AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 7 - Questions |
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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 Oct 10, 2005 3:50 am |
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Polarman
Joined: 24 Jun 2005
Posts: 55
Location: Barbados |
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1. Was DAT originally intended as a professional or a domestic recording medium?
DAT was intended for a domestic recording medium replacing the cassette standard.
2. What is the sampling rate of standard DAT?
The standard sampling rate of a DAT is 48 KHz.
3. What is the resolution of standard DAT?
The standard resolution of a DAT is 16 bits.
4. What is 'azimuth recording'?
Other words for azimuth recording is guard-bandless or chevron recording. With azimuth recording tracks can be recorded side by side without a spacing in this way it increases recorded density. The read head gap can be wider than the recorded track so the read head over scans the track, this simplifies tracking requirements. The drum’s head pairs are angled differently from each other with respect to the tape. The two tracks A and B has a azimuth angle of plus minus 20 degrees. The A head then reads the A track and the B track but greatly attenuated due to phase cancellation. This reduces crosstalk between close tracks there is no need for guard band between tracks.
5. Describe the head wheel in DAT recorder.
With the DAT rotary head its possible to have slow linear tape speed while achieving high recording bandwidth. Each track is discontinuously recorded as the tape runs past the angled drum which spins rapidly in the same direction as tape travel. A DAT drum rotates with 2000rpm and has two or four heads 180 or 90 apart. The record/playback signal is recorded or played back 50% of the time and interrupted 50 % of the time.
6. What is SCMS?
Consumer DAT recorders contain a Serial Copy Management System (SCMS) circuit to prevent multiple –generation digital copying of protected material.
SCMS was developed in response to record companies' needs to protect their copyright. SCMS restricts digital copying by setting certain bits in the data stream. There are three options:
- Copy permit - a recording can be copied freely. Only professional machines allow this code to be set.
- Copy restrict - a recording can be copied once, then its SCMS bits are set to Copy prohibit
- Copy prohibit - Digital copying is not allowed.
7. What is the distinguishing feature of a DAT machine capable of near-simultaneous off-tape monitoring?
The distinguishing feature of a DAT machine capable of near-simultaneous off-tape monitoring is that it the recorder has an extra pair of heads mounted on the drum.
8. What is the sub-code area of the DAT tape used for?
Sub Code areas allow extra data to be recorded alongside the audio information. This include:
A-time, which logs the time taken since the beginning of the tape
P-time, which logs the time taken since the last Start ID.
Start ID marks the beginning of each item;
Skip ID tells the machine to go directly to the next Start ID
9. What is 'interleaving'?
Interleaving is when the data bits a are dispersed through the data stream prior to storage or transmission. Upon play back the data stream is de-interleaved or replaced in order. The reason for this is if a burst error occurs it damages a continuous section of data but with interleaving the errors gets distributed through the bit stream and make it possible for error correction.
10. What is the width of the tape used for 24-track DASH?
The width of the tape used for 24-track DASH is ½ inch (12.55 mm) open reel
11. What is the width of the tape used for 48-track DASH?
The width of the tape used for 48-track DASH is ½ inch (12.55 mm) open reel.
12. Describe how 24-track and 48-track DASH machines are compatible.
The 24-track is totally two-way compatible with 48-track They are both forward and reverse compatible. If you record your project on a 24-track and feel that you don’t have enough tracks you can just change to a 48-track and get 24 extra tracks. If you start on a 48 track and only use 24 track you can change to a 24 track and do you mix.
13. How are DASH tapes edited?
DASH was designed to be a cut-and-splice editing format like on a analogue tape but the most used way to edit on DASH is by copying between two machines synchronized together with an offset.
14. In DASH, why does a playback head come before the record head in the tape path?
The playback head on a DASH comes before the record head because digital operations take time while analog processes take place instantaneously. Since you have a delay processing the signal to put it on the tape, when overdubbing the delay is avoided with having the playback head before the record head.
15. Comment on the cleaning requirements of DASH How many tracks does a modular digital multitrack (MDM) have?
A DASH should be cleaned by an expert. To clean a DASH you need special tools and a DASH should not be cleaned with cotton like an analogue tape recorder. A modular digital multitrack (MDM) have 8 tracks.
16. How can more tracks be obtained?
Multiple machines can be easily synchronized to give more tracks.
17. Comment on the types of usage of ADAT and DTRS machines?
ADAT is used in pro-, semipro- and budget music recording studios
DTRS is used in broadcast and film post-production
18. Why is perceptual coding necessary?
Perceptual coding necessary to reduce the amount of data needed to reconstruct a waveform. The PCM is not suitable to for example internet. Perceptual coding uses a psychoacoustic model of the human auditory system to identify imperceptible signal content to remove irrelevant parts (not hear able by human ear) of the waveform. The signal is then coded efficiently to avoid redundancy.
19. Describe briefly the use of perceptual coding in the following:
Internet audio
MPEG Layer III (MP3)
MP3 is developed by Fraunhofer Institute and Thomson Multimedia. Mp3 is used to decrease file size prior to electronic distribution. Files can be uploaded and downloaded over the internet or attached to email. The data must be put through an MP3 decoder for playback, once downloaded the file can be transferred to a solid state playback device such an MP 3 player. Typical compress ratio is 10:1 depending on which compression level is used. There is also MP3 pro which enhances the sound quality and improves the compression scheme. MP3 Prof. splits the coding process in two parts. First part analyses the low frequency band information and encodes it in to a normal MP3 stream the second part analyses the high frequency content. The result is a more compact MP3 file with higher sound quality.
MWA
WMA (Windows Media Audio) is Microsoft’s response to Mp3. WMA can encode high-quality audio at low bit-rate and file size settings. WMA is also used to real time stream audio and lot of stations on the internet is using that to stream to Windows Media Player. WMA also provides a degree of content copy protection.
AAC
AAC (Advanced Audio Coding) is developed by Dolby Labs, Sony, ATT and Fraunhofer Institute. AAC is a secure digital music distribution over the internet and is also good for multi channel formats. It can encode up to 48 channels up to 24/96 in a single bit stream.
Ogg Vorbis
Ogg Vorbis was designed as a substitute to MP3 and WMA. One big advantage with this format is that it is free from royalties. Ogg Vorbis is capable to deliver audio in a different channel formats at both constant and variable e bit-rates.
Real Audio
Real Networks was one of the first companies to stream audio with its Real Player on the Internet. There are several compression levels to choose from. The Real audio server can automatically recognize which modem, cable or network connection speed is currently in use and transit data in the best possible format.
Film sound
Dolby Digital
Dolby Digital (AC-3) is a perceptual coding technique used in film sound. Dolby AC-3 is used in the cinema at a bit rate of 640 kbps. AC-3 can code 1 to 7 channels.
DTS
DTS or Digital Surround is an alternative and competing format to Dolby Digital is DTS Digital Surround, or just "DTS". Like Dolby Digital, DTS is another 5.1-channel surround sound format that is available in movie theaters, and as an optional soundtrack on some DVD-Video movies for home theater viewing. But unlike Dolby Digital, DTS is not a standard soundtrack format for DVD-Video, and is not used by HDTV or digital satellite broadcasting.
THX
The THX Surround EX format is jointly developed by Lucasfilm THX and Dolby Laboratories, and is the home theater version of "Dolby Digital Surround EX, an Extended Surround sound format used by state-of-the-art movie theaters. Lucasfilm THX licenses the THX Surround EX format for use in receivers and preamplifiers.
SDDS
SDDS (Sony Dynamic Digital Sound), SDDS it splits the audio data in three bands below 5.5kHz, 5.5-11kHZ and above 11kHZ and individually uses perceptual codeing for each band.
DVD-Video
Dolby Digital is the standard audio format on DVD
Digital television
Dolby digital
The major DTV standards are ATSC (North America), DVB (Europe) and ISDB (Japan). All three use MPEG-2 video compression and Dolby Digital audio compression. DVB and ISDB also include MPEG audio compression.
Personal stereo
Addtional devices in a personal stereo that uses perceptual coding is Minidisk and DCC (Digital Compact Cassette). Mini disk uses ATRAC (Adaptive Transform Acoustic Coding) and compresses to 1:5.
20. What can be done, other than perceptual coding, to reduce bitrate?
There are two main categories of audio coding lossless and perceptive. The principles of predictive recording is a lossless method which means that the signal being played back is exactly as the signal recorded (minus errors ).
Audio signals are largely repetitive, which is why predictive coding works. The technique involves a 'predictor' which has knowledge of typical audio signal behaviour. By looking at the preceding audio signal, it tries to anticipate what will happen next and, because of audio's repetitive nature, the prediction is generally quite accurate. If this prediction is subtracted from the original signal, only a small difference signal remains, and this is recorded or transmitted as the data-reduced result.
Both the coder and the decoder use the same predictor 'knowledge' to generate/regenerate the predicted signal. The accuracy of this system is entirely dependent on the predictor algorithm. Ideally if the residual is transmitted intact there is no loss of information but in practice typically around 98% of the original signal is retrieved.
The technique works less well in anticipating essentially random signals in noise-like sounds, or in predicting highly unpredictable (but crucial) transients. To improve the precision of the system, therefore, many coders use band-splitting techniques (splitting the whole audio spectrum into four separate frequency bands). This allows multiple predictors to work on simpler band-limited signals with far greater accuracy than they would if handling the complete signal.
One drawback of predictive coding is that since the decoder must use exactly the same predictor as the encoder, improvements to the 'intelligence' of the encoder can only be useful if your decoder is updated too, otherwise the accuracy of the decoded signal will actually suffer.
In general, this kind of system works very well, providing a typical reduction ratio of around 4:1. However, it can prove fatiguing to the listener over long periods, because damaged transient signals require more 'brain power' from the listener to interpret the sound. Multiple passes through the encoding/decoding process also lead to rapid loss of signal quality, very similar to that experienced when copying analogue cassettes.
21. What is masking?
Frequency masking
Our brains do not treat the audio spectrum as a continuum the human hearing we perceive sound through around 25 distinct critical bands of varying bandwidths (at 100Hz the critical band is about 160Hz wide, but at 10kHz it is 2500Hz wide). You can say that humans listen through a 25 band EQ. Quieter sounds will be masked away from louder signals in the same band. This phenomenon is called frequency masking. Although our hearing is incredibly perceptive of simple signals in isolation, in the presence of complex sounds it effectively runs out of 'hearing resources' and so can only perceive the most dominant parts at any particular moment in time.
The hum from a bass guitar amplifier is inaudible while the guitar is playing, although quite evident on its own. Tape hiss is inaudible in the presence of full-range music, but obvious between tracks.
Time masking
A loud sound affects our perception of quieter signals both before and after it sounds. A quiet signal that occurs 10-20 milliseconds before a louder signal, for example, may be masked by the louder signal, this is called backwards masking. The squeak of a kick-drum pedal might be plainly audible on its own, but can be masked by the presence of a much louder bass drum thump which happens a few milliseconds later. The hearing mechanism also takes time to recover from a loud sound, and this creates a masking effect which extends up to 100-200 milliseconds after the masking signal has ceased -- this is called forward masking. The length of the masking is related to the amplitude of the masking signal.
22. How do perceptual coding systems handle signals that are probably going to be masked by other audio?
Perceptual coding removes all signals that are not perceivable to the human ear.
23. What is Huffman coding?
There are many different reasons for and ways of encoding data, and one of these ways is Huffman coding. This is used as a compression method in digital imaging and video as well as in other areas. The idea behind Huffman coding is simply to use shorter bit patterns for more common characters, and longer bit patterns for less common characters.
The Huffman code shares the same principles as the Morse code where frequent letters has short codes and infrequent letters has long codes. In the Huffman coding the probability of different codes values to be transmitted is studied and the most frequent codes are arranged to be transmitted with short word length symbols and less frequent with longer word length symbols.
24. What is the typical bitrate for an MP3 file intended for Internet distribution?
A typical bit rate for an MP3 file intended for Internet distribution is 128 kbit but since more and more people get faster connection 160 to 192 kbps I getting more common.
25. What is the bitrate of Dolby AC3 as used in film sound?
The bitrate of Dolby AC3 as used in film theatre sound is 640 kbps.
26. What is metadata?
In short metadata can be described as data about data. Media content such as audio, video, graphics and son on is sometimes known as essence. Related data to this is metadata and it can hold parameters such as sampling frequency, down mixing and number of channels it can describe how to decode the essence, it can contain intellectual property information such as copyright and ownership.
27. Describe and explain the following disk editing systems concepts and their operation:
Comparison with analogue and digital tape editing
In a digital mixer all incoming analogue signals are converted to digital. In that way all functions on the signals will be done in the digital domain. Digital inputs and outputs make it possible to connect recording devices, processors and effects digitally with no need to convert to analogue. This is a big advantage since the digital signal is more robust then a analogue signal; no crosstalk, unaffected by lead capacitance, electromagnetic fields, distortion and noise. Processing and effects can also be carried out in the digital domain without to convert to analogue. The digital mixer is also more ergonomically different from analog mixers. The digital mixers are normally smaller because there is no need to have as many faders as you have channels, any fader can be assigned to any channel. Digital mixers are normally fully automated and have a total recall which means that you can easily work on many projects at the same time.
Since one knob controls the EQ for all the channels, digital consoles have fewer controls than analog consoles. One knob or switch can have several functions. This makes digital consoles harder to operate than analog ones because, with a digital console, you can’t just reach for an EQ knob for a particular channel. You have to do several button presses to set the EQ parameters.
Digital consoles have built-in effects and automated mixing. You can set up different mixes, store each mix in the mixer’s memory, and recall each mix with the press of a button. When you recall a mix, some mixers make the faders move into the positions you set up. Other mixers do not move the faders when you recall a mix. You have to set them manually by looking at a display, which is a disadvantage.
In an analog mixer, all of the connections, inputs and outputs, are "hardwired". Once the design is finished, the mixer can never change. In a digital mixer, once the audio is inside the mixer, there is virtually total freedom to move it around, add effects, and configure its paths anyway you need to for your application. For example, any input can go to any or all channels. And the various paths within a digital mixer can be routed to many different destinations as well as different physical outputs.
Features, functions and parameters of disk editing systems
A digital mixer usually have 24, 48, 72 or 96 channels and can operate in 44.1 KHz, 48 KHz, 88.2 KHz, 96 KHz and 192 KHz. They work internally with 32 up to 56 bits internally. Most features are totally automated.
Because the audio in a digital console has entered the digital domain, it is easy to add very high quality, on-board digital effects processing. Since this processing is an integral part of the mixer, you avoid the audio losses of cabling, external patching and audio conversions inherent in an analog console.
The internal processing power in a digital mixer can also be reconfigured. For example, you could have a Real Time Analyzer for live concert work or when you are first tweaking the sound in your studio. You could configure your console so you have compressors on 20 or 30 inputs or configure your effects to be a Mastering Tool Kit with multi-band compressors and expanders as you mix digitally to CD or DAT.
To help speed up working on a project, a digital mixer may have libraries of different EQ settings or dynamics presets. These libraries are available to use in any project, so you can store your favorite EQ for recording a bass or a drums. You can also use the presets to help you get started processing your audio.
Scenes are pictures of every parameter of your digital mixer: from the levels to the EQ to the FX settings to the bus routings; everything. Scenes let you recall a mix just as you left it, so you can get right to work on it again. Scenes allow you to bring back the exact settings you used when you recorded a vocal, so you can overdub with the same EQ and levels. Scenes let you store your current mix idea, work on another great idea you just had, then compare the two. They can also let you instantly reconfigure your mixer for live concerts if you need different levels, FX or mic settings between songs, or from one band to another. Scenes can be used whether you are mixing or tracking or overdubbing or doing live concerts.
Compatibility between systems of different manufacturers, and between earlier and later versions of the same system
Different manufactures uses different standards on plugins and session files so moving one project from one platform could make it impossible to use on another platform. Different system uses different data reduction parameters so that could also be an issue if the system doesn’t have the opportunity to use the same data reduction. With earlier and later versions of the same system, there could be some problems if the different versions have different inputs and outputs. This issues are valid between earlier an later versions also since the technology develops in high pace.
Operational procedures
Connect all instruments to the line level inputs like electric guitar, samplers, synths etc. Connect the mics to the microphones inputs. Connect the analogue outboard gear to analogue send and return and your digital outboard to digital outputs and inputs. Set up a send group for the foldback. Connect the mixer to your multitrack recorder.
Automation and recall
Digital consoles can to store and then recall all of the settings of a mix, even including the effects. This allows you to work with a project, store its settings and later come back to the mix exactly as you left it. A mixer with memory has lots of other advantages. For example, you could recall a vocal’s EQ and level for a new take or to punch in over an old version with matching audio quality. The speed of recall also allows you to try one mixing approach, store it, try a completely different idea, and then compare the two.
There are two basic types of automation: snapshot and dynamic.
Snapshots are complete pictures of your mixer: all of the effects settings, levels, etc. Some digital mixers let you customize which mixer parameters are stored in your snapshots. You place a snapshot where you need an effects patch change on your guitar part, an instant volume change for verse to chorus vocal or anywhere you want an immediate mixer adjustment the change happens automatically. Snapshots are also used for internal FX to have different effects at different times.
Dynamic automation is used for gradual changes of parameters such as a fade out. For example, you could use dynamic automation to recall every little level move you make to your vocal track. Once you have it right, you can then move on and work on the drums, knowing that the vocal track is now perfect. Some digital consoles have moving faders that help you see exactly what is happening during your mix. The memory for this automation in your digital mixer can be on board or it can be transmitted via MIDI. If it is MIDI based automation, then you would use a sequencer to play back the automation data during your mix.
Plug-ins
Digital mixer often comes with internal effects and processors. It’s also possible to upgrade these. In addition to this some mixer has the ability to use third party plug-ins, like audio metering tools, de-essers, distortion and speaker cabinets modelling, delay, filter effects, pitch correction etc.
28. What was the original purpose of timecode?
The original purpose of timecode was to use MIDI information to synchronize devices within a system. This can for example be synchronising a sequencer to a drum machines sequencer which lets the drum machines start and stop with the master sequencer.
29. Briefly describe a synchronized system, using the words 'master', 'slave' and 'chase' in your answer.
In a synchronized system one device is the master and the rest of the devices are the slaves. What ever the master does the slaves follows. The devices are connected through MIDI cables.
I fast wind modes a chase synchroniser will not read timecode so when it goes to playmode the synchroniser has to chase the master.
30. Why is the reliable synchronization of audio not possible using MTC (MIDI timecode)?
The reason why it is not reliable to synchronise audio with MTC is that it takes two frames to transmit a complete frame value and the receiver is only updated every two frames. Even if the system keeps on offset for this you can only sync to every second frame.
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.
A digital recorder generally achieves synchronous lock by adjusting its playback sample rate which means its speed and pitch ratio. It does this to precisely match the relative playback speed of the master transport.
32. From where does a digital recorder derive its clock in a synchronized system? Give two answers.
A digital recorder can sync from another device in the system or to an external separate clock which controls the whole system.
33. Briefly describe the synchronization of multiple ADAT machines (or DTRS).
Multiple ADAT machines are synchronized by connecting the sync out connector of the first ADAT to the sync in connector of the next ADAT and so on.
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?
The drum machine has to be set to external sync allowing it to sync to MIDI sync.
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?
The SPP (Song Position Pointers) includes the position from start in a song not in clocks but in MIDI beats. Before this was introduced the slaves normally started where they stopped the last time that you stopped the master. So when you for example fast forward or played again from start the slaves just started to play where they where.
36. What information do song position pointers provide?
The SPP (Song Position Pointers) includes the position from start in a song not in clocks but in MIDI beats. Before this was introduced the slaves normally started where they stopped the last time that you stopped the master. So when you for example fast forward or played again from start the slaves just started to play where they where.
37. Briefly, what is 'LTC'?
In 1967, The US Society of Motion Picture and Television Engineers introduced SMPTE ("simpty") time code. The audio sync tone version of SMPTE is called linear or longitudinal time code or LTC and is a time code recorded on to an analog or video track and encodes a modulated square wave at a bit rate of 2400 bits/second. LTC can’t be read slower than 1/10th to 1 /20th of the normal tape speed. LTC code is preferred for audio, electronic music, and mid-level video production.
38. What is 'latency'?
Latency means the delay that occurs between one event and another. In workstations latency normally means the delay between outputs and inputs of the audio hardware. Low latency is very important when using a workstation channels as foldback when overdubbing.
39. What is 'zero latency monitoring'?
Zero latency monitoring is a feature that let you monitor your input signal directly on the output. This is used when too much latency occurs when recording and over dubbing.
40. What is the drawback to zero latency monitoring?
The draw back of zero latency monitoring is that because you are monitoring the input signal directly the signal is dry and unprocessed.
41. What is DSP?
DSP (Digital Signal Processing) works by directly altering the samples of a sound file or defined region according to a program algorithm to achieve desired results. These processing functions can be performed either in real time or non-real time (offline process). The processes or effects can be: equalisation, dynamic range, delay, reverb, chorus, flanging etc.
42. What is the advantage of using a system with DSP?
The advantage to use DSP is that there is no need to convert the signal to analogue while adding effects or processing.
43. Describe the relationship between the number of plug-ins in use and the available processing power of the computer?
Plugins can be very heavy for the CPU. The more plugins used the more power from the CPU is used. There are two solutions to this:
a) Use offline process, instead of having the effects or processing working in real-time CPU power can be saved to let the DSP calculate the results in separate files. Many DAW uses offline processing or a function called Freeze.
b) Instead of have effects or processing internally in the computer add DSP card or accelerator cards. Example of this is Protools, UAD-1, Waves APA32 or APA44-M.
44. In relation to plug-ins, what is an 'instance'?
Instance in relation to plugins means the number of plugins used. For example when mixing you use the same kind of compressor on different tracks or if you use a VST instrument on different tracks.
45. What is an 'insert effect'?
Insert effect is an effect used on the insert of a channel. The whole signal will be affected. This is normally referred to processing like EQ, gates, limiters, expanders etc.
46. What is a 'bus effect'?
Bus effects are effects that different tracks can be assigned to. A part of the signal can be tapped out and blended with the dry signal. Normally you use delays and reverb as bus effect.
47. What problem might be caused by the latency of a plug-in?
problem that might be caused by the latency of a plug-in is that if the plugin is an insert effect the track will be delayed. I f it is on a send/rerun effect the wet signal will be heard delayed according to the dry signal.
48. What is an 'interleaved' sound file?
Interleaved signal is a stereo file Left and right is join in to one stereo file.
49. What is 'bouncing'?
Bouncing is when two or more tacks are mixed down to one mono or stereo tack. This was very common when the limit of tracks was an issue. Bouncing when relating to final mix means to actual expert or save the whole mix in to a mono or stereo file.
_______________________________________________________________________
SOURCES:
Huber, D.M., & Runstein, R.E. (2005). Modern Recording Techniques, 6th ed. Focal Press: Burlington
Pohlmann, Ken C. (2005). Principles of digital audio, 5th ed. New York, McGraw Hill
Rumsey, F. (2004). Desktop Audio Technology: digital audio and MIDI principles.Focal Press: Oxford
Rumsey, F. & McCormick, T. (2004). Sound and Recording: An Introduction, 4th ed. Oxford, Focal Press
Watkinson, J. (2001). The Art of Digital Audio, 3rd ed. Oxford, Focal Press
http://www.dvdaust.com/film_sound.htm
http://www.timefordvd.com/tutorial/SurroundSound.shtml
http://www.answers.com/topic/dtv
http://www.nhk.or.jp/strl/publica/bt/en/le0010-1.html
http://www.soundonsound.com/sos/aug98/articles/datacompression.html
http://www.si.umich.edu/Classes/540/Readings/Encoding%20-%20Huffman%20Coding.htm
http://www.mp3-tech.org/ |
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Mon Oct 17, 2005 3:25 am |
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rachelh
Joined: 16 Jan 2005
Posts: 35
Location: Trinidad WI |
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1. Was DAT originally intended as a professional or a domestic recording medium?
DAT or Digital Audio Tape was designed for the domestic market as a replacement for the compact cassette it was the first mass market digital audio tape recorder. The dimensions of a DAT cassette are small: 73 x 54 mm and 10.5 mm in depth, the weight of the cassette approximating to 20 grams. Professional users have an affinity to DAT due to its effectiveness as a recording medium; the most notable downside would be the size of the cassette. [1]
2. What is the sampling rate of standard DAT?
DAT supports a standard sampling rate of 44.1 or 48 kHz although some manufacturers have developed recorders that operate at a sampling rate of 96 kHz – where the tape speed is doubled and playing time is halved. [1]
3. What is the resolution of standard DAT?
DAT or Digital Audio Tape supports standard resolution linear quantization of 16-bits at a sampling rate of 44.1 or 48 kHz. [1]
4. What is 'azimuth recording'?
DAT data tracks are recorded using azimuth recording also known as guard-bandless or chevron recording. Azimuth refers to the head’s tilt in the plane parallel to the tape. The head gap should be perpendicular to the tape so that all track gaps are electrically in phase with each other. DAT uses two heads one is set at –20 degrees and the other at 20 degrees and tracks are laid down in an alternative manner with no spacing. This allows for alternate tracks, which are misaligned at 40 degrees to be bypassed in the playback and increases the recording density. [1]
5. Describe the head wheel in DAT recorder.
DAT uses two heads one is set at –20 degrees and the other at 20 degrees and tracks are laid down in an alternative manner. This allows for alternate tracks, which are misaligned at 40 degrees to be bypassed in the playback. The rotary head permits sloe linear tape speed while achieving a high recording bandwidth. [1]
6. What is SCMS?
SCMS or Serial Copy Management System 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 options: –
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. [1]
7. What is the distinguishing feature of a DAT machine capable of near-simultaneous off-tape monitoring?
Near simultaneous off tape recording can be achieved via a DAT recorder as done in a three head analogue recorder by mounting an extra pair of heads on the drum of the recorder.
8. What is the sub-code area of the DAT tape used for?
The sub-code are of the DAT tape is used for storing information which allows for data and its respective audio information to be recorded beside each other. Information pertaining to times and events are logged into the sub code area if the DAT tape.
In DAT, each recorded scan track consists of 196 blocks contained within 16 tracks. Sub-code is contained in two tracks: tracks 3 and 14; each of which is bordered with a PLL [Phase-Locked Loop] and Post amble barrier track. The sub-code data is recorded redundantly to assure accuracy and is written in 8 blocks for each sub code area. The PLL preambles, which is written in 2 blocks are contained in tracks 2 and 13 respectively and contain data, which label fields and lock the data within the fields, which in turn allows for independence in sub code writing.
Each sub code block contains a synchronisation byte, ID code [W1] byte, block address code [W2] byte, parity byte, and 32 data bytes.
The W1 byte contains PCM ID sub code data, which cannot be edited, sampling frequency, quantization level, tape speed, copy-inhibit flag, channel number as well as other parameters are included in this ID.
The W2 block identifies if the data block contains audio or data sub-code data the MSB is 0 for audio and 1 for sub code. In sub code blocks the least significant four bits contain block addresses [0 to 15] the other 3 bits contain sub code ID data.
The parity byte is used to detect errors in the ID and block address bytes, the parity byte itself is the EXCLUSIVE OR sum of the ID and block address bytes.
The remainder of the sub code block contains 256 bits, which equate to 32 data bytes of either audio or sub code data.
The majority of the DAT sub code itself contains data, which is used for program timing, and numbering - called sub code ID and is written along the scan track and in turn providing greater resistance to dropouts. The ID bits contain flags such as Start ID, Skip ID, TOCID [Table of contents ID], Data ID and Format ID information that state how many packs of data have been recorded. Sub code ID data is very complex and flexible in nature and within its 64 bit parameter contains information pertaining to program time, absolute time, running time, table of contents, catalogue number, ISRC [International Standard Recording Code], professional time code, as well as parity data.
In general, the sub code area of DAT allows for data to be recorded alongside the Audio in a DAT track, the data of which pertains to parameters of the audio itself and help to ensure error free and accurate recording and playback. [1]
9. What is 'interleaving'?
In DAT interleaving is the technique by which the system is protected from dropouts. Interleaving works by dispersing data so that if any section of data is lost, the data, which is dispersed, will contain enough information in which to reconstruct the diminished damaged signal. Interleaving itself is a method of digital data error reduction. This system works by distributing and intermingling successive bits over a wide area on the storage media. Hence, scattering potential error sources which prevents them from being consecutive on playback. if any section of data is lost, the data, which through interleaving is dispersed, will contain enough information in which to reconstruct the diminished damaged signal are employed to help minimise the consequences of errors. The minimisation of error consequences would result in the salvaging of work that has been damaged. [3][1]
10. What is the width of the tape used for 24-track DASH?
DASH or Digital Audio Stationary Head uses tape that is ½ inch in width.
11. What is the width of the tape used for 48-track DASH?
The width of tape used for 48-track DASH is ¼ inch.
12. Describe how 24-track and 48-track DASH machines are compatible.
24 and 48-track DASH machines are compatible due to the fact that if you are recording on a 24 – track DASH machine and you maximise your tracks and would like additional tracks to work with, you can simply transfer your material to a 48 track DASH machine and proceed as normal. A 48 track can be transferred to a 24 track DASH also; only difference being that only the first 24 tracks will be useable.
13. How are DASH tapes edited?
Digital Audio Stationary Head [DASH] tapes are edited by playing back the material and copying the data onto another tape and editing out the unwanted parts and splicing together the remaining parts.
14. In DASH, why does a playback head come before the record head in the tape path?
In DASH, the playback head comes before the record head in the tape path because there is an inherent delay associated with digital operations thus the playback head has to be in function to allow the record head to know how to operate and to allow for synchronous overdubs and recording.
15. Comment on the cleaning requirements of DASH How many tracks does a modular digital multitrack (MDM) have?
Unlike Analogue tape recorders DASH requires a specially trained technician to come in and clean the machine. On an average of every 6 months the head should be aligned, once again, a specially trained technician should do this. If these procedures are not adhered to, considerable damage to the system could occur amounting to thousands of dollars in damages. An MDM Modular Digital Multitrack is capable of having 8 tracks.
16. How can more tracks be obtained?
An MDM Modular Digital Multitrack is capable of having 8 tracks. More tracks can be obtained for a MDM by synchronising machines.
17. Comment on the types of usage of ADAT and DTRS machines?
ADAT [Alesis Digital Audio Tape] machines are mainly used in recording studios whilst DTRS [Digital Tape Recording System] is commonly used in postproduction for film as well as broadcasting.
18. Why is perceptual coding necessary?
Perceptual coding is based on psychoacoustic principles surround the phenomenon of masking. “Perceptual coding reduces the bit rate of a signal by implementing these psychoacoustic principles based on critical bands and the masking phenomenon. The signal's sample rate is maintained, but the word length is selectively decreased dynamically based on signal conditions. Masking is considered so that the increase in quantization noise is rendered as inaudible as possible.”[3]
“Perceptual coders analyse the frequency and amplitude content of a signal and compares it to a human auditory model. Using the model, the coder removes statistically irrelevant or redundant material. Although lossy, theoretically, the listener will not perceive the loss.
Using digital filtering, the audio is split into a number of critical bands. Each band can then be re-quantized using fewer bits. Only levels above the threshold of perception are quantized. The higher the level, the more bits that are used. Re-quantizing effects are constrained within the bands, and are more effectively masked by the band's program material.”[3]
19. Describe briefly the use of perceptual coding in the following:
Predictive or perceptive coding refers to codes that are based on the psychoacoustics of sound and hearing. This type of coding relies on the principles of masking – where our auditory perception is less sensitive to sound at one frequency whilst another frequency of close value is being heard. Hence the higher frequency masks the lower frequency. In perceptive coding, the greater the compression factor, the more accurately the human senses must be modelled. And more quantization can take place and will be masked by the greater frequency. So, previously decoded data can be used to predict current data. So, data can be transmitted with omissions and a predictive codec can be used to accurately predict the missing data by examining the previous data values and estimating what the omitted value will be, this value is then subtracted from the post omitted data value and produces a prediction residual error that is transmitted from the encoder to the decoder which in turn interprets this data and produces an output value which is used to replace the omitted data. [5]
Internet audio
An example of perceptual coding that is used for Internet audio would be MPEG Layer-3 (MP3) or MPEG-4 AAC which are based on the psychoacoustics of hearing and the perception of sound to achieve a size reduction by a factor of 10-12 with little or no perceptible loss of quality.
http://www.iis.fraunhofer.de/amm/techinf/basics.html
Film sound
An example of perceptual coding used for film sound would be the Dolby Digital AC-3, which was the first perceptual coder designed specifically to process multichannel digital audio. This system also benefits from the design of the Dolby AC-1 and AC-2 and from the development of analogue perceptual coding systems. In general the fewer the bits used to describe an audio signal, the greater the quantizing noise that can exist
http://www.headwize.com/tech/dolby2_tech.htm
http://www.roxio.com/dvd_forum/glossary.jhtml
DVD-Video
An example of a perceptual coder used for DVD Video would be MPEG 1 which encodes video in accordance with the ISO/IEC 11172 specification.
http://www.roxio.com/dvd_forum/glossary.jhtml
Digital television
An example of a perceptual coder used in digital television is the MPEG 2 perceptual coder.
This coder is backwards compatible with MPEG 1 and video is encoded in accordance with the ISO/IEC 13818 specification.
http://www.roxio.com/dvd_forum/glossary.jhtml
Personal stereo
An example of perceptual coding that is used for personal stereos is the MiniDisc.
Through a combination of various techniques including psychoacoustics, subband coding and transform coding, ATRAC [Adaptive Transform Acoustic Coding] succeeds in coding digital audio with virtually no perceptual degradation in sound quality... “ Listening tests indicate that the difference between ATRAC sound and the original source is not perceptually annoying nor does it reduce the sound quality. Furthermore, the system is sufficiently compact to be installed in portable consumer products. Using ATRAC, the MiniDisc provides a practical solution for portable digital audio.”
http://www.minidisc.org/aes_atrac.html
20. What can be done, other than perceptual coding, to reduce bitrate?
Whereas perceptual coding operates mainly on data irrelevancy in the signal, Huffman coding or Entropy coding uses the probability of occurrence to code messages. For example when data is analysed, samples that contain information least likely to occur is coded with longer codewords whilst samples that occur most often are assigned shorter codewords. Huffman coding is lossless due to the fact that information is not lost and the process itself is completely reversible. In general, Huffman coding is noiseless and uses statistical techniques to represent a message with the shortest possible code length. [1]
21. What is masking?
Masking refers to the phenomenon by which soft signals are ‘covered up’ due to the presence of loud signals, which are occurring at the same time. The greatest masking occurs when the frequency of the sound and the frequency of the masking noise are close to each other. Masking can be also be caused by harmonics of the masking tone. Equalisation might be required to make the instruments sound different enough to overcome any masking effects. [4]
“Research shows that masking occurs with tones inside of frequency bands; a given tone will mask another tone within that band, but will not affect tones outside of that band; these are known as critical bands. The bandwidth of these bands increases as frequency increases, but can be approximated to be about 1/3 octave for frequencies between 300-20,000 Hz. The bands are not fixed, but are continuously variable and any audible tone will create a band centred on it. The masking tone raises the threshold of perceived hearing around that tone. Sound beneath that threshold is masked; however, sound outside of the tone's critical band will not be affected.”[3]
22. How do perceptual coding systems handle signals that are probably going to be masked by other audio?
“Perceptual coders analyse the frequency and amplitude content of a signal and compares it to a human auditory model. Using the model, the coder removes statistically irrelevant or redundant material. Although lossy, theoretically, the listener will not perceive the loss.
Using digital filtering, the audio is split into a number of critical bands. Each band can then be re-quantized using fewer bits. Only levels above the threshold of perception are quantized. The higher the level, the more bits that are used. Re-quantizing effects are constrained within the bands, and are more effectively masked by the band's program material.”[3]
23. What is Huffman coding?
Huffman coding or Entropy coding uses the probability of occurrence to code messages. For example when data is analysed, samples that contain information least likely to occur is coded with longer codewords whilst samples that occur most often are assigned shorter codewords. Huffman coding is lossless due to the fact that information is not lost and the process itself is completely reversible. In general, Huffman coding is noiseless and uses statistical techniques to represent a message with the shortest possible code length. [1]
24. What is the typical bitrate for an MP3 file intended for Internet distribution?
The typical bitrate for an mp3 file intended for Internet distribution is 128 bps [bits per second]
25. What is the bitrate of Dolby AC3 as used in film sound?
The bitrate of the Dolby Digital Plus Enhanced AC3 as used in film sound is 640 kbps. [1]
26. What is metadata?
Metadata is a relatively new digital practice that is supposed to combat the over reliance of compression which is necessary to allow television and radio audio so that they will have widespread reception. It allows the user to define the sonic composition of the signal received.
Essence refers to content such as audio, video, still pictures, graphics as well as text whilst metadata refers to content such as edit lists or other related data, which describes the data. Metadata can hold parameters such as sampling frequency, down mixing, and channels all of which describe how the essence should be decoded. Metadata can be also used to search for essence and contain intellectual property information such as copyright and ownership, which in turrn is, needed to access the essence. Metadata can also play an essential role in storing data that provides insight into how certain elements should be assembled – also known as ‘composition’ as well provides information for synchronisation. [1]
27. Describe and explain the following disk editing systems concepts and their operation:
• Comparison with analogue and digital tape editing
Digital consoles have the same features as their analogue counterparts only that these devices use large-scale integrated circuits and central processors to convert, process, route and interface to external audio and computer related devices to convert and store audio in the digital domain. Although digital systems interface with the signal path in ways that differ from analogue consoles, the digital signal path is conceptually the same as the analogue signal path. Console features can be controlled by physically placing a readout display at each central point on the input strip at locations that are similar to the way it would be done on an analogue console. In contrast to analogue consoles, digital input strips can gain access to the central control panel if channel select is pressed on the desired channel [1]
• Features, functions and parameters of disk editing systems
1. Digital consoles have the same features as their analogue counterparts only that these devices use large-scale integrated circuits and central processors to convert, process, route and interface to external audio and computer related devices to convert and store audio in the digital domain. Although digital systems interface with the signal path in ways that differ from analogue consoles, the digital signal path is conceptually the same as the analogue signal path.
2. Digital consoles have centralised control panels that is used to control and vary the particular channel setting parameters such as auxiliary send, dynamics, EQ or track assignment settings which contrasts to the way these settings are changed as stated in the analogue console features above. This panel can be multipurpose in its operation allowing itself to be reconfigured via the use of software [soft] buttons, pan and level controls.
3. Console features can be controlled by physically placing a readout display at each central point on the input strip at locations that are similar to the way it would be done on an analogue console
4. In contrast to analogue consoles, digital input strips can gain access to the central control panel if channel select is pressed on the desired channel
[1]
• Compatibility between systems of different manufacturers, and between earlier and later versions of the same system
When one tries to interface digital signal processing equipment and digital storage media with analogue mixing circuitry negates some of the advantage of digital as a recording medium. The combination of higher fidelity standards, greater versatility and easier operation and enhanced interfacing has lead designers to the digital mixing console as a more cost effective solution. In the dynamics section of a mixer, signals can be processed without the use of outboard devices; compressors, limiters, expanders and noise gates are located here. At the insert point of a digital mixer a direct send/ return or insert access point can be sued to send the line level audio signal out to an external processing device or recorder. One vital point to note is that plugging in a dynamics, EQ or effects processor will only affect the signal that is passing through the selected I/O channel. The output of a console also has an insert point making it easy for this signal to be processed further.
[2][1]
• Operational procedures
The recording console is the centre of all studio operations it is the central entity to which all signals pass through and can be routed to other devices, the console itself may have control over outboard devices it also has control over automation as well as tape machines.
Operation wise an audio signal is inserted into a channel input, then, it can be altered using effects devices, sent to various monitors or effects devices -altered sonically, outputted and finally mixed down to its final form.
• Automation and recall
Automation can be described as the storage of dynamic fader positions against time. Mixing desk automation is designed to help the engineer simplify automation when the number of simultaneously used faders is too many to handle by just one person. This is done by creating sub-areas of the mix at each pass; this gradually builds up the finished product [under computer control] making it easy for the engineer to edit. In practice, an automated system can range from able to only sense only the position of a volume fader and level-related switching functions [this is often called snapshot automation due to the representation of the mixer or console setting at only one point in time] to being a fully automated system that can store and recall all the dynamic functions of a production console. Console automation allows for faders to be easily arranged into one or more groups. Fader grouping is done via the use of a single control voltage or digital value, which is used to control the relative balance of several, grouped channels and/ or tracks. These sub areas are then edited during the mix down process and executed under computer control during a final mix pass. Since digital consoles already speak the language for automation that is, digital, it is much easier and cost effective to code and decode gain, effects, routing and other automation functions.
Write/ update/ read refers to the modes of automation. During the write mode, the mix moves are recorded into memory in real-time. Update mode allows for the already stored data can be updated and modified with only changes being recorded [update/offset mode]. Read mode refers to playback of the mixed information directly from the system’s automation computer.
Recall in automation describes the process by which the operator/ engineer is told the correct placement of controls, leaving it up to the user to reset the controls if desired. This term namely Total Recall was coined by SSL for its system, recall helps in saving time as the user is saved from writing down the positions of every knob and button allowing for the console to be easily reset during sessions. It is to be observed that Total Recall differs from True Total Reset as the latter allows for an interface to be positioned between the automation system and every control located on the console.
A section of automated data may be re-recorded via the use of the scans, which observe the various controls of a mixing surface during the course of a mix. In Write mode, the automation system will detect the associated changes and convert them in a series of corresponding digital words that can be stored directly into the computers automation memory. After the tracks are written, update mode can be used to go back at a later time to alter mix settings originally written to memory, once a control is altered or if automation is re-recorded or reversed in the input strips the updated settings can be changed by adding or subtracting from the track data rather than completely re-writing it. In this way complex moves can be obtained and remain intact while other changes are implemented.
The update mode allows automated punch ins to be blended smoothly with existing data and can be used to go back at a later time to alter mix settings originally written to memory, once a control is altered or if automation is re-recorded or reversed in the input strips the updated settings can be changed by adding or subtracting from the track data rather than completely re-writing it. In this way complex moves can be obtained and remain intact while other changes are implemented. The data is blended smoothly due to the process of level matching which ensures that the punched in data levels correspond to the pre-existing ones, allowing for seamless transition. When using a moving fader this is automatic, for VCA this is done manually by observing the nulling indicator.
The concept of total automation resounds around the principles of an assignable mixing console. This assignable mixing console allows for ease in the implementation of storing switch closures and settings into memory [via the use of a microprocessor] therefore allowing for reiteration to be quickly done. Because a digital console inherently speaks the ‘digital language’ total recall automation is more effective than using an analogue console. With digital consoles it is much easier and cost effective to code and decode gain, effects, routing and other automation functions thus, making assigning a digital console for total automation an easy task. Analogue consoles call for converting the positions and moves of the various dynamic controls on a mixing surface [such as volume, Equalisation and Sends] into a DC voltage that can then be converted into digital data which is a longer process than that employed by a digital console [1][6][7]
• Plug-ins
Plug-ins, recording equipment and software in the digital realm emulate their analogue counterparts because it is a sound that users are accustomed to. External plug-ins can be inserted at the Insert Point of the console where a direct send/ return or insert access point can be used to send the line level audio signal out to an external processing device [such as noise gates, filters, meters, de–essers etc] or recorder. One vital point to note is that plugging in a dynamics, EQ or effects processor will only affect the signal that is passing through the selected I/O channel
[3] [2]
28. What was the original purpose of timecode?
Timecode is the standard method of interlocking audio; video and film transports that makes use of a code developed by the Society of Motion Picture and television Engineers [STMPE]. The use of this STMPE time code allows for the identification of an exact position on a tape or within a media program by assigning a digital address to each specified length. This address code cannot slip, and always retains its accuracy between 1/24th and 1/30th of a second [dependant on the media type and the standards being used]. The specified tape segments are called frames, a term taken from film production. And the time code address is the tag given to every audio and video frame and is represented by an 8 digit code highlighting hours:minutes:seconds:frames i.e. 00:00:00:00 [4]
29. Briefly describe a synchronized system, using the words 'master', 'slave' and 'chase' in your answer.
When two devices are synchronized to one another it is necessary to have one be the master and the other the slave. The slave unit responds to commands or information from the master and is thus controlled by it. This is the basic principle behind all synchronization in audio and video. For example, if a computer system is following an analogue tape machine (or video deck) it can be said to be "slaved" to it hence the slave ‘chases’ its master.
30. Why is the reliable synchronization of audio not possible using MTC (MIDI timecode)?
MTC or Musical Instrument Digital Interface Time Code was developed for electronic music studios, project studios and all other production environments to have a cost effective and easily implemented way to translate time code into time-stamped MIDI messages and back. It allows for time code to be distributed throughout the MIDI chain to devices that are able to understand and execute MTC commands. The reliable synchronisation audio is not possible using MTC [MIDI time code] because MTC is meant or was devised as a timing agent and is not able to hold the synchronisation between devices, as it does not contain a clock. [4]
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.
If the speed of a digital recorder were varied due to being controlled by a synchroniser, the sampling rate of the output would adjust accordingly- a standard such as drop code, non-drop code or EBU time code would be applied.
32. From where does a digital recorder derive its clock in a synchronized system? Give two answers.
A digital recorder derives its clock in a synchronised system from either:
1. SMPTE abbreviates for the Society of Motion Picture and Television Engineers. SMPTE, much like the Audio Engineering Society and other organizations, provide a coherent place for keeping professionals (in this case television and film audio engineers) up to date with current information as well as formalizing and documenting necessary standards from time to time. It was SMPTE who devised the classical method of measuring intermodulation distortion, but one of their most noteworthy achievements is the formalization and standardization of SMPTE Time Code, which is the standard method of interlocking audio, video and film transports that makes use of a code developed by the Society of Motion Picture and television Engineers [STMPE]. The use of this STMPE time code allows for the identification of an exact position on a tape or within a media program by assigning a digital address to each specified length. This address code cannot slip, and always retains its accuracy between 1/24th and 1/30th of a second [dependant on the media type and the standards being used]. The specified tape segments are called frames, a term taken from film production. And the time code address is the tag given to every audio and video frame and is represented by an 8 digit code highlighting hours:minutes:seconds:frames i.e. 00:00:00:00 [4]
2. The Master clock - 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 system 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. [4]
33. Briefly describe the synchronization of multiple ADAT machines (or DTRS).
ADAT- Alesis Digital Audio Tape has a recording time of 60 minutes, is used in budget recording studios, is capable of 8 tracks, multiple machines can be synchronised to produce more tracks, S-VHS tapes are used for recording, the tape needs to be formatted before use, high resolution versions are available. DTRS- Digital Tape Recording System has a recording time of 108 minutes, is used namely for film post-production and broadcast post production, is capable of 8 tracks, multiple machines can be synchronised to produce more tracks, Hi-8 tapes are used for recording, the tape needs to be formatted before use, high resolution versions are available.
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?
An audio sequencer is a digital device that is used to record, edit and output MIDI messages in a sequential fashion. These sequential messages are generally arranged in track based format that follows the modern production concept of locating separate instruments and/ or instrument voices onto separate tracks. In a synchronised system consisting of a sequencer a drum machine where the drum machine is the slave, the setting that needs to be altered on the drum machine to allow it to synchronise to the sequencer is that the external sync setting has to be selected, this setting allows for the drum machine to be controlled by an external device. [4]
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?
Song position pointers (SPPs) allow a sequencer or drum machine to be synchronised to an external source such as a tape machine from any measure position within a song. Although SSPs are used less often than MIDI Time Code [MTC], it can synchronise a location in a MIDI sequence in measure to a matching position point on an external device such as a drum machine or a tape recorder, by providing timing reference that increments once for every six MIDI clock messages with respect to the beginning of a song. Before the introduction of SPP’s if the master was started at some point other than at the start of the sequence the slave devices would continue playing from the last point that it was stopped and would not correlate or re calibrate to the start point of the master. [4]
FSK of Frequency shift Keying is an audio tone (frequency) modulated by a square wave, which is used both for data transfer and also for sequencer and drum machine synchronization. FSK is the sound that you hear when your fax or modem making as it establishes communication. In the early days of electronic music, before MIDI, drum machines or sequencers were synchronized to each other or to a tape machine via this method. Back then the only information transmitted was a rate which was interpreted as tempo by the machines. There was no location information included so the song always had to be started from the very beginning in order to achieve proper sync. If there was any drop out or glitch along the way one had to go back and start at the very beginning of the song to re-establish sync. It was cumbersome and unreliable to say the least, and that is why formats such as SFSK, DTL, SMPTE, and MTC were later adopted. [2]
36. What information do song position pointers provide?
Song position pointers (SPPs) allow a sequencer or drum machine to be synchronised to an external source such as a tape machine from any measure position within a song. Although SSPs are used less often than MIDI Time Code [MTC], it can synchronise a location in a MIDI sequence in measure to a matching position point on an external device such as a drum machine or a tape recorder, by providing timing reference that increments once for every six MIDI clock messages with respect to the beginning of a song. [4]
37. Briefly, what is 'LTC'?
LTC or longitudinal Time Code is a major system that exists for encoding time code onto magnetic tape. LTC encodes a biphase time code signal onto the analogue audio or cue track in the form of a modulated square wave at a bit rate of 2400 bits/second. In most situations LTC code is preferred for audio. Electronic music and midlevel video production and it’s a more accessible and cost effective protocol. VITC is used by major video production houses, it makes use of the same SMPTE and user code structure as LTC but is encoded onto videotape in an entirely different manner that is, it actually encodes the time code information into the video signal itself. [4]
38. What is 'latency'?
Latency is also known as delay. It occurs in the exchange of packets of data from one place to another. In audio, latency is the time that the audio is retrieved from the hard drive then heard or the time that audio is processed then heard with real time effects with respect to hardware, latency is the time taken for a signal to pass through to an electronic device [the time taken for a device to respond to the request / command at hand] or the time taken for a request to be processed and applied.
There is no latency associated with a hardware sampler as the samples are stored in RAM [Random Access Memory] and sometimes the cache section of the memory which has a faster access time than normal RAM. Propagation delay, describes the initial delay that will occur that is, the time it will take the signal to pass through a signal processing box, thus propagation delay contributes to latency, in general, the higher the latency of the device, the longer it would take for processing. A device with a high latency takes a longer time to execute a task than a low latency would take.
39. What is 'zero latency monitoring'?
Latency is also known as delay. It occurs in the exchange of packets of data from one place to another. In audio, latency is the time that the audio is retrieved from the hard drive then heard or the time that audio is processed then heard with real time effects – it is the time taken for an electronic device to execute the task requested. A device with a high latency takes a longer time to execute a task than a low latency would take.
The term ‘Zero latency monitoring’ was introduced in 1998 by RME with the DIGI96 series of audio interfaces and refers to the technique of routing the input signal directly to the output on the audio card and has become one of the most important features of modern, host based hard disk recording. [2]
40. What is the drawback to zero latency monitoring?
The drawback to zero latency monitoring would be that workflow has to be compromised to achieve the effect. The only easy answer to this is to go with costly solutions until processing speeds allow the power and flexibility of dedicated systems to be truly replicated with host based systems. [2]
41. What is DSP?
DSP [Direct Signal Processing] refers to digital signal processing, DSP processing is much faster than host based or computer based processing, as DSP is synonymous with lower incidences of latency. Because DSP based systems are designed around music production [incorporating specially designed hardware chips and dedicated processors], they are designed to cope with audio signals as opposed to normal data that is written and stored onto a computer. Thus making DSP a more compatible medium when recording and playing back music.
42. What is the advantage of using a system with DSP?
The advantage of using a system with DSP is that the system would run faster than a host based or computer based processing, as DSP is synonymous with lower incidences of latency. Because DSP based systems are designed around music production [incorporating specially designed hardware chips and dedicated processors], they are designed to cope with audio signals as opposed to normal data that is written and stored onto a computer. Thus making DSP a more compatible medium when recording and playing back music.
43. Describe the relationship between the number of plug-ins in use and the available processing power of the computer?
Plug -ins are additional software that are needed to make programs work with the OS, it is integrated/ installed so that the program can be used alongside with the OS. Keeping in mind that the processing power of a computer is utilised for every command, and every process that takes place, it can be seen that plug-ins especially those that utilise vast amounts of a systems resources can eventually slow the system down – during the time it takes to process the command. [2]
44. In relation to plug-ins, what is an 'instance'?
In relation to plug-ins an instance refers to the number of times in which a plug in is utilised
45. What is an 'insert effect'?
An insert effect is one that is routed through the insert point of a console.
Insert Point – at this point a direct send/ return or insert access point can be used to send the line level audio signal out to an external processing device or recorder. One vital point to note is that plugging in a dynamics, EQ or effects processor will only affect the signal that is passing through the selected I/O channel [1]
46. What is a 'bus effect'?
A bus effect is one, which originates from the output bus.
Output Bus – a bus is a single electrical conduit that runs the horizontal length of a console or mixer, signals can be injected into the bus line and be routed off the bus to one or more output destinations [1]
47. What problem might be caused by the latency of a plug-in?
The problem that may be caused by the latency of a plug-in is dropouts, which would indicate that the processor has reached its critical limit with respect to the tasks at hand. Dropouts may occur due to the process being completed by the plug-in or by background processes that are being run by the OS or another program occurring in conjunction with the ‘plug-in process’
48. What is an 'interleaved' sound file?
An interleaved sound file is one that has been combined in terms of panning hence, the left and right channels are mixed down into one single waveform or data block in the digital realm
49. What is 'bouncing'?
Bouncing refers to the process by which similar tracks are combined onto one to free up other tracks and make them available for additional recording. In bouncing, the inferiority of the sync signal is notable because it cannot be ‘bounced’ as it is inferior in sound quality.
Bouncing in the context of mixing refers to the process by which similar tracks are combined onto one to free up other tracks and make them available for additional recording.
Reference:
1. Principles of Digital Audio 5th Edition – Ken C. Pohlman
2. www.sweetwater.com
3. www.mtsu.edu
4. Modern Recording Techniques – DM Huber, R Runstein
5. The Art Of Digital Audio – John Watkinson
6. Sound and Recording an Introduction 4th edition – Francis Rumsey, Tim McCormick
7. The Art Of Digital Audio 3rd edition– John Watkinson |
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Tue Oct 18, 2005 5:15 am |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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That is excellent work both of you, really is superb detail you are both writing, this will pay seriously good odds for the examination.
So impressed!
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Sun Oct 30, 2005 2:31 pm |
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