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path: root/Alc/mixer.c
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* Combine related members into a structChris Robinson2016-08-241-1/+1
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* Don't store the looping state in the voiceChris Robinson2016-07-311-1/+1
| | | | | Certain operations on the buffer queue depend on the loop state to behave properly, so it should not be deferred until the async voice update occurs.
* Fix use of a loop varChris Robinson2016-07-251-9/+11
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* Move the input channel array out of the DirectParams and SendParamsChris Robinson2016-07-131-20/+18
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-111-14/+14
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* Avoid function calls to get the HRTF sample rate and IR sizeChris Robinson2016-07-071-1/+1
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* Make the source position calues atomicChris Robinson2016-05-191-7/+7
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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 the real output's left and right channels with HRTFChris Robinson2016-03-111-1/+7
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* Only calculate steps for the used coefficientsChris Robinson2016-02-141-1/+1
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* Calculate HRTF stepping params right before mixingChris Robinson2016-02-141-4/+36
| | | | | This means we track the current params and the target params, rather than the target params and the stepping. This closer matches the non-HRTF mixers.
* Calculate channel gain stepping just before mixingChris Robinson2016-02-141-8/+91
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* Mix to multichannel for effectsChris Robinson2016-01-281-1/+1
| | | | | | This mixes to a 4-channel first-order ambisonics buffer. With ACN ordering and N3D scaling, this makes it easy to remain compatible with effects that only care about mono input since channel 0 is an unattenuated mono signal.
* Move the bsincTable to a separate fileChris Robinson2015-11-101-988/+0
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* Update the bsinc tableChris Robinson2015-11-101-966/+967
| | | | Precision is increased to cover the full 32-bit float range.
* Cast a double->float return to silence MSVCChris Robinson2015-11-061-1/+1
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* Implement a band-limited sinc resamplerChris Robinson2015-11-051-1/+997
| | | | | | | | This is essentially a 12-point sinc resampler, unless it's resampling to a rate higher than the output, at which point it will vary between 12 and 24 points and do anti-aliasing to avoid/reduce frequencies going over nyquist. Code provided by Christopher Fitzgerald.
* Change the Kaiser rippling limit to -60dBChris Robinson2015-11-041-2/+2
| | | | | This improves the transition cutoff, shortening its width and reducing the amount of error.
* Replace the Lanczos window with Kaiser for the sinc resamplerChris Robinson2015-11-041-17/+87
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* Update filter histories even when they're not usedChris Robinson2015-10-241-0/+4
| | | | | | If the filter properties are continually updated, and the HF or LF gain goes from <1, to 1, and later back to <1, the history shouldn't hold stale values from before it was at 1.
* Set XYZ channel gains for source sends to 0Chris Robinson2015-10-231-18/+0
| | | | | It's cleaner to just set the gains to 0 rather than to special-case B-Format in the mixer.
* Use one send gain per buffer channelChris Robinson2015-10-231-1/+1
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* Use a constant value for the post-position paddingChris Robinson2015-10-151-33/+20
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* Store the source's previous samples with the voiceChris Robinson2015-10-151-92/+32
| | | | | | This helps avoid different results when looping is toggled within a couple samples of the loop point, or when a processed buffer is removed while the source is only a couple samples into the next buffer.
* Replace the sinc6 resampler with sinc8, and make SSE versionsChris Robinson2015-10-111-13/+23
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* Move the FIR4 from SSE2 to SSE3Chris Robinson2015-10-111-3/+3
| | | | | SSE3 can avoid the slow _MM_TRANSPOSE_PS4 call thanks to the inclusion of horizontal adds.
* Use doubles to calculate the Lanczos coefficientsChris Robinson2015-10-091-20/+20
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* Combine two arraysChris Robinson2015-10-011-13/+11
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* Move the resampler stuff to mixer.c where it's usedChris Robinson2015-10-011-0/+53
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* Implement a 6-point sinc-lanczos filterChris Robinson2015-09-291-12/+26
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* Replace the cubic resampler with a 4-point sinc/lanczos filterChris Robinson2015-09-271-10/+19
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* Don't keep selecting the mixer to useChris Robinson2015-09-271-30/+32
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* Increase the max pitch to 255Chris Robinson2015-09-261-0/+3
| | | | | | | Note that this is the multiple above the device sample rate, rather than the source property limit. It could theoretically be increased to 511 by testing against UINT_MAX instead of INT_MAX, since the increment and positions are using unsigned integers. I'm just being paranoid about overflows.
* Move HRTF params and state closer togetherChris Robinson2015-02-091-2/+2
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* Do up to 256 samples at a time with multi-step loopsChris Robinson2014-12-181-2/+2
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* Inline a couple functionsChris Robinson2014-12-181-2/+2
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* Offset to the buffer's channel start firstChris Robinson2014-12-181-3/+6
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* Assert that there's a buffer for mixingChris Robinson2014-12-171-0/+1
| | | | For Clang's static analysis.
* Add SSE2 and SSE4.1 cubic resamplersChris Robinson2014-12-151-0/+8
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* Use a lookup table to do cubic resamplingChris Robinson2014-12-151-0/+17
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* Remove IrSize from DirectParamsChris Robinson2014-11-291-1/+4
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* Partially revert "Use a different method for HRTF mixing"Chris Robinson2014-11-231-2/+23
| | | | | | | | | | | | The sound localization with virtual channel mixing was just too poor, so while it's more costly to do per-source HRTF mixing, it's unavoidable if you want good localization. This is only partially reverted because having the virtual channel is still beneficial, particularly with B-Format rendering and effect mixing which otherwise skip HRTF processing. As before, the number of virtual channels can potentially be customized, specifying more or less channels depending on the system's needs.
* Rename Voice's NumChannels to OutChannelsChris Robinson2014-11-221-1/+1
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* Store the number of output channels in the voiceChris Robinson2014-11-221-1/+1
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* Remove an unnecessary union containerChris Robinson2014-11-221-1/+1
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* Use a different method for HRTF mixingChris Robinson2014-11-221-23/+2
| | | | | | | | | | | | | | | | | | | | | | | This new method mixes sources normally into a 14-channel buffer with the channels placed all around the listener. HRTF is then applied to the channels given their positions and written to a 2-channel buffer, which gets written out to the device. This method has the benefit that HRTF processing becomes more scalable. The costly HRTF filters are applied to the 14-channel buffer after the mix is done, turning it into a post-process with a fixed overhead. Mixing sources is done with normal non-HRTF methods, so increasing the number of playing sources only incurs normal mixing costs. Another benefit is that it improves B-Format playback since the soundfield gets mixed into speakers covering all three dimensions, which then get filtered based on their locations. The main downside to this is that the spatial resolution of the HRTF dataset does not play a big role anymore. However, the hope is that with ambisonics- based panning, the perceptual position of panned sounds will still be good. It is also an option to increase the number of virtual channels for systems that can handle it, or maybe even decrease it for weaker systems.
* Rename speakers to channels, and remove an old incorrect commentChris Robinson2014-11-071-1/+1
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* Use a separate macro for the max output channel countChris Robinson2014-11-071-1/+1
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* Use the copy resampler only when there's no sub-sample offsetChris Robinson2014-11-021-7/+6
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* Add preliminary AL_EXT_BFORMAT supportChris Robinson2014-10-311-0/+17
| | | | | Currently missing the AL_ORIENTATION source property. Gain stepping also does not work.
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