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* af: account for queued frames in audio position calculationwm42015-02-121-0/+2
| | | | | | af_rubberband exposed this issue. (cherry picked from commit d85aa35ffbd978c7ae86bf84ebf9ac7686312e8f)
* af: remove old filter compatibility hackwm42015-01-152-42/+1
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* audio/filter: switch remaining filters to refcountingwm42015-01-154-44/+58
| | | | | All of these filters are very similar in frame management, and copy data to a new frame during filtering.
* audio/filter: switch remaining in-place filters to refcountingwm42015-01-159-127/+134
| | | | | | | | | | | | | | | | | | Adds about 7 lines of boilerplate per filter. This could be avoided by providing a different entrypoint (something like af->filter_inplace), which would basically mirror the old interface exactly for this kind of filter. But I feel like it would just be a hack to support all those old, useless filters better. (The ideal solution would be using a language that can do closures to provide a compat. wrapper, but whatever.) af_bs2b has terribly repetitious code for setting up filter functions for each format (most of them useless, in addition to bs2b being useless), so I did something terrible with macros. af_sinesuppress had commented code for float filtering (maybe it was broken; it has been commented every since it was added in 2006). Remove this code.
* 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.
* af_lavcac3enc: use refcounted frameswm42015-01-141-89/+95
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* af_lavfi: use refcounted frameswm42015-01-141-44/+57
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* audio/filter: actually set fmt_in/fmt_out fieldswm42015-01-141-0/+2
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* af_scaletempo: use refcounted frameswm42015-01-141-11/+23
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* af_lavrresample: use refcounted frameswm42015-01-141-23/+46
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* af_convert24: use refcounted frameswm42015-01-131-8/+13
| | | | | This requires allocating a fully new frame. 32->24 could be in-place, but this is not possible for 24->32.
* audio/filters: use refcounted frames for some in-place filterswm42015-01-133-7/+31
| | | | | These are also quite simple, but require requesting write access to the frames. The error handling (for OOM) is a bit annoying.
* audio/filters: use refcounted frames for some simple filterswm42015-01-134-10/+18
| | | | These are read-only, and very trivial to convert.
* af_volume: use refcounted frameswm42015-01-131-8/+15
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* audio: use refcounted frames in the filter chainwm42015-01-132-53/+190
| | | | | | | | | | | | | | | | | | | 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()wm42015-01-136-31/+2
| | | | | | | | | | | | 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.)
* af_volume: dump applied replaygain in verbose modewm42015-01-041-1/+5
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* win32: add mmap() emulationwm42014-12-262-3/+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.
* af_hrtf: Fix out-of-range read.reimar2014-12-061-2/+7
| | | | | | | Based on patch by Yuriy Kaminskiy [yumkam gmail]. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37330 b3059339-0415-0410-9bf9-f77b7e298cf2 Signed-off-by: wm4 <wm4@nowhere>
* audio: make mp_audio_config_to_str return a stack-allocated stringwm42014-11-251-8/+3
| | | | Simpler overall.
* af_scaletempo: use float division for ratewm42014-11-211-1/+1
| | | | | | | | From what I understand the division is to align the dimension of the value from seconds to milliseconds. Hard to tell whether the "rounding" was intentional or not; I'm tipping on "not". Found by Coverity.
* Remove some unneeded NULL checkswm42014-11-211-5/+6
| | | | Found by Coverity; also see commit 85fb2af3.
* af: remove redundant functionwm42014-11-121-9/+2
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* 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 frameswm42014-11-102-10/+38
| | | | | | | | | | | | | | This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of estimating how much audio we need to filter, we always filter full frames. Maybe this should be adjusted later: in case filtering increases the volume of the audio data, we should try not to buffer too much filter output by reducing the input that is fed at once. For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it doesn't seem worth the trouble.
* audio: change how filters are inserted on playback speed changeswm42014-11-103-0/+72
| | | | | | | | | | Use a pseudo-filter when changing speed with resampling, instead of somehow changing a samplerate somewhere. This uses the same underlying mechanism, but is a bit more structured and cleaner. It also makes some of the following changes easier. Since we now always use filters to change audio speed, move most of the work set_playback_speed() does to recreate_audio_filters().
* af_format: remove redundant message prefixeswm42014-11-101-2/+2
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* af_lavcac3enc: fix byte orderwm42014-10-121-2/+2
| | | | | | | | Oops. Fixes #1172. CC: @mpv-player/stable
* audio/filter: allow removing filters by labelwm42014-10-022-1/+33
| | | | | | | | Although the "af" command already could do this, it seems it's better to introduce a lower level mechanism for now. This avoids some messy issues, since that code would recursive call reinit_audio_chain(). To be used by the next commit.
* audio: refactor some aspects of filter chain setupwm42014-10-022-13/+15
| | | | | | | | | | | There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
* audio/filter: don't wipe full filter chain if adding a filter failswm42014-10-021-2/+5
| | | | | There's no need for that, and in fact makes it more likely that it recovers normally.
* audio: cleanup spdif format definitionswm42014-09-232-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before this commit, there was AF_FORMAT_AC3 (the original spdif format, used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS and DTS-HD), which was handled as some sort of superset for AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used IEC61937-framing, but still was handled as something "separate". Technically, all of them are pretty similar, but may use different bitrates. Since digital passthrough pretends to be PCM (just with special headers that wrap digital packets), this is easily detectable by the higher samplerate or higher number of channels, so I don't know why you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs. AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is just a mess. Simplify this by handling all these formats the same way. AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3). All AOs just accept all spdif formats now - whether that works or not is not really clear (seems inconsistent due to earlier attempts to make DTS-HD work). But on the other hand, enabling spdif requires manual user interaction, so it doesn't matter much if initialization fails in slightly less graceful ways if it can't work at all. At a later point, we will support passthrough with ao_pulse. It seems the PulseAudio API wants to know the codec type (or maybe not - feeding it DTS while telling it it's AC3 works), add separate formats for each codecs. While this reminds of the earlier chaos, it's stricter, and most code just uses AF_FORMAT_IS_IEC61937(). Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to include special formats, so that it always describes the fundamental sample format type. This also ensures valid AF formats are never 0 (this was probably broken in one of the earlier commits from today).
* audio: drop swapped-endian audio formatswm42014-09-233-90/+30
| | | | | | | | | | | | | | | | | | | | 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.
* audio: remove swapped-endian spdif formatswm42014-09-231-4/+12
| | | | | | | | | | | | | | | | | | | | | | IEC 61937 frames should always be little endian (little endian 16 bit words). I don't see any apparent need why the audio chain should handle swapped-endian formats. It could be that some audio outputs might want them (especially on big endian architectures). On the other hand, it's not clear how that works on these architectures, and it's not even known whether the current code works on big endian at all. If something should break, and it should turn out that swapped-endian spdif is needed on any platform/AO, swapping still could be done in-place within the affected AO, and there's no need for the additional complexity in the rest of the player. Note that af_lavcac3enc outputs big endian spdif frames for unknown reasons. Normally, the resulting data is just pulled through an auto- inserted conversion filter and turned into little endian. Maybe this was done as a trick so that the code didn't have to byte-swap the actual audio frame. In any case, just make it output little endian frames. All of this is untested, because I have no receiver hardware.
* af_hrtf: initialize coefficient arrayswm42014-09-191-0/+25
| | | | | | | | | | | | | | | Sometimes, --af=hrtf produces heavy artifacts or silence. It's possible that this commit fixes these issues. My theory is that usually, the uninitialized coefficients quickly converge to sane values as more audio is filtered, which would explain why there are often artifacts on init, with normal playback after that. It's also possible that sometimes, the uninitialized values were NaN or inf, so that the artifacts (or silence) would never go away. Fix this by initializing the coefficients to 0. I'm not sure if this is correct, but certainly better than before. See issue #1104.
* af_lavrresample: fix crash with size 0wm42014-09-151-2/+3
| | | | | | | | | | | | | | The filter output size can be 0. Due to how filtering works, this is nothing unusual, but avresample_convert() will return 0. The same case is already handling with "normal" resampling (this commit fixes the reordering code). Additionally, don't use an assert(). avresample_convert() failing is unusual, but might also happen due to e.g. internal out of memory conditions, so we shouldn't just crash on it. Curiously observed with --ao=oss --audio-channels=5.1 when changing speed.
* af_hrtf: request required samplerate, instead of erroring outwm42014-09-051-8/+1
| | | | | | | | It seems hrtf works in 48khz only - and if that wasn't the input, the filter just exited with an error. Make it request the 48khz instead. The player will insert a resampling filter. Not sure why it wasn't done like this in the first place.
* af_hrtf: cosmetics: reindent misaligned code blockwm42014-09-051-8/+8
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* Move compat/ and bstr/ directory contents somewhere elsewm42014-08-291-1/+1
| | | | | | | | | bstr.c doesn't really deserve its own directory, and compat had just a few files, most of which may as well be in osdep. There isn't really any justification for these extra directories, so get rid of them. The compat/libav.h was empty - just delete it. We changed our approach to API compatibility, and will likely not need it anymore.
* af_lavrresample: minor cosmeticswm42014-08-171-4/+2
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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/+2
| | | | | | | No reason to use less. Since the name "default" is misleading now, replace it with "auto" (still recognize the old name).
* Improve setting AVOptionswm42014-08-022-13/+9
| | | | | | | | Use OPT_KEYVALUELIST() for all places where AVOptions are directly set from mpv command line options. This allows escaping values, better diagnostics (also no more "pal"), and somehow reduces code size. Remove the old crappy option parser (av_opts.c).
* audio: remove unused metadata fieldwm42014-07-212-3/+0
| | | | | This was used for replaygain at some point, until replaygain info was passed through explicitly.
* Remove some mp_msg calls with no trailing \nwm42014-07-131-6/+6
| | | | | | | The final goal is all mp_msg calls produce complete lines. We want this because otherwise, race conditions could corrupt the terminal output, and it's inconvenient for the client API too. This commit works towards this goal. There's still code that has this not fixed yet, though.
* af_volume: fix calculations including replay-gainMohammad Alsaleh2014-06-281-2/+2
| | | | | | | | | | | | | rgain is not an additive value. It's a multiplier/gain. Previous behaviour produced negative level values in some cases (when rgain < 1.0) which caused volume to be louder when its value was lowered. CC: @mpv-player/stable Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com> Signed-off-by: wm4 <wm4@nowhere>
* Add more constwm42014-06-1126-60/+60
| | | | | | | While I'm not very fond of "const", it's important for declarations (it decides whether a symbol is emitted in a read-only or read/write section). Fix all these cases, so we have writeable global data only when we really need.
* af_lavcac3enc: detach on any passthrough format, not just ac3wm42014-04-161-1/+1
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* Kill all tabswm42014-04-1321-524/+524
| | | | | | | | | | | I hate tabs. This replaces all tabs in all source files with spaces. The only exception is old-makefile. The replacement was made by running the GNU coreutils "expand" command on every file. Since the replacement was automatic, it's possible that some formatting was destroyed (but perhaps only if it was assuming that the end of a tab does not correspond to aligning the end to multiples of 8 spaces).
* af_volume: fix clang -Wsometimes-uninitializedKevin Mitchell2014-04-131-1/+1
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* af_lavfi: fix graph parse deprecation warningKevin Mitchell2014-04-131-1/+1
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* demux: move metadata-based replaygain decoding out of af_volumeAlessandro Ghedini2014-04-041-80/+9
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* af_volume: use replaygain side dataAlessandro Ghedini2014-04-041-7/+19
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* af: add replaygain_data field to af_stream and af_instanceAlessandro Ghedini2014-04-042-0/+3
| | | | Closes #664
* af_volume: fix replaygainwm42014-03-271-2/+3
| | | | | | | | This was accidentally broken in commit b72ba3f7. I somehow made the wild assumption that replaygain adjusted the volume relative to 0% instead of 100%. The detach suboption was similarly broken.
* af_lavcac3enc: use new AVFrame APIwm42014-03-161-3/+3
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* build: simplify libavfilter configure checkswm42014-03-161-1/+1
| | | | | This is all not needed anymore. In particular, remove all configure switches except --enable-libavfilter.
* Remove some more unneeded version checkswm42014-03-161-13/+3
| | | | | All of these check against things that happened before the latest supported FFmpeg/Libav release.
* af_lavrresample: remove avresample_set_channel_mapping() fallbackswm42014-03-161-24/+0
| | | | | | | This function is now always available. Also remove includes of reorder_ch.h from some AOs (these are just old relicts).
* af_volume: don't print missing replaygain tags as errorwm42014-03-141-1/+1
| | | | There's no reason to. Audio files often lack them.
* af_volume: add detach optionwm42014-03-141-0/+4
| | | | | | Maybe this should be default. On the other hand, this filter does something even if the volume is neutral: it clips samples against the allowed range, should the decoder or a previous filter output garbage.
* af_volume: separate softvol volume control from replaygain levelwm42014-03-141-5/+8
| | | | | | | | | Currently, both replaygain adjustment and user volume control (if softvol is enabled) share the same variable. Sharing the variable would cause especially if --volume is used; then the replaygain volume would always be overwritten. Now both gain values are simple added right before doing filtering.
* af_volume: remove double-negated suboptionwm42014-03-141-3/+3
| | | | | You had to use "no-replaygain-noclip" to set this option. Rename it, so that only one negation is needed.
* af_volume: add support for replaygain pre-amp and clipping preventionAlessandro Ghedini2014-03-131-11/+74
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* af_volume: add replaygain supportAlessandro Ghedini2014-03-131-0/+22
| | | | | | | | | This adds the options replaygain-track and replaygain-album. If either is set, the replaygain track or album gain will be automatically read from the track metadata and the volume adjusted accordingly. This only supports reading REPLAYGAIN_(TRACK|ALBUM)_GAIN tags. Other formats like LAME's info header would probably require support from libav.
*