An Adaptive Forward Error Control Method for Voice Communication

Wang, Lizhong; Muqing, Wu; Lulu, Wei; Li, Mojia

doi:10.1109/icnds.2010.5479338

Cited by 7 publications

(3 citation statements)

References 4 publications

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“…Considering that the primary objective of FEC is to reduce packet loss as perceived by application, packet loss rate (Ppl) after reconstruction was used to decide the required redundancy level in seminal works, such as Bolot and Vega-García [1996], and in later works, such as Johansson et al [2002] and Lizhong et al [2010]. However, Padhye et al [2000] warn that packet loss "may not be representative under all network conditions," such as during burst losses, when many consecutive packets are lost.…”

Section: Forward Error Correction (Fec) Controlmentioning

confidence: 99%

Survey on application-layer mechanisms for speech quality adaptation in VoIP

Carvalho

Mota

2013

ACM Comput. Surv.

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VoIP calls are sensitive to several impairments, such as delay and packet loss. One way to overcome these problems is by adaptively adjusting application-layer parameters to keep a minimum speech quality level. At the heart of self-adaptive systems lies a feedback loop, which consists of four key activities: monitoring, analysis, planning, and execution. Nevertheless, the existing adaptive approaches to QoS control of VoIP do not explicitly exhibit this feedback loop. Bringing it to surface can help developers in designing more robust and human-independent VoIP systems. This survey presents a comprehensive review of the current state-of-the-art research on speech quality adaptation of VoIP systems at the application layer and some research challenges on this subject.

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Section: Forward Error Correction (Fec) Controlmentioning

confidence: 99%

Survey on application-layer mechanisms for speech quality adaptation in VoIP

Carvalho

Mota

2013

ACM Comput. Surv.

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“…Ses iletiminde SNR değerinin düşük olması, hem ses kalitesini hem de haberleşme menzilini negatif yönde etkilemektedir. FEC koruması altındaki veriler gürültülü ortamlarda bozularak iletilse dahi kullanılan FEC tarafından düzeltilerek kayıplar engellenebilir [9]. Dolayısıyla FEC optimizasyonu, paket kayıplarını azaltmada ve ses kalitesini arttırmada büyük önem arz etmektedir [10,11].…”

Section: Gi̇ri̇ş (Introduction)unclassified

ETSI TS 102 361 sayısal mobil telsiz standardına uyumlu telsizlerin ses kalitesinin ses şiddeti optimizasyonu ve daha iyi ileri hata düzeltme planı kullanılarak arttırılması

Yetik

Ekşim

2018

Gazi Üniversitesi Mühendislik-Mimarlık Fakültesi Dergisi

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Voice quality and intelligibility enhancement in digital mobile radios • Performance improvement of advanced multi-band excitation speech codec • Voice volume optimization Block diagram of our work is shown below. Purpose: Our purpose in this work is to enhance voice quality and provide deterministic voice quality in all conditions to increase intelligibility and clearness of the transmitted sound in digital radios. Theory and Methods: We proposed and presented two different improvements that are easy to implement and are isolated from the protocol itself and mainly focused on vocoder utilized in the radios. Primary proposition is to replace the block code based forward error correction with a better error correction schemes like convolution codes. Thus, hybrid of Golay and 5/6 punctured convolution codes are utilized instead of using solely block codes. Other proposed method is based on voice volume optimization using vocoder features like gain parameters. Smoothed and hard limited voice gain parameters offer a clear and understandable voice. Results: In this work, we achieved 2.95 dB coding gain with the help of proposed forward error correction. Also, smooth and clear voice reconstruction is guaranteed by volume optimization and voice gain smoothing. Conclusion: Proposed volume optimization and voice gain balancing help the reconstructed voice to have better dynamic range, increased intelligibility and more natural representation of the source voice. By replacing pure block codes with the hybrid of block and convolutional code, we achieved 2.95 dB coding gain for a BER value of Pb=10-5 with respect to the AMBE FEC. Due to the fact that proposed improvements are vocoder oriented and isolated from protocol, they can be applied to any newer and already manufactured digital mobile radios to enhance voice quality. Not only achieved coding gain offers better communication range and voice quality but also it may extend battery life of radios.

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“…These methods are typically classified into either sender-based schemes or receiver-based schemes. Senderbased schemes consist of a collection of packet loss protection methods that provide error-robust transmission methods such as interleaving, forward error correction (FEC), and redundant speech transmission (RST) [8][9][10][11][12]. On the other hand, receiver-based schemes consist of a collection of packet loss concealment (PLC) methods that compensate for lost speech signals using substitutable signals, for example, silence, previous good speech, or regenerated speech according to the analysis-by-synthesis criterion [13][14][15][16].…”

Section: Introductionmentioning

confidence: 99%

Adaptive Speech Streaming Based on Speech Quality Estimation and Artificial Bandwidth Extension for Voice over Wireless Multimedia Sensor Networks

Kang

Park

Kim

et al. 2015

International Journal of Distributed Sensor Networks

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In this paper, an adaptive speech streaming method is proposed to improve the perceived speech quality (PSQ) of voice over wireless multimedia sensor network (WMSNs). First of all, the proposed method estimates the PSQ of the received speech data under different network conditions that are represented by the packet loss rates (PLRs). Simultaneously, the proposed method classifies the speech signal as either an onset or a nononset frame. Based on the estimated PSQ and the speech class, it determines an appropriate bit rate for the redundant speech data (RSD) that are transmitted with the primary speech data (PSD) to help reconstruct the speech signals of any lost frames. In particular, when the estimated PLR is high, the bit rate of the RSD should be increased by decreasing that of the PSD. Thus, the bandwidth of the PSD is changed from wideband to narrowband, and an artificial bandwidth extension technique is applied to the decoded narrowband speech. It is shown from the simulation that the proposed method significantly improves the decoded speech quality under packet loss conditions in a WMSN, compared to a decoder-based packet loss concealment method and a conventional redundant speech transmission method.

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An Adaptive Forward Error Control Method for Voice Communication

Cited by 7 publications

References 4 publications

Survey on application-layer mechanisms for speech quality adaptation in VoIP

Survey on application-layer mechanisms for speech quality adaptation in VoIP

ETSI TS 102 361 sayısal mobil telsiz standardına uyumlu telsizlerin ses kalitesinin ses şiddeti optimizasyonu ve daha iyi ileri hata düzeltme planı kullanılarak arttırılması

Adaptive Speech Streaming Based on Speech Quality Estimation and Artificial Bandwidth Extension for Voice over Wireless Multimedia Sensor Networks

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