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manpage: update af_lavrresample entry
Reflects the changes over the last few commits.
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@ -30,15 +30,15 @@ filter list.
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Available filters are:
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lavrresample[=option1:option2:...]
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Changes the sample rate of the audio stream to an integer <srate> in Hz.
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Can be used if you have a fixed frequency sound card or if you are stuck
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with an old sound card that is only capable of max 44.1kHz.
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This filter uses libavresample (or libswresample, depending on the build)
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to change sample rate, sample format, or channel layout of the audio stream.
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This filter is automatically enabled if the audio output doesn't support
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the audio configuration of the file being played.
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This filter is automatically enabled if necessary. It only supports the
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16-bit integer native-endian format.
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It supports only the following sample formats: u8, s16ne, s32ne, floatne.
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srate=<srate>
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the output sample rate (defaut: 44100)
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the output sample rate
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length=<length>
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length of the filter with respect to the lower sampling rate (default:
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16)
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