AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 7 - Big week |
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A big week!
1. Was DAT originally intended as a professional or a domestic recording medium?
2. What is the sampling rate of standard DAT?
3. What is the resolution of standard DAT?
4. What is 'azimuth recording'?
5. Describe the head wheel in DAT recorder.
6. What is SCMS?
7. What is the distinguishing feature of a DAT machine capable of near-simultaneous off-tape monitoring?
8. What is the sub-code area of the DAT tape used for?
9. What is 'interleaving'?
10. What is the width of the tape used for 24-track DASH?
11. What is the width of the tape used for 48-track DASH?
12. Describe how 24-track and 48-track DASH machines are compatible.
13. How are DASH tapes edited?
14. In DASH, why does a playback head come before the record head in the tape path?
15. Comment on the cleaning requirements of DASH How many tracks does a modular digital multitrack (MDM) have?
16. How can more tracks be obtained?
17. Comment on the types of usage of ADAT and DTRS machines?
18. Why is perceptual coding necessary?
19. Describe briefly the use of perceptual coding in the following:
Internet audio
Film sound
DVD-Video
Digital television
Personal stereo
20. What can be done, other than perceptual coding, to reduce bitrate?
21. What is masking?
22. How do perceptual coding systems handle signals that are probably going to be masked by other audio?
23. What is Huffman coding?
24. What is the typical bitrate for an MP3 file intended for Internet distribution?
25. What is the bitrate of Dolby AC3 as used in film sound?
26. What is metadata?
27. Describe and explain the following disk editing systems concepts and their operation:
• Comparison with analogue and digital tape editing
• Features, functions and parameters of disk editing systems
• Compatibility between systems of different manufacturers, and between earlier and later versions of the same system
• Operational procedures
• Automation and recall
• Plug-ins
28. What was the original purpose of timecode?
29. Briefly describe a synchronized system, using the words 'master', 'slave' and 'chase' in your answer.
30. Why is the reliable synchronization of audio not possible using MTC (MIDI timecode)?
31. If the speed of a digital recorder is varied due to being controlled by a synchronizer, what must happen to the sampling rate of the output? Give two answers.
32. From where does a digital recorder derive its clock in a synchronized system? Give two answers.
33. Briefly describe the synchronization of multiple ADAT machines (or DTRS).
34. In a synchronized system consisting of a sequencer and a drum machine, where the drum machine is the slave, what setting needs to be made on the drum machine to allow it to synchronize to the sequencer?
35. Before the introduction of song position pointers, what was likely to happen if the master was started at some point other than the start of the sequence?
36. What information do song position pointers provide?
37. Briefly, what is 'LTC'?
38. What is 'latency'?
39. What is 'zero latency monitoring'?
40. What is the drawback to zero latency monitoring?
41. What is DSP?
42. What is the advantage of using a system with DSP?
43. Describe the relationship between the number of plug-ins in use and the available processing power of the computer?
44. In relation to plug-ins, what is an 'instance'?
45. What is an 'insert effect'?
46. What is a 'bus effect'?
47. What problem might be caused by the latency of a plug-in?
48. What is an 'interleaved' sound file?
49. What is 'bouncing'?
_________________
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 Apr 18, 2005 3:11 am |
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Rico1210
Joined: 03 Aug 2004
Posts: 39
Location: Newcastle, UK |
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Hi,
Heres my answers for week 7. Had a bit of trouble with number 31 
1. Was DAT originally intended as a professional or a domestic recording medium?
DAT was originally intended as a consumer-recording medium. It was to replace the cassette recorder.
2. What is the sampling rate of standard DAT?
DAT recorders offer a range of sampling rates that include professional rates of 44.1 and 48 kHz. An original consumer DAT recorder operates at 48 kHz to avoid digital copying of CDs, whereas professional and modern machines offer both.
3. What is the resolution of standard DAT?
Standard DAT resolution is 16 Bit.
4. What is 'azimuth recording'?
Azimuth describes the orientation of the head gap with respect to the tape. The gap should be perpendicular to the edge of the tape otherwise high frequencies are not efficiently recorded or replayed because the head gap becomes wider. If the azimuth is incorrect, the relative phase between tracks is changed.
Azimuth of the replay head is adjusted quite regularly, especially when replaying tapes that have been recorded on a different machine with a different azimuth. Azimuth of the recording head is not modified unless there is reason to believe that it has changed.
5. Describe the head wheel in DAT recorder.
A DAT recorder uses a rotary head (hence the name R-DAT) like a video recorder. When recording, it lays the tracks diagonally across the width of the tape unlike analogue tape, which records the signal parallel to the tape edge. Unlike a video recorder, on which tape is wrapped almost all of the way around the head so that there is always contact between the two, tape on a DAT is only wrapped a quarter of the way around the head. At times the tape does not touch the head, collecting data in a buffer to be read out at a constant speed compensates for this. The head wheel in a DAT recorder spins at 2000 rpm.
6. What is SCMS?
SCMS(Serial Copy Management System) is a system used to enforce copyright laws of certain material. It enables copyrighted material to only be copied once. DAT recorders in professional use are exempt from SCMS as they accept a different type of digital input.
7. What is the distinguishing feature of a DAT machine capable of near-simultaneous off-tape monitoring?
The distinguishing feature of a DAT machine is an extra pair of heads can be mounted on the drum. Simultaneous off-tape monitoring can be performed during recording just like a three-head analogue tape recorder
8. What is the sub-code area of the DAT tape used for?
Subcode is an auxiliary data stream, which is merged with audio samples, and has numerous functions. A DAT can carry 273 kbits per second of subcode, which is about four times the amount a CD contains. The sub-code area of the DAT tape is where the time is recorded. The time from the beginning of the tape is recorded in every track, therefore if you start recording in the middle of a tape, the machine reads the time at that spot and keeps recording time sub-code appropriately. ID's are also recorded in the sub-code. These are points that contain digital code that can be found easily. They are Start ID's, Skip ID's (instruct the deck to find the next start point) and End ID's. Program numbers can also be recorded in the sub-code. These are numbers searched for during previous play operations. Sometimes Program numbers are included with each Start ID. The subcode area of a DAT also contains the AES/EBU channel status message.
9. What is 'interleaving'?
Interleaving is when the order of samples is shuffled. Samples that were previously adjacent to each other in real time, are separated from each other on the tape. Interleaving is used in digital recording to rearrange the original order of samples, which can convert burst errors into random errors when the samples are de-interleaved. This helps to conceal errors.
10. What is the width of the tape used for 24-track DASH?
The width of the tape used for 24-track DASH(Digital Audio Stationery Head) is half an inch.
11. What is the width of the tape used for 48-track DASH?
The width of the tape used for 48-track DASH(Digital Audio Stationery Head) is half an inch.
12. Describe how 24-track and 48-track DASH machines are compatible.
24-track and 48-track DASH machines are compatible because they use the same tape. If the tape is originally recorded on a 24-track machine it will leave guard bands between the tracks. If it is decided that more tracks are needed the tape can be moved to a 48-track machine, which will record tracks in the 24 guard bands.
13. How are DASH tapes edited?
There are two ways to edit DASH tapes. The first is the cut-and-splice technique. The tape should be edited with a 90-degree butt joint. Circuits are incorporated in the machine to detect a splice and perform an electronic cross-fade from one side to the other, to conceal the edit. The other method of editing is Electronic editing. This is achieved using two machines plus a control unit. A master tape is assembled from takes of the player machine copied in real time.
14. In DASH, why does a playback head come before the record head in the tape path?
The playback head comes before the record head in the tape path because where an analogue process will happen instantaneously; a digital process takes a little time. The playback head comes first to overcome the delay that would be created whilst the signal is being processed, when recording overdubs. There must be a playback head upstream of the record head otherwise multitrack recording wouldn't work.
15. Comment on the cleaning requirements of DASH How many tracks does a modular digital multitrack (MDM) have?
DASH machines should be cleaned by an expert in the field of DASH cleaning as someone can do thousands of pounds worth of damage if they don’t know what they are doing. The heads are cleaned with a chamois-leather cleaning tool, wiping in a horizontal motion only. Cotton buds, which are used for analogue cleaning, will clog the head with their fibres. The alignment of a DASH machine should be done every 6 months.
A modular digital multitrack (MDM) has 8 tracks.
16. How can more tracks be obtained?
More tracks can be obtained by synchronizing multiple machines.
17. Comment on the types of usage of ADAT and DTRS machines?
ADAT machines are very affordable and are popular in budget music recording studios. DTRS machines are used in broadcasting and post-production of films.
18. Why is perceptual coding necessary?
Perceptual coding is necessary to reduce the bit rate of a signal and improve the representation . Perceptual coding removes parts of the audio signal that cannot be heard by the human ear. These parts are unnecessary as they are beyond the limitations of human perception.
19. Describe briefly the use of perceptual coding in the following:
Internet audio
MPEG Layer-3(MP3) and MPEG-4 AAC are modern perceptual coding techniques that are used in internet audio. These techniques exploit the limitations of the human ear and the perception of sound to reduce the bit rate with little or no perceptible loss of quality.
Film sound
The ATRAC(Adaptive Transform Acoustic Coding) system is a perceptual coding system that can be found in SDDS(Sony Dynamic Digital Sound), SDDS is used widely in cinema audio. ATRAC encodes the audio data, making a substantial saving in bit rate. It does this by splitting the incoming signal into three bands (below 5.5kHz, 5.5-11kHZ and above 11kHZ) and individually analyzes the frequency content and level of each. Dolby Digital (AC-3) is another perceptual coding technique used in film sound. Dolby AC-3 is used in the cinema at a bit rate of 640 kbps.
DVD-Video
The perceptual coding systems used in DVD-Video are: Dolby Digital AC-3, DTS (Digital Theatre Sound) and MPEG-2. THX is also in 95% of the cases based on AC-3 installations, and is used on laserdiscs and DVDs at a bit rate of 384 kbps. All NTSC DVD players will replay Dolby Digital AC-3, and newer models will also replay DTS. DVD players in Europe may use MPEG-2 multi channel audio for encoding up to six channels.
Digital television
The MPEG 2 perceptual coding system is the standard in the digital television industry. It is used to broadcast digital video accross cable, satellite and other channel.
Personal stereo
Sony's MiniDisc player uses a form of perceptual coding in its personal stereos. The ATRAC(Adaptive Transform Acoustic Coding) system compresses compact disc audio to approximately 1/5 of the original data rate with virtually no loss in sound quality. ATRAC encodes the audio data, making a substantial saving in bit rate. It does this by splitting the incoming signal into three bands (below 5.5kHz, 5.5-11kHZ and above 11kHZ) and individually analyzes the frequency content and level of each. (PASC)Precision Adaptive Sub-band Coding is a perceptual coding system used in Phillips Digital Compact Cassette(DCC) that achieves 75% data reduction.
20. What can be done, other than perceptual coding, to reduce bitrate?
Predictive coding can be used to reduce bit rate. In all signals, part of the signal is obvious from what has gone before and what may come after. Most signals have a degree of predictability, for example a sign wave is highly predictable because all cycles look the same. A signal can be transmitted with parts omitted(encoded), provided that there is a suitable decoder that can predict the omissions from the previous and next data. All encoders must contain a model of the decoder to be safe in the knowledge that the information will be correctly re-created. Predictive codecs contain two identical predictors, one in the coder and one in the decoder. These predictors examine the previous data values and estimate what the next value will be. The estimated value is then subtracted from the actual next value to produce a prediction error(residual) that is transmitted from the encoder. The decoder receives the prediction error and adds it to its own prediction, which produces the output code value. There will be no loss of information provided the prediction error(residual) is transmitted intact. However, not all signals can be correctly predicted. All codecs have difficulties with noise, as noise is totally unpredictable.
21. What is masking?
The human hearing system is not equally sensitive at all frequencies; it is less so at low and high frequencies. The perception threshold of hearing is raised at a particular frequency in the presence of another sound at a similar frequency. In simpler terms, when two sounds of a similar frequency are played but only one can be heard, the first sound 'masks' the second. The perception threshold is raised, so a sound must be louder to be heard. Masking has its uses in audio engineering and is widely used in noise reduction. For instance, low-level noise that exists in the same frequency band as a high-level music signal will be masked by the music. Masking is also used in digital compression systems, as it allows the use of lower resolution in frequency bands where the noise will be masked by the signal.
22. How do perceptual coding systems handle signals that are probably going to be masked by other audio?
Perceptual coding systems remove the parts of the signal that are unperceivable to the human ear. The remaining signal is then processed with more available bits, as there is no wastage on masked parts of the signal . This process is called dynamic bit allocation.
23. What is Huffman coding?
Huffman coding uses a system similar to morse code where variable-length coding is used in which frequently used values are allocated short codes and values that occur infrequently are allocated long codes. The probability of the code value is studied, and the most frequent values are transmitted with short wordlength symbols. As the probability of the value falls, it will be allocated a longer wordlength.
24. What is the typical bitrate for an MP3 file intended for Internet distribution?
The typical bit rate for an MP3 file intending for internet distribution is 128kbit. Although the bit rate of 192kbit is becoming popular in file sharing networks.
25. What is the bitrate of Dolby AC3 as used in film sound?
Dolby AC-3 is used in the cinema at a bit rate of 640 kbps. THX is also in 95% of the cases based on AC-3 installations, and is used on laserdiscs and DVDs at a bit rate of 384 kbps.
26. What is metadata?
Metadata can be described as data about data. For instance, the index of a book or the indexes of books in a library are metadata.
27. Describe and explain the following disk editing systems concepts and their operation:
• Comparison with analogue and digital tape editing
Digital consoles have the benefit of automation systems with motorised faders, which is a feature rarely seen on analogue consoles.
Unlike analogue consoles, digital consoles can be in constant use as any settings may be stored in the scene memory or on disk to be recalled later.
Digital mixers often include an LCD video display of the settings, up to a high-resolution 320 x 240. This is very rare in analogue mixers, as the magnetic field from the scan coils tends to break into the audio circuits.
• Features, functions and parameters of disk editing systems
Digital Mixing Consoles come in all different sizes. 24, 36, 56, and 72 channel models with moving fader automation and come with a choice of working at 44.1 kHz, 48 kHz, 88.2 kHz, or 96 kHz. The preferred operation being 24-bit/96kHz. They come with auxilary/insert sends for effects processors, surround monitoring capabilities and motor driven, touch sensitive faders. Input channels usually have independent compression and gating/ducking processors for dynamics control, as well as 4-band parametric channel equalizers. Digital mixers usually come with an LCD display, expensive models may have touch screen technology.
Digital Consoles often come with a preset library of effects, compression, gating, EQ, I/O patching, and channel setups. User setups can be added to the libraries for instant recall whenever they are needed. Studio Manager software is also usually included, which makes it easy to relate on-screen controls to the corresponding console functions. Digital Mixing Consoles also tend to have a small number of user-defined keys that can be assigned to software control functions.
Digital mixers can have an internal wordlength of up to 32 bits to accommodate sample attenuations. If a sample is attenuated, it will develop bits that are below the radix point. If 24dB attenuate an eight-bit sample, the sample value will shift four places. To accommodate such a shift, extra bits must be available.
• Compatibility between systems of different manufacturers, and between earlier and later versions of the same system
Manufacturers approach disk editing systems/mixing consoles in different ways. Compatibility problems do occur between systems from different manufacturers. This is because not all systems will use the same data reduction techniques, if a project is transferred to a system that does not use the same techniques then it will not be usable. Also, effects plug-ins can cause compatibility problems. If a project contains certain types of plug-ins that are not available on another system, the outcome will be a lot different.
• Operational procedures
Instruments, drum machines, samplers, synthesisers are connected to the line-level inputs of the channel. Performers microphones are connected to microphone preamplifiers of the channel. Auxiliary/insert sends can be used to route the signal to effects units and processors, or to performers headphones for foldback. Ouptputs can be sent from each channel to multitrack software or a multitrack recorder.
• Automation and recall
The automation of an analogue mixing console uses a system where the faders produce a varying voltage and this is converted to a digital code or gain coefficient in an ADC(Analogue to Digital Converter) and recorded alongside the audio tracks. On replay the coefficients are converted back to analogue voltages which control VCAs(Voltage Controlled Amplifiers) in sequence with the analogue audio channels. A digital mixer uses a similar system to that of an analogue mixer, whereby the coefficients are obtained in the same way, but are not treated the same upon replay. In a digital mixer the coefficients are not converted back to analogue to control VCAs, they remain in the digital domain and control multipliers in the audio channels directly.
Digital audio processing is by program control, therefore the desk configuration can be changed by running programs for various functions in a different order.
The configuration and the settings of all controls can be stored in memory or on disk, and recalled instantly.
• Plug-ins
Disk editing systems often have built in effects processors that include, eq’s, compressors, reverbs and delays and some have on board support for VST plug-ins. Disk editing systems often have expansion slots for more plug-ins cards and networking cards.
28. What was the original purpose of timecode?
The original purpose of timecode is as a synchronising mechanism, to enable multiple devices to all work together in sync.
29. Briefly describe a synchronized system, using the words 'master', 'slave' and 'chase' in your answer.
Synchronising is the process of making two devices operate together as one. One of the devices will tell the second device when to start, when to stop, and how fast to play. This is the master device. The device that is being controlled by the master is called the slave. The slave will follow in sync with the master device, stopping and starting when told to do so. If the master device is started from a midway point in a sequence, the slave device will have to chase the master device to reach the start point.
30. Why is the reliable synchronization of audio not possible using MTC (MIDI timecode)?
Reliable synchronisation of audio is not possible using MIDI timecode(MTC) because it does not contain a clock pulse. MTC consists of messages sent every quarter-frame that build up over two frames into a complete frame address. MIDI timecode is not as reliable as SMPTE timecode.
31. If the speed of a digital recorder is varied due to being controlled by a synchronizer, what must happen to the sampling rate of the output? Give two answers.
Had a bit of trouble with this one.
32. From where does a digital recorder derive its clock in a synchronized system? Give two answers.
In a synchronised system a digital recorder will sync to the other device/s in the system, which could be a computer or sequencer. A digital recorder can sync to SMPTE timecode or MTC (MIDI Timecode).
33. Briefly describe the synchronization of multiple ADAT machines (or DTRS).
A S/PDIF cable can be used to synchronise multiple ADAT machines. The cable should be connected to the sync output on the master machine, to the sync input on the slave machine. ADAT recorders can be synchronised to obtain more tracks.
34. In a synchronized system consisting of a sequencer and a drum machine, where the drum machine is the slave, what setting needs to be made on the drum machine to allow it to synchronize to the sequencer?
In a synchronized system consisting of a sequencer and a drum machine, where the drum machine is the slave, the MIDI sync option should be set to allow the drum machine to synchronize to the sequencer.
35. Before the introduction of song position pointers, what was likely to happen if the master was started at some point other than the start of the sequence?
If the master were started at some point other than the start of the sequence before the introduction of SPPs, the sequence would start from the beginning. A song position pointer (SPP) represents a position in a song, in terms of MIDI beats from the start. Before this was introduced there was only one position the audio could play from.
36. What information do song position pointers provide?
A song position pointer (SPP) represents the position in a song, in terms of MIDI beats from the start. Song position pointers are messages sent from the master device to inform instruments at which position of the song they should be.
37. Briefly, what is 'LTC'?
LTC (Longitudinal TimeCode) is a system created by the Society of Motion Picture and Television Engineers that is also known as SMPTE code. It is basically an encoded signal recorded onto the audio track of a tape that contains registered time information in hours, minutes, seconds and frames.
38. What is 'latency'?
Latency can be defined as the time it takes a packet of data to get from one designated point to another. The time it takes the data to reach the designated point then return to the sender is considered to be the latency. In other words, latency is delay.
39. What is 'zero latency monitoring'?
Zero Latency Monitoring refers to the technique of routing the input signal directly to the output on the audio card. This enables a performer to obtain perfect feedback without latency problems. It has become one of the most important features of modern, host based hard disk recording.
40. What is the drawback to zero latency monitoring?
The drawback of using zero latency monitoring is that because the signal is sent directly from the input to the output of the soundcard, the signal can't be processed by effects software or plug-ins.
41. What is DSP?
Digital Signal Processing. DSP converts analogue signals to digital for them to be manipulated or processed. DSP chips are widely used in sound cards, fax machines, modems, mobile phones and digital TVs.
42. What is the advantage of using a system with DSP?
DSP chips process information almost instantaneously. The advantage of using a system with Digital Signal Processing is the speed that signals are processed is a lot quicker, which reduces latency.
43. Describe the relationship between the number of plug-ins in use and the available processing power of the computer?
Host based processing uses a PC's processor to complete the calculations needed for using plug-ins such as effects. The amount of plug-ins that can be used at any one time is limited by the power of the host-based processor, although more can be added if the buffer is increased.
44. In relation to plug-ins, what is an 'instance'?
An instance, in relation to plug-ins, is the number of times that plug-in is being used. The same plug-in may be used twice or three times, therefore there are two or three instances of that plug-in.
45. What is an 'insert effect'?
Insert effects are those effects that you route the instrument's sound through before going to the mixer. Typical insert effects are compression, distortion and filters.
46. What is a 'bus effect'?
Bus or Send effects are connected to the mixer's effects send so that more than one device can be routed to it. Reverb and delay are the typical send effects.
47. What problem might be caused by the latency of a plug-in?
Problems with the synchronisation of audio tracks can be caused by the latency of plug-ins. If a plug-in is affected by excess latency, it will be delayed slightly compared to the other audio tracks.
48. What is an 'interleaved' sound file?
An interleaved sound file is a digital recording where the left and right channels are mixed together as one block of data.
49. What is 'bouncing'?
Bouncing is when a number of tracks are grouped together onto a smaller number of tracks. For instance, a drum kit may have been recorded separately onto 5 or 6 tracks. These tracks can be grouped together into smaller tracks, maybe 2 tracks. This enables more playback tracks to be available for other instruments, instead of taking up 5 or 6 tracks they now only take up 2. V-tracks of the original recordings can be kept incase it is later felt that individual changes are needed. These can again be grouped or 'bounced' down to the required number of tracks.
http://www.mp3-tech.org/ac3.html
http://www.ti.com/corp/docs/investor/ar97/advantages.htm
http://www.minidisc.org/MaskingPaper.html |
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Sun Apr 24, 2005 4:29 pm |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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quote:
4. What is 'azimuth recording'? Azimuth describes the orientation of the head gap with respect to the tape. The gap should be perpendicular to the edge of the tape otherwise high frequencies are not efficiently recorded or replayed because the head gap becomes wider. If the azimuth is incorrect, the relative phase between tracks is changed.
Azimuth of the replay head is adjusted quite regularly, especially when replaying tapes that have been recorded on a different machine with a different azimuth. Azimuth of the recording head is not modified unless there is reason to believe that it has changed.
Caution here, this is referring to a digital audio systme of tape head alignment. I'd like you to check this one again.
quote:
31. If the speed of a digital recorder is varied due to being controlled by a synchronizer, what must happen to the sampling rate of the output? Give two answers.
Had a bit of trouble with this one.
OK this still needs to be worked out....
quote:
42. What is the advantage of using a system with DSP?
DSP chips process information almost instantaneously. The advantage of using a system with Digital Signal Processing is the speed that signals are processed is a lot quicker, which reduces latency.
Good though also mention the reduction in processing needed for say the host CPU.
--
An excellent week there Rico, lots of work there I can see, you are getting very good at answering. Just clear up the above comments and we are good.
_________________
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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Tue Apr 26, 2005 4:39 am |
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