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* Accept a "narrow" layout for 7.1 with mmdevapiChris Robinson2015-01-151-2/+3
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* Fix the flags typeChris Robinson2015-01-111-1/+2
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* Update a commentChris Robinson2014-12-241-2/+2
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* Don't update the channel format if all JACK ports were allocatedChris Robinson2014-12-231-9/+11
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* Try the device-specified channel configuration firstChris Robinson2014-12-231-3/+15
| | | | | This does basically no checking that the channel config is correct, but should be good enough for when non-stereo modes are requested.
* Use the custom ringbuffer instead of JACK'sChris Robinson2014-12-231-41/+20
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* Don't set unused pointers to NULL, don't try to write 0 samplesChris Robinson2014-12-231-7/+8
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* At least measure the ringbuffer size for JACK's latencyChris Robinson2014-12-231-1/+17
| | | | | | We should also add the port latency, but there's currently no way to ensure it's synchronized with the ringbuffer (the ringbuffer will update before the port latency gets updated).
* Add an option to prevent spawning a JACK serverChris Robinson2014-12-221-2/+6
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* Add an option for an increased ringbuffer size with JACKChris Robinson2014-12-221-2/+10
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* Fill out the JACK playback backendChris Robinson2014-12-221-15/+326
| | | | | | | | | | | | | | | | | | | | | | | | A few notes about it: The OpenAL device's requested buffer metrics are ignored, and instead the device will keep one JACK-sized buffer's worth of audio prepared for JACK's next process request. Output is restricted to 32-bit float stereo. Part of this is because JACK requires a buffer size that's a power of 2 (measured in samples), and the ringbuffer requires a buffer size that's a power of 2 (measured in bytes). A channel count of 6 (5.1) or 7 (6.1) will not work without causing a sample to split over the edge of the ringbuffer. Additioanlly, JACK doesn't provide information about what channel configuration a device or set of ports has, so there's no way to know what ports 3 and up map to (even the first two ports are unknown, but assuming stereo seems to work well enough). There is no device latency measurement (for AL_SOFT_source_latency) due to the non-atomicity between the ringbuffer's read size and port latency. A method is needed to ensure the ringbuffer's read size and port latency can both be measured between the end of a JACK update cycle (when the port latency has been updated with the newly-retrieved audio) and the following ringbuffer read update.
* Add a skeleton backend for JACKChris Robinson2014-12-212-0/+262
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* Trace lparam and wparam message valuesChris Robinson2014-12-211-1/+1
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* Check the PROPVARIANT type before useChris Robinson2014-12-211-2/+8
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* Use VECTOR_FOR_EACH instead of a manual loopChris Robinson2014-12-211-9/+7
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* Use a macro for the record thread nameChris Robinson2014-12-213-3/+3
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* Support capture with mmdevapiChris Robinson2014-12-191-1/+551
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* Set a couple pulse callbacks to NULL during reset, and minor cleanupsChris Robinson2014-12-171-9/+9
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* Don't kill pulseaudio's mixer thread if it's already killedChris Robinson2014-12-171-1/+1
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* Check mmdevice endpoint for being headphonesChris Robinson2014-11-271-1/+31
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* Avoid unnecessary uses of ALCdevice_Lock and ALCdevice_UnlockChris Robinson2014-11-271-8/+8
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* Set headphones when dsound reports headphonesChris Robinson2014-11-271-1/+3
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* Support B-Format output with the wave file writerChris Robinson2014-11-257-1/+33
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* Require at least pulse client 0.9.16Chris Robinson2014-11-241-48/+8
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* Trace PulseAudio's active portChris Robinson2014-11-241-2/+5
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* Set headphones when pulse reports using the headphones portChris Robinson2014-11-231-0/+3
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* Don't force UpdateSize to a multiple of 4 after buffer metrics were setChris Robinson2014-11-171-2/+0
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* Always get the pulseaudio sink info on device resetChris Robinson2014-11-171-16/+14
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* Fix 5.1 surround soundChris Robinson2014-11-077-33/+38
| | | | | | | | | | | | | Apparently, 5.1 surround sound is supposed to use the "side" channels, not the back channels, and we've been wrong this whole time. That means the "5.1 Side" is actually the correct 5.1 setup, and using the back channels is anomalous. Additionally, this means the 5.1 buffer format should also use the the side channels instead of the back channels. A final note: the 5.1 mixing coefficients are changed so both use the original 5.1 surround sound set (with the surround channels at +/-110 degrees). So the only difference now between 5.1 "side" and 5.1 "back" is the channel labels.
* 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 Robinson2014-09-087-37/+9
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* Convert the winmm backend to the new backend APIChris Robinson2014-09-082-285/+381
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* Only pass nano seconds to al_nssleepChris Robinson2014-09-084-5/+5
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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.
* Check the given CoreAudio capture device nameChris Robinson2014-08-261-0/+5
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* Convert the wave writer backend to the new APIChris Robinson2014-08-242-111/+181
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* Update COPYING to the latest ↵François Cami2014-08-1814-28/+28
| | | | https://www.gnu.org/licenses/old-licenses/lgpl-2.0.txt to fix the FSF' address Fix the FSF' address in the source
* Use VECTOR_FIND_IF and VECTOR_FOR_EACH instead of manual loopsChris Robinson2014-08-093-105/+56
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* Use the default input device for portaudio's default capture deviceChris Robinson2014-08-081-1/+4
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* Use pulseaudio's write callback to signal a mixer proc wakeupChris Robinson2014-07-261-3/+11
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* Don't require pre-declaring vector typesChris Robinson2014-07-065-12/+8
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* Don't print an ERR if pulse fails to get latency info due to no dataChris Robinson2014-05-261-3/+8
| | | | It just means it was called too quickly after starting.
* Add a needed header for the CoreAudio backendChris Robinson2014-05-221-0/+1
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* Initialize a pointer to NULL instead of a stringChris Robinson2014-05-151-2/+1
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* Initialize some variables when declaring themChris Robinson2014-05-151-4/+2
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* Don't try to read the internal capture buffer into itselfChris Robinson2014-05-141-6/+9
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* Avoid freeing an in-use capture bufferChris Robinson2014-05-141-5/+3
| | | | | When stopping, ALSA may capture into its own storage buffer. Do not free the storage buffer if it first reads from it.
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