From c8b8fe9981c654c0539ca77056ed6451a3da7367 Mon Sep 17 00:00:00 2001 From: wm4 Date: Sun, 30 Jun 2019 01:10:44 +0200 Subject: audio: remove unreferenced af_lavrresample This filter wasn't referenced anywhere and thus was dead code. It should have been in the audio filter list in user_filters.c. This was intended as compatibility wrapper (to avoid breaking old command lines and config files), and has no real use. Apparently I forgot to add it to the filter list (did I even test this shit?), and so it was rotting around for 1.5 years doing nothing (just like myself). Note that users can just use the libavfilter provided filter to force resampling, just that it has a different name and different options. There's also af_format to force inserting auto conversion through the internal f_swsresample filter. --- DOCS/man/af.rst | 38 -------------------------------------- 1 file changed, 38 deletions(-) (limited to 'DOCS') diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst index 4e806d482e..aebf76431a 100644 --- a/DOCS/man/af.rst +++ b/DOCS/man/af.rst @@ -24,44 +24,6 @@ See ``--vf`` group of options for info on how ``--af-defaults``, ``--af-add``, Available filters are: -``lavrresample[=option1:option2:...]`` - 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 does not support - the audio configuration of the file being played. - - .. warning:: - - Deprecated. Either use the ``--audio-resample-...`` options to customize - resampling, or the libavfilter ``--af=aresample`` filter, which has its - own options. - - It supports only the following sample formats: u8, s16, s32, float. - - ``filter-size=`` - Length of the filter with respect to the lower sampling rate. (default: - 16) - ``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) - ``no-detach`` - Do not detach if input and output audio format/rate/channels match. - (If you just want to set defaults for this filter that will be used - even by automatically inserted lavrresample instances, you should - prefer setting them with the ``--audio-resample-...`` options.) This - does not do anything anymore and the filter will never detach. - ``normalize=`` - Whether to normalize when remixing channel layouts (default: auto). - ``auto`` uses the value set by ``--audio-normalize-downmix``. - ``o=`` - Set AVOptions on the SwrContext or AVAudioResampleContext. These should - be documented by FFmpeg or Libav. - ``lavcac3enc[=options]`` Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is -- cgit v1.2.3