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Conectando hardware e definindo níveis.

De Ax-Fx II Wiki.

Lista de portas de E / S

INSTR (frontal): plugue de fone de 1/4 ”, desbalanceado, condicionado para uso de guitarra, comutação automática, 1 Megaohm (ajustável), nível de instrumento de + 16dBu ENTRADA 1 (traseira): plugue de 1/4”, desbalanceado, condicionado para uso de guitarra, comutação automática, 1 Megaohm (ajustável), nível de instrumento de + 16dBu ENTRADA 2 (traseira): XLR Fêmea e 1/4 "combo, L / R, balanceada, 1 Megaohm, nível de linha de + 20dBu INPUT 3 (traseira) : Plugue para fone de 1/4 ”, L / R, balanceado, projetado para aplicações de ganho unitário como 4CM, inserções estéreo duplas ou para uso geral, 1 Megaohm, nível de linha + 20dBu INPUT 4 (traseira): plugue de 1/4” L / R, balanceado, projetado para aplicações de ganho unitário como 4CM, inserções estéreo duplas ou de uso geral, 1 Megaohm, nível de linha 20dBu SAÍDA 1: XLR, L / R, balanceado, comutador de aterramento, linha de 600 Ohm, + 20dBu nível SAÍDA 1: tomada de telefone de 1/4 ", L / R, HumBuster, comutador de aterramento, nível de linha de 600 Ohm, + 20dBu SAÍDA 2: XLR, L / R, balanceada, comutador de aterramento, nível de linha de 600 Ohm e 20dBu Saída 3: 1/4 "jack de telefone, L / R, HumBuster, 600 Ohm, Nível de linha de + 20dBu SAÍDA 4: Tomada de 1/4 ", L / R, HumBuster, nível de linha de 600 Ohm, + 20dBu Conversão A / D e D / A: 48 kHz, 24 bits, faixa dinâmica de 114 dB, 20 Hz - 20 Resposta de freqüência kHz (0 / -1 dB) E / S digital: S / PDIF (coaxial RCA), AES (XLR), áudio USB 8x8, taxa de amostragem 48 kHz MIDI: IN, OUT, THRU Saída de fone de ouvido: 1/4 " jack estéreo, 35 Ohm Portas do pedal Expresssion: 2x TRS de 1/4 ", 10-100 kOhm, porta momentânea ou com travamento FASLINK II: XLR Fêmea.

As portas XLR e 1/4 "da saída 1 podem ser usadas simultaneamente. Elas são armazenadas em buffer.

As saídas XLR são protegidas contra phantom power do console.

O Ax-Fx III possui vários blocos de Entrada e Saída que podem ser colocados em qualquer lugar da grade.

IN 1 (INSTRUMENT): plugue de 1/4 ”, desbalanceado, máximo + 16dBu (condicionado ao uso de guitarra), nível de instrumento IN 1 (traseiro): plugue de 1/4”, desbalanceado, máximo + 20dBu IN 2 (FX RTN ): 1/4 ", L / R, balanceado, 1 Megaohm, máx. 20dBu OUT 1 PRINCIPAL: XLR, balanceado, 600 ohms, saída máxima + 20dBu OUT 1 PRINCIPAL: jaque do telefone de 1/4", desbalanceado cancelamento) OUT 2 (FX SEND): 1/4 ", L / R, desequilibrado, Humbuster, 600 ohm, max. E / S Digital de 20 dBu: S / PDIF (RCA Coaxial), AES (XLR), Áudio USB.

As portas XLR e 1/4 "da saída 1 podem ser usadas simultaneamente. Elas são armazenadas em buffer.

As saídas XLR são protegidas contra phantom power do console.

"Ambas as saídas devem funcionar simultaneamente. Elas são, na verdade, armazenadas em buffer, portanto, mesmo que você produza uma em curto, ela não deve afetar a outra." fonte.

IN 1 (INSTRUMENTO): 1/4 ", mono, desbalanceado, 1 Megaohm (fixo), máx. 16dBu, nível de instrumento OUT 1 (MAIN): XLR, E / D, balanceado, 600 ohm, max.20dBu OUT 1 ( PRINCIPAL): 1/4 ", L / R, desequilibrado, Humbuster, 600 ohm, max. 20dBu IN 2 (RTN FX): 1/4 ", E / D, equilibrado, 1 Megaohm, max. 20dBu OUT 2 (ENVIO FX): 1/4", E / D, não balanceado, Humbuster, 600 ohm, max. Saída digital 20dBu S / PDIF, 24 bits, 48 kHz (fixa)

As portas XLR e 1/4 "da saída 1 podem ser usadas simultaneamente. Elas são armazenadas em buffer.

As saídas XLR são protegidas contra phantom power do console.

IN [PRE] (INSTR): 1/4 ", mono, nível de instrumento, desbalanceado, 1 Megaohm (dependendo da configuração de Impedância de entrada), máx. 16dBu, nível de instrumento OUT [PRE]: 1/4", L / R, ( L / Mono), desbalanceado, Humbuster, 600 ohm, máx 20dBu IN [POST]: 1/4 ", L / R, entrada de nível de linha (+ 4dBu), 1 Megaohm, balanceado, max 20dBu OUT [POST]: 1 / 4 ", L / R, desbalanceado, HumBuster, 600 ohms, max 20dBu.

Coisas que você deve saber.

Equilibrado versus desequilibrado.

Sinal de áudio não balanceado: transportado por um cabo de dois condutores. O cabo mais comum é um cabo de guitarra de 1/4 ", onde o fio terra envolve o fio positivo. Cabos desequilibrados geralmente são bons para executar sinal até vários metros (& lt; 10). Sinal de áudio balanceado: carregado em três cabo condutor que conecta uma entrada balanceada e saída balanceada. Os dois fios de sinal transportam cópias idênticas do sinal, com um dos fios 180 graus fora de fase com o outro, criando um diferencial. No lado receptor os dois sinais são trazidos de volta em fase um com o outro, e o ruído induzido será cancelado. Estes cabos geralmente usam XLR ou terminais tipo conector de ponta de anel (TRS).O fio terra envolve os fios de sinal e atua como uma blindagem. suporta o uso de um interruptor de aterramento e de longas distâncias de cabos sem ruído.

Uma caixa DI converte sinais entre equilibrados / desequilibrados.

Mais informações neste artigo da Wicked Wiki e na Wikipedia.

"O XLR só é necessário para cabos longos, onde existe o perigo de interferência no cabo. Qualquer coisa menor do que, digamos, 10 metros e desbalanceamento é suficiente." source (FX8) "Para a maioria dos usos, um cabo TS desequilibrado é bom. As entradas são balanceadas para que você possa obter ainda mais rejeição por meio do uso de um cabo TS-to-TRS a partir do envio do amplificador." source (Ax-Fx II) "As saídas XLR não balanceadas são do mesmo nível no Ax-Fx II. No Ax-Fx original as saídas XLR eram 6 dB mais quentes. Este não é o caso do II." fonte.

Nível do instrumento versus nível da linha.

Sinais de áudio operam em:

Nível do microfone: nível de saída mais baixo, usado com microfones e entradas de microfone em mixers. Nível do instrumento: nível de guitarras, baixos e pedais de efeitos. Nível de linha: nível de saída mais alto. O nível de linha pode ser: Nível do consumidor: -10dBv. Nível profissional: + 4dBu. Comumente usado em processadores de 19 ", entradas de linha em mixers e monitores.

As saídas na engrenagem do Fractal Audio operam no nível da linha. Alguns são ajustáveis entre +4 e -10 (veja abaixo).

O Ax-Fx III, Ax-Fx II, AX8 e FX8 suportam saídas Humbuster. Mais Informações.

Ganho de unidade.

O que é ganho de unidade? Ganho de unidade significa que o nível de entrada é igual ao nível de saída.

Quando a unidade ganha importância? Não é importante ao conectar o dispositivo a um amplificador ou console de mixagem. É importante em configurações onde o dispositivo está sendo usado como um processador somente de efeitos (por exemplo, como uma pedaleira ou loop de efeitos de um amplificador) ou ao usar o método de quatro cabos (4CM) para conectar a um amplificador de guitarra.

Como configurar para ganho de unidade? Para configurar uma entrada / saída para ganho unitário, ajuste o botão Output correspondente em sua posição máxima. Para testar: preencha a grade com shunts e você deve obter exatamente o mesmo sinal na saída que você colocou.

"Modo de ganho de unidade é um modo especial projetado para uso com o 4CM. Quando você gira os níveis de saída para cima o que quer que você coloque em você sai (assumindo todos os blocos de ganho de unidade na cadeia). Se você tiver um bloco de amplificador na cadeia, então você tem toneladas de ganho e, portanto, não tem mais ganho de unidade ". source "Com o volume Ax-Fx até o máximo, você deve empurrar + 20dBu para o amplificador, o que poderia prender as entradas ao amplificador. O modo de ganho da unidade é desejável apenas para o método de 4 cabos." fonte.

FX8 e ganho de unidade (do Manual do Proprietário):

Por que eu me importo que o FX8 é projetado para ganho de unidade?

R: O FX8 torna fácil obter ganho de unidade. Isso pode ser importante porque o tom do amplificador, a quantidade de distorção, a dinâmica e o ruído dependem do nível. Com ganho de unidade:

O nível do sinal da sua saída de guitarra pode alcançar sua entrada de amplificador sem ser alterado. Portanto, sua interação com o amplificador de guitarra soa e se sente da mesma forma, oferecendo uma experiência de reprodução transparente ao usar o FX8. O nível do seu FX SEND pode atingir o seu retorno FX sem ser alterado. Todo o sistema pode, portanto, executar de forma otimizada, sem alterações imprevisíveis ao nível, dinâmica ou ruído, quando você ativar o True Bypass ou ignorar todos os efeitos posteriores.

P: Como configuro o FX8 para ganho de unidade?

A: Você não precisa! Basta configurar de acordo com as instruções básicas da Seção 3. Uma predefinição vazia padrão deve soar no mesmo nível do True Bypass Mode.

P: O que eu posso fazer para, inadvertidamente, perturbar o ganho de unidade?

A: Muitos parâmetros SETUP e EFFECT alteram o nível de ganho. Alguns deles têm a intenção de alterar os níveis de ganho (afinal, como é que um aumento deve funcionar, afinal?) Aqui está uma pequena lista de coisas a serem consideradas:

O parâmetro LEVEL de cada efeito aumenta ou diminui o nível geral. Mudar MIX em certos efeitos altera os níveis seco e molhado. Isso evita que os sinais “empilhem” e causem cortes. Você pode compensar com seus ouvidos ativando e desativando os efeitos e comparando o nível com o True Bypass ativado. Se você quiser alterar o BYPASS MODE de um bloco da configuração padrão de THRU, é melhor verificar seus níveis ao ativar / desativar o efeito ANTES de alternar para algo como MUTE FX IN. Os parâmetros de nível na página OUTPUT do menu do modo principal aumentam ou diminuem os níveis gerais. Configurações incorretas na página E / S: AUDIO podem resultar em mudanças de ganho. O NOISE GATE tem um controle de nível. Se o seu equipamento for MONO, todos os controles BALANCE ou PAN podem afetar os níveis. Os equalizadores gráficos globais afetam o nível geral. As configurações da página NÍVEL DE I / O NÃO afetam o ganho de unidade. Cada configuração é compensada internamente.

Q: Alguma última palavra de conselho?

R: Use o interruptor TRUE BYPASS como uma forma de garantir que suas predefinições e cenas estejam no caminho certo. Em geral, é melhor estar no controle de seus níveis do que ser fixado no “conceito” de ganho de unidade. Faça o que soa melhor para você e aprenda o máximo que puder sobre seu equipamento.

Notas técnicas de Cliff sobre decibéis:

"O decibel é uma unidade de medida que dá a razão da potência de um sinal em relação a outro. A fórmula para o decibel é dB = 10 * log_10 (P1 / P2) onde P1 e P2 são medições de potência. A razão é chamado de decibel é porque é 10 bels. Um bel seria log_10 (P1 / P2). O importante é entender que o decibel é uma RATIO de poderes. Um dB não tem sentido sem um poder de referência. Então, se alguém diz " esse sinal é 86 dB "é um número sem sentido, pois não tem referência. Os decibéis são convenientes porque convertem a percepção logarítmica em uma escala linear. A audição humana, por exemplo, é logarítmica. Muitos outros fenômenos naturais são logarítmicos, o que significa que os fenômenos existe no "domínio de multiplicação" em oposição ao "domínio de adição". Por exemplo, a visão humana é logarítmica. Percebemos a luz de tal forma que a luz deve dobrar para que ela apareça duas vezes mais brilhante. uma curva exponencial de intensidade de luz vs. brilho percebido. Se tomarmos o logaritmo da intensidade, obteremos uma linha reta. É por isso que as câmeras usam f-stops, que são um logaritmo de base 2. Então, de volta aos níveis de referência. Existem muitos níveis de referência usados em dB: dBm, dBu, dBV, dB re. kPa, etc. dBm refere-se à potência referenciada a um miliwatt. Se a potência medida for, digamos, 100 mW, então isso seria 10 * log10 (100/1) = 10 * log10 (100) = 20 dBm. dBV é uma relação de tensão e não é realmente um verdadeiro dB, mas, independentemente disso, ainda é comumente usado. A fórmula para dBV é 20 * log10 (V1 / V2), pois precisamos ajustar a tensão para obter a potência. Em áudio, uma unidade comum é o dBu. dBu é a potência relativa à tensão em um resistor de 600 ohms que está se dissipando 1 mW. Isso é aproximadamente 0,77 volts. Nos primórdios da telecom, 600 ohms era a impedância de terminação padrão, daí o dBu. A maioria das artes de áudio profissional é executada em +4 dBu. O que isso significa? 0 dBu é 0,77 volts, então +4 dBu seria 4 dB maior, ou cerca de 1,22 volts. Para ir de dB para volts a fórmula é 10 ^ (dB / 20). A engrenagem de áudio do consumidor geralmente é executada a -10dBV, ou aproximadamente 0,32 volts. Ao gravar o seu objetivo é obter o seu nível de sinal perto do nível de sinal nominal do equipamento a ser utilizado. Isso garante a melhor relação S / N. Muitos consoles de gravação usam medidores VU que são calibrados de tal forma que "0 dB" é +4 dBu. O objetivo é obter seu nível de sinal em torno de 0 dB. Engrenagem bem projetada tem alguma quantidade de "altura livre". Headroom é a diferença entre o nível máximo do sinal e o nível do sinal nominal. Por exemplo, o Ax-Fx II tem um nível de sinal máximo de +18 dBu. Se operar a +4 dBu nominal, isso dá 14 dB de headroom, o que significa que qualquer pico de sinal pode ser mais de quatro vezes maior. Na engrenagem digital, encontramos o dBFS, que é dB em relação à escala total. Full-scale é um termo que indica o nível máximo de sinal dentro ou fora de um conversor A / D ou D / A, respectivamente. Com os conversores digitais, o melhor desempenho é obtido operando o conversor de forma que o nível de sinal nominal esteja próximo da escala total. A tensão exata é desconhecida e irrelevante. A maioria das engrenagens digitais terá indicadores que medem os níveis em relação ao valor de escala total do conversor. Por exemplo, os medidores de entrada no Ax-Fx indicam o sinal de entrada relativo ao valor de escala total do conversor A / D. O conselho "tickle the red" tem como objetivo operar o conversor A / D perto de seu valor de escala completa, pois os LEDs vermelhos acendem a 6 dB abaixo da escala total ou -6 dBFS. "Source" Decibels são decibéis. Não existe tal coisa como "decibéis de poder de raiz". Por definição, um decibel (dB) é uma relação de dois poderes. A fórmula é 10 * log10 (P1 / P2), onde P1 e P2 são a potência de dois sinais, respectivamente. Em eletrônica, no entanto, geralmente manipulamos e medimos os níveis de tensão. É conveniente representar a proporção de dois níveis de tensão em dB. Para fazer isso, você precisaria ajustar a tensão para obter a potência (desde P = V ^ 2 / R). Também assumimos R = 1 por conveniência. Com um pouco de matemática você obtém dB = 20 * log10 (V1 / V2). Portanto, se reduzirmos o nível de tensão de um sinal por um fator de 0,1, então o sinal é agora -20 dB em relação a antes. dB é simplesmente um mapeamento logarítmico para linear de fácil leitura. Música, percepção humana e muitas outras coisas na natureza tipicamente têm uma resposta logarítmica. O decaimento de, por exemplo, um prato é logarítmico. Se você plotar isso em um eixo linear, é difícil exibir por causa do intervalo dinâmico. Mas se você usar um eixo logarítmico, "comprimirá" os dados em algo mais fácil de visualizar. Decibéis são apenas um mapeamento amplamente aceito. Você poderia usar qualquer base para o log; log2, ln, etc, mas desde que temos 10 dedos log10 é bom. O ponto é que X dB é X dB. Se você reduzir um sinal em 20 dB você reduziu sua voltagem para 10% do que era anteriormente. Você também reduziu seu poder para 1% do que era anteriormente. Estas são as mesmas coisas: 20 * log10 (0.1) = 10 * log10 (0.01).

Conectando instrumentos e outros dispositivos.

Entrada do instrumento.

Ax-Fx III: INSTRUMENTO (dianteiro e traseiro, comutação automática). Ax-Fx II: INSTR (frente). AX8: IN 1 (INSTRUMENTO). FX8: IN 1 (PRE).

A entrada do instrumento usa um circuito proprietário e um conversor A / D dedicado para reduzir o ruído. É condicionado para guitarra através de hardware e software ("Secret Sauce"). Para obter melhores resultados, use a entrada de instrumento para guitarra, com ou sem fio, elétrica ou acústica, exceto quando estiver executando um sinal de nível de linha.

"O buffer de entrada foi projetado para.

Ax-Fx III: O Ax-Fx III possui duas entradas de instrumentos: frontal e traseira. As especificações são as mesmas. A parte traseira deve ser usada com racks, unidades sem fio e outros. Usando a entrada frontal, por exemplo, com um cabo, SEMPRE anula a entrada traseira. Isso não requer configuração na instalação.

Molho Secreto: O AX8, Ax-Fx II XL e XL + apresentam “Secret Sauce III”. O Ax-Fx III possui um circuito "Secret Sauce IV" nas entradas dos instrumentos dianteiro e traseiro. Isso reduz o nível de ruído usando uma técnica proprietária, juntamente com circuitos especiais de entrada analógica.

"O" Special Sauce III "usa uma combinação de coisas para obter um piso de ruído mais baixo. Uma dessas coisas é o novo op-amplificador Burr-Brown no caminho do sinal que tem um ruído e distorção extremamente baixos (e são muito caros) Como sempre eu não desenho coisas para serem baratas, eu o desenho para ser bom ". fonte "O espectro de uma guitarra é rosa (ish). Acima de 800 Hz, a energia rola dramaticamente. Por sorte, os humanos percebem que o ruído acima de 800 Hz é mais censurável, pois se manifesta como assobio. a entrada frontal pré-enfatiza as altas freqüências e, em seguida, faz o inverso no software. Isso tem o efeito líquido de uma resposta de freqüência plana, mas empurra o piso de ruído para baixo pela quantidade de pré-ênfase. É um truque antigo, usado em FM registros de rádio e vinil. A premissa básica é otimizar a conversão de dados para o conteúdo informativo da fonte. " (Ax-Fx II) "Você tem que definir a seleção de entrada para coincidir com a entrada que você está usando. Se você estiver usando a entrada frontal, então você deve definir a seleção de entrada para frente e vice-versa. Se você ligar alguma coisa a frente e defina a seleção de entrada para a traseira será MUITO mais brilhante. A entrada frontal é otimizada para entradas de nível de guitarra e tem modelagem espectral e mais ganho do que a entrada traseira. A entrada frontal é otimizada para captadores de guitarra. Esta é uma combinação de processamento de hardware e software. Se você definir a fonte de entrada para Analog Rear, isso desativa a parte de processamento do software. Se você estiver conectado à frente, ele mudará o tom, uma vez que você ainda está passando pelo processamento do hardware. É por isso que eu digo você deve combinar a seleção de entrada com a entrada que você está usando. As entradas traseiras são entradas de nível de linha padrão e podem ser usadas com qualquer material de programa. A entrada frontal, como dito acima, é otimizada para captadores de guitarra. ganho e menos espaço e pode cortar se você sed para material de programa não-violão. Se você conectar um violão diretamente na parte traseira, poderá descobrir que não tem nível de sinal suficiente. "

Ax-Fx II, III e FX8: impedância de entrada ajustável na entrada do instrumento (Ax-Fx III: frontal e traseira). AX8: impedância de entrada fixa (1 Mohm).

Alternando entre diferentes guitarras.

Ao alternar entre guitarras, haverá diferenças de nível e tom. Isso pode ser bom, porque por que você usaria guitarra diferente? Mas se você quiser adaptar predefinições para acomodar diferentes violões, várias coisas são possíveis:

Use presets diferentes para diferentes guitarras. Use uma entrada separada para cada guitarra e atribua a cada uma sua própria cadeia de sinal. Ajuste o nível no bloco de entrada. Este parâmetro controla o volume do sinal que entra na grade. Foi especificamente introduzido para este fim: compensando as diferenças de nível de saída entre as guitarras. Ele funciona apenas por pré-ajuste, portanto, ele precisa de ajuste por pré-configuração, a menos que a configuração seja armazenada como parte de um bloco global (apenas Ax-Fx II). Não é possível anexar um modificador. Configure X / Y ou canais no bloco AMP para diferentes guitarras, usando diferentes valores para o Input Trim, por exemplo. Anexe um modificador ao Input Trim no bloco AMP, conectado a um pedal ou switch. Configure blocos AMP diferentes (não no AX8) para cada guitarra. Use cenas e controladores de cena, anexados ao Input Trim no bloco AMP. Ajustar o ganho de amperagem no menu Global (não no Ax-Fx III). Somente Ax-Fx II: adicione um bloco de baixa CPU a cada pré-ajuste. Como FILTER ou VOL ou PEQ (PEQ e FILTER permitem EQ-ing adicional). Coloque-o no início da grade para que isso afete a quantidade de ganho no bloco Amp. Mantenha o bloco neutro e defina seu nível em, por exemplo, -6. Torne-o um bloco global, para que você possa alterar facilmente uma configuração e aplicá-la em todas as predefinições imediatamente. Anexe seu parâmetro Bypass a um controlador externo. Envolva esse bloco indo para I / O & gt; MIDI e alternando EXT CTRL xx INIT VAL entre 0% e 100%. Ou atribua um footswitch de função geral ao CC do controlador externo e use-o para alternar (defina a opção como Global: Sim no MFC). Funciona em todas as predefinições.

Consulte a seção 4 do Manual do Proprietário para obter um exemplo predefinido de uma guitarra de saída dupla, como magnetic + piezo.

Vários instrumentos simultaneamente.

Ax-Fx III: A guitarra nº 1 conecta-se à entrada do instrumento (frente da traseira), a guitarra nº 2 conecta-se à entrada 2. Até um terceiro e quarto instrumento podem ser conectados. Cada um pode ter sua própria cadeia de sinal na grade e sua própria saída, se desejado. Existem dois blocos de Amp, então dois instrumentos podem usar a modelagem de amplificadores. Mas você não precisa de um bloco de amplificador para um violão, ou um piezo, ou um elétrico que soe como um violão. Talvez nem para baixo, com a ajuda do modelo de drive B7K. Você pode até mesmo se safar com um bloco Drive e um bloco Cab para tons de guitarra limpos (ao custo da dinâmica). Veja também a seção 4 do manual do proprietário.

Definir entrada 1 para estéreo. Conecte um instrumento à frente e o outro à parte traseira da Entrada 1 à direita. Use duas linhas na grade. Adicione um bloco VOL a cada linha. Defina um para Entrada esquerda (para o instrumento que entra na entrada frontal ou traseira para entrada 1 esquerda) e outro para entrada direita (para o instrumento que entra na entrada traseira 1 direita). Adicione um bloco AMP após cada bloco VOL, se necessário. Você também pode deixar de fora os blocos VOL e ajustar Amp 1 para Input Left, e Amp 2 para Input Right; isso só funciona com os blocos do Amp na primeira coluna. Continue as linhas até o final, adicionando um CAB e efeitos a cada um, se necessário, ou mescle-os, se desejar. Mantenha os sinais separados usando os controles Balance.

AX8: Para usar o AX8 com 3 dispositivos: guitarra e dois outros dispositivos como piezo ou synth:

Adicione um Amp e Cab. Coloque o bloco FX Loop após o bloco Cab. Isso envia o som de guitarra regular para a Saída 2 e permite que os sinais da Entrada 2 (esquerda e direita) entrem na grade. Dividir o sinal após o bloco de loop de FX em duas linhas e adicionar um bloco de volume para cada um. Definir Entrada Selecione em um bloco de Volume para Apenas Direita. Defina Pan e Balance conforme desejado. Set Input Select no outro bloco de volume para somente à esquerda. Defina Pan e Balance conforme desejado.

Instrumentos acústicos.

O instrumento ou a entrada 2 podem ser usados para conectar um instrumento acústico. Na entrada Ax-Fx III 2 também pode ser usado.

O IR de um corpo acústico pode adicionar ressonância acústica ao tom. Sempre use IRs Ultra-Res ao usar IRs acústicos quando disponíveis. Não há IRs acústicos entre os gabinetes de estoque. Você pode encontrar alguns aqui:

Uma predefinição para um violão com bons captadores pode ser mantida simples: compressor, EQ e reverb. Um bloco de Amp não é necessário, mas o modelo TUBE PRE pode ajudar a aquecer o tom.

Você também pode usar o Tone Matching com ótimos resultados. Tutoriais no G66.eu.

Baixo.

O Ax-Fx III, II e AX8 fornecem cabines de baixo de estoque e alguns modelos de amplificador de baixo.

O sintonizador suporta ajuste de guitarra baixo. No Ax-Fx III, a detecção de pitch em guitarras baixo foi melhorada.

Entre guitarra e processador: Se você quiser conectar um pedal de efeitos ao Ax-Fx II, III ou AX8, com o processador configurado para modelagem de amp e cabina, conecte-o entre a guitarra e a entrada do instrumento no processador. Lembre-se de verificar a impedância de entrada no Ax-Fx II, III e FX8, e também certificar-se de que a saída do pedal não corta a entrada do Ax-Fx III.

No loop de efeitos: Como alternativa, insira o pedal em um loop de efeitos. Certifique-se de ajustar os níveis quando necessário. Você pode incluir / excluir o pedal por cena ou usar o loop de efeitos como um comutador de áudio.

(sobre a execução de pedais de nível de instrumento nos loops do Ax-Fx III)

"As saídas têm pads ajustáveis." source (Ax-Fx III) "A entrada pode suportar até +/- 5V, o que é maior do que qualquer pedal de 9V pode fornecer." fonte.

Entre AX8 ou FX8 e amplificador: Um pedal também pode ser colocado entre o AX8 e o FX8 e um amplificador, ao usar o processador como uma pedaleira virtual. Certifique-se de ajustar os níveis onde for necessário.

Um receptor sem fio pode se conectar à entrada do instrumento ou a outra entrada.

(Ax-Fx II) "A entrada frontal tem um SNR melhor, mas se você estiver usando uma conexão sem fio, a melhor SNR da entrada frontal não será perceptível, já que o ruído da rede sem fio dominará." fonte.

Reproduza som do computador ou outro dispositivo.

Através do Áudio USB: O áudio do computador pode ser reproduzido através do Ax-Fx II e III através do Áudio USB. O sinal não entrará na grade e será transmitido para a Saída 1. Para ajustar o volume, ajuste o volume no aplicativo de áudio. O áudio não pode ser roteado para outra saída do Ax-Fx II. Para direcionar o sinal para outra saída do Ax-Fx III, consulte a seção 3 do Manual do Proprietário (Mac) ou use o ASIO (Windows).

O AX8 e o FX8 não suportam áudio USB (entrada nem saída).

Através de Entradas: O áudio externo pode entrar no processador conectando um computador ou tocador de música às entradas.

Ax-Fx III: saída 3 L / R ou saída 4 L / R seria a primeira escolha. Entrada 1 (traseira) no AX8 e Ax-Fx II: certifique-se de ajustá-lo para estéreo, use um preset com apenas shunts ou configure o modo Ax-Fx II para Bypass. Entrada 2 no AX8 e Ax-Fx II: o bloco FX Loop deve ser usado e conectado à saída da rede.

Microfones

Um microfone só deve ser conectado à entrada 2 (ou 3 ou 4). O Ax-Fx II, III e AX8 não tem pré-amplificadores de microfone, então haverá uma incompatibilidade de nível. O sinal do microfone está muito baixo e a relação sinal-ruído será alta. Você precisará aumentar o nível da fonte para obter força de sinal suficiente na unidade. Isso pode ser feito usando o pré-amplificador de microfone externo de amplificador ou um dispositivo como o adaptador A85F da Shure.

(Duas opções impressionantes entre um microfone e o Ax-Fx III são o Summit Audio TBA-221 e o FMR-RNP. Uma boa opção barata é o Rolls MP13. Com um ou dois destes, você pode vender a interface. " .

Sustentar e feedback.

É tão fácil obter feedback de sua guitarra como com um amplificador e gabinete regulares (com exceção dos fones de ouvido). Se você não conseguir, experimente o parâmetro Output Phase no menu I / O.

Não há efeito "sustainer" dedicado ou feedback (como o FreqOut da Digitech) no Ax-Fx e no AX8. O membro do fórum Simeon criou uma predefinição de simulação de feedback, imitando o feedback controlado.

Pedais e interruptores.

Definir o nível de entrada principal.

Ax-Fx III: E / S & gt; Entrada. Ax-Fx II: E / S & gt; Entrada & gt; Entrada de Instr / Entrada 1 / Entrada 2. AX8: E / S & gt; Níveis & gt; IN 1 (Instrumento) Pad / In 2 (Fx Rtn) Nível Nominal. FX8: E / S & gt; Níveis & gt; Entrada 1 (Pre) Pad.

O que é Input Level ou Input Pad para?

O nível de entrada e o painel de entrada não são controles GAIN. Eles NÃO afetam o nível geral de volume, ou o ganho de recorte ou de amplificador (a menos que você fique abaixo de 5%). Ele apenas otimiza a relação sinal-ruído dos conversores analógico-digital. O ajuste é aplicado antes do conversor A / D e é compensado por um impulso correspondente mas oposto à saída do conversor.

Como definir o nível de entrada ou o painel de entrada.

Defina Input Level ou Input Pad para fazer com que o LED vermelho de entrada pisque apenas ocasionalmente enquanto toca ("tickle the red"). Strum duro, no captador mais alto! Há um intervalo substancial entre laranja e vermelho. Se você não consegue piscar vermelho, não se preocupe.

A luz vermelha acende ANTES de os grampos de entrada do instrumento. Significa "Atenção, você está se aproximando de um recorte", ao contrário de "Atenção, você está recortando". No Ax-Fx II, AX8 e FX8, a luz vermelha acende a -6dB do ponto em que o sinal é limitado / limitado. No Ax-Fx III os LEDs vermelhos no painel frontal acendem a -1 dBFS!

Enquanto a palavra "clipping" é usada aqui, na realidade o sinal de entrada nunca é realmente clipe, porque um limitador entra em ação.

AX8 e FX8: maior valor de Pad de Entrada significa uma entrada mais baixa (preenchimento). Execute-o o mais baixo possível, porque o preenchimento aumenta o nível de ruído. Ax-Fx III: a entrada do instrumento no Ax-Fx III é um pouco mais sensível do que no Ax-Fx II, mas tem mais espaço livre / dinâmico. Não o defina abaixo de 5%, porque neste momento o ganho pode ser afetado. Também esteja ciente de que os LEDs vermelhos no painel frontal acendem a -1 dBFS. A ponte do medidor LED do painel frontal fornece status visual instantâneo para as entradas.

Nota: ao usar um instrumento mono, não use uma entrada definida como Estéreo ou Soma L + R, use Apenas Esquerda (padrão)

Os níveis de entrada podem ser controlados via MIDI CCs. No Ax-Fx III, defina os CCs em Setup & gt; MIDI / Remote & gt; De outros.

(Ax-Fx II) "Para um Strat, perto de 100% no nível de entrada não é incomum. Eu corro meu Strat por lá. Ele tem pickups do tipo vintage." source (Ax-Fx II) "Para obter o melhor desempenho de ruído, é importante que o trim instr In esteja configurado corretamente no menu Entrada / Saída. Configure isso o mais alto possível sem cortar a entrada." source (Ax-Fx II) "Você não tem que fazer cócegas nos vermelhos. Ajuste para o seu violão mais quente e deixe-o." source (Ax-Fx II) "O AFXII possui potenciômetros controlados digitalmente antes e depois dos conversores A / D e D / A. Portanto, sabe quais são os ganhos de entrada e saída. Compensa esses ganhos no caminho digital." source (Ax-Fx II) "Full-scale é um termo que indica o nível máximo de sinal dentro ou fora de um conversor A / D ou D / A, respectivamente. Com conversores digitais, o melhor desempenho é alcançado operando o conversor o nível de sinal nominal está próximo da escala completa. A tensão exata é desconhecida e irrelevante. A maioria das marchas digitais terá indicadores que medem os níveis em relação ao valor de escala total do conversor. Por exemplo, os medidores de entrada no Ax-Fx indicam o sinal de entrada relativo ao valor de escala completa do conversor A / D. O conselho "tickle the red" visa operar o conversor A / D perto de seu valor de escala completa quando os LEDs vermelhos acenderem a 6 dB abaixo da escala total, ou -6 dBFS. " source "O controle Input Trim no menu I / O está antes do A / D. Você pode usar isso para reduzir o nível para o A / D. Se você quiser 4 dB de redução de ganho: A = 10 ^ (- 4 / 20) = 0,63 Então você precisa reduzir seu pad de entrada em 37% O novo valor é 0.243 * 0.63 = 0.153 = & gt; 15.3% "source (Ax-Fx III)" Não existe uma configuração otimizada única. captadores quentes com cordas grossas e uma mão pesada, você pode precisar configurá-lo em 10% ou menos. Se você tiver single-coils vintage com cordas finas e um toque leve, 100%. Ajuste-o para agradar o vermelho ao jogar duro. " fonte.

Definindo o nível de saída principal.

Botões de nível de saída no painel: Os níveis de saída principais no Ax-Fx II, III e AX8 são controlados diretamente com os botões de nível de saída no painel.

(sobre os botões de níveis de saída no AX8) "A saída não chega a zero. Isso foi feito devido à grande quantidade de problemas de suporte em que as pessoas diriam que não estavam obtendo nenhum som e que foi simplesmente devido a o fato de que eles tinham o botão virado para baixo. Então agora você recebe um pequeno sinal e recebemos menos chamadas de suporte. " (fonte) (sobre o Ax-Fx II) "Testamos a saída como plana dentro de +/- 1 dB ao longo do alcance do botão. Na verdade, eu ficaria surpreso se houvesse alguma variação mensurável." source (Ax-Fx II) "O pote de saída" é na verdade uma escada de resistores discretos que é controlada remotamente pelo botão no painel frontal. Outros produtos simplesmente reduzem o sinal digital que entra no conversor D / A, mas isso é sub - O máximo possível para reduzir sua faixa dinâmica ao fazer isso O Ax-Fx II se esforça para manter o sinal no D / A o mais alto possível para uma faixa dinâmica ideal e então controla o nível de saída usando um ganho de saída programável. essa abordagem é que você ouvirá um pequeno ruído quando a saída alternar entre os resistores na escada. " source (Ax-Fx II) "Colocar um pote após D / A requer a passagem de cabos de / para o painel frontal. Esses cabos podem degradar a qualidade do sinal e aumentar o ruído. Os potenciômetros no painel frontal do II são controles remotos para the digital pots. The signal never passes through them. The digital pots also allow us to boost the level from the D/A and then attenuate it precisely to improve output SNR. The Output X Boost/Pad feature would be impossible without digital pots. " source.

Prevent clipping the mixing console or amplifier inputs : The nominal level of the main outputs of the Axe-Fx II and AX8 is line level: +4dBu default on the AX8 and Axe-Fx II, and -10dBV default on the Axe-Fx III. This is adjustable on the Axe-Fx III and AX8 in I/O > Audio.

Reminder: the III defaults to -10 dB nominal output level, so its output level is lower than the II and the AX8.

"The default Output Level for Output 1 and 2 is -10 dBv. This was done to reduce the number of support cases due to people overloading the inputs on consumer-grade interfaces, mixers, etc. (IOW cheap stuff). Most professional gear runs at +4 dBu so if using a pro-grade interface, mixer, etc. you may want to go into the Global menu and change the level to +4 dBu." source.

ALWAYS connect the device to a line level input on the board, when available. The output signal is too hot for a MIC input on the board.

(Axe-Fx II and AX8) "The XLR output is balanced but it's +4dBu nominal. The problem is people connect it to a mic input which is way too sensitive for that level signal. If the board has a mic/line switch you want to set it to line level. Or if it has a pad switch turn that on. Otherwise turn the level knob way down. The thing to remember is that XLR is just a connector. It doesn't imply microphone levels. Most pro stuff like eq's, etc. have line-level XLR's."

If only MIC inputs are available on the mixing console, try this:

use a pad switch on the mixer to attenuate the signal and prevent clipping. decrease input gain on the channel strip. decrease the output level from the device by turning down the Output knob at the front.

(Axe-Fx II) "Optimal gain staging would be with the level knob around noon. Higher than this and you risk clipping the inputs of the downstream device. With the level knob at full the Axe-Fx II will probably incinerate a Soundblaster or other low-cost stuff. The max level out of the Axe-Fx II is +20dBu. Most pro gear can easily handle that but lots of gear cannot and the trend in newer gear is towards lower and lower maximum input levels (due to single-ended designs and low-voltage/low-power constraints). In the old days, +20dBu was routine. Everything could put out and handle +20. Not so much anymore." source (Axe-Fx II) "The II actually has more output than the I. The II can do about +20 dBu, the I was about +18." source (Axe-Fx II) "Start with amp volume at noon. Bring up Axe-Fx volume until desired level is reached. If you need more, turn up amp. With the Axe-Fx volume all the way up you would be pushing +20 dBu into the amp which could clip the inputs to the amp." source.

No DI box needed : There's NO NEED to use a DI box to connect the Axe-Fx II, III or AX8 to a mixing board directly (with amp and cab modeling).

Controlling output levels via MIDI : MIDI CCs can control output levels. To reset them without the help of a MIDI controller, change the assignment to "none" in I/O.

Preset level and Global EQ Gain : The main output level is also affected by the output level of the selected preset, and by the Global EQ's Gain control.

Output meters on the Axe-Fx III :

A front panel LED meter bridge provides instant visual status for the inputs and outputs. The red LEDs on the front panel come on at -1 dBFS. This is different than on previous hardware. There's a Meters page in the Home menu showing all I/O levels. THe Utility lets you check the performance of the four outputs.

Input 2 and Output 2.

Axe-Fx II and AX8 : The Axe-Fx II and AX8 have a single stereo effects loop: Input 2 (Effects Return) / Output 2 (Effects Send). Read this: FX Loop block.

Input 2 : Input 2 on the Axe-Fx III provides a set of Combi connectors (XLR and 1/4" L/R). Use the Input 2 block on the grid to handle the incoming signal. These connectors support high-impedance sources such as guitars and basses, besides other gear. Because of this, there will be some white noise when Input 2 is connected to an output, and nothing is plugged into Input 2 (this does not apply to ports 3 and 4). Input 2 level is adjusted through I/O > Input > Input Trim, see comments above. Mono/stereo processing is set in I/O > Audio > Input 2 Mode. Output 2 : XLR L/R connectors. Use the Output 2 block on the grid to handle the signal. Input 2 and Output 2 are LINE level ports.

Output level can be controlled via a MIDI CC. On the Axe-Fx III set the CC in Setup > MIDI/Remote > De outros.

Input/Output 3 and Input/Output 4 (Axe-Fx III)

These stereo pairs on the Axe-Fx III are designed for inserting outboard gear, for the Four-Cable-Method (4CM) and for other purposes. Use the corresponding Input and Output blocks on the grid to handle the signal.

These connectors support high-impedance sources such as guitars and basses, besides other gear.

Mono/stereo processing is set in I/O > Audio > Input 3/4 Mode.

Input 2 and Output 2 are LINE level ports. Input levels are adjusted through I/O > Input > Input Trim.

To set these loops to unity gain (their primairy purpose), set their OUT levels to maximum.

"Outputs 3 and 4 are primarily intended for unity gain applications, i. e. fx loops. You can use them as general-purpose outputs as well. When doing this you may need to increase the Output Level of the associated Output block." source.

Levels can be controlled via MIDI CCs. On the Axe-Fx III set the CCs in Setup > MIDI/Remote > De outros.

I/O parameters.

Main Input Source / Input 1 Source.

Applies to: Axe-Fx II, Axe-Fx III.

This lets you set select the source of input 1: ANALOG (default), SPDIF/AES or USB (Axe-Fx III: USB channels 5 and 6).

Note that Input 1 is mono.

Word Clock.

Applies to: Axe-Fx II, Axe-Fx III.

See Digital audio (below).

SPDIF/AES Select.

Applies to: Axe-Fx II, Axe-Fx III.

Only one can be active at any time.

Input 1 Left Select.

Applies to: Axe-Fx II.

Input 1 is split: it appears at the front (mono) as well as on the rear (mono or stereo). This parameter tells the Axe-Fx II if the front input or the rear input is used to connect the instrument. Default: front.

The rear port has the same impedance as the front input (1 Mohm, for guitars) but it operates at LINE level instead of instrument level.

The front input does not disable Input 1 left at the rear. Use the menu to select either the front input or rear input. You can leave everything plugged in.

"Input 1 on the rear is a very high impedance input (1 Mohm). It is compatible with low impedance outputs. Most people don't understand the real meaning of impedance and think you need to connect low impedance to low impedance but that's only with passive devices and is a relic of the old days when transformers were used to get the best power transfer. Nowadays we have active inputs with very high impedance which are compatible with a broad range of source impedances." source.

Input Mode.

Applies to: Axe-Fx II, Axe-Fx III, AX8, FX8.

Axe-Fx II : when connecting a guitar to the front instrument input, set this parameter to Left Only (default). Using Sum L+R can introduce noise (from the disconnected right Input 1) and attenuates the signal level (6 dB). You'd only use Sum L+R or Stereo when connecting a stereo instrument to Input 1 (rear).

Input 2 Mode controls the same for Input 2 on the Axe-Fx II, AX8 and FX8. Input 3 Mode and Input 4 Mode control the same for inputs 3 and 4 on the Axe-Fx III.

Output 1 Level, IN and OUT Nominal Level, Post Level.

Applies to: Axe-Fx III, AX8, FX8.

IN 2 Nominal Level: use -10dB (default) when this input connects to the output of a guitar or stomp box. Use +4dB when it connects to the output of a pro level device.

OUT 2 (Main) Nominal Level: use +4dB (default) when the output connects to a mixing console or line level input on another pro audio device. Use -10dB when the output connects to a traditional amplifier or pedal.

Reminder: the III defaults to -10 dB nominal output level, so its output level is lower than the II and the AX8.

Post Level.

This switches between +4dB (pro line level, default) and -10dB (consumer-grade equipment). Use +4dB when the unit connects to a line level on another device. Use -10dB when the unit connects to a guitar amplifier's instrument input or a pedal.

If you're using the FX8 in a Four Cable Method (4CM) setup, and you're experiencing hiss, try switching to another Post Level value.

NOTE: some users report that the FX8 needs a reboot after changing the value before the changed value takes effect. source.

Output Mode.

Applies to: Axe-Fx II, Axe-Fx III, AX8, FX8.

This determines if the output signal is stereo, or one of two mono modes: Sum L+R or Copy L>R.

Output Phase.

Applies to: Axe-Fx II, Axe-Fx III, AX8.

This lets you invert the phase of the signal if necessary.

Output Boost/Pad.

Applies to: Axe-Fx II, Axe-Fx III, AX8.

The Boost/Pad parameter optimizes the output level and reduces noise in certain scenarios, such as a 4CM setup (Four Cable Method) or when placing the device in front of a traditional amp.

When putting the Axe-Fx II or III (output 3 or 4) in front of a real amp, maximize Boost/Pad to make sure the full range of the D/A converter is used.

On the Axe-Fx III, Boost/Pad is not available on Outputs 1 and 2.

With 4CM: increase Boost/Pad, and prevent clipping. Turn OUT on the front panel fully open. Then use Input Level in I/O to finetune the signal level.

Javajunkie: "Boost/pad is not intended for boosting USB output (although it does have that effect). It is for the four cable method (4CM)".

"Usually you will use this when you send the effect loop out before the amp block. The signal will be weak going to the D/A converters. This allows you to boost the signal to the D/A converters w/ maintaining the same output level." source.

Output 2 | Copy Output 1, Output 2 Echo.

Applies to: Axe-Fx II, Axe-Fx III, AX8.

This determines what signal is sent through Output 2.

ON/OFF: set to ON a copy of the Output 1 signal is sent to Output 2, UNLESS the preset contains an Output 2 block.

Nenhum. Output 1: an exact copy of the Output 1 signal, UNLESS the preset contains an FX Loop block. Input 1: an exact copy of the direct input signal (unprocessed). This sends a DI signal through Output 2, UNLESS the preset contains an FX Loop block.

Output 3/4 | Copy Input 1.

Applies to: Axe-Fx III.

This determines whether a copy of Output 1 is sent through Output 3 or 4. For example for re-amping.

This setting only works if the preset doesn not contain an Output 3 block or Output 4 block.

Output 1 (PRE) Headroom.

This parameter can supply more headroom if the signal through Output 1 PRE is too hot. Downside is that the noise floor is increased. Default is 6 dB, for the least headroom and lowest noise floor. The increased headroom is offset by a corresponding but opposite adjustment internally, so “what you hear” remains the same at all settings. Maximum headroom is 18 dB.

USB/Digi Out Source.

Applies to: Axe-Fx II.

This lets you determine what is sent out through USB Audio/AES/SPDIF: Output 1, Output 2, or Input (which is the unprocessed DI signal).

USB Buffer Size.

Applies to: Axe-Fx II, Axe-Fx III.

See USB Audio below.

USB Return Level.

Applies to: Axe-Fx II.

A level control for the incoming USB Audio signal.

Setups: Axe-Fx and AX8.

See the rig diagram in Section 4 of the Owner's Manual.

FRFR - Direct - Studio Monitors - DAW.

What is FRFR: full range flat response.

Amplifying a signal with cabinet modeling requires a power amp and speaker system that covers the entire frequency spectrum: 20 Hz - 20 kHz. Also, it should not add any tonal coloring of its own. Those systems are called FRFR: full range flat response. Also referred to as: neutral. What goes in comes out. Tone shaping is entirely left to the input device, which is a Fractal Audio processor here.

From the Axe-Fx III Owner's Manual:

"A Full-Range Flat Response (“FRFR”) system aims to reproduce the entire audio spectrum without compromise. In comparison, most guitar speakers are narrow range, with no ability to accurately reproduce extended lows and highs. A 1×12 open-back combo is never going to sound like a 4×12 stack. In comparison, full-range flat response studio monitors, high‐quality PA speakers, and FRFR speakers designed specifically for guitar should be able to reproduce anything you play through them."

Getting accustomed to the FRFR sound may take some time. It is different from listening to a traditional guitar amp. You may also miss the "thump" of a traditional guitar cabinet.

FRFR amplification means: portability, no tone coloring, reduced stage volume, consistent tone at all volume levels and in all venues, ability to work with cab modeling, ability to produce synth and acoustic tones, and the musician hears exactly what the audience hears.

Which systems are FRFR.

FRFR systems include:

Studio monitors. Active (powered) FRFR cabs and wedges. Passive FRFR cabs and wedges, powered by a separate amplifier. Headphones. DAW. High-quality PA systems.

Popular manufacturers of FRFR solutions for stage use include Atomic, RCF, Matrix, Meyer, Friedman, XiTone, Mission Engineering, EAW, QSC and others. Quality studio monitor brands include Focal, Adam, Genelec.

Close miking.

Using amp and cab modeling with FRFR amplification means that the audio signal includes a virtual cabinet. These cab models use impulse responses (IRs). These are sampled sounds of cabinets and speakers, with the recording microphone(s) placed within inches of the speaker cap or cone ("near-field"). This is referred to as a "close-miked" sound. When compared to the sound of a traditional amp and cabinet (aka "amp in the room" or "far-field" recording), close-miked sounds haven much more bass and treble, due to the microphone's proximity to the speaker.

That sound is often EQ'd at the mixing board to fit into the entire mix. Fractal Audio's Axe-Fx and AX8 modelers provide lots of EQ tools to use.

"You're not going to hear the same thing through FRFR that you heard from guitar cabs. Your audience will hear something very similar but you won't. What you're hearing through FRFR is a mic'd representation of the cabs. It takes some getting used to. You have to start thinking like a producer/engineer rather than a guitar player. If you start trying to dial out what you call "fizz" and "artifacts" you're going to end up with a tone that doesn't cut. It might sound good to you but it won't fit in the mix. That fizz and sizzle is what makes those classic rock tones work. Listen to some isolated tracks of VH and AC/DC and you'll hear a ton of high-end sizzle. In the mix, however, it's not noticeable. If you remove it then the guitar sounds dead." source "The sound of an amp in the "far field" is quite different than what you get with close-miking. IR's are made using close-miking and therefore sound nothing like listening to a guitar cab at distance from the cone. Your audience does not hear the far field tone, they hear the close-miked tone as that's what is put through the FOH. It can be quite an adjustment coming from far field amp tone to close-miked tone. Some people just never adjust. Fortunately the Axe-Fx was designed to give you the best of both worlds. You can use the FX Loop and Output 2 to a power amp and conventional guitar cab while routing the fully processed tone with IR to the FOH. See the manual for full details. Rather than using your amp you can use a lightweight solid-state power amp and any of the new, lightweight guitar cabs that use Neodymium speakers. This gives you the classic far field amp tone for yourself in a lightweight package and the polished sound for the FOH direct from Output 1." source "Close-miked IRs typically have a lot more high frequencies than what you hear at a distance and off-axis from the speaker." source "All speakers "move air", that's the entire point of their design. Guitar speakers are inherently directional at higher frequencies. So when you stand off to the side you hear less highs. If you have two or four speakers the directivity gets even worse. FRFR speakers have less directivity. This combined with IR technology that almost invariably uses samples of a close-miked speaker and you end up with a different listening experience. To confuse the issue further many combo amps have an open back which changes the frequency response at the listening position even more. Now, if you connect your Axe-Fx to a power amp and traditional 1x12, 2x12, etc. then you will get "amp in the room" but the "moving air" statement has no basis in fact." source "You can't compare what you are used to hearing "in the room". The close-miked sound ALWAYS has more highs and lows. This is due to the physics of near-field micing. And this is why a highpass and lowpass are frequently employed at mixdown." source "The classic method is "1W / 1m" which is to apply 1W and measure 1 meter away. When you get the microphone close to the speaker the response is much different and you usually get more highs and lows. This is "close miked" and is the technique normally used in studio recordings. During mixdown the producer/engineer will then often highpass and lowpass the signal to remove these excess highs and lows and to make the guitar "sit in the mix". IRs are almost always made using the same close-miked technique and, hence, will sound like a raw recording. Far-field IRs are possible but very difficult to obtain requiring a large facility and special techniques. Our primary goal is to model an amplifier and speaker as accurately as possible and the latest modeling is astonishingly accurate. We do not purport to be producers or mix engineers and leave the choice of low cut and high cut frequencies up to the user. Furthermore many users rely on the soundman to apply the filtering at the board, just as they would when mic'ing a "real" amp. More importantly the choice of frequencies is highly dependent upon the IR used." source "IRs are equivalent to close-mic'ing an amp. When you close mic an amp you almost always get more bass and treble than an "amp in the room". The extra bass is due to the proximity effect of the microphone. The extra treble is primarily due to the directivity of the speaker. During mixdown engineers/producers will typically incorporate a low cut and high cut to help the sound "sit in the mix". The thing to take away from all this is that an IR represents the close mic'd sound (unless using far-field IRs which are rare) and the close mic'd sound of an amp is much different than the "amp in the room" sound. As such it is common to use frequency shaping on a close-mic'd amp." source "The Axe-Fx is extremely accurate in duplicating the sound of a mic'd amp. Your monitoring thus becomes an essential part of the chain and accuracy is paramount. Many "FRFR" monitors are neither FR nor FR." source "FRFR is just not the same. Traditional head/cab you hear the sound from a bandwidth-restricted speaker at, say, 10 ft. In a typical modeler setup you are hearing what the "mic heard" when the IR was made and that mic was pushed up against the grill cloth. One approach is to use "far field" IRs which are obtained using a measurement mic at a typical listening distance and angle. These are rare. There are a couple stock far-field IRs. They are indicated by (JM) for Jay Mitchell, who created them. Even then it's still not the same because when you are using a traditional setup you move around while playing and the tone changes based on the angle. With a far-field IR the tone doesn't change with angle. When I was gigging I used a power amp and cab behind me and sent the XLR outputs to FOH. More gear to lug but best of both worlds: traditional backline sound, consistent FOH sound." source "You're never going to get a full-range monitor to sound like an amp in the room regardless of the IR used. One reason for this is dispersion. A traditional guitar cabinet has a beam pattern that decreases with increasing frequency. This means less high frequencies when listening off-axis. A full-range monitor will have more highs. Now some will argue that if you capture the traditional cab off-axis in the far field then you'll get the same thing but you won't because the monitor is not interacting with the environment in the same way. The traditional cab will send less frequency content to off-axis which is then reflected off the floor, walls and ceiling. The monitor will send more highs off-axis that are reflected. Our hearing relies a LOT on the spatial cues of reflection and the reflections will not be the same. Compound the above with the fact that 99.9% of IRs are near field captures which sound nothing like the far field. I believe trying to get a monitor to do amp in the room is a lesson in futility. If you really want that sound use a traditional guitar cab." source.

Fletcher-Munson.

Another audio phenomenom to be aware of, especially when using FRFR amplification, is: Fletcher–Munson. This refers to scientific findings that human ears perceive sound at low volume levels differently than at higher levels. This is VERY important when dialing in tones.

At low volume levels people often tend to dial in more treble and bass than at loud levels. The Loudness switch on older home stereo systems does just that. But: when the volume is turned up, e. g. at a gig, those low and high frequencies become much too prominent. The guitar will then compete with cymbals (and lose), and with the bass (and lose). The result: the guitar drowns in the mix. Even turning up the volume does not fix this.

More information about Fletcher-Munson:

How to handle harsh, boomy tones and getting lost in the mix.

To avoid harsh and boomy tones when using FRFR amplification, and to be heard in the mix, you need to do this:

Always dial in your "live" guitar tone at gig levels, which is 90 dB and higher. Do NOT expect excellent "bedroom" or headphones tones to translate well to a rehearsal room or stage. Use EQ! The guitar is mainly a "mid frequency" instrument, so shave off superfluous top and bottom end and add mids. This will help focus your tone.

EQ can be applied in several ways:

Use the Low Cut and High Cut parameters in the Cab block to block undesirable top and bottom end. Common values are low-cutting (high-passing) between 80-150 Hz and high-cutting (low-passing) between 6-10 kHz. Now this may seem to make your guitar sound worse by itself, but it will improve its sound within the entire mix! Put a PEQ at the end of the grid and block the low and high frequencies. Use the Global EQ or a GEQ for similar results. Adjust Depth/Bass and Treble/Presence in the Amp block. Boost the mids. For example: put a PEQ at the end of the grid, set a band (use Peaking when using the first or last band) to 770 hz, Q at 0.35, Gain between 2 and 4 dB. source Use the Cut switch in the Amp block.

"Resist the temptation to add bass and treble. The amp designers knew what they were doing (well most of them). If you are applying heavy EQ then you will be disappointed at gig volumes. What sounds midrangey and bland at low volumes will sound great at high volumes. Do some research on Fletcher-Munson to understand this." source "People often talk about applying low cuts and high cuts. This is because the cabinet models used in modelers are almost always (with a couple exceptions) based on near-field samples of guitar cabinets. IOW, the mic is pushed up against the grill cloth. This just happens to be the way that record producers/engineers mic a cabinet in the studio and the way guitar cabs are mic'd on stage. This is done primarily for isolation reasons. The downside of this approach is that the resulting tone will have a lot more lows and highs than when listening to the amp+cab "in the room". What the mic "hears" when pushed up against the grill cloth is not the same thing that we hear standing 10 feet away. The most common technique to deal with this is to simply cut out the lows and highs using blocking filters, e. g. highpass and lowpass filters. Producers routinely do this when mixing as excessive amounts of lows and highs will cause the guitar tracks to get "lost in the mix". Live sound engineers often do the same thing. The Cabinet block has blocking filters built in for just this very reason. You can also use a couple dedicated filter blocks or a parametric EQ block. For now let's use the Cabinet block. My personal settings are Low Cut around 80 Hz and High Cut around 7500 Hz and Filter Slope set to 12 dB/octave but these are just a starting point. Far-field IRs are available but they are rare due to the difficulty in obtaining them. They require a large facility and special techniques making the process impractical in most cases. So, until an abundant source of far-field IRs are available we need to think like a producer/engineer who is dealing with the mic pushed up against the grill cloth. This means shaping the tone with EQ to remove unwanted frequencies." source.

Do not use a microphone on a FRFR speaker on stage.

When you're using FRFR amplification on stage and you need to provide a signal for FOH, do NOT place a microphone in front of the FRFR monitor. That would make no sense because the signal already contains miked cab simulation. Direct-to-FOH is the right way to do it: run a cable from the output(s) to the mixing console. For long distances use the balanced outputs.

Create an amp-in-the-room tone with FRFR amplification.

To create the amp-in-the-room sound using FRFR amplification:

Select a "farfield" stock cab type. There are the ones with "JM" in the name. Or select a stock cab captured with a neutral mic, such as the Red Wires ones, and set Proximity to its lowest value for far-field coloring. If you have stereo amplification: use two IRS in stereo and use a very short delay in the Cab block on one of them to create the HAAS effect. Use Room Reverb in the Cab block (not on AX8). Increase De-Phase in the Cab block. It adds amp-in-the-room characteristics to the FR sound (not on AX8). Add a mid-boost, as described above.. Turn up Speaker Compression (Amp block) or Motor Drive (Cab block).

Of course, if you want a real "amp in the room" tone from your modeler, use a power amp and a traditional guitar speaker cabinet.

Tweeter squeal.

Some FRFR speakers can cause very high-pitched loud feedback.

"Tweeter squeal is magnetic feedback from the speaker's tweeter. Move further away from the speakers. This is a phenomenon unique to FRFR solutions." source "Magnetic feedback is an issue unique to FRFR amplification. The tweeter creates a magnetic feedback loop with the pickups. The closer you get to the speaker the more feedback until the point it squeals. The only solution is to move away from the speaker or turn down the gain/volume." source "The high-pitched feedback is pickup squeal and is caused by electromagnetic feedback from the speaker to your pickups. FRFR tends to exacerbate this since you have a tweeter feeding back high frequencies. A noise gate can help but the best solution is to move away from the speaker." source.

Power amp and guitar speaker.

See the rig diagram in Section 4 of the Owner's Manual.

Why use a power amp and a guitar speaker.

When you need amplification, and FRFR (see above) is not your thing, you can amplify the processor through a power amp and a guitar speaker.

Traditional cabs move "air" and deliver a punch, which is not always achievable through FRFR. That's why many players still use a traditional cab on stage (backline), even with a direct signal going to the PA system.

Tube power amp for guitar (or head or combo)

When using the processor with a tube power amp designed for guitars (e. g. Mesa, VHT, Fryette) and a traditional cabinet:

Switch off power amp modeling in the processor. Disable cab modeling. Set the controls on the power amp as neutral as possible. Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab.

This also applies when connecting the processor to an effects return port of a guitar combo amp or head.

"If you use a tube power amp and don't turn off power amp modeling in the Axe-Fx you will get the impression that the tube power amp sounds "bigger" and "warmer". This is because the tube power amp will have more bass (and highs) than the solid-state power amp since a tube power amp's response follows the speaker impedance. People will ALWAYS find that more bass and treble sounds "better" when listening alone but in a band context that tone will get lost. Speaker designers have been exploiting this fact of human perception for decades. Many "hi-fi" speakers exaggerate the bass and treble because the uneducated customer will think they sound "better". A truly flat speaker will sound dull in comparison to one with exaggerated lows and highs. Over time, however, those exaggerated frequencies lead to fatigue. It's only in comparison that exaggerated bass and treble sound "better". In an isolated context this aspect of human perception is not evident." source "If you are using a tube power amp you should set any Presence, Depth, Resonance, etc. controls to their minimum positions on that amp (assuming they are conventional controls). On a Mesa power amp, set them to noon. The Presence control on Mesa amps is most neutral around noon. If you turn it up it boosts the highs, if you turn it down it cuts the highs. On most other power amps it only boosts. source.

Tube power amp (neutral)

When using the processor with a "neutral" tube power amp (e. g. Fryette's Power Station) and a traditional cabinet:

Power amp modeling should be turned on. Disable cab modeling. The tube amp interacts with the speaker automatically. It is not necessary to simulate impedance/resonance, so set Low and High Resonance in the Amp block to zero. Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab.

Solid-state power amp.

When using the processor with a "neutral" solid-state power amp (no tubes, e. g. Matrix, Seymour Duncan) and a traditional cabinet:

Power amp modeling should be turned on. Disable cab modeling. Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab. The solid-sate amp doesn't automatically interact with the speaker like a tube amp does. That's where the Resonance parameters in the Amp block are for. The defaults may not get the best results. Optimize the bass response (Low Frequency Resonance) by finding the resonant frequency of the cabinet, like this: Put a Filter block after the Amp block. Set the type to Peaking, Q to 5 or so and Gain to 10 dB. Start with a Frequency of around 50 Hz. Play some chugga-chugga and slowly adjust the Frequency until you hear and feel the cabinet resonate. You need to do this at loud volume level to notice it. Make a note of the frequency. Remove the Filter block and set the Amp block's Low Resonance to match.

Add a Synth block (after the Amp block) to the preset and make sure it is connected to the grid output. Select Sine wave. Turn off Tracking. Turn up the volume of your rig. Adjust Frequency until you hear and feel the cabinet resonate. You need to do this at loud volume level to notice it. Make a note of the frequency. Remove the Synth block and set the Amp block's Low Resonance to match.

Gain-staging.

Make sure not to overload the input of a connected power amp or active monitor.

"The II actually has more output than the I. The II can do about +20 dBu, the I was about +18." source "Start with amp volume at noon. Bring up Axe-Fx volume until desired level is reached. If you need more, turn up amp. With the Axe-Fx volume all the way up you would be pushing +20 dBu into the amp which could clip the inputs to the amp." source.

Wet/Dry/Wet setup (W/D/W)

Speaker wire.

Combining FRFR and power amp + guitar cab.

The Axe-Fx series and AX8 let you combine amplification methods, such as sending a signal WITH cab modeling (FRFR) to one output (like a PA system), and a signal WITHOUT cab modeling to another output to feed a power amp and guitar speaker on stage.

Axe-Fx II and AX8 :

Echo Out2 = Out1 : Enabling this setting in I/O will duplicate the Output 1 signal to Output 2. The Global EQ on Output 2 lets you tailor the tone, independent of Output 1. The Output 1 knobcontrols the level of Output 1, and the Output 2 knob does the same with Output 2. This method DOES NOT WORK with presets that contain a FXL block. This method is great when you want to control the level of your personal monitoring (Out2) separately from the signal that's being sent to FOH (Out1). Tutorial by AxeFxTutorials. FXL block : Insert an FXL block and make it part of the routing but don't connect it to the grid output. The signal before the FXL block will be sent to Output 2. This method is more flexible than the one above, because the position of the FXL block determines which part of the signal is being sent to Output 2. For example, placing FXL before or after a Cabinet block determines whether the Output 2 signal includes cabinet modeling or not. Use this when you want your FOH signal to be "direct" (including cab modeling) and your stage sound to come from a traditional cabinet (without cab modeling). Among the factory presets is a template. The Axe-Fx II lets you put FXL in series or parallel, but the AX8 requires FXL in a parallel row to prevent a feedback loop. Here's a tutorial. Left/Right : Split the signal at the end of the grid into a row with a Cab block and a row with a shunt. In the Output Mixer pan those rows 100% left (Cab) and right (shunt). Now OUT1 Left is the signal with cabinet modeling, and OUT1 Right is the signal without cabinet modeling. This method allows you to use the stereo effects loop for other purposes. source.

Echo Out2 = Out1 : see above. Separate outputs : you can use multiple Output blocks and signal chains to handle multiple outgoing signals. See the rig diagrams in Section 4 of the Owner's Manual.

Four Cable Method (4CM)

Axe-Fx III : See the rig diagram in Section 4 of the Owner's Manual. Use I/O 3 or I/O 4, set to unity gain, adjust Boost/Pad in the I/O menu for optimal SNR, and adjust levels where needed.

Axe-Fx II : Adjust Boost/Pad and Input Level in the I/O menu to optimize the signal. Also, turn the Output Level knob fully open for unity gain. You can't combine 4CM with cab modeling.

"The very early Axe-Fx II's had more bandwidth than necessary on Output 2. The frequency response extended to hundreds of kHz. When used with certain tube amps this would cause instability in the output drivers. The solution was to limit the bandwidth to a "normal" range of 20 to 20 kHz. We provided the update for free and all units shipped after the first 100 or so had this update included. The Axe-Fx II Mark II, XL and XL+ have a redesigned output circuit that is immune from any of these issues." source "It is very difficult to minimize the hiss when putting a digital processor in front of a high-gain amp due to the A/D and D/A conversions. The XL is probably one of the quietest processors made but there will still be some residual hiss when using high gain. The Output 2 Boost/Pad feature was specifically intended to minimize hiss in these scenarios by running the D/A converter as "hot" as possible and then reducing the signal level after the converter with an analog pad." source "The XL+ shares the same amazing low-noise architecture of the FX8. I regularly use my XL+ in 4CM as this is part of the modeling process. It's the quietest device I've ever tried in 4CM." source.

AX8 : Unlike the Axe-Fx and FX8, the AX8 is not optimized for 4CM, but it will work. The process is the same as with the Axe-Fx II.

Effects only.

POST-effects.

When using the Axe-Fx III, II or AX8 as an effects-only device in an amp's effects loop, you probably want it to send and receive line level signals, at unity gain.

Use Input 2 and Output 2 on the Axe-Fx II and AX8, and Input/Output 3 or 4 on the Axe-Fx III. Adjust Input Level for an optimal signal-to-noise ratio. Select the correct input and output settings in I/O. Set the Output 2 knob to it maximum for unity gain.

Test the setup by creating a preset with only shunts. The level should be the same as when leaving out the processor. Then start adding effect blocks (no Amp or Cab).

Axe-Fx III: See the rig diagram in Section 4 of the Owner's Manual.

"You should NOT use Boost/Pad in this configuration." source.

Some amps require inserting a dummy jack into the effects loop's Send to activate the effects loop.

PRE-effects.

Use Output 2 on the Axe-Fx II and AX8, and Input/Output 3 or 4 on the Axe-Fx III. In I/O set output level to -10 dB if possible. Crank Boost/Pad to make sure the full range of the D/A converter is used and turn up the Output knob all the way for unity gain.

Test the setup by creating a preset with only shunts. The level should be the same as when leaving out the processor. Then start adding effect blocks (no Amp or Cab).

Axe-Fx III: See the rig diagram in Section 4 of the Owner's Manual.

As an IR loader.

The Axe-Fx and AX8 can be used as IR loaders.

Connect the guitar to an amp head. Connect the amp's speaker output to a load box, such as Fractal Audio's X-Load. Connect the loadbox to the AX8 or Axe-Fx. Use a preset with a Cab block in it, and without an Amp block.

Digital I/O.

Digital I/O: supported by which Fractal Audio products.

Axe-Fx III : USB, SPDIF, AES. Axe-Fx II : USB, SPDIF, AES. AX8 : SPDIF output only. FX8 : no.

To connect the SPDIF port to an optical port, a SPDIF-to-TOSLINK adapter is required.

Why use digital I/O.

A digital connection skips the analog/digital conversion of the source signal. The analog outputs of the Axe-Fx and AX8 however deliver high-quality results too, similar to digital audio.

Sample rate fixed at 48kHz.

The sample rate of the Axe-Fx series, FX8 and AX8 is fixed at 48kHz (24-bit).

Digitally connected devices and DAW software always need to be set to the same sample rate. Exemplo.

If required, resampling can be handled by software.

"The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC). So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's." "IMHO, the ideal sample rate is 64 kHz but that's not a standard. The nice thing about 64 kHz is that you can have a gentle transition band from 20 kHz to Nyquist which results in shorter filters, lower latency, less phase shift, etc. I was very tempted to make the Axe-Fx II run at 64 kHz but people probably would have freaked out." source "I've long maintained that 64 kHz is the ideal sample rate for audio. But I can't get the industry to change." source "48 kHz is considered "pro" sampling rate. The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment. IMO 44.1 kHz is insufficient for professional audio. Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme. The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz. There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over - sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits. MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs. MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information." source "If the Axe-Fx were running at 44.1 all the cab IRs would need to be resampled, or there would need to be an SRC chip on the digital I/O. Nao tem almoço gratis. The problem isn't the Axe-Fx, the problem is studios stubbornly sticking to 44.1 when 48 is a much better rate." source Xiph: about sampling rates. SonicScoop: The Science of Sample Rates (When Higher Is Better — And When It Isn’t). Discussion. Discussion. Discussion. Discussion.

Axe-Fx II and USB Audio.

Windows : The Axe-Fx II is an Audio Class 2.0 compliant device. A class-compliant device requires no drivers. The drivers are provided by the OS manufacturer. Audio Class 2.0 also encompasses MIDI-over-USB. Microsoft does not support Audio Class 2.0. Therefore FAS provides a driver for Windows systems. The driver for Windows contains both the firmware installer and the audio drivers.

Apple : Apple does support Audio Class 2.0, but poorly. To overcome this, you can increase the buffer size in the Axe-Fx II's I/O > Audio menu. The driver for Macs is NOT an audio driver. It is a firmware installer. The Axe-Fx II uses a "soft" USB controller. It gets its code from the host computer. When you turn the Axe-Fx II on it requests firmware from the host. This is superior to a hard-coded controller in that updates merely require a new host image rather than reflashing the controller.

See the parameter I/O > Audio > Buffer size.

Lower USB Buffer Size in I/O > Audio for less latency, increase when experiencing distorted audio. You should stop USB audio streaming when changing this value so as to allow the buffer to reset properly. Streaming can be stopped by closing the application sending data to the Axe-Fx or by disconnecting the USB cable.

The meters in the Utility menu display the USB performance. Ideally the bar should be at around 50%. If the bar sinks all the way to the bottom or goes all the way to the top, then the buffer may under/overflow and the USB buffer size should be increased. The number of buffer errors that have occurred since the last buffer reset is indicated above the bar graph.

USB Level in I/O > Audio sets the level of the USB input signal sent to the main outputs. If you don't hear anything when monitoring the Axe-Fx through a computer, check this parameter. Also verify the USB/DIGI OUT setting.

USB Audio rate is fixed at 48 kHz, 24-bits.

"The Axe-Fx II USB is 24 bits. This is 144.7 dB of dynamic range. Full-scale is about +20 dBu. So even if your guitar is -20 dBu (-40 dB re. FS) you still have over 100 dB of dynamic range. A typical single coil pickup can easily exceed -20 dBu. A humbucker can easily exceed 0 dBu. Full-scale of 20 dBu gives you a few bits of headroom in case of very hot pickups. The self noise of a guitar pickup and associated electronics limits its dynamic range to less than 100 dB typically." And: "The digital bit depth on the USB and Digital I/O exceeds both the dynamic range of the Axe-Fx itself and certainly that of any guitar. Furthermore the bit depth is sufficient to fully capture the dynamic range of a guitar while still maintaining +20 dBu as full-scale." source "The hardware is incapable of doing 4x4. The only choices are 3x3 or 4x2 and Logic doesn't work with 3x3. We also had some issues with 3x3 in Windows 7 IIRC." source.

Axe-Fx III and USB Audio.

The Axe-Fx III has a dedicated 16-core, 500 MHz USB microcontroller, providing 16 (8x8) channels of USB audio (8 in, 8 out) through USB 2.0.

USB Audio rate is fixed at 48 kHz, 24-bit.

USB In (from Axe-Fx III to Computer) :

1+2: Output 1 (regular stereo output).

3+4: Output 2 (regular stereo output).

5+6: Input 1 (copy of signal at front/rear Instrument input, for reamping, mono).

7+8: Input 2 (copy of signal at Input 2, stereo).

USB Out (from computer to Axe-Fx III) :

1+2: Routed to physical Output 1 L+R (audio from computer, added to OUT1).

3+4: Routed to physical Output 2 L+R (audio from computer, i/e/ backing tracks that can be processed seperately).

5+6: Routed to the Grid via INPUT 1 block when its source is set to USB (for reamping).

7+8: Routed to the Grid via the dedicated INPUT USB block (for additional computer audio).

A USB audio sound source can be placed anywhere on the grid with its own dedicated block.

Latency when using software monitoring is low.

USB performance can be monitored on the Meters page of the Home menu, or in the Utilities menu.

Incoming sound from USB Audio is mixed with the signal that comes out at Output 1. To change this, adjust the sound settings on the computer.

Mac: see section 3 of the Owner's manual. Windows: use ASIO.

"The effective throughput of USB 2.0 is roughly 280 Mb/s. One channel of audio is 48000 samples/s * 24 bits/sample = 1.152 Mb/s. Theoretically you could transfer over 200 channels of audio on USB 2.0." source "The Axe-Fx III has a new driver. I just tested it under Reaper and was able to set the buffer size to 8 with no problems. I typically have it at 256 because I monitor directly from the output but it seems to be working fine on the lowest setting." source.

AES and SPDIF can not be used simultaneously.

AX8 : SPDIF output only. The strength of the SPDIF signal level depends on the position of the front panel output knob (unlike the Axe-Fx II).

"The SPDIF is a digital representation of OUTPUT 1." source.

AES and SPDIF can not be used simultaneously.

Latency when monitoring.

When monitoring audio through the computer's output, latency (the time between playing a note and hearing it) depends on the computer and the USB driver. Higher buffer sizes = higher latency. The Axe-Fx II and III allow you to adjust the USB buffer size.

"The Axe-Fx III has a new driver. I just tested it under Reaper and was able to set the buffer size to 8 with no problems. I typically have it at 256 because I monitor directly from the output but it seems to be working fine on the lowest setting." source.

When monitoring audio directly from the Fractal Audio hardware or hardware audio interface, latency is below 2 ms.

Master or slave.

Word Clock.

The clock source for the A/D and D/A converters is either AUTO/INTERNAL or SPDIF/AES.

Axe-Fx III : Word Clock is recovered from the SPDIF/AES input signal.

"Yes via SPDIF/AES in (which actually works better as a word clock than a word clock input)." source.

AX8 and FX8 : not supported.

Auto : uses the internal clock if the input source is Analog or USB, uses the recovered SPDIF/AES clock if the input is SPDIF/AES. SPDIF/AES IN : uses the recovered clock for all input sources. A valid 48 kHz data stream must be present at the AES or SPDIF input. If a valid stream is not detected, the unit will fall back to the internal clock and display "NO INPUT CLOCK!". The SPDIF/AES select must be set to the appropriate value, i. e. if the data stream is input to the XLR jack then SPDIF/AES SELECT must be set to AES. "Set Word Clock to SPDIF/AES In. Connect a cable from the ULN-8 to AES In or SPDIF In. Set SPDIF/AES Select to appropriate input used." source "The Axe-Fx II will derive its clock from the AES/SPDIF when using Digital In. In Analog In it uses its internal clock." source.

Simultaneous analog and digital audio.

Axe-Fx II : you can use digital input together with analog output and simultaneous digital out. Also, you can use Input/Output 2 while using the digital input. source.

How to mute hardware when listening to DAW.

When the Axe-Fx II is connected to a DAW through USB, and you have monitors connected to the Axe-Fx and you're recording, you may want to monitor just the DAW signal, not the signal from the Axe-Fx.

To accomplish this, use one of these methods:

Set USB/DIGI Out Source to Output 2, set Output 2 Echo to Output 1 and lower gain in Output 1's Global EQ gain slider. Set USB/DIGI Out Source to Output 2, set Output 2 Echo to Output 1 and send a 0 value for Output 1 Volume's MIDI CC. Set USB/DIGI Out Source to Output 2, put an FX Loop block at the end of the grid chain and connect your signal row(s) to it instead of the output block.

The Axe-Fx III's expanded I/O capabilities allow easier control.

Use the processor as an A/D converter.

To use the device as an analog-to-digital converter: (source)

Create a preset with nothing but shunts from input to output. Or set Input 1 Left Select to to Rear and plug the device into Input 1 Left on the back (Axe-Fx II). Or connect In and Out and engage Bypass Mode (Axe-Fx II).

Connecting the Axe-Fx to a DAW drops the volume or puts it into bypass.

Some DAWs send MIDI commands (intended for other devices) which lower the volume of the Axe-Fx or put it into Bypass mode. To solve this, adjust the DAW settings. If not possible, force the Axe-Fx to ignore those commands by setting MIDI CCs 10 and 13 in I/O > Control to "none".

More information about digital I/O.

Headphones.

Axe-Fx II and III : Use the Output 1 knob to set the headphones level. If you have both monitors and headphones connected and want to listen through headphones only, use one of these methods:

Switch off the monitors. Feed the studio monitors through another output. Use a line level attenuator to turn down the monitors' level without affecting the headphones level.

AX8 : The AX8 does not have a dedicated headphones output. You can use a Y-cable with the outputs. If you connect your headphones directly to the AX8, you may get less signal level than from a dedicated headphones output on other devices. A popular solution is to use a small headphones amp, e. g. from M-Audio or Rolls.

"It does not have a headphone output but the outputs should be able to drive phones with ease. You'd just need a Y-cable adapter." source "The output impedance of the 1/4" outputs is 600 ohms IIRC. This may be too high for some headphones. We always use a small output impedance on our designs to help protect the output devices against improper connections, ESD, etc. The outputs were not really designed to drive headphones, they are designed to drive high-impedance inputs (>10K). Headphones will work but it won't be optimum. For optimum results use a dedicated headphone amp." source.

Impedance : If the volume level through your headphones is very low, switch to lower impedance headphones (such as 32 Ohm), use a headphone amp or use headphones with a built-in amplifier (such as Blue Mo-Fi). The Axe-Fx III is better at feeding sufficient signal level to hard-to-drive headphones than the Axe-Fx II. Article about headphones and impedance.

Sound through headphones can be dull.

"Because there's no string and body reinforcement. When you play through speakers the sound couples into the guitar body and strings. With headphones you don't get this so the sound is very sterile and lifeless. Now, if you use speakers during recording and then playback through headphones it will sound fine." "It's lack of acoustic reinforcement. I did a test a few years ago and I don't remember the actual numbers but having a speaker aimed at the guitar adds many dBs of power to the lower mids coming out of the guitar. IOW, if you measure the spectrum of the signal coming out of a guitar alone and then compare that to the signal coming out with a cab or monitor in proximity at a reasonable volume there are a LOT more lower mids with the speaker present. This results in a "thin" sound without the speaker." source "The problem with headphones is that there is no acoustic reinforcement of the guitar. There is zero coupling between the speakers (inside the headphones) and the guitar. Without that coupling, which is a type of positive feedback, the sound is lifeless, thin and harsh. When your heroes recorded their guitar parts that weren't using headphones. On "Appetite for Destruction" Slash recorded his guitar parts in the control room. To get even more coupling into the guitar a combo amp was in the control room with him pointed at the guitar. A volume pedal allowed him to adjust the volume of the combo amp so he could control the coupling. With the volume pedal all the way up he could get controlled feedback. I've actually done tests comparing the spectrum out of the guitar when there is no coupling (i. e. monitors turned off) and with typical coupling (monitors loud or using a conventional cab). The boost in the low midrange is significant. I forget the actual numbers but it was at least several dB." source "I did some studies years ago and having a speaker in proximity to the guitar actually changes the final tone considerably. I compared the frequency response with the amp in isolation to the frequency response with the amp in proximity and measured several dB difference in the lows and mids. It was clearly audible when the recordings were played back." source.

Tips for improving sound quality through headphones:

Use far-field Impulse Responses. Increase Proximity in the Cabinet block. Increase Room Level in the Cabinet block. Always use a stereo input signal. Add the Stereo Enhancer to presets, or add Micro Delay in the Cab block. Increase De-Phase in the Cab block. Use Speaker Compression in the Amp block.

More information about headphones:

In-Ear Monitoring (IEM)

In-Ear Monitoring (IEM) provides a way for musicians to monitor sound through earbuds, instead of floor wedges etc. While this provides a superior listening environment, it takes getting used to the direct sound into your ears and the absence of surround sounds.

The expanded I/O architecture of the Axe-Fx III makes it easy to send and receive IEM-specific signals, without the need for an external mixer.

Warning: always use the built-in limiter of your IEM system to protect your ears against sudden spikes and peaks!

The same tips as for headphones apply.

Setups: FX8.

Pedalboard (PRE effects)

Guitar goes into IN [PRE] / INSTR. Note: this input only feeds effect blocks designated as PRE. OUT [PRE] LEFT goes into the amplifier's guitar input. Use a Humbuster cable to prevent noise. You can use default FX8 settings. Exception: change the output mode in I/O > Audio (see manual for stereo operation)to Mono (see manual for stereo operation).

You can also use this setup to connect an FX8 to the Axe-Fx.

More information in the Owner's Manual, including a description of the cables required.

In amplifier's effects loop (post-effects)

Guitar goes straight into the amplifier. Amp's effects loop SEND goes into IN [POST] LEFT. Amp's effects loop RETURN goes into OUT [POST] LEFT. Use a Humbuster cable to prevent noise. You can use default FX8 settings. Exceptions: Change the output mode in I/O > Audio to Mono (see manual for stereo operation). Change Global Looper Location to OUT POST. Change Global Detector to IN [POST]. "Are the outputs buffered for long cable runs? Yes." source.

More information in the Owner's Manual, including a description of the cables required.

Four Cable Method (4CM)

The FX8 can be set up to put effects before the amp as well as in the amp's effects loop.

Guitar goes into IN [PRE] / INSTR. OUT [PRE] LEFT goes into the amp's guitar's input. Use a Humbuster cable to prevent noise. The amp's effects loop SEND goes into IN [POST] LEFT. The amp's effects loop RETURN goes into OUT [POST] LEFT. Use a Humbuster cable to prevent noise. You can use default FX8 settings. Exceptions: Change the output mode in I/O > Audio to Mono (see manual for stereo operation). Change Global Looper Location to OUT POST. "Are the outputs buffered for long cable runs? Yes." source.

If you have the FX8 set up for 4CM and want to change this, for example to put the FX8 before a computer, just use a jumper cable to connect OUT PRE L MONO to IN POST L, with OUT POST L going to the computer, amp or whatever. All effects will work and there's no need to change stuff in the configuration.

If you're using the FX8 in a 4CM setup and you're experiencing hiss, try another Post Level value.

More information in the Owner's Manual, including a description of the cables required.

Combine with Axe-Fx II, Axe-Fx III or AX8.

You can use the FX8 for "pre" effects (plug guitar into FX8) and the Axe-Fx or AX8 for post-effects, including amp and cabinet modeling (plug FX8 into Axe-Fx or AX8). By adding a MIDI connection you can change Axe-Fx and AX8 presets from the FX8.

Relays: switch amp channels and more.

CAUTION: Do NOT connect anything to the relays jacks until you've read the warnings in the manual!

What are relays? Relays are electrically operated switches/connectors, which can be used to switch channels on an amplifier and switch other stuff.

How many relays does the FX8 have? The FX8 has two relays.

How can I control these relays? These are controlled through:

Scenes : you can use scenes to switch amp channels through relays. This is configured on the preset's Config page. Footswitches : you can assign footswitches to the relays per preset, for manual control. Assign the footswitch and configure it on the Footswitch page.

IMPORTANT: a Relay block in the preset will disable the scene's Relay settings.

Do the relays support X/Y switching? The relays support X/Y switching.

What are the possible settings? The relay states are:

Off: nothing connected. Tip: tip to Sleeve. Ring: ring to sleeve. Both: tip AND ring to sleeve.

What are the switch modes of the relays? The switch modes of the relays are:

Latching: the selected RELAY ON state remains connected and the switch LED remains ON as long as the switch is engaged. Nothing is connected when the switch is OFF. Auto-Off: the selected RELAY ON state remains connected only for a moment when you press the footswitch. The relay then automatically turns OFF, as does the LED.

Which cables can be used? Depending on the amp, you can use TS or TRS cables.

"The FX8 will short tip-to-sleeve, ring-to-sleeve, or both. The circuit is designed to handle 200mA of current. If the current generated by that voltage drop is 200mA or less, then the FX8 will not have a problem." source "The relays of the FX8 are designed for use ONLY with amplifiers that use “short-to-sleeve” type switching. Do NOT connect the FX8 relays to the switch jacks of an amp that uses voltage differential switching or any other type of switching aside from short-to-sleeve, or serious damage can occur to both units. If you are not 100% sure, contact your amp manufacturer to determine whether your amp is compatible with short-to-sleeve switching. The FX8 relay jacks are compatible with TRS cables, TS cables, or TRS-to-dual-TS split cables. The relays are also fully isolated from the electrical ground of the FX8." "The FX8 features two TRS (Tip-Ring-Sleeve) relays that can be used to switch the channel or other functions of a connected amplifier or device. If the warning above seems stern, that’s because the last thing we want is for anyone to damage their amp or FX8. In fact, short-to-sleeve relay switched amps are quite common, and your amp may well be perfectly compatible. We need to trust and require you however, to understand how your amp works and make the right choices about connecting it to the FX8 relay jacks. Your amp manufacturer should be able to help if you read them the warning above." "The FX8 relay outputs employ a "Short-to-Sleeve" connection. Each relay output can short Tip-to-Sleeve, Ring-to-Sleeve, and Both. If the pedal connection uses a voltage drop to power an LED, the relay circuit on the FX8 is rated for a maximum of 200mA." source.

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Axe Change - The Official Site for Fractal Audio Presets, Cabs and More.

The "Setup" filter helps you to focus on those presets best-suited for how you use your Axe-Fx.

FRFR/Direct presets (like all of the Factory presets) are designed for "full-range flat-response" systems such as studio monitors, PAs, headphones, etc.

Pwr. Amp + Cab presets use real power amps and guitar speakers.

4-Cable Method (aka :4CM") presets require a special setup with connectors both in front of and in the effects loop of a real amp.

Fx Loop presets in the loop of a real amp. A loop may be SERIES or PARALLEL and the preset needs to be built accordingly.

Gtr Amp Input presets should be used in front of a (real) head or combo. Guitar->Axe-Fx->Amp Input.

Other setups also exist (e. g. combinations of the above).

The Axe-Fx II manual (chapter three) includes more information about setups.

"Cab" search results ignore the settings of this filter.

Search Results: Periphery.

Presets (13) Cabs (0) Advanced Search?

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