Projetos de Filtro Digital - (Digital Filter Design)

Digital Filter Design
Captulo 1: Introduo

1.1. Conceitos bsicos de filtragem digital 1.2. Tipos de filtros digitais
1/1 Conceitos bsicos de filtragem digital

Filtragem digital tem caractersticas especficas que voc precisa dedicar especial ateno. O sinal analgico deve preencher certos requisitos. Alm disso, para converter um sinal digital de sada em formato analgico, necessrio realizar mais processamento de sinais, a fim de obter o resultado adequado. Figura 1-1 mostra o diagrama do bloco processo de filtragem digital.
Figura 1-1. Filtragem digital O processo de converso de um sinal analgico em formato digital realizado por amostragem com uma frequncia de amostragem finita fs. Se um sinal de entrada contm frequncias componentes superior a metade da frequncia de amostragem (fs / 2), que ir provocar distores ao original espectro. Esta a razo pela qual necessrio realizar primeiro filtragem de um sinal utilizando um filtro passa-baixa que elimina alta freqncia componentes de entrada de espectro. Este filtro chamado filtro anti-aliasing em que impede aliasing. Aps o processo de filtragem e de amostragem, um sinal digital est pronto para posterior processamento, que, neste caso, a adequada filtragem utilizando filtro digital. O sinal de sada tambm um sinal digital, que, em alguns casos, tem de ser convertido de volta em formato analgico. Aps digital para o analgico converso, sinal contm alguns componentes frequncias superiores fs / 2, que deve ser eliminada. Novamente, necessrio utilizar um filtro passa-baixa com a frequncia de amostragem fs / 2. As caractersticas especficas de converso esto afetando o sinal ultrapassa o mbito deste livro. Filtro digital atenuao normalmente expressa em termos de escala logartmica decibis (dB). A atenuao medidos em decibis pode ser encontrado utilizando a seguinte expresso: a = 20 * log (H (f)) Cut-off freqncias so utilizadas para filtrar especificao, que ser discutido posteriormente. A frequncia de corte do passband uma freqncia em que a transio da passband para a regio de transio ocorre. A frequncia de corte do stopband uma freqncia em que a transio da regio para a transio stopband ocorre. Estas duas freqncias so equivalentes apenas ideal para o filtro que no possvel perceber na prtica. Em outras palavras, eles so sempre diferentes.
1.2.Types de filtros digitais
Os filtros podem ser classificados em vrios grupos, dependendo de quais so os critrios utilizados para a classificao. Os dois principais tipos de filtros digitais so finitas impulso resposta filtros digitais (FIR filtros) e infinito impulso resposta filtros digitais (IIR). Ambos os tipos tm algumas vantagens e desvantagens que devem ser cuidadosamente considerado quando se projetar um filtro. Alm disso, necessrio ter em conta todas as caractersticas fundamentais de um sinal a ser filtrado, uma vez que estes so muito importantes ao decidir qual filtro de usar. Na maioria dos casos, apenas uma caracterstica que realmente importa e que a de saber se necessrio que tenha filtro linear fase caracterstica ou no. Discurso sinal, por exemplo, podem ser transformados em sistemas com a fase no-linear caracterstica. A fase caracterstica do sinal de fala no da essncia e, como tal, pode ser negligenciada, o que resulta na possibilidade de utilizao muito mais ampla gama de sistemas para a sua prosessing. H tambm sinais de que a fase caracterstica da essncia. Um exemplo tpico so sinais obtidos a partir de vrios sensores na indstria. Portanto, necessrio que tenha um filtro linear fase caracterstica para evitar perder informaes importantes. Quando um sinal a ser filtrado analisado desta forma, fcil decidir qual tipo de filtro digital melhor usar. Assim, se a fase caracterstica da essncia, FIR filtros devem ser usados como eles tm linear fase caracterstica. Esses filtros so de ordem superior e mais complexo, portanto. Caso contrrio, quando apenas frequncia de resposta que interessa, prefervel utilizar filtros IIR digitais que tm muito mais baixa ordem, ou seja, so menos complexos e, portanto, muito mais fcil de perceber. As caractersticas bsicas dos Finite Impulse Response (FIR) filtros so:

fase caracterstica linear; filtro alta ordem (mais complexos circuitos) e estabilidade.
As caractersticas bsicas do Infinite Impulse Response (IIR) so:

no-linear fase caracterstica; filtro para baixo (menos complexos circuitos) e resultando filtro digital tem potencial para se tornar instvel.
Captulo 2: Finite Impulse Response (FIR) Filtro

2.1. 2.2. 2.3. 2.4. 2.5.
Introduo FIR filtro conceber mtodos Janela funes Exemplos Finitas palavra de comprimento efeitos
2.1 Introduo
FIR filtros so filtros digitais com resposta impulso finita. Eles tambm so conhecidos como norecursivo filtros digitais como eles no tm o feedback (uma parte de um filtro recursive), apesar de algoritmos recursivos pode ser usado para filtrar FIR realizao.
Figura 2-1-1. Bloco diagramas de filtros FIR e IIR FIR filtros podem ser concebidas utilizando mtodos diferentes, mas a maioria deles so baseados em ideais filtro aproximao. O objetivo no ideal para alcanar caractersticas, como impossvel qualquer forma, mas para atingir suficientemente boas caractersticas de um filtro. A funo de transferncia do filtro FIR abordagens como o ideal para o filtro aumenta, aumentando assim a complexidade e quantidade de tempo necessrio para o processamento de entrada amostras de um sinal a ser filtrado.
Figura 2-1-2. Ideal filtro passa-baixa aproximao A frequncia de resposta resultante pode ser uma funo ou um monotone oscilatrios funo dentro de uma determinada faixa de freqncia. A onda de frequncia de resposta depende do mtodo utilizado no processo de concepo, bem como sobre os seus parmetros. Este livro descreve o mtodo mais popular para o design que utiliza filtro FIR janela funes. As caractersticas da funo de transferncia, bem como o seu desvio em relao frequncia de resposta ideal depender do filtro ordem e janela funo em uso. Cada filtro categoria apresenta vantagens e desvantagens. Esta a razo pela qual to importante para a categoria e escolher cuidadosamente um tipo de filtro durante o processo de concepo. FIR filtros podem ter linear fase caracterstica, que no como filtros IIR, que ser discutida no captulo 3. Obviamente, em tais casos, quando necessrio ter uma fase linear caracterstica, FIR filtros so a nica opo disponvel. Se a caracterstica linear fase no necessrio, como o caso com o processamento discurso sinais, FIR filtros no so boa soluo a todos. Figura 2-1-3 ilustra entrada e sada de sinais de sistemas de fase no-linear.
Figura 2-1-3. O efeito do no-linear caracterstica fase O sistema introduz um desfasamento de 0 radianos na freqncia de , e radianos em trs vezes a mesma frequncia. Sinal de entrada constituda por uma frequncia natural e harmnico com a mesma amplitude em trs vezes a mesma frequncia. Figura 2-1-3. mostra o esquema de bloqueio de sinal de entrada ( esquerda) e sinal de sada (direita). bvio que estes dois sinais tm diferentes
ondas. A potncia do sinal no alterado, nem as amplitudes de harmnicos, apenas a fase da segunda harmnica alterado. Se partirmos do princpio de que a entrada um sinal discurso cuja fase caracterstica no da essncia, tal distoro na fase do sinal seria insignificante. Neste caso, o sistema preenche todos os requisitos necessrios. No entanto, se a fase caracterstica de grande importncia, tais uma grande distoro no deve ser permitido. A fim de que a fase caracterstica de um filtro FIR linear, o impulso resposta deve ser simtrica ou anti-simtrica, que expressa da seguinte forma: h [n] = h [Nn-1]; symmetric impulso resposta (sobre o seu meio elemento) h [n] =-h [Nn-1]; anti-simtrica impulso resposta (sobre o seu meio elemento) Um dos inconvenientes de filtros FIR uma ordem de alta concebido filtro. A ordem do filtro FIR notavelmente mais elevados em comparao com um filtro IIR com a mesma frequncia de resposta. Esta a razo porque to importante para usar filtros FIR apenas quando a fase linear caracterstica muito importante. Um certo nmero de atrasos linhas contidas em um filtro, ou seja, um nmero de amostras de entrada que devem ser guardados para efeitos de clculo da sada amostra, determina a ordem de um filtro. Por exemplo, se o filtro se presume ser de ordem 10, isso significa que necessrio para salvar 10 amostras de entrada anterior a atual amostra. Todos os onze amostras afetar a produo amostra de filtro FIR. A funo de transformar uma tpica FIR filtro pode ser expressa como um polinmio de uma varivel complexa z-. Todos os plos da funo de transferncia esto localizados na origem. Por esta razo, filtros FIR so garantidos para ser estvel, enquanto filtros IIR tem potencial para se tornar instvel.
2,2 Finitas impulso resposta (FIR) filtro desenho mtodos

A maioria das FIR filtro so baseados em mtodos de concepo ideal filtro aproximao. O filtro resultante aproxima do ideal caracterstica como o filtro de ordem aumenta, tornando, assim, o filtro ea sua execuo mais complexa. O filtro desenho processo comea com as especificaes e exigncias do desejvel FIR filtro. Qual o mtodo para ser utilizado no filtro desenho processo depende do filtro especificaes e implementao. Este captulo discute o filtro FIR desenho mtodo janela funes. Cada um dos mtodos tem dado as suas vantagens e desvantagens. Assim, muito importante para cuidadosamente escolher o mtodo de filtro FIR desenho. Devido sua simplicidade e eficincia, a janela mtodo mais comumente utilizada para projetar filtros. A frequncia de amostragem mtodo fcil de usar, mas filtros concebidos desta forma tm pequena atenuao no stopband. Como j mencionado anteriormente, o processo de desenho comea com a especificao do desejvel FIR filtro.
2.2.1 Conceitos bsicos e FIR filtro especificao

Primeiro de tudo, necessrias para aprender os conceitos bsicos que sero usados ainda neste livro. Voc deve estar ciente de que, sem estar familiarizado com estes conceitos, no possvel compreender a anlise e sntese de filtros digitais. Figura 2-2-1 ilustra um filtro digital passa-baixa especificao. O termo refere-se especificao realmente frequncia de resposta especificao.
Figura 2 .-2.-1a. Filtro digital passa-baixa especificao
Figure2-2-1b. Filtro digital passa-baixa especificao

p - normalizou frequncia de corte na passband; s - normalizou frequncia de corte na stopband; 1 - mxima ripples na passband; 2 - atenuao mnima na stopband [dB]; AP - mxima ripples na passband; e como - no mnimo atenuao stopband [dB].
Frequncia de normalizao pode ser expresso da seguinte forma:
onde:
fs uma frequncia de amostragem; f a frequncia de normalizar e normalizado frequncia.
Especificaes para a alta-pass, band-pass e banda-stop filtros so definidos quase da mesma forma que aqueles para os filtros passa-baixa. Figura 2-2-2 ilustra um filtro passa-alta especificao.
Figura 2 .-2.-2a. High-pass filtro digital especificao
Figura 2 .-2.-2b. High-pass filtro digital especificao Comparando estes dois nmeros 2-2-1 e 2-2-2, bvio que passa-baixo e alto-filtros passa ter especificaes semelhantes. Os mesmos valores so definidos em ambos os casos, com a diferena que, no caso, o mais tarde passband substituded pela stopband e vice-versa. Figura 2-2-3 ilustra um filtro passa-banda especificao.
Figura 2-2-3-A. Band-pass filtro digital especificao
Figura 2-2-3-B. Band-pass filtro digital especificao Figura 2-2-4 ilustra uma banda de balco filtro digital especificao.
Figura 2-2-4-A. Band-stop filtro digital especificao
Figura 2-2-4-B. Band-stop filtro digital especificao
2.2.2 Z-transform
O Z-transform realizada mediante a tempo discreto sinais. Ele converte um sinal discreto tempo de domnio em um domnio complexo frequncia-representao. muito adequado para a anlise tempo-domnio discretos sinais e sistemas. O z-transformada proveniente do Fourier discreta tempo de domnio transformao e considerado o funcionamento bsico do filtro digital design processo. O Z-transform definido como:
onde Z o nmero complexo. Exemplo: Assuma que as amostras de um sinal em tempo discreto x (n) so conhecidos. necessrio transformar este sinal atravs da ztransform e Fourier fransform.
x (n) = (1,2,3,4,5,4,3,2,1); 0 n 8
z-transformada definida como segue:
Torna-se:
A ltima expresso representa a transformar-z do sinal. A transformada de Fourier pode ser encontrado por reescrever a expresso anterior, em termos de z como z = ej. Alm disso, torna-se:
Figura 2-2-5 ilustra o espectro de frequncias do sinal.
Figura 2-2-5. Freqncia espectro do sinal Comparando Z Fourier e transforma, fcil notar algumas semelhanas entre eles:
No sistema de coordenadas polares, o nmero complexo z pode ser expressa como:
As duas ltimas expresses nos levam concluso de que a transformada de Fourier apenas uma forma especial de o transformar-z para r = 1. No plano z, a transformada de Fourier representado como uma unidade crculo, que pode ser visto na figura abaixo 2-2-6.
Figura 2-2-6. Transformada de Fourier no plano z
2.2.2.1 Transformada discreta funo de sistemas de tempo

O Z-transformar-se principalmente utilizado para encontrar a funo de transferncia de sistemas lineares a tempo discreto. Quando a funo de transferncia encontrado, necessrio considerar a sua zeros e plos na z avio. A funo de transferncia discreta de sistemas de tempo definido como:
onde:

bi so os coeficientes feedforward filtro (no-recursivo parte); aj so o feedback filtro coeficientes (recursive parte); H0 uma constante; qi so os zeros da funo de transferncia e pj so os plos da funo de transferncia.
A recursive parte da funo de transferncia realmente um feedback do sistema a tempo discreto. Como referido anteriormente, FIR filtros recursivos no tm esta parte da funo de transferncia, de modo que a expresso anterior pode ser simplificado da seguinte forma:
O impulso de resposta do sistema em tempo discreto obtido a partir de z-transformada inversa da funo de transferncia ou seja, a funo de transferncia do sistema em tempo discreto , na realidade, o Z-transform impulso de resposta:
onde h (n) o impulso de resposta do sistema em tempo discreto.
Figura 2-2-7. Esquema de um sistema linear a tempo discreto No domnio do tempo, o sistema a tempo discreto, mostrado na Figura 2-2-7, pode tambm ser expressa como a convoluo do sinal de entrada x (n) e resposta ao impulso h (n) do sistema:
No domnio da freqncia, o sistema a tempo discreto, mostrado na Figura 2-2-7, pode ser expressa como a multiplicao do sinal de entrada transformar-Z X (z) e funo de transferncia H (z) do sistema:
que ainda d:
A primeira maneira de representar a tempo discreto sistemas mais adequado para a implementao do software em si, enquanto a posterior mais adequado para analisar, implementao hardware (descrito mais tarde) e de sntese, ou seja, a tempo discreto sistema de design. Exemplo: O impulso de uma resposta de 10 ordem FIR filtro criado, atravs do Hamming janela (discutido no prximo captulo) :
h (n) = (0, - 0,0127 - 0,0248, 0,0638, 0,2761, 0,4, 0,2761, 0,0638 - 0,0248 - 0,0127, 0)
A funo de transferncia desse filtro encontrada atravs da ztransform impulso de resposta:
Utilizando a seguinte expresso:
possvel produzir a funo de transferncia do fixo normalizado frequncia. Se, por exemplo = 0.2 ento:
Um exemplo de hardware realizao deste filtro ilustrado na Figura 2-2-8.
Figura 2-2-8. FIR filtro realizao Software realizao requer um buffer de comprimento mnimo 9. Buffers so geralmente circular e seu comprimento pode ser expressa como 2 ^ n, que neste caso significa que o buffer circular de comprimento 16 = 2 ^ 4.
2.2.2.2 Efeito dos plos e zeros da funo de transferncia

A localizao de zeros e plos da funo de transferncia muito importante para discretetime sistema de anlises e snteses. De acordo com a sua posio, possvel testar a estabilidade de um sistema a tempo discreto, detectar arredondar por erros cometidos devido implementao do software de um filtro, bem como erros nos coeficientes encontrados durante a implementao de um filtro de hardware.
Para que um sistema a tempo discreto estvel, todos os plos do sistema em tempo discreto funo de transferncia deve estar localizado no interior do crculo unitrio, como mostrado na Figura 2-2-6. Se esta exigncia no for satisfeita, o sistema torna-se instvel, o que muito perigoso. A localizao de zeros no afeta a estabilidade dos sistemas de tempo discreto. Recordando que a FIR flters no tem um feedback, o que significa que a funo de transferncia no tem plos. Isto provoca um filtro FIR de ser sempre estvel. Filtro de estabilidade ser discutido em mais detalhes, juntamente com filtros IIR que tm potencial para tornar-se instvel devido ao feedback que eles tm. Esta propriedade da FIR filtros realmente representa os seus componentes essenciais adventage. A partir de agora, s os zeros da funo de transferncia sero discutidos neste captulo. Um erro no coeficiente de representao sempre produzidos devido a implementao de software e hardware. Na implementao do software, um erro desencadeado pela palavra de comprimento finito efeito, enquanto que, na implementao de hardware, ocurrs devido impossibilidade de representar os coeficientes com absoluta preciso. O resultado em ambos os casos que o valor dos coeficientes difere do seu valor obtido no processo de concepo. Tais erros causar desvio de frequncia a tempo discreto sistema concebido. Desvio de frequncia depende do espaamento entre os zeros da funo de transferncia FIR filtro. FIR filtro coeficiente erro afeta mais a freqncia caracterstica como o espaamento entre os zeros da funo de transferncia estreitam. Esta propriedade particularmente tpica dos filtros highorder porque os seus zeros esto muito prximos uns dos outros. No entanto, uma ligeira erros na representao coeficiente frequncia pode causar grandes desvios. Figura 2-2-9 ilustra o requerido e obtido freqncia caracterstica de um filtro FIR. Finito A palavra de comprimento efeito sobre a funo de transformar um filtro FIR claramente marcado. Suponhamos que um 50-th fim FIR filtro passa-baixa com normalizou frequncia de corte de 0,25 Hz projetado usando o Hann janela.
Figura 2-2-9. Desvio de frequncia caracterstica necessria A freqncia desvio apresentado na Figura 2-2-9 basicamente ligeiro desvio, embora seja muito elevado em alguns pontos. O mnimo atenuao ea largura da regio de transio resultante IIR filtro permanecem inalteradas, de modo que tal desvio aceitvel. No entanto, como este no um caso comum, necessrio ter muito cuidado na concepo fim filtros elevado porque a funo de transferncia zeros se aproximar, enquanto que mais afectam as resultantes freqncia caracterstica.
2.2.3 Ideal filtro aproximao

O ideal filtro frequncia de resposta utilizado na concepo FIR filtros usando janela funes. O objectivo consiste em calcular o ideal filtro amostras. FIR filtros tm impulso finita resposta, o que
significa que o filtro ideal frequncia de amostragem deve ser realizada em um nmero finito de pontos. Como a frequncia de resposta ideal filtro infinito, fcil produzir erros amostrais. O erro menor que o filtro de ordem aumenta. Figura 2-2-10 ilustra a transferncia das funes do padro ideal quatro filtros.
Figura 2-2-10. Transferir funes de quatro filtros padro ideal O ideal filtro frequncia de resposta pode ser computado atravs da transformada de Fourier inversa. Os quatro filtros padro ideal frequncia respostas esto contidas na tabela abaixo 2-2-1.
Tabela 2-2-1. A freqncia das respostas dos quatro filtros padro ideal O valor da varivel n varia entre 0 e N, onde N o filtro de ordem. A constante M pode ser expressa como M = N / 2. Equivalently, N pode ser expressa como N = 2M. A constante M um inteiro, se o filtro de ordem N mesmo, o que no o caso de filtros de ordem mpar. Se M um inteiro (mesmo filtro fim), a frequncia de resposta ideal filtro simtrico Mth sobre a sua amostra, que encontrado atravs da expresso mostrada na tabela acima 2-2-1. Se M no um inteiro, o ideal filtro frequncia de resposta ainda simtrica, mas no sobre a resposta certa freqncia amostra. Dado que a varivel n varia entre 0 e N, a frequncia de resposta ideal filtro tem N +1 amostra. Se for necessrio para encontrar uma frequncia de resposta do filtro no-padro ideal, a expresso da transformada de Fourier inversa deve ser utilizado:
No-padro filtros so raramente utilizados. No entanto, se existe a necessidade de utilizao de alguns deles, a integral acima deve ser computado atravs de vrios mtodos numricos.
2.2.4 Filtro FIR desenho utilizando janela funes

A FIR filtro desenho processo atravs janela funes podem ser divididas em vrias etapas: 1. 2. 3. 4. 5. 6. Definir filtro especificaes; Especificando uma janela funo de acordo com o filtro de especificaes; Computao do filtro ordem requerida para um determinado conjunto de especificaes; Computao pela janela funo coeficientes; Computao o ideal de acordo com os coeficientes de filtro para filtrao; Computao FIR filtro de acordo com os coeficientes obtidos janela funo e filtro ideal coeficientes; 7. Quando o filtro tem demasiado grande ou demasiado estreito regio de transio, necessrio mudar o filtro por ordem crescente ou decrescente em funo das necessidades, e depois que os passos 4, 5 e 6 so iterada tantas vezes quanto necessrio. O objectivo final de definio de especificaes filtro para encontrar o desejado normalizou frequncias (c, c1, c2), transio de largura e stopband atenuao. A janela funo e filtrar fim so ambos especificados de acordo com esses parmetros. Assim, a janela seleccionada funo deve satisfazer as especificaes dadas. Este ponto ser discutido mais im pormenor no prximo captulo (2.3). Aps este passo, isto , quando a janela funo conhecida, podemos calcular o filtro ordem requerida para um determinado conjunto de especificaes. Uma das tcnicas de computao est prevista no captulo 2.3. Quando a janela funo e filtro ordem so conhecidos, possvel calcular a funo coeficientes janela w [n] usando a frmula para a funo determinada janela. Esta questo igualmente abrangido no prximo captulo. Aps estimar a janela coeficientes funo, necessrio encontrar a freqncia ideal filtro amostras. As expresses utilizadas para calcular estas amostras so discutidos na seo 2.2.3 sob Ideal filtro aproximao. O objectivo final desta etapa a obteno dos coeficientes hd [n]. Dois sequencies W [n] e HD [n] tm o mesmo nmero de elementos. O prximo passo calcular a frequncia de resposta do filtro projetado h [n] utilizando a seguinte expresso:
Por ltimo, a funo de transferncia do filtro projetado ser encontrado por transformar impulso resposta via transformada de Fourier:
ou atravs de transformar-Z:
Se a regio de transio da concebido filtro mais amplo do que necessria, necessrio aumentar o filtro de ordem, a janela reestimate funo coeficientes e ideal filtro frequncia amostras, multipliclas, a fim de obter a frequncia de resposta do filtro e projetado reestimate funo de transferncia como bem. Se a regio de transio mais reduzido do que necessrio, o filtro de ordem pode ser diminuda com a finalidade de otimizar o hardware e / ou software recursos. tambm necessrio reestimate o filtro de freqncia coeficientes depois disso. Por razes de estimativas precisas, o filtro de ordem deve ser diminudo ou aumentado de 1.
2.2.5 realizao filtro FIR

FIR filtro funo de transferncia pode ser expressa como:
A frequncia de resposta realizados no domnio do tempo de maior interesse para a realizao filtro FIR (tanto de hardware e software). A funo de transferncia pode ser encontrada atravs do z-filtro FIR transformar de uma frequncia de resposta. FIR filtro de sada amostras pode ser calculado utilizando a seguinte expresso:
onde:

x [k] so FIR filtro de entrada amostras; h [k] so os coeficientes do filtro FIR frequncia de resposta, e y [n] so FIR filtro de sada amostras.
Uma boa propriedade de filtros FIR que eles so menos sensveis exactido dos filtros IIR constantes do que do mesmo modo. Existem vrios tipos de filtro FIR realizao. Este captulo abrange direta, direta transpor cascata e realizaes que so muito convenientes para o hardware implementao de um filtro. Quanto implementao do software, direto e otimizado realizations sero discutidas neste livro.
2.2.5.1 direto realizao

Direct realizao de filtro FIR baseado na aplicao directa desta expresso:
Direct realizao tambm conhecido como um filtro transversal. Figura 2-2-11 ilustra o bloco diagrama descreve o hardware directo realizao de um filtro FIR.
Figura 2-2-11. FIR filtro directo realizao
Para a realizao directa estrutura, a multiplicao constantes so os mesmos que os coeficientes de funo de transferncia, ou seja, o filtro FIR frequncia de resposta coeficientes. Tal como para o software direta realizao da FIR filtro, necessrio fornecer um mnimo de buffer N amostras, onde N o nmero de filtro FIR coeficientes. Pela sua simplicidade e rapidez, mais comumente utilizado tampo chamado buffer circular cujo comprimento pode ser expressa como 2 ^ k. O valor da constante k um valor mnimo para o qual a expresso N 2 ^ k vlido. Assim:
onde o operador
representa arredondamento para um valor menor.
Figura 2-2-12. Circular buffer de comprimento 16 = 2 ^ 4 O algoritmo utilizado para a realizao de software directo FIR filtro constitudo por: 1. 2. 3. Lendo as amostras de um sinal a ser filtrado; Armazenar uma nova amostra no primeiro lugar disponvel, e Realizando uma operao convoluo sobre filtro coeficientes (frequncia de resposta coeficientes), resultando em uma amostra FIR filtro de sada.
Uma vez que o buffer de 16 bits de largura, abordando buffer circular realizada utilizando endereamento modo 16:
Exemplo: Supondo-se que um filtro usado neste exemplo uma ordem 5. FIR filtro. Design este filtro usando direta com a realizao circular buffer. Tampo O comprimento deve ser de 2 ^ k. O buffer comprimento obtido da seguinte forma:
Isto significa que o comprimento mnimo de buffer circular de 2 ^ 3 = 8. O contedo do buffer aps receber as primeiras 10 amostras mostrado na tabela 2-2-2. Entrada de amostras so indicadas por x [n], enquanto a sombreado clulas denotar tampo locais que tenham sido alteradas. Etapa 0 1 2 3 4 5 6 7 8 9 10 x [7] x [7] x [7] x [6] x [6] x [6] x [6] x [5] x [5] x [5] x [5] x [5] x [4] x [4] x [4] x [4] x [4] x [4] x [3] x [3] x [3] x [3] x [3] x [3] x [3] x [2] x [2] x [2] x [2] x [2] x [2] x [2] x [2] x [1] x [1] x [1] x [1] x [1] x [1] x [1] x [1] x [9] x [0] x [0] x [0] x [0] x [0] x [0] x [0] x [0] x [8] x [8] addr. 7 addr. 6 addr. 5 addr. 4 addr. 3 addr. 2 addr. 1 addr. 0
Tabela 2-2-2. Entrada circular buffer depois de receber 10 amostras Para a realizao do software, a filtragem das amostras de entrada realizada como por frmula abaixo:
2.2.5.2 direto transpor realizao

Direct transpor realizao direta semelhante a realizao de muitas maneiras. Falando sobre a implementao do software, a realizao direta transpor tambm devem ter um tampo de comprimento mnimo N, onde N o nmero de filtro FIR coeficientes. Figura 2-2-13 ilustra o bloco diagrama descrevendo hardware directo transpor realizao de um filtro FIR.
Figura 2-2-13. Direct transpor realizao de um filtro FIR No existem diferenas significativas entre diretos e diretos transpor realizations. Ambas as estruturas tm o mesmo nmero de atraso elementos, o mesmo nmero de agentes multiplicadores e os mesmos coeficientes para realizar mediante multiplicao.
2.2.5.3 realizao Cascade

Cascade realizao, muito conveniente para a sua estrutura modular, comumente utilizado para filtro FIR hardware realizao. Ao utilizar esta realizao, um filtro dividido em vrias seces de baixa ordem. O segundo fim-seces so mais comumente utilizados. Seces individuais so na sua maioria na forma directa realizao, embora eles tambm podem ser na forma direta transpor realizao. A cascata normalmente utilizada para a realizao de ordem superior filtro realizao. A funo de transferncia da cascata realizao tem o seguinte aspecto:
onde:

M o nmero de sees e ak1, ak2 so os coeficientes de multiplicao da seco K.
Figura 2-2-14 ilustra o bloco diagrama descreve o hardware cascata realizao de um filtro FIR.
Figura 2-2-14. FIR filtro cascata realizao O nmero de multiplicadores, adicionais e atrasos o mesmo que para a realizao direta. A principal vantagem da cascata realizao a sua modularidade, caso contrrio, muito conveniente para a implementao de hardware. A cascata de segunda ordem seces importante para a realizao dos filtros de forma arbitrria.
2.2.5.4 Optimizado realizao

Optimizado realizao tem menos, mas mais exigente multiplicadores para realizao. Essa percepo mais comumente usado para implementao de software filtros FIR, porque a reduo do nmero de multiplicadores refora o processo de convoluo (amostras de filtragem processo). Optimizado realizao utiliza a simetria de frequncia de resposta coeficientes. H tambm filtros anti-simtrica FIR que esto alm do escopo deste livro. De qualquer forma, a realizao otimizados podem ser utilizados para estes filtros tambm. A simetria dos coeficientes do filtro FIR frequncia de resposta pode ser expressa pela equao abaixo:
Esta simetria torna possvel a transferncia para a funo a ser expresso da seguinte forma:
Figura 2-2-15 ilustra o diagrama de bloco para a realizao otimizada ainda N, que figura que ilustra 2-2-16 para mpar N.
Figura 2-2-15. Optimizado para a realizao mpar frequncia de resposta
Figura 2-2-16. Optimizado para a realizao ainda frequncia de resposta
Janela 2/3 funes

A janela mtodo mais comumente utilizada para projetar filtros FIR. A simplicidade do processo de concepo torna este mtodo muito popular. Uma janela um array consistindo finito de coeficientes selecionados para satisfazer os requisitos desejveis. Este captulo apresenta alguns mtodos para estimar coeficientes e as caractersticas bsicas da janela em si, bem como o resultado filtros concebidos utilizando estes coeficientes. O ponto encontrar esses coeficientes denotado por w [n]. Ao conceber filtros digitais FIR utilizando janela funes, necessrio especificar:

Uma janela funo a ser utilizados, e O filtro de acordo com a ordem especificaes exigidas (seletividade e stopband atenuao).
Esses dois requisitos esto interligados. Cada funo uma espcie de compromisso entre os dois seguintes requisitos:

Quanto maior a seletividade, ou seja, a transio da regio mais restrita, e A maior supresso da indesejvel espectro, ou seja, quanto maior o stopband atenuao.
Tabela 2-3-1 janela abaixo contm todas as funes mencionadas neste captulo e brevemente compara sua seletividade e stopband atenuao. Normalized comprimento do lbulo principal para N = 20 0.1 Mnima Transio para stopband a regio N = 20 atenuao da janela funo 0.041 13 dB Mnima stopband atenuao do filtro projetado 21 dB
Janela funo Retangular
Triangular (Bartlett) Hann Bartlett, Hanning Hamming BOHMAN Blackman Blackman-Harris
0.2 0.21 0.21 0.23 0.31 0.32 0.43
0.11 0.12 0.13 0.14 0.2 0.2 0.32
26 dB 31 dB 36 dB 41 dB 46 dB 58 dB 91 dB
26 dB 44 dB 39 dB 53 dB 51 dB 75 dB 109 dB
Tabela 2-3-1. Comparao das funes janela Deve ser dada especial ateno para o facto de a atenuao mnima da janela e que a funo do filtro concebido usando essa funo so diferentes, na maioria dos casos. A diferena, isto , atenuao adicional ocorre no mbito do processo de concepo de um filtro utilizando janela funes. Isso afeta a stopband atenuao tornar-se ainda maior, que muito desejvel. No entanto, uma desvantagem deste mtodo que o mnimo stopband atenuao fixada para cada funo. A excepo a Kaiser janela descritos mais adiante neste captulo. Os seguintes conceitos tais como os principais lobo, lbulo principal largura, lobos laterais, regio de transio, mnimo stopband atenuao da funo janela e mnimo stopband atenuao do filtro projetado so descritas em mais detalhe na Figura 2-3-1.
Figura 2-3-1. Principais lobo, lbulo principal largura, lobos laterais, regio de transio Como pode ser visto na tabela acima 2-3-1, o stopband atenuao destas janelas no regulvel. S possvel a afectar a regio de transio, aumentando o filtro de ordem. Por esta razo, prefervel comear a concepo processo, especificando a funo apropriada janela com base no stopband atenuao. mais prefervel para especificar uma janela com a menor atenuao stopband dado que satisfaz os requisitos. Isto permite que o filtro projetado para ter o mais reduzido transio regio. Exemplo:
Desenhe um filtro com as seguintes caractersticas: 1. 2. 3. Mnima stopband atenuao 40dB; Regio de transio entre 2KHz e 3kHz e Frequncia de amostragem 10KHz.
Analisando a tabela acima 2-3-1 (ltima coluna), bvio que a exigida atenuao pode ser alcanado usando a janela Hann. Alm disso, essa atenuao pode ser obtida por meio de qualquer outra janela seguinte Hann janela (todas as funes so classificadas segundo a sua atenuao stopband), assim a maior stopband atenuao seria a um custo de toda a regio de transio para o mesmo fim filtro. Uma vez que o filtro de especificao tambm inclui regio de transio, esse resultado seria ainda mais para aumentar o filtro de ordem e sua complexidade, tambm. Aps especificar o adequado janela funo, necessrio calcular o filtro de ordem. O corte de frequncias de transio regies so normalizadas em primeiro lugar.

A frequncia de amostragem fs = 10KHz; A menor frequncia de corte de transio regio f1 = 2KHz; e A alta frequncia de corte de transio regio f2 = 3kHz.
Normalized as freqncias so obtidas no seguinte maneira:
Transio regio do Hann janela 0.12 para um 20. Fim filtro. Para satisfazer as especificaes dadas, a necessria transio regio expressa como:
fn2 - fn1 = 0.2

Uma vez que a necessria transio regio maior do que a de um 20. Fim filtro, o filtro de ordem deve ser inferior a 20. encontrado atravs iterao. Cerca de estimativa, a primeira soluo, neste caso, pode ser um fim 12th filtro. The required order is somewhat higher and is found after performing several iterations. The resulting filter order is 16. Filter Designer Tool is used for obtaining a first-order filter. The table 2-3-2 provides performed iterations and other important information. Number of iteration 1 2 3 4 Filter order 12 14 16 15 Filter attenuation at 3KHz 25 dB 33 dB 49 dB 38 dB
Table 2-3-2. Calculating filter order After specifying the window function and filter order, it is necessary to compute window coefficients w[n] using expressions for the specified window.
2.3.1 Rectangular Window

The rectangular window is rarely used for its low stopband attenuation. The first lobe (refer to Figure 2-3-2) has attenuation of 13dB and the narrowest transition region, therefore. A filter designed using this window has minimum stopband attenuation of 21 dB. Unlike other window functions being a kind of compromise between requirements for as narrow transition region as possible and as high stopband attenuation as possible, this window is characterized by extreme values. Namely, the minimum transition region is achieved, but at a cost of stopband attenuation. It is easy to find rectangular window coefficients as all coefficients between 0 and N-1 (N-filter order) are equal to 1, which can be expressed in the following way:
w[n] = 1; 0 n N1
Note that the rectangular window performs selection of N samples from a sequence of input samples, but it does not perform sample scaling. Figure 2-3-2 illustrates coefficients in the time-domain.
Figure 2-3-2. Rectangular window in the time domain Figure 2-3-3 illustrates the frequency domain of rectangular window
Figure 2-3-3. Rectangular window in the frequency domain (spectrum) For its less stopband attenuation, the rectangular window is not preferable for digital filter design. Such a less attenuation is a result of cut-off samples within a window (a sequence of sampled frequencies). Up to a zero sample (from which sampling starts), all sampled frequencies are equal to zero. The first sample represents a sudden jump to some value (non-zero sample). Exactly these sudden jumps result in producing relatively sharp high-frequency components which lessen the stopband attenuation. The attenuation gets higher by making cut-off samples less sharp, which results in reducing filter selectivity, ie wider transition region. Since initial requirements of a digital filter are predefined and due to less selectivity, it is necessary to increase the filter order to narrow the transition region. Note the fact that the transition region is inversely proportional to the filter order N. The transition region narrows as the filter order increases. Increase in filter order affects the filter to become more complex and need more time for sample processing. This is why it is very important to be careful when specifying the window function and filter order as well.
2.3.2 Triangular (Bartlett) window

The triangular (Bartlett) window is one among many functions that lessens the effects of final samples. Due to it, the stopband attenuation of this window is higher than that of the rectangular window, whereas the selectivity is less. Namely, filters designed using this window have wider
transition region than those designed using the rectangular window. As a result, it is necessary to have a higher order filter in order to keep the same tansition region as that of the filters designed with the rectangular window. This is the price to pay for producing the higher attenuation. This function also represents a kind of compromise between requirements for as narrow transition region as possible and as higher stopband attenuation as possible, where the transition region is considered more important characteristic. One of the advantages of designing filter using the triangular window is the simplicity of computing coefficients. The triangular window coefficients can be expressed as:
Figure 2-3-4 illustrates coefficients in the time-domain.
Figure 2-3-4. Triangular window coefficients in the time-domain Figure 2-3-5. illustrates the coefficients spectrum of the triangular window shown in Figure 2-3-4.
The attenuation of triangular window is low for most digital filter applications, but it is considerably higher than that for rectangular window. In some cases, when high attenuations are not needed, this filter can be used because it provides an easy way of computing coefficients.
2.3.3 Hann Window

The Hann window is used to lessen bad effects on frequency characteristic produced by the final samples of a signal being filtered. Digital filters designed with this window have higher stopband attenuation than those designed with triangle function. The first side lobe in the frequency domain of this filter has 31dB attenuation, whereas it amounts to 44dB in the designed filter. The transition region is the same as for triangular window, which makes this function one of the most desirable for designing. Another advantage of this window is the ability to realtively fast increase the stopband attenuation of the following lobes. Already the second lobe has 41dB attenuation, whereas it amounts to 54dB for the designed filter. Refer to Figure 2-3-7 illustrating the coefficients of the Hann window. The Hann window belongs to a class of generalized cosine windows which will be discussed in more details at the end of this chapter. The Hann window coefficients can be expressed as:
Figure 2-3-6. illustrates the Hann window coefficients in the time domain.
As seen from the table 2-3-1, the transition region of Hann window is wider than that of triangular window. The difference is that the transition region of triangular window is computed in terms of -26dB attenuation, and in terms of -31dB attenuation for Hann window Accordingly, the conclusion is that Hann window has sharper fall than triangular one, which is considered as its advantage. For the same requirements for minimum attenuation, the Hann window will have a narrower transition region.
2.3.4 Bartlett-Hanning Window

The Bartlett-Hannign window is another compromise between requirements for as narrow transition region as possible and as higher stopband attenuation as possible. The stopband attenuation of a filter designed with this window amounts to 39dB which is not at all a large value. The Bartlett-Hanning window coefficients are expressed as:
This window is in fact a combination of triangular (Bartlett) and Hann window and has higher minimum stopband attenuation than both of them. Considering that the transition region is almost the same as for triangular and Hann window, this function is one of most commonly used windows. Figure 2-3-8. illustrates the Bartlett-Hanning window coefficients in the time domain.
When the stopband attenuation doesnt need to be higher than 39dB, the Bartlett-Hanning window is one of the best solutions to use. As seen in Figure 2-3-9, the side lobes increase attenuation, which is one of very desirable characteristics of digital filters.
2.3.5 Hamming Window

The Hamming window is one of the most popular and most commonly used windows. A filter designed with the Hamming window has minimum stopband attenuation of 53dB, which is sufficient for most implementations of digital filters. The transition region is somewhat wider than that of the Hann and Bartlett-Hanning windows, whereas the stopband attenuation is considerably higher. Unlike minimum stopband attenuation, the transition region can be changed by changing the filter order. The transition region narrows, whereas the minimum stopband attenuation remains unchanged as the filter order increases. The Hamming window coefficients are expressed as:
The Hamming window belongs to a class of generalized cosine functions which are described in more details at the end of this chapter. Figure 2-3-10 illustrates the Hamming window coefficients in the time domain.
Figure 2-3-11 illustrates the Hamming window in the frequency domain. As seen, the first side lobe is attenuated, so that the minimum stopband attenuation is defined in terms of the second side lobe and amounts to 41dB. All side lobes have almost the same maximum values (about -45dB).
2.3.6 Bohman Window

The Bohman window is a convolution of two semi-periods of a cosine function. The transition region and the main lobe are wider than those for Hamming window, but the stopband attenuation is higher, therefore. The attenuation of the first side lobe for Bohman window is 46dB, whereas the filters designed with Bohman window have the stopband attenuation of 51dB. The Bohman window coefficients are expressed as:
Figure 2-3-12 illustrates the Bohman window coefficients in the time domain, whereas Figure 2-3-13 illustrates its coefficients in the frequency domain.
Figure 2-3-13 illustrates the Bohman window coefficients in the frequency domain. As seen, the first side lobe has minimum attenuation of 46dB, whereas for designed filter it amounts to 51dB. The attenuation increases relatively fast. The second side lobe in the Bohman window frequency domain has the attenuation of 61dB, whereas for the filters designed with this window it amounts to 65dB. It should be noted that side lobes of such filters are different than shown in Figure 2-3-13. With regard to minimum attenuation as well as to mid-wide transition region, it can be concluded that Bohman window is suitable for most applications.
2.3.7 Blackman Window

The Blackman window is, along with Kaiser, Hamming and Blackman-Harris windows, considered most commonly used and the most popular windows. Relatively high attenuation makes this window very convenient for almost all applications. The minimum stopband attenuation of a filter designed with this window amounts to 75dB. The Blackman window coefficients are expressed as:
Figure 2-3-14 illustrates the Blackman window coefficients in the time domain, whereas Figure 2-315. illustrates its coefficients in the frequency domain.
As seen from Figure 2-3-15, the Blackman window frequency domain reminds of the Hann window frequency domain. The difference is in attenuation of the first side lobe which amounts to 51dB, as well as in the main lobe which is somewhat wider. The side lobes, following the first one, cause additional stopband attenuation.
2.3.8 Blackman-Harris Window

The Blackman-Harris window is definately one of the most well-known and most commonly used windows. It is characterized by high stopband attenuation and the widest transition region comparing to all windows mentioned in this chapter. However, increase in filter order (more complex filter) cannot affect the minimum stopband attenuation, but affects the transition region. The Blackman-Harris window has almost twice as wide transition region than, say, the Hamming window. Such a drawback (wide transition region) can be overcome by increasing the filter order. The result is a higher order filter (comparing to one designed with Hamming window) with twice as high stopband attenuation. It should be noted that the transition region of the Hamming window ranges between 3dB and 41dB (decrease of 38dB), whereas for Blackman-Harris window it is decreased by 88dB (from 3dB to 91dB ). The Blackman-Harris window has minimum stopband attenuation of 91dB, whereas for the filters designed with this window it amounts to 109dB. Such an attenuation is sufficient for any digital filter application. If it is possible to support a filter order that will be required by this window, then the Blackman-Harris window is probably the best soultion.
The Blackman-Harris window coefficients are expressed as:
Figure 2-3-16. illustrates the Blackman-Harris window coefficients in the time domain, whereas Figure 2-3-16. illustrates its coefficients in the frequency domain.
Figure 2-3-17. illustrates the main characteristics of the Blackman-Harris window in the frequency domain. As seen, the main lobe is wide and the first side lobe is suppressed. Minimum stopband attenuation is determined by the second side lobe having the minimum attenuation of 91dB. The attenuation of other side lobes gradually increases.
2.3.9 Blackman-Harris Window

The generalized cosine functions are a type of window functions whose coefficients can be obtained via expression:
There are some window functions that also belong to this class. These are:

Rectangular window; Hann window; Hamming window; and Blackman window.
Each of these window functions is only a variation of the expression above. Their coefficients a, b and c are given in the table 2-3-3 below. Window function Rectangular Hann Hamming Blackman A 1 0.5 0.54 0.42 B 0 0.5 0.46 0.5 C 0 0 0 0.08
Table 2-3-3. The coefficients of generalized cosine window functions
2.3.10 Kaiser Window

As you know, each of the windows described above is a kind of compromise between requirements for as narrow transition region as possible (greater selectivity) and as higher stopband attenuation as possible. Comparing Hann and Bartlett-Hanning windows, it is obvious that both of them have the same transition region, but the Bartlett-Hanning window has higher attenuation. There is one more thing of concern which says that the minimum stopband attenuation depends on the specified window, whereas an increase in filter order affects the transition region. All this leads us to the conclusion that the windows described here are not optimal. An optimal window is a function that has maximum attenuation according to the given width of the main lobe. The optimal window is also known as Kaiser window. Its coefficients are expressed as:
where aa is the minimum stopband attenuation, and is the width of (normalized) transition region. The order of band-pass and band-stop filters, obtained from the expression above, should be multiplied by 2.
The value of parameter can be obtained from the table 2-3-4. aa less than 21 between 21 and 50 more than 50 Table 2-3-4. Values of parameter I0(*) is a modified zero order Bessel function of the first kind. It can be approximated via expression: 0 0.5842(aa - 21)^0.4 + 0.07886(aa - 21) 0.1102(aa - 8.7)
In most practical cases it is sufficient to consider the first 20 elements of this order. As can be seen from all mentioned above, in order to design an optimal Kaiser filter it is necessary to know normalized width of transition region as well as minimum desirable stopband attenuation. Example: What would a filter design with Kaiser window look like if we take into consideration the requirements given at the beginning of this chapter? The result of a filter designed with the Hann window is a 16th order filter. Assuming that it is required to design a filter with the following characteristics: 1. Minimum stopband attenuation is 40dB (aa); 4. Transition region is between 2KHz and 3KHz (f1, f2); and 5. Sampling frequency is 10KHz (fs). Transition region:
The specified filter order is N=12. It means that the needed filter order is less by 4 than that obtained with the Hann window. It is important to say that all the expressions above are obtained in an empirical way, which means that there will be some exceptions and variations in practice. For example, the expression used to compute the filter order gives accurate results in approximately 98% cases. Otherwise, the resulting filter order should be changed. Fortunately, the changes to be made are slight, and the filter order is increased or decreased by 2 at most.
2.4 Examples
This chapter discusses various FIR filter design methods. It also provides examples of all types of filters as well as of all methodes described in the previous chapters. The four standard types of filters are used here:

low-pass filter; high-pass filter; band-pass filter; and band-stop filter.
The design method used here is known as the window method. The FIR filter design process can be split into several steps as described in Chapter 2.2.4 entitled Designing FIR filters using window functions . These are: 1. Defining filter specifications;

Specifying a window function according to the filter specifications; Computing the filter order according to the filter specifications and specified window function; Computing the coefficients of the window;
Computing the ideal filter coefficients according to the filter order; Computing the FIR filter coefficients according to the obtained window function and ideal filter coefficients; and If the resulting filter has too wide or too narrow transition region, it is necessary to change the filter order. The specified filter order is increased or decreased according to needs, and steps 4, 5 and 6 are repeated after that as many times as needed.
Depending on the window function in use, some steps will be skipped. If the filter order is known, step 3 is skipped. If the window function to use is predetermined, step 2 is skipped. In every given example, the FIR filter design process will be described through these steps in order to make it easier for you to note similarities and differencies between various design methodes, window functions and design of various types of filters as well.
2.4.1 Filter design using Rectangular window 2.4.1.1 Example 1

Step 1: Type of filter low-pass filter Filter specifications:
Filter order N=10
2. Sampling frequency fs=20KHz 3. Passband cut-off frequency fc=2.5KHz Step 2: Method filter design using rectangular window Step 3: Filter order is predetermined, N=10; A total number of filter coefficients is larger by one, ie N+1=11; and Coefficients have indices between 0 and 10. Step 4: All coefficients of the rectangular window have the same value equal to 1.
w[n] = 1 ; 0 n 10
Step 5: The ideal low-pass filter coefficients (ideal filter impulse response) are expressed as:
where M is the index of middle coefficient.
Normalized cut-off frequency c can be calculated using the following expression:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c with expression for the impulse response coefficients of the ideal low-pass filter:
The middle element is found via the following expression Step 6: The designed FIR filter coefficients are obtained via the following expression:
The FIR filter coefficients h[n] rounded to 6 digits are:
Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-1 illustrates the direct realization of designed FIR filter, whereas Figure 2-4-2 illustrates the optimized realization of designed FIR filter, which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.1.2 Example 2
Step 1: Type of filter high-pass filter Filter specifications:

Filter order N=8 Sampling frequency fs=20KHz Passband cut-off frequency fc=5KHz
Step 2: Method filter design using rectangular window Step 3: Filter order is predetermined, N=8; A total number of filter coefficients is larger by 1, ie N+1=9; Coefficients have indices between 0 and 8. Step 4: All coefficients of the rectangular window have the same value equal to 1.
w[n] = 1 ; 0 n 8
Step 5: The ideal high-pass filter coefficients (ideal filter impulse response) are expressed as:
Normalized cut-off frequency c can be calculated using the following expression:
Step 6: The designed FIR filter coefficients are found via expression:
Step 7:
The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-3 illustrates the direct realization of designed FIR filter, whereas figure 2-4-4 illustrates the optimized realization of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.1.3 Example 3
Step 1: Type of filter band-pass filter Filter specifications:
Filter order N=14
2. Sampling frequency fs=20KHz 3. Passband cut-off frequency fc1=3KHz, fc2=5.5KHz Step 2: Method filter design using rectangular window Step 3: Filter order is predetermined, N=14 A total number of filter coefficients is larger by 1, ie N+1=15. Coefficients have indices between 0 and 14. Step 4: All coefficients of the rectangular window have the same value equal to 1.
w[n] = 1 ; 0 n 14
Normalized cut-off frequencies c1 and c2 can be found using the following expressions:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c1 and c2 with expression for the impulse response coefficients of the ideal band-pass filter:
Step 7: Filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-5 illustrates the direct realization of designed FIR filter, whereas figure 2-4-6 illustrates optimized realization of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.1.4 Example 4
Step 1: Type of filter band-stop filter Filter specifications:
Filter order N=14
Sampling frequency fs=20KHz Stopband cut-off frequency - fc1=3KHz, fc2=5.5KHz
Step 2: Method filter design using rectangular window Step 3: Filter order is predetermined, N=14; A total number of filter coefficients is larger by 1, ie N+1=15; and Coefficients have indices between 0 and 14. Step 4: All coefficients of the rectangular window have the same value equal to 1.
w[n] = 1 ; 0 n 14
Normalized cut-off frequencies c1 and c2 can be found using expressions:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c1 and c2 with expression for the impulse response coefficients of the ideal band-stop filter:
Note that, excepting the middle element, all coefficients are the same as in the previous example (band-pass filter with the same cut-off frequencies), but have the opposite sign. Step 6: The designed FIR filter coefficients are found via expression:
The FIR filter coefficients h[n], rounded to 6 digits, are:
2.4.2 Filter design using Bartlett window 2.4.2.1 Example 1


Filter order Nf=9 Sampling frequency fs=20KHz Passband cut-off frequency fc=2.5KHz
Step 2: Method filter design using Barlett window Step 3: Filter order is predetermined, Nf=9; A total number of filter coefficients is larger by 1, ie N=Nf+1=10; and Coefficients have indices between 0 and 8. Step 4: The coefficients of Bartlett window are expressed as:
Step 5: The ideal low-pass filter coefficients (ideal filter impulse response) are given in the expression below:
Since the value of M is not an integer, the middle element representing a center of coefficients symmetry doesnt exist. Normalized cut-off frequency c can be calculated using expression:
h[n] = w[n] * hd[n] ; 0 n 9

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-9 illustrates the direct realization of designed FIR filter, whereas figure 2-4-10 illustrates optimized realization of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.2.2 Example 2
Step 1:
Type of filter high-pass filter Filter specifications:

Filter order Nf=8 Sampling frequency fs=20KHz Passband cut-off frequency fc=5KHz
Step 2: Method filter design using Bartlett window Step 3: Filter order is predetermined, Nf=8; A total number of filter coefficients is larger by 1, ie N=Nf+1=9; Coefficients have indices between 0 and 8. Step 4: The Bartlett window function coefficients are found via expression:
Normalized cut-off frequency c may be calculated via the following expression:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c with expression for the impulse response coefficients of the ideal high-pass filter:
h[n] = w[n] * hd[n] ; 0 n 8

2.4.2.3 Example 3

Filter order Nf=14; Sampling frequency fs=20KHz; and Passband cut-off frequencies fc1=3KHz, fc2=5.5KHz.
Step 2: Method filter design using Bartlett window Step 3: Filter order is predetermined, Nf=14; A total number of filter coefficients is larger by 1, ie N=Nf+1=15; and Coefficients have indices between 0 and 14. Step 4: The Balett window coefficients are found via expression:
Normalized cut-off frequencies c1 and c2 can be calculated using expressions:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c1 and c2 with expression for the impulse response coefficients of the ideal band-pass filter:
h[n] = w[n] * hd[n] ; 0 n 14

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-13 illustrates the direct realization of designed FIR filter, whereas figure 2-4-14 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.2.4 Example 4
Step 1:
Type of filter band-stop filter Filter specifications:

Filter order Nf=14; Sampling frequency fs=20KHz; and Stopband cut-off frequencies fc1=3KHz, fc2=5.5KHz.
Step 2: Method filter design using Bartlett window Step 3: Filter order is predetermined, Nf=14; A total number of filter coefficients is larger by 1, ie N=Nf+1=15; and Coefficients have indices between 0 and 14. Step 4: The coefficients of Bartlett window are found via expression:
The values of coefficients (rounded to six digits) are obtained by combining the values of M, c1 and c2 with expression for the impulse response coefficients of the ideal band-stop filter:
Note that, excepting the middle element, all the coefficients are the same as in the previous example (band-pass filter with the same cut-off frequencies), but have the opposite sign. Step 6: The designed FIR filter coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 14

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization:
Figure 2-4-15 illustrates the direct realization of designed FIR filter, whereas figure 2-4-16 illustrates optimized realization of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
It is determined on purpose that FIR filters, explained in examples 3 and 4, have the same order. The similarity between the coefficients of band-pass and band-stop FIR filters is obvious. All coefficients of the band-stop FIR filter have the same absolute values as the corresponding coefficients of the band-pass FIR filter. The only difference is that they are of the opposite sign. The middle element of the band-stop filter is defined as:
bbs = 1 - bbp
where:

bbs is the middle coefficient of the band-stop filter; and bbp is the middle coefficient of the band-pass filter.
Because of such similarity, it is easy to convert a band-pass FIR filter into a band-stop FIR filter having the same cut-off frequencies, sampling frequency and filter order. Besides, low-pass and high-pass FIR filters are interrelated in the same way, which can be seen in examples describing Hann window.
2.4.3 Filter design using Hann window 2.4.3.1 Example 1


Filter order Nf=10; Sampling frequency fs=20KHz; and Passband cut-off frequency fc=2.5KHz.
Step 2: Method filter design using Hann window Step 3: Filter order is predetermined, Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10. Step 4: The Hann window function coefficients are found via expression:
Normalized cut-off frequency c can be calculated using expression:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c with expression for the impulse reaponse coefficients of the ideal low-pass filter:
h[n] = w[n] * hd[n] ; 0 n 10

2.4.3.2 Example 2
Step 1: Filter type high-pass filter Filter specifications:

Filter order Nf=10; Sampling frequency fs=20KHz; and Passband cut-off frequency fc=2.5KHz.
Step 2: Method filter design using Hann window Step 3: Filter order is predetermined, Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10. Step 4: The Hann window function coefficients are found via expression:
Step 5:
The ideal high-pass filter coefficients (ideal filter impulse response) are expressed as:
h[n] = w[n] * hd[n] ; 0 n 10

Step 7: The filter order is predetermined. There is no need to additionally change it.
Filter realization: Figure 2-4-19 illustrates the direct realization of designed FIR filter, whereas figure 2-4-20 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.3.3 Example 3

Filter order Nf=14; Sampling frequency fs=20KHz; and Passband cut-off frequency fc1=3KHz, fc2=5.5KHz.
Step 2: Method filter design using Hann window Step 3: Filter order is predtermined, Nf=14; A total number of filter coefficients is larger by 1, ie N=Nf+1=15; and Coefficients have indices between 0 and 14. Step 4: The Hann window function coefficients are found via expression:
The values of coefficients (rounded to six digits) are obtained by combining the values of M, c1 and c2 with expression for the impulse response coefficients of the ideal band-pass filter:
h[n] = w[n] * hd[n] ; 0 n 14

2.4.3.3 Example 3
Step 1: Type of filter band-stop filter Filter specifications:

Filter order Nf=14; Sampling frequency fs=20KHz; and Passband cut-off frequency fc1=3KHz, fc2=5.5KHz.
Step 2: Method filter design using Hann window Step 3: Filter order is predetermined, Nf=14; A total number of filter coefficients is larger by 1, ie N=Nf+1=15; Coefficients have indices between 0 and 14. Step 4: The Hann window function coefficients are found via expression:
Note that, excepting the middle element, all coefficients are the same as in the previous example (band-pass filter with the same cut-off frequencies), but have the opposite sign. Step 6: The designed FIR filter coefficients are found via expression:
h[n]=w[n] * hd[n] ; 0 n 14
Step 7:
The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-23 illustrates the direct realization of designed FIR filter, whereas figure 2-4-24 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
It is specified on purpose that FIR filters, explained in examples 1 and 2, have the same order. The similarity between low-pass and high-pass FIR filter coefficients is obvious. All coefficients of the low-pass FIR filter have the same absolute values as the corresponding coefficients of the high-pass FIR filter. The only difference is that they are of the opposite sign. The middle element is defined as:
blp = 1 - bhp
where:

blp is the middle coefficient of a low-pass filter; and bhp is the middle coefficient of a high-pass filter.
Because of such similarity, it is easy to convert a low-pass FIR filter into a high-pass FIR filter having the same cut-off frequencies, sampiling frequency and filter order.
2.4.4 Filter design using Bartlett-Hanning window 2.4.4.1 Example 1


Filter order Nf=9; Sampling frequency fs=22050Hz; and Passband cut-off frequency fc=4KHz.
Step 2: Method filter design using Bartlett-Hanning window Step 3:
Filter order is predetermined, Nf=9; A total number of filter coefficients is larger by 1, ie N=Nf+1=10; and Coefficients have indices between 0 and 9. Step 4: The Bartlett-Hanning window function coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 9

2.4.4.2 Example 2
Step 1:
Type of filter high-pass filter Filter specifications:

Step 2: Method filter design using Bartlett-Hanning window Step 3: Filter order is predetermined, Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10. Step 4: The Bartlett-Hanning window function coefficients are found via expression:
Normalized cut-off frequency c may be calculated using expression:
h[n] = w[n] * hd[n] ; 0 n 10

2.4.4.3 Example 3
Step 1:
Type of filter band-pass filter Filter specifications:

Filter order Nf=12; Sampling frequency fs=22050Hz; and Passband cut-off frequency fc1=2KHz, fc2=5KHz.
h[n] = w[n] * hd[n] ; 0 n 12

2.4.4.4 Example 4
Step 1:
Type of filter band-stop filter Filter specifications:

Filter order Nf=12; Sampling frequency fs=22050Hz; and Passband cut-off frequencies fc1=2KHz, fc2=6KHz.
The values of coefficients (rounded to six digits) are obtained by combining the values of M, c1 and c2 with expression for the impulse reaponse coefficients of the ideal band-stop filter:
h[n] = w[n] * hd[n] ; 0 n 12

2.4.5 Filter design using Hamming window 2.4.5.1 Example 1

Step 1: Type of filter low-pass filter Filte specifications:

Sampling frequency fs=22050Hz; Passband cut-off frequency fc1=3KHz; Stopband cut-off frequency fc2=6KHz; and Minimum stopband attenuation 40dB.
Step 2: Method filter design using Hamming window Step 3: For the first iteration, the filter order can be determined from the table 2-4-1 below. Normalized length of the main lobe for N=20 0.1 0.2 0.21 0.21 0.23 0.31 0.32 0.43 Minimum stopband attenuation of window function 13 dB 26 dB 31 dB 36 dB 41 dB 46 dB 58 dB 91 dB Minimum stopband attenuation of designed filter 21 dB 26 dB 44 dB 39 dB 53 dB 51 dB 75 dB 109 dB
Window function Rectangular Triangular (Bartlett) Hann Bartlett-Hanning Hamming Bohman Blackman Blackman-Harris
Transition region for N=20 0.041 0.11 0.12 0.13 0.14 0.2 0.2 0.32
Table 2-4-1. Comparison of window functions Using the specifications for the transition region of the required filter, it is possible to compute cut-off frequencies:
The required transition region of the filter is:
The transition region of the filter to be designed is approximately twice that of the filter given in the table above. For the first iteration, the filter order can be half of that.

Filter order is Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10.
Step 4: The Hamming window function coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 10

Step 7: Analyse in the frequency domain is performed using the Filter Designer Tool program.
Figure 2-4-33 illustrates the frequency characteristic of the resulting filter. It is obtained in the Filter Designer Tool program. As seen, the resulting filter doesnt satisfy the required specifications. The attenuation at the frequency of 6KHz amounts to 32.96dB only, which is not sufficient. It is necessary to increase the filter order. Another way is to compute the attenuation at the frequency of 6KHz. Starting from the impulse response, the first thing that should be done is the Z-transform. It is explained, along with Fourier transformation, in chapter 2-2-2.
It is easy to obtain the Fourier transformation via the Z-transform:
According to the analyse performed using Filter Designer Tool, it is confirmed that the filter order has to be incremented.
The filter order is incremented by two. The whole process of designing filter is repeated from the step 3. Step 3: Filter order is Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The Hamming window function coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

Figure 2-4-34 illustrates the frequency characteristic of the resulting filter. As seen, the resulting filter doesnt satisfy the given specifications. The attenuation at the frequency of 6KHz amounts to 45.26dB only, which is not sufficient. It is necessary to change the filter order. Filter realization: Figure 2-4-35 illustrates the direct realization of designed FIR filter, whereas figure 2-4-36 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.5.2 Example 2
Step 1:
Type of filter high-pass filter Filte specifications:

Step 2: Method filter design using Hamming window Step 3: Filter order is predetermined, Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10. Step 4: The Hamming window function coefficients are found via expression:
The Hamming window function is one of rare standard windows where w[0] > 0 is in effect. Step 5: The ideal high-pass filter coefficients (ideal filter impulse response) are expressed as:
h[n] = w[n] * hd[n] ; 0 n 10

2.4.5.3 Example 3
Step 1:
Type of filter band-pass filter Filte specifications:

Filter order Nf=12; Sampling frequency fs=16KHz; Passband cut-off frequency fc1=2KHz, fc2=5KHz.
Step 2: Method filter design using Hamming window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The Hamming window function coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-39 illustrates the direct realization of designed FIR filter, whereas figure 2-4-40 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle elements.
2.4.5.4 Example 4
Step 1:
Type of filter band-stop filter Filte specifications:

Step 2: Method filter design using Hamming window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The Bartlett-Hanning window function coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

The first example (low-pass filter designed using Hamming window) explains the algorithm used to compute the needed filter order when it is unknown. The filter order can also be found using Kaiser window, after which the number of iterations, ie correction steps is reduced. The forth example explains the way of designing a band-stop filter. As can be seen, the impulse response of the resulting filter contains large number of zero values, which results in reducing the number of multiplication operations in design process. These zeros appear in impulse response because of the stopband width which amounts to 0.5 = /2. If it is possible to specify the sampling frequency from a certain frequency range, you should tend to specify the value representing a multiple of the passband width. The number of zeros contained in an impulse response is larger in this case, whereas the number of multiplications, otherwise the most demanding operation in filtering process, is less. In the given example, only 5 multiplication operations are performed in direct realization of a twelfthorder FIR filter, ie 3 multiplication operations in optimized realization structure.
2.4.6 Filter design using Bohman window 2.4.6.1 Example 1


Filter order Nf=10; Sampling frequency fs=20KHz; and Passband cut-off frequency fc=5KHz.
Step 2: Method filter design using Bohman window Step 3: Filter order is Nf=10; A total number of filter coefficients is larger by 1, ie N=Nf+1=11; and Coefficients have indices between 0 and 10. Step 4: The Bohman window function coefficients are found via expression:
Normalized cut-off frequency c may be computed using expression:
Step 6:
The designed FIR filter coefficients are found via expression:
h[n] = w[n] * hd[n] ; 0 n 10

2.4.6.2 Example 2
Step 1:

Sampling frequency fs=22050Hz; Passband cut-off frequency fc1=1.5KHz; Stopband cut-off frequency fc2=4KHz; and Minimum stopband attenuation 35dB.
Step 2: Method filter design using Bohman window Step 3: The needed filter order is determined via iteration. It is necessary to specify the initial value of filter order that is to be changed as many times as needed. This value is specified according to the data contained in the table 2-4-2 below: Normalized length of the main lobe for N=20 0.1 0.2 0.21 0.21 0.23 0.31 0.32 0.43 Minimum Transition stopband region for N=20 attenuation of window function 0.041 0.11 0.12 0.13 0.14 0.2 0.2 0.32 13 dB 26 dB 31 dB 36 dB 41 dB 46 dB 58 dB 91 dB Minimum stopband attenuation of designed filter 21 dB 26 dB 44 dB 39 dB 53 dB 51 dB 75 dB 109 dB
Window function Rectangular Triangular (Bartlett) Hann Bartlett-Hanning Hamming Bohman Blackman Blackman-Harris
Table 2-4-2. Comparison of window functions According to the specifications for the transition region of required filter, it is possible to compute cut-off frequencies:
The required transition region is:
The transition region of the filter to be designed is somewhat wider than that of the filter given in table 2-4-2. For the first iteration, during filter design process, the filter order can be lower. Unlike the low-pass FIR filter, the high-pass FIR filter must be of even order. The same applies to band-pass and band-stop filters. It means that filter order can be changed in odd steps. The smallest change is 2. In this case, the filter order, comparing to that from the table (20), can be decreased by 2 for the purpose of defining initial value. Filter order is Nf=18; A total number of filter coefficients is larger by 1, ie N=Nf+1=19; and Coefficients have indices between 0 and 18. Step 4: The coefficients of Bohman window are found via expression:
Normalized cut-off frequency c is equal to passband cut-off frequency:
h[n] = w[n] * hd[n] ; 0 n 18

Figure 2-4-45 illustrates the frequency characteristic of the resulting filter. It is obtained in the Filter Designer Tool program. As seen, the resulting filter satisfies the required specifications. The attenuation at the frequency of 1500Hz amounts to 37.04dB only, which is more than enough. However, the final objective when designing a filter is to find a minimum filter order that satisfies the filter specifications. Since the filter order must be changed by an even number, the specified value is -2. The filter order is decreased by 2. The whole process of designing filters is repeated from the step 3 on. Step 3: Filter order is Nf=16; A total number of filter coefficients is larger by 1, ie N=Nf+1=17; and Coefficients have indices between 0 and 16. Step 4: The coefficients of Bohman window are found via expression:
Normalized cut-off frequency c is equal to the passband cut-off frequency:
h[n] = w[n] * hd[n] ; 0 n 16

Step 7:
Analyse in the frequency domain is performed using the Filter Designer Tool program.
Figure 2-4-46 illustrates the frequency characteristic of the resulting filter. The figure is obtained in the Filter Designer Tool program. As seen, the resulting filter satisfies the required specifications. The objective is to find the minimum filter order. Since the attenuation is close to the required attenuation, the correct order is probably 16. However, it is necessary to check it. Since the filter order must be changed by an even number, the specified value is -2. The filter order is decreased by 2, therefore. The whole process of designing filter is repeated from the step 3 on. Step 3: Filter order is predetermined, Nf=14; A total number of filter coefficients is larger by 1, ie N=Nf+1=15; and Coefficients have indices between 0 and 14. Step 4: The coefficients of Bohman window are found via expression:
Normalized cut-off frequency c is equal to passband cut-off freqency:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c with expression for the impulse reaponse coefficients of the ideal high-pass filter:
Step 6: The designed FIR filter coefficients are found via expression: h[n] = w[n] * hd[n] ; 0 n 14 The FIR filter coefficients h[n] rounded to 6 digits are:
Figure 2-4-47 illustrates the frequency characteristic of the resulting filter. The figure is obtained in the Filter Designer Tool program. As seen, the resulting filter doesnt satisfy the required specifications. The attenuation at the frequency of 1500KHz amounts to 26.24dB only, which is not sufficient. The previous value (Nf=16) represents the minimum FIR filter order that satisfies the given specifications. The filter order is Nf=16, whereas impulse response of the resulting filter is as follows:
2.4.6.3 Example 3
Step 1:
Type of filter band-pass filter Filte specifications:

Filter order Nf=12; Sampling frequency fs=44100Hz; and Passband cut-off frequency fc1=4KHz, fc2=15025Hz.
Step 2: Method filter design using Bohman window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The coefficients of Bohman window are found via expression:
Step 6: The coefficients of designed FIR filter are found via expression: h[n] = w[n] * hd[n] ; 0 n 12 The FIR filter coefficients h[n] rounded to 6 digits are:
2.4.6.4 Example 4
Step 1:

Step 2: Method filter design using Bohman window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The coefficients of Bohman window are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

2.4.7 Filter design using Blackman window 2.4.7.1 Example 1

Step 1:
Type of filter low-pass filter Filte specifications:

Filter order Nf=12; Sampling frequency fs=44100KHz; Passband cut-off frequency fc=15KHz; and Attenuation of 0dB at 0Hz 0dB.
Step 2: Method Filter design using Blackman window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman window are found via expression:
Normalized cut-off frequency c can be computed using expression:
The values of coefficients are obtained (rounded to six digits) by combining the values of M and c with expression for the impulse response coefficients of the ideal low-pass filter:
h[n] = w[n] * hd[n] ; 0 n 12

The resulting coefficients must be scaled in order to provide attenuation of 0dB at 0Hz. In order to provide attenuation of 0 dB, the following condition must be met:
The sum of the previously obtained coefficients is:
As the sum is greater than one, it is necessary to divide all coefficients of the impulse response by 1.000274. After division, these coefficients have the following values:
The sum of scaled coefficients is equal to 1, which means that attenuation at 0Hz frequency amounts to 0dB. Note that these coefficients cannot be used in designing a FIR filter safe from filtering overflow. In order to prevent a filtering overflow from occuring it is necessary to satisfy the condition below:
The resulting filter doesnt meet this condition. Negative coefficients in impulse response make that both conditions cannot be met. The sum of apsolute values of coefficients in the resulting filter is:
The sum of coefficients apsolute values before scaling amounts to 1.37001 (1.3696361.000274). After scaling, it is somewhat less, so it is less likely that an overflow occurs. In such cases, possible filtering overflows are not dangerous. Namely, most processors containing hardware multipliers (which is almost necessary for filtering) have registers with extended band. In this case, it is far more important to faithfully transmit a direct signal to a FIR filter output. Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-54 illustrates the direct realization of designed FIR filter, whereas figure 2-4-55 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.7.2 Example 2
Step 1:

Filter order Nf=12; Sampling frequency fs=22050Hz; Passband cut-off frequency fc=4KHz; Prevention of possible filtering overflows.
Step 2: Method filter design using Blackman window Step 3: Filter order is Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman window function are found via:
Step 6: The designed FIR filter coefficients are found via expression: h[n] = w[n] * hd[n] ; 0 n 12 The FIR filter coefficients h[n] rounded to 6 digits are:
In order to prevent filtering overflow, the following condition must be met:
The sum of absolute values of the resulting FIR filter coefficients is:
The obtained coefficients must be scaled (divided) by 1.34807. After that, their values are:
2.4.7.3 Example 3
Step 1: Type of filter band-pass filter Filte specifications:

Filter order Nf=12; Sampling frequency fs=44100Hz; Passband cut-off frequency fc1=4KHz, fc2=15025Hz; Prevention of possible filtering overflow.
Step 2: Method filter design using Blackman window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman window function are found via expression:
Normalized cut-off frequencies c1 and c2 can be computed using expressions:
h[n] = w[n] * hd[n] ; 0 n 12

In order to prevent filtering overflows, the following condition must be met:
The obtained coefficients must be scaled (divided) by 1.102056. After this, their values are:
2.4.7.4 Example 4
Step 1:

Filter order Nf=12; Sampling frequency fs=16000Hz; Passband cut-off frequencies fc1=2KHz, fc2=6KHz; Prevention of possible filtering overflows.
Step 2: Method filter design using Blackman window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman window are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

2.4.8 Filter design using Blackman-Harris window 2.4.8.1 Example 1


Filter order Nf=12; Sampling frequency fs=44100KHz; Passband cut-off frequency fc=15KHz; Attenuation of 0dB at 0Hz.
Step 2: Method filter design usng Blackman-Harris window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman-Harris window are found via expression:
Normalized cut-off frequency c may be calculated using expression:
h[n] = w[n] * hd[n] ; 0 n 12

The resulting coefficients must be scaled in order to provide attenuation of 0dB at 0Hz. To provide 0 dB attenuation, the following condition must be met:
The sum of the previously obtained coefficients is:
As the sum is greater than one, it is necessary to divide all the impulse response coefficients by 0.977947. After this, the values of these coefficients are:
The sum of scaled coefficients is equal to 1, which means that attenuation at 0Hz frequency amounts to 0dB. Note that these coefficients cannot be used in designing a FIR filter safe from filtering overflow. In order to prevent a filtering overflow from occuring it is necessary to satisfy the condition below:
The resulting filter doesnt meet this condition. Negative coefficients in impulse response indicate that both conditions cannot be met. The sum of apsolute values of coefficients in the resulting filter is:
The sum of coefficients apsolute values before scaling amounts to 1.37001 (1.3696361.000274). After scaling, the sum of coefficients apsolute values is somewhat less, so it is less possible that an overflow occurs. In such cases, possible filtering overflows are not dangerous. Namely, most processors containing hardware multipliers (which is almost necessary for filtering) have registers with extended band. In this case, it is far more important to faithfully transmit a direct signal to a FIR filter output. Step 7: The filter order is predetermined. There is no need to additionally change it.
2.4.8.2 Example 2
Step 1: Type of filter high-pass filter Filte specifications:

Filter order Nf=12; Sampling frequency fs=22050Hz; Passband cut-off frequency fc=4KHz; Prevention of filtering overflows.
Step 2: Method filter design using Blackman-Harris window Step 3: Filter order is Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman-Harris window are found via:
Step 5:
The ideal high-pass filter coefficients (ideal filter impulse response) are expressed as:
The values of coefficients (rounded to six digits) are obtained by combining the values of M and c with the expression for the impulse response coefficients of the ideal high-pass filter:
h[n] = w[n] * hd[n] ; 0 n 12

In order to prevent filtering overflow, the following condition must be met:
2.4.8.3 Example 3
Step 1:
Type of filter band-pass filter Filter specification:

Filter order Nf=12; Sampling frequency fs=44100Hz; Passband cut-off frequencies fc1=4KHz, fc2=15025Hz; and Prevention of possible filtering overflows.
Step 2: Method filter design using Blackman-Harris window Step 3: Filter order is predetermined, Nf=12; A total number of filter coefficients is larger by 1, ie N=Nf+1=13; and Coefficients have indices between 0 and 12. Step 4: The coefficients of Blackman-Harris window are found via expression:
h[n] = w[n] * hd[n] ; 0 n 12

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter reaization: Figure 2-4-66 illustrates the direct realization of designed FIR filter, whereas figure 2-4-67 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element.
2.4.8.4 Example 4
Step 1:
Type of filter band-stop filter Filter specification:

Filter order Nf=20; Sampling frequency fs=16000Hz; Passband cut-off frequency fc1=2KHz, fc2=6KHz; and Prevention of possible filtering overflows.
Step 2: Method filter design using Blackman-Harris window Step 3: Filter order is predetermined, Nf=20; A total number of filter coefficients is larger by 1, ie N=Nf+1=21; and Coefficients have indices between 0 and 20. Step 4: The coefficients of Blackman-Harris window are found via expression:
h[n] = w[n] * hd[n] ; 0 n 20

Step 7: The filter order is predetermined. There is no need to additionally change it. Filter realization: Figure 2-4-68 illustrates the direct realization of designed FIR filter, whereas figure 2-4-69 illustrates optimized realization structure of designed FIR filter which is based on the fact that all FIR filter coefficients are, for the sake of linear phase characteristic, symmetric about their middle element. This FIR filter is an excellent example showing the importance of the sampling frequency. It is specified to give the passband amounting to 0.5. This causes most impulse response coefficients of the resulting FIR filter to be zeros. It further makes the filter realization structure simpler. As for optimized FIR filter design, there are only 4 multiplications, even though the filter is of 20th order. Unfortunatelly, the buffer length cannot be minimized. It is fixed and corresponds to the filter order. However, it is possible to affect design complexity, whether it is hardware or software implementation.
2.5 Finite word-length effects

There are hardware and software FIR filter realizations. Regardless of which of them is used, a problem known as the finite word-length effect exists in either case. One of the objectives, when designing filters, is to lessen the finite word-length effects as much as possible, thus satisfying the initiative requirements (filter specifications). On software filter implementation, it is possible to use either fixed-point or floating-point arithmetic. Both representations of numbers have some advantages and disadvantages as well. The fixed-point representation is used for saving coefficients and samples in memory. Most commonly used fixed-point format is when one bit denotes a sign of a number, ie 0 denotes a positive, whereas 1 denotes a negative number, and the rest of bits denote the value of a number. This is mostly used to represent numbers in the range -1 to +1. Numbers represented in the fixed-point format are equidistantly quantized with the quantization step 1/2N-1, where N is the number of a bit used for saving the value. As one bit is a sign bit, there are N-1 bits available for value quantization. The maximum error that may occur during quantization is 1/2 quantization step, that is 1/2N. It can be noted that accuracy increases as the number of bits increases. Table 2-5-1 shows the values of quantization steps and maximum errors made due to quantization process in the fixed-point presentation. Number of exact decimal points 1 2 4 9 19
Bit number 4 8 16 32 64
Range of numbers (-1, +1) (-1, +1) (-1, +1) (-1, +1) (-1, +1)
Quantization step 0.125 0.0078125 3.0517578125*10-5
Mx. quantization error 0.0625 0.00390625 1.52587890625*10-5
4.6566128730774*10-10 2.3283064365387*10-10 1.0842021724855*10-19 5.4210108624275*10-20
Table 2-5-1. Quantization of numbers represented in the fixed-point format The advantage of this presentation is that quantization errors tend to approximate 0. It means that errors are not accumulated in operations performed upon fixed-point numbers. One of disadvantages is a smaller accuracy in coefficients representation. The difference between actual sampled value and
quantized value, ie the quantization error, is smaller as the quantization level decreases. In other words, the effects of the quantization error are negligible in this case. The floating-point arithmetic saves values with better accuracy due to dynamics it is based on. Floating-point representations cover a much wider range of numbers. It also enables an appropriate number of digits to be faithfully saved. The value normally consists of three parts. The first part is, similar to the fixed-point format, represented by one bit known as the sign bit. The second part is a mantissa M, which is a fractional part of the number, and the third part is an exponent E, which can be either positive or negative. A number in the floating-point format looks as follows:
where M is the mantissa and E is the exponent. As seen, the sign bit along with mantissa represent a fixed-point format. The third part, ie exponent provides the floating-point representation with dynamics, which further enables both extremely large and extremely small numbers to be saved with appropriate accuracy. Such numbers could not be represented in the fixed-point format. Table 2-5-2 below provides the basic information on floatingpoint representation for several different lengths. Number of exact Bit number Mantissa size Exponent size Band decimal points 16 32 7 23 8 8 2.3x10-38 .. 3.4x1038 1.4x10-45 .. 3.4x1038 2 6-7
Table 2-5-2. Quantization of numbers represented in the floating-point format It is not possible to determine the quantization step in the floating-point representation as it depends on exponent. Exponent varies in a way that the quantization step is as small as possible. In this number presentation, special attention should be paid to the number of digits that are saved with no error. The floating-point arithmetic is suitable for coefficient representation. The errors made in this case are considerably less than those made in the fixed-point arithmetic. Some of disadvantages of this presentation are complex implementation and errors that do not tend to approximate 0. The problem is extremely obvious when the operation is performed upon two values of which one is much less than the other. Exemplo FIR filter coefficients:
{0.151365, 0.400000, 0.151365}

Coefficients need to be represented as 16-bit numbers in the fixed-point and floating-point formats. If we suppose that numbers range between -1 and +1, then quantization level amounts to 1 / 2^16 = 0.0000152587890625. After quantization, the filter coefficients have the following values:
{0.1513671875, 0.399993896484375, 0.1513671875}

Quantization errors are:
{-0.0000021875, 0.000006103515625, -0.0000021875}

If filter coefficients are represented in the floated-point format, it is not possible to determine quantization level. In this case, the coefficients have the following values:
{0.151364997029305, 0.400000005960464, 0.151364997029305}

Quantization errors produced while representing coefficients as 16-bit numbers in the floating-point format are:
{0.000000002970695, -0.000000005960464, 0.000000002970695}

As seen, a coefficient error is less in the floating-point representation. Floating-point arithmetic can also be expressed in terms of fixed-point arithmetic. For this reason, the fixed-point arithmetic is more often implemented in digital signal processors. The finite word-length effect is the deviation of FIR filter characteristic. If such characteristic still meets the filter specifications, the finite word-length effects are negligible. As a result of greater error in coefficients representation, the finite word-length effects are more prominent in fixed-point arithmetic. These effects are more prominent for IIR filters for their feedback property than for FIR filters. In addition, coefficient representation can cause IIR filters to become instable, whereas it cannot affect FIR filters that way. FIR filters keep their linear phase characteristic after quantization. The reason for this is the fact that the coefficients of a FIR filter with linear phase characteristic are symmetric, which means that the corresponding pairs of coefficients will be quantized to the same value. It results in the impulse response symmetry remaining unchanged. After all mentioned, it is easy to notice that finite word length, used for representing coefficients and samples being processed, causes some problems such as:

Coefficient quantization errors; Sample quantization errors (quantization noise); and Overflow errors.
2.5.1 Coefficient Quantization

The coefficient quantization results in FIR filter changing its transform function. The position of FIR filter zeros is also changed, whereas the position of its poles remains unchanged as they are located in z=0. Quantization has no effect on them. The conclusion is that quantization of FIR filter coefficients cannot cause a filter to become instable as is the case with IIR filters.
Even though there is no danger of FIR filter destabilization, it may happen that transfer function is deviated to such an extent that it no longer meets the specifications, which further means that the resulting filter is not suitable for intended implementation. The FIR filter quantization errors cause the stopband attenuation to become lower. If it drops below the limit defined by the specifications, the resulting filter is useless. Transfer function changes occurring due to FIR filter coefficient quantization are more effective for high-order filters. The reason for this is the fact that spacing between zeros of the transfer function get smaller as the filter order increases and such slight changes of zero positions affect the FIR filter frequency response.
2.5.2 Samples Quantization

Another problem caused by the finite word length is sample quantization performed at multipliers output (after filtering). The process of filtering can be represented as a sum of multiplications performed upon filter coefficients and signal samples appearing at filter input. Figure 2-5-1 illustrates block diagram of input signal filtering and quantization of result as well.
Multiplication of two numbers each N bits in length, will give a product which is 2N bits in length. These extra N bits are not necessary, so the product has to be truncated or rounded off to N bits, producing truncation or round-off errors. The later is more preferable in practice because in this case the mid-value of quantization error (quantization noise) is equal to 0. In most cases, hardware used for FIR filter realization is designed so that after each individual multiplication, a partial sum is accumulated in a register which is 2N in length. Not before the process of filtering ends, the result is quantized on N bits and quantization noise is introduced, thus drastically reduced. Quantization noise depends on the number of bits N. The quantization noise is reduced as the number of bits used for sample and coefficient representation increases. Both filter realization and position of poles affect the quantization noise power. As all FIR filter poles are located in z=0, the effect of filter realization on the quantization noise is almost negligible.
2.5.3 Overflow
Overflow happens when some intermediate results exceed the range of numbers that can be represented by the given word-length. For the fixed-point arithmetic, coefficients and samples values are represented in the range -1 to +1. In spite of the fact that both FIR filter input and output samples are in the given range, there is a possibility that an overflow occurs at some point when the results of multiplications are added together. In other words, an intermediate result is greater than 1 or less than -1. Example: Assume that it is needed to filtrate input samples using a second-order filter.
Such filter has three coefficients. These are: {0.7, 0.8, 0.7}. Input samples are: { ..., 0.9, 0.7, 0.1, ...} By analyzing the steps of the input sample filtering process, shown in the table 2-5-3 below, it is easy to understand how an overflow occurs in the second step. The final sum is greater than 1. Filter coefficients 0.7 0.8 0.7 Table 2-5-3. Overflow As the range of values, defined by the fixed-point presentation, is between -1 and +1, the results of the filtering process will be as shown in the table 2-5-4. Filter coefficients 0.7 0.8 0.7 Table 2-5-4. Overflow effects As mentioned, an overflow occurs in the second step. Instead of desired value +1.19, the result is an undesirable negative value -0.81. This difference of -2 between these two values is explained in Figure 2-5-2 below. Input sample 0.9 0.7 0.1 Intermediate result 0.63 0.63 + 0.56 - 2 = -0.81 -0.81 - 0.07 = -0.88 Input sample 0.9 0.7 0.1 Intermediate result 0.63 0.63 + 0.56 = 1.19 1.19 + 0.07 = 1.26
However, if some intermediate result exceeds the range of presentation, it does not necessarily cause an overflow in the final result. The apsolute value of the result is less than 1 in this case. In other words, as long as the final result is within the word-length, overflow of partial results is not of the essence. This situation is illustrated in the following example. Example: The second-order filter has three coefficients. These are: {0.7, 0.8, 0.7} Input samples are: { ..., 0.9, 0.7, -0.5, ...} The desired intermediate results are given in the table 2-5-5. Filter coefficients 0.7 0.8 0.7 Table 2-5-5. Desired intermediate results Input sample 0.9 0.7 -0.5 Intermediate result 0.63 0.63 + 0.56 = 1.19 1.19 - 0.35 = 0.84
As seen, some intermediate results exceed the given range and two overflows occur. Refer to the table 2-5-6 below. Filter coefficients 0.7 0.8 0.7 Input sample 0.9 0.7 -0.5 Intermediate result 0.63 0.63 + 0.56 - 2 = -0.81 -0.81 - 0.35 + 2 = 0.84
Table 2-5-6. Obtained intermediate results So, in spite of the fact that two overflows have occured, the final result remained unchanged. The reason for this is the nature of these two overflows. The first one has decremented the final result by 2, whereas the second one has incremented the final result by 2. This way, the overflow effect is annuled. The first overflow is called a positive overflow, whereas the later is called a negative overflow. Note: If the number of positive overflows is equal to the number of negative overflows, the final result will not be changed, ie the overflow effect is annuled. Overflow causes rapid oscillations in the input sample, which further causes highfrequency components to appear in the output spectrum. There are several ways to lessen the overflow effects. Two most commonly used are scaling and saturation. It is possible to scale FIR filter coefficients to avoid overflow. A necessary and sufficient condition required for FIR filter coefficients in this case is given in the following expression:
where: bk are the FIR filter coefficients; and N is the number of filter coefficients. If, for any reason, it is not possible to apply scaling then the overflow effects can be lessened to some extend via saturation. Figure 2-5-3 illustrates the saturation characteristic.
When the saturation characteristic is used to prevent an overflow, the intermediate result doesnt change its sign. For this reason, the oscillations in the output signal are not so rapid and undesirable high-frequency components are attenuated. Lets see what happens if we apply the saturation characteristic to the previous example: Exemplo Again, it is needed to filtrate input samples using a second-order filter. Such filter has three coefficients. These are: {0.7, 0.8, 0.7} Input samples are: { ..., 0.9, 0.7, 0.1, ...} The desirable intermediate results are shown in the table 2-7-7 below. Filter coefficients 0.7 0.8 0.7 Input sample 0.9 0.7 0.1 Intermediate result 0.63 0.63 + 0.56 = 1.19 1.19 + 0.07 = 1.26
Table 2-5-7. Desirable intermediate results
As the range of values, defined by the fixed-point presentation, is between -1 and +1, and the saturation characteristic is used as well, the intermediate results are as shown in the table 2-5-8. Filter coefficients 0.7 0.8 0.7 Input sample 0.9 0.7 0.1 Intermediate result 0.63 0.63 + 0.56 = 1 1 + 0.07 = 1
Table 2-5-8. Intermediate results and saturation characteristic The resulting sum is not correct, but the difference is far smaller than when there is no saturation: Without saturation: = 1.26 - (-0.88) = 2.14 With saturation: = 1.26 - 1 = 0.26 As seen from the example above, the saturation characteristic lessens an overflow effect and attenuates undesirable components in the output spectrum.

Projetos de Filtro Digital - (Digital Filter Design)

Uploaded by

Document Information

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Projetos de Filtro Digital - (Digital Filter Design)

Uploaded by

Copyright:

Available Formats

Digital Filter Design

1.1. Conceitos bsicos de filtragem digital 1.2. Tipos de filtros digitais

1/1 Conceitos bsicos de filtragem digital

1.2.Types de filtros digitais

As caractersticas bsicas do Infinite Impulse Response (IIR) so:

Captulo 2: Finite Impulse Response (FIR) Filtro

2.1. 2.2. 2.3. 2.4. 2.5.

2,2 Finitas impulso resposta (FIR) filtro desenho mtodos

2.2.1 Conceitos bsicos e FIR filtro especificao

Figura 2 .-2.-1a. Filtro digital passa-baixa especificao

Figure2-2-1b. Filtro digital passa-baixa especificao

Frequncia de normalizao pode ser expresso da seguinte forma:

fs uma frequncia de amostragem; f a frequncia de normalizar e normalizado frequncia.

Figura 2 .-2.-2a. High-pass filtro digital especificao

Figura 2-2-3-A. Band-pass filtro digital especificao

Figura 2-2-4-A. Band-stop filtro digital especificao

Figura 2-2-4-B. Band-stop filtro digital especificao

Figura 2-2-5 ilustra o espectro de frequncias do sinal.

No sistema de coordenadas polares, o nmero complexo z pode ser expressa como:

Figura 2-2-6. Transformada de Fourier no plano z

2.2.2.1 Transformada discreta funo de sistemas de tempo

onde h (n) o impulso de resposta do sistema em tempo discreto.

Utilizando a seguinte expresso:

Um exemplo de hardware realizao deste filtro ilustrado na Figura 2-2-8.

2.2.2.2 Efeito dos plos e zeros da funo de transferncia

2.2.3 Ideal filtro aproximao

2.2.4 Filtro FIR desenho utilizando janela funes

2.2.5 realizao filtro FIR

2.2.5.1 direto realizao

Figura 2-2-11. FIR filtro directo realizao

representa arredondamento para um valor menor.

2.2.5.2 direto transpor realizao

2.2.5.3 realizao Cascade

M o nmero de sees e ak1, ak2 so os coeficientes de multiplicao da seco K.

2.2.5.4 Optimizado realizao

Figura 2-2-15. Optimizado para a realizao mpar frequncia de resposta

Figura 2-2-16. Optimizado para a realizao ainda frequncia de resposta

Janela 2/3 funes

Janela funo Retangular

Triangular (Bartlett) Hann Bartlett, Hanning Hamming BOHMAN Blackman Blackman-Harris

0.2 0.21 0.21 0.23 0.31 0.32 0.43

0.11 0.12 0.13 0.14 0.2 0.2 0.32

Normalized as freqncias so obtidas no seguinte maneira:

fn2 - fn1 = 0.2

2.3.1 Rectangular Window

2.3.2 Triangular (Bartlett) window

Figure 2-3-4 illustrates coefficients in the time-domain.

2.3.3 Hann Window

2.3.4 Bartlett-Hanning Window

2.3.5 Hamming Window

2.3.6 Bohman Window

2.3.7 Blackman Window

2.3.8 Blackman-Harris Window

The Blackman-Harris window coefficients are expressed as:

2.3.9 Blackman-Harris Window

Rectangular window; Hann window; Hamming window; and Blackman window.

Table 2-3-3. The coefficients of generalized cosine window functions

2.3.10 Kaiser Window

low-pass filter; high-pass filter; band-pass filter; and band-stop filter.

2.4.1 Filter design using Rectangular window 2.4.1.1 Example 1

Filter order N=10

where M is the index of middle coefficient.

Normalized cut-off frequency c can be calculated using the following expression:

The FIR filter coefficients h[n] rounded to 6 digits are: