Update translation_pt_PT.ts
This commit is contained in:
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</message>
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<message>
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<location filename="../../audiomixerboard.cpp" line="133"/>
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<source>Sets the panning position from Left to Right of the channel. Works only in stero or preferably mono in/stereo out mode.</source>
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<source>Sets the panning position from Left to Right of the channel. Works only in stereo or preferably mono in/stereo out mode.</source>
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<translation type="unfinished"></translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientdlg.cpp" line="88"/>
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<source>Push this button to connect a server. A dialog where you can select a server will open. If you are connected, pressing this button will end the session.</source>
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<source>Push this button to connect to a server. A dialog where you can select a server will open. If you are connected, pressing this button will end the session.</source>
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<translation>Pressione este botão para se ligar a um servidor. Uma janela será aberta onde pode selecionar um servidor. Se já estiver ligado a um servidor, pressionar este botão encerrará a sessão.</translation>
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</message>
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<message>
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@ -623,7 +623,7 @@
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</message>
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<message>
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<location filename="../../clientdlg.cpp" line="125"/>
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<source>The reverberation effect requires significant CPU so that it should only be used on fast PCs. If the reverberation level fader is set to minimum (which is the default setting), the reverberation effect is switched off and does not cause any additional CPU usage.</source>
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<source>The reverberation effect requires significant CPU so it should only be used on fast PCs. If the reverberation level fader is set to minimum (which is the default setting), the reverberation effect is switched off and does not cause any additional CPU usage.</source>
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<translation>O efeito de reverberação requer uma utilização do CPU significativa, de forma a que só deve ser usado em PCs rápidos. Se o atenuador do nível de reverberação estiver definido como mínimo (que é a configuração padrão), o efeito de reverberação será desativado e não causará nenhum uso adicional do CPU.</translation>
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</message>
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<message>
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@ -663,7 +663,7 @@
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</message>
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<message>
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<location filename="../../clientdlg.cpp" line="172"/>
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<source>The sound card buffer delay (buffer size) is set to a too small value.</source>
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<source>The sound card buffer delay (buffer size) is set to too small a value.</source>
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<translation>O atraso do buffer da placa de som (buffer size) está definido para um valor demasiado baixo.</translation>
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</message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="41"/>
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<source>The jitter buffer size can be manually chosen for the local client and the remote server. For the local jitter buffer, dropouts in the audio stream are indicated by the light on the bottom of the jitter buffer size faders. If the light turns to red, a buffer overrun/underrun took place and the audio stream is interrupted.</source>
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<source>The jitter buffer size can be manually chosen for the local client and the remote server. For the local jitter buffer, dropouts in the audio stream are indicated by the light below the jitter buffer size faders. If the light turns to red, a buffer overrun/underrun took place and the audio stream is interrupted.</source>
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<translation>O tamanho do jitter buffer pode ser escolhido manualmente para o cliente local e o servidor remoto. Para o jitter buffer local, as interrupções no fluxo de áudio são indicadas pela luz na parte inferior dos faders do jitter buffer. Se a luz ficar vermelha, ocorreu um excesso/déficit do buffer e o fluxo de áudio é interrompido.</translation>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="56"/>
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<source>In case the auto setting of the jitter buffer is enabled, the network buffers of the local client and the remote server are set to a conservative value to minimize the audio dropout probability. To tweak the audio delay/latency it is recommended to disable the auto setting functionality and to lower the jitter buffer size manually by using the sliders until your personal acceptable limit of the amount of dropouts is reached. The LED indicator will visualize the audio dropouts of the local jitter buffer by a red light.</source>
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<source>If the auto setting of the jitter buffer is enabled, the network buffers of the local client and the remote server are set to a conservative value to minimize the audio dropout probability. To tweak the audio delay/latency it is recommended to disable the auto setting functionality and to lower the jitter buffer size manually by using the sliders until your personal acceptable limit of the amount of dropouts is reached. The LED indicator will visualize the audio dropouts of the local jitter buffer with a red light.</source>
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<translation>Caso a configuração automática do jitter buffer estiver ativada, os buffers de rede do cliente local e do servidor remoto são configurados com um valor conservador para minimizar a probabilidade de perda de áudio. Para ajustar o atraso/latência do áudio, é recomendável desativar a funcionalidade de configuração automática e diminuir o tamanho do jitter buffer manualmente usando os controles deslizantes até que a quantidade de perdas de áudio lhe sejam pessoalmente aceitáveis. O indicador LED representará as interrupções de áudio do jitter buffer local através de uma luz vermelha.</translation>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="96"/>
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<source>In case the ASIO4ALL driver is used, please note that this driver usually introduces approx. 10-30 ms of additional audio delay. Using a sound card with a native ASIO driver is therefore recommended.</source>
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<source>If the ASIO4ALL driver is used, please note that this driver usually introduces approx. 10-30 ms of additional audio delay. Using a sound card with a native ASIO driver is therefore recommended.</source>
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<translation>Caso o driver ASIO4ALL seja usado, note que esse driver geralmente introduz aprox. 10-30 ms de atraso de áudio adicional. Dado isto, é recomendável usar uma placa de som com um driver ASIO nativo.</translation>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="106"/>
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<source>In case the selected sound card device offers more than one input or output channel, the Input Channel Mapping and Output Channel Mapping settings are visible.</source>
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<source>If the selected sound card device offers more than one input or output channel, the Input Channel Mapping and Output Channel Mapping settings are visible.</source>
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<translation>Caso o dispositivo selecionado da placa de som ofereça mais que um canal de entrada ou saída, as configurações de Mapeamento de canais de entrada e de saída estarão visíveis.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="128"/>
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<source> samples. The smaller the network buffers, the smaller the audio latency. But at the same time the network load increases and the probability of audio dropouts also increases.</source>
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<source> samples. The smaller the network buffers, the lower the audio latency. But at the same time the network load increases and the probability of audio dropouts also increases.</source>
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<translation> amostras. Quanto menor o buffer da rede, menor a latência do áudio. Mas, ao mesmo tempo, a carga da rede e a probabilidade de interrupção do áudio também aumentam.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="141"/>
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<source>64 samples: This is the preferred setting since it gives lowest latency but does not work with all sound cards.</source>
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<source>64 samples: This is the preferred setting since it provides the lowest latency but does not work with all sound cards.</source>
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<translation>64 amostras: esta é a configuração preferida, pois oferece menor latência, mas não funciona com todas as placas de som.</translation>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="143"/>
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<source>128 samples: This setting should work on most of the available sound cards.</source>
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<source>128 samples: This setting should work for most available sound cards.</source>
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<translation>128 amostras: esta configuração deve funcionar na maioria das placas de som disponíveis.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="147"/>
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<source>Some sound card driver do not allow the buffer delay to be changed from within the </source>
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<source>Some sound card drivers do not allow the buffer delay to be changed from within the </source>
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<translation>Alguns drivers da placa de som não permitem que o atraso do buffer seja alterado pelo cliente </translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="158"/>
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<source>The actual buffer delay has influence on the connection status, the current upload rate and the overall delay. The lower the buffer size, the higher the probability of red light in the status indicator (drop outs) and the higher the upload rate and the lower the overall delay.</source>
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<source>The actual buffer delay has influence on the connection status, the current upload rate and the overall delay. The lower the buffer size, the higher the probability of a red light in the status indicator (drop outs) and the higher the upload rate and the lower the overall delay.</source>
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<translation>O atraso do buffer influencia o estado da ligação, a taxa de upload atual e a latência geral. Quanto menor o atraso do buffer, maior a probabilidade de a luz vermelha no indicador de estado (interrupções), maior a taxa de upload e menor a latência geral.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="200"/>
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<source>Select the number of audio channels to be used. There are three modes available. The mono and stereo modes use one and two audio channels respectively. In the mono-in/stereo-out mode the audio signal which is sent to the server is mono but the return signal is stereo. This is useful for the case that the sound card puts the instrument on one input channel and the microphone on the other channel. In that case the two input signals can be mixed to one mono channel but the server mix can be heard in stereo.</source>
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<source>Select the number of audio channels to be used. There are three modes available. The mono and stereo modes use one and two audio channels respectively. In mono-in/stereo-out mode the audio signal which is sent to the server is mono but the return signal is stereo. This is useful if the sound card has the instrument on one input channel and the microphone on the other channel. In that case the two input signals can be mixed to one mono channel but the server mix can be heard in stereo.</source>
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<translation>Selecione o número de canais de áudio a serem usados. Existem três modos disponíveis. Os modos mono e estéreo usam um e dois canais de áudio, respectivamente. No modo Entrada Mono/Saída Estéreo, o sinal de áudio enviado ao servidor é mono, mas o sinal de retorno é estéreo. Isso é útil quando a placa de som coloca o instrumento e o microfone em canais diferentes. Nesse caso, os dois sinais de entrada podem ser misturados num canal mono, mas a mistura do servidor pode ser ouvida em estéreo.</translation>
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<message>
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</message>
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<message>
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<location filename="../../clientsettingsdlg.cpp" line="261"/>
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<source>The ping time is the time required for the audio stream to travel from the client to the server and backwards. This delay is introduced by the network. This delay should be as low as 20-30 ms. If this delay is higher (e.g., 50-60 ms), your distance to the server is too large or your internet connection is not sufficient.</source>
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<source>The ping time is the time required for the audio stream to travel from the client to the server and back again. This delay is introduced by the network. This delay should be as low as 20-30 ms. If this delay is higher (e.g., 50-60 ms), your distance to the server is too large or your internet connection is not sufficient.</source>
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<translation>A latência da ligação é o tempo necessário para o fluxo de áudio viajar do cliente para o servidor e vice-versa. Esta latência é introduzida pela rede. Esta latência deve ser tão baixa quanto 20-30 ms. Se esta latência for maior (por exemplo, 50-60 ms), a distância até ao servidor é muito grande ou sua ligação à Internet não é suficiente.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../util.cpp" line="705"/>
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<source>Set your name or an alias here so that the other musicians you want to play with know who you are. Additionally you may set an instrument picture of the instrument you play and a flag of the country you are living. The city you live in and the skill level of playing your instrument may also be added.</source>
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<source>Set your name or an alias here so that the other musicians you want to play with know who you are. Additionally you may set an instrument picture of the instrument you play and a flag of the country you are living in. The city you live in and the skill level playing your instrument may also be added.</source>
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<translation>Defina o seu nome ou um pseudônimo aqui para que os outros músicos com quem quer tocar saibam quem você é. Além disso, pode definir uma imagem do instrumento que toca e uma bandeira do país em que vive. A cidade em que vive e o nível de habilidade com o seu instrumento também podem ser adicionados.</translation>
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<message>
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</message>
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<message>
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<location filename="../../serverdlg.cpp" line="68"/>
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<source> users can see the server in the connect dialog server list and connect to it. The registering of the server is renewed periodically to make sure that all servers in the connect dialog server list are actually available.</source>
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<source> users can see the server in the connect dialog server list and connect to it. The registration of the server is renewed periodically to make sure that all servers in the connect dialog server list are actually available.</source>
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<translation> possam ver o servidor na lista do diálogo de ligação e ligar-se a ele. O registo do servidor é renovado periodicamente para garantir que todos os servidores na lista de diálogo de ligação estejam realmente disponíveis.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../serverdlg.cpp" line="75"/>
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<source>If the Make My Server Public check box is checked, this will show the success of registration with the central server.</source>
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<source>If the Make My Server Public check box is checked, this will show whether registration with the central server is successful.</source>
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<translation>Se a caixa de seleção Tornar Servidor Público estiver marcada, isto mostrará o sucesso ou insucesso do registo no servidor central.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../../windows/sound.cpp" line="121"/>
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<source>The audio device does not support to set the required sampling rate. This error can happen if you have an audio interface like the Roland UA-25EX where you set the sample rate with a hardware switch on the audio device. If this is the case, please change the sample rate to </source>
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<source>The audio device does not support setting the required sampling rate. This error can happen if you have an audio interface like the Roland UA-25EX where you set the sample rate with a hardware switch on the audio device. If this is the case, please change the sample rate to </source>
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<translation>O dispositivo de áudio não suporta definir a taxa de amostragem (sample rate) necessária. Este erro pode ocorrer se você tiver uma interface de áudio como o Roland UA-25EX, onde se define a taxa de amostragem através de um interruptor de hardware no dispositivo de áudio. Se for esse o caso, altere a taxa de amostragem para </translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../../windows/sound.cpp" line="519"/>
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<source> software requires the low latency audio interface ASIO to work properly. This is no standard Windows audio interface and therefore a special audio driver is required. Either your sound card has a native ASIO driver (which is recommended) or you might want to use alternative drivers like the ASIO4All driver.</source>
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<source> software requires the low latency audio interface ASIO to work properly. This is not a standard Windows audio interface and therefore a special audio driver is required. Either your sound card has a native ASIO driver (which is recommended) or you might want to use alternative drivers like the ASIO4All driver.</source>
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<translation> requer que a interface de áudio de baixa latência ASIO funcione corretamente. Esta não é uma interface de áudio padrão do Windows e, portanto, é necessário um driver de áudio especial. Ou a sua placa de som possui um driver ASIO nativo (recomendado), ou pode usar drivers alternativos, como o driver ASIO4All.</translation>
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</message>
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<message>
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</message>
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<message>
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<location filename="../../soundbase.cpp" line="141"/>
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<source>The audio driver properties have changed to a state which is incompatible to this software. The selected audio device could not be used because of the following error:</source>
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<source>The audio driver properties have changed to a state which is incompatible with this software. The selected audio device could not be used because of the following error:</source>
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<translation>As propriedades do driver de áudio foram alteradas para um estado incompatível com este programa. O dispositivo de áudio selecionado não pôde ser usado devido ao seguinte erro:</translation>
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</message>
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<message>
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