Regular Pulse Excitation Research Articles

Fast and robust 3D T1 mapping using spiral encoding and steady RF excitation at 7 T: application to cardiac manganese enhanced MRI (MEMRI) in mice.

Mapping longitudinal relaxation times in 3D is a promising quantitative and non-invasive imaging tool to assess cardiac remodeling. Few methods are proposed in the literature allowing us to perform 3D T1 mapping. These methods often require long scan times and use a low number of 3D images to calculate T1 . In this project, a fast 3D T1 mapping method using a stack-of-spirals sampling scheme and regular RF pulse excitation at 7 T is presented. This sequence, combined with a newly developed fitting procedure, allowed us to quantify T1 of the whole mouse heart with a high spatial resolution of 208 × 208 × 315 µm(3) in 10-12 min acquisition time. The sensitivity of this method for measuring T1 variations was demonstrated on mouse hearts after several injections of manganese chloride (doses from 25 to 150 µmol kg(-1) ). T1 values were measured in vivo in both pre- and post-contrast experiments. This protocol was also validated on ischemic mice to demonstrate its efficiency to visualize tissue damage induced by a myocardial infarction. This study showed that combining spiral gradient shape and steady RF excitation enabled fast and robust 3D T1 mapping of the entire heart with a high spatial resolution.

Implementation of variable bitrate data hiding techniques on standard and proposed GSM 06.10 full rate coder and its overall comparative evaluation of performance

Paper deals with implementation of variable bit rate steganographic data transmission over ETSI GSM 06.10 FR coder at five different bitrates. Then, few modifications are suggested in Regular Pulse Excitation section of ETSI GSM FR coder which ultimately claims to produce state of the art proposed GSM FR coder. In contrast with ETSI GSM FR coder, proposed coder also exhibits same bit rate steganographic data transmission. Here, in order to facilitate the same, few RPE pulses are identified and being utilized for embedding and hiding the information bits into them. Key element of this research is to allow for joint speech coding and data hiding and that is accomplished with two different approaches like Fixed and Joint Approach. These both approaches are implemented on both Standard and Proposed coders for their overall analytical evaluation of performance using Subjective (Mean opinion Score and Degraded MOS) and Objective (Perceptual Evaluation of Speech Quality) analysis. Small data information is represented as stego signal which can be embedded over different encoded wave files (chosen from NOIZEUS corpus) that serve as carrier signal. Simulation results for both coders reveal the trade off between data embedding rate and recovered speech quality (for both approaches). It is quite evident from both Subjective and Objective analysis that proposed coder offers comparable performance at the same time with lesser simulation delay because of its inherent constructional difference. It remains the fact that for both the coders, Joint approach performs better but at the cost of more simulation delay.

A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE) to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC).

Harmonic chirp imaging method for ultrasound contrast agent

Coded excitation is currently used in medical ultrasound to increase signal-to-noise ratio (SNR) and penetration depth. We propose a chirp excitation method for contrast agents using the second harmonic component of the response. This method is based on a compression filter that selectively compresses and extracts the second harmonic component from the received echo signal. Simulations have shown a clear increase in response for chirp excitation over pulse excitation with the same peak amplitude. This was confirmed by two-dimensional (2-D) optical observations of bubble response with a fast framing camera. To evaluate the harmonic compression method, we applied it to simulated bubble echoes, to measured propagation harmonics, and to B-mode scans of a flow phantom and compared it to regular pulse excitation imaging. An increase of approximately 10 dB in SNR was found for chirp excitation. The compression method was found to perform well in terms of resolution. Axial resolution was in all cases within 10% of the axial resolution from pulse excitation. Range side-lobe levels were 30 dB below the main lobe for the simulated bubble echoes and measured propagation harmonics. However, side-lobes were visible in the B-mode contrast images.

Modelling long-term correlations in broadband speech and audio pulse coders

A new idea to model long-term correlations in broadband speech and audio coders based on regular pulse excitation is presented. It involves expanding the excitation sequence with additional pulses targeting this type of correlation while being bit rate efficient. Comparison with the standard method (long-term prediction) highlights the advantages of the proposed technique.

Extension of proposal of standards for intelligibility tests of Chinese speech: CDRT-tone

The published Chinese diagnostic rhyme test (CDRT) has been proposed as a standard method for evaluating the intelligibility of coded, synthesised or distorted Chinese speech over wired or wireless voice channels in a communication system. Inheriting the philosophy of the English context DRT test, the CDRT only assessed six groups of phonetic attribute distortion. However, pitch distortion is a highly important element for evaluating intelligibility of Chinese speech, which is characterised by Chinese tones, an important element of Chinese language which is absent in western languages. An extension to the CDRT named ‘CDRT-tone’ for intelligibility test of Chinese tones is proposed. The relative features of Chinese tones are outlined and the principles of choosing phonetic units for the proposed standard, CDRT-tone, are presented. Then the validity of the CDRT-tone is discussed. Finally, CDRT-tone was applied to evaluate the pitch performance of the GSM regular pulse excitation with a long-term prediction (RPE-LTP) coding algorithm.

Objective speech quality assessment and the RPE-LTP coding algorithm in different noise and language conditions

The formulation of reliable signal processing algorithms for speech coding and synthesis require the selection of a prior criterion of performance. Though coding efficiency (bits/second) or computational requirements can be used, a final performance measure must always include speech quality. In this paper, three objective speech quality measures are considered with respect to quality assessment for American English, noisy American English, and noise-free versions of seven languages. The purpose is to determine whether objective quality measures can be used to quantify changes in quality for a given voice coding method, with a known subjective performance level, as background noise or language conditions are changed. The speech coding algorithm chosen is regular-pulse excitation with long-term prediction (RPE-LTP), which has been chosen as the standard voice compression algorithm for the European Digital Mobile Radio system. Three areas are considered for objective quality assessment which include: (i) vocoder performance for American English in a noise-free environment, (ii) speech quality variation for three additive background noise sources, and (iii) noise-free performance for seven languages which include English, Japanese, Finnish, German, Hindi, Spanish, and French. It is suggested that although existing objective quality measures will never replace subjective testing, they can be a useful means of assessing changes in performance, identifying areas for improvement in algorithm design, and augmenting subjective quality tests for voice coding/compression algorithms in noise-free, noisy, and/or non-English applications.

A comparison of two speech coders for digital mobile radio applications

This paper attempts to compare the performance of two popular speech coders, namely regular pulse excitation with long-term predictor (RPE-LTP) and code-excited linear prediction (CELP), in both random and burst error environments. In simulation, the bit stream generated by each coder is corrupted with error patterns obtained from the burst error model. The burst error model consists of two parts: a Rayleigh fading envelope and an M-ary in differential phase shift keying reciever model in which the signal level is varied according to the Rayleigh fading envelope. The performance of each coder is evaluated both objectively and subjectively. The simulation results indicate that the RPE-LTP coder provides more consistent performance in the burst error environment. This paper also discusses the comparison of four channel coding techniques designed for the CELP coder.

A reduced complexity multipulse compression system

A reduced complexity multipulse coder is proposed in this paper. A 33% complexity reduction affects the overall speech output quality only slightly as is demonstrated in both informal listening and SNR computation. On the other hand, it makes possible the implementation of the coder with cheap commercial signal processing chips. The TMS32020 has been used for real time implementation of the proposed algorithm at 16 kbit/s with only fixed point computations. The method studied here can be considered as an alternative to the regular pulse excitation technique.

A regular-pulse excited linear predictive codec

A 16 kbit/s speech codec with low complexity and low signal delay is presented which is a special version of the Regular-Pulse Excitation LPC approach (RPE-LPC). This proposal is the basis for the codec standard which will be used in the future Pan-European digital mobile telephone system. An experimental hardware model is described.

Regular-pulse excitation--A novel approach to effective and efficient multipulse coding of speech

This paper describes an effective and efficient time domain speech encoding technique that has an appealing low complexity, and produces toll quality speech at rates below 16 kbits/s. The proposed coder uses linear predictive techniques to remove the short-time correlation in the speech signal. The remaining (residual) information is then modeled by a low bit rate reduced excitation sequence that, when applied to the time-varying model filter, produces a signal that is close to the reference speech signal. The procedure for finding the optimal constrained excitation signal incorporates the solution of a few strongly coupled sets of linear equations and is of moderate complexity compared to competing coding systems such as adaptive transform coding and multipulse excitation coding. The paper describes the novel coding idea and the procedure for finding the excitation sequence. We then show that the coding procedure can be considered as an optimized baseband coder with spectral folding as high-frequency regeneration technique. The effect of various analysis parameters on the quality of the reconstructed speech is investigated using both objective and subjective tests. Further, modifications of the basic algorithm, and their impact on both the quality of the reconstructed speech signal and the complexity of the encoding algorithm, are discussed. Using the generalized baseband coder formulation, we demonstrate that under reasonable assumptions concerning the weighting filter, an attractive low-complexity/high-quality coder can be obtained.