Topic
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.
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05 May 1996
TL;DR: This paper presents a modified multiband excitation (M/sup 2/BE) speech coding algorithm which can achieve high-quality synthesized speech at 4.8, 3.6, 2.4 and 1.2 kb/s.
Abstract: This paper presents a modified multiband excitation (M/sup 2/BE) speech coding algorithm which can achieve high-quality synthesized speech at 4.8, 3.6, 2.4 and 1.2 kb/s. The computer simulation results show that the synthesized speech is quite natural even at rate as low as 2.4 kb/s. This algorithm has been successfully implemented on a single TMS320C31 floating point processor and informal listening test demonstrates that the speech quality is quite good between 1.2-4.8 kb/s.
1 citations
05 Aug 2008
TL;DR: A single chip H.264/AVC codec LSI was newly developed for digital HD camcorder, which integrates a video codec, an audio codec, graphics, video interfaces, peripherals, a host processor, and an SDRAM interface.
Abstract: A single chip H.264/AVC codec LSI was newly developed for digital HD camcorder, which integrates a video codec, an audio codec, graphics, video interfaces, peripherals, a host processor, and an SDRAM interface. In order to realize low-power real-time receding and playing back system, the LSI uses a software-based codec architecture and a flexible SDRAM control architecture.
1 citations
12 Dec 2013
TL;DR: The results show that the codec bit rate and level of packet loss have a significant impact on the performance of speech recognition over IP.
Abstract: Nowadays, VoIP has become the core communications on the internet. One of the crucial applications on VoIP is the automated IVR system that interacts with the user automatically. Speech recognition plays an important role behind this kind of system. This paper studies the effect of codec bit rate and network packet loss on Thai speech recognition systems over an IP network. We encoded the speech samples of male, female and artificial voice with various bit rates of Speex codec. The speech sample was sent by RTP through the IP network with packet loss simulation. The speech quality was measured by PESQ and compared to word error rate of speech recognition. The results show that the codec bit rate and level of packet loss have a significant impact on the performance of speech recognition over IP.
1 citations
19 Apr 2015
TL;DR: This paper introduces a novel algorithm for improved transform coding of harmonic audio that does not deploy the conventional scheme of splitting the input signal into a spectrum envelope and a residual, but models the spectral peak regions.
Abstract: Audio coding of harmonic signals is a challenging task for conventional MDCT coding schemes. In this paper we introduce a novel algorithm for improved transform coding of harmonic audio. The algorithm does not deploy the conventional scheme of splitting the input signal into a spectrum envelope and a residual, but models the spectral peak regions. The presented coding scheme is part of the recently standardized 3GPP EVS codec.
1 citations
01 Jun 1998
TL;DR: A 1.9 kbps Zinc-function excited, wave-form interpolated speech codec is proposed and its bit error sensitivity is analysed.
Abstract: A 1.9 kbps Zinc-function excited, wave-form interpolated speech codec is proposed and its bit error sensitivity is analysed. This codec is incorporated in a standard GSM-like system, employing either convolutional or turbo coding. Ironically, the higher complexity turbo codec provides only a modest robustness gain over the standard convolutional code due to the short interleaver constraint imposed by the highly bandwidth-efficient speech codec.
1 citations