• Journal of rehabilitation welfare engineering & assistive technology
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• v.11 no.2
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• pp.143-146
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• 2017

In this paper, a novel approach is proposed to improve the performance of speech/music classification using the recurrent neural network (RNN) in the enhanced voice services (EVS) of 3GPP for hearing aids. Feature vectors applied to the RNN are selected from the relevant parameters of the EVS for efficient speech/music classification. The performance of the proposed algorithm is evaluated under various conditions and large speech/music data. The proposed algorithm yields better results compared with the conventional scheme implemented in the EVS.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1E
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• pp.83-87
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• 1995

The implementation of the voice dodec is imjplemented by using TMS320C30, which is the floating point DSP chip from Texas Instrument. QCELP (Qualcomm Code Excited Linear Prediction) is used to encode and decode the voice. The QCELP code is implemented by the TMS320C30 C-dode. The DSP board is controlled by the PC. The PC program tranfors the voice file from and to the DSP board, which is also implemented by C-code. The voice is encoded by the DSP board and the encoded data is transferred to PC to be stored as a file. To hear the voice. the voice data file is sent to DSP board and decoded to synthesize audible voice. Two flags are used by both programs to notify the status of the operation. By checking the flags, DSP and PC decides when the voice data is transferred between them.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.6
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• pp.1327-1332
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• 2015

In this paper, we studied several aspects that affect the quality of VoIP using WiFi network. It's clear that the background data traffic within an AP, the end-to-end delay and the traffic loss of TCP/IP network gives serious effects on the voice quality. A kind of access control for the VoIP connection within an AP should be done for the acceptable voice quality.

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.295-298
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• 2003

In this paper, a novel maximum a posterion (MAP) estimation for the channel decoding of H.264 codes in the presence of transmission error is presented. Arithmetic codes with a forbidden symbol and trellis search techniques are employed in order to estimate the best transmitted. And, there has been growing interest of communication, the research about transmission of exact data is increasing. Unlike the case of voice transmission, noise has a fatal effect on the image transmission. The reason is that video coding standards have used the variable length coding. So, only one bit error affects the all video data compressed before resynchronization. For reasons of that, channel needs the channel codec, which is robust to channel error. But, usual channel decoder corrects the error only by channel error probability. So, designing source codec and channel codec, Instead of separating them, it is tried to combine them jointly. And many researches used the information of source redundancy In received data. But, these methods do not match to the video coding standards, because video ceding standards use not only one symbol but also many symbols in same data sequence. In this thesis, We try to design combined source-channel codec that is compatible with video coding standards. This MAP decoder is proposed by adding semantic structure and semantic constraint of video coding standards to the method using redundancy of the MAP decoders proposed previously. Then, We get the better performance than usual channel coder's.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• The Journal of Engineering Research
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• v.3 no.1
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• pp.43-46
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• 1998

Considerable research in the last three decades has examined the problem of enhancement of speech degraded by additive background noise. We compare traditional methods such as spectral subtraction and Wiener filter, recently proposed psychoacoustic model based methods such as perceptual filter and noise suppression in EVRC in terms of performance and complexity.

• Proceedings of the KSPS conference
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• 2003.10a
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• pp.109-112
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• 2003

In real time packetized voice applications, missing frames is a major source of voice quality degradation. Thus packet loss concealment(PLC) algorithms are needed to guarantee the QoS of the VoIP. Still current speech codecs for VoIP work poor when consecutive packet losses are issued. In this paper, we proposed a new PLC algorithm for the G.729 codec. Our algorithm works better especially when the consecutive packet loss occurs mainly because it adopts an adaptive gain controller utilizing the number of missing packet information combined with a fixed codebook vector estimation algorithm and LPC bandwidth expansion.

• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.21-24
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• 2001

Bluetooth wireless technology is a publicly available specification proposed for Radio Frequency (RF) communication for short-range and point-to-multipoint voice and data transfer. It operates in the 2.4GHz ISM(Industrial, Scientific and Medical) band and offers the potential for low-cost, broadband wireless access for various mobile and portable devices at range of about 10 meters. In this paper, we describe the structure and the test results of the bluetooth baseband module we have developed. This module has a UART interface for HCI and a audio codec for voice. The interface between controller and this module supports common control interface. An FPGA implementation of this module is tested for file and bit-stream transfers between PCs.