'''Translation note :'''Cassette == Recording a sound ==Tape.
A sound is recorded by making a measurement of the amplitude of the sound at regular intervals which are defined by the "sample rate". 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 "sound sample" or "audio sample".
The "sample rate" defines the rate/frequency at which the measurements are taken. The higher the sample rate, the faster the measurements are taken, and the lower the sample rate, the slower the measurements are taken. The sample rate is described by the "Hz" unit of measurement. The "Hz" unit of measurement means "per second". Therefore, == Recording a sample rate of 44010Hz means 44010 measurements are taken each second (this is one measurement every 1/44010 th of a second).sound ==
If the "sample rate" is too low, then changes in the A sound which occur between each is recorded by making a measurement will not be measured. Therefore of the faster amplitude of the measurements sound at regular intervals which are taken, defined by the more accurate ''sampling rate'' and to a vertical resolution (between the recording will be, lowest and therefore highest points on the higher wave) that is called the quality ''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 that can be recordedsample, audio sample, ''etc.''
It is worth noting that The sampling rate defines the rate/frequency at high sample rates, because there which the measurements are many taken. The higher the sampling rate, the faster/more frequently the measurements are taken, and the resulting size of higher the file (containing maximal frequency that can be represented by the audio data) signal. Conversely, the lower the sampling rate, the slower the measurements are taken, and the maximal frequency that can be largestored is lower. The sample rate is described by the "Hz" unit of measurement. The "Hz" unit of measurement means "per second". Therefore, a sampling rate of 44100 Hz, a.k.a. 44.1 kHz, means that 44100 measurements are taken each second (in other words, one measurement every 1/44100 th of a second).
A "sample" which uses 8-bits for storage can represent 256 distinct amplitude levels and a sample If the sampling is too low, then changes in the sound which uses 16-bits for storage can describe 65535 distinct amplitude levelsoccur between each measurement will not be measured. The Because higher the number of bits used audio frequencies are defined by each sample for storageoscillating more rapidly, the larger the range of distinct amplitude levels this means that lower sampling rates can be representedstore only lower frequencies. Therefore, the higher faster the number of bits used by each sample for storagemeasurements are taken, the more accurate the recording will be, and 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.
All modern sound cards As should support be familiar to CPC users with a little technical knowledge, a sample that uses 8-bit bits for storage can represent 256 distinct amplitude levels, and a sample which uses 16-bit samples and bits for storage can describe 65536 distinct amplitude levels. The higher the number of bits used by each sample rates for storage, the larger the range of 22050Hz distinct amplitude levels that can be represented. Therefore, the higher the number of bits used by each sample for storage, the higher the quality of sound that can be recorded. 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 44100Hzloudest sounds that it can represent. 16-bit signals provide a nominal 96 dB of dynamic range. All of that sound theory is, of course, irrelevant to the CPC as it only gets 1-bit of information out of the tape signal level.
Some sound cards will support a greater range of recording rates which can be lower and higher than these values.===What settings should you use?===
For samp2cdt you All modern sound cards should save the file as "PCM" ("Pulse Code Modulation")support 8-bit and 16-bit samples and sample rates of 22050 Hz and 44100 Hz. This is Some sound cards will support a uncompressed, unencoded storage representationgreater range of recording rates which can be lower and higher than these values. Each sample is The familiar format of CD audio uses a single measurement 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 amplitude 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.The hardware tape data separator inside the CPC only extract 1-bit of information out of the sound taken at a measurement point signal that comes in time. So, using 16-bit instead of 8-bit samples provides no gain at all.
Other representations such as "ADPCM" ("Amplitude Delta Pulse Code Modulation"), encode or compress the data to reduce the size ===Illustrations and explanations of the digital audio file. Samp2cdt can't understand these representations, so please use "PCM" only.===
[[Image:wave1.gif]]
[[Image:wave4.gif]]
''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.''
[[Image:wave5.gif]]
[[Image:wave7.gif]]
''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 sampling rate, the more information is lost. The higher ; in other words, the sample ratemaximal frequency that the signal can represent is lower. Similarly, lower bit-depths mean that the less information signal is lostless 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; CD audio's 44.1 kHz and 16-bit should be more than adequate for most uses. Due to the way the CPC hardware process the sound signal, 16-bit has zero advantage over 8-bit for CPC cassettes. To sum it up, 44.1kHz and 8-bit is recommended for storage of CPC cassettes.''
Notes:
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).
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, 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.On a side note, the WAV sound container only allows 8-bit unsigned samples, so there is no ambiguity as to how to interpret 8-bit samples.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!The fact there is no 'centre' value is actually a good thing, as the CPC has to convert the sound signal that comes in to a single bit, determining whether the signal is low or high.
== Duplication of cassettes ==
A loader on the computer must therefore be able to identify the actual sound of the data from other sounds that are on the cassette. If it can't do this, then there will be loading errors.
If you are transfering a cassette using samp2cdtCSW2CDT, then you are advised to use an original (i.e. a cassette created directly from a master cassette), or a first generation copy (i.e. a cassette copied from an original).
== Loader ==
1. The CPC464 and CPC464+ have a cassette player built in. To connect a cassette player to the CPC664, CPC6128 or KC Compact then you must use a lead.
2. It is not known exactly how the amplitude of the sound from the cassette corresponds to the final "0" or "1" measurement.
samp2cdt uses a crude method to perform this conversion.
For a 8-bit signed sample:
* if the amplitude is 0..128, then the final measurement will be "1".
* if the amplitude is -128..0 then the final measurement will be "0".
=== Writing ===
If the state of bit 5 is changed at a fixed frequency, then the graph of the state of bit 5 over time will be a square wave. However, the resulting audio written on the cassette will not be a perfect square wave because nature will attempt to convert the waveform into a sine wave.
Loading system audio waveform
Every loading system on the Amstrad uses a serial bit-stream. i.e. a single bit of information is read at a time.
This serial bit-stream is grouped into blocks of audio sound.
Every loading system uses a basic structure to describe each audio block in the following order:
|pilot|sync|data|trailer|
'''pilot'''
This is also refered to as "leader" by some documents.
This is constructed from a repeated waveform often with a fixed number of repetitions defined by the loading system.
The shape of the waveform is known by the loader program and this is used to identify the pilot waveform from other waveforms that may be present (e.g. noise).
The pilot is often long, so that the loader doesn't need to see the start of the pilot waveform in order to load the block.
The loader program will test the incoming waveform, checking it against the parameters defined for the pilot, before the waveform is accepted as the pilot waveform. (e.g. the number of repetitions must be some defined minimum value). The incoming waveform must fall within these specifications otherwise the waveform is not accepted as a pilot waveform.
'''sync ("synchronisation")'''
The sync is a waveform which is different to the pilot, and this defines the end of the pilot and the start of the data. When this sync has been detected, the loader knows that there is data following, and that the loader is always at the same point in the data stream. i.e. the loader program is synchronised to a specific point in the data waveform.
'''data'''
This is the actual data which is composed of waveforms defining "0" and "1" data bits.
The first element of the data may be a marker or id which may, for example, indicate the type of data in the block or the number of the block.
The remaining bits will define the data and zero or more checksums.
The whole data may consist of a single block with a single location and length (e.g. one block for a screen another for data), or multiple blocks each with their own location and length. (e.g. one block for screen and data)
The location and lengths of the blocks may be in the data stream itself, or they may be in a preceeding block, or may be hard-coded into the loader program.
'''trailer'''
The trailer always follows the data. Some loaders may not have a trailer. The two main purposes of the trailer are to ensure that the waveform of the last data bit in the data is constructed correctly and to provide some time in which the loader can prepare for the next block.
The exact definition of the loading systems's audio waveform is defined by the loader program.
samp2cdt has a number of decoder algorithms which recognises the audio waveform of various loading systems. These decoders read the waveform using a similar method to the loader program itself. These decoders have been created by examining the instructions of each loader program and the graph of the waveform in a sound recording package.
== Example of a typical loading system ==
;;
;; B = &F5 (I/O address of PPI 8255 input port B)
;; C = previous data read from PPI port B ld d,0 ;; initialise count to 0 .loop inc d ;; increment count in a,(c) ;; read input to PPI 8255 port B xor c ;; exclusive-or with previous data read from PPI 8255 port B and %10000000 ;; isolate bit 7
;; if result is 0, then the state of bit 7 that has
;; been read is the same as the previous state. i.e. bit 7 has not changed state.
;; if result is not 0, then the state of bit 7 has changed.
;; e.g. if bit 7 was previously 1, it is now 0. if bit 7 was previously 0, it is now 1. jr z,loop
;; when execution reaches here we know that bit 7 has changed state and D
;; contains the number of tests.
== Various Audio file formats ==
There are numerous Audio file formats, each of which can store audio, but each has it's its own structures and representation for the data.
The "format" of a file describes the internal structure, order and encoding of the data within the file.
Here is a list of the audio file formats supported by samp2cdt:
* 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.
==Links==
[[Category:Hardware]][[Category:DATA Storage|*]][[Category:Music and sound]][[Category:CPC Internal Components]]
