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* ao_wasapi: do not use deprecated wchar functionsPedro Pombeiro2017-06-292-2/+2
| | | | These break on UWP. Based on a patch by Pedro Pombeiro.
* Avoid calling close(-1)wm42017-06-291-2/+5
| | | | | | | | | | While this is perfectly OK on Unix, it causes annoying valgrind warnings, and might be otherwise confusing to others. On Windows, the runtime can actually abort the process if this is called. push.c part taken from a patch by Pedro Pombeiro.
* ao_wasapi: remove subtly duplicated codewm42017-06-281-7/+6
| | | | Seems like this can be slightly simplified.
* ao_wasapi: remove duplicate code for creating IAudioClientwm42017-06-281-34/+24
| | | | | | | | | | | | The code accounting for the terrible AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED semantics (which MSDN claims can happen "starting with Windows 7" - so probably on Windows 10 too) duplicated the call for creating the IAudioClient. That's not great, so get rid of it. Let wasapi_thread_init() handle this. It has a retry loop anyway. This redoes device lookup and format negotiation, but potential failures due to race conditions (what if the driver decides to change behavior) shouldn't be worse than before.
* audio/out/pull: detect and log underflowswm42017-06-281-0/+20
| | | | Mostly for debugging, I guess.
* audio/out: require AO drivers to report period size and correct bufferwm42017-06-259-4/+39
| | | | | | | | | | | | | | | | 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.
* ao_openal: change license to LGPLwm42017-06-241-7/+7
| | | | All authors have agreed.
* ao_pulse: reorder format choiceNiklas Haas2017-06-231-2/+2
| | | | | | | | | | | | Right now, the current order pretty much means that pulse defaults to S16 for arbitrary unsupported formats, but fallback to float would make more sense since it's the easiest to convert everything to without requiring dithering, and PA will probably just internally convert things to float anyway. Also move S32 above S16, which essentially means format_maps is sorted by preference. (Although ao_pulse currently ignores this and always picks the first as a fallback)
* build: simplify OSS checks and remove changes by "bugmen0t"wm42017-06-221-6/+3
| | | | | | | | | | | | | | | | | | The user bugmen0t was apparently a shared github account with publicly available login. Thus, we can't get LGPL relicensing permission from the people who used this account. To relicense successfully, we have to remove all their changes. This commit should remove 20d1fc13, f26fb009, defbe48d. It also should remove whatever test fragments were copied from the ancient configure, as well as some configure logic (potentially that device path stuff). I think this change still preserves the most important use-cases of OSS: BSDs, and the Linux OSS emulation (the latter for testing only). According to an OSS user, the 4front checks were probably broken anyway. The SunAudio stuff was probably for (Open)Solaris, which is dead. ao_oss.c itself will remain GPL, and still contains bugmen0t changes.
* encode_lavc: move from GPL 2+ to LGPL 2.1+.Rudolf Polzer2017-06-131-7/+8
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* ao_pcm: change license to LGPLwm42017-05-201-7/+7
| | | | | | | | | | | | | | | | | | | | | All relevant authors have agreed to the relicensing. Problem cases: eca47b1a5edae: someone else gets credited for the "idea" of this change, but it doesn't seem like it was a patch (otherwise reimar would have said "patch"). Also, the associated code got essentially removed again anyway. (The option parsing was rewritten fully.) ffb529e4eb2a9: anonymous/unknown author, but the code was fully removed anyway. The struct was removed, and the modern code does explicit read/write calls. 40789473d215b: author was not contacted, but this code was removed anyway. The magic number (0x7ffff000) is still in the new code, but I don't think that is copyright relevant. c750b8ab2d3c8: the message was entirely removed.
* audio/out: change license of some core files to LGPLwm42017-05-204-28/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* ao_wasapi: set name of event threadJames Ross-Gowan2017-05-181-0/+2
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* ao_pulse, ao_rsound: change license to LGPLwm42017-05-082-14/+14
| | | | | | | | All authors have agreed. One exception is 71247a97b35, whose author was not asked, but we deem the change as trivial. (And technically it was replaced when the audio chain dropped non-native endian sample formats.)
* ao_coreaudio: change license to LGPLwm42017-05-087-49/+49
| | | | | | | | | | | | | | | | All authors have agreed to the relicensing. The code was pretty much rewritten by Stefano Pigozzi. Since the rewrite happened incrementally, and seems to include refactored portions of older code, this relicensing was done on the pre-refactor code do. The original commit adding this AO (as ao_macosx.c) credits Timothy J. Wood as original author. He was asked and agreed to LGPL. It's not entirely sure from which project this code came from, but it's probably libao. In that project, Stanley Seibert made some changes to it (who as a major developer of libao was asked just to be sure), and also Ralph Giles and Ben Hines made two small changes. The latter were not asked, but none of their code survived anyway.
* options: remove remaining deprecated audio device selection optionswm42017-04-236-38/+3
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* ao_openal: kill off device listingwm42017-04-231-41/+1
| | | | | Probably helps with #4311. It surely is not the correct fix, of course. But ao_openal has no business of causing trouble anyway.
* ao_wasapi_changenotify: use %ls instead of %S for wchar_twm42017-04-201-4/+4
| | | | | %ls is C99. %S is supported by some systems, including MinGW/MSVC, but no reason to use it.
* ao_wasapi_changenotify: fix potential race conditionwm42017-04-201-8/+8
| | | | | | | IMMDeviceEnumerator_RegisterEndpointNotificationCallback() will start listening for notifications, and is the point at which callbacks can start firing. These callbacks will read the fields we set after the register calls, which is a potential race condition. Move it upwards.
* audio: lower "Disabling multichannel output." warning to verbosewm42017-04-021-1/+1
| | | | Not sure why it was a warning in the first place.
* ao_wasapi: do not pass nonsense to drivers with doublewm42017-03-291-5/+23
| | | | | | | | | | | | | | | | | | | | | | This tried to use AF_FORMAT_DOUBLE as KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, with wBitsPerSample==64. This is probably not allowed, and drivers appear to react inconsistently to it. (With one user, the format was accepted during format negotiation, but then rejected on actual init.) Remove it, which essentially forces it to fall back to some other format. (Looks like it'll use af_select_best_samplerate(), which would probably make it try S32 next.) The af_fmt_from_planar() is so that we don't have to care about AF_FORMAT_FLOATP. Wasapi always requires packed data anyway. This should actually handle other potentially unknown sample formats better. This changes that set_waveformat() always set the exact format. Now it might set a "close" format instead. But all callers seem to deal with this well. Although in theory, callers should probably handle the fallback. The next cleanup (if ever) can take care of this.
* ao_jack: update latency on buffer_size/graph changeCheng Sun2017-03-181-7/+34
| | | | | | | The buffer_size may be updated before the process callback is called for the first time. Or, the connection graph could change, which changes the latency of the pipeline after mpv's output. Ensure we keep on top of these changes by registering callbacks to update our latency estimation.
* 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.
* options: add M_OPT_FILE to some more file optionsPhilip Sequeira2017-03-061-1/+1
| | | | (Helps shell completion.)
* 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: 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.
* ao_oss: fix mixer channel messageKevin Mitchell2017-02-081-1/+1
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* ao_oss: use --audio-device if --oss-device isn't set.Kevin Mitchell2017-02-081-6/+10
| | | | | | | | | | | | | | Fall back on PATH_DEV_DSP if nothing is set. This mirrors the behaviour of --audio-device / --alsa-device. There doesn't appear to be a general way to list devices with oss, so --audio-device=help doesn't list oss devices except for the default one if the file exists. Previously --audio-device was ignored entirely by ao_oss. fixes #4122
* win32: add COM-specific SAFE_RELEASE to windows_utils.hJames Ross-Gowan2017-01-304-30/+27
| | | | | | | | | | | | | | | See: https://msdn.microsoft.com/en-us/library/windows/desktop/dd743946.aspx Microsoft example code often uses a SAFE_RELEASE macro like the one in the above link. This makes it easier to avoid errors when releasing COM interfaces. It also reduces noise in COM-heavy code. ao_wasapi.h also had a macro called SAFE_RELEASE, though unlike the version above, its SAFE_RELEASE macro accepted a second parameter which allowed it to destroy arbitrary objects other than just COM interfaces. This renames ao_wasapi's SAFE_RELEASE to SAFE_DESTROY, which should more accurately reflect what it does and prevent confusion with the Microsoft version.
* audio/out/push: merge if branches with same conditionwm42017-01-091-4/+3
| | | | Cosmetic change.
* Remove compatibility thingswm42016-12-071-23/+0
| | | | | | Possible with bumped FFmpeg/Libav. These are just the simple cases.
* ao_alsa: print certain ALSA errors as string instead as numberwm42016-12-071-2/+2
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* ao_wasapi: log return code when probing audio formatswm42016-11-302-13/+9
| | | | | | | | | We log a large number of formats, but we rarely log the result of the probing. Change this. The logic in try_format_exclusive() changes slightly, but should be equivalent. EXIT_ON_ERROR() checks for FAILED(), which should be exclusive to SUCCEEDED().
* ao_wasapi_utils: remove unused variablepavelxdd2016-11-271-1/+0
| | | | | Introduced in 1a2319f3e4cc42c680e2fd3ba30022c7a9adf3fe Produced a warning during compilation on Windows.
* options: remove deprecated sub-option handling for --vo and --aowm42016-11-2518-64/+23
| | | | | | | | 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-silencewm42016-11-242-13/+34
| | | | | | | | | | | | | | | | | | | 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.
* audio: fix --audio-stream-silence with ao_wasapiwm42016-11-211-2/+4
| | | | | Seems like wasapi will restart the HDMI stream if resume is called during playback.
* audio: fix --audio-stream-silence with ao_alsawm42016-11-211-2/+3
| | | | | | | ao_alsa.c calls this before the common code sets ao->sstride. Other than this, I'm still not sure whether this works. Seems like no, or depends.
* 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.
* 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.
* 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.
* audio/out: add AudioUnit output driver for iOSAman Gupta2016-11-016-5/+227
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* 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: prevent underruns with spdif under certain conditionswm42016-10-041-0/+2
| | | | | | | | | | | | | | | | | The player tries to avoid splitting frames with spdif (sample alignment stuff). This can in certain corner cases with certain drivers lead to the situation that ao_get_space() returns a number higher than 0 and lower than the audio frame size. The playloop will round this down to 0 bytes and do nothing, leading to a missed wakeup. This can lead to underruns or playback completely getting stuck. It can be reproduced by playing AC3 passthrough with no video and: --ao=null --ao-null-buffer=0.256 --ao-null-outburst=6100 This commit attempts to fix it by allowing the playloop to write some additional data (to get a complete frame), that will be buffered within the AO ringbuffer even if the audio device doesn't want it.
* win32: build with -DINITGUIDJames Ross-Gowan2016-09-282-2/+0
| | | | | | | | | | | | We always want to use __declspec(selectany) to declare GUIDs, but manually including <initguid.h> in every file that used GUIDs was error-prone. Since all <initguid.h> does is define INITGUID and include <guiddef.h>, we can remove all references to <initguid.h> and just compile with -DINITGUID to get the same effect. Also, this partially reverts 622bcb0 by re-adding libuuid.a to the build, since apparently some GUIDs (such as GUID_NULL) are not declared in the source file, even when INITGUID is set.
* ao_openal: enable building on OSXJosh de Kock2016-09-211-0/+14
| | | | Signed-off-by: Josh de Kock <josh@itanimul.li>
* options: simplify M_OPT_EXITwm42016-09-171-1/+1
| | | | | | | | | | | | | There were multiple values under M_OPT_EXIT (M_OPT_EXIT-n for n>=0). Somehow M_OPT_EXIT-n either meant error code n (with n==0 no error?), or the number of option valus consumed (0 or 1). The latter is MPlayer legacy, which left it to the option type parsers to determine whether an option took a value or not. All of this was changed in mpv, by requiring the user to use explicit syntax ("--opt=val" instead of "-opt val"). In any case, the n value wasn't even used (anymore), so rip this all out. Now M_OPT_EXIT-1 doesn't mean anything, and could be used by a new error code.
* player, ao, vo: don't call mp_input_wakeup() directlywm42016-09-165-23/+35
| | | | | | | | | | | | | 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).
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