• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.413-417
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• 2009

The use of scalable video coding scheme has been regarded as a promising solution for guaranteeing the quality of service of the video streaming over the Internet because it is a capable coding scheme to perform quality adaptation depending on network conditions. In this paper, we use a streaming model that transmits base layer using TCP and enhancement layers using DCCP, which try to provide transmission reliability of the BL and TCP friendliness. Unlike pervious works, the proposed algorithm performs rate adaptation based on playout buffer status. The PoB status of the client is sent back periodically to the server and serves as a network congestion indicator. Experimental results show that our scheme improves streaming quality comparing with pervious scheme in the case of not only constant/dynamic background flows but also VBR-encoded video sequence.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6C
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• pp.423-433
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• 2005

We present power(in frequency domain) and rate adaptation(in time domain) schemes in multicarrier (MC) direct-sequence code-division multiple-access(DS/CDMA) communications. Utilizing channel state information from the receiver, the adaptation schemes allocate power the user's sub-band with the largest channel gain. In the time domain, the transmission data rate is adapted for a desired transmission quality. In the case of single-user channels, a closed-form expression is derived for an optimal time domain power adaptation that minimizes the average bit error rate(BER). Channel inversion power adaptation is found to provide nearly optimal performance in this case, as the number of sub-bands or available average transmission power increase. Analysis and simulation results show the BER performance of the proposed power and rate adaptations with fixed average transmission power significantly improves the performance over the power allocation in the frequency domain only. Also, we compare the performance of the proposed power and rate adaptation schemes in MC-DS/CDMA systems with that of power and rate adapted single carrier DS/CDMA systems with RAKE receiver.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.5
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• pp.765-776
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• 1995

This paper describes a speaker adaptation method to improve the recognition performance of MLP(multiLayer Perceptron) based HMM(Hidden Markov Model) speech recognizer. In this method, we use lst-order linear transformation network to fit data of a new speaker to the MLP. Transformation parameters are adjusted by back-propagating classification error to the transformation network while leaving the MLP classifier fixed. The recognition system is based on semicontinuous HMM's which use the MLP as a fuzzy vector quantizer. The experimental results show that rapid speaker adaptation resulting in high recognition performance can be accomplished by this method. Namely, for supervised adaptation, the error rate is signifecantly reduced from 9.2% for the baseline system to 5.6% after speaker adaptation. And for unsupervised adaptation, the error rate is reduced to 5.1%, without any information from new speakers.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.539-544
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• 2003

This paper describes how to reduce the effect of an occupation threshold by that the transform of mixture components of HMM parameters is controlled in hierarchical tree structure to prevent from over-adaptation. To reduce correlations between data elements and to remove elements with less variance, we employ PCA (Principal component analysis) and ICA (independent component analysis) that would give as good a representation as possible, and decline the effect of over-adaptation. When we set lower occupation threshold and increase the number of transformation function, ordinary MLLR adaptation algorithm represents lower recognition rate than SI models, whereas the proposed MLLR adaptation algorithm represents the improvement of over 2% for the word recognition rate as compared to performance of SI models.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.8
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• pp.680-687
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• 2013

We investigate a generalized combined power and rate adaptation scheme in direct-sequence (DS) code-division multiple-access (CDMA) communications over Nakagami fading channels. The transmission power allocated to user i is proportional to $G^p_i$, where $G_i$ is the channel gain of user i and p is a real number, and the data rate (i.e., spreading gain) is jointly adapted so that a desired QoS is maintained. We analyze the average data rate of the proposed adaptation scheme subject to fixed average and peak transmission power constraints. Our results show that the proposed joint adaptation scheme provides a significant performance improvement over power-only and rate-only adaptation.

• Phonetics and Speech Sciences
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• v.7 no.3
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• pp.131-138
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• 2015

In this paper, we propose L1-norm regularization for state vector adaptation of subspace Gaussian mixture model (SGMM). When you design a speaker adaptation system with GMM-HMM acoustic model, MAP is the most typical technique to be considered. However, in MAP adaptation procedure, large number of parameters should be updated simultaneously. We can adopt sparse adaptation such as L1-norm regularization or sparse MAP to cope with that, but the performance of sparse adaptation is not good as MAP adaptation. However, SGMM does not suffer a lot from sparse adaptation as GMM-HMM because each Gaussian mean vector in SGMM is defined as a weighted sum of basis vectors, which is much robust to the fluctuation of parameters. Since there are only a few adaptation techniques appropriate for SGMM, our proposed method could be powerful especially when the number of adaptation data is limited. Experimental results show that error reduction rate of the proposed method is better than the result of MAP adaptation of SGMM, even with small adaptation data.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2A
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• pp.128-133
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• 2010

We investigate the optimal chip rate of power or rate adapted direct-sequence code division multiple access (DS/CDMA) communication systems in Nakagami fading channels. We find that the optimal chip rate that maximizes the spectral efficiency depends upon both the channel parameters, such as multipath intensity profile (MIP) and line-of-sight (LOS) component, and the adaptation scheme itself. With the rate adaptation, the optimal chip rate is less than $1/T_m$, irrespective of the channel parameters, where $1/T_m$ is multipath delay spread. This indicates that with the rate adaptation, correlation receiver achieves higher spectral efficiency than RAKE receiver. With the power adaptation, however, the optimal chip rate and the corresponding number of tabs in RAKE receiver are sensitive to MIP and LOS component.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.1
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• pp.29-34
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• 2012

Wireless multicast service can be used for video streaming service to save the network resources by sending the same popular multimedia contents to a group of users at once. For better multimedia streaming multicast service, we propose a quality-aware rate adaptation (QARA) scheme for scalable video multicast in rate adaptive wireless networks. In QARA, transmission rate is determined depending on the content's type and users' channel conditions. First, the base layer is transmitted by a low rate for high reliability. That means we provide basic service quality to all users. On the contrary, the transmission rate for enhancement layer is adapted by using channel condition feedback from a randomly selected node. So, the enhancement layer frames in a multimedia content is sent with various transmission rates. Therefore, each node can be provided with differentiated quality services. Consequently, QARA is capable of serving heterogeneous population of mobile nodes. Moreover, it can utilize network resources more efficiently. Our simulation results show that QARA outperforms utilization of the available transmission rate and reduces the data transmission time.

• International journal of advanced smart convergence
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• v.7 no.4
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• pp.92-100
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• 2018

This paper presents a link adaptation scheme for adaptive full duplex (AFD) systems. The signal modulation levels and communication link patterns are adaptively selected according to the changing channel conditions. The link pattern selection process consists of two successive steps such as a transmit-receive antenna pair selection based on maximum sum rate or minimum maximum symbol error rate, and an adaptive modulation based on maximum minimum norm. In AFD systems, the antennas of both nodes are jointly determined with modulation levels depending on the channel conditions. An adaptive algorithm with relatively low complexity is also proposed to select the link parameters. Simulation results show that the proposed AFD system offers significant bit error rate (BER) performance improvement compared with conventional full duplex systems with perfect or imperfect self-interference cancellation under the same fixed sum rate.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.268-272
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• 2009

In this paper, we propose a Gaussian Mixture Model (GMM) based incremental robust adaptation with a forgetting factor for the speaker verification. Speaker recognition system uses a speaker model adaptation method with small amounts of data in order to obtain a good performance. However, a conventional adaptation method has vulnerable to the outlier from the irregular utterance variations and the presence noise, which results in inaccurate speaker model. As time goes by, a rate in which new data are adapted to a model is reduced. The proposed algorithm uses an incremental robust adaptation in order to reduce effect of outlier and use forgetting factor in order to maintain adaptive rate of new data on GMM based speaker model. The incremental robust adaptation uses a method which registers small amount of data in a speaker recognition model and adapts a model to new data to be tested. Experimental results from the data set gathered over seven months show that the proposed algorithm is robust against outliers and maintains adaptive rate of new data.