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* Make ComputeAngleGains use ComputeDirectionalGainsChris Robinson2014-10-027-291/+176
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* Use helpers to set the gain step valuesChris Robinson2014-10-021-142/+73
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* Use VECTOR_FIND_IF instead of manual loopsChris Robinson2014-09-301-42/+38
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* Add a cast for MSVCChris Robinson2014-09-301-1/+1
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* Copy the null terminator from the string instead of appending itChris Robinson2014-09-301-4/+1
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* Use size_t for the vector size and capacityChris Robinson2014-09-304-30/+23
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* Use an ambisonics-based panning methodChris Robinson2014-09-302-130/+121
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For mono sources, third-order ambisonics is utilized to generate panning gains. The general idea is that a panned mono sound can be encoded into b-format ambisonics as: w[i] = sample[i] * 0.7071; x[i] = sample[i] * dir[0]; y[i] = sample[i] * dir[1]; ... and subsequently rendered using: output[chan][i] = w[i] * w_coeffs[chan] + x[i] * x_coeffs[chan] + y[i] * y_coeffs[chan] + ...; By reordering the math, channel gains can be generated by doing: gain[chan] = 0.7071 * w_coeffs[chan] + dir[0] * x_coeffs[chan] + dir[1] * y_coeffs[chan] + ...; which then get applied as normal: output[chan][i] = sample[i] * gain[chan]; One of the reasons to use ambisonics for panning is that it provides arguably better reproduction for sounds emanating from between two speakers. As well, this makes it easier to pan in all 3 dimensions, with for instance a "3D7.1" or 8-channel cube speaker configuration by simply providing the necessary coefficients (this will need some work since some methods still use angle-based panpot, particularly multi-channel sources). Unfortunately, the math to reliably generate the coefficients for a given speaker configuration is too costly to do at run-time. They have to be pre- generated based on a pre-specified speaker arangement, which means the config options for tweaking speaker angles are no longer supportable. Eventually I hope to provide config options for custom coefficients, which can either be generated and written in manually, or via alsoft-config from user-specified speaker positions. The current default set of coefficients were generated using the MATLAB scripts (compatible with GNU Octave) from the excellent Ambisonic Decoder Toolbox, at https://bitbucket.org/ambidecodertoolbox/adt/
* Combine some fields into a structChris Robinson2014-09-108-132/+127
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* Invert the ChannelOffsets arrayChris Robinson2014-09-102-55/+58
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* Use a wave file channel mask based on the actual formatChris Robinson2014-09-101-14/+12
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* Remove some unnecessary config optionsChris Robinson2014-09-081-18/+8
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* Use a vector instead of a manual dynamic arrayChris Robinson2014-09-081-150/+90
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* Don't modify a capture device's formatChris Robinson2014-09-081-168/+7
| | | | | | OpenAL's capture API guarantees the application gets the format requested, or else the device will fail to open. The only valid change is that the capture buffer can be larger than requested.
* Remove the GetLatency method from the old BackendFuncsChris Robinson2014-09-088-43/+10
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* Convert the winmm backend to the new backend APIChris Robinson2014-09-083-286/+382
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* Only pass nano seconds to al_nssleepChris Robinson2014-09-084-5/+5
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* Use a standard pointer-sized integer typeChris Robinson2014-09-041-1/+1
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* Make the fontsound's buffer and link fields atomicChris Robinson2014-09-031-2/+4
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* Fix Neon mixer definitionChris Robinson2014-08-311-2/+2
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* Use al_calloc/al_free to allocate contexts and voicesChris Robinson2014-08-301-6/+6
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* Setup the HRTF format before tracing the pre-reset formatChris Robinson2014-08-291-17/+17
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* Check mmdevapi device ids to match the default deviceChris Robinson2014-08-281-19/+36
| | | | | Seems Windows can return different IMMDevice object pointers for the same endpoint.
* Return the correct default capture device nameChris Robinson2014-08-261-1/+1
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* Check the given CoreAudio capture device nameChris Robinson2014-08-261-0/+5
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* Remove a couple unnecessary typedefsChris Robinson2014-08-241-2/+2
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* Convert the wave writer backend to the new APIChris Robinson2014-08-243-112/+182
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* Rename activesource to voiceChris Robinson2014-08-213-158/+157
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* Use an array of objects for active sources instead of pointersChris Robinson2014-08-212-18/+11
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* Use a NULL source for inactive activesourcesChris Robinson2014-08-213-16/+19
| | | | Also only access the activesource's source field once per update.
* Support brace-enclosed environment variable namesChris Robinson2014-08-191-0/+9
| | | | | | | This makes it possible to append alpha-numeric characters directly to an environment variable value, e.g. ${FOO}bar will use "FOO" as the variable name and keep the "bar" as-is, whereas $FOObar will take "FOObar" as the variable name.
* Update COPYING to the latest ↵François Cami2014-08-1834-68/+68
| | | | https://www.gnu.org/licenses/old-licenses/lgpl-2.0.txt to fix the FSF' address Fix the FSF' address in the source
* ALC_SOFT_pause_device is finishedChris Robinson2014-08-121-1/+1
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* Assume SSE is available if building with support and no run-time checkingChris Robinson2014-08-111-0/+12
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* Try the __cpuid intrinsic if GCC's __get_cpuid isn't availableChris Robinson2014-08-111-10/+42
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* Check for GCC's __get_cpuid before using itChris Robinson2014-08-111-2/+2
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* Simplify some vector size range checksChris Robinson2014-08-102-17/+19
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* Give NULL for null-vector iteratorsChris Robinson2014-08-102-3/+5
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* Pass pointer-to-vector types as char* instead of void*Chris Robinson2014-08-102-17/+16
| | | | | C aliasing rules only allow char* to alias an otherwise-incompatible type, rather than void*.
* Use VECTOR_FIND_IF and VECTOR_FOR_EACH instead of manual loopsChris Robinson2014-08-094-106/+57
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* Use the default input device for portaudio's default capture deviceChris Robinson2014-08-081-1/+4
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* Disable the autowah effectChris Robinson2014-08-061-0/+4
| | | | | | | | There's apparently some issues with it causing noise or killing the output. It might be due to the per-sample changes being too harsh for the filter to keep up with, but it's not something I can take care of in time for release. This commit should be reverted after release when work on fixing it can resume.
* Make the DYNLOAD LoadFSynth function non-inlineChris Robinson2014-08-051-1/+1
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* Load fluidsynth dynamically when possibleChris Robinson2014-08-051-2/+107
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* Use an ATOMIC_INIT macro instead of ATOMIC_LOAD_UNSAFEChris Robinson2014-08-031-9/+9
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* Use atomics for the device and context list headsChris Robinson2014-08-012-61/+98
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* Make the source's buffer queue head and current queue item atomicChris Robinson2014-07-312-16/+17
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* Rename ATOMIC_COMPARE_EXCHANGE to ATOMIC_COMPARE_EXCHANGE_STRONGChris Robinson2014-07-311-1/+1
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* Use pulseaudio's write callback to signal a mixer proc wakeupChris Robinson2014-07-261-3/+11
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* Explicitly pass the address of atomics and parameters that can be modifiedChris Robinson2014-07-262-22/+22
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* Support C11 atomicsChris Robinson2014-07-231-1/+3
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