No comment. Sorry.

	initialization
	initialize initializeRecordingState

	object fileIn

	primitives
	primGetActualRecordingSampleRate primRecordSamplesInto:startingAt: primSetRecordLevel: primStartRecordingDesiredSampleRate:stereo:semaIndex: primStopRecording

	private
	allocateBuffer emitBuffer: emitPartialBuffer meterFrom:count:in: recordLoop samplesPerFrame

	recording controls
	clearRecordedSound pause playback resumeRecording startRecording stopRecording

	results
	condensedSamples condensedStereoSound recordedSound soundSegments

	trimming
	copyFrom:to:normalize:dcOffset: copyTo:from:to:from:startingAt:normalize:dcOffset: endPlace firstSampleOverThreshold:dcOffset:startingAt: normalizeFactorFor:min:max:dcOffset: place:plus: scanForEndThreshold:dcOffset:minLull:startingAt: scanForStartThreshold:dcOffset:minDur:startingAt: segmentsAbove:normalizedVolume: suppressSilence trim:normalizedVolume:

	class initialization
	initialize

	instance creation
	new

	accessing	top
	codec:
	desiredSampleRate: use of this method indicates a strong desire for the specified rate, even if the OS/hardware are not cooperative
	isActive Return true if I have a recordProcess running.
	isPaused Return true if recording is paused.
	meterLevel Return the meter level, an integer in the range [0..100] where zero is silence and 100 represents the maximum signal level possible without clipping.
	recordLevel
	recordLevel: Set the desired recording level to the given value in the range 0.0 to 1.0, where 0.0 is the lowest recording level and 1.0 is the maximum. Do nothing if the sound input hardware does not support changing the recording level.
	samplingRate
	samplingRate:

	initialization	top
	initialize SoundRecorder new
	initializeRecordingState

object fileIn

top

	primitives	top
	primGetActualRecordingSampleRate Return the actual sample rate being used for recording. This primitive fails unless sound recording is currently in progress.
	primRecordSamplesInto:startingAt: Record a sequence of 16-bit sound samples into the given array starting at the given sample index. Return the number of samples recorded, which may be zero if no samples are currently available.
	primSetRecordLevel: Set the desired recording level to the given value in the range 0-1000, where 0 is the lowest recording level and 1000 is the maximum. Do nothing if the sound input hardware does not support changing the recording level.
	primStartRecordingDesiredSampleRate:stereo:semaIndex: Start sound recording with the given stereo setting. Use a sampling rate as close to the desired rate as the underlying platform will support. If the given semaphore index is > 0, it is taken to be the index of a Semaphore in the external objects array to be signalled every time a recording buffer is filled.
	primStopRecording Stop sound recording. Does nothing if recording is not currently in progress. Do not fail if plugin is not available

	private	top
	allocateBuffer Allocate a new buffer and reset nextIndex.
	emitBuffer: since some sound recording devices cannot (or will not) record below a certain sample rate, trim the samples down if the user really wanted fewer samples
	emitPartialBuffer
	meterFrom:count:in: Update the meter level with the maximum signal level in the given range of the given buffer.
	recordLoop Record process loop that records samples.
	samplesPerFrame Can be overridden to quantize buffer size for, eg, fixed-frame codecs

	recording controls	top
	clearRecordedSound Clear the sound recorded thus far. Go into pause mode if currently recording.
	pause Go into pause mode. The record level continues to be updated, but no sound is recorded.
	playback Playback the sound that has been recorded.
	resumeRecording Continue recording from the point at which it was last paused.
	startRecording Turn of the sound input driver and start the recording process. Initially, recording is paused.
	stopRecording Stop the recording process and turn of the sound input driver.

	results	top
	condensedSamples Return a single SoundBuffer that is the contatenation of all my recorded buffers.
	condensedStereoSound Decompose my buffers into left and right channels and return a mixed sound consisting of the those two channels. This may be take a while, since the data must be copied into new buffers.
	recordedSound Return the sound that was recorded.
	soundSegments

	trimming	top
	copyFrom:to:normalize:dcOffset: Return a new SoundBuffer containing the samples in the given range.
	copyTo:from:to:from:startingAt:normalize:dcOffset: Copy samples from buf to resultBuf removing the DC offset and normalizing their volume in the process.
	endPlace
	firstSampleOverThreshold:dcOffset:startingAt: Beginning at startPlace, this routine will return the first place at which a sample exceeds the given threshold.
	normalizeFactorFor:min:max:dcOffset: Return a normalization factor for the range of sample values and DC offset. A normalization factor is a fixed-point number that will be divided by 1000 after multiplication with each sample value.
	place:plus: Return the place that is nSamples (may be negative) beyond thisPlace.
	scanForEndThreshold:dcOffset:minLull:startingAt: Beginning at startPlace, this routine will find the last sound that exceeds threshold, such that if you look lull samples later you will not find another sound over threshold within the following block of lull samples. Return the place that is lull samples beyond to that last sound. If no end of sound is found, return endPlace.
	scanForStartThreshold:dcOffset:minDur:startingAt: Beginning at startPlace, this routine will find the first sound that exceeds threshold, such that if you look duration samples later you will find another sound over threshold within the following block of duration samples. Return the place that is duration samples prior to that first sound. If no sound is found, return endPlace.
	segmentsAbove:normalizedVolume: Break the current recording up into a sequence of sound segments separated by silences.
	suppressSilence
	trim:normalizedVolume: Remove the leading and trailing parts of this recording that are below the given threshold. Remove any DC offset and scale the recording so that its peaks are the given percent of the maximum volume.

	accessing	top
	anyActive Return true if any sound recorder is actively recording
	canRecordWhilePlaying Return true if this platform supports simultaneous sound recording and playback.

	class initialization	top
	initialize SoundRecorder initialize

	instance creation	top
	new Answer a new instance of the receiver (which is a class) with no indexable variables. Fail if the class is indexable.