Commit message (Collapse) | Author | Age | Files | Lines | |
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* | Combine related members into a struct | Chris Robinson | 2016-08-24 | 1 | -10/+12 |
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* | Make a function definition static | Chris Robinson | 2016-08-23 | 1 | -1/+1 |
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* | Hold updates for both listener and source updates | Chris Robinson | 2016-08-23 | 1 | -11/+0 |
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* | Don't pass the context's distance model as the source's | Chris Robinson | 2016-08-23 | 4 | -13/+18 |
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* | Avoid resupplying unneeded source updates | Chris Robinson | 2016-08-23 | 4 | -62/+38 |
| | | | | | The source's voice holds a copy of the last properties it received, so listener updates can make sources recalculate internal properties from that stored copy. | ||||
* | Use a more specialized mixer function for B-Format to HRTF | Chris Robinson | 2016-08-12 | 2 | -2/+6 |
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* | Decode directly from B-Format to HRTF instead of a cube | Chris Robinson | 2016-08-11 | 1 | -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. | ||||
* | Avoid checking DeferUpdates for each source state change | Chris Robinson | 2016-08-08 | 1 | -8/+24 |
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* | Use al_calloc/al_free in more places | Chris Robinson | 2016-08-03 | 2 | -9/+9 |
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* | Add 'restrict' to another parameter | Chris Robinson | 2016-08-03 | 2 | -2/+2 |
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* | Don't store the looping state in the voice | Chris Robinson | 2016-07-31 | 2 | -13/+22 |
| | | | | | 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. | ||||
* | Remove DevFmtBFormat3D, which is covered by DevFmtAmbi1 | Chris Robinson | 2016-07-31 | 1 | -1/+1 |
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* | Add an option to specify the ambisonic output configuration | Chris Robinson | 2016-07-31 | 1 | -0/+12 |
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* | Add a stand-alone upsampler for higher-order ambisonic oputput | Chris Robinson | 2016-07-30 | 1 | -1/+4 |
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* | Add a config to output first-, second-, or third-order ambisonics | Chris Robinson | 2016-07-29 | 1 | -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. | ||||
* | Avoid a NULL deref when creating 0 auxiliary effect slots | Chris Robinson | 2016-07-26 | 1 | -12/+9 |
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* | Remove the last use of ALfilterState_processSingle | Chris Robinson | 2016-07-26 | 2 | -18/+0 |
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* | Remove broken autowah effect code | Chris Robinson | 2016-07-26 | 5 | -21/+0 |
| | | | | | It's been disabled forever, and I have no idea how to make it work properly. Better to just redo it when making something that works. | ||||
* | Add some more restrict keywords | Chris Robinson | 2016-07-26 | 1 | -2/+2 |
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* | Rename input_gain to b0 | Chris Robinson | 2016-07-26 | 2 | -4/+3 |
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* | Increase the default thunk array size | Chris Robinson | 2016-07-25 | 1 | -1/+1 |
| | | | | | This value should be enough to hold IDs for most apps without needing to reallocate it, while not being unnecessarily large (4KB). | ||||
* | Make a MAX_AMBI2D_COEFFS macro instead of a magic number | Chris Robinson | 2016-07-17 | 1 | -1/+8 |
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* | Modify bs2b_cross_feed to do multiple samples at once | Chris Robinson | 2016-07-13 | 1 | -36/+5 |
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* | Move the input channel array out of the DirectParams and SendParams | Chris Robinson | 2016-07-13 | 3 | -18/+16 |
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* | Store the voice output buffers separate from the params | Chris Robinson | 2016-07-11 | 2 | -6/+10 |
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* | Reorder some source fields | Chris Robinson | 2016-07-07 | 2 | -24/+27 |
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* | Avoid using memcpy to copy a single struct | Chris Robinson | 2016-07-06 | 2 | -17/+15 |
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* | Remove a slightly outdated comment | Chris Robinson | 2016-07-06 | 1 | -1/+0 |
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* | Remove the VirtOut buffer alias | Chris Robinson | 2016-07-05 | 1 | -6/+3 |
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* | Use separate arrays for UIntMap keys and values | Chris Robinson | 2016-07-04 | 5 | -10/+10 |
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* | Ensure voices has been updated once before mixing them | Chris Robinson | 2016-06-16 | 1 | -1/+8 |
| | | | | | | | | Sometimes the mixer is temporarily prevented from applying updates, when multiple sources need to be updated simultaneously for example, but does not prevent mixing. If the mixer runs during that time and a voice was just started, it would've mixed the voice without any internal properties being set for it. | ||||
* | Rename MaxNoOfSources for consistency | Chris Robinson | 2016-06-08 | 1 | -1/+1 |
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* | Only define 8 HRTF filter states and params for the device | Chris Robinson | 2016-06-04 | 1 | -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 clock | Chris Robinson | 2016-06-03 | 1 | -0/+2 |
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* | Remove some unnecessary volatile keywords | Chris Robinson | 2016-06-03 | 2 | -33/+33 |
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* | Make a function static | Chris Robinson | 2016-06-01 | 2 | -7/+7 |
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* | Use a macro to specify the ambisonic periphonic channel mask | Chris Robinson | 2016-06-01 | 1 | -1/+10 |
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* | Use SSE for applying the HQ B-Format decoder matrices | Chris Robinson | 2016-05-31 | 1 | -0/+3 |
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* | Hold the effectslot map lock while handling it | Chris Robinson | 2016-05-29 | 3 | -2/+38 |
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* | Use a linked list for active effect slots | Chris Robinson | 2016-05-29 | 3 | -37/+34 |
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* | Avoid the mixer lock when getting the plain source offset | Chris Robinson | 2016-05-28 | 1 | -11/+15 |
| | | | | i.e. without the latency | ||||
* | Avoid an explicit mixer lock for getting the source offset and latency | Chris Robinson | 2016-05-28 | 1 | -27/+72 |
| | | | | | The only mixer locking involved is with the backend, as determined by it's ability to get the device clock and latency atomically. | ||||
* | Change the backend getLatency method to return the clock time too | Chris Robinson | 2016-05-28 | 1 | -3/+8 |
| | | | | | | This will also allow backends to better synchronize the tracked clock time with the device output latency, without necessarily needing to lock if the backend API can allow for it. | ||||
* | Use the backend lock for the effectstate's deviceUpdate call | Chris Robinson | 2016-05-27 | 1 | -8/+4 |
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* | Use a specific lock for the backend's stop/reset/play calls | Chris Robinson | 2016-05-27 | 1 | -0/+2 |
| | | | | | This helps protect against the device changing unexpectedly from multiple threads, instead of using the global list/library lock. | ||||
* | Remove a couple unneeded functions | Chris Robinson | 2016-05-25 | 1 | -4/+7 |
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* | Avoid using realloc in a number of places | Chris Robinson | 2016-05-21 | 3 | -13/+27 |
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* | Make the source position calues atomic | Chris Robinson | 2016-05-19 | 2 | -21/+22 |
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* | Avoid redundantly storing distance model settings | Chris Robinson | 2016-05-17 | 5 | -26/+10 |
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* | Use floats for the listener transforms | Chris Robinson | 2016-05-16 | 2 | -26/+1 |
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