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* Convert the Null backend to the ALCbackend styleChris Robinson2013-10-282-61/+101
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* Add a backend factory base typeChris Robinson2013-10-281-0/+43
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* Add a default getLatency to ALCbackendChris Robinson2013-10-281-0/+1
| | | | And make sure the backend is properly deleted.
* Move the lock/unlock methods to the backendChris Robinson2013-10-281-0/+12
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* Add missing header to gitChris Robinson2013-10-271-0/+62
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* Rework threading functionsChris Robinson2013-10-2711-43/+46
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* Set a name for the mixer and recording threadsChris Robinson2013-10-2711-1/+18
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* Use the UNUSED macro in the backendsChris Robinson2013-10-075-59/+25
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* Use a simpler U64 macro to make 64-bit constantsChris Robinson2013-10-061-1/+1
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* Only rest as long as needed in the Null rendererChris Robinson2013-10-051-5/+4
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* Compile using -std=c99 when availableChris Robinson2013-10-033-3/+3
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* Silence some clang warningsChris Robinson2013-06-051-2/+2
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* Recognize DSSPEAKER_5POINT1_SURROUND (and 7POINT1) with DSoundChris Robinson2013-05-221-4/+11
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* Fix a potential leak when mmdevapi fails to openChris Robinson2013-05-191-0/+3
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* Report the actual ALSA error if setting the buffer metrics failsChris Robinson2013-03-181-2/+2
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* Use force_align_arg_pointer for QSAChris Robinson2013-03-151-0/+8
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* Add a QSA backend for QNXChris Robinson2013-03-141-0/+1170
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* Build an appropriate channel map for PulseAudioChris Robinson2013-02-081-2/+28
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* Destroy the ring buffer when closing down a portaudio deviceChris Robinson2013-02-011-0/+3
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* Filter unneeded thread messages with MMDevApiChris Robinson2012-12-241-1/+3
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* Add support for OpenSL ES 1.0.1 on AndroidChris Robinson2012-12-121-1/+8
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* Add support for float32 output to CoreAudioChris Robinson2012-12-031-10/+13
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* Use the correct multiplier for int32 output with CoreAudioChris Robinson2012-12-031-2/+2
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* Lock the device before calling aluHandleDisconnectChris Robinson2012-12-027-0/+26
| | | | | | PulseAudio causes an assert if being relocked inside a callback on the worker thread, where aluHandleDisconnect is called. We can assume it's already locked there, so just make sure the device is locked before being calling it.
* Avoid locking PulseAudio's mainloop when starting and stopping captureChris Robinson2012-11-191-4/+0
| | | | | The mainloop is already locked by the caller, and double-locking interferes with the subsequent wait.
* Use the stored buffer metrics for the mmdevapi buffer sizeChris Robinson2012-11-041-9/+1
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* Add rudimentary latency tracking for mmdevapiChris Robinson2012-11-041-1/+16
| | | | | | | | This won't be as granular as it could be, since it only updates when the wakeup event trips (which may or may not happen more often than OpenAL's mix updates). A more correct method would be to query GetCurrentPadding directly, but that would require sending a message to the processing thread and waiting for a reply, since we can't guarantee COM on the calling thread.
* Rename some struct fields for DirectSoundChris Robinson2012-11-041-48/+48
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* Avoid recreating the DSound primary bufferChris Robinson2012-11-041-1/+1
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* Lock the device using the pulseaudio main loopChris Robinson2012-10-071-8/+15
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* Define WIN32_LEAN_AND_MEAN when including windows.h in alMain.hChris Robinson2012-10-062-0/+6
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* Scale and round NumUpdates when PulseAudio changes the sample rateChris Robinson2012-10-041-7/+7
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* Reset maxlength if PulseAudio updates the playback rateChris Robinson2012-10-021-0/+1
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* Round when calculating NumUpdates from PulseAudio's minreqChris Robinson2012-09-251-7/+8
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* Make sure PulseAudio sets an update size that's a multiple of 4 samples with SSEChris Robinson2012-09-201-2/+5
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* Don't include alu.h in alMain.hChris Robinson2012-09-1414-24/+23
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* Free the global ALSA config after opening capture devices tooChris Robinson2012-09-021-1/+4
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* alsa: Free the global config cache to remove a lot of reported leaksLauri Kasanen2012-09-021-0/+6
| | | | Signed-off-by: Lauri Kasanen <[email protected]>
* Print the error when reporting snd_pcm_delay failureChris Robinson2012-08-271-1/+1
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* Add a macro to help make a 64-bit valueChris Robinson2012-08-191-1/+1
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* Use a default method to provide a dummy latencyChris Robinson2012-08-1912-98/+12
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* Hold the device lock with PulseAudio between mixing and writingChris Robinson2012-08-181-0/+2
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* Hold the device lock with ALSA between mixing and writingChris Robinson2012-08-181-0/+4
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* Use non-blocking access for ALSA playbackChris Robinson2012-08-181-17/+37
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* Move the device lock into the backend function tableChris Robinson2012-08-1814-0/+28
| | | | | For backend-specific implementations: this should hold the audio mixer loop for playback devices, and provide recursive mutex behavior.
* Add a device method to retrieve the active latencyChris Robinson2012-08-1714-14/+154
| | | | | This is effectively the time until the next update will be heard by the user, or the closest approximation thereof, in nanoseconds.
* Avoid dynamic handling of libsndioChris Robinson2012-08-161-81/+0
| | | | This makes it safer and easier to handle as OpenBSD updates it
* Don't unlock the pulseaudio mainloop when mixingChris Robinson2012-08-121-2/+0
| | | | | | Lock contention could cause a lengthy delay between mixing and writing the audio. There shouldn't be any risk of dead-locking as the device lock should never be held when the mainloop gets locked.
* Add missing includeChris Robinson2012-06-191-0/+1
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* Use a proper method to wait for a PulseAudio operationChris Robinson2012-06-191-18/+19
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