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* mpv_talloc.h: rename from talloc.hDmitrij D. Czarkoff2016-01-111-1/+1
| | | | This change helps avoiding conflict with talloc.h from libtalloc.
* audio: fix format function consistency issueswm42015-06-261-3/+3
| | | | | | | | | | | 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".
* Update license headersMarcin Kurczewski2015-04-131-6/+5
| | | | Signed-off-by: wm4 <wm4@nowhere>
* ao_pcm: add append modewm42015-01-141-1/+3
| | | | | Pretty useful for debugging, although a bit useless or possibly misleading too (see comments in the manpage).
* ao_pcm: simplifywm42014-11-211-17/+13
| | | | Also shuts up Coverity.
* audio: cleanup spdif format definitionswm42014-09-231-2/+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 formatswm42014-09-231-7/+14
| | | | | | | | | | | | | | | | | | | | 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 formatswm42014-09-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | 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.
* ao_pcm: minor simplificationwm42014-09-061-1/+1
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* ao_pcm: fix message stringsAmos Onn2014-06-151-2/+2
| | | | Signed-off-by: wm4 <wm4@nowhere>
* af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriateMarcoen Hirschberg2014-05-281-1/+1
| | | | | | In most places where af_fmt2bits is called to get the bits/sample, the result is immediately converted to bytes/sample. Avoid this by getting bytes/sample directly by introducing af_fmt2bps.
* af_lavrresample: remove avresample_set_channel_mapping() fallbackswm42014-03-161-1/+0
| | | | | | | This function is now always available. Also remove includes of reorder_ch.h from some AOs (these are just old relicts).
* audio/out: make draining a separate operationwm42014-03-091-1/+1
| | | | | | | | | | | | Until now, this was always conflated with uninit. This was ugly, and also many AOs emulated this manually (or just ignored it). Make draining an explicit operation, so AOs which support it can provide it, and for all others generic code will emulate it. For ao_wasapi, we keep it simple and basically disable the internal draining implementation (maybe it should be restored later). Tested on Linux only.
* audio/out: make ao struct opaquewm42014-03-091-0/+1
| | | | | | We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-1/+1
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* Move options/config related files from mpvcore/ to options/wm42013-12-171-1/+1
| | | | | | | | | Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
* audio/out: prepare for non-interleaved audiowm42013-11-121-3/+4
| | | | | | | | | | | | | | | | | | | This comes with two internal AO API changes: 1. ao_driver.play now can take non-interleaved audio. For this purpose, the data pointer is changed to void **data, where data[0] corresponds to the pointer in the old API. Also, the len argument as well as the return value are now in samples, not bytes. "Sample" in this context means the unit of the smallest possible audio frame, i.e. sample_size * channels. 2. ao_driver.get_space now returns samples instead of bytes. (Similar to the play function.) Change all AOs to use the new API. The AO API as exposed to the rest of the player still uses the old API. It's emulated in ao.c. This is purely to split the commits changing all AOs and the commits adding actual support for outputting N-I audio.
* audio/out: reject non-interleaved formatswm42013-11-121-0/+3
| | | | | | | | | | No AO can handle these, so it would be a problem if they get added later, and non-interleaved formats get accepted erroneously. Let them gracefully fall back to other formats. Most AOs actually would fall back, but to an unrelated formats. This is covered by this commit too, and if possible they should pick the interleaved variant if a non-interleaved format is requested.
* audio: replace af_fmt2str_short -> af_fmt_to_strwm42013-11-071-1/+1
| | | | Also, remove all af_fmt2str usages.
* audio/out: remove useless info struct and redundant fieldswm42013-10-231-6/+4
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* ao_pcm: big endian AC3 in wav doesn't workwm42013-10-221-1/+0
| | | | | | | | At least not with ffmpeg. Honestly, I have no idea how little endian AC3 works at all, since ao_pcm doesn't do anything special about it, and treats it like s16le. Maybe it's broken and ffmpeg has special logic to detect it.
* audio/out: do some mp_msg conversionswm42013-08-221-10/+6
| | | | | | | Use the new MP_ macros for some AOs instead of mp_msg. Not all AOs are converted, and some only partially. In some cases, some additional cosmetic changes are made.
* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-061-2/+2
| | | | Followup commit. Fixes all the files references.
* audio/out: remove options argument from init()wm42013-07-221-1/+1
| | | | Same as with VOs in the previous commit.
* ao_pcm: use new option APIwm42013-07-211-16/+12
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* audio/out: remove ao->outburst/buffersize fieldswm42013-06-161-2/+1
| | | | | | | | | | | | | | | The core didn't use these fields, and use of them was inconsistent accross AOs. Some didn't use them at all. Some only set them; the values were completely unused by the core. Some made full use of them. Remove these fields. In places where they are still needed, make them private AO state. Remove the --abs option. It set the buffer size for ao_oss and ao_dsound (being ignored by all other AOs), and was already marked as obsolete. If it turns out that it's still needed for ao_oss or ao_dsound, their default buffer sizes could be adjusted, and if even that doesn't help, AO suboptions could be added in these cases.
* audio/out: channel map selectionwm42013-05-121-1/+4
| | | | | | | | | Make all AOs use what has been introduced in the previous commit. Note that even AOs which can handle all possible layouts (like ao_null) use the new functions. This might be important if in the future ao_select_champ() possibly honors global user options about downmixing and so on.
* ao: remove ao_driver.is_new fieldwm42013-05-121-1/+0
| | | | Is unused, is completely pointless.
* audio/out: switch to channel mapwm42013-05-121-26/+8
| | | | | | This actually breaks audio for 5/6/8 channels. There's no reordering done yet. The actual reordering will be done inside of af_lavrresample and has to be made part of the format negotiation.
* ao_pcm: fix references to -novideowm42012-12-031-8/+1
| | | | | | | | | | | The option is -no-video. Remove the deprecated "fast" suboption, which did nothing and instructed the user to use "-novideo" instead. Fix a reference to -novideo in encoding.rst. Add a "generic" entry about -no-* to the list of renamed options. The change is already explicitly mentioned in the text above the table, but even if it's redundant, it makes it harder to overlook.
* Rename directories, move files (step 2 of 2)wm42012-11-121-5/+5
| | | | | | | | | | | | Finish renaming directories and moving files. Adjust all include statements to make the previous commit compile. The two commits are separate, because git is bad at tracking renames and content changes at the same time. Also take this as an opportunity to remove the separation between "common" and "mplayer" sources in the Makefile. ("common" used to be shared between mplayer and mencoder.)
* Rename directories, move files (step 1 of 2) (does not compile)wm42012-11-121-0/+256
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
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