summaryrefslogtreecommitdiffstats
path: root/audio/filter/af.c
Commit message (Collapse)AuthorAgeFilesLines
* audio: rewrite filtering glue codewm42018-01-301-824/+0
| | | | Use the new filtering code for audio too.
* af_lavrresample: deprecate this filterwm42018-01-131-2/+4
| | | | | | 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).
* audio: add global options for resampler defaultswm42018-01-131-1/+2
| | | | | | | | This is part of trying to get rid of --af-defaults, and the af resample filter. It requires a complicated mechanism to set the defaults on the resample filter for backwards compatibility.
* audio: add audio softvol processing to AOwm42017-11-291-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
* af: remove deprecated audio filterswm42017-11-291-9/+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).
* audio: move libswresample wrapper out of audio filter codewm42017-09-211-0/+2
| | | | | | | | | Move it from af_lavrresample.c to a new aconverter.c file, which is independent from the filter chain code. It also doesn't use mp_audio, and thus has no GPL dependencies. Preparation for later commits. Not particularly well tested, so have fun.
* af, vf: improvements to libavfilter bridgewm42017-05-311-1/+7
| | | | | | Add the "lavfi-" prefix (details see manpage additons). Tag the filter name as "(lavfi)" in the verbose filter list output.
* af: implement generic lavfi option bridge toowm42017-04-041-2/+25
| | | | | | Literally copy-pasted from the same commit for video filters. (Once new code for filters is implemented, this will all go away or at least get unified anyway.)
* command: add better runtime filter toggling methodwm42017-03-251-0/+3
| | | | | | | | | | 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 Janssen2017-03-251-2/+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>
* m_config: add helper function for initializing af/ao/vf/vo suboptionswm42016-09-021-4/+4
| | | | | | | | Normally I'd prefer a bunch of smaller functions with fewer parameters over a single function with a lot of parameters. But future changes will require messing with the parameters in a slightly more complex way, so a combined function will be needed anyway. The now-unused "global" parameter is required for later as well.
* audio: improve aspects of EOF handlingwm42016-08-181-0/+6
| | | | | | | | | | | The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
* audio/filter: remove delay audio filterPaul B Mahol2016-08-121-2/+0
| | | | Similar filter is available in libavfilter.
* af: avoid rebuilding filter chain in another minor casewm42016-07-151-0/+3
| | | | | No need to create additional noise of we can trivially see that rebuiding the chain won't change anything.
* audio: fix code for adjusting conversion filterswm42016-07-111-4/+5
| | | | | | | | | | | | | | | This code was supposed to adjust existing conversion filters (to make them output a different format). But the code was just broken, apparently a refactoring accident. It accessed af instead of af->prev. The bug tended to add new conversion filters, even if an existing one could have been used. (Can be tested by inserting a dummy lavrresample filter followed by a format filter which forces conversion.) In addition, it's probably better to return the actual error code if reinitializing the filter fails. It would then respect an AF_FALSE return value, which means format negotiation failed, instead of a generic error.
* audio: add heuristic to move auto-downmixing before other filterswm42016-07-101-7/+66
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Normally, you want downmixing to happen first thing in the filter chain. This is reflected in codec downmixing, which feeds the filter chain downmixed audio in the first place. Doing this has the advantage of needing less data to process. But the main motivation is that if there is a drc filter in the chain, you want to process it the downmixed audio. Add an idiotic heuristic to achieve this. It tries to detect whether the audio was indeed automatically downmixed (or upmixed). To detect what the output format is going to be, it builds the filter chain normally, and then retries with the heuristic applied (and for extra paranoia, retries without the heuristic again if it fails to successfully rebuild the filter chain for unknown reasons). This is simple and will work in almost all cases. Doing it in a more complete way is rather hard, because filters are so generic. For example, we know absolutely nothing about the behavior of af_lavfi, which creates an opaque filter graph with libavfilter. We don't know why a filter would e.g. change the channel layout on its output. (Our heuristic bails out in this case.) We're also slave to the lowest common denominator of how our format negotiation works, and how libavfilter's works. In theory, we could make this mechanism explicit by introducing a special dummy filter. The filter chain would then try to convert between input and output formats at the dummy filter, which would give the user more control over how downmix happens. On the other hand, the user could just insert explicit conversion filters instead, so this would probably have questionable value.
* audio: add auto-inserted flag to filter list loggingwm42016-07-101-0/+2
| | | | Like the video filter chain.
* audio: cleanup audio filter format negotiationwm42016-07-101-123/+62
| | | | | | | | | | | | | | | | | | The algorithm and functionality is the same, but the code becomes much simpler and easier to follow. The assumption that there is only 1 conversion filter (lavrresample) helps with the simplification, but the main change is to use the same code for format/channels/rate. Get rid of the different AF_CONTROL_SET_* controls, and change the af->data parameters directly. (af->data is badly named, but essentially is a placeholder for the output format.) Also, instead of trying to use the af_reinit() loop to init inserted conversion filters or filters with changed output formats, do it inline, and move the common code to a filter_reinit() function. This gets rid of the awful retry variable. In general, this should not change any runtime behavior.
* audio: insert audio-inserted filters at end of chainwm42016-07-091-34/+1
| | | | | | This happens to be better for the af_volume filter (for softvol), and saves some code too. It's "better" because you want to affect the final filtered audio, such as after a manually inserted drc filter.
* vf, af: print filter labels in verbose modewm42016-07-061-0/+2
|
* Fix misspellingsstepshal2016-06-261-1/+1
|
* build: make libavfilter mandatorywm42016-02-051-2/+0
| | | | | | The complex filter support that will be added makes much more complex use of libavfilter, and I'm not going to bother with adding hacks to keep libavfilter optional.
* command: add af-command commandwm42016-01-221-0/+11
| | | | Similar to vf-command. Requested. Untested.
* audio: change downmix behavior, add --audio-normalize-downmixwm42016-01-201-0/+1
| | | | | | This is probably the 3rd time the user-visible behavior changes. This time, switch back because not normalizing seems to be the more expected behavior from users.
* af: prevent endless loop when removing filters due to spdifwm42015-10-261-1/+2
| | | | | | | | | | | | | | | | | | | | | This code removes filters which can not take spdif inout. This was made so that PCM filters are transparently dropped in spdif mode. This entered an endless loop with: --af=lavcac3enc:::2 --audio-channels=5.1 The forced number of output channels is incompatible with spdif. It's trying to insert af_lavrresample as conversion filter to compensate for it. Of course this doesn't work, which triggers the PCM filter removal. Then it goes on normally - since the new state is exactly as before, it will try the same thing again, forever. Fix by reusing the retry counter, which is a very dumb but very effective measure against these cases of filter negotiation failure. We could try to be more clever (for example, if the removed filter is a conversion filter, we can be sure this won't work, and error out immediately). But better keep it simple and robust.
* audio/filter: remove reentrancy flagwm42015-09-201-14/+1
| | | | | | | | | | This flag was used by some filters and made sure none of these filters were inserted twice. This triggers only if the user explicitly tries to add multiple filters (and not e.g. due to auto-insertion), so at best this warned the user from doing something potentially pointless. At worst, it blocked some (mildly) legitimate use-cases. Get rid of it. Also see #2322.
* af_lavfi: implement af-metadata propertywm42015-09-111-0/+12
| | | | | | | Works like vf-metadata. Unfortunately requires some code duplication (even though it's not much). Fixes #2311.
* audio/filter: remove af_bs2b toowm42015-09-041-3/+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 filterswm42015-09-031-23/+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.
* audio: remove af_dummywm42015-08-011-2/+0
| | | | Was used internally once; has no function anymore.
* af: fix behavior with pathologic filter chainswm42015-07-071-0/+2
| | | | | | | | Some filter chains require a huge number of auto-inserted conversion filters. There is an overly stupid safeguard against infinite filter insertions, which counts the number of conversion filters inserted. This triggered accidentally in this case. Fix by resetting this counter after a non-conversion filter was successfully configured.
* audio: fix format function consistency issueswm42015-06-261-2/+2
| | | | | | | | | | | Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
* af: restore detaching of PCM filters when using spdifwm42015-06-221-1/+6
| | | | | | Basically, af_fix_format_conversion() behaves stupid you insert a conversion filter that won't work, and adding back the conversion test function is the simplest fix to it.
* Various spelling fixesMarcin Kurczewski2015-06-181-1/+1
| | | | Signed-off-by: wm4 <wm4@nowhere>
* af: remove conversion filter searchwm42015-06-161-86/+4
| | | | | | This attempted to find a minimal filter graph for a format conversion involving multiple conversion filters. With the last 2 commits it becomes dead code - remove it.
* af_convert24: remove this filterwm42015-06-161-3/+0
|
* audio: remove S8, U16, U24, U32 formatswm42015-06-161-2/+0
| | | | | | | | | | | | | They are useless. Not only are they actually rarely in use; but libavcodec doesn't even output them, as libavcodec has no such sample formats for decoded audio. Even if it should happen that we actually still need them (e.g. if doing direct hardware output), there are better solutions. Swapping the sign is a fast and lossless operation and can be done inplace, so AO actually needing it could do this directly. If you wonder why we keep U8 instead of S8: because libavcodec does it.
* af: fix an aspect of filter chain flushingwm42015-06-151-0/+11
| | | | | Even if we flush the current filter, we have to read the remaining output from the frame we previously fed to the filter.
* af: don't attempt to remove last filter for spdif filter removalwm42015-05-051-1/+1
| | | | | | | | | | | | | | | Some time ago, a mechanism was added for automatically removing PCM-only filters if the input format is spdif. This could cause an infinite loop if the AO did not support spdif, but was falling back to some PCM format. Then this code tried to remove the last filter, which is a dummy filter for receiving and queuing filter output. af_remove() simply fails gracefully in this case, so this happens over and over again. Fix by explicitly checking whether the filter to remove is a dummy filter. (af_remove() also fails only if the dummy filters are attempted to be removed - checking this directly is simpler.)
* Update license headersMarcin Kurczewski2015-04-131-5/+4
| | | | Signed-off-by: wm4 <wm4@nowhere>
* audio/filter: fully renegotiate audio formats on every reconfigwm42015-04-121-0/+10
| | | | | | | | | | | | | | | | | | | It could happen that a lavrresample filter would keep its old output format when the decoder changed its output format. This simply happened because the output format was never reset. Normally, this was not an issue, because lavrresample filters only inserted for format conversion were removed on format changes. But if --no-audio-pitch-correction is set and playback speed is changed, then there is a "permanent" lavrresample filter in the filter chain, which shows this behavior. Fix by explicitly resetting output formats for all filters which support it. Note: this can crash with libswresample in some cases. I'm not sure if this is mpv's fault or libswresample's, but since it works with libavresample, I'm going to assume it's not our's.
* audio: automatically deatch filters if spdif prevents their usewm42015-04-071-0/+17
| | | | Fixes #1743 and partially #1780.
* audio: change a detail about filter insertionwm42015-04-071-7/+10
| | | | | | | | | | The af_add() function has a problem: if the inserted filter returns AF_DETACH during init, the function will have a dangling pointer. Until now this was avoided by making sure none of the used filters actually return AF_DETACH, but it's getting infeasible. Solve this by requiring passing an unique label to af_add(), which is then used instead of the pointer.
* audio: change playback speed directly in resamplerwm42015-03-021-2/+0
| | | | | | | | | | | | | Although the libraries we use for resampling (libavresample and libswresample) do not support changing sampelrate on the fly, this makes it easier to make sure no audio buffers are implicitly dropped. In fact, this commit adds additional code to drain the resampler explicitly. Changing speed twice without feeding audio in-between made it crash with libavresample inc ertain cases (libswresample is fine). This is probably a libavresample bug. Hopefully this will be fixed, and also I attempted to workaround the situation that crashes it. (It seems to point in direction of random memory corruption, though.)
* af: account for queued frames in audio position calculationwm42015-02-111-0/+2
| | | | af_rubberband exposed this issue.
* af_rubberband: pitch correction with librubberbandwm42015-02-111-0/+4
| | | | | | | | | 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.
* af: remove old filter compatibility hackwm42015-01-151-37/+1
|
* af: verify filter input formatswm42015-01-151-1/+4
| | | | | | | | | | | Just to make sure all filters get the correct format. Together wih the check in af_add_output_frame(), this asserts that af->prev->fmt_out == af->fmt_in This also requires setting the "in" pseudo-filter (s->first) formats correctly. Before this commit, the fmt_in/fmt_out fields weren't used for this filter.
* audio/filter: actually set fmt_in/fmt_out fieldswm42015-01-141-0/+2
|
* audio: use refcounted frames in the filter chainwm42015-01-131-49/+166
| | | | | | | | | | | | | | | | | | | The goal is switching the whole audio chain to using refcounted frames. This brings the architecture closer to FFmpeg, enables better integration with libavfilter, will reduce useless copying somewhat, and will probably allow better timestamp tracking. For now, every filter goes through a semi-awful wrapper in af_do_filter(), though. This will be fixed step by step, and the wrapper should eventually be removed. Another thing that will have to be done is improving the timestamp handling and avoiding extra copies for the AO. Some of the new code is rather similar to the video filter code (the core filter code basically just has types replaced). Such code duplication is normally very unwanted, but in this case there's probably no other choice. On the other hand, this code is pretty simple (even if somewhat tricky). Maybe there will be unified filter code in the future, but this is still far away.
* audio/filter: remove unused af_calc_filter_multiplier()wm42015-01-131-17/+0
| | | | | | | | | | | | The purpose of this function was to filter only as much audio input as needed to produce a certain amount of audio output. This could (in theory) avoid excessive buffering when e.g. changing playback speed with resampling. Use of this was already removed in commit 5fd8a1e0. No problems were experienced, so let's assume this feature is practically worthless. (Though it's possible that it was quite useful over a decade ago, or in some cornercases with evil files.)
* win32: add mmap() emulationwm42014-12-261-2/+0
| | | | | | | | Makes all of overlay_add work on windows/mingw. Since we now don't explicitly check for mmap() anymore (it's always present), this also requires us to make af_export.c compile, but I haven't tested it.
* audio: make mp_audio_config_to_str return a stack-allocated stringwm42014-11-251-8/+3
| | | | Simpler overall.
* af: remove redundant functionwm42014-11-121-9/+2
|
* af: check audio params for validitywm42014-11-121-0/+5
| | | | Normally, these should be valid anyway, so this is just being cautious.
* audio: make decoders output refcounted frameswm42014-11-101-9/+35
| | | | | | | | | | | | | | This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". R