Compressed domain noise reduction and echo suppression for network speech enhancement

Chandran, Reshma; Marchok, D.J.

doi:10.1109/mwscas.2000.951575

Cited by 11 publications

(3 citation statements)

References 1 publication

Supporting

Mentioning

Contrasting

Order By: Relevance

“…Code-Excited Linear Prediction (CELP) is the most popular model in low bit-rate speech coding, and most of the research is focused on this model. R. Chandran proposed the first compressed domain method [1] in 2000. In this method, the intensity of noise is estimated with the aid of the codec's voice activity detector.…”

Section: Introductionmentioning

confidence: 99%

Compressed domain speech enhancement based on the joint modification of codebook gains

Xia

Bao

Liang

et al. 2011

2011 IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

View full text Add to dashboard Cite

A compressed domain speech enhancement method based on the joint modification of adaptive and algebraic codebook gains for the codec of ITU-T G.722.2 is proposed in this paper. First the power of excitation signal corresponding to the noise is estimated by the method of minimum statistics. Then the decision-directed approach is used to get an estimate of the a priori SNR. And the algebraic codebook gain is modified by multiplying a Wiener-type modification factor. In order to solve the problem of power loss in voiced segment, the modified adaptive codebook gain is got by keeping the power of the modified excitation signal equal to the scaled version of the noisy one. The result of performance evaluation under ITU-T G.160 shows that, in comparison with the method that only modifies the algebraic codebook gain, the proposed method could provide larger amount of noise reduction in both white and colored noise with smaller attenuation on the speech level, and the objective speech quality is improved evidently.

show abstract

Section: Introductionmentioning

confidence: 99%

Compressed domain speech enhancement based on the joint modification of codebook gains

Xia

Bao

Liang

et al. 2011

2011 IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

View full text Add to dashboard Cite

show abstract

“…However, the echo attenuation can be severely degraded by the non linearity and/or unpredictability of the effective acoustic echo path [2,3]. An alternative approach [4] involves modifying the coded parameters. In [5] a complete noise reduction system was implemented by modifying the fixed codebook gain of a CELP coder.…”

Section: Introductionmentioning

confidence: 99%

Acoustic Echo Cancellation Embedded in Smart Transcoding Algorithm between 3GPP AMR‐NB Modes

Fapi¹,

Pastor

Beaugeant³

et al. 2010

Journal of Electrical and Computer Engineering

View full text Add to dashboard Cite

Acoustic Echo Cancellation (AEC) is a necessary feature for mobile devices when the acoustic coupling between the microphone and the loudspeaker affects the communication quality and intelligibility. When implemented inside the network, decoding is required to access the corrupted signal. The AEC performance is strongly degraded by nonlinearity introduced by speech codecs. The Echo Return Loss Enhancement (ERLE) can be less than 10 dB for low bit rate speech codecs. We propose in this paper a coded domain AEC integrated in a smart transcoding strategy which directly modifies the Code Excited Linear Prediction (CELP) parameters. The proposed system addresses simultaneously problems due to network interoperability and network voice quality enhancement. The ERLE performance of this new approach during transcoding between Adaptive Multirate-NarrowBand (AMR-NB) modes is above 45 dB as required in Global System for Mobile Communications (GSM) specifications.

show abstract

“…Several works for processing speech codec parameters were reported for speech recognition, noise reduction and echo suppression [1][2][3]. In these applications, one of the most useful information is the estimated speech energy in frame or subframe level.…”

Section: Introductionmentioning

confidence: 99%

Frame energy estimation based on speech codec parameters

Kim¹,

Cao²,

Tarraf³

2008

2008 IEEE International Conference on Acoustics, Speech and Signal Processing

View full text Add to dashboard Cite

This paper proposes an efficient method for estimating frame energy of speech from Enhanced Variable Rate Coder (EVRC) bitstream for network-based speech processing applications in Transcoder Free Operation (TrFO) environments, where speech signals are represented as speech coding parameters. A frame of speech energy is decomposed into the energy of excitation and vocal tract filter, and the frame energy estimation method is derived for each component. Among many parameters of EVRC bitstream, the fixed codebook gain and adaptive codebook gain are used for the estimation of excitation energy, and Line Spectrum Pair (LSP) information is used to estimate the energy of vocal tract filter. Experimental results demonstrated the novelty of the proposed method. The correlation coefficient between the actual and estimated frame energy can be maintained at a value of 0.994 with just 5% multiplicative operations of full decoding.

show abstract

Compressed domain noise reduction and echo suppression for network speech enhancement

Cited by 11 publications

References 1 publication

Compressed domain speech enhancement based on the joint modification of codebook gains

Compressed domain speech enhancement based on the joint modification of codebook gains

Acoustic Echo Cancellation Embedded in Smart Transcoding Algorithm between 3GPP AMR‐NB Modes

Frame energy estimation based on speech codec parameters

Contact Info

Product

Resources

About