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Week 3 - Interconnecting and Processes

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Forum Index > Digital Audio Operations 01 2005


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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Week 3 - Interconnecting and Processes  Reply with quote  

Hi Rico!

There are two parts this week.

1. The questions found below
2. A practical, which is a software review.


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.
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It's all in the ears. - Learn the concepts not the software. Audio Courses is a way into the music business for you
Post Mon Mar 21, 2005 6:51 am
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Rico1210



Joined: 03 Aug 2004
Posts: 39
Location: Newcastle, UK
 Reply with quote  

Hi,

answers for this week. I have a diagram on time compression uploaded to the ftp, going to try and link it, but i'll probably do it wrong and it won't work Rolling Eyes

1. Explain the function of the of the following devices

(a) An Anti alias Filter prevents aliaising from occurring. It does so by preventing frequencies of more than half the sampling rate from reaching the sampling stage. This is achieved by having a low-pass filter at the input.

(b) Sample & Hold is a device used in analog to digital convertors that extends instantaneous samples. It is also known as a ‘track-hold circuit’

(c) A Dither Generator produces low level noise that is added to an input signal to reduce errors in quantisation.

2.Explain the operation of copying digitally when the devices are equipped with SCMS

When copying digitally using devices that are equipped with SCMS (Serial Copying Management System), the receiving device determines whether to allow or disallow the recording. It determines this by the information received from the category code in the channel-status byte. Byte 15 (the last byte) indicates whether the signal is from an original recording (0), or a from a first-generation copy (1). If a copy is made, a 'no further copying' flag is place on the copy to prevent further copying .

3.Describe the functions of the contents of a sub frame of MADI

A sub frame of MADI(Multi-channel Audio Interface) is almost identical to that of an AES/EBU sub frame. The last 28 bits are identical to AES/EBU with only the synchronizing in the first four bits being different. Bit 0 is used as channel 0. The 56 channels of of MADI follow each other in numerical order, therefore it is necessary to identify channel 0 to avoid the channels mixing up. Bit 1 is used to indicate active channels. When less than 56 channels are being transmitted, active channels are identified by 1 and must be consecutive starting at channel 0. Channels that are inactive are set to 0 and follow the active channels. AES/EBU signals may contain two audio channels. Bits 2 and 3 are used to tell the receiver whether a channel contains the A or B leg, and when the AES/EBU channel status block sync occurs.

Reference. The Art of Digital Audio - John Watkinson


4.Describe the Master Clock system of synchronisation of a digital signal chain.

The Master Clock synchronisation system of a digital signal chain controls each individual component from a master clock. This stops any accumulation of sample frequency jitter if the clock was transferred sequentially through the chain


5.With regard to a CD-Recordable system explain the following operations

A Single session, or Disc at Once, writes the whole CD in a 'single session'. There are 'lead in' and 'lead out' areas written to the disc that contain navigational information, aswell as the program area that contains the data. A Table Of Contents(TOC) is written on completion, and enables playback on any machine.

Track at Once is a session where each track is written to the CD individually. The writer recalibrates after each track. The CD can't be played on a conventional machine until the Table Of Contents (TOC) has been written.

A Multi Session disc can support several recording sessions, it is simply a chain of single sessions each recorded with 'lead in' and 'lead out' areas. The first session can be read in any drive, but subsequent sessions can only be accessed by a multisession capable drive. The TOC is not written until all sessions have been added.


6.Explain what information concerning a CD can be transferred to a DAT copy disregarding any SCMS considerations

If SCMS is disregarded, all the information from a CD can be transferred to a DAT. The information includes start times and Table of Contents(TOC) which are transferred within the User bits of a SPDIF interface, this is used by a DAT machine to create its own start IDs.


7.With the aid of diagrams explain time compression. List two applications of time compression.

Time compression allows a sample stream to broken into blocks for convenient handling. The diagram shows an Analogue to Digital Conversion(ADC) feeding a pair of RAMs. Whilst one of the RAMs is being written by the ADC, the other RAM can be read. As soon as the first RAM is full, the ADC output switches to the input of the other RAM so that there is no sample loss. The RAM is now read in less time than it takes to write, and the output then pauses until the second RAM is full. This is because the RAM can now be read at a higher clock rate than the sampling rate. The samples are now arranged in blocks with conveniant pauses in between them, rather than an unbroken stream which is difficult to handle. The pauses in between blocks can be used to record error correction words, synchronising patterns or a hard disk could use the pause to move to another track.



Reference. The Art of Digital Audio - John Watkinson


8.Explain the principles of error concealment.

Errors can occur in the digital recording of binary data, a bit is either correct or incorrect with no middle ground. Dropouts or burst errors can occur, which cause a large number of bits in one place to be incorrect. Whether an error is audible depends on which bit of the sample is involved. If the Least Significant Bit(LSB) is in error in a loud piece of music, the effect would not be heard as it would be masked. If the Most Significant Bit(MSB) is in error in a quiet piece of music, the effect would clearly be heard. Error correction works by reversing the state of the binary number, this works because binary only has two states. If the error can't be corrected then the process of Error Concealement is used. Concealement is a process where the value of the missing sample is estimated from nearby samples. The nearby sample may not necessarily be the the same as the original, which in some cases can cause the concealement to be audible. Concealement is made possible by shuffling the sample sequence to separate odd and even numbers prior to recording. The odd and even samples are recorded in different places and recombined into their natural sequence upon replay. Therefore if there is a burst error, only either even or odd numbers will be lost, the result being that every other sample is correct. Only half of the waveform can now be described, but it can be reproduced, though with a little loss of accuracy.

Reference. The Art of Digital Audio - John Watkinson


9.Describe in simple terms the main advantage of an over sampled D-A

The main advantage of an oversampled D-A is that sound will be clearer or better. The Signal to Noise Ratio(SNR) of a CD is fixed at approximately 96dB within a bandwidth of 20 to 20 kHz. Over sampling D-As recalculate intermediate sample values which has the effect of stretching the same SNR out over a much larger bandwidth. The effective noise in the important 20 to 20kHz section is therefore reduced giving a clearer or better sound.


10.Compare the results of too high recording levels in a analogue and digital tape systems.

If the recording level is too high in a digital tape system, clipping occurs. Clipping occurs where the top and bottom of a waveform are chopped off, producing sudden excess distortion. If the recording level is high in an analogue system, the sound can be modified by harmonic distortion. Slight compression also takes place on the analogue signal. Harmonic distortion can be desirable as it can give the output signal a warm sound. Clipping however is not desirable as it does not add anything of use to the signal.
Post Fri Mar 25, 2005 12:15 pm
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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Good  Reply with quote  

Good stuff rico, going well.

For images use:

code:
[img]http://www.pathtoimage[/img]


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Post Thu Apr 07, 2005 7:37 am
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