Digital audio techniques

Task 1 The conversion of an analogue wave to a digital wave is a far from simple process. Here is a basic diagram to explain this concept.

As you can see in this diagram when I record a sound in to logic it goes through a big conversion process, I will now go into more depth about what these single processes are. The first process of this is dependent on how your recording you digital audio, if your like me and your were using a microphone than the first process would be this. There are many different types of mic the process I will explain is the one of a basic dynamic mic. A dynamic microphone takes advantage of electromagnetic effects. When a magnet moves past a wire (or coil of wire), the magnet induces current to flow in the wire. In a dynamic microphone, the diaphragm moves either a magnet or a coil when sound waves hit the diaphragm, and the movement creates a small current. This is then sent down the wire to whatever is connected to. This signal is normally small and is needed to be amplified. This is where our second process comes in, the use of a line amplifier, this is were the small current that is generated by the coil in the microphone in amplified to a bigger level, as I said before the current that is produced by a microphone is normally round about a mill-volt level (a thousandth of a volt) this is the amplified about a 1000 times to match the level of the preamps, when the current is amplified about 1000 times it goes to about 60db Before the anti-aliasing filter there is a dither generator, this is employed to add small amounts of white noise to reduce signal-to-error and distortion. thus making it possible to encode signals that are less than the least significant bit. This is smaller than a quantisation amount. The next part of the process is the anti aliasing filter. This is the Nyquist theory takes place, although it made be used later as well. All the aliasing filter does is get rid of unwanted frequencies, above the Nyquist frequencies. And let frequencies below this level through. I will go into the process in more depth later.

The process flowing this is the sample and hold process. The sample and hold circuit is used to change analogue signals to a subsequent system such as an AD converter (analogue to digital converter). The purpose of this circuit is to hold the analogue value steady for a short time while the converter or other following system performs some operation that takes a little time. Timing information, which makes up the sample rate, is generated by a crystal reference clock. Any variation in this timing can cause jitter. jitter adds noise and distortion. Jitter is worst for high amplitude high frequency signals. Here is a smaller diagram to show what the sample and hold circuit does.-

As you can see this is where the sample is converted from digital to analogue, you can see here that the sound wave is Turing from an analogue wave (curved) to a digital wave (squared). During the sample (and hold) process the analogue-to-digital conversion process begins. This is a process which is really important to component of your sound wave sound once converted. In this there are some important changes which I will explain. The sample rate is how well the digital converter reads and understands the sound waves.

A lower sample rate will have not read the sound wave as accurately as the higher one. This is because it will automatically square off parts of the wave. I higher sample rate will read the sound wave and square off smaller parts off it, giving a better conversion To a digital wave. Bit depth Bit depth is the analyse of the difference of dynamic range between one small sound wave and a large sound wave.

This is not to be confused with volume it is just how the well the conversion reads between a small sound a and a large sound. Bit rate. Bit rate is a calculation between the sample rate and bit depth. The calculation is X x Y x 2 =B X=Bit depth Y=Sample rate B=bit rate Note: bit rate is a term for finding the overall quality of the audio sample once converted to a digital wave. Multiplexing is a process where multiple analog signals or digital data streams are combined into one signal. I combing multi signal into one. Error correction is were bits of data have been damaged or lost, the main way that this is fixed in a digital environment is that they are Parity checked, this is were bits of code that are corrupted or missing are searched, taken out and replaced of squared off. This is normally done by a algebraic calculation.

In coding theory, a parity-check matrix of a linear block code C is a generator matrix of the dual of the code. As such, a codeword c is in C if and only if the matrix-vector product Hct=0. Quote from wikapedia Record modulation is a form of encoding the data so it can be sorted, this is were the digital waves are stored as binary code, (1s and 0s)

Each bit is a code for an electrical or optical pulse: 1 = high-level pulse 0 = low-level pulse If the binary number 1101 represents analogue amplitude at a certain point in time, four pulses are sent: high, high, low, high

After this it can be stored on to a digital devise such as a hard-drive.

Task 2 Above I have described the analogue to digital process, now I will describe the digital to analogue process. Here is yet anther diagrammed to explain this

The first process that audio has to go through when being reconstructed to original straight is reproduction demodulation, this is where the digital bitstream gos back to its originally modulated binary state. When this process occurs the following happens Waveform shaping of the recorded coded signal to reconstruct the recorded code Extraction of time based code to synchronise individual frames Demodulation of the group code typically to NRZ code

The next process is error correction. This is the same as I have metion in the last task. (Error correction is were bits of data have been damaged or lost, the main way that this is fixed in a digital environment is that they are Parity checked, this is were bits of code that are corrupted or missing are searched, taken out and replaced of squared off. This is normally done by a algebraic calculation.) De-multiplexing is where the pervious signals that were converted into one main signal are re-constricted to two its original sate where there is more than one signal. The next process is where digital signals are converted back to analogue signals

The digital-to-analogue converters must have greater dynamic range (bit depth) than the audio signal itself. Most modern digital recording and processing devices use sigmadelta converters and a method known as oversampling. Oversampling is a process commonly used in digital audio systems to improve Nyquist anti-aliasing filter characteristics, as well as further reducing intermodulation and other forms of distortion. After this anther output sample-and-hold circuit is used. This devise samples and holds the analogue output from the DAC until it is stable; in order to allow the removal of switching glitches (transient oscillations arising as the various bits switch at different times). The output should be a perfect analogue PAM staircase signal; however errors in the form of Jitter can become apparent in poorly constructed devices.

Fig1, forms of jitter. From this there is then anther filter, this is called smoothing filter the basic concept of this is to remove all frequency content above the half-sampling frequency, thereby converting the now analogue PAM staircase waveform into a smoothly continuous final waveform. Over sampling techniques have eliminated the need for brick-wall filters and with 24-bit DACs a more gradual filter slope can be applied. After this, it is then ready to come back through to speaker/headphones, I a full audio signal. Task 3 There are many reason why you should why hard drives are better to use than cds, the main reasons, is size and speed, hard-drives, generally started at 40gb compared to a cd which only 700meg as standard. Also hard-drives generally are faster. Here are some speeds of 2 standard hard-drives
Maximum external transfer rate Average seek time Average rotational latency Spindle speed Available capacities Cache size Recording technology 300 MB/s 11 ms 4.16 ms 7,200 RPM 200, 250, 300, 320, 400, 500, 80, 120, 160, 200, 250, 300, 750 GB 400, 500 GB 8 MB (200 GB) 8 MB (80-250 GB) 8/16 MB (250-750 GB) 16 MB (300-500 GB) Perpendicular Longitudinal NA

As you can see, the maximum external rate is 300mega bites per sound, this means it can play up to 300 mega bits of music per second without stalling, also the seeking, time is 11 Mila seconds, which means it will takes 11millsecound to find your audio files, this is very fast and means you can play large amounts of audio, without compromising your speed.

On this graph you ca see the speed that cds, can read, the average speed you can read a cd , is about 2.47 per second, compared to hard drives, 300 meg per second, saying this, this was running and x16 ( the speed the computer read the speed) I know that the max a computer can read at is x60, but this would only mean that the cd would read at 10.5 megabytes a second, which is still no were near the reading rate of the hard drive.

As you can see from both diagrams, cds are much slower off reading data, and when you take into consideration that you might be playing up to 20 audio tracks at a time, it is easy to understand why hard drives, are a much more efficient. Task 4 When recording my song in logic, I used many connecters, the main being xlr and jack leads, these lead are standard microphone/instruments leads. (xlr Seen left) Here is a basic diagram of how xlr leads work and look like. As you can see from this diagram, xlr leads, have a ground wire, a positive wire, and neutral.

Jack lead diagram.

Jack leads and xlr leads, work very similar to each other, they just carry currents, to the source were its going to be amplified.

Over leads that I use are leads such as fire wire and usb. FireWire (based on technology originally developed by Apple), was adopted in 1995 as an official industry standard. This original version of FireWire is a high-speed, hot-swappable peripheral interface that supports data transfer rates of up to 400 Mbit/sec. FireWire 800 has been released recently, doubling the transfer rate to 800 Mbps. Due to it's speed and hot-swap capabilities, FireWire has become widely used on DV camcorders, external hard disk drives, external DVD burners and more. FireWire based audio interfaces are taking over from USB in popularity due to their better performance on latency issues. USB 2.0 operates at 480 Mbps and can be found in over 2 billion PC, CE, and mobile device s. In addition to high performance and ubiquity, USB enjoys strong consumer brand recognition and a reputation for ease-of-use. Here is some test results I found on fire vs usb.

FireWire, built from the ground up for speed, uses a "Peer-to-Peer" architecture in which the peripherals are intelligent and can negotiate bus conflicts to determine which device can best control a data transfer

As you can see fire-wire is slightly faster and more USB 2.0 uses a "Master-Slave" architecture in which efficient saying the computer handles all arbitration functions and this there are dictates data flow to, from and between the attached peripherals (adding additional system overhead and promblems with it. resulting in slower, less-efficient data flow control) Fire draws Performance Comparison - FireWire vs. USB 2.0 electricity from Read and write tests to the same IDE hard drive connected computer, so not using FireWire and then USB 2.0 show: ejecting devises Read Test: properly can lead, 5000 files (300 MB total) FireWire was 33% faster than USB 2.0 to seriously damaging them. Usb also draws power but no way near as much, so the risk of it damaging your devise is a lot lower.
160 files (650MB total) FireWire was 70% faster than USB 2.0 Write Test: 5000 files (300 MB total) FireWire was 16% faster than USB 2.0

160 files (650MB total) FireWire was 48% faster than USB 2.0

Other connections I use include world clock

Although digital audio has opened a vast array of sonic possibilities for musicians and engineers alike, it has not come without problems. One of these is the issue of word clocks, which basically control the functioning of AD and DA converters. The recording of audio sound has come a remarkably long way in regards quality and ease of storage. I am sure the debate over what medium is better, analogue or digital will continue. But the fact there are converters capable of 192khz, and ever increasing bit rates means that digital recording is now capable of the quality of recording that can match and hereby surpass the analogue method can achieve.