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* Check that a source is actually playing before setting pausedChris Robinson2017-03-071-28/+35
| | | | | Also slightly refactor setting playing state when the device is disconnected or there's no buffers to play.
* Store the channel count and sample size in the voiceChris Robinson2017-03-073-17/+13
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* Don't modify the source state in the mixerChris Robinson2017-03-072-8/+25
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* Remove an unused functionChris Robinson2017-03-072-7/+0
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* Merge pull request #97 from adrianbroher/ci-backendskcat2017-03-072-6/+72
|\ | | | | Enable and enforce dependencies on CI services.
| * Only download and strip Android NDK when not cachedMarcel Metz2017-03-061-10/+12
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| * Use TravisCI cache to store stripped Android NDK toolchainMarcel Metz2017-03-061-0/+3
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| * Unpack only required files from Android NDKMarcel Metz2017-03-061-1/+8
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| * Configure CMake to require available backends on CI servicesMarcel Metz2017-03-062-3/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Configure CMake to require the installed backend libraries. This should help to find build system regressions. On TravisCI with Linux this requires the ALSA, PulseAudio, PortAudio, OSS and JACK backend. On TravisCI with Android cross compile Linux this requires the OpenSL backend. On TravisCI with MacOSX this requires the CoreAudio backend. ON AppVeyor with Windows this requires the WinMM, DSound and MMDevAPI backend.
| * Add Android cross-compile to TravisCI test matrixMarcel Metz2017-03-061-2/+27
| | | | | | | | The test entry adds the ability to test the OpenSLES backend.
| * Explicit declare test matrix for TravisCIMarcel Metz2017-03-061-5/+6
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| * Delete Xamarin.Common.targets on AppVeyorMarcel Metz2017-03-061-0/+5
| | | | | | | | | | | | | | This is a workaround for a Xamarin build script bug specific to AppVeyor. For more details see: http://help.appveyor.com/discussions/problems/4569
| * Install dependencies on TravisCI to enable more featuresMarcel Metz2017-03-051-0/+14
| | | | | | | | | | | | | | Install Ubuntu development packages for PulseAudio, PortAudio, ALSA and JACK to enable the building of most Linux backends on TravisCI. Intall Ubuntu development packages for Qt5 to enable `alsoft-config`.
* | Make the voice's source pointer atomicChris Robinson2017-03-054-15/+17
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* | Update alffplay for newer ffmpeg and convert to C++Chris Robinson2017-03-053-1585/+1558
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* | Remove ex-common and test-common static libsChris Robinson2017-03-041-21/+14
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* | Remove unnecessary wrappers around SDL_soundChris Robinson2017-03-0410-629/+253
| | | | | | | | Also remove wrappers for the now-unsupported buffer_samples extension.
* | Use the LINK_FLAGS property instead of abusing libs for flagsChris Robinson2017-03-041-4/+8
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* Merge pull request #95 from adrianbroher/export-configkcat2017-03-042-56/+61
|\ | | | | Enable exporting of CMake import targets
| * Use Ubuntu 14.04 in TravisCI to get a less antique CMake versionMarcel Metz2017-03-051-0/+1
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| * Export cmake import targets for project build treeMarcel Metz2017-03-051-0/+3
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| * Export cmake import targets for project install treeMarcel Metz2017-03-051-1/+6
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| * Rename logical CMake target `openal` to `OpenAL`Marcel Metz2017-03-051-35/+37
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| * Compile `common` library within dependent targetsMarcel Metz2017-03-041-22/+15
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| * Make logical target name `openal` uniform accross all platformsMarcel Metz2017-03-041-41/+42
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* | Add a boolean to specify if a voice should be playingChris Robinson2017-03-023-28/+53
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* | Stretch out some GUI elements for the decoder configurationsChris Robinson2017-03-011-12/+12
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* | Increment MixCount in UpdateClockBaseChris Robinson2017-02-281-1/+4
| | | | | | | | This is to protect clocktime reads since the backend lock won't protect it.
* | Dynamically allocate the channel delay buffersChris Robinson2017-02-284-8/+66
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* | Remove unused function declarationsChris Robinson2017-02-281-3/+0
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* | Remove an unneeded functionChris Robinson2017-02-283-24/+7
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* | Start a ALC_SOFT_loopback2 extensionChris Robinson2017-02-282-55/+218
|/ | | | | | | | | | | | | | | | | | | | | | 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 a variable counter for an array size limitChris Robinson2017-02-281-21/+13
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* Don't use the mutex in the base getClockLatency implementationChris Robinson2017-02-281-3/+8
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* Print WARNs when a device or context error is generatedChris Robinson2017-02-272-0/+6
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* Avoid standard malloc for buffer queue entriesChris Robinson2017-02-272-7/+10
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* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-274-17/+38
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* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-275-127/+137
| | | | | | | | | | | | | | This has a couple behavioral changes. First and biggest is that querying AL_BUFFERS_PROCESSED from a source will always return all buffers processed when in an AL_STOPPED state. Previously all buffers would be set as processed when first becoming stopped, but newly queued buffers would *not* be indicated as processed. That old behavior was not compliant with the spec, which unequivocally states "On a source in the AL_STOPPED state, all buffers are processed." Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return 0. Previously it would return the current "active" buffer in the queue, but there's no basis for that in the spec.
* Ensure a non-playing or -paused source does not use a mixing voiceChris Robinson2017-02-254-46/+62
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* Set CMP0020 for QtChris Robinson2017-02-251-0/+3
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* Improve handling of source state readsChris Robinson2017-02-242-72/+89
| | | | | | | This avoids using seq_cst for loading the source state when either inside the mixer, or otherwise protected from inconsistencies with async updates. It also fixes potential race conditions with getting the source offset just as a source stops.
* Remove an unused functionChris Robinson2017-02-232-6/+0
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* Remove CalcXYZCoeffs and inline CalcAngleCoeffsChris Robinson2017-02-236-34/+22
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* Alter how panpot/pair-wise panning worksChris Robinson2017-02-233-32/+32
| | | | | | | | | | | | | | | | | | | This change allows pair-wise panning to mostly go through the normal ambisonic panning methods, with one special-case. First, a term is added to the stereo decoder matrix's X coefficient so that a centered sound is reduced by -3dB on each output channel. Panning in front creates a similar gain response to the typical L = sqrt(1-pan) R = sqrt(pan) for pan = [0,1]. Panning behind the listener can reduce (up to) an additional -10dB, creating a audible difference between front and back sounds as if simulating head obstruction. Secondly, as a special-case, the source positions are warped when calculating the ambisonic coefficients so that full left panning is reached at -30 degrees and full right at +30 degrees. This is to retain the expected 60-degree stereo width. This warping does not apply to B-Format buffer input, although it otherwise has the same gain responses.
* Rename stereo-panning option to stereo-encodingChris Robinson2017-02-225-37/+40
| | | | Also rename the 'paired' value to 'panpot', and make it the default.
* Limit filter gains to -24dBChris Robinson2017-02-225-40/+34
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* Update ChangeLogChris Robinson2017-02-221-0/+8
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* Reduce the default period count to 3Chris Robinson2017-02-223-3/+3
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* Don't remove a period from the OSS bufferChris Robinson2017-02-221-11/+4
| | | | | Since we're now waiting for space to be available before mixing, the mixing buffer isn't adding another period.
* Fix OpenSL latency calculationChris Robinson2017-02-221-2/+2
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