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* m_option: drop support for -del for list optionsDudemanguy2023-10-301-1/+1
| | | | | | 5f74ed58286a1339412554932f31844ec1b64280 deprecated this many years ago. The utility is questionable at best given that -remove exists and is more natural to use. Free up some code and drop it.
* various: remove trailing whitespaceGuido Cella2023-10-301-1/+1
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* af_scaletempo2: better defaultsChristoph Heinrich2023-10-151-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Why a bigger search-interval is required: scaletempo2 doesn't do a good job when the signal contains frequencies less then 1/search_interval. With a search interval of 30ms that means anything below 33.333Hz sounds bad. Depending on the genre it's very for music to contain frequencies down to 30Hz, and sometimes even a little bit below that. Therefore a higher default value is needed to handle such cases. Based on that an argument can be made for a value of 50, as that should work down to 20Hz, or something even higher because movies sometimes have some infrasonic content. However the downside of big search intervals is increased CPU usage and intelligibility at higher speeds, as it effectively leads to parts of the audio being skipped. A value of 40 can handle frequencies down to 25Hz, enough for all music except very rare edge cases, while still providing decent intelligibility. Why a smaller window-size is required: Large values reduce intelligibility at high speeds and therefore small values are preferred. However when values get too small it starts to sound weird (similar to librubberband). In my testing a value of 10 already works well, but adding a small safety margin seems like a good idea, especially since it made no noticeable difference to intelligibility, which is why 12 was chosen.
* af_scaletempo: overlap is a factor not a percentageChristoph Heinrich2023-10-071-2/+2
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* af_scaletempo2: raise max playback rate to 8.0llyyr2023-09-271-1/+1
| | | | | | 4.0 was too low and copied from Chromium defaults when the filter was initially written, there's no good reason for it to be so low, so double it.
* DOCS/af: update audio filters sectionllyyr2023-09-241-6/+22
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* options: remove --vf-defaults and --af-defaultsDudemanguy2023-09-211-2/+2
| | | | | | | These were deprecated a long time ago and apparently didn't even work with lavfi filters. Go ahead and remove them and additionally clean up some code related to them. m_config_from_obj_desc_and_args becomes much simpler now and a couple of arguments can be completely removed.
* various: remove trailing whitespaceGuido Cella2022-05-141-4/+4
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* DOCS: convert 3rd party http links to httpsLaserEyess2021-08-171-1/+1
| | | | | | | | | | Discovered with: find . -type f \( -name '*.md' -o -name '*.rst' \) -exec grep -n 'http://' {} + All real, i.e. non-example, links found were moved to https. There are some dead links and websites with no https available which were not converted.
* audio: add scaletempo2 filter based on chromiumDorian Rudolph2020-07-271-0/+22
| | | | | | | | scaletempo2 is a new audio filter for playing back audio at modified speed and is based on chromium commit 51ed77e3f37a9a9b80d6d0a8259e84a8ca635259. It sounds subjectively better than the existing implementions scaletempo and rubberband.
* audio: redo video-sync=display-adropwm42020-05-231-0/+7
| | | | | | | | | | | | | | | | | This mode drops or repeats audio data to adapt to video speed, instead of resampling it or such. It was added to deal with SPDIF. The implementation was part of fill_audio_out_buffers() - the entire function is something whose complexity exploded in my face, and which I want to clean up, and this is hopefully a first step. Put it in a filter, and mess with the shitty glue code. It's all sort of roundabout and illogical, but that can be rectified later. The important part is that it works much like the resample or scaletempo filters. For PCM audio, this does not work on samples anymore. This makes it much worse. But for PCM you can use saner mechanisms that sound better. Also, something about PTS tracking is wrong. But not wasting more time on this.
* audio: remove unreferenced af_lavrresamplewm42019-09-191-38/+0
| | | | | | | | | | | | | | 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.
* f_lavfi: add an option to use old audio PTS handling for af_lavfiwm42018-04-151-0/+13
| | | | | The fix-pts option basically uses the old af_lavfi's (before filter rewrite) timestamp logic. The rest is explained in the manpage.
* manpage: clarify some --vf optionswm42018-01-311-1/+2
| | | | In particular, mention deprecated things.
* audio: rewrite filtering glue codewm42018-01-301-4/+3
| | | | Use the new filtering code for audio too.
* af_rubberband: add af-command to multiply current pitchVobe2018-01-151-0/+6
| | | | | | | | | | | | | This commit introduces the multiply-pitch af-command. Users may bind keys to this command in order to incrementally adjust the pitch of a track. This will probably mostly be useful for musicians trying to transpose up and down by semi tones without having to calculate the correct ratio beforehand. As an example, here is an input.conf to test this feature: { af-command all multiply-pitch 0.9438743126816935 } af-command all multiply-pitch 1.059463094352953
* af_lavrresample: deprecate this filterwm42018-01-131-0/+6
| | | | | | The future direction might be not having such a user-visible filter at all, similar to how vf_scale went away (or actually, redirects to libavfilter's vf_scale).
* af: remove deprecated audio filterswm42017-11-291-176/+0
| | | | | | | | | | | | These couldn't be relicensed, and won't survive the LGPL transition. The other existing filters are mostly LGPL (except libaf glue code). This remove the deprecated pan option. I guess it could be restored by inserting a libavfilter filter (if there's one), but for now let it be gone. This temporarily breaks volume control (and things related to it, like replaygain).
* af, vf: improvements to libavfilter bridgewm42017-05-311-0/+3
| | | | | | Add the "lavfi-" prefix (details see manpage additons). Tag the filter name as "(lavfi)" in the verbose filter list output.
* audio: move replaygain control to top-level optionswm42017-04-261-0/+3
| | | | | | | | | | | | | | | | | | | | | af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
* command: add better runtime filter toggling methodwm42017-03-251-23/+4
| | | | | | | | | | Basically, see the example in input.rst. This is better than the "old" vf-toggle method, because it doesn't require the user to duplicate the filter string in mpv.conf and input.conf. Some aspects of this changes are untested, so enjoy your alpha testing.
* af_drc: removeJan Janssen2017-03-251-23/+0
| | | | | | | | | | | | | | | | Remove low quality drc filter. Anyone whishing to have dynamic range compression should use the much more powerful acompressor ffmpeg filter: mpv --af=lavfi=[acompressor] INPUT Or with parameters: mpv --af=lavfi=[acompressor=threshold=-25dB:ratio=3:makeup=8dB] INPUT Refer to https://ffmpeg.org/ffmpeg-filters.html#acompressor for a full list of supported parameters. Signed-off-by: wm4 <wm4@nowhere>
* man: fix PDF buildMartin Herkt2016-10-061-4/+0
| | | | | It really doesn’t want to pagebreak that box, so I removed a useless example.
* af_rubberband: add af-command and option to change the pitchHector Martin2016-09-191-2/+13
| | | | | This allows both fixed and dynamic control over the audio pitch using librubberband, which was previously not exposed to the user.
* af_pan: add af-command support to change the matrixHector Martin2016-09-191-0/+7
| | | | | This allows for seamless changes in the downmixing matrix without having to reinitialize the filter chain.
* audio/filter: remove delay audio filterPaul B Mahol2016-08-121-28/+0
| | | | Similar filter is available in libavfilter.
* af_lavcac3enc: make encoder configurablewm42016-06-231-1/+5
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* manpage: minor correctionswm42016-01-241-7/+3
| | | | Pointed out by der_richter on IRC.
* manpage: fix broken examplewm42015-09-271-1/+1
| | | | Fixes #2353.
* manpage: add a note about libavfilterwm42015-09-111-0/+4
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* af_lavrresample: add normalize suboptionwm42015-09-081-0/+7
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* audio/filter: remove af_bs2b toowm42015-09-041-22/+0
| | | | | | | Some users still use this filter, so the filter was going to be kept. But I overlooked that libavfilter provides this filter. Remove the redundant wrapper from mpv. Something like --af=lavfi=bs2b should work and give exactly the same results.
* audio/filter: remove some useless filterswm42015-09-031-158/+0
| | | | | | | | | | | | | | | | | | | | | | | | All of these filters are considered not useful anymore by us. Some have replacements in libavfilter (useable through af_lavfi). af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub, af_surround, af_sweep: pretty simple and useless filters which probably nobody ever wants. af_ladspa: has a replacement in libavfilter. af_hrtf: the algorithm doesn't work properly on most sources, and the implementation was buggy and complicated. (The filter was inherited from MPlayer; but even in mpv times we had to apply fixes that fixed major issues with added noise.) There is a ladspa filter if you still want to use it. af_export: I'm not even sure what this is supposed to do. Possibly it was meant for GUIs rendering audio visualizations, but it couldn't really work well. For example, the size of the audio depended on the samplerate (fixed number of samples only), and it couldn't retrieve the complete audio, only fragments. If this is really needed for GUIs, mpv should add native visualization, or a proper API for it.
* af_volume: add a replaygain fallback optionwm42015-06-231-0/+4
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* af_convert24: remove this filterwm42015-06-161-7/+0
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* manpage: clarify --af=formatKevin Mitchell2015-04-151-14/+16
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* manpage: remove extra newline at end of af.rstKevin Mitchell2015-04-151-1/+0
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* manpage: correct option minchn->minch for lavcac3encKevin Mitchell2015-03-301-3/+3
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* man: fix PDF buildMartin Herkt2015-03-151-3/+4
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* af_rubberband: make all librubberband options configurablewm42015-02-111-0/+9
| | | | | | | librubberband exports a big load of options. Normally, the default settings (whether they're librubberband defaults or our defaults) should be sufficient, but since I'm not so sure about this, making it configurable allows others to figure it out for me.
* af_rubberband: pitch correction with librubberbandwm42015-02-111-0/+5
| | | | | | | | | If "--af=rubberband" is used, librubberband will be used to speed up or slow down audio with pitch correction. This still has some problems: the audio delay is not calculated correctly, so the audio position jitters around by a few milliseconds. This will probably ruin video timing.
* manpage: fix af_scaletempo suggested commandswm42015-02-101-3/+3
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* audio: drop swapped-endian audio formatswm42014-09-231-3/+2
| | | | | | | | | | | | | | | | | | | | Until now, the audio chain could handle both little endian and big endian formats. This actually doesn't make much sense, since the audio API and the HW will most likely prefer native formats. Or at the very least, it should be trivial for audio drivers to do the byte swapping themselves. From now on, the audio chain contains native-endian formats only. All AOs and some filters are adjusted. af_convertsignendian.c is now wrongly named, but the filter name is adjusted. In some cases, the audio infrastructure was reused on the demuxer side, but that is relatively easy to rectify. This is a quite intrusive and radical change. It's possible that it will break some things (especially if they're obscure or not Linux), so watch out for regressions. It's probably still better to do it the bulldozer way, since slow transition and researching foreign platforms would take a lot of time and effort.
* man: fix a whole bunch of typosMartin Herkt2014-09-011-14/+14
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* af_lavcac3enc: lower minimum channel number to 3wm42014-08-121-1/+1
| | | | It seems only stereo PCM should be passed through.
* af_lavcac3enc: change default bitrate to 640wm42014-08-121-1/+3
| | | | | | | No reason to use less. Since the name "default" is misleading now, replace it with "auto" (still recognize the old name).
* DOCS: remove en/ sub-directorywm42014-06-201-0/+616
This additional sub-directory doesn't serve any purpose anymore. Get rid of it.