From d2c20f4cfbfe01bdf63c2a76f20d39212598440e Mon Sep 17 00:00:00 2001 From: wm4 Date: Fri, 29 Mar 2013 22:41:47 +0100 Subject: manpage: update af_lavrresample entry Reflects the changes over the last few commits. --- DOCS/man/en/af.rst | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'DOCS') diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst index 7babf874d4..5c08b5f048 100644 --- a/DOCS/man/en/af.rst +++ b/DOCS/man/en/af.rst @@ -30,15 +30,15 @@ filter list. Available filters are: lavrresample[=option1:option2:...] - Changes the sample rate of the audio stream to an integer in Hz. - Can be used if you have a fixed frequency sound card or if you are stuck - with an old sound card that is only capable of max 44.1kHz. + This filter uses libavresample (or libswresample, depending on the build) + to change sample rate, sample format, or channel layout of the audio stream. + This filter is automatically enabled if the audio output doesn't support + the audio configuration of the file being played. - This filter is automatically enabled if necessary. It only supports the - 16-bit integer native-endian format. + It supports only the following sample formats: u8, s16ne, s32ne, floatne. srate= - the output sample rate (defaut: 44100) + the output sample rate length= length of the filter with respect to the lower sampling rate (default: 16) -- cgit v1.2.3