Audio filter

Precise definition of receiver bandwidth by means of audio filters (improves sensitivity). 

In the directly acquired dimension the spectral window can be opened up to cover all frequencies of interest on both sides of the carrier with a proportional increase in the overall data size. As an alternative, it may be sufficient to open the (audio)filters to allow signals to alias to empty regions (if any) with no intensity loss. Carrier shift using time domain tools in data processing can be applied to restore a more convenient arrangement of the spectrum [10]. 

Most audio filters are typically HR filters because a). They are directly transformed from their analog counterparts via the bilinear transform b). They are faster to compute... 

Any peak outside the spectral window will be aliased ( folded ) into the spectral window at a position the same distance from the edge of the window. Aliased peaks are usually reduced in intensity (by the audio filter) and impossible to correctly phase increasing the spectral width will eliminate them and reveal the peak in its correct position. The manner of aliasing depends on the type of acquisition. With the Bruker-type acquisition (alternating acquisition of real and imaginary data samples), aliased peaks appear reflected at equal... 

Normally, the audio filter/amplifier gains are slightly different, the receiver electronics introduces offsets to each channel, and the quadrature detection is not perfect, making the NMR signal in the practice looks like ... 

DLS sets are similar in many ways to the proprietary SoundFonts from Creative Labs and Ensoniq. SoundFonts can be converted to the DLS format with the right software (such as Audio Compositor or Awave Studio), but some quality may be lost as many SoundFonts use audio filters (for example, compression or low-pass filters to make the various notes in a set more uniform). 

These are meant to be used with a capacitor to tune a filter circuit, with resonances in the audio frequency range for reducing and filtering the harmonics or communication frequencies. They provide a near short-circuit for the required harmonics to filter them out of circuit. They may be single-phase or three-phase and connected in series or parallel of the capacitor circuit and may have a fixed or variable reactance, rated continuously with saturated magnetic characteristics. They may incur heavy losses. 

The filter bank is the deciding factor for the basic structure of a perceptual coding system. Figure 2.6 shows the basic block diagram of an static n-channel analysis/synthesis filter bank with downsampling by k. If k = n, it is called a filter bank with critical sampling. A number of basic parameters can be used to describe filter banks used for audio coding ... 

Over the past years, two main types of filter banks have been used for high quality audio coding ... 

The following section gives a short overview of filter banks which are currently used for audio coding purposes. 

QMF filter banks. Quadrature mirror filters (QMF, see [Esteban and Galand, 1977]) have often been proposed for audio coding. The most common configuration is the tree of filters with a two-way split. In one of the early examples [Theile et al., 1987] the 64d filter design from [Johnston, 1980] has been used. The decomposition tree is set up so that the filter bands resemble critical bands. The QMF halfband filters are non-perfect reconstruction, but with perfect alias cancellation by design. The reconstruction error of the analysis/synthesis pair can be held at small amplitudes by increasing the filter length. 

The MDCT is known under the name Modulated Lapped Transform ([Malvar, 1990]) as well. Extensions using an overlap of more than a factor of two have been proposed [Vaupelt, 1991, Malvar, 1991] and used for coding of high quality audio [Vaupelt, 1991]. This type of filter banks can be described within the framework of cosine-modulated filter banks ([Koilpillai and Vaidyanathan, 1991][Ramstadt and T anem, 1991, Malvar, 1992]). 

Adaptive filter banks. In the basic configuration, all filter banks described above feature a time/frequency decomposition which is constant over time. As mentioned above, there are possibilities to switch the characteristics of a filter bank, going from one time/frequency decomposition to another one. We explain the basic principle using the example of MPEG Audio Layer 3 ... 

Filterbanks. There is still continued research on filter banks for high quality audio coding. Topics include wavelet based filter banks, low delay filter banks [Schuller, 1995] or variable filter banks allowing a higher degree of variability than classic window switching [Princen and Johnston, 1995],... 

It is easy to assume that the filter coefficients in equation 5.15 (a / and b j]) are constant. In fact, there are many instances when this is not the case. For example, in real-time audio processing a user moves a slide potentiometer (either physical or possibly on the display) this is digitized and the host processor must change the coefficients in various filters. 

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