests according to ITU-T recommendation BS-1116 indicates a significant improvement compared to the standard MPEG-4 AAC codec. The paper concludes with a discussion of future research directions.

An embedded speech and audio coding method based on bit-plane coding and SQVH

Abstract: A multi-layer embedded speech and audio coding algorithm based on bit-plane coding and Scalar Quantized Vector Huffman Coding (SQVH) is proposed in this paper. In this codec the signal sampled at 32 kHz can be coded in terms of scalable bit rates. The core codec is International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.729.1 which can process signal with 7 kHz bandwidth. Besides there are five extra bit-rates added and the bandwidth is extended to 14 kHz. The additional bit-rates include 36, 40, 48, 56, and 64kb/s. Some new methods used in the additional layers are proposed in this paper. The objective and subjective listening tests show that this codec has good performance compared with reference codec.

An MPEG-4 Twin-VQ based high quality audio codec design

Abstract: This paper presents a high quality audio codec design based on a transform-domain weighted interleave vector quantization (Twin-VQ) scheme adopted in the MPEG-4 audio standard. Three novel techniques are employed in this scheme to compress the data, ie, (1) flattening of MDCT coefficients by the spectrum of linear predictive coding (LPC) coefficients; (2) further flattening of MDCT coefficients by the Bark envelope; and (3) weighted interleave vector quantization. This paper examines the related design issues in implementing an efficient Twin-VQ codec. Fast computation algorithms are derived for the computationally intensive modules. Design parameters of each module are determined and the codebooks for weighted interleave vector quantization are constructed. Experimental results show that the designed codec can compress natural audio efficiently and reproduce high quality outputs.

Adaptive prediction order scheme for AMR-WB+

Abstract: In this paper, we present an adaptive linear prediction order scheme - a simple and effective method to facilitate variable rate coding. We have applied it to the Extended Adaptive Multi-Rate Wideband (AMR-WB+), a state-of-the-art generic speech and audio coder with fixed-length coding designed for wireless transmission. By introducing this scheme into the codec, bit rate can be reduced with a negligible degradation of the quality. The order of linear prediction is adapted based on the audio content, and can be determined by a closed-loop or an open-loop optimal order selection module. The resulted audio quality is examined by average segmental SNR and Perceptual Evaluation of Audio Quality (PEAQ) measure. For this evaluation, speech, music and mixed content audio signals are used. The result proves the effectiveness of this method.