AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 3 |
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OK.
Firstly this week do please check over week2 again, there are unanswered concepts I would like some closure on. I will keep that thread open, we really must get those issues done to progress.
It doesn't matter if one person tackles all the questions at once, you may both add something by doing so. So please do not feel the work should be split in half, I want you BOTH to tackle ALL the concepts each week.
Ok here is week 3, I'm throwing theory in here which you will have to get very busy at un-packing. These are not easy.
1. Explain the function of the of the following devices
(a) Anti alias Filter
(b) Sample & Hold
(c) Dither Generator
2. Explain the operation of copying digitally when the devices are equipped with SCMS
3. Describe the functions of the contents of a sub frame of MADI
4. Describe the Master Clock system of synchronisation of a digital signal chain
5. With regard to a CD-Recordable system explain the following operations
(a) Single Session
(b) Track at Once
(c) Multi Session
6. What is over-sampling?
7. With the aid of diagrams explain time compression. List two applications of time compression.
8. Explain the principles of error concealment.
9. Describe in simple terms the main advantage of an over sampled D-A
10. Compare the results of too high recording levels in a analogue and digital tape systems.
_________________
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Sun Mar 18, 2007 7:19 pm |
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resol69
Joined: 31 Dec 2002
Posts: 69
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| from nancy |
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1.Explain the function of the of the following devices
(a) Anti Alias Filter
Let’s look at aliasing first:
According to the Nyquist sampling theorem the sampling rate should be at least twice the maximum frequency component of the signal. In other words, the maximum frequency of the input signal should be less than or equal to half of the sampling rate. Even if you are sure that the signal being measured has an upper limit on its frequency, pickup from stray signals (such as the powerline frequency or from local radio stations) could contain frequencies higher than the Nyquist frequency. These frequencies may then alias into the appropriate frequency range and give you erroneous results when sampling.
How to reduce aliasing:
To reduce aliasing, an anti-aliasing filter is needed whenever analog signals are sampled, or when a digital signal is sample rate converted from a high sample rate to a lower sampling rate. An anti-aliasing filter is a low-pass/high-cut filter that is set to cut of frequencies above the Nyquist Frequency.
How an anti aliasing filter works:
To be sure that the frequency content of the input signal is limited, a low pass filter (a filter that passes low frequencies but attenuates the high frequencies) is added before the sampler and the analog-to-digital converter. This filter is an anti-alias filter because by attenuating the higher frequencies (greater than the Nyquist frequency), it prevents the aliasing components from being sampled. Because at this stage (before the sampler and the analog-to-digital converter) you are still in the analog world, the anti-aliasing filter is an analog filter.
The major issue
with an anti-aliasing filter is that while its job is to cut of frequencies above the Nyquist frequency it is nearly impossible to create a usable Brickwall Filter. Since the anti-alias filter is set close to the audible frequency range of human hearing the filter can begin to filter out audible frequencies. In practice this is only a problem if you are recording at a 44.1 kHz Sampling Frequency because the Nyquist Frequency for that sampling rate is 22.05 kHz which is close to the human hearing threshold of roughly 20 kHz. This is why many people record and/or mix at 48 kHz sampling frequency or higher.
(b) Sample & Hold
What is a sample and hold circuit and what does it do?
A sample and hold circuit is used to interface changing analogue signals to an analog-to-digital converter. The purpose of this circuit is to hold the analogue value steady for a short time while the converter performs some operation that takes a little time. In most circuits, a capacitor is used to store the analogue voltage, and an electronic switch or gate is used to alternately connect and disconnect the capacitor from the analogue input. The rate at which this switch is operated is the sampling rate of the system.
Sample and hold circuits are often used when multiple samples need to be measured at the same time. Each value is sampled and held, using a common sample clock. In order that the input voltage is held constant for all practical purposes, it is essential that the capacitor has very low leakage (gradual energy loss), and that it is not loaded to any significant degree which calls for a very high input impedance (the impedance actually experienced by a signal which is connected to its input).
Why do you need a sample and hold circuit?
In some kinds of analog-to-digital converters for example, the circuit tries a series of values, and stops converting once the voltages are "the same" within some defined error margin. If the input value changed during this comparison process, the resulting conversion would be inaccurate. Analog-to-digital converters will often incorporate internal sample and hold circuitry to ensure more accurate conversion of the analog signal.
(c) Dither Generator
First, what is dithering?
Dithering is actually adding noise to an audio signal, on purpose!
Dither is a form of noise, or 'erroneous' signal or data which is deliberately added to sample data for the purpose of minimizing quantization error. So, a dither generator creates noise to add to an audio signal.
Why dithering is needed.
When converting from an analog signal to a digital signal, there is always some degree of error. Low level signals are difficult for digital gear to record; the sampling machine simply has difficulty deciding whether the necessary bits should be turned on or off, creating "quantization noise." By adding a small amount of very controlled noise to the original signal, the bits can be made to positively switch on or off, improving low level sound resolution
An analog signal is continuous, with ideally infinite accuracy, while the digital signal's accuracy is dependent on the quantization resolution, or number of bits of the analog to digital converter. The difference between the actual analog value and approximated digital value due to the "rounding" that occurs while converting is called quantization error. We use a fixed number of bits (16, for example) to accurately represent our sample points, but we can’t really use 16 bits all the time, so we get distortion. If you adjust the recording to allow for peaks that hit the full sixteen bits that means much of the music is recorded at a much lower volume—using fewer bits. Plus components that are smaller than the level of one bit won't be recorded at all. Dithering, or adding noise to the signal, fixes this problem.
How to dither
Dithering is done by adding noise of a level less than the least-significant bit before rounding to 16 bits. The added noise has the effect of spreading the many short-term errors across the audio spectrum as broadband noise. Dither is added using a dither generator.
What does dithering actually do?
Besides reducing the distortion of the low-level components, dither let's us hear components below the level of our least-significant bit. By changing a signal that's not large enough to cause a bit transition on its own, the added noise pushes it over the transition point for an amount statistically proportional to its actual amplitude level, allowing us to pull the weak signal out of the noise.
To dither, or not to dither (not to be confused with Jitter)
Dither must be added before any quantization or re-quantization process, to prevent distortion. The lesser the bit depth, the greater the dither must be. The results of the process still yield distortion, but the distortion is of a random nature so its result is effectively noise.
Digital audio editing software should dither automatically when appropriate. Dithering does require some computational power itself, so the software is more likely to take shortcuts when doing "real-time" processing as compared to processing a file in a non-real-time manner. So, an application shows a live on-screen mixer with live effects for real-time control of digital track mixdown is likely to skimp in this area, whereas an application that must complete its process before you can hear the result doesn't need to.
If we use high enough resolution, dither becomes unnecessary. For audio, this means 24 bits. Audio digital signal processors usually work at this resolution, so they can do their intermediate calculations without fear of significant errors, and dither only when its time to deliver the result as 16-bit values.
2. Explain the operation of copying digitally when the devices are equipped with SCMS
What is SCMS
SCMS, or Serial Copy Management System was an early form of digital rights management, or copy protection. SCMS was created as a compromise between electronics manufacturers, which wanted to make DAT machines available in the United States, and the RIAA, which previously hampered the availability of DAT machines in the US via lawsuit threats. The RIAA did not want low-cost digital recorders readily available, since it felt that such technology would result in widespread piracy. SCMS was created to prevent DAT recorders from making second-generation or serial copies.
How it works
SCMS sets a "copy" bit in all copies, which prevents anyone from making further copies of those first copies. Any number of copies can be made from the master. This helped protect copyrighted material from being duplicated to some extent.
The copy protection looks for some bits written on the subcode data. There are three options:
Copy allowed (00)
Copy once (11)
Copy prohibited (10)
If the source has the copy bits 00, and you make a copy of this, the copy will have the bit set as 00 too, allowing copies of the copies. If the source has the copy bits set as 11, every copy of this material will have the bits set to 10 and the copy from the copy would be prohibited.
3. Describe the functions of the contents of a sub frame of MADI
First, what is MADI?
MADI (Multi-channel Audio Digital Interface) is an interface that's usually found in higher-end equipment for transporting multiple streams of digital audio in a single direction between two devices, such as a mixing console and multitrack recorder.
MADI is made up 28 AES/EBU (2-channel) signals in serial, i. e. after one another. MADI allows two devices to exchange up to 56 channels of 24-bit digital audio using a single 75 Ohm BNC coaxial cable, or FDDI optical link, along with a separate synchronization signal. MADI adopts the standard AES/EBU interface and uses Time Division Multiplexing to squeeze all the channels into one normal frame.
About Frames and Subframes
The MADI transmission is broken down into a series of frames, where the sample data for all channels in one single sample period is stored in one frame. MADI frame is broken down into subframes, where each subframe stores a single 32-bit sample for a single audio channel. Of the 32 bits available to each subframe, 24 bits are available to carry audio data, the next four bits carry validity, user, status and parity (defined below), while the last four bits carry mode information for frame synchronisation and channel active/inactive status. When less than 21-bit audio is used, the auxiliary sample bits can be used for other applications, such as carrying talkback or cueing audio.
validity bit — defines the payload as audio
user data bit — any private data
channel status bit — used to convey channel status information (metadata)
parity bit — used for error checking
4. Describe the Master Clock system of synchronization of a digital signal chain
Using a master clock system, one device is the master timekeeper, and other devices (slaves) respond to the master’s timing data rather than follow their own internal timing.
Digital audio signals need to be synchronized to a master clock to prevent certain types of "jitter" that can degrade sound quality.
5. With regard to a CD-Recordable system explain the following operations:
(a) Single Session: The number of sessions a compact disk has refers to the number of different continuously-written chunks of data that are placed on the disk. Traditional CD formats such as standard CD audio and CD data, are said to be single session; everything that is ever going to be on the disk is placed there at once when the disk is manufactured.
(b) Track at Once: Each time a track is finished, the recording laser is stopped, and two run-out blocks are written. When the laser is started again to write another track, one link block and four run-in blocks are written.
(c) Multi Session: Newer CD formats however use more than one session, and are called multi-session drives. Using multiple sessions means that information can be written to the first part of the disk, and then later more information can be appended to it in the unused space left after the first session. Each session has its own lead in, program area, and lead out. This takes up about 20 megabytes of space, sot it is less efficient than recording data all at once.
6. What is over-sampling?
Oversampling involves using a sample frequency that is much higher than the Nyquist sample rate. This reduces the work the anti aliasing filter needs to do. This means that the analogue filter in the system can be gentler, giving a better response from less expensive components. Oversampling can be used in the analogue to digital conversion process (easing the design of the anti-aliasing filter) and in the digital to analogue conversion process (easing the design of the reconstruction filter). The use of oversampling in one conversion process is completely independent of the other conversion process. In other words, oversampling can be used at the digital to analogue converter, regardless of whether oversampling has been used in the analogue to digital conversion process.
7. With the aid of diagrams explain time compression. List two applications of time compression.
Time compression speeds up the playback of audio-video content without causing the pitch to change. This includes removing pauses and overlapping the audio.
Applications of time compression:
Skim and browse files quickly
Make audio files smaller (to take up less storage space and transmission time as in for use on the Internet or Mp3 files)
8. Explain the principles of error concealment.
A digital to analog decoder incorporates three levels of protection against damaged data:
Error correction;
an error is detected in the data and completely corrected by using the additional error-correction data specifically put there for the purpose.
Error concealment;
an error is detected but it is too severe to be corrected. Missing data becomes a “guess” based on the surrounding data and the result hopefully will be inaudible. However, if you ever get chance to see a CD player that has correction and concealment indicator lights, you will notice that an awful lot of concealment goes on just to play an average disc. How well concealment is done is one of the factors that make different digital systems sound different.
Muting;
in this case the error is so bad that the system cant’ even make a guess at it and shuts down momentarily rather than output what could be an exceedingly loud glitch.
9. Describe in simple terms the main advantage of an over sampled D-A
Analogue anti-aliasing filters are expensive due to the high quality analogue components needed. In an oversampling system, the analogue anti-aliasing filter can have a very slow roll off and be very cheap.
10. Compare the results of too high recording levels in analogue and digital tape systems.
Audio that exceeds 0 dB is called 'overdriven' audio. It's at 0 dB that analog audio starts to distort. The higher the dB reading the more distortion. But analog audio is very forgiving. Going +2 or +3 dBs over 0 results in very minimal distortion, and can even be warm or fuzzy. Digital audio is totally unforgiving in this respect. Once your volume level hits 0 you experience is called 'digital clipping' (square wave clipping). You must keep the very loudest part of your audio (spikes) below 0 dBs. |
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Tue Mar 20, 2007 2:09 pm |
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