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.

A low complexity audio coding scheme for wideband audio

Abstract: Differential encoding is a well known low complexity coding technique. Its use in the coding of wideband audio is limited by its inability to follow rapid changes in the signal. This is a serious drawback when coding high fidelity audio where this inability can seriously degrade the perceptual quality of the reconstruction. This overload problem can be remedied by using a recursively indexed quantizer. In this paper we present some empirical results for the differential coding of audio signals. >

A high-quality mixed-signal audio codec in 65-nm CMOS

Abstract: This paper presents the implementation of an audio codec suitable for 3rd generation mobile telecom standards in a state-of-the-art 65-nm CMOS process. The overall design of the codec together with measurement results of key features are presented. The codec is a mixed-signal design, containing A/D and D/A converters, as well as the digital signal processing blocks needed to support them. Buffers capable of driving low-ohmic loads are also included.

A Particle Swarm Optimization-Based Approach to Speaker Segmentation Based on Independent Component Analysis on GSM Digital Speech

Abstract: Adaptive Multi-Rate (AMR) codec was standardized for GSM in 1999. AMR offers substantial improvement over previous GSM speech codecs in error robustness by adapting speech and channel coding depending on channel conditions. The Adaptive Multi-Rate speech codec is adopted as a standard for IMT-2000 by ETSI and 3GPP and consists of eight source codecs with bit rates from 4.75 to 12.2 kbit/s. In this paper, we present an approach comprising of particle swarm optimization (PSO), which encodes possible segmentations of an audio record, and measures mutual information between these segments and the audio data. This measure is used as the fitness function for the PSO. A compact encoding of the solution for PSO which decreases the length of the PSO individuals and enhances the PSO convergence properties is adopted. The algorithm has been tested on two actual sets of data with AMR format for speaker segmentation, obtaining very good results in all test problems. The results have been compared to the widely used a genetic algorithm-based in several practical situations. No assumptions have been made about prior knowledge of speech signal characteristics. However, we assume that the speakers do not speak simultaneously and that we have no real-time constraints.

Electronic apparatus and method for designing multi-format decoder

Abstract: Disclosed are a method of designing to transform an a single-format decoder, which performs image processing on an image signal using a first codec, to a multi-format decoder, which performs image processing on an image signal using a second codec, and an electronic apparatus thereof. The method includes processing first information about the first codec and second information about the second codec to produce a first transform coefficient associated with the first codec and a second transform coefficient associated with the second codec, through a predetermined transform method; determining a similarity between the first transform coefficient associated with the first codec and the second transform coefficient associated with the second codec; and converting the single-format decoder to the multi-format decoder by adding at least one of a shifter, an adder, and a subtractor, based on the determined similarity. Consequently, a reduction in chip size and cost may be achieved.