Changes

['''Fig 1. An amplitude/time graph showing the waveform of the original sound]'''

['''Fig 2. An amplitude/time graph showing the waveform of the original sound. The crosses indicate the amplitude measured at each sample time and the dotted lines indicate the the time of each measurement. The duration of time between each dotted line, defined by the sample rate, is equal to the duration of a sample. From this it can be seen that each sample has a finite and equal duration.''

['''Fig 3. An amplitude/time graph showing the waveform of the original sound. As in Fig 2, the crosses indicate the amplitude measured at each sample time. The dotted line shows the waveform generated by sampling. The final value of each sample is defined to be the amplitude measured at the time of measurement.'''

['''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.'''

['''Fig 5. An amplitude/time graph showing the waveform of the original sound. As in Fig 2, this graph shows the amplitude of each measurement, and the dotted line indicates the time of measurement. This graph was created using a low sample rate. Notice that the time between each measurement is longer compared to Fig 2.'''

[This image shows the same information as Fig 3, but using a low sample rate ['''Fig 6. An amplitude/time graph showing the waveform of the original sound. As in Fig 3, the crosses indicate the amplitude measured at each sample time, and the dotted line shows the waveform generated by sampling. This graph shows the resulting waveform generated using a low sample rate.'''

['''Fig 7. An amplitude/time graph showing the sampled waveform. 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 rate, the more information is lost. The higher the sample rate, the less information is lost. Therefore, to record a sound, it is best to use a high sample rate]'''

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 twice that frequency. i.e. to record a sound of 3000Hz, you must record using 6000Hz. If you use a lower frequency (e.g. 5000Hz), then the sound is not recorded accuratly. Most Amstrad loaders are between 300 to 2500Hz, therefore you should use a recording sample rate of at least 5000Hz. It is recommended to use one of the "common" sample rates. e.g. 22050Hz (22Khz) or 44100Hz (44.1Khz)

i.e2. There are two different representations to record store the amplitude of the sample in a sound of 3000Hz, you must record using 6000HzPCM audio file: unsigned or signed.

If you use a lower frequency (e* A 8-bit unsigned sample has values between 0 and 255.g. 5000Hz)In this range, 0 represents a low amplitudes, 255 a high amplitude, then and the sound is not recorded accuratlyamplitudes increase linearly from 0 to 255.

Most Amstrad loaders are between 300 to 2500Hz, therefore you should use a recording sample rate of at least 5000Hz. It is recommended to use one of the "common" sample rates. e.g. 22050Hz (22Khz) or 44100Hz (44.1Khz) 2. There are two different representations to store the amplitude of the sample in a PCM audio file: unsigned or signed. * A 8-bit unsigned sample has values between 0 and 255. In this range, 0 represents a low amplitudes, 255 a high amplitude, and the amplitudes increase linearly from 0 to 255. * A 8-bit signed sample has values between -128 and 127. In this range, -128 represents a low amplitude, and 127 high amplitude, and the amplitudes increase linearly from -128 to 127.

Both methods can represent the same data, 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 both methods of data storage.

1. # system loader recognises and loads "program for fast-loader". 2. # "program for fast loader" takes control and loads the data from the fast-loader block(s) into computer memory 3. # when loading has completed, the program is executed.

1. # a CPC without a disc interface (e.g. a CPC464, CPC464+ or KC Compact) will start-up in cassette mode. For a CPC with a disc interface attached (or internal), you must type |TAPE to enter cassette mode.

You can test if the computer is operating in cassette mode by typing RUN". If you see "Press PLAY then any key", then the computer is operating in cassette mode. If there is an error, then the computer is not operating in cassette mode.

1. a stored "Checksum".

This is the result of the "checksum calculation" calculated from the correct data. This is then stored with the data (e.g. before or after the data) when the master cassette is created. 2. a calculated "Checksum".

This is the result of the "checksum calculation" calculated from the data read from the cassette. After the calculation is complete it is compared against the stored checksum.

* Windows Wave file (a file which has the ".wav" file extension) is the common file format used for audio sounds on computers running "Windows". * A Voice Wave file (a file which has the ".voc" file extension) was created by Creative for the original Soundblaster ISA sound card. This file format is used by the original voc2tzx utility which samp2cdt was developed from. * A Audio Interchange file (a file which has the ".aiff" or ".aif" file extension) is the common file format used for audio sounds on the Mac computer. * A file which has the ".iff" file extension is the common file format used for audio sounds by the Amiga computer.