Implementation and Overall Performance Evaluation of CELP Based GSM AMR NB Coder over ABE

Quality and intelligibility of narrowband telephone speech can be improved by artificial bandwidth extension (ABE), which extends the speech bandwidth using only information available in the narrowband speech signal. This paper reports a three-language evaluation of an ABE method that has recently been launched in several of Nokia's mobile telephone models. The method extends the speech bandwidth to frequencies above the telephone band by first utilizing spectral folding and then modifying the magnitude spectrum of the extension band with spline curves. The performance of the method was evaluated by formal listening tests in American English, Russian, and Mandarin Chinese. The results of the listening tests indicate that ABE processing improved the subjective quality of coded narrowband speech in all these languages. Differences between bandwidth-extended American English test sentences and their original wideband counterparts were also evaluated using both an objective distance measure that simulates the characteristics of human hearing and a conventional spectral distortion measure. The average objective error was calculated for different categories of speech sounds. The error was found to be smallest in nasals and semivowels and largest in fricative sounds.

Because of its limited bandwidth, telephone speech is poorly intelligible. Artificial bandwidth extension (ABWE) reconstructs themissing frequencies aiming at, e.g., higher intelligibility. It was recently demonstrated that hearing-impaired persons wearing a hearing aid benefit from ABWE-enhanced telephone speech. However, it is unclear, whether persons without hearing impairment also take profit from ABWE in the same test conditions and if so, to what extent. This paper presents a subjective listening test with normal-hearing subjects based on meaningless German syllables simulating narrowband (NB), ABWE-enhanced and wideband (WB) telephone speech in two noisy listening conditions. The test results reveal a clear impact of hearing impairment on the ABWE capability to improve telephone intelligibility. For a signal-to-noise ratio (SNR) of 0 dB, subjects with and without hearing impairment similarly benefit from ABWE. At 20 dB SNR, hearing-impaired subjects take even more profit in contrast to normal-hearing subjects.

Telephone networks normally transmit narrowband (NB) speech with a bandwidth restricted to 300 Hz to 3.4 kHz. While human speech comprises frequencies even further than the NB, there is a degradation of speech quality and intelligibility in such networks. This can be enhanced by recovering the speech components existing beyond the telephone bandwidth using Quadrature Mirror Filter (QMF) to give a perceptually superior wideband (WB) speech signal. QMF banks utilized as the fundamental mathematical tools to analyze and synthesize speech signals to coding of speech signal in sub-bands. Utilization of QMF's allows evading the aliasing effects due to samples decimation when signal is divide into sub-bands. In this paper, the performance of bandwidth extension (BWE) of speech through QMF was assessed by objective parameters like Log Spectral Distortion (LSD) and Itakura-Saito distance (ISD) and subjective parameters like Mean opinion score (MOS), Absolute Category Rating (ACR), Comparison Category Rating (CCR) and Perceptual Evaluation of Speech Quality (PESQ) which demonstrates that the recovered WB speech signal provides speech quality compared by original WB speech signal.

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.

Recently we have explored the use of a Gaussian mixture model (GMM) based global transformer for artificial bandwidth extension (ABWE) for improving the automatic recognition of children's speech in mismatched condition. As the spectral characteristic of the speech varies significantly from one sound class to another so the global transformation would be sub-optimal for that purpose. Motivated by that in this work, we explore the use of class specific GMM based ABWE transformers for the bandwidth extension of the narrowband speech. For the deriving the class specific ABWE transformers an existing unsupervised hidden Markov model (HMM) based method is used. Further for contrast purpose an supervised class specific GMM based ABWE transformers are also explored. The unsupervised and supervised class specific ABWE approaches have resulted in 21.30% and 26.37% relative improvement in word error rate on digit recognition task. The effectiveness of class specific ABWE is also explored in terms of the mutual information between narrowband and the extended higherband speech as well as a group of other speech quality measures.

Many artificial speech bandwidth extension (ABE) approaches perform source-filter decomposition of the input narrowband speech, with subsequent computation of upper frequency band (UB) spectral envelope posteriors. In this paper we perform a direct comparison of HMM- and deep neural network (DNN)-based modeling of likelihoods or posteriors for ABE UB envelope estimation. DNN-based approaches turn out to significantly exceed GMM-based ones in speech quality. Further analysis reveals that this is not due to a better shape of the estimated UB spectral envelope, but primarily due to a much better estimate of the energy ratio of the upper band vs. the lower band — an important result with significant impact on ABE speech quality particularly for fricative sounds.

Performance study of stochastic speech coders

In this paper we investigate embedded coding of speech in the CELP (code-excited-linear-prediction) framework. Compared to other known approaches of variable bit rate speech coding, such as the adaptive multi-rate (AMR) codec, embedded coding systems allow bit rate reductions at any point along the communication network, without changes of the encoder and the decoder. Thus, the quality of the decoded speech increases with the amount of received bits. Aiming at a coding scheme, that produces such a hierarchically-structured bit stream, we focus on a decomposition of the excitation signal by means of pyramid coding. To achieve reasonable speech compression, analysis-by-synthesis (AbS) based quantization of the pyramid layers is designed in a CELP-type fashion, called P-CELP (pyramid CELP). Besides an efficient design of fixed algebraic codebooks in each pyramid layer, special attention is payed to the integration of an adaptive codebook. To achieve maximum performance, the proposed coding scheme is used for wideband (50 Hz-7 kHz) speech.

In this paper, an artificial bandwidth extension (ABE) algorithm from narrowband to wideband is proposed in order to improve the quality of narrowband speech. The proposed ABE algorithm is based on spectral band replication in the modified discrete cosine transform (MDCT) domain with no additional bits. In particular, the patch index search for the replication band is restricted so that the harmonic structure of the wideband speech is maintained after ABE. In the proposed ABE algorithm, we first determine whether the current analysis frame of speech signals is voiced or unvoiced. A harmonic spectral replication or a correlation-based replication approach is then applied for the voiced or unvoiced frame, respectively. The proposed ABE algorithm is finally embedded into the G.729 speech decoder as a post-processor. It is shown from the subjective evaluation using a MUSHRA test that the mean opinion score of the wideband speech signals extended by the proposed ABE method is measured as 75.5, which is higher of around 14% than that of narrowband speech signals.KeywordsBandwidth extensionartificial bandwidth extensionnarrowband speechwideband speechperceptual quality improvement

The quality and intelligibility of narrowband telephone speech can be enhanced by artificial bandwidth extension. This study combines Gaussian mixture model-based (GMM) mel spectrum extension with a filter bank implementation for generating the missing spectral content in the highband at 4-8 kHz. The narrowband mel spectrum is calculated from input speech and the GMM is used to estimate the mel spectrum in the highband. An excitation signal for the highband is generated as a combination of upsampled linear prediction residual and modulated noise. The excitation is divided into sub-bands that are weighted and summed to realize the estimated mel spectrum. The bandwidth-extended output is obtained as the sum of the artificial highband signal and narrowband speech. Listening tests indicate that this method is preferred over narrowband speech and over a previously presented artificial bandwidth extension method which is implemented in some mobile phone models.

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. >

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. >

In recent years research in the area of artificial bandwidth extension (BWE) of speech signals has made significant progress. The intention of BWE is to produce wideband speech with a cut-off frequency of for instance 7 kHz from a narrowband version (e.g., with telephone bandwidth, i.e., 300 Hz − 3.4 kHz). The respective algorithms are based on the estimation of parameters of a source model for speech production given the knowledge of thenarrowband signal. Atheoretical performance bound on this estimation has been formulated in [1]. In order to overcome this boundary we propose the transmission of (compact) side information which can support the parameter estimation. Since a common additional channel would conflict with the requirement of backwards compatibility in narrowband communication systems, the side information is embedded as digital watermark into the narrowband speech.

Dual rate G.723.1 speech codec has been extensively used for real-time video and teleconferencing applications where abridged bandwidth and great voice quality is essential. This paper proposes an effective execution of G.723.1 speech codec. This coder has two bit rates related with it, which is, 5.3 kbit/s and 6.3 kbit/s. This paper addresses implementation analysis of G.723.1 codec carried out on 6.3 kbit/s. This permits the system designer some adaptability by giving a great quality portrayal of the voice information with a fixed level of algorithm difficulty. Encoding of speech is performed with linear predictive analysis-by-synthesis coding. For the excitation signal, the 6.3-kbit/s coder utilizes Multipulse Maximum Likelihood Quantization (MP-MLQ). The 5.3-kbit/s coder utilizes Algebraic-Code-Excited Linear Prediction (ACELP). The quality of decoded speech is evaluated by subjective and objective parameters like Mean opinion score(MOS) and Perceptual evaluation of speech quality(PESQ).

Voice over Internet Protocol (VoIP) is a technology that transports voice data packets across packet switched networks using the Internet Protocol (IP). Perceived speech quality is the key metric for QoS for VoIP applications. The most popular objective measurement method for VoIP voice is E-Model. The primary aim of this work is to investigate the how the delay affect the VoIP quality and how to combined the affection in to standard E-Model. In this paper, a testing environment to measure the impairment of different delay jitter is setup. The Mean Opinion Scores (MOS) were calculated, Perceptual evaluation of speech quality (PESQ) algorithm is used. We use stand deviation of delay to describe the delay effect on voice quality, and give a new Ij formula, which is added to the standard E-model, as jitter impairment factor. Good accuracy is achieved by our new extended E-model.