• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.293-303
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• 2002

This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.101-104
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• 2004

음성속도변환(TSM : Time Scaling Modification) 알고리즘은 시간축에서 음성 신호의 속도를 변환할 수 있는 방식이다. 이러한, 방법으로는 OLA(Overlap Add), SOLA (Synchronized Overlap Add) 알고리즘 등이 연구 되어 왔다. 2 가지 방식 중에도 동기화를 시켜 overlap 을 시키는 SOLA 알고리즘이 OLA 방법에 비해 음질이 우수하다. 본 논문에서는 TMS320C5416 DSP 에 계산량이 감소된 SOLA 알고리즘을 실시간 구현하였다. 기존의 SOLA 알고리즘에서 동기화를 위해 사용하고 있는 cross-correlation 함수는 곱셈연산에서 발생하는 bit 의 dynamic range 가 커서 나눗셈 연산에서도 과도한 연산량을 필요로 한다. 따라서 이러한 계산량의 감소를 위해 기존의 cross-correlation 함수가 대신 더하기와 빼기의 연산으로 수행되는 NAMDF 함수를 사용하여 계산량을 줄였다. 제안한 방법을 SOLA 알고리즘에 적용하여 성능 평가를 실시하였다. TMS320C5416 DSP 에 실시간으로 실험한 결과 NAMDF 함수를 사용하였을 경우 음질의 저하가 거의 없었으며, 계산량을 기존의 cross-correlation 방식에 비해 6.22MIPS 가까이 감소시킬 수 있었다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.84-89
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• 2003

In this paper, we real-time implemented to the TMS320C5416 the vocoder of variable bit rate applied the SOLA-B algorithm by Henja to the ITU-T G.729A vocoder of 8kbps transmission rate. This proposed method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. At this time, we bandied that the interval of cross correlation function if skipped every 3 sample for decreasing the computational amount of SOLA-B algorithm. The real-time implemented vocoder of C.729A and SOLA-B algorithm is represented the complexity of maximum that is 10.2MIPS in encoder and 2.8MIPS in decoder of 8kbps transmission rate. Also, it is represented the complexity of maximum that is 18.5MIPS in encoder and 13.1MIPS in decoder of 6kbps, it is 18.5MIPS in encoder and 13.1MIPS in decoder of 4kbps. The used memory is about program ROM 9.7kwords, table ROM 4.5kwords, RAM 5.1 kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, for evaluation of speech quality of the vocoder of real-time implemented variable bit rate, it is estimated the MOS score of 3.69 in 4kbps.

• Speech Sciences
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• v.10 no.3
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• pp.241-250
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• 2003

In this paper, we implemented the vocoder of variable rate by applying the SOLA-B algorithm to the G.729A to the TMS320C5416 in real-time. This method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. But the method applied to the existed G.729A and SOLA-B algorithm is caused the loss of speech quality in G.729A which is not reflected about length variation of speech. Therefore the proposed method is encoded according as it is modified the structure of LSP quantization table about the length of speech is reduced by using the SOLA-B algorithm. The vocoder of variable rate by applying the G.729A and SOLA-B algorithm is represented the maximum complexity of 10.2MIPS about encoder and 2.8MIPS about decoder in 8kbps transmission rate. Also it is evaluated 17.3MIPS about encoder, 9.9MIPS about decoder in 6kbps and 18.5MIPS about encoder, 11.1MIPS about decoder in 4kbps according to the transmission rate. The used memory is about program ROM 9.7kwords, table ROM 4.69kwords, RAM 5.2kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, the result of MOS test for evaluation of speech quality of the vocoder of variable rate which is implemented in real-time, it is estimated about 3.68 in 4kbps.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.11-19
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• 2000

This research deals with the implementation procedures of a chorus processing DSP for karaoke system. It is necessary to compress the chorus data to store as many choruses as we can. We apply MPEG-1 audio algorithm to compress the chorus data. And the chorus system must be accompanied with the karaoke that can change the key and the tempo. So the chorus DSP must be able to change the key and tempo of the chorus data. We apply SOLA (Synchronized Overlap and Add) to do it. We designed the chorus DSP that can compress the chorus, change the key and tempo. And we verified the chorus DSP logic using FPGA. The used FPGA are two FLEX10K100s made by ALTERA. Finally we make the ASIC chip of chorus DSP and verify its operation.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.207-209
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• 2000

인터넷전화 서비스는 저렴한 가격과, 타 서비스와 통합 및 가치부가(Value Added)면에서 기존의 전화에 비해 많은 장점을 가지고 있으나, 상대적으로 낮은 음질로 인하여 사용자의 요구를 만족시키지 못하고 있다. 이것은 현재 인터넷은 best-effort형 패킷 전달 서비스만을 제공하고 있기 때문에 전송지연, 패킷손실, 지터등을 보장할 수 있는 방법이 없기 때문이다. 본 논문에서는 인터넷전화에서 패킷손실이나 전송지연으로 인한 음질 저하문제를 SOLA 알고리즘을 이용해 보완하였다. 제시된 알고리즘에서는 송신측에서 패킷을 전송하면 수신측에서는 수신 패킷에 SOLA 알고리즘을 적용하여 수신 패킷을 사람이 인지하지 못하는 수준에서 확장하여 전송지연으로 인한 패킷손실을 감소시킨다. 시뮬레이션 결과 전송지연으로 인한 패킷 손실 확률이 상당히 감소되었고 음질 또한 상당히 개선되었다.

• MALSORI
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• no.59
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• pp.69-87
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• 2006

Among the conventional time-scale modification (TSM) methods, the synchronized overlap and add (SOLA) method is widely used due to its good performance relative to computational complexity But the SOLA method remains complex due to its synchronization procedure using the normalized cross-correlation function. In this paper, we introduce a computationally efficient SOLA method utilizing 3 level center clipping method, as well as zero-crossing and level-crossing information. The result of subjective preference test indicates that the proposed method can reduce the computational complexity by over 80% compared with the conventional SOLA method without serious degradation of synthesized speech quality.

• Journal of Broadcast Engineering
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• v.1 no.2
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• pp.85-95
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• 1996

The purpose of this paper Is to propose a dubbed signal time-synchroniztion technique based on the SOLA(Synchronized Over-Lap and Add) method which has been widely used to modify the time scale of speech signal. In broadcasting audio recording environments, the high degree of background noise requires dubbing process. Since the time difference between the original and the dubbed signal ranges about 200mili seconds, process is required to make the dubbed signal synchronize to the corresponding image. The proposed method finds he starting point of the dubbing signal using the short-time energy of the two signals. Thereafter, LPC cepstrum analysis and DTW(Dynamic Time Warping) process are applied to synchronize phoneme positions of the two signals. After determining the matched point by the minimum mean square error between orignal and dubbed LPC cepstrums, the SOLA method is applied to the dubbed signal, to maintain the consistency of the corresponding phase. Effectiveness of proposed method is verified by comparing the waveforms and the spectrograms of the original and the time synchronized dubbing signal.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1
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• pp.1-9
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• 2009

This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.