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* ao_coreaudio: refactor chmap detectionStefano Pigozzi2013-07-223-91/+96
| | | | | | b2f9e0610 introduced this functionality with code that was quite 'monolithic'. Split the functionality over several functions and ose the new macros to get array properties.
* ao_coreaudio: refactor properties codeStefano Pigozzi2013-07-224-146/+111
| | | | | | | | Introduce some macros to deal with properties. These allow to work around the limitation of CoreAudio's API being `void **` based. The macros allow to keep their client's code DRY, by not asking size and other details which can be derived by the macro itself. I have no idea why Apple didn't design their API like this in the first place.
* ao_coreaudio: move utils functions to snake_caseStefano Pigozzi2013-07-223-39/+34
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* ao_coreaudio: split ao_coreaudio_common in two filesStefano Pigozzi2013-07-225-160/+287
| | | | | | | | | * ao_coreaudio_utils: contains several utility function * ao_coreaudio_properties: contains functions to set and get audio object properties. Conflicts: audio/out/ao_coreaudio.c
* ao_coreaudio: store asbd only when selectedStefano Pigozzi2013-07-221-4/+1
| | | | | Previous code needlessly stored the input asbd before actually testing it's support against the hardware.
* ao_coreaudio: fallback to waveext on non surround inputsStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: set channel layout based on hardware queryStefano Pigozzi2013-07-221-2/+106
| | | | this is a wip
* ao_coreaudio: fix regression in digital stream selectionStefano Pigozzi2013-07-221-1/+1
| | | | | | The condition was checked wrongly on asbd which is the input format description. This lead to the condition always being true, thus selecting lpcm streams for digital input.
* ao_coreaudio: return errors instead false in init functionsStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: remove useless function declarationStefano Pigozzi2013-07-221-3/+0
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* ao_coreaudio: only set chmap_sel info for lpcmStefano Pigozzi2013-07-221-5/+7
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* ao_coreaudio: set channel layout bitmapStefano Pigozzi2013-07-221-1/+17
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* ao_coreaudio: move digital detection before asbd creationStefano Pigozzi2013-07-221-10/+9
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* ao_coreaudio: remove chmap selection if format is digitalStefano Pigozzi2013-07-221-1/+0
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* ao_coreaudio: remove volume multiplication by 4Stefano Pigozzi2013-07-221-2/+2
| | | | | kHALOutputParam_Volume is the linear gain so it should be at maximum 1 to keep the audio quality good. No idea why it was more than that.
* ao_coreaudio: remove device property listener on uninitStefano Pigozzi2013-07-222-12/+32
| | | | Also extract this functionality inside a function in coreaudio_common
* ao_coreaudio: print help string in one goStefano Pigozzi2013-07-221-5/+11
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* ao_coreaudio: change all ++var to var++Stefano Pigozzi2013-07-222-12/+8
| | | | | Luckily they all were inside for loops so the functionality does not actually change.
* ao_coreaudio: change private vars names to match mpv conventionsStefano Pigozzi2013-07-221-91/+101
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* ao_coreaudio: remove packetSize private variableStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: refactor play/pauseStefano Pigozzi2013-07-221-27/+15
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* ao_coreaudio: refactor uninitStefano Pigozzi2013-07-221-17/+6
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* ao_coreaudio: remove a fixme since that seems fixedStefano Pigozzi2013-07-221-2/+0
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* ao_coreaudio: ca_msg: add trailing \n where missingStefano Pigozzi2013-07-222-6/+6
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* ao_coreaudio: refactor playStefano Pigozzi2013-07-221-16/+8
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* ao_coreaudio: extract mixmode set/unset in utility functionsStefano Pigozzi2013-07-222-46/+58
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* ao_coreaudio: move AudioStreamChangeFormat to common file and refactorStefano Pigozzi2013-07-222-86/+74
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* ao_coreaudio: extract methods to lock/unlock device for digital outputStefano Pigozzi2013-07-222-28/+26
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* ao_coreaudio: lpcm: remove buffer size calculation depending on audio unitStefano Pigozzi2013-07-221-8/+0
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* ao_coreaudio: refactor initializationStefano Pigozzi2013-07-222-396/+282
| | | | | | | | | | | | | The initialization is split more clearly between compressed and lpcm case. For the compressed case, format selection is simplified a lot and negotiation removed. The way it was written it just passed back to the core the original requested format, not what was found available on hardware. Since this is most likely useless for the compressed case, I didn't bother with this. In the future I'd like to split this AO in two one that only uses the AUHAL and the other with direct access to the hardware so that even passthrough of lcpm can be possible. This would decrease the latency, audiophiles would like that.
* ao_coreaudio: refactor print_helpStefano Pigozzi2013-07-221-26/+21
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* ao_coreaudio: split out some utility functions and refactor themStefano Pigozzi2013-07-222-310/+315
| | | | | | | Split out some utility functions that use the CoreAudio API but are not related the main task of the AOs (which is to move data correctly to the ringbuffer). These are mainly need for the verbosity of the CoreAudio API and are just obscuring the 'real' code.
* ao_coreaudio: make variable names shorterStefano Pigozzi2013-07-221-30/+30
| | | | property_address -> p_addr
* ao_coreaudio: add error check function and macroStefano Pigozzi2013-07-221-41/+68
| | | | WIP
* ao_coreaudio: dry up ca_msg and use it everywhereStefano Pigozzi2013-07-221-61/+57
| | | | | Change the ca_msg macro to pass along MSGT_AO automatically. Also use it for every output for consistency.
* ao_coreaudio: simplify digital render callbackStefano Pigozzi2013-07-221-4/+3
| | | | | It was reported that it also works by not setting the read size in the AudioBuffer (now idea how, but I will discover it later).
* ao_coreaudio: rewrite spdif render callbackStefano Pigozzi2013-07-221-39/+20
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* ao_coreaudio: simplify render callbackStefano Pigozzi2013-07-221-23/+9
| | | | | | | Read only the requested amount by the AUHAL (instead of all the buffered data). No idea what the deal is with pausing the audio units if there is no audio to play, maybe to avoid underruns of some sort. Anyway from my tests this condition never occurred so I'm removing it all.
* af_bs2b: use new option APIwm42013-07-221-84/+28
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* af_lavfi: switch to new option APIwm42013-07-221-8/+24
| | | | | This makes it actually possible to use the filter with more complicated filter graphs (such as graphs containing the "," character).
* af_scaletempo: use new option APIwm42013-07-221-84/+44
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* af_lavrresample: switch to new option APIwm42013-07-221-51/+45
| | | | | Also add a "o" suboption, which should allow fine control over libavresample.
* af_force: use new option APIwm42013-07-221-34/+21
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* audio/filter: use new option APIwm42013-07-225-163/+99
| | | | | | | | | | | | | Make the VF/VO/AO option parser available to audio filters. No audio filter uses this yet, but it's still a quite intrusive change. In particular, the commands for manipulating filters at runtime completely change. We delete the old code, and use the same infrastructure as for video filters. (This forces complete reinitialization of the filter chain, which hopefully isn't a problem for any use cases. The old code forced reinitialization too, but it could potentially allow a filter to cache things; e.g. consider loaded ladspa plugins and such.)
* af_force: add option that causes filter to fail at initializationwm42013-07-221-1/+9
| | | | This is useful for debugging.
* af: fix recovery code for filter insertion (changing volume with spdif crash)wm42013-07-221-4/+2
| | | | | | | | | | | | This code is supposed to run if dynamic filter insertion (such as when inserting a volume filter in mixer.c) fails. Then it removes all filters and recreates the default list of filters. But the code just blew up and entered an endless loop, because it removed even the sentinel in/out filters. This could happen when trying to use softvol controls while using spdif, but also other situations. Fix it by calling the correct code. Also remove these obnoxious yoda-conditions.
* audio/decode: remove macro crapwm42013-07-227-115/+61
| | | | | Declare decoders directly, instead of using the LIBAD_EXTERN macro. This is simpler (no weird magic) and more extensible.
* ao_wasapi: Make default on Windows.Diogo Franco (Kovensky)2013-07-221-3/+3
| | | | Ahead of OSS because cygwin provides OSS.
* ao_wasapi: Fix S/PDIF passthrough initDiogo Franco (Kovensky)2013-07-221-3/+4
| | | | | | | | | | MSDN tells me to multiply the samplerates by 4 (for setting up the S/PDIF signal frequency), but doesn't mention that I'm only supposed to do it on the new, NT6.1+ IEC 61937 structs. Works on my Realtek Digital Output, but as I can't connect any hardware to it I can't hear the result. Also, always ask for little-endian AC3. I'm not sure if this is supposed to be LE or NE, but Windows is LE on all platforms, so we go with LE.
* ao_wasapi: Log the passthrough format in MSGL_VDiogo Franco (Kovensky)2013-07-221-0/+5
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* ao_wasapi: Also do passthrough for AF_FORMAT_MPEG2Diogo Franco (Kovensky)2013-07-221-1/+1
| | | | That's the sample format ad_spdif uses when the source is MP3.
* ao_wasapi: Support S/PDIF passthroughDiogo Franco (Kovensky)2013-07-221-0/+45
| | | | | | | Entirely untested as this troper has no S/PDIF hardware. Refuses trying any other format if we can't use passthrough, or we would end up sending white noise at the user.
* ao_wasapi: Fix double free on uninitDiogo Franco (Kovensky)2013-07-221-2/+0
| | | | | Caused by incorrect conversion to the m_option API: since we don't allocate the state ourselves, we also don't free it ourselves.
* ao_wasapi: Support loading devices by nameDiogo Franco (Kovensky)2013-07-221-17/+65
| | | | | | | | | Do an strstr match against the device description and, if we have only a single match, take it. This works as long as the devices in the system don't change, but it's not supposed to be reliable; if one wants reliability, one uses the device ID string. Formatting.
* ao_wasapi: Don't search for devices as part of validationDiogo Franco (Kovensky)2013-07-221-23/+6
| | | | | | | | | | | This could turn valid parameters into syntax errors by the mere presence or abscence of a device (e.g. USB audio devices), so don't do that. We do validate that, if the parameter is an integer, it is not negative. We also respond to the "help" parameter, which does the same as the "list" suboption but exits after listing. Demote the validation logging to MSGL_DBG2.
* ao_wasapi: Change function macros to require semicolon after invocationDiogo Franco (Kovensky)2013-07-221-29/+29
| | | | Add semicolons where they were missing.
* ao_wasapi: Use OPT_STRING_VALIDATE for device suboptionDiogo Franco (Kovensky)2013-07-221-22/+82
| | | | | | | | | | | | | | | | | Validates by trying to pick the device using the device enumerator and aborting with out of range on failure. Refactors find_and_load_device to not use the wasapi_state; it might be called during validation. Adds missing CoInitialize/CoUninitialize calls. Remove unused variables (the SAFE_RELEASE macros keep them referenced so compiler warnings don't help finding them...). Remove the IMMDeviceEnumerator from the wasapi_state, it's only needed during initialization and initialization is now well factored enough to get rid of it. Try and connect to unplugged devices as well when using the device ID string.
* ao_wasapi: Fully convert to m_option APIDiogo Franco (Kovensky)2013-07-221-20/+10
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* ao_wasapi: Don't leak the default device's ID when listing devicesDiogo Franco (Kovensky)2013-07-221-3/+4
| | | | Also remove unused variable.
* ao_wasapi: Annotate the default device when listing devicesDiogo Franco (Kovensky)2013-07-221-4/+17
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* ao_wasapi: Refactor device listing/loadingDiogo Franco (Kovensky)2013-07-221-42/+97
| | | | | | | | | | | | Omit "{0.0.0.00000000}." on devices that start with that substring, re-add when searching for devices by ID. Log the device ID of the default device. Log the friendly name of the used device. Consistently refer to endpoints/devices as devices, as this is more consistent with mpv terminology.
* ao_wasapi0: Rename to ao_wasapiDiogo Franco (Kovensky)2013-07-222-46/+46
| | | | | Nobody knows what the 0 was for. There's no "WASAPI version 0". Just take it out.
* ao_wasapi0: Use the mix format directly in try_mix_formatDiogo Franco (Kovensky)2013-07-221-0/+2
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* ao_wasapi0: Don't complain about failed init during AO probingDiogo Franco (Kovensky)2013-07-221-8/+10
| | | | Only if the user specifically asked for ao_wasapi0.
* ao_wasapi0: Don't fail init when listing devicesDiogo Franco (Kovensky)2013-07-221-2/+0
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* ao_wasapi0: Demote "negotiation failed" message to MSGL_VDiogo Franco (Kovensky)2013-07-221-2/+2
| | | | | Could spam the console with what may be harmless in some cases. We already complain loudly if we're stuck checking this too many times.
* ao_wasapi0: Support shared mode, better format guessing methodDiogo Franco (Kovensky)2013-07-221-72/+258
| | | | | | | | | | | | | | | | | | | | | | Uses WASAPI in shared mode by default, add :exclusive flag to choose exclusive mode (duh). WASAPI works somewhat different in shared mode: the OS suggests the sample format to use, and the GetBuffer call is done slightly differently. The shared mode driver does not consume audio as fast as it notifies the thread; we need to check how much we're allowed to write. Not doing this correctly results in spamming the console with AUDCLNT_E_BUFFER_TOO_LARGE errors. When guessing formats for exclusive mode, try several sample size and sample rate combinations instead of just falling back to s16le@44100hz. If none of the rates are accepted, tries remixing >6 channels to 5.1 channels. Failing that, tries remixing to stereo. Failing everything, including the CD Red Book format, what else is left to test? Calculate buffer_block_size based on the configured channels and bytes per sample; MSDN docs say nBlockAlign is not guaranteed to be set for anything but integer PCM formats.
* ao_wasapi0: Support device enumeration and selectionDiogo Franco (Kovensky)2013-07-221-3/+215
| | | | | | | | | Adds the :list suboption to ao_wasapi0, which enumerates the audio endpoints in the system. Adds the :device=<n> suboption, which either takes an ID string (as output by list) or a device number and uses the requested device instead of the system default.
* ao_dsound: use new option APIwm42013-07-221-14/+14
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* ao_jack: use new option APIwm42013-07-221-72/+34
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* ao_oss: switch to new option APIwm42013-07-211-31/+30
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* options: remove --mixer and --mixer-channel, turn them into alsa/oss suboptswm42013-07-212-36/+19
| | | | | | These two options were supported by ALSA and OSS only. Further, their values were specific to the respective audio systems, so it doesn't make sense to keep them as top-level options.
* ao_rsound: use new option APIwm42013-07-211-27/+17
| | | | | Untested. I don't even know if this compiles. I have no clue what rsound even is.
* ao_sdl: use new option APIwm42013-07-211-29/+19
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* ao_openal: use new option APIwm42013-07-211-42/+29
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* ao_pulse: use new option APIwm42013-07-211-17/+14
| | | | Untested, but should be fine.
* options: hide encoding AO/VO in help outputwm42013-07-211-0/+1
| | | | | These can't be used manually. Encoding is enabled with -o instead, and the encoding AO/VO is selected using internal mechanisms.
* ao_alsa: use new option API (changes syntax)wm42013-07-211-66/+25
| | | | | | | | | | | | This changes how device names are handled. Before this commit, device names were mangled in strange ways to avoid clashing with the option parser syntax. "." was replaced with ",", and "=" with ":" (the user had to do the inverse to get the correct device name). The "new" option parser has multiple ways to escape option strings, so we don't need this confusing hack anymore. Add an explicit note to the manpage as well.
* ao_pcm: use new option APIwm42013-07-211-16/+12
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* ao_portaudio: use new option APIwm42013-07-211-32/+35
| | | | | This basically serves as example. All other AOs should be ported as well.
*