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A sound is recorded by making a measurement of the amplitude of the sound at regular intervals which are defined by the "sample ''sampling rate"'' and to a vertical resolution (between the lowest and highest points on the wave) that is called the ''bit-depth''. The act of taking the measurement is often called "''sampling" '' and each measurement unit is called a "''sample"''. A file which contains samples is often called a "waveform, sound sample" or ", audio sample", ''etc.''

The "sample sampling rate" defines the rate/frequency at which the measurements are taken. The higher the sample sampling rate, the faster /more frequently the measurements are taken, and the higher the maximal frequency that can be represented by the signal. Conversely, the lower the sample sampling rate, the slower the measurements are taken, and the maximal frequency that can be stored is lower. The sample rate is described by the "Hz" unit of measurement. The "Hz" unit of measurement means "per second". Therefore, a sample sampling rate of 44010Hz 44010 Hz means 44010 measurements are taken each second (this is in other words, one measurement every 1/44010 th of a second).

If the "sample rate" sampling is too low, then changes in the sound which occur between each measurement will not be measured. Because higher audio frequencies are defined by oscillating more rapidly, this means that lower sampling rates can store only lower frequencies. Therefore , the faster the measurements are taken, the more accurate the recording will be, and therefore thus the higher the quality of sound that can be recorded. Of course, at high sample rates, because there are many more measurements taken, the resulting size of the file (containing the audio data) can be large.

It is worth noting As should be familiar to CPC users with a little technical knowledge, a sample that at high uses 8-bits for storage can represent 256 distinct amplitude levels, and a sample which uses 16-bits for storage can describe 65536 distinct amplitude levels. The higher the number of bits used by each sample ratesfor storage, because there are many more measurements takenthe larger the range of distinct amplitude levels that can be represented. Therefore, the resulting size higher the number of bits used by each sample for storage, the file (containing higher the audio data) quality of sound that can be largerecorded. Moreover, the number of bits is directly related to the dynamic range of the resulting signal; that is, how much of a difference there is between the quietest and loudest sounds that it can represent. 16-bit signals provide a nominal 96 dB of dynamic range.

All modern sound cards should support 8-bit and 16-bit samples and sample rates of 22050Hz 22050 Hz and 44100Hz44100 Hz. Some sound cards will support a greater range of recording rates which can be lower and higher than these values. The familiar format of CD audio uses a sampling rate of 44100 Hz and a bit-depth of 16-bits. These values are more than adequate to represent almost all real-world signals for listening by humans - and also, conveniently, are fine for Amstrad tapes, too! In fact, in theory, because the standard Amstrad tape routines have a maximal frequency of 2500 Hz, settings as low as 8000 Hz and 8 bits would probably be fine. However, you will probably want to use higher settings, just in case and/or to keep in line with more common formats such as CD audio, especially if you intend to archive your recordings.

Some For samp2cdt, you should save the file as "PCM" ("Pulse Code Modulation"). This is a uncompressed, unencoded storage representation. Each sample is a single measurement of the amplitude of the sound cards will support taken at a greater range measurement point in time. Other representations such as "ADPCM" ("Amplitude Delta Pulse Code Modulation"), encode or compress the data to reduce the size of recording rates which the audio file. Samp2cdt can be lower and higher than 't understand these valuesrepresentations, so please use "PCM" only.

For samp2cdt you should save the file as "PCM" ("Pulse Code Modulation"). This is a uncompressed, unencoded storage representation. Each sample is a single measurement ===Illustrations and explanations of the amplitude of the sound taken at a measurement point in time. Other representations such as "ADPCM" ("Amplitude Delta Pulse Code Modulation"), encode or compress the data to reduce the size of the digital audio file. Samp2cdt can't understand these representations, so please use "PCM" only.===

''Fig 4. An amplitude/time graph showing the sampled waveform. This waveform was generated at a high sample rate, and therefore the resulting waveform has a shape which is similar to the original. This waveform is the type you can see in a audio recording program like Goldwave. '''Note''', however, that this distinctively square signal is not what would be output by any barely decent sound-card! Audio hardware has built-in filters to smooth waveforms as they are converted from digital to analogue.''

''Fig 7. An amplitude/time graph showing the sampled waveform. As explained in the note for Figure 4, this is only a visual representation of the digitally stored audio, '''not''' of the signal that would be output by any competent audio card. However, it does illustrate how low sampling rates reduce the bandwidth of frequencies: This waveform was generated at a low sample rate, and therefore the resulting waveform is much more coarse compared to Fig 4. Notice that although the general shape is similar to the original waveform, and much of the smoothness is is lost between the time of each measurement. The loss of smoothness also means loss of information since this waveform is not the same as the original. If you compare this graph against Fig 4 then you will see that : the lower the sample raterating, the more information is lost. The higher the sample rate; in other words, the less information maximal frequencies that the signal can represent is lostlower. Similarly, lower bit-depths mean that the signal is less accurate, and in extreme cases can generate audible noise, Therefore, to record a sound, it is best to use a relatively high sample sampling rateand bit-depth.''

1. The "Nyquist theory" states that in order to accuratly record a sound of a known frequency, you must use a recording frequency which is more than twice that frequency(note "more than", not equal to). i.e. Example: to record a sound of 3000Hz3000 Hz, you must record using 6000Hz>6000 Hz. If you use a lower frequency sampling rate (e.g. 5000Hz5000 Hz), then frequencies less than or equal to half of the sound is not recorded accuratlysampling rate cannot be properly represented and will be altered into lower-sounding frequencies. Most Amstrad loaders are between 300 to 2500Hz2500 Hz, therefore you should use a recording sample rate of at least 5000Hz>5000 Hz. It is recommended to use one of the "common" sample rates. e.g. 22050Hz 22050 Hz (22Khz22.05 Khz) or 44100Hz 44100 Hz (44.1Khz1 Khz).

 Both methods can represent the same data, just in different ways (techies will be able to compare this to their knowledge of Z80 assembly), so there is no advantage to using either. The original reason for the two methods is due to the original method to playback the sound. Modern sound cards can play audio stored in both methods ways.Note that both (albeit more obvious in the latter) share a feature typical of data storagebinary-encoded numbers: there is no exact 'centre' value, because the total number of possible values is even. In the context of audio, this means that, if the signal spanned the entire range, its centre (average) would be slightly off-zero (in this case, below), which is known as a DC offset. However, even if this did occur, it would be negligible and certainly not audible by humans!