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* ao_coreaudio_chmap: change license to LGPLwm42016-01-192-14/+14
| | | | | | While the situation is not really clear for the other rewritten coreaudio code, it's very clear for the channel mapping code. It was all written by us. (MPlayer doesn't even have any channel map handling.)
* Relicense some non-MPlayer source files to LGPL 2.1 or laterwm42016-01-197-49/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This covers source files which were added in mplayer2 and mpv times only, and where all code is covered by LGPL relicensing agreements. There are probably more files to which this applies, but I'm being conservative here. A file named ao_sdl.c exists in MPlayer too, but the mpv one is a complete rewrite, and was added some time after the original ao_sdl.c was removed. The same applies to vo_sdl.c, for which the SDL2 API is radically different in addition (MPlayer supports SDL 1.2 only). common.c contains only code written by me. But common.h is a strange case: although it originally was named mp_common.h and exists in MPlayer too, by now it contains only definitions written by uau and me. The exceptions are the CONTROL_ defines - thus not changing the license of common.h yet. codec_tags.c contained once large tables generated from MPlayer's codecs.conf, but all of these tables were removed. From demux_playlist.c I'm removing a code fragment from someone who was not asked; this probably could be done later (see commit 15dccc37). misc.c is a bit complicated to reason about (it was split off mplayer.c and thus contains random functions out of this file), but actually all functions have been added post-MPlayer. Except get_relative_time(), which was written by uau, but looks similar to 3 different versions of something similar in each of the Unix/win32/OSX timer source files. I'm not sure what that means in regards to copyright, so I've just moved it into another still-GPL source file for now. screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but they're all gone.
* ao_wasapi: use share_mode value instead of raw option opt_exclusiveKevin Mitchell2016-01-181-1/+1
| | | | | | | Previously used opt_exclusive option to decide which volume control code to run. The might not always reflect the actual state, for example if passthrough is used. Admittedly, none of the volume controls will work anyway with passthrough, but this is the right thing to do.
* ao_openal: wipe out global context on init errorKevin Mitchell2016-01-181-0/+1
| | | | | Previously this would break all further attempts to init the driver after one had failed.
* mpv_talloc.h: rename from talloc.hDmitrij D. Czarkoff2016-01-117-7/+7
| | | | This change helps avoiding conflict with talloc.h from libtalloc.
* ao_coreaudio: replace fourcc_repr()wm42016-01-113-35/+7
| | | | Replace with the more general mp_tag_str().
* ao_wasapi: move out some utility functionswm42016-01-112-76/+1
| | | | | | Note that hresult_to_str() (coming from wasapi_explain_err()) is mostly wasapi-specific, but since HRESULT error codes are unique, it can be extended for any other use.
* ao_dsound: remove this audio outputwm42016-01-062-711/+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_wasapi: remove unnecessary header fileKevin Mitchell2016-01-055-74/+31
| | | | | All the wasapi files were including both ao_wasapi.h and ao_wasapi_utils.h. Just merge them into a single file.
* ao_wasapi: initialize change notify in main threadKevin Mitchell2016-01-052-3/+6
| | | | This is something else that has nothing to do with audio rendering.
* ao_wasapi: remove old vistablob prototypeKevin Mitchell2016-01-051-2/+0
| | | | this function was removed earlier, but the prototype was missed
* ao_wasapi: make find_deviceID read only wrt struct aoKevin Mitchell2016-01-053-19/+20
| | | | This makes it clearer that state->device is being allocated.
* ao_wasapi: move device selection to main threadKevin Mitchell2016-01-053-8/+10
| | | | In attempt to simplify the audio event thread, this can now be moved out.
* ao_wasapi: avoid some redundant error messages in device selectionKevin Mitchell2016-01-051-15/+7
| | | | | If these error conditions are triggered, the called function will have already output a sufficiently informantive error message.
* ao_wasapi: alloc later to avoid free on errorKevin Mitchell2016-01-051-2/+1
| | | | | In get_device_desc, don't alloc the return value until we know there wasn't an error.
* ao_pulse: attempt to fall back to an arbitrary sample formatwm42016-01-051-2/+7
| | | | | | | | | | | | | | | | | | | | | | | Normally, PulseAudio accepts any combination of sample format, sample rate, channel count/map. Sometimes it does not. For example, the channel rate or channel count have fixed maximum values. We should not fail fatally in such cases, but attempt to fall back to a working format. We could just send pass an "unset" format to Pulse, but this is not too attractive. Pulse could use a format which we do not support, and also doing so much for an obscure corner case is not reasonable. So just pick a format that is very likely supported. This still could fail at runtime (the stream could fail instead of going to the ready state), but this sounds also too complicated. In particular, it doesn't look like pulse will tell us the cause of the stream failure. (Or maybe it does - but I didn't find anything.) Last but not least, our fallback could be less dumb, and e.g. try to fix only one of samplerate or channel count first to reduce the loss, but this is also not particularly worthy the effort. Fixes #2654.
* ao_pulse: check for sample rate boundswm42016-01-051-1/+1
| | | | | | | pa_format_info_valid() does not do this. (Although there is a proposed patch on the PulseAudio mailing list.) See #2654.
* ao_pulse: move format setting into a functionwm42016-01-051-20/+26
| | | | No real functional changes.
* ao_wasapi: remove +x flag from fileswm42016-01-044-0/+0
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* ao_wasapi: just use a pointer to the deviceID in change_notifyKevin Mitchell2016-01-041-3/+1
| | | | | Rather than creating a new string from the device instance. This will allow moving the change_init to the main thread before the device is loaded.
* ao_wasapi: correctly name the IMMNotificationClientVtblKevin Mitchell2016-01-041-2/+2
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* ao_wasapi: make persistent enumerator local to change_notifyKevin Mitchell2016-01-044-21/+12
| | | | This is no longer required by anything else
* ao_wasapi: rewrite device listing and selectionKevin Mitchell2016-01-042-225/+187
| | | | | | | | | | | | | Unify and clean up listing and selection. Use common enumerator code for both operations to avoid duplication or inconsistencies. Maintain, but significatnly simplify manual device selection by id, name or number. This actually fixes loading by name which didn't really work before since the "name" displayed by --audio-device=help differed from that used to match the selection, which used the device "description" instead. Save the selected deviceID in the private structure for later loading. This will permit moving the device selection into the main thread in a future commit.
* ao_wasapi: fix delay calculation againKevin Mitchell2016-01-021-6/+12
| | | | | | | | Apparently it's only wine where the qpc_position returned by IAudioClock_GetPosition can be overflowed. So actually do the rescaling correctly, but throw away the result if it looks unreasonable. this fixes a regression in 5afa68835ade9f21f9c709f791319bf9d2e35265
* ao_wasapi: fix delay calculationKevin Mitchell2015-12-211-21/+36
| | | | | | | | | | | | | | | | | | | Make sure that subtraction of performance counters is done correctly. Follow the *exact* instructions for converting performance counter to something comparable to the QPCposition returned by IAudioClient::GetPosition https://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx Also make sure that subtraction of unsigned integers is stored into a signed integer to avoid nastiness. Also be more careful about overflow in the conversion of the device position into number of samples. Avoid casting mp_time_us() to a double, and use llrint to convert the double precision delay_us back to integer for ao_read_data. Finally, actually check the return value of ao_read_data and add a verbose message if it is not the expected value. Unfortunately, there is no way to tell WASAPI when this happens since the frame_count in ReleaseBuffer must match GetBuffer.
* Fix some typos in code commentsAman Gupta2015-12-211-1/+1
| | | | Signed-off-by: wm4 <wm4@nowhere>
* ao_wasapi: move volume control init to it's own functionKevin Mitchell2015-12-211-21/+34
| | | | also make failure non-fatal
* ao_wasapi: correctly handle audio session display failureKevin Mitchell2015-12-211-10/+14
| | | | | In particular, try and release/null the interface so that it won't be marshalled.
* ao_wasapi: non-fatal error handling for COM marshallingKevin Mitchell2015-12-213-55/+65
| | | | | Also make sure that CoReleaseMarshalData is called if errors occur before unmarshalling.
* ao_wasapi: wrap long lines and use only c99 comment styleKevin Mitchell2015-12-214-119/+172
| | | | | also remove a log message in AOCONTROL_UPDATE_STREAM_TITLE since none of the other controls have one.
* ao_wasapi: reorganize private structureKevin Mitchell2015-12-211-37/+36
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* ao_wasapi: remove useless buffer_block_sizeKevin Mitchell2015-12-212-7/+0
| | | | this was only ever used for a verbose message
* ao_wasapi: move exclusive and shared-specific controls to functionsKevin Mitchell2015-12-211-75/+87
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* ao_wasapi: call the class-specific release functionsKevin Mitchell2015-12-201-6/+6
| | | | | IUnknown_Release() might be alright, but stay on the safe side.
* ao_wasapi: check for proxy availability in controlKevin Mitchell2015-12-201-22/+37
| | | | | Make sure that the proxy has been created before using it. This will be used when a future commit makes proxy setup optional.
* ao_wasapi: actually use hw volume support information for exclusive modeKevin Mitchell2015-12-202-17/+30
| | | | | | | | | Do not try and set/get master volume in exclusive if there is no hardware support. This would just uselessly change the master slider, but have no effect on the actual volume. Furthermore if getting hardware volume support information fails, then assume it has none.
* ao_wasapi: don't cast control arg to something it isn'tKevin Mitchell2015-12-201-9/+14
| | | | | the ao_control_vol_t cast was happening outside AOCONTROL_GET/SET_VOLUME which is the only place that would be valid
* ao_wasapi: remove volume "restore" on exitKevin Mitchell2015-12-203-48/+10
| | | | | | It was complicated and not even very intuitive to the user. If you are controlling the master volume, you just have to be prepared to deal with the consequences.
* ao_wasapi: split exclusive/shared specific ao controlsKevin Mitchell2015-12-201-37/+60
| | | | | this avoids having to check if we're exclusive or shared for every control
* ao_wasapi: add E_NOINTERFACE to error listKevin Mitchell2015-12-201-0/+1
| | | | this is encountered trying to set up COM proxies in wine
* 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.)
* 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_wasapi: get rid of Vistablob hackKevin Mitchell2015-11-244-51/+6
| | | | | This was required to work around XP linking issues and is no longer required.
* ao_wasapi: only report per-app volume in shared modeKevin Mitchell2015-11-191-1/+2
| | | | | otherwise we were incorrectly adjusting the hardware master volume in exclusive mode with softvol=auto
* ao_wasapi: work around DTS passthrough failurewm42015-11-191-1/+18
| | | | | | | | | | Apparently, some audio drivers do not support the DTS subtype, but passthrough works anyway if the AC3 subtype is set. Just retry with AC3 if the proper format doesn't work. The audio device which exposed this behavior reported itself as "M601d-A3/A3R (Intel(R) Display Audio)". xbmc/kodi even always passes DTS as AC3.
* ao_openal: fix sign of speaker angle in commentKevin Mitchell2015-11-181-1/+1
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* ao_openal: fix virtual speaker positioningJustas LaviĊĦius2015-11-181-10/+10
| | | | | | Place speakers in standard positions equidistant from the listener. use standard coordinate system
* ao_openal: accommodate more sample formatsKevin Mitchell2015-11-171-7/+44
| | | | | | Try and and choose the closest sample format to the one requested. fixes #2494
* ao_openal: move uninit before initKevin Mitchell2015-11-171-12/+12
| | | | the next commit will use uninit within init
* ao_jack: remove "alsa" std-channel-layout choicewm42015-11-071-5/+1
| | | | | Same deal as with previous commit. "waveext" is less arbitrary and at least supports 3/7 channels.
* ao_alsa: remove the last bits of legacy channel map fallbackwm42015-11-071-3/+1
| | | | | | | | Essentially we'd use something random, just because it's part of the srt of traditionally used ALSA channel mappings. But each driver can do its own things. This doesn't let me sleep at night, so remove it.
* ao_alsa: fix 7.1 over HDMIwm42015-11-041-0/+26
| | | | | | | | | | | | | | | | | We need to effectively swap the last channel pair. See commit 4e358a96 and 5a18c5ea for details. Doing this seems rather strange, as 7.1 just extends 5.1 with 2 new speakers, and 5.1 doesn't need this change. Going by the HDMI standard and the Intel HDA sources (cited in the referenced commits), it also looks like 7.1 should simply append two channels to 5.1 as well. But swapping them is apparently correct. This is also what XBMC does. (I didn't find any other applications doing 7.1 PCM using the ALSA channel map API. VLC seems to ignore the 7.1 case.) Testing reveals that at least the end result is correct. "Normal" ALSA 7.1 is unaffected by this, as it reports a different (and saner) channel layout.
* ao_alsa: map mp_chmaps back to ALSA in a different waywm42015-11-041-39/+54
| | | | | | | | | | | | | | | | Instead of constructing an ALSA channel map from mpv ones from scratch, try to find the original ALSA channel map again. Th result is that we need to convert channel maps only in one direction. If we need to map a mp_chmap to ALSA, we fetch the device's channel map list, convert each entry to mp_chmap, and find the first one which fits. This seems helpful for the following commit. For now, this only gets rid of mapping back the trivial MONO mapping, which alone would still be acceptable, but with other channel layout mogrifications it gets messy fast. While we need to do something awkward to keep our channel map reordering for VAR chmaps (which basically gives nicer output and possibly slightly better performance), this is still the better solution.
* ao_alsa: print more chmap info at debug verbositywm42015-11-041-9/+11
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* ao_alsa: minor cleanupswm42015-11-031-34/+18
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* ao_alsa: simplify dmix non-NA hackwm42015-11-031-58/+43
| | | | | There's really no need to do this deep in the chmap sslection code. This will setup the device further than before, but that doesn't matter.
* ao_alsa: move channel map setting code out of main init functionwm42015-11-031-104/+125
| | | | This grew way too large.
* ao_alsa: make failure of buffer parameter setting non-fatalwm42015-11-031-4/+12
| | | | | | | | | | | | | | | | | | | | These calls actually can leave the ALSA configuration space empty (how very useful), which is why snd_pcm_hw_params() can fail. An earlier change intended to make this non-fatal, but it didn't work for this reason. Backup the old parameters, so we can retry with the non-empty configuration space. (It has to be non-empty, because the previous setters didn't fail.) Note that the buffer settings are not very important to us. They're a leftover from MPlayer, which needed to write enough data to the audio device to not underrun while decoding and displaying a video frame. In mpv, most of these things happen asynchronously, _and_ there is a dedicated thread just for feeding the audio device, so we should be pretty imune even against extreme buffer settings. But I suppose it's still useful to prevent PulseAudio from making the buffer too large, so still keep this code.
* ao_alsa: disable resampling first thingwm42015-11-031-7/+7
| | | | | | Again, this could have bad access, is unlikely, and has no bad consequences. It's noteworthy that vlc and the ALSA PCM example both do this first, even if they set the sample rate later.
* ao_alsa: set access type before formatwm42015-11-031-11/+13
| | | | | | | | I'm worried that not restricting the access type before restricting the format will cause problems. While it's unlikely, it might prevent failures in some corner cases. Also, since we by default always use interleaved access (buggy ALSA plugins), this will have no effects at all.
* ao_alsa: handle channel count mismatch safeguard after chmap negotiationwm42015-11-031-12/+12
| | | | | | | | If the API doesn't list padded channel maps, but the final device channel map is padded, and if unpadded output is not possible (unlike in the somewhat similar dmix case), then we shouldn't apply the channel count mismatch fallback in the beginning. Do it after channel map negotiation instead.
* ao_alsa: apply non-NA fallback only if input is stereowm42015-11-031-0/+1
| | | | | | Doesn't matter much; effectively this prevents just log spam in some cases where the map is legitimately padded. Normally this is really only needed for the dmix ALSA case. (See git blame for details.)
* ao_alsa: treat SND_CHMAP_UNKNOWN as NA channel toowm42015-11-031-0/+1
| | | | | | Apparently required by nVidia HDMI. It should not be, and NA would definitely be more correct here, so this could be considered a driver bug. Maybe.
* ao_alsa: remove log message on pausingwm42015-11-031-1/+2
| | | | This was annoying.
* ao_coreaudio_exclusive: check for maximum channel countwm42015-10-261-0/+6
| | | | | | Until recently, the channel layout code happened to catch this, but now an explicit check is needed. Otherwise, it'd try to pad the missing channels with NA in the channel map fallback code.
* ao_coreaudio_exclusive: fallback to stereo on unknown channel layouts