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.

Simulasi global system for mobile communication adaptive multi rate speech codec

Abstract: As the wireless technology continuing developing enormously, within it the development of speech signal compression joining too, which is implemented in digital communication system. Among the aim is privileged to minimize the channel capacity or bandwidth usage and the speed of transmission. So the use of smaller bits, which brings maximised voice quality, are the most important factor to this technology. One of the speech signal compression techniques which is implemented in GSM network is called Adaptive Multi Rate (AMR) speech codec or known as GSM-AMR speech codec. There are 8 (eight) bit rates 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbps which is used by GSM-AMR. The bit rates have abilities to adapt to the situation and condition of the network and also ability to tailor the operator needs and requirement. The result of a non real-time simulation is shown that GSM-AMR has a good enough quality of speech signal reconstruction with range of SNR between 18.78 and 20.35 dB and also range of MSE between 0.053 and 0.079. The smaller the bitrate, the better the SNR and the MSE. It was also proven with the comparison of original signal and constructed signal which is shown that the reconstructed signal has similarities figure with the original signal. Overall, the performance of the speech codec gives a good speech quality. Keyword: Speech Codec, GSM-AMR, Adaptive Multi Rate, SNR, MSE.

Turbo-detected unequal protection audio and speech transceivers using serially concantenated convolutional codes, trellis coded modulation and space-time trellis coding

Abstract: The MPEG-4 TwinVQ audio codec and the AMR-WB speech codec are investigated in the context of a jointly optimised turbo transceiver capable of providing unequal error protection. The transceiver advocated consists of serially concatenated Space-Time Trellis Coding (STTC), Trellis Coded Modulation (TCM) and two different-rate Non-Systematic Convolutional codes (NSCs) used for unequal error protection. A benchmarker scheme combining STTC and a single-class protection NSC is used for comparison with the proposed scheme. The audio and speech performance of both schemes is evaluated, when communicating over uncorrelated Rayleigh fading channels. An $E_b/N_0$ value of about 2.5 (3.5)~dB is required for near-unimpaired audio (speech) transmission, which is about 3.07 (4.2)~dB from the capacity of the system.

A progressive approach for perceptual audio coding

Abstract: A progressive approach to perceptual audio coding is presented in this paper. In contrast with existing perceptual audio coding schemes where the digital audio signal is encoded at a fixed bit rate, the progressive approach achieves fully embedded audio coding whose bit rate can be controlled dynamically, making it useful in audio networking applications. In the proposed scheme, efficient successive quantization and bit layer coding are applied for the compression of an audio signal representation obtained from traditional subband decomposition. An improved successive approximation quantization (ISAQ) is introduced to perform bit allocation and quantization based on the psychoacoustic analysis of input audio signals.