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* Use an atomic instead of volatile to tell a thread to quitChris Robinson2018-02-011-6/+5
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* Use semaphores to signal for more samples with JACK and OpenSLChris Robinson2018-02-012-54/+16
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* Don't bother with an explicit stop backend methodChris Robinson2018-01-2917-340/+208
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* Call the backend close method in the destructorChris Robinson2018-01-2813-79/+125
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* Avoid using alloca in the CoreAudio backendChris Robinson2018-01-161-12/+10
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* Properly cast some parameters for MSVCChris Robinson2018-01-161-5/+5
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* Use GetProcBinary to get the executable name for PulseAudioChris Robinson2018-01-131-4/+7
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* Make a couple functions inlineChris Robinson2018-01-122-0/+9
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* Use atomic variables in place of volatileChris Robinson2018-01-111-16/+17
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* Define a function where it's usedChris Robinson2018-01-111-0/+7
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* Fix up some types for MSVCChris Robinson2018-01-112-2/+2
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* Move the ringbuffer declarations to a separate headerChris Robinson2018-01-119-1/+10
| | | | And rename alcRing.c to ringbuffer.c for consistency.
* Move the config function declarations to their own headerChris Robinson2018-01-117-0/+7
| | | | And rename alcConfig.c to alconfig.c for consistency.
* Restore the original JACK message callback when possibleChris Robinson2017-09-231-1/+9
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* Downgrade some ERRs to TRACEsChris Robinson2017-07-231-2/+2
| | | | | These don't exist outside OSSv4, e.g. with OSS/Free, padsp, or aoss, so no need to be concerned.
* Make sure OSS device files exist before adding themChris Robinson2017-07-231-2/+10
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* Set the float PCM GUID for wave files only when outputting floatChris Robinson2017-07-151-2/+3
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* Store the QSA backend's ExtraData in the wrapper structChris Robinson2017-06-291-118/+121
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* Add casts to silence MSVCChris Robinson2017-06-291-1/+1
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* Trace the message name in the message handler loopChris Robinson2017-06-261-1/+14
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* Clean up some messy rounding codeChris Robinson2017-06-261-7/+9
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* Ensure the mmdevapi capture buffer is at least 100msChris Robinson2017-06-261-0/+4
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* Improve traces for the mmdevapi capture conversionsChris Robinson2017-06-231-10/+8
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* Trace the capture converter formats for mmdevapiChris Robinson2017-06-221-0/+6
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-183-173/+169
| | | | | | | | | | I say "convert" because it takes the lazy way and essentially just embeds the wrappers into the backend. It's done this way because I lack the means to check any changes, even syntactically. This also means the device's ExtraData field is still needed. However, this does mean all the backends are now using the new API. Code related to the old interface can now be removed.
* Limit device buffer based on PulseAudio's tlengthChris Robinson2017-06-151-10/+9
| | | | | | Unfortunately PulseAudio has a habit of limiting tlength, and trying to calculate the device's buffer length to write regardless of tlength could result in some amount always being writable.
* Add a config option to specify custom ALSA devicesChris Robinson2017-05-231-5/+40
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* Add an env var to specify a default pulse deviceChris Robinson2017-05-161-0/+7
| | | | | Some apps don't allow selecting an audio device, and due to problems with KDE, PulseAudio isn't allowed to move the stream after being created by default.
* Remove unnecessary functions in the JACK backendChris Robinson2017-04-171-19/+8
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* Implement capture support in the OpenSL backendChris Robinson2017-04-151-1/+379
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* Use separate atomic macros for pointersChris Robinson2017-04-143-5/+5
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* Store the ambisonic order separate from the channel enumChris Robinson2017-04-1214-167/+202
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* Use the converters to enable mmdevapi captureChris Robinson2017-04-111-42/+171
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* Convert the CoreAudio backend to the updated backend APIChris Robinson2017-04-092-178/+283
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* Clean up some formattingChris Robinson2017-04-081-5/+5
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* Try to write the full configured buffer length with PulseAudioChris Robinson2017-04-081-15/+37
| | | | | This basically ignores tlength even if it's smaller than what was requested. It keeps up-to-date with minreq changes too now, in case that happens.
* Rename al_string_* functions to alstr_*Chris Robinson2017-04-0416-133/+133
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* Don't use the mutex in the base getClockLatency implementationChris Robinson2017-02-281-3/+8
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* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-271-1/+2
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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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* Print warnings about missing libraries and functionsChris Robinson2017-02-213-2/+24
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* Always lock the device backend before calling aluMixDataChris Robinson2017-02-189-48/+59
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* Return some device latency by defaultChris Robinson2017-02-181-2/+7
| | | | | | | A device will never have 0 latency. OpenAL Soft itself uses a sample buffer length of UpdateSize*NumUpdates, and during playback will have about (NumUpdates-1) periods filled, more or less. Without a more accurate measurement from the playback system, this is better than reporting 0.
* Use select() to wait for audio with OSS and SolarisChris Robinson2017-02-182-85/+137
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* Use the correct IID for the opensl buffer queueChris Robinson2017-02-071-2/+3
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* Convert the OpenSL backend to the new backend APIChris Robinson2017-02-072-176/+433
| | | | | This also removes the buffer queue callback's call to aluMixData, which could potentially block on a mutex.
* Fix for NULL JNIEnvChris Robinson2017-02-051-5/+3
| | | | Which can happen with native-only apps
* Call getSystemService as a non-static functionChris Robinson2017-02-031-4/+4
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* Android's AudioManager.getProperty(String) returns a StringChris Robinson2017-02-021-1/+1
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