Dynamic range: Its the range in dB between the lowest and the highest program levels.

The lowest acoustic level likely to be encountered is about 20 dBA; the highest can be 110 dBA or may be more. The acoustic dynamic range is thus 90dB. Digital-recording systems such as compact discs can handle dynamic ranges of around 90dB; FM radio, about 45 dB, while AM radio is restricted to 30 dB or even less. The Dynamics Unit is likely to be a limiter/compressor device with possible a noise gate on the more advanced mixers. The dynamic range is determined by the bit resolution. The analog signal is encoded voltage values, which are stored as bits. Each represents a voltage threshold that is half of the previous. The top of the dynamic range is the highest number and is termed full-scale digital. It is important to note that the relationship of full scale digital to analog voltage is not fixed and can range from +20dBV down to 0dBV. The number of bits, i.e. the resolution, determines how far down we go, before the encoding process no longer differentiates the signal. Each additional bit gives us an additional 6 dB of signal to work with, at the bottom of the range. Some advanced consoles incorporate dynamics control on every module, so that each signal can be treated without resorting to external devices. The functions available on the best designs rival the best external devices, incorporating compressor and expander sections, which can act as limiters and gates respectively, if required. One system allows the EQ to be placed in the side-chain of the dynamics unit, providing frequency-sensitive limiting, among other things, and it is usually possible to link the action of one channels dynamics to the next in order to gang stereo channels, so that the image does not shift when one channel has a sudden change in level while the other does not. When dynamics are used on stereo signals it is important that left and right channels have the same settings, otherwise the image may be affected.

Compressors 1. A compressor is a device, which reduces the dynamic range of an audio signal by a controllable amount, with no significant waveform distortion. 2. Threshold. The output level of a compressor is the same as the input level up to the threshold. 3. Compression ratio. It determines the amount of gain reduction. A ratio of 5:1 means that if input level is 5dB over the threshold, the output signal level will be 1 dB over the threshold. The gain (level) has been reduced by 4 dB. 4. Limiting. A very high (ideally infinite) compression ratio. In practice compression ratios higher than about 20:1 are regarded as limiting. On some units the threshold is raised by a fixed amount, typically 8dB, when limiting is selected. Limiters are voltage regulation devices that reduce the dynamic range of the signal passing through. They can be applied at any point in the signal chain including individual input channels, output channels or the post-frequency-divider signal driving a power amplifier. Limiters have a threshold controlled variable voltage gain. The behavior of the device is characterized by its two operating ranges: linear and nonlinear and by the timing parameters associated with the transition between the states: attack and release. The ranges are separated by the voltage threshold and the associated time constants that govern the transition between them. If the input signal exceeds the threshold for a sufficient period of time, the limiter gain becomes non-linear. The voltage gain of the limiter decreases because the output becomes clamped at the threshold level, despite rising levels at the input. If the input level recedes to below the threshold for a sufficient duration, the "all clear" sounds and the device returns to linear gain. Some of the popular software compressors and limiters include the Oxford Dynamics and the Waves L1 Ultramaximizer (see Figures 1 and 2 below).

Figure 1. Oxford dynamics compressor

Figure 2. L1 Ultramaximizer (Waves Limiter)

Crest Factor The 3dB difference between the peak and RMS values only holds as long as the input signal is a continuous single frequency, i.e. a simple sine wave. If the signal has multiple frequencies, the peak to RMS ratio is no longer constant. It is highly volatile or dynamic. The presence of multiple frequencies creates momentary confluences of signals that can sum together for a fleeting moment into a peak that is higher than any of the individual parts. This is known as a transient peak. Most audio signals are transient by nature since we can't dance to sine waves. A strong transient, like a pulse, is one that has a very high peak value and a minimal RMS value. Transient peaks are the opposite extreme of peak-to-RMS ratio from the sine wave. The term to describe the variable peak-to-RMS ratio found in different program materials is the crest factor. The lowest possible crest factor is 3 dB (sine wave). There is no limit to the maximum. The typical crest factor for musical signals is 12 dB. Since our system will be transmitting transients and continuous signals, we must ensure that the system has sufficient dynamic range to allow the transient peaks to stay within the linear operating range. The presence of additional dynamic range above the ability to pass a simple sine wave is known as headroom, with 12dB being a common target range.

Crest factor, RMS, peak and peak-to-peak values. (A) A sine wave has the lowest crest factor of 3 dB. (B) An example complex waveform with transients with a 12 dB crest factor

Noise gates: This term is applied to expanders when the expansion ratio is high 20:1 or more.

About compression Too much buss compression or over-limiting, either when mixing or mastering, results in whats become known as hypercompression. Hypercompression is to be avoided at all costs because: 1. It cant be undone later. 2. It can suck the life out of a song, making it weaker-sounding instead of punchier. Lossy codecs, such as MP3 have a hard time encoding hypercompressed material and insert unwanted side effects as a result. 3. It leaves the mastering engineer (who may be you) with no room to work. Its known to cause listener fatigue, so the consumer wont listen to your record for as long or as many times. 4. A hypercompressed track can actually sound worse over the radio because of the behavior of broadcast processors at the station. A hypercompressed track has no dynamics, leaving it loud but lifeless and unexciting. On a DAW, its a constant waveform that fills up the DAW region. Taste in loudness has changed over the years. In fact, music industries have imposed a louder-and-louder way of listening to the music products available. This happens, because louder music makes always greater impression on the listener and in the long run, that is their main purpose: impression. When the listener gets impressed, he rarely pays attention to the small audible technical or aesthetical details. Consequently, tactics concerning the loudness of the produced music has changed dramatically over the years. The following scheme shows this great change, which starts from very little compression to hypercompression. The years range is from 1985 to 2005.

On the other hand, it is possible to maximize the subjective dynamic range of digital audio signals during the process of requantization. This is particularly useful when mastering high-resolution recordings for CD, because the reduction to 16 bit word lengths would normally result in increased quantizing noise. It is in fact possible to retain most of the dynamic range of a higher resolution recording, even though it is being transferred to a 16bit medium. This remarkable feat is achieved by a noiseshaping process, similar to that described earlier. During requantization, digital filtering is employed to shape the spectrum of the quantizing 0 noise, so that as much of it as possible is shifted into the least 10 audible parts of the spectrum. This usually involves moving the noise away from the 4 kHz region, where the ear is most sensitive and increasing it at the high- frequency end of the spectrum. The result is often quite high levels of noise at high frequency, but still lying below the audibility threshold. In this way CDs can be made to sound almost as if they had the dynamic range of 20bit recordings. Some approaches allow the mastering engineer to choose from a number of shapes of noise until one is found that is subjectively the most pleasing for the type of music concerned, whereas others stick to one theoretically derived correct shape.

Truncation of audio samples results in distortion. (a) Shows the spectrum of a 1kHz signal generated and analyzed at 20 bit resolution. In (b) the signal has been truncated to 16 bit resolution and the distortion products are clearly noticeable.

Lets see now, some cases, where controlling dynamics can be really creative:

Whatever you do, just dont make excessive and unnecessary use of compression. Just make the best of it, trying to make your sound more clear, detailed and why not, impressive. What is always true in music production, is that loudness (i.e. impression in music industry) cannot really compensate for poor recording or audio handling. Bibliography Douglas S., (2010). Audio engineering explained, 1st ed, Focal Press. McCarthy B., (2007). Sound systems: design and optimization, Focal Press. Russ M., (2004). Sound Synthesis and sampling, 2nd ed, Focal Press. Owsinski B., (2008). The mastering engineers handbook, 2nd ed, Thomson Course Technology McCormick T., Rumsey F., (2009). Sound and recording, 6th ed, Focal Press.

Thank you for your time!!!

Afrodite Karantonaki (Greece)