AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
|
| Week 3 |
 |
|
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
|
Mon Sep 12, 2005 4:23 am |
|
|
rachelh
Joined: 16 Jan 2005
Posts: 35
Location: Trinidad WI |
|
|
|
1. Explain the function of the of the following devices
(a) Anti alias Filter
Aliasing is occurrence of frequencies greater than half of the sampling frequency entering the conversion process. Aliasing is audible during playback and can be identified as harmonic distortion. An anti-aliasing filter can be placed before the analogue to digital conversion to combat aliasing. The anti alias filter itself is a low- pass filter. Theoretically the ideal filter would pass all frequencies up to the Nyquist cut off frequency or Nyquist rate and have an infinite attenuation thereafter. However, in reality such a brick wall filter does not exist hence, a sample rate of a slightly higher value must be applied in order to compensate for the attenuation slope that is necessary for the filter to be effective – example of which is choosing a sample rate of 44.1 kHz to accurately encode a bandwidth of up to 20 kHz. In oversampling the anti-aliasing filter is replaced by a digital decimation filter which delivers more accurate frequency and phase response. [1][3]
(b) Sample & Hold
The term sample and hold describes a function or circuit found in early synthesisers that enable the instantaneous value voltage of a waveform to be captured. This captured sample is then transmitted or held within the set parameters until another course of action is taken. This voltage can be used to control some other parameter in the synthesiser such as a filter, dependant on the usage, a sample and hold circuit can produce pretty random sounding fluctuations in one or more aspects of a sound. [2]
(c) Dither Generator
Signal-to-Error distortion is due to the fact that analogue is continuous and digital is not, hence, in analogue to digital conversion the representative digital word can only be a close approximation to the original signal level – as the quantization process is limited by the number of discrete steps that can be encoded.
White noise- a signal that includes a random distribution of all frequencies at equal levels across the spectrum, introduced in small amounts can reduce the signal-to-error and distortion in a process called dither. With the introduction of dither it is possible to encode signals that are smaller than the Least Significant Bit/ LSB of digital word that is, it is possible to encode signals less than a single quantization step. The process of introducing dither into the conversion process yields better results than if the quantization distortion that would otherwise result was left as is. [1]
2. Explain the operation of copying digitally when the devices are equipped with SCMS
SCMS or Serial Copy Management System [pronounced scums] arose out of the need to protect recordings from being unlawfully re-recorded. On a consumer DAT machine, the 44.1 kHz sampling frequency is often reserved for pre recorded DAT tapes and is designed to discourage this illegal recording. It is actually a digital protection flag that is encoded in Byte 0 [bits 6 and 7] of the S/PDIF’s sub code area. SCMS restricts digital copying by allowing for three flag states –
Status 00 -copy permit [allows for a recording to be freely copied],
Status 11: Copy restrict [allows for only one copy of the recording to be made] and
Status 10: Copy prohibit [which forbids digital copying]
When copying digitally with devices that are equipped with SCMS the flag status plays an important role in allowing or disallowing the copy and if copying is permitted – how many copies can be made.
3. Describe the functions of the contents of a sub frame of MADI
MADI – Multichannel Audio Digital Interface is used for communicating eight or more digital audio channels from one device to another. 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 192 bit)
[4]
4. Describe the Master Clock system of synchronisation of a digital signal chain
A master clock is a dedicated master synchronization signal, which controls all timing clocks in a signal chain. The master clock transmits pulses at a rate of 24 times per quarter note [24 pqq] the purpose of this clock is to transmit this timing to all devices in the digital signal chain to improve the system’s timing resolution and simplify timing when working with non-standard meters such as 3/8, 5/16, 5/32 and so on. The purpose of the master clock as a digital audio source is to ensure jitter free performance during the recording and playback processes. Hence the master clock helps prevent deterioration in the signal clock path caused by either voltage loss or variations in impedance, which lead to a reduction of quality at the conversion process. [1][2]
5. With regard to a CD-Recordable system explain the following operations
(a) Single Session
A single session or Disc at Once is recorded uninterrupted, the data is written to the disc in one session then the table of contents is written after which the disc is closed and ready to be played on a conventional player
(b) Track at Once
Track at once refers to the process by which each track is written on the disc individually - the laser recalibrates after each track, only after the Table of Contents is written can the disc be played on a conventional player.
(c) Multi Session
Multi Session refers to the process by which data can be written onto the disc in more than one session. Each track is written separately the first session can be read in any drive, but successive sessions can only be accessed by a multi-session capable drive. The table of contents 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
The information concerning a CD that can be transferred to a DAT copy disregarding any SCMS considerations would be the table of contents, the start times of each individual track, as well as any other information.
7. With the aid of diagrams explain time compression. List two applications of time compression.
Time compression is when the data rate of the disk drive exceeds the sampling rate this allows for the hard disk to be capable of recording audio. Time compression allows for data to be saved and read in a fraction of real-time and resultantly provides for more disk usage. Two applications of time compression would be to record samples that take up vast amounts of memory as usually one accrues hundreds of samples, also in the editing process as time compression greatly speeds up data retrieval.
8. Explain the principles of error concealment.
Errors occur in the analogue to digital conversion process because in quantization, the number of discrete steps that con be encoded within a digital word limits the accuracy of the process. Dither - White noise- a signal that includes a random distribution of all frequencies at equal levels across the spectrum, introduced in small amounts can reduce the signal-to-error and distortion. Linear Interpolation also known as averaging is another method of error concealment. This method looks at the values before and after the bad digital word and replaces the error with the average of those values. This method works well when there is only a single error, multiple consecutive errors become problematic when using this means. [1][2]
9. Describe in simple terms the main advantage of an over sampled D-A
The main advantage of an over sampled D-A would be minimised intermodulation and distortion at the conversion process resulting in a product that is more accurate that is of a better sound quality. The more samples taken of the original source the better chance of sonic accuracy.
10. Compare the results of too high recording levels in an analogue and digital tape systems.
The results of too high recording levels in analogue and digital tape systems are as follows:
In analogue too high recording levels would lead to harmonic distortion whilst in its digital counterpart clipping [when the dynamic range of the sound exceeds the dynamic range of the equipment] will occur. Recording levels can be adjusted to combat these results in both the analogue and digital domains pre recording, after which compression can be used to combat the problem.
+++++++++++++++++++++++++++
Reference:
1. Modern Recording Techniques – DM Huber, R Runstein
2. www.sweetwater.com
3. Sound and cording [an introduction] F Rumsey, T McCormick
4. http://www.nt-instruments.com/X0-ASP-pLngCateId_219-pIntLevel_4-pLngPageId_661-X1-default.htm |
|
Sun Sep 18, 2005 8:10 am |
|
|
|
|
Polarman
Joined: 24 Jun 2005
Posts: 55
Location: Barbados |
| Week 3 |
|
|
Hi Cris! I really have problems with questions number 6. I suppose you must get down in detail of what is really transfered from the CD to the DAT. Or am I making easy things hard? Maybe Rachel can help.
1. Explain the function of the of the following devices
Anti alias Filter
Anti alias Filter is a low-pass filter applied to the input signal of a digital system prior to the digitisation process. This filter ensures that spurious signals (alias signals) resulting from the digitisation process are not contributing components of the sampled signal.
Example
If your sampling system operated at a 40kHz sampling rate and the original audio input happened to contain a signal at 30kHz (which would be inaudible to humans, but could still be present), the lower image of that signal would appear at 10kHz. Not only would this be clearly audible, it would also be impossible to separate the unwanted image frequency from the wanted audio band.
http://www.fhwa.dot.gov/environment/noise/measure/chap2.htm
http://www.maxim-ic.com/appnotes.cfm/appnote_number/928
http://www.soundonsound.com/sos/may98/articles/digital.html
http://www.tvhandbook.com/support/pdf_files/audio/Chapter8_2.pdf
Sample & Hold
The sample and hold is an analog device required to create a very short time window during which the input-voltage level is tracked and then sampled, and a much longer window during which the sampled voltage is held at the sample-and-hold output. The ADC turns the held voltage into a quantised number.
http://www.tvhandbook.com/support/pdf_files/audio/Chapter8_2.pdf
Dither Generator
The dither generator produces a noise signal that is added to the input so that the effects of quantisation error are minimized. The dither must be added to the signal prior to filtering and digitisation. Since it is broadband, it must be filtered along with the signal.
A properly designed dithering can greatly improve the perceived image or sound quality, but a poorly selected one can make the quality worse.
http://www.mtsu.edu/~djbrown/Conversion.html
http://mlab.uiah.fi/~eye/mediaculture/noise.html
2. Explain the operation of copying digitally when the devices are equipped with SCMS
SCMS stands for "Serial Copy Management System" and is the way copies of digital music are regulated in the consumer market. It prevents more than one generation of digital copying. It is implemented through information that is added to the stream of data that contains the music when one makes a digital copy (a "clone"). When making an analog copy only the music is transferred so there is no SCMS, and copying is totally unrestricted.
Example
You can make 20 digital copies of a CD - you just can't copy digitally any of the 20 copies.
This protection is achieved through the use of the channel statue bit in the consumer format (SPDIF). The receiving device decide if you can copy or not. The codes for the status bites:
00=unlimited copies,
11=1 more copy,
10=no more copies.
http://www.dilettantesdictionary.com/index.php?start=26&let=s
http://www.solorb.com/dat-heads/pbscms.html
http://www.minidisc.org/faq_sec_6.html
http://www.digitalproducer.com/2001/09_sep/features/09_24/cdlaw4.htm
3. Describe the functions of the contents of a sub frame of MADI
MADI was designed for digital connection between multitrack recorders and mixing consoles. The standard provides 56 simultaneous digital audio channels. MADI takes the sub frame structure of the AES/EBU interface and multiplexes 56 of these into one sampling period instead of the original two. MADI has a constant bit rate of 100 megabits per second instead of the variable bit rate found in the AES/EBU. The sub frame data is identical to AES/EBU interface but the first 4 bits differs.
Bit 0 =Frame Sync (Channel 0 = 1, Channel X = 0)
Bit 1 = Channel On/Off (Channel on =1, Channel off= 0)
Bit 2 = Channel A/B (Transparent to AES/EBU, left or right channel)
Bit 3 = Block sync (AES/EBU channel block sync)
4. Describe the Master Clock system of synchronisation of a digital signal chain
The major difference between high-end and budget converters is the quality, stability, and consistency of the internal clock circuitry. Its very important that a digital chain works with the same reference clock. The master clock is giving pulses to all digital devices in the digital chain. You should assign one clock as the master clock and the rest to slave. If you don’t have a stable clock you will have jitter.
The master clock that drives an A/D converter must be very stable. A jittery master clock in an A/D converter can cause irrevocable distortion and/or noise which cannot be cancelled out or eliminated at further stages in the chain.
http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=28
http://www.soundonsound.com/sos/Apr03/articles/digitalclocking.asp
5. With regard to a CD-Recordable system explain the following operations
Single Session
Everything that is ever going to be on the disk is placed there at once. All tracks are recorded without ever stopping the laser, and the disc is closed. Hence no link blocks and no clicks. A Table Of Content (TOC) is written when closed/finalised, this enables playback on any machine.
Since standard audio CDs and data CDs (CD-ROM disks) are stamped with a predefined image, they are single session. Single Session is also called Disc-at-Once.
Track at Once
Track-at-Once recording is what most recorders and software support today. Each time a track is finished, the recording laser is stopped, and two run-out blocks are written. When the laser is started again to write another track, one link block and four run-in blocks are written.
These blocks don't affect data tracks because you never read *between* data tracks, but they are a problem for audio because in some audio players you might hear a click when the link and run blocks are encountered between tracks. These link blocks may also cause problems if a disc is to be mastered and duplicated at the factory, and many disc replicators refuse or remaster Track-at-Once discs.
http://www.cd-info.com/CDIC/Technology/CD-R/Premastering/DAO.html
The problem with track-at-once CDs is that there is a digital gap before each track selection and this interruption in digital data causes excessive errors during the glass mastering.
http://musicguy.com/mastering_articles.html
Multi Session
Using multiple sessions means that information can be written to the first part of the disk, and then later more information can be appended to it in the unused space left after the first session.
When a disk is written a piece at a time, it is necessary to "update" the table of contents (TOC) each time a new section is written, to reflect the new contents of the disk. Drives that support multi-session disks are programmed to seek out these multiple tables of contents that can occur in various places on the disk. The first session can be read on all drives but not the subsiding ones.
6. Explain what information concerning a CD can be transferred to a DAT copy disregarding any SCMS considerations
I had a lot of headache with this question…..since
DAT sub codes are organized and stored in an entirely different manner then on a CD. They are formatted differently on the SPDIF interface as well.
I suppose you get the start IDs but not the tracking number or TOC.
7. With the aid of diagrams explain time compression. List two applications of time compression.
The ADC outputs an unbroken stream of real-time samples. This stream can be hard to handle so time compression is used to break up the stream into arranged blocks with pauses in between them.
The stream that comes from the ADC is written into one of two RAMs. The stream is written with the sampling rate by the write clock. When the first RAM is full it starts to write into the second RAM and at the same time reads out the first RAM with a higher frequency clock. Since the RAM is read faster then written the first RAM will be empty before the second RAM is full. These time slots are used in different ways.
Examples
1. In a DAT recorder to reduce the tape wrap
2. A hard disk might move to another track
8. Explain the principles of error concealment.
When recording digital audio errors can occur. Either it can be a small noise impulse that can cause some individual bits to be in error or it can be a dropout that causes a large number of bits to be in error. This bigger error is called burst error. The audibility of a bit error depends of course which bit of the sample is involved. If it is LSB on a loud passage the error would not be noticeable but if it is the MSB in a quite passages you could hear the error quite clearly. In digital audio the amount of errors that can be corrected depends on the amount of redundancy. Corrected samples are indistinguishable from the original. If the errors are more then your redundancy can handle a correction is not possible but to do as small damage as possible error concealment is used.
Prior to recording the sample stream the odd and even samples are separated. They may even be recorded in separate places to avoid that a burst error affects both the odd and the even samples. On replay the odd and even samples are put together as the original sequence. If an error has occurred you will at least have every other sample correct. The incorrect samples are recalculated. If sample 7 and 9 are correct and sample 8 is incorrect you can calculate sample 8 as (7+9)/2.
9. Describe in simple terms the main advantage of an over sampled D-A
The D-A step is very important in digital audio.
There are two main advantages to over sample a DA.
A) Oversampling is widely used in the DAC. The effects of over sampling at the DAC are advantageous to the design of the analog reconstruction filter that must be built. By having a high sample rate out of our DAC we can use a very simple analog filter. This is important since we will be able to design an analog filter that is not only cheap hardware wise, but also has a nice linear phase response over the pass band.
B) Another reason for over sampling is to reduce the effects of quantisation noise. By over sampling, we can spread any quantisation noise over a larger bandwidth while keeping our signal of interest in the same band. Our filter will serve to cut out the out-of-band quantization noise while keeping our original signal and thereby increasing our SNR.
http://www.earlevel.com/Digital%20Audio/Oversampling.html
10. Compare the results of too high recording levels in a analogue and digital tape systems.
In a digital recording system there is no “gradually” distortion when it clips it clips. This happens when there are no more bits available at 0dBFS. The top or the bottom of the wave form is chopped of / becomes square.
An analog tape gives gradually more harmonic distortion and level compression.
_____________________________________________
Books
Rumsey, F & McCormick. (2004)
Sound and Recording, An introduction
, Fourth Edition, Focal Press
Watkinson, John. (2005)
The Art of Digital Audio,
Third Edition, Focal Press |
|
Sun Sep 18, 2005 3:31 pm |
|
|
AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
|
| CD to DAT |
|
|
ok GOOD "Hi Cris! I really have problems with questions number 6. I suppose you must get down in detail of what is really transfered from the CD to the DAT. Or am I making easy things hard? Maybe Rachel can help. "
Did you become familiar with this now?
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
|
|
Sun Oct 30, 2005 2:40 pm |
|
|
|
|
|
|
All times are GMT. The time now is Fri May 16, 2008 3:47 pm
|
|
|
|
| |
