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* options: change option macros and all option declarationswm42020-03-181-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* options: change how option range min/max is handledwm42020-03-131-1/+1
| | | | | | | | | | | | | | | | | Before this commit, option declarations used M_OPT_MIN/M_OPT_MAX (and some other identifiers based on these) to signal whether an option had min/max values. Remove these flags, and make it use a range implicitly on the condition if min<max is true. This requires care in all cases when only M_OPT_MIN or M_OPT_MAX were set (instead of both). Generally, the commit replaces all these instances with using DBL_MAX/DBL_MIN for the "unset" part of the range. This also happens to fix some cases where you could pass over-large values to integer options, which were silently truncated, but now cause an error. This commit has some higher potential for regressions.
* options: split m_config.c/hwm42020-03-131-1/+1
| | | | | | | | | | | | | | | | | Move the "old" mostly command line parsing and option management related code to m_config_frontend.c/h. Move the the code that enables other part of the player to access options to m_config_core.c/h. "frontend" is out of lack of creativity for a better name. Unfortunately, the separation isn't quite clean yet. m_config_frontend.c still references some m_config_core.c implementation details, and m_config_new() is even left in m_config_core.c for now. There some odd functions that should be removed as well (marked as "Bad functions"). Fixing these things requires more changes and will be done separately. struct m_config is left with the current name to reduce diff noise. Also, since there are a _lot_ source files that include m_config.h, add a replacement m_config.h that "redirects" to m_config_core.h.
* audio: slightly simplify pull underrun message printingwm42020-02-131-4/+1
| | | | | | | | | | | | | | | A previous commit moved the underrun reporting to report_underruns(), and called it from get_space(). One reason was that I worried about printing a log message from a "realtime" callback, so I tried to move it out of the way. (Though there's little justification other than a bad feeling. While an older version of the pull code tried to avoid any mutexes at all in the callback to accommodate "requirements" from APIs like jackaudio, we gave up on that. Nobody has complained yet.) Simplify this and move underrun reporting back to the callback. But instead of printing the message from there, move the message into the playloop. Change the message slightly, because ao->log is inaccessible, and without the log prefix (e.g. "[ao/alsa]"), some context is missing.
* player: consider audio buffer if AO driver does not report underrunswm42020-02-131-5/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | AOs can report audio underruns, but only ao_alsa and ao_sdl (???) currently do so. If the AO was marked as not reporting it, the cache state was used to determine whether playback was interrupted due to slow input. This caused problems in some cases, such as video with very low video frame rate: when a new frame is displayed, a new frame has to be decoded, and since there it's so much further into the file (long frame durations), the cache gets into an underrun state for a short moment, even though both audio and video are playing fine. Enlarging the audio buffer didn't help. Fix this by making all AOs report underruns. If the AO driver does not report underruns, fall back to using the buffer state. pull.c behavior is slightly changed. Pull AOs are normally intended to be used by pseudo-realtime audio APIs that fetch an audio buffer from the API user via callback. I think it makes no sense to consider a buffer underflow not an underrun in any situation, since we return silence to the reader. (OK, maybe the reader could check the return value? But let's not go there as long as there's no implementation.) Remove the flag from ao_sdl.c, since it just worked via the generic mechanism. Make the redundant underrun message verbose only. push.c seems to log a redundant underflow message when resuming (because somehow ao_play_data() is called when there's still no new data in the buffer). But since ao_alsa does its own underrun reporting, and I only use ao_alsa, I don't really care. Also in all my tests, there seemed to be a rather high delay until the underflow was logged (with audio only). I have no idea why this happened and didn't try to debug this, but there's probably something wrong somewhere. This commit may cause random regressions. See: #7440
* ao: avoid unnecessary wakeupswm42020-02-131-7/+12
| | | | | | | | | | | | | If ao_add_events() is used, but all events flags are already set, then we don't need to wakeup the core again. Also, make the underrun message "exact" by avoiding the race condition mentioned in the comment. Avoiding redundant wakeups is not really worth the trouble, and it's actually just a bonus in the change making the ao_underrun_event() function return whether a new underrun was set, which is needed by the following commit.
* audio: react to --ao and --audio-buffer runtime changeswm42019-12-271-3/+3
| | | | | | Before this commit, runtime changes were only applied if something else caused audio to be reinitialized. Now setting them reinitializes audio explicitly.
* audio: add ao_audiotrack for androidAman Gupta2019-11-191-0/+4
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* input: add gamepad support through SDL2Stefano Pigozzi2019-10-231-1/+1
| | | | | | | | | | | | | | | The code is very basic: - only handles gamepads, could be extended for generic joysticks in the future. - only has button mappings for controllers natively supported by SDL2. I heard more can be added through env vars, there's also ways to load mappings from text files, but I'd rather not go there yet. Common ones like Dualshock are supported natively. - analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an activation threshold. - only supports one gamepad at a time. the feature is intented to use gamepads as evolved remote controls, not play multiplayer games in mpv :)
* ao: add API for underrun reportingwm42019-10-111-0/+14
| | | | | | | | | | | | | | AOs can now call ao_underrun_event() (in any context) if an underrun has happened. It will print a message. This will be used in the following commits. But for now, audio.c only clears the underrun bit, so that subsequent underruns still print the warning message. Since the underrun flag will be used in fragile ways by the playback state machine, there is the "reports_underruns" field that signals strong support for underrun reporting. (Otherwise, underrun events will not be used by it.)
* ao: use a local option structwm42018-05-241-3/+22
| | | | Instead of accessing MPOpts.
* build: make encoding mode non-optionalwm42018-05-031-2/+0
| | | | Makes it easier to not break the build by confusing the ifdeffery.
* encode: get rid of the output packet queuewm42018-05-031-1/+1
| | | | | | | | | | | | Until recently, ao_lavc and vo_lavc started encoding whenever the core happened to send them data. Since audio and video are not initialized at the same time, and the muxer was not necessarily opened when the first encoder started to produce data, the resulting packets were put into a queue. As soon as the muxer was opened, the queue was flushed. Change this to make the core wait with sending data until all encoders are initialized. This has the advantage that we don't need to queue up the packets.
* ao: do not allow actual buffer size of 0wm42018-03-081-0/+1
| | | | | | | | | You can use --audio-buffer=0 to minimize the audio buffer size. But if the AO reports no device buffer size (like e.g. ao_jack does), then the buffer size is actually 0, and playback can never work properly. Make it fallback to a size of 1, which is unlikely to work properly, but you get what you asked for, instead of a freeze.
* build: drop support for SDL1wm42018-02-131-1/+1
| | | | | For some reason it was supported for ao_sdl because we've only used SDL1 API.
* ao: minor simplification to gain processing codewm42017-11-301-4/+3
| | | | | Cosmetic move of a variable, and consider an adjustment below 1/256 or so not worth applying (even in the float case).
* ao: simplify hack for float atomicswm42017-11-301-14/+2
| | | | | | | stdatomic.h defines no atomic_float typedef. We can't just use _Atomic unconditionally, because we support compilers without C11 atomics. So just create a custom atomic_float typedef in the wrapper, which uses _Atomic in the C11 code path.
* audio: add audio softvol processing to AOwm42017-11-291-0/+66
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* command: drop "audio-out-detected-device" propertywm42017-10-091-7/+0
| | | | | | Coreaudio stopped setting it a few releases ago (66a958bb4fa). There is not much of a user- or API-visible change, so remove it without deprecation.
* audio: introduce a new type to hold audio frameswm42017-08-161-6/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
* audio/out: fix comment typoKevin Mitchell2017-07-091-1/+1
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* audio/out: add helper code to do 24 bit conversion in AOwm42017-07-071-0/+69
| | | | | | | | | | | | | | | | | | | | | | | | | I plan to remove the S24 sample formats in mpv. It seems like we should still support this _somehow_ in AOs though. So the idea is to convert the data to more obscure representations (that would not be useful for filtering etc. anyway) within the AO. This commit adds helper to enable this. ao_convert_fmt is meant to provide mechanisms for this, rather than a generic audio format description (as the latter leads only to overly generic misery). The conversion also supports only cases which we think will be needed at all. The main advantage of this approach is that we get S24 out of sight, and that we could support other crazy formats (like S20). The main disadvantage is that usually S32 will be selected (if both S32 and S24 are available), and there's no user control to force S24. That doesn't really matter though, and at worst makes testing harder or will lead to unpleasant arguments with audiophiles (they'd be wrong anyway). ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in which playing S32 with data in the LSBs breaks when playing it as padded 24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the unused bits to 0 if wValidBitsPerSample implies LSB padding.)
* audio/out: require AO drivers to report period size and correct bufferwm42017-06-251-0/+7
| | | | | | | | | | | | | | | | 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.
* audio/out: change license of some core files to LGPLwm42017-05-201-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | All contributors of the current code have agreed. ao.c requires a "driver" entry for each audio output - we assume that if someone who didn't agree to LGPL added a line, it's fine for ao.c to be LGPL anyway. If the affected audio output is not disabled at compilation time, the resulting binary will be GPL anyway, and ootherwise the code is not included. The audio output code itself was inspired or partially copied from libao in 7a2eec4b59f4 (thus why MPlayer's audio code is named libao2). Just to be sure we got permission from Aaron Holtzman, Jack Moffitt, and Stan Seibert, who according to libao's SVN history and README are the initial author. (Something similar was done for libvo, although the commit relicensing it forgot to mention it.) 242aa6ebd40: anders mostly disagreed with the LGPL relicensing, but we got permission for this particular commit. 0ef8e555735: nick could not be reached, but the include statement was removed again anyway. 879e05a7c17: iive agreed to LGPL v3+ only, but this line of code was removed anyway, so ao_null.c can be LGPL v2.1+. 9dd8f241ac2: patch author could not be reached, but the corresponding code (old slave mode interface) was completely removed later.
* audio: lower "Disabling multichannel output." warning to verbosewm42017-04-021-1/+1
| | | | Not sure why it was a warning in the first place.
* ao: never set ao->device = ""Kevin Mitchell2017-02-201-2/+3
| | | | | | | For example, previously, --audio-device='alsa/' would provide ao->device="" to the alsa driver in spite of the fact that this is an already parsed option. To avoid requiring a check of ao->device[0] in every driver, make sure this never happens.
* ao: fix potential NULL deref in ao_device_list_add()wm42017-02-201-2/+2
| | | | | | Probably didn't happen in practice, but anyway. Found by coverity.
* options: remove deprecated sub-option handling for --vo and --aowm42016-11-251-13/+10
| | | | | | | | 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: make empty device ID mean default devicewm42016-11-141-7/+14
| | | | | | | | | This will make it easier for AOs to add explicit default device entries. (See next commit.) Hopefully this change doesn't lead accidentally to bogus "Default" entries to appear, but then it can only happen if the device ID is empty, which would mean the underlying audio API returned bogus entries.
* audio: avoid returning audio-device-list entries without descriptionwm42016-11-141-0/+2
| | | | | | Use the device name as fallback. This is ugly, but still better than skipping the description entirely. This can be an issue on ALSA, where the API can return entries without proper description.
* audio/out: add AudioUnit output driver for iOSAman Gupta2016-11-011-0/+4
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* player, ao, vo: don't call mp_input_wakeup() directlywm42016-09-161-17/+27
| | | | | | | | | | | | | Currently, calling mp_input_wakeup() will wake up the core thread (also called the playloop). This seems odd, but currently the core indeed calls mp_input_wait() when it has nothing more to do. It's done this way because MPlayer used input_ctx as central "mainloop". This is probably going to change. Remove direct calls to this function, and replace it with mp_wakeup_core() calls. ao and vo are changed to use opaque callbacks and not use input_ctx for this purpose. Other code already uses opaque callbacks, or has legitimate reasons to use input_ctx directly (such as sending actual user input).
* options: add automagic hack for handling sub-option deprecationswm42016-09-051-0/+1
| | | | | | | | | | | | | | | | | I decided that it's too much work to convert all the VO/AOs to the new option system manually at once. So here's a shitty hack instead, which achieves almost the same thing. (The only user-visible difference is that e.g. --vo=name:help will list the sub-options normally, instead of showing them as deprecation placeholders. Also, the sub-option parser will verify each option normally, instead of deferring to the global option parser.) Another advantage is that once we drop the deprecated options, converting the remaining things will be easier, because we obviously don't need to add the compatibility hacks. Using this mechanism is separate in the next commit to keep the diff noise down.
* options: add a mechanism to make sub-option replacement slightly easierwm42016-09-051-0/+1
| | | | | | | | 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 optionswm42016-09-021-0/+1
| | | | | | Same deal as with vo_opengl. Also edit the outdated information about multichannel output a little.
* 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.
* vo, ao: disable positional parameter suboptionswm42016-09-011-0/+1
| | | | | | | | | | | | | | | | Positional parameters cause problems because they can be ambiguous with flag options. If a flag option is removed or turned into a non-flag option, it'll usually be interpreted as value for the first sub-option (as positional parameter), resulting in very confusing error messages. This changes it into a simple "option not found" error. I don't expect that anyone really used positional parameters with --vo or --ao. Although the docs for --ao=pulse seem to encourage positional parameters for the host/sink options, which means it could possibly annoy some PulseAudio users. --vf and --af are still mostly used with positional parameters, so this must be a configurable option in the option parser.
* player: add --audio-stream-silencewm42016-08-091-0/+2
| | | | | Completely insane that this has to be done. Crap for compensating HDMI crap.
* audio: use --audio-channels=auto behavior, except on ALSAwm42016-08-041-5/+29
| | | | | | | | | | | | | | | | | | | | | | | 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.
* audio: don't add default entry to audio-device-list if AO support listingwm42016-06-291-3/+2
| | | | | | | In such cases there isn't really a reason to do so, and using such an entry would probably fail anyway. Also convenient for the following commit.
* ao: initial OpenSL ES supportIlya Zhuravlev2016-02-271-0/+4
| | | | | | | | OpenSL ES is used on Android. At the moment only stereo output is supported. Two options are supported: 'frames-per-buffer' and 'sample-rate'. To get better latency the user of libmpv should pass values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER) and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).
* mpv_talloc.h: rename from talloc.hDmitrij D. Czarkoff2016-01-111-1/+1
| | | | This change helps avoiding conflict with talloc.h from libtalloc.
* ao_dsound: remove this audio outputwm42016-01-061-4/+0
| | | | | | | It existed for XP-compatibility only. There was also a time where ao_wasapi caused issues, but we're relatively confident that ao_wasapi works better or at least as good as ao_dsound on Windows Vista and later.
* ao: disambiguate default device list entrieswm42015-11-271-2/+6
| | | | | | | | | | | If there were many AO drivers without device selection, this added a "Default" entry for each AO. These entries were not distinguishable, as the device list feature is meant not to require to display the "raw" device name in GUIs. Disambiguate them by adding the driver name. If the AO is the first, the name will remain just "Default". (The condition checks "num > 1", because the very first entry is the dummy for AO autoselection.)
* audio: always log channel maps before determining final mapwm42015-10-261-11/+4
| | | | | Until now, this was done only in debug verbosity, while some AOs logged equivalent information in verbose mode.