AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 3 - Interconnecting and Processes |
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Hi there, this week I am "front loading" with more in depth concepts that will also appear again further into the course, I am front loading as I feel this a better way for you to work.
In the following text you will find
"keywords"
in bold with a brief definition under those headings. The keywords are essential to understand.
The outlines here are to serve as a mere introduction to the concepts and it is suggested you research in much more detail.
There are two parts this week.
1. The questions found below
2. A practical, which is a software review.
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An Overview of Digital Audio Processes
Overview
In a digital recording of binary data, a bit is either correct or wrong, with no intermediate stage.
Small amounts of noise are rejected, but inevitably, infrequent noise impulses cause some individual bit to be in error.
Dropouts cause a larger number of bits in one place to be in error, (burst error).
If a binary digit is wrong, it is only necessary to reverse the state and it must be correct.
The main difficulty is in identifying the bits which are in error.
Adding redundancy helps here.
The purpose of error correction is to render errors from the medium inaudible.
Concealment
Concealment is a process where the value of a missing sample is estimated from those nearby.
Note
The estimated sample value is not necessarily exactly the same as the original.
In some circumstances concealment can be audible.
In a well-designed system, concealment's occur with negligible frequency.
How it works
Concealment is made possible by rearranging the sample prior to recording.
Odd numbered samples are separated from even numbered samples.
Odd and even sets of samples are recorded in different places on the medium.
A burst error will only affect one set.
On replay the samples are combined into their natural sequence.
The error is now split up so that it results in every other sample being lost.
The waveform is now described half as often but can still be reproduced with some loss of accuracy.
Almost all digital recorders use such an odd/even shuffle for concealment.
Interleaving
In high-density recorders, more data are lost in a given sized dropout. The efficiency of the system can be raised using interleaving.
How it works
Sequential samples from the ADC are assembled into codes, but these are not recorded in their natural sequence.
A number of codes are assembled along rows in a memory.
When the memory is full, it is copied to the medium by reading down Columns.
On replay, the samples need to be de-interleaved to return them to their natural sequence.
This is done by writing samples from tape into a memory in columns, and when full, the memory read in rows.
If a burst error occurs on the medium, it will damage sequential samples in a vertical direction in the de-interleave memory.
When the memory is read, a single large error is broken down into a number of small errors.
Food for thought
The presence of an error correction system means that the audio quality is independent of the tape/head quality within limits.
There is no point in trying to assess the health of a machine by listening to it as this will not reveal whether the error rate is normal or within a whisker of failure. The only useful procedure is to monitor the frequency with which errors are being corrected, and compare it with normal figures.
Channel coding
Data words representing audio samples contain many bits and so they have to be recorded serially, a bit at a time.
Why Channel Codes?
Recording data serially isn't as simple as connecting the serial output of a shift register to the head.
Serialised raw data needs to be modulated into a waveform which contains an embedded clock irrespective of the values of the bits in the sample.
On replay a circuit called a data separator can lock to the embedded clock and use it to separate strings of identical bits.
The process of modulating serial data to make it self-locking is called channel coding.
Channel coding also shapes the spectrum of the serialised waveform to make it more efficient.
With a good channel code, more data can be stored on a given medium.
Spectrum shaping is used in CD to prevent the data from interfering with the focus and tracking servos, and in RDAT to allow re-recording without erase heads.
Channel coding is also needed to broadcast digital audio where shaping of the spectrum is a requirement to avoid interference with other services.
Data Reduction
One of the purposes of the subconscious processing is to limit the amount of information presented to the conscious mind, to prevent stress and to make life safer and easier.
Auditory masking is a process which selects only the most important frequencies from the spectrum applied to the ear.
Data reduction takes advantage of this process to reduce the amount of data needed to carry sound of a given subjective quality.
The data reduction process mimics the operation of the hearing mechanism as there is little point in recording information only for the ear to discard it.
Uses
Data reduction is essential for DAB whose bandwidth is important.
It is used to reduce consumption of the medium in consumer recorders such as DCC and Mini-disc.
Reduction to around one-quarter of the PCM rate can be virtually inaudible on high-quality systems.
Greater compression factors inevitably result in quality loss which may be acceptable for certain applications such as communications but not for quality music reproduction.
The output of a data reduction unit is still binary, but it is no longer regular PCM, so it cannot be fed to a normal DAC without passing through a matching decoder which provides a conventional PCM output.
The combination of a data reduction unit and a decoder is called a codec.
Hard Disc Recorders
Data is stored on concentric tracks which it accesses by moving the head radially.
Time compression allows the hard disk to be an audio recorder.
The data rate of the disk drive is far higher than the sampling rate (time compression).
Data can be read and placed in the timebase corrector in a fraction of real time.
The timebase corrector moves through the memory and the disk drive has time to change tracks.
When memory is available, the drive is commanded to read, and fills the space.
The buffer memory provides an unbroken stream of samples to the DAC and so continuous audio is obtained.
Recording samples uses the same process but the inverse.
Advantages
Access is much quicker than with tape.
This speeds up editing.
The use of data reduction allows the recording time of a disc to be extended considerably.
This technique is often employed in plug-in circuit boards which are used to convert a personal computer into a digital audio recorder.
An Open Reel Digital Recorder
Overview
Analogue inputs are converted to the digital domain by converters
One converter for every channel to be recorded.
There is not necessarily one tape track per audio channel (unlike analogue).
In stereo machines the two channels of audio samples may be distributed over a number of tracks each in order to reduce the tape speed and extend the playing time.
Operation
The samples from the ADC will be converted into odd and even for concealment purposes.
Usually one set of samples will be delayed before recording.
The continuous stream of samples from the converter will be broken into blocks by time compression prior to recording.
Time compression allows the inclusion of edit gaps, addresses and redundancy bits into the data stream.
An interleaving process is also necessary to re-order the samples prior to recording.
The raw data is then channel coded - before recording.
On replay the data separator reverses the channel coding to give the original raw data with the addition of some errors.
Following de-interleave, the errors are reduced in size and are more easily correctable.
The memory required for de-interleave may double as the timebase correction memory, so any variations in the speed of the tape are rendered undetectable.
Any errors which are beyond the power of the correction system will be concealed after the odd-even shift is reversed.
Following conversion in the DAC an analogue output emerges.
On replay a digital recorder works differently to an analogue recorder, which simply drives the tape at a constant speed.
In contrast, a digital recorder drives the tape at a constant sampling rate.
Main points
The timebase corrector works by reading samples out to the converter at a constant frequency.
This reference frequency comes typically from a crystal oscillator.
If the tape goes to fast, the memory will be written faster than it is being read, and will eventually overflow.
Conversely, if the tape goes too slow, the memory will become exhausted of data.
In order to avoid these problems, the speed of the tape is controlled by the quantity of data in the memory.
If the memory is filling up, the tape slows down; and vice versa.
As a result, the tape will be driven at whatever speed is necessary to obtain the constant sampling rate.
Rotary-Head Recorder
One of the main advantages of rotary head recorders is its extremely high packing density: the number of bits that can be recorded in a given space. In a digital audio recorder packing density directly translates into playing time available for a given size of the medium.
Overview
The heads are mounted in a revolving drum.
The tape is wrapped around the surface of the drum in a helix
Slanting tracks are laid.
The discontinuous tracks of the rotary head recorder are naturally compatible with time compressed data.
As Figure 1.16 illustrates, the audio samples are time compressed into blocks, each of which can be contained in one slant track.
In a medium such as RDAT there are two heads mounted on opposite sides of the drum.
One rotation of the drum lays down two tracks.
Effective concealment can be had by recording odd numbered samples on one track of the pair and even numbered samples on the other.
A rotary-head recorder contains the same basic steps as any digital recorder.
Digital Compact Cassette
Overview
DCC is a consumer recorder that uses data reduction.
Data is reduced to a quarter of its size by data reduction processing.
This allows a reasonable tape consumption similar to that achieved by a rotary head recorder.
Operation
DCC uses stationary heads in a conventional tape transport which can also play analogue cassettes.
Data is distributed over 9 parallel tracks which occupy half the width of the tape.
At the end of the tape the head rotates and plays the other tracks in reverse.
The advantage of the conventional approach with linear tracks is that tape duplication can be carried out at high speeds.
This makes DCC attractive to record companies.
Further Points
Data reduction relies on the phenomenon of auditory masking and this may effectively restrict DCC to being a consumer format.
It will be seen from Figure 1.19 that the data reduction unit adjacent to the input is complimented by the expansion unit or decoder prior to the DAC.
References and further reading:
1. John Watkinson, "The Art of Digital Audio", Pub. Focal press, 1995
2. Bloom, P.J., "High-quality digital audio in the entertainment industry": an overview of achievements and challenges. IEEE Acoust. Speech Signal Process. Mag., 2, 2-25 (1985)
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QUESTIONS
1. Explain the function of the of the following devices
(a) Anti alias Filter
(b)Sample & Hold
(c) Dither Generator
2. Explain the operation of copying digitally when the devices are equipped with SCMS
3. Describe the functions of the contents of a sub frame of MADI
4. Describe the Master Clock system of synchronisation of a digital signal chain
5. With regard to a CD-Recordable system explain the following operations
(a) Single Session
(b) Track at Once
(c) Multi Session
6. Explain what information concerning a CD can be transferred to a DAT copy disregarding any SCMS considerations
7. With the aid of diagrams explain time compression. List two applications of time compression.
8. Explain the principles of error concealment.
9. Describe in simple terms the main advantage of an over sampled D-A
10. Compare the results of too high recording levels in a analogue and digital tape systems.
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Mon Sep 20, 2004 10:32 am |
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iNSTiNCT2765
Joined: 05 Nov 2003
Posts: 60
Location: Denmark |
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Week 3 Questions
-haven't found enough on time compression yet but i'll get that question answered soon..hopefully
1. Explain the function of the following devices
(a) Anti alias Filter this is a filter that limits the frequency range of an analogue signal to half the sample-rate (Nyquist frequency) before it goes through an A/D converter so the sample is accurately created with no aliasing occurring.
(b) Sample & Hold this circuit holds the analogue value steady while the A/D converter outputs the corresponding digital word. If the value isnt held steady then the conversion could be inaccurate.
(c) Dither Generator this generator produces a noise signal that is added to the input signal to minimize the effects of quantisation error.
2. Explain the operation of copying digitally when the devices are equipped with SCMS.
The CD that is to be copied must be the original, as SCMS (Serial Copy Management System) does not allow copies to be made from copies when the data is streamed using SPDIF. When an original CD is copied, a no further copying message is added to the copy so it cannot be used to make further copies.
3. Describe the functions of the contents of a sub frame of MADI.
A MADI frame consists of 56 sub frames, each carrying one audio channel with associated channel status and user data. The first four bits of each sub frame are different from those of an AES/EBU sub frame and function as follows:
Bit 0 mark for channel 0 (1 for first channel, otherwise 0)
Bit 1 channel on/off (active channels marked with 1)
Bit 2 channel A/B (marking of a stereo channel where A = 0)
Bit 3 channel status block sync. (marks start of a new channel status block)
4. Describe the Master Clock system of synchronisation of a digital signal chain.
Each individual piece of equipment in a digital signal chain is synchronized to and controlled by the same Master Clock
5. With regard to a CD-Recordable system explain the following operations:
(a) Single Session the table of contents is written first and then the data is written to the CD in one session and the CD is closed right after to make is playable on all conventional players. This is done in one uninterrupted session.
(b) Track at Once the laser stops writing after each track and is then recalibrated before writing the next track. The table of contents is not written until the last track has been written.
(c) Multi Session this operation consists of writing data to a CD in more than one session. Each track is written separately and you can add new data to the disc over a longer period of time until it is full and then it can be closed and used.
6. Explain what information concerning a CD can be transferred to a DAT copy disregarding any SCMS considerations.
The table of contents and the start times of each track are transferred with the audio data from a CD to DAT if SCMS considerations are disregarded.
7. With the aid of diagrams explain time compression. List two applications of time compression.
8. Explain the principles of error concealment.
Error concealment is a technique used to reduce the effect of a digital error if the error cannot be corrected using error correction. It involves making a smooth transition from the last good block and first good block before and after the error respectively. This is done using some form of crossfading. This is the reason why a digital copy may not be an exact duplicate of the original master.
9. Describe in simple terms the main advantage of an over sampled D-A.
The main advantage of an oversampled D-A is plainly a better linear phase response over the audio spectrum, in other words, a more detailed sound.
10. Compare the results of too high recording levels in an analogue and digital tape systems.
When a signal is recorded too high or too hot in an analogue system, harmonic distortion and compression is added to the signal, which can be a pleasant characteristic. In a digital system, if the signal is recorded too hot, a very unpleasant distortion also called clipping occurs, which is to be avoided as it does not benefit the audio signal in any way. |
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Sun Sep 26, 2004 5:24 am |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| More Detaisl Needed |
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Some good stuff here iNSTiNCT2765
I'll indicate a few bits that you need to get a little more depth on.
quote:
4. Describe the Master Clock system of synchronisation of a digital signal chain.
Each individual piece of equipment in a digital signal chain is synchronized to and controlled by the same Master Clock
Yes a little more expansion needed here.
quote:
9. Describe in simple terms the main advantage of an over sampled D-A.
The main advantage of an oversampled D-A is plainly a better linear phase response over the audio spectrum, in other words, a more detailed sound.
This is asking you about your understanding of oversampling, so you need to explain what you know. What are the benefits here?
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Mon Sep 27, 2004 12:09 pm |
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iNSTiNCT2765
Joined: 05 Nov 2003
Posts: 60
Location: Denmark |
4. Describe the Master Clock system of synchronisation of a digital signal chain.
The master clock system synchronises to each component individually. This eliminates timing delays or jitter if the clock was sent through the signal chain sequentially.
9. Describe in simple terms the main advantage of an over sampled D-A.
The main advantage of an oversampled D-A is plainly a better linear phase response over the audio spectrum, in other words, a more detailed sound. This means that the signal to noise ratio due to quantisation on a CD, which is 96dB, can be spread over a larger bandwidth meaning the noise in the 20-20kHz frequency range is reduced. |
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Tue Sep 28, 2004 12:01 pm |
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