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authorwm4 <wm4@nowhere>2013-03-29 22:41:47 +0100
committerwm4 <wm4@nowhere>2013-04-13 04:21:29 +0200
commitd2c20f4cfbfe01bdf63c2a76f20d39212598440e (patch)
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parentff6342a3112e194c228660516cc19dd1ce4de6d9 (diff)
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manpage: update af_lavrresample entry
Reflects the changes over the last few commits.
Diffstat (limited to 'DOCS')
-rw-r--r--DOCS/man/en/af.rst12
1 files changed, 6 insertions, 6 deletions
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 <srate> 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=<srate>
- the output sample rate (defaut: 44100)
+ the output sample rate
length=<length>
length of the filter with respect to the lower sampling rate (default:
16)