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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24 May 1999
TL;DR: A new test-bench developed by Telital S.p.A. is presented, in co-operation with the Electronics Department of the University of Trieste, in order to evaluate the quality of the speech coding and decoding algorithms for the newest generation of GSM terminals.
Abstract: This paper presents a new test-bench developed by Telital S.p.A., in co-operation with the Electronics Department of the University of Trieste, in order to evaluate the quality of the speech coding and decoding algorithms for the newest generation of GSM terminals. A Data Acquisition System (DAS) has been implemented to obtain, in addition to subjective speech quality measurements (as usually in literature), also objective speech quality measurements. In the paper, some important features of GSM speech codecs are highlighted and summarised through the analysis of the above mentioned objective quality measurements. These measurements have been performed on the Telital GSM terminal in different simulated fading conditions. This allows one to test not only the efficiency of the particular channel "coding+interleaving" scheme employed for different carrier-to-interference (C/I) conditions, but also the efficiency of such scheme for different fading rates.
3 citations
03 Apr 2012
TL;DR: It has been shown that the loss due to GSM-AMR codec is very significant for speaker verification compared to undecoded speech, though the packet loss and bit rate may degrade the quality of speech but it is not significant to detection of speaker's identity.
Abstract: Automatic Speaker Verification (ASV) is a challenging task over the mobile/IP based system as the coding introduces some loss in system performance This paper reports on the work in progress to examine the impact of GSM-AMR codec used in mobile at its various bit rates and G729 codec for VoIP, along with different kind of noise and packet loss scenario for the speech signal PURE YOHO database has been used for the evaluation of this task Respective encoder and decoders are used back to back on wideband clean microphone speech to simulate the real-life situation Evaluation of performance is done through the measurement of Equal Error Rate (EER) It has been shown that the loss due to GSM-AMR codec is very significant for speaker verification compared to undecoded speech Though the packet loss and bit rate may degrade the quality of speech but it is not significant to detection of speaker's identity
3 citations
Proceedings Article•
01 Sep 2007TL;DR: A flexible framework is presented which performs multiple-description coding of speech signals with two or more channels without excessive increase in complexity through the use of forward error correction codes and a layered speech codec.
Abstract: A flexible framework is presented which performs multiple-description coding of speech signals with two or more channels. The use of forward error correction codes together with a layered speech codec permits encoding into more than two descriptions without excessive increase in complexity. Results of a formal MOS listening test reveal considerable improvements in robustness as long as base layer quality and the number of descriptions are chosen appropriately. A modification of the original encoding scheme allows trading off bit rate savings against robustness to extreme channel conditions. Different coding schemes can easily be compared using a real-time demonstrator software.
3 citations
01 Jan 2011
TL;DR: A system for the transmission of binaural wideband speech signals over a standard telephone network is proposed, which is backwards compatible with the 3GPP Adaptive Multirate (AMR) codec.
Abstract: A system for the transmission of binaural wideband speech signals over
a standard telephone network is proposed. It is backwards compatible with the
(single channel and narrowband) 3GPP Adaptive Multirate (AMR) codec. The
required information about the source location and for audio bandwidth extension is
transmitted over a steganographic communication channel that is embedded within
the bitstream of the AMR codec. A legacy receiver can still decode the single
channel narrowband signal without noticeable quality loss.
3 citations
30 Jun 2014
TL;DR: Simulation results show that AMR-WB ITU-T G.722.2 based Arnold cat Map encryption is very efficient since the encrypted speech is similar to a white noise.
Abstract: Speech encryption is becoming more and more essential as the increasing importance of multimedia applications and mobile telecommunications However, multimedia encryption and decryption are often computationally demanding and unpractical for power-constrained devices and narrow bandwidth environments In this paper an encryption scheme for AM-WB ITU-T G 7222 speech based Arnold cat Map is presented analyzed and evaluated using objective and subjective tests for the 8 modes of the AMR-WB ITU-T G7222 Simulation results show that AMR-WB ITU-T G7222 based Arnold cat Map encryption is very efficient since the encrypted speech is similar to a white noise The perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD) tests for speech speech extracted from TIMIT database confirm the efficiency of the presented scheme
3 citations