Excitation Codebook Research Articles

Low Delay Sparse and Mixed Excitation CELP Coders for Wideband Speech Coding

Code Excited Linear Prediction (CELP) algorithms are proposed for compression of speech in 8 kHz band at switched or variable bit rate and algorithmic delay not exceeding 2 msec. Two structures of Low-Delay CELP coders are analyzed: Low-delay sparse excitation and mixed excitation CELP. Sparse excitation is based on MP-MLQ and multilayer models. Mixed excitation CELP algorithm stems from the narrowband G.728 standard. As opposed to G.728 LD-CELP coder, mixed excitation codebook consists of pseudorandom vectors and sequences obtained with Long-Term Prediction (LTP). Variable rate coding consists in maximizing vector dimension while keeping the required speech quality. Good speech quality (MOS=3.9 according to PESQ algorithm) is obtained at average bit rate 33.5 kbit/sec.

Proposed modifications in ITU-T G.729 8 kbps CS-ACELP speech codec and its overall performance analysis

This paper proposes modification in the transmission of excitation codevector and its non-zero pulse sign magnitude using "codebook partition and label assignment" approach, which in turn reduces the number of bits required to transmit it through the communication channel in legacy CS-ACELP 8 kbps speech codec. The proposed approach uses the excitation codebook structure of forward mode standard G.729E 11.8 kbps with two non-zero pulses per track which avoids the use of two algebraic codebook structure for forward mode as well as for backward mode of G.729E with least significant pulse replacement approach for finding optimized excitation codevector. Proposed modification in legacy 8 kbps CS-ACELP (80 bits/10 ms) speech codec actuates the bit rate of 10.6 kbps (106 bits/10 ms) with a better objective and subjective analysis in stark contrast with legacy 8 kbps CS-ACELP speech coder and also avoids the switching of codebook modes of standard 11.8 kbps (G.729E) CS-ACELP speech coder. This paper also aims to propose the reduction in the number of searches in the final codevector of excitation structure by considering initial codevector as a final codevector which improves the quality of the speech compared to the output speech quality of legacy G.729 CS-ACELP working at 8 kbps. Both legacy CS-ACELP 8 kbps speech codec and proposed CS-ACELP 10.6 kbps are implemented in MATLAB. Subjective and objective analysis are carried out on a proposed CS-ACELP 10.6 kbps speech codec in order to evaluate its performance and the results obtained are then cross- compared with the results of legacy CS-ACELP (8 kbps) using set of tables and graphs. It is evident from obtained results that both PESQ and MOS scores are quite comparable for each set of wave files even though bitrates are reduced. Consistency and efficiency of proposed algorithm is assured by calculating the population mean of 95% confidence interval based on obtained objective and subjective parameter results.

Bitrate Scalability for Multi-Pulse Based Code Excited Linear Prediction Speech Coder

Problem statement: The bitrate scalability is an important functionality that is defined in the Moving Picture Expert Group (MPEG-4) requirements. Since the traffic in a communication network varies with time, the capability of varying the bitrate of a speech coder is needed. Approach: This study proposes an approach to vary the bitrate by using a bitrate scalable tool which is attached with the core section of Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder. The bitrates scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder with the bitrate scalable tools is improved above the speech quality of the conventional coder. Conclusion: From the study, the proposed approach gives a tool to improve the speech compression and can also support a variety of coder bitrates.

Efficient Candidate Scheme for Fast Codebook Search in G.723.1

In multimedia communication, due to the limited computational capability of the personal information machine, a coder with low computational complexity is needed to integrate services from several media sources. This paper presents two efficient candidate schemes to simplify the most computationally demanding operation, the excitation codebook search procedure. For fast adaptive codebook search, we propose an algorithm that uses residual signals to predict the candidate gain-vectors of the adaptive codebook. For the fixed codebook, we propose a fast search algorithm using an energy function to predict the candidate pulses, and we redesign the codebook structure to twin multi-track positions architecture. Overall simulation results indicate that the average perceptual evaluation of speech quality (PESQ) score is degraded slightly, by 0.049, and our proposed methods can reduce total computational complexity by about 67% relative to the original G.723.1 encoder computation load, and with perceptually negligible degradation. Objective and subjective evaluations verify that the more efficient candidate schemes we propose can provide speech quality comparable to that using the original coder approach.

Designing Algebraic Trellis Vector Code as an Efficient Excitation Codebook for ACELP Coder

Designing Algebraic Trellis Vector Code as an Efficient Excitation Codebook for ACELP Coder

High Pitch Delay Resolution Technique for Tonal Language Speech Coding Based on Multi-Pulse Based Code Excited Linear Prediction Algorithm

Problem statement: In spontaneous speech communication, speech coding is an important process that should be taken into account, since the quality of coded speech depends on the efficiency of the speech coding algorithm. As for tonal language which tone plays important role not only on the naturalness and also the intelligibility of the speech, tone must be treated appropriately. Approach: This study proposes a modification of flexible Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder with multiple bitrates and bitrate scalabilities for tonal language speech in the multimedia applications. The coder consists of a core coder and bitrate scalable tools. The High Pitch Delay Resolutions (HPDR) are applied to the adaptive codebook of core coder for tonal language speech quality improvement. The bitrate scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder is improved above the speech quality of the conventional coder without pitch-resolution adaptation. Conclusion: From the study, it is a strong evidence to further apply the proposed technique in the speech coding systems or other speech processing technologies.

On the Study of Noise Allocation for Speech Signal in Low Bit-Rate Audio Coding

This letter proposes a new masking threshold adjustment method to improve the quality for the speech signals in low bit-rate audio coding. The Enhanced aacPlus (EAAC) audio codec increases the masking threshold of all frequency bands to be suitable for the given encoding rate by considering equal loudness noises only, which is a representative way for implementing the adjustment technique. The proposed method, however, dynamically adjusts the masking threshold of each frequency band based on the energy ratio of each band to the average band energy. More quantization noises are added to formant regions that have relatively large energy ratio values, but less distortion is allowed in spectral valley regions, which eventually helps to enhance perceptual quality for speech signals. The proposed idea reflects the spectral weighting criterion in searching optimal excitation codebooks used in many speech coding algorithms. Simulation results confirm that the proposed method implemented on the EAAC coder improves quality for the speech input signals at the same bit-rate while keeping equivalent quality for music contents.

New excitation codebook search methods to reduce perceptual degradation of CELP

AbstractIn this paper, the authors propose three new excitation codebook search methods for suppressing the selection of an excitation code vector that has a high likelihood of producing pulse‐like degraded sound in a low‐bit‐rate CELP coder that uses a pulse‐like excitation code vector. The proposed methods are (1) adaptive weighting for coding distortion based on the state of the input speech, (2) weighting for coding distortion due to amplitude distribution distortion in the time direction, and (3) setting an upper limit for the coding distortion rate for pulse‐like excitation codebook searches. Although an excitation code vector with large coding distortion is selected if these proposed methods are set from the coding distortion minimization standard that is used in a conventional CELP excitation code vector search, there is an improvement in the quality that is perceived by the sense of hearing. The authors applied the proposed methods to a 4‐kbps CELP coder to verify their effectiveness according to subjective quality evaluation experiments. The results verified the effectiveness of the proposed methods by showing that the introduction of the proposed methods improved the MOS value by approximately 0.3 for clean speech and by approximately 0.4 for noise‐added speech. © 2007 Wiley Periodicals, Inc. Syst Comp Jpn, 38(8): 1–10, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/scj.20739

An energy extrapolation-based concealment algorithm for an erased excitation signal

We propose an energy extrapolation-based concealment algorithm for an excitation signal of a CELP decoder. In the concealment algorithm, adaptive codebook excitation is generated to maintain continuity of excitation energy on consecutive frames, and high-frequency regions are replaced with random excitation processed through a high-pass filter. The cut-off frequency of the filter is switched according to mode information. The algorithm was implemented on the 3GPP AMR standard decoder, and its performance was compared with that of the standard algorithm. Listening test results suggested that improved subjective quality was produced by the new algorithm.

Watermarking Combined with CELP Speech Coding for Authentication

This letter presents a speech watermarking scheme that is combined with CELP (Code Excited Linear Prediction) speech coding for speech authentication. The excitation codebook of CELP is partitioned into three parts and labeled '0', '1' and 'any' according to the private key. Watermark embedding process chooses the codebook whose label is the same as the watermark bit and combines it with the codebook labeled 'any' for CELP coding. A statistical method is employed to detect the watermark, and the watermark length for authentication and detection threshold are determined by false alarm probability and missed detection probability. The new codebook partition technique produces less distortion, and the statistical detection method guarantees that the error probability can be controlledunder prescribed level.

Pitch adaptive windows for improved excitation coding in low-rate CELP coders

A novel paradigm based on pitch-adaptive windows is proposed for solving the problem of encoding the fixed codebook excitation in low bit-rate CELP coders. In this method, the nonzero excitation in the fixed codebook is substantially localized to a set of time intervals called windows. The positions of the windows are adaptive to the pitch peaks in the linear prediction residual signal. Thus, high coding efficiency is achieved by allocating most of the available FCB bits to the perceptually important segments of the excitation signal. The pitch-adaptive method is adopted in the design of a novel multimode variable-rate speech coder applicable to CDMA-based cellular telephony. Results demonstrate that the adaptive windows method yields excellent voice quality and intelligibility at average bit-rates in the range of 2.5-4.0 kbps.

An experimental assessment of personal speech coding

In the speech coders in common use today, all quantizer codebooks are designed to suit the statistical and perceptual characteristics of speech signals of a population of speakers. However, an individual's speech signal does not exhibit, even over a long time, the entire range of characteristics of the population. With the advent of personal communication systems, personal information might become available and hence may be exploited to improve the rate-distortion performance of speech coders. In this paper, we experimentally assess the potential gain of personal speech coding by designing codebooks for individual speakers. Experiments are performed using linear prediction based analysis-by-synthesis (LPAS) speech coders. The spectral envelope, excitation, and pitch lag quantization codebooks of LPAS coders are redesigned. The gains appear to be modest, suggesting the need to use a coding framework that models personal characteristics more explicitly, e.g. by using prosodic features. Amongst the coder components, the spectral quantizer is found to be most amenable to personalization.

Comb filter speech coding with preselected excitation code vectors

In a code excited speech encoder, an input speech signal is segmented into speech samples at first intervals and a spectral parameter is derived from the speech samples that occur at second intervals longer than the first intervals, the spectral parameter representing the characteristic spectral feature. Each speech sample is weighted with the spectral parameter for producing weighted speech samples. The pitch period of the speech signal is determined from the weighted speech samples. A predetermined number of excitation code vectors having smaller amounts of distortion are selected from excitation codebooks as candidate code vectors. The candidate vectors are comb-filtered with a delay time set equal to the pitch period. One of the filtered code vectors having a minimum distortion is selected. The selected filtered code vector is calculated for minimum distortion and, in response thereto, a gain code vector is selected from a gain codebook. Index signals representing the spectral parameter, the pitch period, the selected excitation and gain code vectors are multiplexed for transmission or storage.

Method and apparatus for efficient multiband celp wideband speech and music coding and decoding

A method of digitally compressing speech and music by use of multiple band ("multiband") fixed excitations stored in codebooks. The use of multiband fixed excitations, along with a coupling method for interconnecting the excitation codebooks and adaptive codebooks and for generating the composite excitation signal, improve the long-term and short-term prediction, and the use of voice-music classification allows the coding structure to be adapted to the statistical character of the audio signal.

Code excitation linear predictive (CELP) encoder and decoder and code excitation linear predictive coding method

There is provided a code excitation linear predictive (CELP) coding or decoding apparatus in which a code vector, which is transmitted by a codebook such as a stochastic codebook, is converted adaptively in accordance with vocal tract analysis information (LPC) so that a high quality reproduction speech is obtained at a low coding rate. Further, in order to obtain a similar effect, a pulse-like excitation codebook formed of an isolated impulse is provided in addition to the adaptive excitation codebook and stochastic excitation codebook so that either the stochastic excitation codebook or the pulse-like excitation codebook is selectively used to provide a vocal tract parameter as a linear spectrum pair parameter.

A 2-D vector excitation coding technique

We present a VQ-based technique for coding image data that adopts an analysis by synthesis approach. We define a new type of spatial interaction model for image data, called prediction pattern, which we use along with a excitation vector, to generate an approximation of an input block of pixels. A prediction pattern is a k× k array with each element representing a prediction scheme from a given set of predictors. A prediction pattern captures the spatial dependences present in an image block. Given a codebook of prediction patterns and a codebook of excitation vectors, we encode an image by partitioning it into blocks and for each block identifying the prediction pattern from within the codebook that best models the spatial dependences that are present in the block. We then search the excitation codebook for a code vector that in combination with the already chosen prediction pattern results in the synthesis of the closest approximation to the current image block. We present algorithms for codebook design and give implementation results. The proposed technique gives promising results.

Voice coder and a method for searching codebooks

After dividing voice signals into subframes, a voice coder calculates auditory sense masking threshold values for each subframe with a masking threshold value calculating circuit, and transforms the auditory sense masking threshold values to auditory sense weighting filter coefficients. An auditory sense weighting circuit performs auditory sense weighting to the signals using the auditory sense weighting filter coefficients and searches excitation codebooks or multipulses using auditory sense weighted signals.

A 4.2 kb/s low-delay speech coder with modified CELP

In this letter, we propose a low-delay speech coder at 4.2 kb/s. In the proposed coder, we remove short-term prediction and use three-tap long term prediction to exploit the speech redundancy. A "dual-decision" pitch estimation algorithm and an adaptive sinusoidal excitation codebook are introduced to improve the performance of the proposed coder. The synthesized speech of this coder has comparable quality to that of conventional CELP coders at similar bit rates, while it has a coding delay of only 5 ms.

Pitch synchronous multi-band (PSMB) coding of speech signals

A novel speech coding algorithm, named pitch synchronous multi-band (PSMB), is proposed. The new coding algorithm uses the multi-band excitation (MBE) model to generate a representative pitch-cycle waveform (PCW) for each frame. The representative PCW of a frame is encoded by two out of three codebooks depending upon whether the frame is related or unrelated to the previous frame. When a frame is related to its previous frame, the PCW is encoded by a length-converted-excitation (LCE) codebook and a stochastic codebook. The codevectors of the LCE codebook are derived from the previous PCW. When a frame is unrelated to its previous frame, it is encoded by a bandlimited single pulse excitation (BSPE) codebook and the stochastic codebook. The new speech coder introduces a pitch-period-based coding feature. It overcomes some weaknesses existing in the improved MBE (IMBE) speech coder. The PSMB coder operating at 4 kbps outperforms the Inmarsat 4.15 kbps IMBE coder by a clear margin. Our listening tests also indicate that it is slightly better than the FS1016 4.8 kbps code excited linear predictive (CELP) coder in terms of perceptual quality. Fast search algorithms for the three codebooks used in PSMB are also developed. The fast algorithms render the new speech coder comparable to the FS1016 CELP coder, in terms of computational complexity.

Voice coding apparatus with synthesized speech LPC code book

A voice coding apparatus has a first linear prediction analyzer for acquiring linear prediction coefficients based on a received input speech sampled at a given time interval. A synthesized speech LPC code book stores linear prediction coefficients of a speech resynthesized based on an old input speech. An excitation code book has predetermined excitation vectors. A first error minimizer receives a signal representing an error between the linear prediction coefficient from the first linear prediction analyzer and one linear prediction coefficient of the synthesized speech LPC code book and acquires an index of the synthesized speech LPC code book which minimizes the error. A linear predictor computes a predictive speech based on the index, acquired by the first error minimizer, and an excitation vector of the excitation code book. A second error minimizer receives a signal representing an error between the input speech and the predictive speech from the linear predictor, and acquires the predictive speech that minimizes the error and an index of the excitation code book at that time while scanning indexes of the excitation code book. A second linear prediction analyzer converts the predictive speech from the second error minimizer into a linear prediction coefficient again and supplies the converted linear prediction coefficient to the synthesized speech LPC code book.