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* options: change option macros and all option declarationswm42020-03-181-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Change all OPT_* macros such that they don't define the entire m_option initializer, and instead expand only to a part of it, which sets certain fields. This requires changing almost every option declaration, because they all use these macros. A declaration now always starts with {"name", ... followed by designated initializers only (possibly wrapped in macros). The OPT_* macros now initialize the .offset and .type fields only, sometimes also .priv and others. I think this change makes the option macros less tricky. The old code had to stuff everything into macro arguments (and attempted to allow setting arbitrary fields by letting the user pass designated initializers in the vararg parts). Some of this was made messy due to C99 and C11 not allowing 0-sized varargs with ',' removal. It's also possible that this change is pointless, other than cosmetic preferences. Not too happy about some things. For example, the OPT_CHOICE() indentation I applied looks a bit ugly. Much of this change was done with regex search&replace, but some places required manual editing. In particular, code in "obscure" areas (which I didn't include in compilation) might be broken now. In wayland_common.c the author of some option declarations confused the flags parameter with the default value (though the default value was also properly set below). I fixed this with this change.
* ao_alsa: use AO underrun reportingwm42019-10-111-1/+3
| | | | This enables the change introduced in the previous commit for ao_alsa.
* ao_alsa: handle underruns in get_space() toowm42019-10-111-0/+2
| | | | | This is essentially optional. But it will give the higher level code a better guarantee that underruns were tested.
* ao_alsa: mess with underrun handling againwm42019-10-111-6/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit tries to prepare for better underrun reporting. The goal is to report underruns relatively immediately. Until now, this happened only when play() was called. Change this, and abuse that get_delay() is called "relatively often" - this reports the underrun immediately in practice. Background: In commit 81e51a15f7e1 (and also e38b0b245ed4), we were quite confused about ALSA underrun handling. The commit message showed uncertainty how case 3 happened, but it's blindingly obvious and simple. Actually reading the code shows that ALSA does not have a concept of a "final chunk" (or we don't use it). It's obvious we never pass the AOPLAY_FINAL_CHUNK flag along to the ALSA API in any way. The only thing we do is simply writing a partial fragment. Of course this will cause an underrun. Doing a partial write saves us the trouble to pad the last frame with silence, or so. The main reason why the underrun message was avoided was that play() was never called with a non-0 sample count again (except if reset() was called before that). That was OK, at least the goal of avoiding the unwanted message was reached. (And the original "bogus" message at end of playback was perfectly correct, as far as ALSA goes.) If network stalls, play() will called again only once new data is available. Obviously, this could take a long time, thus it's too late.
* ao_alsa: don't silence legitimate underrun if final chunk underrunswm42019-10-061-4/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | It turns out that case 2) mentioned in the previous commit happened quite often when playback ended normally. There is probably a legitimate underrun with normal buffer sizes (100 ms, 4 fragments, gapless audio in "weak" mode). This is a result of the player waiting for video to end, and/or the time needed to kill the video window. The former case means that it depends on your test case whether it happens (a file where video ends slightly before audio is less likely to trigger it). This in turn is due to how gapless playback works. Achieving not having a "gap" requires queuing the audio of the next file without playing a partial chunk (as AOPLAY_FINAL_CHUNK would do). The partial chunk is then played as part of the first chunk played from the next file. But if it detects "later" that there is no next file, it still needs to get rid of the last fragment with AOPLAY_FINAL_CHUNK. At this point it's too late, and an underrun may have actually happened. The way the player uninits and reinits the entire playback engine for the next file in a "serial" manner means it cannot know in advance whether this works. This is the reason why the idiot who added the underrun exception for the last chunk in play() was wrong (I wrote that btw., before you accuse me of being rude). Yes, it's a real underrun, and you could probably hear it.
* ao_alsa: remove sometimes bogus XRUN messagewm42019-10-061-9/+2
| | | | | | | | | | | | | | | | | | | This XRUN (aka underrun) message was printed in the following situations: 1) legitimate underrun during playback 2) legitimate underrun when playing final chunk 3) bogus underrun when playing final chunk The old underrun case (in play()) happens in cases 1) and 2) as well, but 3) did not happen. It appears 3) is indeed something that happens, although it's not known for sure. It's still pretty annoying, so remove the new XRUN message. When testing, care should be taken to play with buffer sizes, video versus no video, and gapless enabled/disabled. Also, suspending the player with Ctrl+Z in the terminal (SIGSTOP) and then resuming is a good way to trigger a "normal" underrun.
* ao_alsa: log the ALSA state if we get a non-XRUN errorJan Ekström2018-09-291-2/+4
| | | | | The ALSA state generally can tell us more information in case we get an unexpected error.
* ao_alsa: handle XRUNs separately from other errorsJan Ekström2018-09-291-2/+7
| | | | | | | | | According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN, and that is a state that we should attempt to automatically recover from. In case recovery fails, log an error and return zero. A warning message will still be output for each XRUN since those are not something we should generally be receiving.
* ao_alsa: early exit get_space if paused or ALSA is not readyJan Ekström2018-09-291-0/+5
| | | | | | | | | | | This has been way too long coming, and for me to notice that a whole lot of ao_alsa functions do an early return if the AO is paused. For the STATE_SETUP case, I had this reproduced once, and never since. Still, seems like we can start calling this function before the ALSA device has been fully initialized so we might as well early exit in that case.
* ao_alsa: simplify get_space()Jan Ekström2018-06-041-6/+10
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* ao_alsa: replace snd_pcm_status() with snd_pcm_avail() in get_space()Muhammad Faiz2018-06-041-5/+4
| | | | | | | | | | Fixes a bug with alsa dmix on Fedora 29. After several minutes, audio suddenly becomes bad and muted. Actually, I don't know what causes this. Probably this is a bug in alsa. In any case, as snd_pcm_status() returns not only 'avail', but also other fields such as tstamp, htstamp, etc, this could be considered a good simplification, as only avail is required for this function.
* ao_alsa: actually report underruns to userwm42018-04-151-5/+5
| | | | | | | | | Print them as a warning. Note that there may be some cases where it underruns, without being a bad condition. This could possibly happen e.g. if the last chunk is written, and then it resumes playback some time after that. Eventually I want to add more code to avoid such spurious warnings.
* ao_alsa: reduce requested buffer sizewm42018-04-151-2/+2
| | | | | | There is a dedicated thread for feeding audio to the ALSA API from a buffer with a larger size. There is little reason to have such a large device buffer.
* ao_alsa: add options for controlling period/buffer sizewm42018-04-151-8/+16
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* ao_alsa: reduce verbosity at -vwm42018-01-251-10/+10
| | | | | | | Always make the hw params dump function use MSGL_DEBUG, and remove the MSGL_V use. That means you need -v -v to see them. The detailed information is usually not very interesting, so this reduces the log noise.
* audio: fix annyoing af_get_best_sample_formats() definitionwm42018-01-251-1/+1
| | | | | | | | | | | | | | | | The af_get_best_sample_formats() function had an argument of int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0 terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't actually write this many formats (since some formats are fundamentally incompatible), but it still feels annoying and incorrect. So fix it, and require that callers pass an AF_FORMAT_COUNT+1 array. Note that the array size has no meaning in C function arguments (just another issue with C static arrays being weird and stupid), so get rid of it completely. Not changing the af_lavcac3enc use, since that is rewritten in another branch anyway.
* Fix undefined preprocessor behaviorwm42018-01-181-2/+5
| | | | | | | | | | | This commit eliminates the following clang warning: warning: macro expansion producing 'defined' has undefined behavior [-Wexpansion-to-defined] Going by the clang commit message, this seems to be explicitly specified as UB by the standard, and they added this warning because MSVC apparently results in different behavior. Whatever, we can just avoid the warning with some small changes.
* ao_alsa: change license to LGPLwm42017-11-231-27/+15
| | | | | | | | | | | | | | | | | | | | | | | | Looks like this is covered by LGPL relicensing agreements now. Notes about contributors who could not be reached or who didn't agree: Commit 7fccb6486e has tons of mp_msg changes look like they are not copyrightable (even if they were, all mp_msg calls were rewritten in mpv times again). The additional play() change looks suspicious, but the function was rewritten several times anyway (first time after that commit in 4f40ec312). Commit 89ed1748ae was rewritten in commit 325311af3 and then again several times after that. Basically all this code is unnecessary in modern mpv and has been removed. No code survived from the following commits: 4d31c3c53, 61ecf838f2, d38968bd, 4deb67c3f. At least two cosmetic typo fixes are not considered as well. Commit 22bb046ad is reverted (this wasn't a valid warning anyway, just a C++-ism icc applied to C). Using the constants is nicer, but at least I don't have to decide whether that change was copyrightable.
* ao_alsa: don't convert twice on retrywm42017-11-231-2/+1
| | | | Obscure corner case.
* Add checks for HAVE_GPL to various GPL-only source fileswm42017-10-101-0/+4
| | | | | | | | This should actually cover all of them, if you take into account that some unchanged GPL source files include header files with such checks. Also this was done already for the libaf derived code. This is only for "safety" and to avoid misunderstandings.
* ao_alsa: drop use of AF_FORMAT_S24wm42017-07-071-21/+56
| | | | | | | | | | | | | Instead of the infrastructure added in the previous commit to do the conversion within the AO. If this is used, and snd_pcm_status_get_avail() returns more frames than snd_pcm_write*() actually accepts, you will get some nice audio corruption. Also, this mutates the data passed via play(), which is rather fishy, but sort of doesn't matter for now. Surely this will cause unintended bugs and WTFs.
* audio/out: require AO drivers to report period size and correct bufferwm42017-06-251-0/+1
| | | | | | | | | | | | | | | | Before this change, AOs could have internal alignment, and play() would not consume the trailing data if the size passed to it is not aligned. Change this to require AOs to report their alignment (via period_size), and make sure to always send aligned data. The buffer reported by get_space() now always has to be correct and reliable. If play() does not consume all data provided (which is bounded by get_space()), an error is printed. This is preparation for potential further AO changes. I casually checked alsa/lavc/null/pcm, the other AOs might or might not work.
* options: remove remaining deprecated audio device selection optionswm42017-04-231-4/+0
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* ao_alsa: fix device filtering, add another exceptionwm42017-03-141-1/+3
| | | | | | | The "return false;" was debugging code. In addition, filter a plain "default", because it's not going to do anything interesting and just looks ugly.
* ao_alsa: filter fewer deviceswm42017-03-141-4/+2
| | | | | | | | | | | | It appears some device can be missing if we filter too many. In particular, I've seen devices starting with "front" and "sysdefault" being mapped to different hardware. I conclude that it's not sane trying to present a nice device list to users in ALSA. It's fucked. (Although kodi appears to attempt some intense "beautification" of the device list, which includes parsing parameters from the device name and such. Well, let's not.) No other audio API requires such ridiculous acrobatics.
* ao_alsa: POLLERR can be set even if the device is not lostwm42017-03-141-1/+5
| | | | | | | | | | Apparently POLLERR can be set if poll is called while the device is in the SND_PCM_STATE_PREPARED state. So assume that we can simply call snd_pcm_status() to check whether the error is because the device went away (i.e. we expect it to return ENODEV if this happened). This avoids sporadic device lost warnings and AO reloads. The actual device lost case is untested.
* ao_alsa: close audio device if polling returns POLLERRwm42017-02-271-1/+3
| | | | | | | | This is apparently what happens in this situation: Turn off display with DPMS, turn back on with DPMS. MPV is hung. See #4189.
* ao_alsa: fix an error checkwm42017-02-271-1/+1
| | | | Fixes #4188 as pointed out in the issue.
* ao_alsa: print certain ALSA errors as string instead as numberwm42016-12-071-2/+2
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* options: remove deprecated sub-option handling for --vo and --aowm42016-11-251-10/+0
| | | | | | | | Long planned. Leads to some sanity. There still are some rather gross things. Especially g_groups is ugly, and a hack that can hopefully be removed. (There is a plan for it, but whether it's implemented depends on how much energy is left.)
* audio/out/push: play silence on --audio-stream-silencewm42016-11-241-6/+0
| | | | | | | | | | | | | | | | | | | Until now, this was only implemented for ao_alsa and AOs not using push.c. ao_alsa.c relied on enabling funny underrun semantics for avoiding resets on lower levels, while other AOs using push.c didn't do anything. Change this and at least make push.c copy silent data to the AO. This still isn't perfect as keeping track of how much silence was played when seems complex, so we don't do it. The consequence is that frame-stepping will essentially randomize the A/V offset (it'll recover immediately when unpausing, but still ugly). Also, in order to empty the currently buffered audio on seeks etc., we still call ao_driver->reset and so on, so the AO driver will still need to handle this specially. The intent is to make behavior with ALSA less weird (for one we can remove the code in ao_alsa.c that tries to trigger an initial underflow). Also might help with #3754.
* ao_alsa: explicitly add default device manuallywm42016-11-141-1/+4
| | | | | | | | | | | | | | The "default" entry (which is and always was mpv/mplayer's default) does not have a description set in the ALSA API. (While "sysdefault" strangely has.) Instead of an empty description, this should show something nice, so reuse the ao.c code for naming default devices (see previous commit). It's still a bit ugly that audio-device-list will have a default entry for "Autoselect device" and "Default (alsa)", but then again we probably want to allow the user to force ALSA (i.e. prevent fallbacks to other AOs) just because ALSA is so flaky and makes this a legitimate feature.
* ao_alsa: fill unused ALSA channels with silencewm42016-11-081-0/+5
| | | | | | | | | | This happens when ALSA gives us more channels than we asked for, for whatever reasons. It looks like this wasn't handled correctly. The mpv and ALSA channel counts could mismatch, which would lead to UB. I couldn't actually trigger this case, though. I'm fairly sure that drivers or plugins exist that do it anyway. (Inofficial ALSA motto: if it can be broken, then why not break it?)
* ao_alsa: strictly disable chmap use for mono/stereowm42016-11-081-12/+21
| | | | | | | | | | If the input is already mono or stereo, or if channel map selection results in mono or stereo, then disable further use of the champ ALSA API (or rather, stop trusting its results). Then we behave like a simple application that only wants to output mono or stereo. See #3045 and #2905. I couldn't actually test these cases, but this commit is supposed to fix them.
* ao_alsa: _really_ disable chmap API use in cases where we shouldwm42016-11-081-3/+7
| | | | | | | | | | | set_chmap() skipped _setting_ the ALSA chmap if chmap use was requested to be disabled by setting dev_chmap.num=0 by the caller, but it still queried the current ALSA channel map. We don't trust it that much, so disable that as well. But we still query and log it, because that could be helpful for debugging. Otherwise we could skip the entire set_chmap() call in these cases.
* ao_alsa: slightly better debug loggingwm42016-11-081-6/+12
| | | | | | | Try to make it more compact, and also always list the reordered layout, but only if it's actually different. Should be the same functionally.
* ao_alsa: try to fallback to "hdmi" before "iec958" for spdifwm42016-10-071-5/+10
| | | | | | | | | | | | | If the "default" device refuses to be opened as spdif device (i.e. it errors due to the AES0 etc. parameters), we were falling back to the iec958 device. This is needed on some systems for smooth operation with PCM vs. spdif. Now change it to try "hdmi" before "iec958", which supposedly helps in other situations. Better suggestions welcome. Apparently kodi does this too, although I didn't check directly.
* audio/out: deprecate device sub-optionswm42016-09-051-1/+1
| | | | | We have --audio-device, which can force the device. Also add something describing to this extent to the manpage.
* options: add a mechanism to make sub-option replacement slightly easierwm42016-09-051-1/+2
| | | | | | | | Instead of requiring each VO or AO to manually add members to MPOpts and the global option table, make it possible to register them automatically via vo_driver/ao_driver.global_opts members. This avoids modifying options.c/options.h every time, including having to duplicate the exact ifdeffery used to enable a driver.
* ao_alsa: change sub-options to global optionswm42016-09-021-31/+51
| | | | | | Same deal as with vo_opengl. Also edit the outdated information about multichannel output a little.
* ao_alsa: log if retrieving supported channel maps failswm42016-08-221-1/+3
| | | | It's a sign that the driver doesn't implement the channel map API.
* ao_alsa: make pause state more robust, reduce minor code duplicationwm42016-08-091-22/+25
| | | | | | | | | With the previous commit, ao_alsa.c now has 3 possible ways to pause playback. Actually all 3 of them need get_delay() to fake its return value, so don't duplicate that code. Also much of the code looks a bit questionable when considering inconsistent pause/resume calls from outside, so ignore redundant calls.
* ao_alsa: handle --audio-stream-silencewm42016-08-091-9/+51
| | | | | | | | | | | | | | | | | push.c does not handle this automatically, and AOs using push.c have to handle it themselves. Also, ALSA is low-level enough that it needs explicit support in user code. At least I haven't found any option that does this. We still can get away relatively cheaply by abusing underflow-handling for this. ao_alsa.c already configures ALSA to handle underflows by playing silence. So we purposely induce an underflow when opening the device, as well as when pausing or resetting the device. This introduces minor misbehavior: it doesn't account for the additional delay the initial silence adds, unless the device has fully played the fragment of silence when the player starts sending data to it. But nobody cares.
* audio: use --audio-channels=auto behavior, except on ALSAwm42016-08-041-1/+1
| | | | | | | | | | | | | | | | | | | | | | | This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
* ao_alsa: add more shitty workaroundswm42016-05-061-9/+25
| | | | | | | | | | | | | | | This reportedly makes it work on ODROID-C2. The idea for this hack is taken from kodi; they unconditionally set some or all of those flags. I don't trust ALSA enough to hope that setting these flags couldn't break something else, so we try without them first. It's not clear whether this is a driver bug or a bug in the ALSA libs. There is no ALSA bug tracker (the ALSA website has had a dead link to a deleted bug tracker fo years). There's not much we can do other than piling up ridiculous hacks. At least I think that at this point invalid API usage by mpv can be excluded as a cause. ALSA might be the worst audio API ever.
* ao_alsa: log final hwparams toowm42016-05-031-1/+2
| | | | snd_pcm_hw_params() updates them.
* ao_alsa: log hwparams while restricting themwm42016-04-281-0/+43
| | | | | | They can sometimes fail, so I want logging to determine what's going on. Most of them are at debug log-level, except the final hwparams.
* ao_alsa: filter audio device listwm42015-11-241-1/+15
| | | | | | | | Remove known useless device entries from the --audio-device list (and corresponding property). Do this because the list is supposed to be a high level list of devices the user can select. ALSA does not provide such a list (in an useable manner), and ao_alsa.c is still in the best position to improve the situation somewhat.
* ao_alsa: list bidirectional devices toowm42015-11-241-1/+1
| | | | | | | | | The ALSA doxygen says: IOID - input / output identification ("Input" or "Output"), NULL means both This bug was blatantly introduced with commit cf94fce4.
* ao_alsa: remove the last bits of legacy channel map fallbackwm42015-11-071-3/+1
| | | | | | | | Essenti