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authorStefano Pigozzi <stefano.pigozzi@gmail.com>2013-03-09 09:30:26 +0100
committerwm4 <wm4@nowhere>2013-03-13 23:51:30 +0100
commit048ceef655bce41bc6e215b5e05cec0fad4d1428 (patch)
treec448eee1a1e8161b6e1c3455b2d87af5660e8e8f /DOCS/man
parent514d8a7c9dfde2acc89ee4d19dd9db6b9db5b882 (diff)
downloadmpv-048ceef655bce41bc6e215b5e05cec0fad4d1428.tar.bz2
mpv-048ceef655bce41bc6e215b5e05cec0fad4d1428.tar.xz
af_lavrresample: add new resampling filter to replace the old ones
Remove `af_resample` and `af_lavcresample`. The former is a mess while the latter uses an API that was long deprecated in libavcodec and is now removed. `af_lavrresample` rougly has the same features and structure of `af_lavcresample`. libswresample fallback by wm4.
Diffstat (limited to 'DOCS/man')
-rw-r--r--DOCS/man/en/af.rst56
1 files changed, 14 insertions, 42 deletions
diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst
index a47a63b8e2..2eaeabed0c 100644
--- a/DOCS/man/en/af.rst
+++ b/DOCS/man/en/af.rst
@@ -29,55 +29,27 @@ filter list.
Available filters are:
-resample[=srate[:sloppy[:type]]]
- Changes the sample rate of the audio stream. 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 is automatically enabled if
- necessary. It only supports 16-bit integer and float in native-endian
- format as input.
-
- <srate>
- output sample frequency in Hz. The valid range for this parameter is
- 8000 to 192000. If the input and output sample frequency are the same
- or if this parameter is omitted the filter is automatically unloaded.
- A high sample frequency normally improves the audio quality,
- especially when used in combination with other filters.
- <sloppy>
- Allow (1) or disallow (0) the output frequency to differ slightly from
- the frequency given by <srate> (default: 1). Can be used if the
- startup of the playback is extremely slow.
- <type>
- Select which resampling method to use.
-
- :0: linear interpolation (fast, poor quality especially when
- upsampling)
- :1: polyphase filterbank and integer processing
- :2: polyphase filterbank and floating point processing
- (slow, best quality)
-
- *EXAMPLE*:
-
- ``mpv --af=resample=44100:0:0``
- would set the output frequency of the resample filter to 44100Hz using
- exact output frequency scaling and linear interpolation.
-
-lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
+lavrresample[=option1:option2:...]
Changes the sample rate of the audio stream to an integer <srate> in Hz.
- It only supports the 16-bit native-endian format.
+ 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 is automatically enabled if necessary. It only supports the
+ 16-bit integer native-endian format.
- <srate>
- the output sample rate
- <length>
+ srate=<srate>
+ the output sample rate (defaut: 44100)
+ length=<length>
length of the filter with respect to the lower sampling rate (default:
16)
- <linear>
- if 1 then filters will be linearly interpolated between polyphase
- entries
- <count>
+ phase_shift=<count>
log2 of the number of polyphase entries (..., 10->1024, 11->2048,
12->4096, ...) (default: 10->1024)
- <cutoff>
+ cutoff=<cutoff>
cutoff frequency (0.0-1.0), default set depending upon filter length
+ linear
+ if set then filters will be linearly interpolated between polyphase
+ entries (default: no)
lavcac3enc[=tospdif[:bitrate[:minchn]]]
Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports