GSM enhanced full rate speech codec

Järvinen, Kari; Vainio, J.; Kapanen, Pekka; Honkanen, T.; Haavisto, P.; Salami, R.; Laflamme, C.; Adoul, J.-P.

doi:10.1109/icassp.1997.596038

Cited by 46 publications

(20 citation statements)

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“…Considering the example of a cellular phone as the client, the common speech codec used is the GSM Enhanced Full-Rate (GSM-EFR) codec, which has a computational complexity of approximately 18 Mflops and a static memory requirement of 5900 words (16-bit) (Jarvinen et al, 1997). MFCC feature extraction requires at most 10 Mflops or less (Adami et al, 2002).…”

Section: Bitrate Scalability Computational Complexity and Memory Reqmentioning

confidence: 99%

Scalable distributed speech recognition using Gaussian mixture model-based block quantisation

Paliwal

2006

Speech Communication

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Section: Bitrate Scalability Computational Complexity and Memory Reqmentioning

confidence: 99%

Scalable distributed speech recognition using Gaussian mixture model-based block quantisation

Paliwal

2006

Speech Communication

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“…An AMR speech codec utilises the Algebraic Code Excitation Linear Prediction (ACELP) algorithm that has also been employed in GSM EFR [9] and D-AMPS EFR [10]. This adaptive coder is also referred to as the Multi-Rate ACELP (MR-ACELP) [11] coder.…”

Section: Introductionmentioning

confidence: 99%

Priority-aware extended eOCSA frame packing algorithm evaluation in WiMAX systems with mixed VoIP and data traffic

Bahmani

Ince

2013

2013 IEEE 20th Symposium on Communications and Vehicular Technology in the Benelux (SCVT)

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Recently developed mobile broadband wireless systems can provide high-quality multimedia services with required levels of QoS. Currently most 4 ℎ generation wireless communication systems employ the Orthogonal Frequency Division Multiple Access (OFDMA) technology in the downlink (DL) part of their physical layer. In this paper we present a simulation-based performance analysis of the Priority-Aware Extended eOCSA (PA-EeOCSA) DL frame packing algorithm under mixed VoIP and uncorrelated data traffic. For VoIP traffic, the ON-SID model has been used and the packing efficiency of both the eOCSA and PA-EeOCSA algorithms have been compared. Simulation results point out that in terms of mean percentage of unallocated bursts and the number of slots allocated on average, the PA-EeOCSA algorithm performs more efficiently in comparison to eOCSA.Results also show that with data users fixed at 20, between 40-45 VoIP users can be admitted if the mean percentage of unallocated bursts is to be maintained below 3.0 %. Recall that, a low mean percentage of unallocated bursts is imperative in order to keep the overall packet loss rate below a tolerable limit. Keywords-Voice over IP(VoIP); ON-SID traffic model;OFDMA; PA-EeOCSA frame packing algorithm; QoS; Mean opinion score.

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“…Many variations of CELP coders have been standardized including [1,19] G.723.1 operating at 6.3/5.3 kbit/s, G.729 operating at 8 kbit/s, G.728 a low delay coder operating at 16 kbit/s and all the digital mobile telephony encoding standards including [20,21,22,23] GSM, IS-54, IS-95 and IS-136. Figure 1.1 shows a simple block diagram of a CELP coder.…”

Section: Hybrid Codersmentioning

confidence: 99%

“…It is difficult to make valid comparisons with existing coders as, to our knowledge, there is no other low rate coder accommodating both speech and music inputs. However, we have compared the quality of the coded signals using NPAC, the RealAudio 6 music coder operating at 8 kbit/s, the RealAudio speech coder operating at 8.5 kbit/s and the G.729 speech coder [20] operating at 8 kbit/s. The quality of the coded signals were evaluated through informal tests.…”

Section: Subjective Evaluationmentioning

confidence: 99%

“…This procedure will go on until the desired number of predictor matrices are found. For each subset of the training set (corresponding to a predictor matrix), we find the optimal predictor matrix through the following optimization procedure: 20) where R j contains the time indexes of the vectors belonging to the jth region. Note that, in order to perform the optimization, we need to have the quantized vectors.…”

Section: Design Of the Predictor Matricesmentioning

confidence: 99%

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Perceptual coding of narrowband audio signals at 8 kbit/s

Najafzadeh-Azghandi

Kabal²

1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speec

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New applications such as Internet broadcast and communications, consumer multimedia products, digital AM broadcast and satellite networks are emerging. Those applications require moderate audio quality without annoying artifacts at bit rates below 16 kbit/s. Although speech coders provide high speech quality at bit rates around 8 kbit/s, they perform poorly when encoding audio signals. In this thesis, we present a novel transform coding paradigm based on the characteristics of the human hearing system. The proposed encoder, i.e., Narrowband Perceptual Audio Coder (NPAC), can accommodate a wide range of narrowband audio inputs without annoying artifacts at bit rates down to 8 kbit/s.NPAC employs a variety of algorithms to remove the perceptually irrelevant parts and statistical redundancies of the input signal. The new algorithms used in NPAC include a perceptual error measure in training the codebooks and selecting the best codewords, perceptually-based bit allocation algorithms and an adaptive predictive scheme to vector quantize the scale factors.The proposed encoder has moderate complexity and delivers good quality for narrowband audio inputs at around 1 bit/sample. Informal subjective tests have been conducted to compare the performance of NPAC with an 8 kbit/s commercially-available audio coder.The tests results show that NPAC performs better for both music and speech inputs. iii RésuméDes nouvelles technologies telles que la diffusion par Internet, la diffusion AM numérique, et les réseaux satellites deviennent de plus en plus populaires et constituent la base de plusieurs nouvelles applications et produits multimédias. La réussite de ces produits sur la marché dépend de la qualité des signaux audio et vidéo ainsi que de la largeur de bande utilisée. Pour le signal audio, il est désirable que le débit soit en bas de 16 kbit/s tout en offrant une qualité acceptable, c'est-à-dire sans de distorsion remarquable.Il està noter que certains codeurs de parole permettent de transmettre le signal de parole au débit de 8 kbit/s avec une très bonne qualité. Toutefois, puisque ces codeurs profitent de la structure particulière de la parole, ils ne peuvent pas offrir la même qualité audio pour d'autres signaux comme la musique.Dans cette thèse, nous présentons une philosophie d'encodage des signaux audio qui tient compte de la structure du système auditif. Le codeur proposé se nomme Codeur Audio Perceptuelà bandeÉtroite (CAPE). CAPE permet d'encoder plusieurs types de signal audioà bandeétroite au débit de 8 kbit/s sans de distorsion remarquable.Plusieurs nouveaux algorithmes sont utilisés dans CAPE afins d'éliminer la redondance statistique ainsi que la partie sans importance perceptuel du signal d'entrée. Parmi les nouveautés de CAPE, il y a une mesure d'erreur perceptuelle qui est utilisée lors de l'entraînement des tableaux de quantification, et pour la sélection du meilleur vecteur de ces tableaux lors de l'encodage. De plus, l'allocation des bits pour les gains du spectre dans différentes bandes de fréque...

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GSM enhanced full rate speech codec

Cited by 46 publications

References 1 publication

Scalable distributed speech recognition using Gaussian mixture model-based block quantisation

Scalable distributed speech recognition using Gaussian mixture model-based block quantisation

Priority-aware extended eOCSA frame packing algorithm evaluation in WiMAX systems with mixed VoIP and data traffic

Perceptual coding of narrowband audio signals at 8 kbit/s

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