• The Journal of the Acoustical Society of Korea
• /
• v.20 no.5
• /
• pp.35-47
• /
• 2001

This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

• Journal of Korea Multimedia Society
• /
• v.10 no.12
• /
• pp.1551-1558
• /
• 2007

The standard of evaluating function of speech coder for mobile telecommunication can be shown in channel capacity, noise immunity, encryption, complexity and encoding delay largely. This study is an algorithm to reduce complexity applying to CDMA(Code Division Multiple Access) mobile telecommunication system, which has a benefit of keeping the existing advantage of telecommunication quality and low transmission rate. This paper has an objective to reduce the computing complexity by controlling the frequency band nonuniform during the changing process of LSP(Line Spectrum Pairs) parameters from LPC(Line Predictive Coding) coefficients used for EVRC(Enhanced Variable-Rate Coder, IS-127) speech coders. Its experimental result showed that when comparing the speech coder applied by the proposed algorithm with the existing EVRC speech coder, it's decreased by 45% at average. Also, the values of LSP parameters, Synthetic speech signal and Spectrogram test result were obtained same as the existing method.

• The Journal of the Acoustical Society of Korea
• /
• v.37 no.2
• /
• pp.110-117
• /
• 2018

In this paper, we propose a low delay window switching MDCT (Modified Discrete Cosine Transform) method for speech/audio coder. The window switching algorithm is used to reduce the degradation of sound quality in non-stationary trasient duration and to reduce the algorithm delay by using the low delay TDAC (Time Domain Aliasing Cancellation). While the conventional window switching algorithms uses overlap-add with different lengths, the proposed method uses the fixed overlap add length. It results the reduction of algorithm delay by half and 1 bit reduction in frame indication information by using 2 window types. We apply the proposed algorithm to G.729.1 based on MDCT in order to evaluate the performance. The propose method shows the reduction of algorithm delay by half while speech quality of the proposed method maintains same as the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.3
• /
• pp.348-354
• /
• 1993

In this research, a multi-tree coder at 9.6Kbits/sec using a novel scheme for adaptation of the short-term coefficients is developed. The overall delay of the tree coder is maintained at 2.5 msec(16 samples at the 6.4KHz sampling frequency). This coder produces good quality speech over ideal channels, and it is very robust to channel errors up to a bit error rate (BER) of $10^{-3}$. This robustness is achieved by using a parallel adaptation scheme in combination with the use of a smoothed version of the received excitation sequence for adaptation of the short-term prediction coefficients. For the multi-tree coder, reconstructed output speech is evaluated using signal-to-quantization noise ratios (SNR), segmental SNRs, and informal listening tests.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.16 no.2
• /
• pp.69-76
• /
• 2023

A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.21 no.6
• /
• pp.1430-1437
• /
• 1996

A low delay code excited linear predictive speech coder(LD-CELP) at bit rates under 8kbps is considered. We try to improve the perfomance of speech coder with frame type dependent modification of synthesis filter. We first classify frames into 3 groups: voiced, unvoiced and onset. For voicedand unvoiced frame, the spectral envelope of the synthesis filter is adapted to the phonetic characteristics. For transition frame from unvoiced to voiced, the synthesis filter which has been interpolated with the bias filter is used. The proposed vocoder produced more clear sound with similar delay level than other pre-existing LD-CELP vocoders.

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.35S no.7
• /
• pp.1-5
• /
• 1998

In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

• The Journal of the Acoustical Society of Korea
• /
• v.30 no.4
• /
• pp.190-196
• /
• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• IEIE Transactions on Smart Processing and Computing
• /
• v.4 no.6
• /
• pp.422-433
• /
• 2015

In this paper, we demonstrate inter-layer prediction tools for scalable video coders. The proposed scalable coder is designed to support not only spatial, quality and temporal scalabilities, but also view scalability. In addition, we propose quad-tree inter-layer prediction tools to improve coding efficiency at enhancement layers. The proposed inter-layer prediction tools generate texture prediction signal with exploiting texture, syntaxes, and residual information from a reference layer. Furthermore, the tools can be used with inter and intra prediction blocks within a large coding unit. The proposed framework guarantees the rate distortion performance for a base layer because it does not have any compulsion such as constraint intra prediction. According to experiments, the framework supports the spatial scalable functionality with about 18.6%, 18.5% and 25.2% overhead bits against to the single layer coding. The proposed inter-layer prediction tool in multi-loop decoding design framework enables to achieve coding gains of 14.0%, 5.1%, and 12.1% in BD-Bitrate at the enhancement layer, compared to a single layer HEVC for all-intra, low-delay, and random access cases, respectively. For the single-loop decoding design, the proposed quad-tree inter-layer prediction can achieve 14.0%, 3.7%, and 9.8% bit saving.