电子与信息学报
電子與信息學報
전자여신식학보
JOURNAL OF ELECTRONICS & INFORMATION TECHNOLOGY
2014年
12期
2896-2901
,共6页
姜开宇%吴超%国雁萌%付强%颜永红
薑開宇%吳超%國雁萌%付彊%顏永紅
강개우%오초%국안맹%부강%안영홍
语音信号处理%声学回声控制%逐级回归%声学回声抵消%声学回声抑制
語音信號處理%聲學迴聲控製%逐級迴歸%聲學迴聲牴消%聲學迴聲抑製
어음신호처리%성학회성공제%축급회귀%성학회성저소%성학회성억제
Speech signal processing%Acoustic echo control%Stage-wise regression%Acoustic echo cancellation%Acoustic echo suppression
传统声学回声控制算法一般采用基于随机梯度法更新的频域分块自适应滤波(PBFDAF)方法,但在以语音为主要回声信号的室内混响环境中,由于回声路径不稳定,往往收敛速度较慢,难以实现足够的回声抑制。该文提出一种基于频域逐级回归的声学回声控制算法。通过逐级回归分析远端信号和麦克风信号之间的线性关系,可以在保持较小的偏差的同时实现收敛较快的系统估计。同时,由于逐级分析了两通道间的短时相干性,因而该算法无需像常见方法一样,额外进行基于通道间相干函数的残余回声抑制或双讲检测,从而保持系统的紧凑性。若进一步假定近端背景噪声准平稳,则可利用基于近端信号非平稳程度的自适应平滑因子,在实现系统估计快速收敛的同时确保其稳定性。实验表明,该方法在常见的近端环境噪声水平下,在收敛速度和稳态误差上相对传统方法有显著优势,非常适合应用在室内远讲模式下的声学回声控制中。
傳統聲學迴聲控製算法一般採用基于隨機梯度法更新的頻域分塊自適應濾波(PBFDAF)方法,但在以語音為主要迴聲信號的室內混響環境中,由于迴聲路徑不穩定,往往收斂速度較慢,難以實現足夠的迴聲抑製。該文提齣一種基于頻域逐級迴歸的聲學迴聲控製算法。通過逐級迴歸分析遠耑信號和麥剋風信號之間的線性關繫,可以在保持較小的偏差的同時實現收斂較快的繫統估計。同時,由于逐級分析瞭兩通道間的短時相榦性,因而該算法無需像常見方法一樣,額外進行基于通道間相榦函數的殘餘迴聲抑製或雙講檢測,從而保持繫統的緊湊性。若進一步假定近耑揹景譟聲準平穩,則可利用基于近耑信號非平穩程度的自適應平滑因子,在實現繫統估計快速收斂的同時確保其穩定性。實驗錶明,該方法在常見的近耑環境譟聲水平下,在收斂速度和穩態誤差上相對傳統方法有顯著優勢,非常適閤應用在室內遠講模式下的聲學迴聲控製中。
전통성학회성공제산법일반채용기우수궤제도법경신적빈역분괴자괄응려파(PBFDAF)방법,단재이어음위주요회성신호적실내혼향배경중,유우회성로경불은정,왕왕수렴속도교만,난이실현족구적회성억제。해문제출일충기우빈역축급회귀적성학회성공제산법。통과축급회귀분석원단신호화맥극풍신호지간적선성관계,가이재보지교소적편차적동시실현수렴교쾌적계통고계。동시,유우축급분석료량통도간적단시상간성,인이해산법무수상상견방법일양,액외진행기우통도간상간함수적잔여회성억제혹쌍강검측,종이보지계통적긴주성。약진일보가정근단배경조성준평은,칙가이용기우근단신호비평은정도적자괄응평활인자,재실현계통고계쾌속수렴적동시학보기은정성。실험표명,해방법재상견적근단배경조성수평하,재수렴속도화은태오차상상대전통방법유현저우세,비상괄합응용재실내원강모식하적성학회성공제중。
Traditional echo control techniques as Partitioned Block Frequency Domain Adaptive Filter (PBFDAF) with stochastic gradient adaptive method usually endure slow convergence and insufficient echo suppression in reverberant room when the echo is speech and the echo path is unstable. An algorithm based on frequency domain stage-wise regression is proposed for acoustic echo control to achieve faster convergence of the system estimation with insignificant bias. Commonly used additional double-talk detector and inter-channel coherence based residual echo suppressor are not needed since short-time coherence analysis is performed in each stage. By further making mild assumptions on the quasi-stationarity of the near-end background noise, both fast convergence and stability of the estimation can be achieved simultaneously with a non-stationarity controlled smoothing factor. Experiments are carried out to show the superiority of the proposed approach in terms of convergence speed and steady state error in distant talking mode in ordinary room environment with various common levels of background noise.