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Quantization and Transformation of the LP Parameters - Report Example

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This paper 'Quantization and Transformation of the LP Parameters' tells that Quantization and transformation of the LP parameters apply effectively the LBG algorithm which states as follows. The usage of the VQ and the LBG methods in the quantization of the residential signals has been used widely…
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Extract of sample "Quantization and Transformation of the LP Parameters"

Name: Instructor’s name: Institution: Course: Date of submission: Title: Quantisation of LP Parameter Introduction Quantisation and transformation of the LP parameters apply effectively the LBG algorithm which states as follows. The usage of the VQ and the LBG methods in the quantisation of the residential signals has been used widely with the aim of transformation. In transmission, the LP parameter is contained as additional information and is also seen as a problem in the process. During LP analysis, the intelligibility and the quality of speech which is coded heavily relies on how accurate the estimated power spectrum envelope is, which is later used in the determination of its LP’s parameters. LP is known to be crucial in an analysis of the range since it describes great peaks or the spectrum in a frequency plane perceptually. Generation of an accurate envelope of a spectrum can be attained after the spectral parameters have been quantized with minimum resolution's degradation (Nanda, 55). Significant amounts of transmission bits are necessary for the LP parameter transmission. Each frame in a 10 LP order and a speech signal of 8 kHz should use 30-60 quantisation bits, on the quantisation methods used. After every quantisation, each transmission frame has a significant information left thus bringing about a continuous study in the quantisation field of the LP parameters transformation. The original theory associated with the methods of change are discussed and their applications using the simulated scalar quantisers. Besides, the conversion applications involving usage of VQ are also discussed. Criteria of quantisation performance. Basing on the fulfillment of the LP parameter, observations have been made and description of the transformations applied with both non-uniform and uniform SQ made. From the simulation, the non-uniform SQ gives less distortion of quantisation as compared to the uniform SQ. Both VQ and non-uniform SQ applies the LBG algorithm in the quantisation of the LP parameters. Spectral distortion is used while the quantisation activity on the LP parameter is computed. In spectral analysis, the distortion results from the LP setting’s quantisation are crucially minimized. After the quantisation process, the design of the all-pole filter must be kept constant. LP parameter quantisation results to minor errors in quantisation that may lead to big mistakes that are related to the LPC spectrum because of the dependency found among parameters, and this might have effects on the stability of the resulting all-pole filter (Nanda, 55). To curb the prospect of relevant quantisation distortion, various bits are used in the quantisation process, an inefficient solution in response to the rates of bits. A necessity, therefore, has been developed that is required in LP parameter transformation during the separation of less sensitive representations to the distortion of the quantisation thus the stability of H (z) ensured. The SD is viewed in duplicate classification in the measurement of the Quantization’s process performance. These two classifications are the Outlier frames’ percentage and the entire data’s SD average with the frame being considered as the Outlier Frame of the. The outlier is further classified into; i. ii. On fulfillment of the spectral transparency, the required performance of the LP parameter’s quantisation is achieved and the spectral transparency is defined when; as outlier frames are, average SD and when the less than 2% SD of the total frames’ number is between 2-4 dB Besides, the applied quantisation for the LP parameter that is defined using the method of autocorrelation. Use of the covariance method in the analysis of LP at times fails to produce stable H (z) synthesis filter although adjustments have been made on the method for the stabilization of the H (z) intending for the degradation of the Spectral envelope’s accuracy (Nanda, 56). Database The TIMIT database has been used in the simulation process with the speech being sampled at 16 bits resolution and 8 kHz for each sample. LP analysis of the 10th order is performed under the autocorrelation process, and 20 ms frames windows applied in accompaniment to the Hamming window. A bandwidth of 10 Hz widening is applicable for the compensation of the sharp peaks of the spectral. In training, the speech takes less than 236 minutes while in the testing, it only takes 28 minutes of the speech thus on combining both, we use 707438 LP vectors in training and for testing the 85353 LP vectors is used (Nanda, 57). Transformation method In conversion method of LP parameter quantisation, autocorrelation method (AM) is used, and an SD average of 1dB is achieved and 80 bits of quantisation. The transparency of the quantisation is only achieved after addition of more pieces since high bits outline the challenges mentioned initially on the necessity for remodeling of the LP parameter into other differing transformation which is more robust aiming at quantisation. Various changes have been implemented for the accommodation of sensitivity of the spectral of the LP parameters. These transformations should be crucially in a position to have its map lying between each LP parameter the coefficients transformed in the absence of information loss. The developed representations which include the Line spectral frequencies (LSF), the coefficient of the arcsine reflection (ASRC) as well as the reflection coefficient have been covered (Nanda, 57). i. Reflection Coefficient. These are transformations coefficients whose development are aimed at the quantisation whose origin in the recursion by Levinson-Durbin. The factors are termed as "reflection coefficients” because they relate to the reflection coefficient of the vocal tract’s acoustic tube models. The transformation's ladder form is called partial correlations (PARCOR), and its spectrum is less sensitive to quantisation as compared to the LP parameter as the stability of the all-pole filter being targeted and achieved. The coefficients of PARCOR are transformed from (aii) LP setting of p order. Besides, the alternate of the PARCOR coefficient’s reverse can also be retrieved. The transformation method called PARCOR is unique for having limitations in response to how it performs under a system of low bit-rates. Also, the quality of PARCOR coefficients that have been quantised is depended on the bits' number allocated for the respective quantisation. The specific number of the bits should be between 60-80 bits thus better method of transformation being researched on for the LP parameters (Nanda, 58). A uniformly-spaced SQ used the mid-level PCM method which is uniform, and a non-uniformed SP of the parameters transformation was carried out, and the coefficients of the uniform SQ on PARCOR proved to be slightly more than those of the non-uniformly spaced SQ. THE NON-UNIFORMLY SQ parameters of the transformation show better performance since the parameters lack flattened behavior of the spectrum. Mainly on the PARCOR, parameter’s distribution is found to have an increased sensitivity of ranging at 1-0 thus leading the verification of the fact that non-uniform quantisations in the quantisation of the parameters are more beneficial. The effects faced during the allocation of the non-uniform bits for the achievement of optimum results. Besides, one determines the optimum bit allocation's quantisation as a way of minimizing the overall distortion of the spectrum. ii. Coefficients of the Arcsine Reflection. This transformation is non-linear, and its development was aimed at the expansion of the regions close to /Ki/ = 1. The transformation is done for the avoidance of sensitivity associated with quantisation in narrow representations of the bandwidth pole, which is of significant disadvantage to the reflection coefficients (Nanda, 60). iii. Ration of Log-Area This type of a transformation was developed for the exploitation of the similar difficulties associated with the factors on reflection. The coefficients of the LAR are represented as Li=Log, as well as the reflective coefficients being Ki=, Therefore, the speech signals that have the filtering emphasis (LAR) are found to give better performance as compared to the PARCOR although the application gets different when used on differently characterized signals. This quickly leads to the unbounding of the LAR's that re-fixed artificially through the values' bounding (Nanda, 60). iv. Frequencies of Line Spectrum (LSP) The LSP was introduced by Itakura and developed further through a series of several implementations, as the method of transformation is a frequent representative of domains thus advantaging the system of human's perception properties. Exploitation of the modeling speech is experienced through the all-pole filter. Besides, the concentration of the location of the LSF parameter is within the resonances of the spectrum of LPC. In the cases where there is no vibration occurring, the coefficients of the LSF are evenly distributed throughout the plane of frequency. Besides, polynomials’ root of two explains further on the LSF. It has been noted that the polynomials correspond to the vocal tracts' lossless model that has its glottis [P (z)] closed and the [Q (z)] open. These two polynomials, [P(z)] and the [Q (z)] are explained by the property that states that all zeros of the polynomial are interlaced to each other thus lying on the unit circle thus an ascending coefficient of the LSF resulting. Also, this ensures that the all-pole filters are stable. Therefore, this transformation method is recorded as the best robust model that applies to the transmission of signals. Besides, the improvements made upon the LSF on the SQ by the perception of the incorporation or techniques of reducing errors have been researched on with the aim of reducing the bit-rates in Quantisation. As a result of the interlacing evident among the coefficients of the LSF, there has been manipulation of an additional transformation method with the aspects developed being called LSFD's (LSF Differences. The method is said to be highly sensitive to channel error. Therefore the production of spectral distortion is high. Despite several types of research having been performed for the improvement of the LSFD’s quantisation performance which is termed as inferior the method is still impractical as well as the lacking computational efficiency. Therefore further LSF coefficients performance of quantisation has been further discussed Nanda, 61). Split Vector Quantisation of LP Parameters Due to the complexity of computation as well as the addition of memory space required in codebook's quantisation, an efficient method of computing is necessary for VQ. This is for the application in the coding speech in future. The computationally of the Fully-search VQ is very high thus more memory space is required for the codebook’s quantisation. The usage of the split VQ design is in the investigation of the LP parameters’ quantisation performance. Despite the Split VQ being suboptimal, there is the reduction of complexity in computing and memory space needed in the management with minimal effects on the fulfillment of the VQ. Besides, the method is found to divide the LP parameters into partitions of the lowest orders. The calculated SD from the LP parameters quantisation is reduced through identification of the least distortion in small-sized partitions (Nanda, 88). In the root domain is where LP setting's separation occurs thus said to be more accurate. The LP polynomials’ separation into two partitions is intended for the LP analysis of a 10th order. Uniform partitioning if less favored to non-uniform partitioning since there is the chances are high that less MSE will be achieved in lowest roots of frequency. Vast bulks of the accuracy of the power spectrum are preserved here as compared to the significant roots frequency. Computing of the quantisation codebook of M-dimension is done from the N input with LP parameters separation into low and high-frequency sources leads to the formation of low-order polynomials. Therefore, computing of the training on the separate codebook must be performed from similar train vectors’ sets since accommodation of the LP parameter’s splitting must be done into the non-uniform partitions (Nanda, 89). Optimum of codebook indices selection is obtained by identification of all possible combinations of the codebook's sets. The range of unique indices through minimizing the MSE is accurate in the selection of quantised LP parameters transformation despite the reaching the excellent selection through SD's minimization. Besides, as large number bits being allocated in quantisation, the computation is expensive to be made on each index combination (Nanda, 90). Searching for high-frequency codebook reduces codebooks' computation time through low-frequency pre-selection of template codebook. Therefore, codebook's template pre-selection id performed through the personal search of the minimum MSE for the individual index of codebooks. The minimum criterion for MSE selection is used in the completion of simulation since it gives much less cost for computation (Nanda, 91). Works cited Nanda P...Robust linear prediction analysis for low bit-rates Speech Coding. Australia: Brisbane. 2002 print p. 55-91. Read More

These two classifications are the Outlier frames’ percentage and the entire data’s SD average with the frame being considered as the Outlier Frame of the. The outlier is further classified into; i. ii. On fulfillment of the spectral transparency, the required performance of the LP parameter’s quantisation is achieved and the spectral transparency is defined when; as outlier frames are, average SD and when the less than 2% SD of the total frames’ number is between 2-4 dB Besides, the applied quantisation for the LP parameter that is defined using the method of autocorrelation.

Use of the covariance method in the analysis of LP at times fails to produce stable H (z) synthesis filter although adjustments have been made on the method for the stabilization of the H (z) intending for the degradation of the Spectral envelope’s accuracy (Nanda, 56). Database The TIMIT database has been used in the simulation process with the speech being sampled at 16 bits resolution and 8 kHz for each sample. LP analysis of the 10th order is performed under the autocorrelation process, and 20 ms frames windows applied in accompaniment to the Hamming window.

A bandwidth of 10 Hz widening is applicable for the compensation of the sharp peaks of the spectral. In training, the speech takes less than 236 minutes while in the testing, it only takes 28 minutes of the speech thus on combining both, we use 707438 LP vectors in training and for testing the 85353 LP vectors is used (Nanda, 57). Transformation method In conversion method of LP parameter quantisation, autocorrelation method (AM) is used, and an SD average of 1dB is achieved and 80 bits of quantisation.

The transparency of the quantisation is only achieved after addition of more pieces since high bits outline the challenges mentioned initially on the necessity for remodeling of the LP parameter into other differing transformation which is more robust aiming at quantisation. Various changes have been implemented for the accommodation of sensitivity of the spectral of the LP parameters. These transformations should be crucially in a position to have its map lying between each LP parameter the coefficients transformed in the absence of information loss.

The developed representations which include the Line spectral frequencies (LSF), the coefficient of the arcsine reflection (ASRC) as well as the reflection coefficient have been covered (Nanda, 57). i. Reflection Coefficient. These are transformations coefficients whose development are aimed at the quantisation whose origin in the recursion by Levinson-Durbin. The factors are termed as "reflection coefficients” because they relate to the reflection coefficient of the vocal tract’s acoustic tube models.

The transformation's ladder form is called partial correlations (PARCOR), and its spectrum is less sensitive to quantisation as compared to the LP parameter as the stability of the all-pole filter being targeted and achieved. The coefficients of PARCOR are transformed from (aii) LP setting of p order. Besides, the alternate of the PARCOR coefficient’s reverse can also be retrieved. The transformation method called PARCOR is unique for having limitations in response to how it performs under a system of low bit-rates.

Also, the quality of PARCOR coefficients that have been quantised is depended on the bits' number allocated for the respective quantisation. The specific number of the bits should be between 60-80 bits thus better method of transformation being researched on for the LP parameters (Nanda, 58). A uniformly-spaced SQ used the mid-level PCM method which is uniform, and a non-uniformed SP of the parameters transformation was carried out, and the coefficients of the uniform SQ on PARCOR proved to be slightly more than those of the non-uniformly spaced SQ.

THE NON-UNIFORMLY SQ parameters of the transformation show better performance since the parameters lack flattened behavior of the spectrum. Mainly on the PARCOR, parameter’s distribution is found to have an increased sensitivity of ranging at 1-0 thus leading the verification of the fact that non-uniform quantisations in the quantisation of the parameters are more beneficial.

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