Embedded speech coding based on pyramid CELP

In today's wireless communication system, quality of decoded speech at receiving end is found muffled and thin, mainly attributed to inherent band limitation (300-3400 Hz) and power constraints. In order to obtain toll quality of recovered speech in terms of intelligibility and naturalness in wireless systems, Narrowband (NB) speech coders should be upgraded to its counterpart Wideband (WB) coders (50-7000Hz). In the meantime, a novel and backward compatible solution is proposed that claims to artificially extend bandwidth of NB speech to WB at receiving end, popular as Artificial Bandwidth Extension (ABE). Out of many techniques which aim to mitigate the effect of the ever unpredictable channel conditions, Adaptive Multi Rate (AMR) NB coder is considered to be one of the potential candidates. Selection of particular bit rate mode (out of all eight bit rate modes between 4.75 kbps and 12.2 kbps) solely depends upon the channel condition. This paper discusses development of ABE algorithm for Code Excited Linear Prediction (CELP) based GSM AMR NB coder, and for the same, MATLAB based e-test bench is created for simulation. Such series of simulations are conducted to discover and gain insight about the overall performance of proposed ABE coder that includes subjective (Mean Opinion Score - MOS) and objective (Perceptual Evaluation of Speech Quality - PESQ) analyses. The evaluated results for both analyses clearly advocate that proposed ABE coder outperforms legacy GSM AMR 06.90 NB coder.

In order to improve the channel bandwidth utilization of voice communication, a variable bit rate speech coding algorithm based on enhanced mixed excitation linear prediction (MELPe) is proposed in the paper. In voice communication, only about 40% of the time is occupied by talking, whereas the rest is engaged by silence or background noise. In addition, unvoiced frame usually requires less transmission rate than the voiced one in low bit rate speech coding algorithms. Therefore, always using the same coding bit rate for speech coding is a waste of channel resource. In this paper, the input signal is divided into speech and silence by using voice activity detection (VAD) technology. And the speech frames are divided into voiced frame or unvoiced frame. They use different coding rates for speech coding and data transmission. All of the parameters are encoded, transmitted and decoded in voiced frame. Only gain parameters, LSF parameters, pitch parameters and overall voicing are encoded, transmitted and decoded in the unvoiced frame. Furthermore, only the gain parameters and the first level LSF parameters are encoded, transmitted and decoded in the silence frame. When about 40% of the time is occupied by talking, compare with the traditional 2.4 kbps MELPe vocoder, the average coding rate of the proposed variable bit rate vocoder can reach 1.33 kbps. But they can achieve the same quality of synthetic speech. Experimental results show that the proposed method reduces the average coding rate, and the synthetic background noise has good comfort on the subjective sense of hearing.

Speech coding algorithms based on Code-Excited Linear Prediction (CELP) [1], have been widely studied in the past few years for low bit rate speech coding and many improvements and variations of the basic algorithm have since emerged. We use Vector Excitation Coding (VXC) as a generic name to represent the class of coders using vector quantization (VQ) coding of filtered excitation signals with “closed-loop” codebook search [2, 3].

The paper presents a speech generative model that provides an efficient way of generating speech waveform from its amplitude spectral envelopes. The model is based on hybrid speech representation that includes deterministic (harmonic) and stochastic (noise) components. The main idea behind the approach originates from the fact that speech signal has a determined spectral structure that is statistically bound with deterministic/stochastic energy distribution in the spectrum. The performance of the model is evaluated using an experimental low-bitrate wide-band speech coder. The quality of reconstructed speech is evaluated using objective and subjective methods. Two objective quality characteristics were calculated: Modified Bark Spectral Distortion (MBSD) and Perceptual Evaluation of Speech Quality (PESQ). Narrow-band and wide-band versions of the proposed solution were compared with MELP (Mixed Excitation Linear Prediction) speech coder and AMR (Adaptive Multi-Rate) speech coder, respectively. The speech base of two female and two male speakers were used for testing. The performed tests show that overall performance of the proposed approach is speaker-dependent and it is better for male voices. Supposedly, this difference indicates the influence of pitch highness on separation accuracy. In that way, using the proposed approach in experimental speech compression system provides decent MBSD values and comparable PESQ values with AMR speech coder at 6,6 kbit/s. Additional subjective listening testsdemonstrate that the implemented coding system retains phonetic content and speaker’s identity. It proves consistency of the proposed approach.

Current developments in high quality, low bit rate speech coders are focused at bit rates in the region of 2.4 kbps and below. This paper presents a source driven variable bit rate coding scheme which operates in conjunction with a prototype interpolation coder (PIC). The proposed SD-VBRC-PIC coder explores effectively the residual redundancy that can be found in the evolution with time trajectories of the PIC model parameters, using a variable synthesis frame size technique. Furthermore, this approach is generic and VBRC can be applied to other prototype interpolation coding systems. Computer simulation experiments show that on average up to a maximum of 30% bit rate savings can be achieved for the same recovered speech quality, as compared to the fixed bit rate of the PIC system under consideration.

Various experiments that have been carried out for the realization of a combined speech and channel coder to be submitted as a candidate for the standardization of the pan-European half-rate channel are described. The speech coder is based on the code-excited linear prediction (CELP) technique and involves binary regular pulse excitation (RPCELP). Fast codebook search procedures are described, and a fast algorithm for closed-loop long-term prediction is introduced. Fractional pitch analysis can be performed with a small computational load. The channel coder uses a convolutional code with an error detection code for the most sensitive bits. The global solution was designed to reduce the effects of transmission errors. The proposed half-rate codec is real-time implemented in fixed point with a very low complexity. Results indicate that speech quality remains close to that of the full-rate codec for all transmission conditions. >

Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coder's data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

The authors introduce two techniques for improving low-bit-rate CELP (code excited linear prediction) speech coders. The sparse-delta stochastic codebook is a recursive codebook design which can be searched using roughly 5% of the computational load required to search a full Gaussian codebook. Pitch sharpening is a class of algorithms which attempt to improve the perceptual quality of CELP by limiting the feedback of low-amplitude noiselike information to the adaptive codebook. Simulation results are presented for sparse-delta, ternary sparse-delta, and simplified-search sparse-delta coders, and for reduced-gain and sparse-adaptive-codebook pitch sharpening algorithms. >

There are many high performance speech coding algorithms, but most of them using bit rates higher than 5kbps. Expected quality of low bit rate speech coding has not been achieved yet, which is an area of interest for military and security applications. There is still room for improvement for low bit rates. Low bit rate speech coding using sinusoidal transform coding is discussed in this paper. The sinusoidal transform coding (STC) is a frequency domain speech compression technique, which can produce speech of high quality at bit rates below 5kbps.In STC, speech signal are represented by linear combination of sine waves with time-vary amplitudes, phase, and frequencies. STC can produce high quality synthesized speech at low bit rates. This paper focuses on the development of a fully speech coder at 2.4 kbps and 4.8 kbps.

Variable Bit Rate (VBR) video traffic modeling is one of the most important challenges in setting up the communication networks. It is applicable to the efficient designing and using of the networks. There are some solutions to the problem of VBR Video traffic modeling. These solutions are divided in to two categories: statistical modeling and deterministic modeling. In this paper, we propose a new deterministic model based on the nonlinear mapping characteristic of fuzzy systems, implemented using a Generalized Regression Neural Network (GRNN). This model is tested with prediction of next 5, 10 and 20 frames of I, P and B frames of the desired movie, using 100 current frames. The model generated data is examined by the matching of autocorrelation function and histogram. This model can regenerate the ACF and histogram of video traces, in order of 10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-3</sup> in error.

The adaptive multi-rate (AMR) is the mandatory speech codec for GSM system, ranging from 12.2 kbps down to 4.75 kbps. To satisfy the requirement of need for both audio and video simultaneously, a kind of hardware and software of AMR speech encoder based on high-performance chips TMS320C64X DSP is proposed. Considering the limited resource of DSP chips, the optimization scheme is referred to later. The result shows that the CPU-load is reduced from 38 percent to 4 percent. And the time of processing each frame is also reduced greatly after optimizing. It is verified with all the test vectors provided by 3 GPP, and the stable operation on the real-time testing board was also confirmed. So, more resource is reserved for video coding on DSP chips. This is helpful for implementing both speech coder and video coder on one DSP chip.

The authors present an algebraic code-excited linear prediction (CELP) speech coder where the innovation codebook comes from the first spherical code of the Barnes-Wall lattice in 16 dimensions. Novel fast optimal algorithms for finding the best sequence in this Barnes-Wall shell innovation codebook are described. This algebraic codebook makes it possible to design a CELP coder at 9.6 kb/s with good quality and still implementable on a current digital-signal-processing chip. >

Postfilters for suppressing coding noises in low bit rate speech coders usually operate in the time domain with high complexity and also introduce nonlinear phase distortion to the speech signals because they are recursive filters. In this paper, a frequency-domain postfiltering technique for multiband excited linear predictive (MBELP) coders is proposed. The idea is to firstly sample the postfilter magnitude spectrum at the pitch harmonics and then scale the signal band magnitudes by the sampled postfilter magnitudes. This approach does not need to specifically perform the filtering process, and the signal energies before and after postfiltering can be easily equalized by simply re-scaling the signal band magnitudes. Another obvious advantage of using this frequency-domain postfiltering technique is that the phase information utilized in multiband synthesis will be retained because only the magnitude spectra are modified. In this paper, a fast method is given to sample the postfilter magnitude spectrum which is characterized by line spectral pair (LSP) parameters. The proposed frequency-domain postfilter has a much lower implementation complexity than its time-domain counterpart.

Vector quantization-based approaches to speech coding have generated new interest in very low bit rate speech coding, that is, speech coded to bit rates below 1200 bits/sec. To achieve such low bit rates, it is necessary to quantize the pitch and energy parameters at rates below 100 bits/sec. Contour quantization is introduced as a technique in which the contour of a given parameter is normalized by a nominal value and vector quantized. Contour quantization is shown to be extremely robust and efficient in encoding the pitch and energy parameters of the LPC vocoder. In this paper, a low rate speech coding system which uses contour quantization to encode the LPC excitation is presented. The system is a fixed bit rate system which is intended to operate at bit rates ranging from 400 bits/s to 800 bits/s. The overall system delay varies from 300 ms at 800 bits/s to 400 ms at 400 bits/s. At 800 bits/s, the system achieved a score of 89 on a three male speaker DRT, and a score of 81 on a three female speaker DRT.

Techniques for improving the performance of CELP (code excited linear prediction) type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function which enhances the perceptual quality of the processed speech is introduced. The combination of harmonic noise weighting and subsample resolution pitch significantly improves the coder performance for voiced speech. A 6.9 kb/s VSELP speech coder which incorporates subsample resolution pitch and harmonic noise weighting is described. Complexity reduction techniques are discussed which allow the coder to be implemented using a single fixed point digital signal processor. >