基于极坐标域主成分分析的无线通信立体声参数编码技术(英文)

High efficiency audio compression is the basic technology in audio involved multimedia communications. Downmixing and parametric coding is efficient coding scheme with wide applications in some up-to-date audio codecs such as Parametric Stereo (PS) in EAAC+ and MPEG-Surround. Principle Component Analysis (PCA) stereo coding followed this idea to map two channels to one channel with maximum energy and parameterize the secondary channel. This paper investigates the conventional PCA method performance under general stereo model with multiple sound sources and different directions, and then proposes a Polar Coordinate based PCA (PC-PCA) stereo coding method. It has been proved that when multiple sound sources exist with different directions, PC-PCA is better than the conventional PCA method when Mean to Standard deviation Ratio (MSR) is large. A stereo codec based on PC-PCA is proposed to validate the performance improvement of proposed method. Objective and subjective tests show the proposed method achieves a comparative quality and saves 50% parameter bit rate comparing with conventional PCA method, and obtains a 4-8 MUSHRA scores improvement comparing with state-of-the-art stereo codec at the same parameter bit rate.     

Frequency shift filter‐based multi‐carrier transceiver

It is well known that orthogonal frequency division multiplexing (OFDM) is sensitive to carrier frequency offset (CFO) and suffers from a high peak‐to‐average ratio. In addition, the performance of OFDM is severely affected by strong co‐channel interference and strong narrowband interference. To mitigate the limitations of OFDM, we propose a new multi‐carrier transceiver based on frequency‐shift filter. A frequency‐shift filter can separate spectrally overlapping sub‐carrier signals by exploiting the spectral correlation inherent in the cyclostationary modulated signals. To increase spectral efficiency, we increase the percentage of spectral overlap between two adjacent sub‐channels. We derive an upper bound and a lower bound on the bit error rate performance of the proposed multi‐carrier transceiver in additive white Gaussian noise channel and frequency‐nonselective Rayleigh fading channel, respectively. Compared with OFDM, our simulation results show that the proposed multi‐carrier transceiver is much less sensitive to CFO and has a lower peak‐to‐average ratio; moreover, without any additional interference suppression technique, the proposed transceiver has the advantage of being able to mitigate strong co‐channel interference with CFO from the intended multi‐carrier signal and mitigate strong narrowband interference in additive white Gaussian noise channel and in Rayleigh fading channel in which a large CFO between the transmitted signal and the received signal often occurs. Copyright © 2011 John Wiley & Sons, Ltd.     

Optimum MVF Estimation‐Based Two‐Band Excitation for HMM‐Based Speech Synthesis

The optimum maximum voiced frequency (MVF) estimation‐based two‐band excitation for hidden Markov model‐based speech synthesis is presented. An analysis‐by‐synthesis scheme is adopted for the MVF estimation which leads to the minimum spectral distortion of synthesized speech. Experimental results show that the proposed method significantly improves synthetic speech quality.     

Filtering of Filter‐Bank Energies for Robust Speech Recognition

We propose a novel feature processing technique which can provide a cepstral liftering effect in the log‐spectral domain. Cepstral liftering aims at the equalization of variance of cepstral coefficients for the distance‐based speech recognizer, and as a result, provides the robustness for additive noise and speaker variability. However, in the popular hidden Markov model based framework, cepstral liftering has no effect in recognition performance. We derive a filtering method in log‐spectral domain corresponding to the cepstral liftering. The proposed method performs a high‐pass filtering based on the decorrelation of filter‐bank energies. We show that in noisy speech recognition, the proposed method reduces the error rate by 52.7% to conventional feature.     

Analysis of new methods for broadcasting digital data to mobileterminals over an FM-channel

Two new methods using an FM-radio channel for transmission of digital data to mobile terminals are examined: 1. A modification of the radio data system (RDS). In RDS, additional digital information is multiplexed with a stereo sound signal. A new system is suggested where the data signal can be multiplexed with a mono audio signal. This causes extension to the bandwidth available for the data signal, and therefore the RDS bitrate can be increased. Error calculations are performed both for the original RDS system and for the new system. 2. Orthogonal frequency division multiplexing (OFDM). OFDM is used in the digital audio broadcasting system (DAB), which is designed to transmit digital audio in the FM band. In OFDM a signal is divided over a large number of 2- or 4-PSK modulated orthogonal subcarriers. The subcarriers of 6 different programmes are multiplexed in one beam to reduce the effects of frequency selectivity of the transmission channel. A new system based on OFDM is proposed, in which the carriers of each programme are transmitted in one FM-channel with a bandwidth of 200 kHz instead of multiplexed with the carriers of other programmes. Error calculations are performed for the subcarriers used in the OFDM modulation method     

Reduced‐Resolution Intra Block Coding Mode

In this letter, a new intra‐block coding mode is presented to improve the coding efficiency for band‐limited signals. A band‐limited block is sub‐sampled, and the sub‐sampled signal is coded on the basis of the conventional prediction/transform coding. The rest of the samples are reconstructed by interpolation at the decoder side without any side information. Experimental results show that the proposed algorithm achieves coding gains of 2.7% for common intermediate format (CIF), 4.29% for quarter CIF, and 6.39% for 720p60 sequences against the H.264/AVC JM10.2 reference software.     

Pairwise subcarriers weighting for suppressing out‐of‐band radiation of OFDM

This paper proposes a subcarrier weighting technique to suppress the out‐of‐band radiation of OFDM signals. By mapping and weighting the same data on an adjacent pair of subcarriers, the spectrum sidelobes are suppressed perfectly through sidelobes mutual cancellation. The optimum weighting factor is derived based on a rectangular pulse‐shaped OFDM spectrum model. Compared with existing out‐of‐band suppression schemes, the proposed scheme not only requires less computational burden but also achieves better spectral roll‐off. For example, when the cyclic prefix of a one‐eighth OFDM‐block length is added, the proposed scheme suppresses the 10‐dB radiation at the center frequency between two subbands which are using cognitive radio. Analytical and simulation results also show that the proposed scheme improves the system carrier‐to‐interference ratio by 10 dB at a normalized frequency offset above 0.1, which leads to the performance improvement in terms of the BER on AWGN channel and multipath fading channel. Copyright © 2011 John Wiley & Sons, Ltd.     

A novel compressed sensing ultra‐wideband channel estimation method based on non‐convex optimization

Due to the low power spectral density and complicated transfer propagation of ultra‐wideband (UWB) signal, it is important to estimate UWB channel accurately. But it is difficult to sample UWB signals directly due to their wider band width. However, compressed sensing (CS) theory provides a feasible way through lower sampling speed. Common CS‐UWB channel estimation methods adopt convex optimization, non‐sparse or non‐restricted form. In order to strengthen the restriction on sparsity of the reconstructed channel vector, a non‐convex optimization method is proposed in this paper to estimate UWB channel. Proposed method sets the objective function as a non‐convex optimization model using l_p–norm. This model is combined as a convex function to approximate the objective function and reconstruct the UWB channel vector iteratively. Because l_p–norm is closer to l₀–norm than l₁ and l₂–norm, its restriction on sparsity of objective vector is stricter. The simulation results show that this method can enhance reconstruction performance compared with existing CS‐UWB channel estimation methods. Copyright © 2013 John Wiley & Sons, Ltd.     

Bandwidth‐Efficient Precoding Scheme with Flicker Mitigation for OFDM‐Based Visible Light Communications

Recently, orthogonal frequency‐division multiplexing (OFDM) was applied to VLC systems owing to its high rate capability. On the other hand, a real‐valued unipolar OFDM signal for VLC significantly reduces bandwidth efficiency. For practical implementation, channel estimation is required for data demodulation, which causes a further decrease in spectral efficiency. In addition, the large fluctuation of an OFDM signal results in poor illumination quality, such as chromaticity changes. This paper proposes a spectrally efficient method based on a hidden‐pilot‐aided precoding technology for VLC with less flickering than a conventional OFDM‐based method. This approach can obtain channel information without any loss of bandwidth efficiency while ensuring illumination quality by reducing the flickering effect of an OFDM‐based VLC. The simulation results show that the proposed method provides a 6.4% gain in bandwidth efficiency with a 4% reduction in flicker compared to a conventional OFDM‐based method.     

Analytical evaluation of downlink interference mitigation in multi‐macrocell/femtocell networks with frequency & cell partitioning

In this paper, we propose an interference mitigation method to suppress the downlink interference in multi‐macrocell/femtocell networks, and analytically evaluate the interference mitigation and average rate performances. Specifically, the proposed interference mitigation method consists of three steps: frequency partitioning, cell partitioning, and sub‐band allocation. In the frequency partitioning step, the whole downlink frequency band is divided into nine non‐overlapping sub‐bands. In the cell partitioning step, each macrocell is divided into four macrocell regions and three femtocell regions for macrocells' and femtocells' communications, respectively. In the sub‐band allocation step, each macrocell or femtocell region is allocated a sub‐band to guarantee that any two neighboring macrocell/femtocell regions use different sub‐bands. Conducted simulation results show that the proposed method is effective in mitigating the downlink interference and improving the average downlink per‐channel rate in multi‐macrocell/femtocell networks. In summary, the major contribution of the proposed interference mitigation method is that the downlink interference can be mitigated without cooperation between macrocells and femtocells, while the full frequency utilization of the macrocell is achieved. Copyright © 2016 John Wiley & Sons, Ltd.     

Speech Enhancement Using Phase‐Dependent A Priori SNR Estimator in Log‐Mel Spectral Domain

We propose a novel phase‐based method for single‐channel speech enhancement to extract and enhance the desired signals in noisy environments by utilizing the phase information. In the method, a phase‐dependent a priori signal‐to‐noise ratio (SNR) is estimated in the log‐mel spectral domain to utilize both the magnitude and phase information of input speech signals. The phase‐dependent estimator is incorporated into the conventional magnitude‐based decision‐directed approach that recursively computes the a priori SNR from noisy speech. Additionally, we reduce the performance degradation owing to the one‐frame delay of the estimated phase‐dependent a priori SNR by using a minimum mean square error (MMSE)‐based and maximum a posteriori (MAP)‐based estimator. In our speech enhancement experiments, the proposed phase‐dependent a priori SNR estimator is shown to improve the output SNR by 2.6 dB for both the MMSE‐based and MAP‐based estimator cases as compared to a conventional magnitude‐based estimator.     

多通路环绕声系统最佳兼容重发的研究

本文利用立体声得发的声像定位理论，提出一种新的在５．１通路环绕声系统（及其它伴随图象的环绕声系统）兼容重发普通双通路立体声信号的方法，并进行了实验验证。理论和实验结果表明：新的方法具有较宽的听音区域，稳定的前方声像，且对中心心位置的倾听者没有声像位置畸变现象，其总的效果过去提提出的方法为优。     

Intra‐ and Inter‐frame Features for Automatic Speech Recognition

In this paper, alternative dynamic features for speech recognition are proposed. The goal of this work is to improve speech recognition accuracy by deriving the representation of distinctive dynamic characteristics from a speech spectrum. This work was inspired by two temporal dynamics of a speech signal. One is the highly non‐stationary nature of speech, and the other is the inter‐frame change of a speech spectrum. We adopt the use of a sub‐frame spectrum analyzer to capture very rapid spectral changes within a speech analysis frame. In addition, we attempt to measure spectral fluctuations of a more complex manner as opposed to traditional dynamic features such as delta or double‐delta. To evaluate the proposed features, speech recognition tests over smartphone environments were conducted. The experimental results show that the feature streams simply combined with the proposed features are effective for an improvement in the recognition accuracy of a hidden Markov model–based speech recognizer.     

Deep Learning for OFDM Channel Estimation in Impulsive Noise Environments

Impulsive noise suppression is essential in orthogonal frequency division multiplexing (OFDM) systems, since impulsive noise may cause a serious decline in channel estimation performance. To solve this problem, a channel estimator based on denoising autoencoder-deep neural network (DAE-DNN) is proposed in this paper. The proposed method is based on a data-driven deep learning framework. Firstly, DAE preprocesses signals to learn damaged data and recover the complete signal are used in the presence of impulsive noise. Then, the transmitted data processed by DAE are used to train the DNN in the offline training process. Finally, the estimated channel state information (CSI) is offered by the proposed DNN model in the online working process. The simulation results demonstrate that the proposed method improves OFDM channel estimation performance significantly. As expected, the proposed method has a better performance than existing ones, such as least squares, minimum mean square error and orthogonal matching pursuit algorithms. Moreover, the proposed method is robust under impulsive noise environments.

     

Two‐Microphone Generalized Sidelobe Canceller with Post‐Filter Based Speech Enhancement in Composite Noise

This paper describes an algorithm to suppress composite noise in a two‐microphone speech enhancement system for robust hands‐free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal‐dominant time‐frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech‐dominant TFBs are identified among the previously detected nonstationary signal‐dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin‐wise output signal‐to‐noise ratio is obtained with these power estimates and a Wiener post‐filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post‐filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.     

Acoustic Event Detection in Multichannel Audio Using Gated Recurrent Neural Networks with High‐Resolution Spectral Features

Recently, deep recurrent neural networks have achieved great success in various machine learning tasks, and have also been applied for sound event detection. The detection of temporally overlapping sound events in realistic environments is much more challenging than in monophonic detection problems. In this paper, we present an approach to improve the accuracy of polyphonic sound event detection in multichannel audio based on gated recurrent neural networks in combination with auditory spectral features. In the proposed method, human hearing perception‐based spatial and spectral‐domain noise‐reduced harmonic features are extracted from multichannel audio and used as high‐resolution spectral inputs to train gated recurrent neural networks. This provides a fast and stable convergence rate compared to long short‐term memory recurrent neural networks. Our evaluation reveals that the proposed method outperforms the conventional approaches.     

Cross‐talk Cancellation Algorithm for 3D Sound Reproduction

If the right and left signals of a binaural sound recording are reproduced through loudspeakers instead of a headphone, they are inevitably mixed during their transmission to the ears of the listener. This degrades the desired realism in the sound reproduction system, which is commonly called ‘cross‐talk.’ A ‘cross‐talk canceler’ that filters binaural signals before they are sent to the sound sources is needed to prevent cross‐talk. A cross‐talk canceler equalizes the resulting sound around the listener's ears as if the original binaural signal sound is reproduced next to the ears of listener. A cross‐talk canceler is also a solution to the problem—how binaural sound is distributed to more than 2 channels that drive sound sources. This paper presents an effective way of building a cross‐talk canceler in which geometric information, including locations of the listener and multiple loudspeakers, is divided into angular information and distance information. The presented method makes a database in an off‐line way using an adaptive filtering technique and Head Related Transfer Functions. Though the database is mainly concerned about the situation where loudspeakers are located on a standard radius from the listener, it can be used for general radius cases after a distance compensation process, which requires a small amount of computation. Issues related to inverting a system to build a cross‐talk canceler are discussed and numerical results explaining the preferred configuration of a sound reproduction system for stereo loudspeakers are presented.     

利用人头动态信息的多通道环绕声编解码方法

针对传统的矩阵环绕声(Matrix Surround)编解码技术存在的不足,提出了新的多通道环绕声编解码方法。在编码端,先将多通道信号转化成前后两对虚拟立体声信号,然后模拟人头微小转动的动态信息,调制前后两对立体声做出相应的变化,左右通道分别相加进行传输,既不影响下混合后普通立体声的收听效果,同时解码端也可以解出动态信息,区分前后立体声,再分配到多通道系统播放,实现良好的环绕声重放。     

Resolving spectral overlap issue of intermediate band solar cells using non‐uniform sub‐bandgap state filling

Intermediate band solar cell provides novel alternative to multi‐junction solar cell, but its efficiency is significantly degraded when spectral overlap exists between different absorption bands. Here, a scheme using non‐uniform sub‐bandgap state filling together with intermediate band transport is proposed to resolve the spectral overlap issue. On the basis of detailed balance calculation, spectrally decoupled devices using low–high state filling is shown to achieve 52.8% conversion efficiency when 4 eV spectral overlap is present between absorption coefficients of different bands, compared with baseline efficiency equal to 35.1% for conventional half‐filled intermediate band devices. If a base material without intermediate band is added to the two section low–high state filling devices, the efficiency is further increased to 61.5%, which approaches efficiency of 63.2% for intermediate band devices with no spectral overlap and 63.8% for unconstrained triple‐junction tandem cells. The junction thermalization loss associated with proposed new structures is shown to be equal to conventional half‐filled intermediate band devices. Copyright © 2013 John Wiley & Sons, Ltd.     

Load‐balancing channel assignment for dual‐band PCS networks

This paper investigates the channel assignment problem of dual‐band PCS systems where single‐band and dual‐band handsets co‐exist. Load‐balancing channel assignment schemes are proposed to improve the system performance. To balance the loads of both bands, the BSC selects a band to serve a call request of a dual‐band handset based on the loads of both bands. In addition, a channel re‐assignment scheme is used to further improve the system capacity. Analytic models and computer simulations have been developed to evaluate the performance of the load‐balancing schemes. The results indicate that both load‐balancing and channel re‐assignment techniques significantly increase the system capacity as the percentage of dual‐band handsets increases. Furthermore, the load‐balancing with channel re‐assignment scheme that combines both techniques achieves the best system performance even when the percentage of dual‐band handset is as low as 25%. In addition, we describe an approach to reduce the signal overhead of the load‐balancing schemes. Copyright © 2002 John Wiley & Sons, Ltd.