Psychtoolbox>PsychPortAudio.{mex*} subfunction

[audiodata absrecposition overflow cstarttime] = PsychPortAudio(‘GetAudioData’, pahandle [, amountToAllocateSecs][, minimumAmountToReturnSecs][, maximumAmountToReturnSecs][, singleType=0]);

Retrieve captured audio data from a audio device. ‘pahandle’ is the handle of
the device whose data is to be retrieved.
‘audiodata’ is a matrix with audio data in floating point format. Each row of
the matrix returns one sound channel, each column one sample for each channel.
Returned samples are in range -1.0 to +1.0, with 0.0 for silence. This is
intentionally a very restricted interface. For lowest latency and best timing we
want you to accept audio data exactly at the optimal format and sample rate, so
the driver can save computation time and latency for expensive sample rate
conversion, sample format conversion, and bounds checking/clipping.
You must call this function once before start of capture operations to allocate
an internal buffer that stores captured audio data inbetween your periodic
calls. Provide ‘amountToAllocateSecs’ as requested buffersize in seconds. After
start of capture you must call this function periodically at least every
‘amountToAllocateSecs’ seconds to drain the internal buffer into your matrix
‘audiodata’. If you fail to call the function frequently enough, sound data will
get lost!
‘minimumAmountToReturnSecs’ optional minimum amount of recorded data to return
at each call. The driver will only return control to your script when it was
able to collect at least that amount of seconds of sound data - or if the
capture engine was stopped. If you don’t set this parameter, the driver will
return immediately, giving you whatever amount of sound data was available -
including an empty matrix if nothing was available.
‘maximumAmountToReturnSecs’ allows you to optionally restrict the amount of
returned sound data to a specific duration in seconds. By default, you’ll get
whatever is available.
If you provide both, ‘minimumAmountToReturnSecs’ and ‘maximumAmountToReturnSecs’
and set them to equal values (but significantly lower than the
‘amountToAllocateSecs’ buffersize!!) then you’ll always get an ‘audiodata’
matrix back that is of a fixed size. This may be convenient for postprocessing
in the scripting language. It may also reduce or avoid memory fragmentation…
‘singleType’ if set to 1 will return a sound data matrix of single() type
instead of double() type. By default, double() type is returned. single() type
matrices only consume half as much memory as double() type matrices, without
loss of audio precision for up to 24-Bit ADC hardware.

Optional return arguments other than ‘audiodata’:

‘absrecposition’ is the absolute position (in samples) of the first column in
the returned data matrix, assuming that sample zero was the very first recorded
sample in this session. The count is reset each time you start a new capture
session via call to PsychPortAudio(‘Start’).
Each call to this function will return a new chunk of recorded sound data. The
‘absrecposition’ provides you with absolute matrix column indices to stitch
together the results of all calls into one seamless recording if you want.
‘overflow’ if this flag is zero then everything went fine. If it is one then you
didn’t manage to call this function frequent enough, the capacity of the
internal recording buffer was exceeded and therefore you lost captured sound
data, i.e., there is a gap in your recording. When initially allocating the
internal buffer, make sure to allocate it big enough so it is able to easily
store all recorded data inbetween your calls to ‘GetAudioData’. Example: You
expect to call this routine once every second in your trial loop, then allocate
a sound buffer of at least 2 seconds for some security headroom. If you know
that the recording time of each recording has an upper bound then you can
allocate an internal buffer of sufficient size and fetch the buffer all at once
at the end of a recording.
‘cstarttime’ this is an estimate of the system time (in seconds) when the very
first sample of this recording was captured by the sound input of your hardware.
This is only to be trusted down to the millisecond level after former careful
calibration of your setup!

###See also: Open GetDeviceSettings