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* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-2/+2
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* Remove unused function declarationsChris Robinson2017-02-281-3/+0
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* Start a ALC_SOFT_loopback2 extensionChris Robinson2017-02-281-5/+28
| | | | | | | | | | | | | | | | | | | | | | This extends the base ALC_SOFT_loopback extension with support for B-Format. When ALC_FORMAT_CHANNELS_SOFT is set to ALC_BFORMAT3D_SOFT, then additional attributes must be specified. ALC_AMBISONIC_LAYOUT_SOFT must be set to ALC_ACN_SOFT or ALC_FUMA_SOFT for the desired channel layout, ALC_AMBISONIC_SCALING_SOFT must be set to ALC_N3D_SOFT, ALC_SN3D_SOFT, or ALC_FUMA_SOFT for the desired channel scaling/normalization scheme, and ALC_AMBISONIC_ORDER_SOFT must be set to an integer value greater than 0 for the ambisonic order (maximum allowed is implementation-dependent). Note that the number of channels required for ALC_BFORMAT3D_SOFT is dependent on the ambisonic order. The number of channels can be calculated by: num_channels = (order+1) * (order+1); /* or pow(order+1, 2); */ In addition, a new alcIsAmbisonicFormatSupportedSOFT function allows apps to determine which layout/scaling/order combinations are supported by the loopback device. For example, alcIsAmbisonicFormatSupported(device, ALC_ACN_SOFT, ALC_SN3D_SOFT, 2) will check if 2nd order AmbiX (ACN layout and SN3D scaling) rendering is supported for ALC_BFORMAT3D_SOFT output.
* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-271-6/+14
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* Dynamically allocate the ALsource Send[] arrayChris Robinson2017-02-211-1/+1
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-211-1/+1
| | | | | The voices are still all allocated in one chunk to avoid memory fragmentation. But they're accessed as an array of pointers since the size isn't static.
* Apply distance compensation when writing to the outputChris Robinson2017-02-191-0/+12
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* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-141-0/+9
| | | | | | ALsourceProps' Send[] array is placed at the end of the struct, and given an indeterminate size. Extra space is allocated at the end of each struct given the number of auxiliary sends set for the device.
* Remove a couple context lock wrapper functionsChris Robinson2017-02-071-6/+0
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* Remove __android_log_print calls for TRACEREFChris Robinson2017-01-271-1/+0
| | | | | TRACEREFs aren't normally important, and for as often as it happens, the added function calls are wasteful even if they end up doing nothing.
* Also log to __android_log_print on AndroidChris Robinson2017-01-261-0/+11
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* Move the B-Format HRTF virtual speaker stuff to InitHrtfPanningChris Robinson2017-01-181-12/+20
| | | | | This keeps the decoder matrices and coefficient mapping together for if it changes in the future.
* Use ALsizei in more placesChris Robinson2017-01-181-8/+8
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* Use ALsizei for sizes and offsets with the mixerChris Robinson2017-01-161-5/+5
| | | | | | Unsigned 32-bit offsets actually have some potential overhead on 64-bit targets for pointer/array accesses due to rules on integer wrapping. No idea how much impact it has in practice, but it's nice to be correct about it.
* Use second-order ambisonics for basic HRTF renderingChris Robinson2017-01-151-2/+2
| | | | | | This should improve positional quality for relatively low cost. Full HRTF rendering still only uses first-order since the only use of the dry buffer there is for first-order content (B-Format buffers, effects).
* Convert the SndIO backend to the updated APIChris Robinson2016-12-211-3/+0
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* Make some pointer-to-array parameters constChris Robinson2016-10-041-0/+42
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* Finalize AL_SOFT_gain_clamp_exChris Robinson2016-10-031-5/+0
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* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-241-0/+2
| | | | | | | | | Designed for apps that either don't change the listener's AL_GAIN, or don't allow the listener's AL_GAIN to go above 1. This allows the volume to still be increased further than such apps may allow, if users find it too quiet. Be aware that increasing this can easily cause clipping. The gain limit reported by AL_GAIN_LIMIT_SOFT is also affected by this.
* Remove some more unnecessary volatilesChris Robinson2016-09-241-5/+5
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* Use unsigned int shifts for device flagsChris Robinson2016-09-071-5/+5
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* Remove use of DECL_CONSTChris Robinson2016-09-061-6/+4
| | | | | No idea if it was really gaining us anything, but removing it fixes a crash I was getting with libs built with Clang.
* Add a query for the maximum source gain limitChris Robinson2016-08-281-0/+5
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* Allow sources to play while alcSuspendContext is in effectChris Robinson2016-08-261-1/+7
| | | | | | | | | | | | | | This appears to be how Creative's Windows drivers handle it, and is necessary for at least the Windows version of UT2k4 (otherwise it tries to play a source while suspended, checks and sees it's stopped, then kills it before it's given a chance to start playing). Consequently, the internal properties it gets mixed with are determined by what the source properties are at the time of the play call, and the listener properties at the time of the suspend call. This does not change alDeferUpdatesSOFT, which will still hold the play state change until alProcessUpdatesSOFT.
* Add a ref count to ALeffectStateChris Robinson2016-08-251-0/+6
| | | | | This is mostly just reorganizing the effects to call the Construct method which initializes the ref count.
* Combine related members into a structChris Robinson2016-08-241-10/+12
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* Use a more specialized mixer function for B-Format to HRTFChris Robinson2016-08-121-2/+2
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* Decode directly from B-Format to HRTF instead of a cubeChris Robinson2016-08-111-2/+3
| | | | | | | | | | | | | | | | | | | | | Last time this attempted to average the HRIRs according to their contribution to a given B-Format channel as if they were loudspeakers, as well as averaging the HRIR delays. The latter part resulted in the loss of the ITD (inter-aural time delay), a key component of HRTF. This time, the HRIRs are averaged similar to above, except instead of averaging the delays, they're applied to the resulting coefficients (for example, a delay of 8 would apply the HRIR starting at the 8th sample of the target HRIR). This does roughly double the IR length, as the largest delay is about 35 samples while the filter is normally 32 samples. However, this is still smaller the original data set IR (which was 256 samples), it also only needs to be applied to 4 channels for first-order ambisonics, rather than the 8-channel cube. So it's doing twice as much work per sample, but only working on half the number of samples. Additionally, since the resulting HRIRs no longer rely on an extra delay line, a more efficient HRTF mixing function can be made that doesn't use one. Such a function can also avoid the per-sample stepping parameters the original uses.
* Remove DevFmtBFormat3D, which is covered by DevFmtAmbi1Chris Robinson2016-07-311-1/+1
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* Add an option to specify the ambisonic output configurationChris Robinson2016-07-311-0/+12
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* Add a stand-alone upsampler for higher-order ambisonic oputputChris Robinson2016-07-301-1/+4
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* Add a config to output first-, second-, or third-order ambisonicsChris Robinson2016-07-291-0/+3
| | | | | | | | | Currently incomplete, as second- and third-order output will not correctly handle B-Format input buffers. A standalone up-sampler will be needed, similar to the high-quality decoder. Also, output is ACN ordering with SN3D normalization. A config option will eventually be provided to change this if desired.
* Make a MAX_AMBI2D_COEFFS macro instead of a magic numberChris Robinson2016-07-171-1/+8
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* Avoid using memcpy to copy a single structChris Robinson2016-07-061-14/+12
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* Remove a slightly outdated commentChris Robinson2016-07-061-1/+0
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* Remove the VirtOut buffer aliasChris Robinson2016-07-051-6/+3
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* Rename MaxNoOfSources for consistencyChris Robinson2016-06-081-1/+1
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* Only define 8 HRTF filter states and params for the deviceChris Robinson2016-06-041-2/+2
| | | | | It will only be used with a cube channel setup, so there's no need to have one for every possible output channel.
* Add property queries to get the device latency with the clockChris Robinson2016-06-031-0/+2
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* Use a macro to specify the ambisonic periphonic channel maskChris Robinson2016-06-011-1/+10
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* Use a linked list for active effect slotsChris Robinson2016-05-291-1/+1
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* Use a specific lock for the backend's stop/reset/play callsChris Robinson2016-05-271-0/+2
| | | | | This helps protect against the device changing unexpectedly from multiple threads, instead of using the global list/library lock.
* Avoid separate updates to sources that should apply togetherChris Robinson2016-05-151-0/+6
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* Provide asynchronous property updates for sourcesChris Robinson2016-05-141-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | This necessitates a change in how source updates are handled. Rather than just being able to update sources when a dependent object state is changed (e.g. a listener gain change), now all source updates must be proactively provided. Consequently, apps that do not utilize any deferring (AL_SOFT_defer_updates or alcSuspendContext/alcProcessContext) may utilize more CPU since it'll be filling out more update containers for the mixer thread to use. The upside is that there's less blocking between the app's calling thread and the mixer thread, particularly for vectors and other multi-value properties (filters and sends). Deferring behavior when used is also improved, since updates that shouldn't be applied yet are simply not provided. And when they are provided, the mixer doesn't have to ignore them, meaning the actual deferring of a context doesn't have to synchrnously force an update -- the process call will send any pending updates, which the mixer will apply even if another deferral occurs before the mixer runs, because it'll still be there waiting on the next mixer invocation. There is one slight bug introduced by this commit. When a listener change is made, or changes to multiple sources while updates are being deferred, it is possible for the mixer to run while the sources are prepping their updates, causing some of the source updates to be seen before the other. This will be fixed in short order.
* Use a lockless method for updating listener and context propertiesChris Robinson2016-05-111-4/+3
| | | | | | | | | | | This uses a separate container to provide the relevant properties to the internal update method, using atomic pointer swaps. A free-list is used to avoid having too many individual containers. This allows the mixer to update the internal listener properties without requiring the lock to protect against async updates. It also allows concurrent read access to the user-facing property values, even the multi-value ones (e.g. the vectors).
* Add a comment about CoeffCount being 0Chris Robinson2016-05-101-1/+3
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* Start AL_SOFT_buffer_samples2 as a replacement for AL_SOFT_buffer_samplesChris Robinson2016-04-251-0/+73
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* Increase max output channels to 16Chris Robinson2016-04-191-1/+8
| | | | This also enables fully periphonic 3rd order HQ decoding.
* Avoid storing channel names for the dry bufferChris Robinson2016-04-161-2/+0
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* More directly map coefficients for ambisonic mixing buffersChris Robinson2016-04-151-4/+12
| | | | | | Instead of looping over all the coefficients for each channel with multiplies, when we know only one will have a non-0 factor for ambisonic mixing buffers, just index the one with a non-0 factor.
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