* Implement `pause` and `play` for ALSA backend
This commit also ensures that the Voice is initially paused when
returned to remain consistent with the rest of the CPAL backends.
Related to #175.
* Remove ineffective pause from end of build_voice method
* ALSA - Change `is_paused` flag from `AtomicBool` to `bool`
* Add pause and play ALSA addition to CHANGELOG
* Use the js! macro from stdweb
* Rework the Buffer::finish method
* Use references from stdweb
* Fix emscripten warnings
* Rework the run() method to use stdweb
* Adjust timings
* Add entry in CHANGELOG
* Rework the API to not use futures anymore
* Add some comments
* Update the MacOS backend
* Restore the null implementation
* Add an emscripten backend
* Remove erroneously added feature
* Fix to_f32 formula
* [WIP] Alsa backend
* Alsa backend compiling
* Working ALSA backend
* Fix tests
* Move WASAPI endpoint to endpoint module
* Fix WASAPI warnings
* Rework the WASAPI backend
* Check overflows for voice ID
* Add comments and minor fixes to WASAPI backend
* Add a changelog
iOS provides three I/O (input/output) units. The vast majority of audio-unit applications use the Remote I/O unit, which connects to input and output audio hardware and provides low-latency access to individual incoming and outgoing audio sample values. For VoIP apps, the Voice-Processing I/O unit extends the Remote I/O unit by adding acoustic echo cancelation and other features. To send audio back to your application rather than to output audio hardware, use the Generic Output unit.
See https://developer.apple.com/library/content/documentation/MusicAudio/Conceptual/AudioUnitHostingGuide_iOS/UsingSpecificAudioUnits/UsingSpecificAudioUnits.html
snd_pcm_sw_params_set_avail_min was being hardcoded to 4096, which
seems to be problematic for lower sample rates. This update sets
the value to the buffer size as supplied by snd_pcm_get_params(),
which is what alsa own sample code does.
This should fix https://github.com/tomaka/cpal/issues/142