Added new initialiser for AEAudioUnitFilter with "useDefaultInputFormat" option to workaround issues with audio units (like the 3D mixer) that misidentify audio format compatibility
* Added currentTime property to AERecorder which is being calculated on every audio callback
Merge pull request #44 from JlUgia/jose/added_current_time_to_AERecorder

* Added currentTime property to AERecorder which is being calculated on ...
Always use 16 bits per channel in AEAudioFileWriter when writing to uncompressed PCM formats
AERecorder: separate startRecording method
Minor TPCircularBuffer update

- Added a couple of extra sanity checks for ABL utilities
- Use const keyword for AudioStreamBasicDescription parameters
Added missing 4096 kAudioUnitProperty_MaximumFramesPerSlice assignment for main output unit
Switched order of audio unit setup calls for AEAudioUnitFilter/Channel

See http://forum.theamazingaudioengine.com/discussion/104/aeaudiounitfilter-couldnt-setup-converter-audio-unit-error/p1 for discussion
Merge pull request #45 from lijon/aerecorder_separate_prepare_and_start

AERecorder: separate startRecording method
Adjusted AERecorder workflow
Updated podspec
Fixed a problem with replacing the IO unit (e.g. when switching voice processing on/off)
Move kAudioUnitProperty_MaximumFramesPerSlice assignment on audio unit channel/filter earlier to avoid initialisation error
Minor tweak to use NSString * const for notifications
Moved an audio unit interaction to main thread to avoid strange deadlock issue with Inter-App Audio use
Changed Audiobus input support to use Audiobus's live mode for simplicity

Previously, AEAudioController's audiobusInputPort was used to receive audio from Audiobus via the "lossless" stream mode. This ensured audio received was identical to audio sent from the Audiobus source, but at the cost of increased latency and unpredictable audio receive intervals, which increased the complexity of receiving apps due to the necessity of dealing with buffer under runs by using a separate "live" stream for monitoring (using AEPlaythroughBuffer or manually).

This change causes AEAudioController to only use Audiobus' "live" mode for input, removing the necessity of dealing with the particulars of "lossless" mode with little tradeoff, as live mode yields excellent results.
AEMixerBuffer: Don't set the mDataByteSize values of buffer lists on return
AEMixerBuffer and AEFloatConverter: Preserve AudioBufferList mDataByteSize fields on return
Allowed time stretching filters by keeping per-channel timeStamps.
Merge pull request #50 from psobot/timestretch_fix

Allowed time stretching filters by keeping per-channel timeStamps.