From 048ceef655bce41bc6e215b5e05cec0fad4d1428 Mon Sep 17 00:00:00 2001 From: Stefano Pigozzi Date: Sat, 9 Mar 2013 09:30:26 +0100 Subject: 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. --- DOCS/man/en/af.rst | 56 ++++++++++++++---------------------------------------- 1 file changed, 14 insertions(+), 42 deletions(-) (limited to 'DOCS/man') 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. - - - 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. - - Allow (1) or disallow (0) the output frequency to differ slightly from - the frequency given by (default: 1). Can be used if the - startup of the playback is extremely slow. - - 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 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. - - the output sample rate - + srate= + the output sample rate (defaut: 44100) + length= length of the filter with respect to the lower sampling rate (default: 16) - - if 1 then filters will be linearly interpolated between polyphase - entries - + phase_shift= log2 of the number of polyphase entries (..., 10->1024, 11->2048, 12->4096, ...) (default: 10->1024) - + 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 -- cgit v1.2.3