History and technical development of the ribbon microphones [1932-2016]

The ribbon microphone was patented in 1932 by Harry F. Olson. It consists of an extremely thin aluminium foil ribbon suspended in a magnetic field. Pressure gradients in the air cause the ribbon to move. The movement of the ribbon in the magnetic field generates a small electromotive force (emf) which can be amplified and recorded. The ribbon microphone quickly gained popularity with audio engineers for its uniform frequency response.

When magnetic tape became the dominant recording media, ribbon microphones became less popular and condenser microphones took over. With recordings and sound mixing processes making use of magnetic media there is always a slight loss of high frequencies. This problem could be remedied by large capsule condenser microphones. These microphones have a number of resonances in the 8 kHz to 12 kHz range that enhances the high frequencies before recording. The capsules of condenser microphones are tensioned tightly, causing the high-frequency resonances. 

The aluminium foil element of the ribbon microphone is only lightly tensioned, causing resonance at very low frequencies. When digital recording became the order of the day, ribbon microphones made a major comeback because high-frequency transfer loss was no longer an issue. Its ability to record fast transients accurately without adding upper-range resonances became again a very positive attribute.

Articles about the ribbon microphone dates back as far as 1931. Most of the early publications were authored by Harry F. Olson who patented the ribbon microphone. Few publications besides those by Olson can be found about the ribbon microphone, but plenty of patents relating to the ribbon microphone are freely available. A selection of those patents is discussed in the paragraphs to follow.

Olson 1932 

Harry Olson filed the first patent for the ribbon microphone in 1931 and the patent was awarded on 25 October 1932 with the title “Apparatus for converting sound vibrations into electrical variations”.

The patent illustrations are of very sturdy mechanical designs as illustrated in picture below:

It describes the ribbon as a relatively small item that is supported in such a way that it resembles the motion of a particle in free air. A device of this nature is classified as a velocity microphone. The combination of mechanical parts surrounding the ribbon is called a baffle. The size of the baffle around the ribbon is calculated according to the highest frequency that the microphone is designed for. The baffle should be designed so that the path length from the front of the ribbon to the rear of the ribbon is half the wavelength of the required frequency. Picture below provides a graph from the patent to illustrate the effect of the baffle size on the frequency response of the microphone. 

The ribbon is made of conducting material that is light in weight and that has low elasticity, for example aluminium foil. The ribbon must not be stretched tightly between its supports. It is crimped in order to suspend it rather loosely between its supports to promote flexibility along its whole length. The supports are made of non-ferro-magnetic conducting material, but it is electrically isolated from the ribbon by non-conductive material. The two signal wire leads are electrically connected to the two ends of the ribbon. The light weight and small restoring force of the ribbon causes its natural vibration frequency to be below the audible range. Tests that were done before the patent submission had shown that a natural vibration frequency of approximately 10Hz produced the most desirable results. The patent states that “When a diaphragm of small mass is suspended in this manner its mechanical reactance is small compared to the impedance of the air. In other words its mass reactance is negligible over a large frequency range compared to the acoustic resistance of the air which it displaces.” 

The ribbon is suspended in the air gap between two poles of a magnet in such an orientation that its surfaces are parallel with the magnetic force lines. The magnet can be a permanent magnet or an electromagnet. The gap between the ribbon and the magnet poles are kept to a minimum to prevent the leakage of air around the ribbon, but the gaps must still be sufficient to prevent frictional contact between the ribbon and the magnet. The patent suggests an air gap of 5.6mm with the ribbon slightly narrower. The ribbon cuts the magnetic field lines while moving in the air gap between the poles because of the sound pressure variations across it. This causes an electromotive force that is proportional to its movement. The electromotive force can be amplified with suitable electronic equipment.  

The magnetic pole pieces with its supporting structure forms the baffle. The baffle increases the path length from the front to the back of the ribbon. The length of this path has an influence on the response of the microphone. The paths around the top and bottom supports of the ribbon are shorter than the paths around the baffle, but its effect is relatively small because it influences only a small part of the ribbon. The clamping structure that secures each end of the ribbon is made of non-magnetic material (ex. copper or brass if it is made from metal). Although it is desirable to make the ribbon as light as possible, it is sometimes necessary to vary its thickness in order to increase its efficiency within the particular baffle setup. The movement of the ribbon is caused by the phase difference of the sound wave between the front and back of the ribbon. The phase difference is determined by the distance that the sound wave has to travel around the baffle from the front to the back of the ribbon. The greatest phase difference occurs when the path length around the baffle is half a wavelength of the sound wave under question. Olson provides a helpful visual representation of this effect in his patent. Picture below shows a horizontal cut through the microphone and a sinusoidal representation of a sound wave. If the path length A-B around the baffle is plotted as E-F on the axis of the sound wave, then the pressure difference between points A and B on the ribbon will be equal to the sum of C-E plus D-F.

The sound intensity at the opposite sides of the baffle is virtually the same for all wavelengths that are longer than twice the distance around the baffle. At wavelengths shorter than this, the intensity on the approaching side of the ribbon increases and the intensity at the retreating side decreases. The reasoning follows that the pressure difference across the ribbon is proportional to the frequency as long as the distance around the baffle is less than half the wave length. Due to the nature of the design the ribbon microphone exhibits very directional characteristics. Sound waves coming from an angle will produce less of a pressure difference across the ribbon. A sound wave directly from the side will produce virtually zero pressure difference. The pressure difference can be calculated with simple trigonometry rules. An alternative design is also provided in the patent. The alternative design does not make use of a ribbon, but of a lightweight diaphragm. This design is however overly complicated and will not be discussed further. 

Olson & Weinberger 1933 

In 1933 Olson, in cooperation with Julius Weinberger, filed a patent that made use of a combination of the pressure gradient (velocity) ribbon microphone and a pressure component microphone to achieve unidirectional operation. Unidirectional operation is desirable in order to improve the ratio of the sound source relative to the sound reflections in the room. The patent achieved its purpose through the use of a normal ribbon (as in the 1931 patent) assembled in series with a modified ribbon. The modified ribbon was adapted to act as a pressure microphone by enclosing the back of the ribbon. The normal and modified ribbons are working in phase for sounds generated in front of the microphone, but are out of phase for sounds originating from the rear side of the microphone. This patent illustrates the influence of the baffle in the extreme case of making the baffle infinitely large by enclosing the rear side of the ribbon completely. 

Anderson 1937 

Leslie Anderson added electronics to Olson and Weinberger‟s unidirectional microphone to file a patent in 1937. The electronics made it possible to adjust the phase differences between the two ribbons, thereby making it possible to adjust the directionality of the microphone from the mixing desk.

Ruttenberg 1938

In 1938 Samuel Ruttenberg filed a patent to address one of the imperfections of the ribbon microphone. When speaking close to the microphone, the low frequencies are over emphasised because the higher frequencies are attenuated. The high frequency attenuation happens because different sections of the ribbon move out of phase. Thus one section cancels out the electrical current that is generated during vibration of another section. The sections move out of phase due to the fact that the ribbon is longer than the wavelength of the higher frequency sound waves. Ruttenberg addressed this problem by designing a special housing for the microphone which closes up the rear of the microphone with an adjustable shutter. When closing up the rear of the microphone, its characteristics are changed from that of a velocity microphone to that of a pressure microphone because the sound wave does not have access to the rear of the ribbon. This idea clearly borrows from the same principles than Olson and Weinberger‟s patent on the unidirectional microphone. The microphone‟s operation in pressure mode tends to minimise the effect of high frequencies being attenuated. This patent clearly illustrates that tampering with the physical surroundings of the ribbon microphone, does have a distinct influence on its operation.

Bostwick 1938

Telephone conferencing has been around for longer than one would expect. In 1938 Lee Bostwick already addressed the problem of feedback between speaker and microphone during teleconferencing. He filed a patent for a device making use of two ribbon microphones and a loudspeaker. The device is constructed with a loudspeaker facing downwards onto a deflector underneath that reflects the sound waves in a horizontal direction toward the conference attendees. Two ribbon microphones are fitted perpendicular on top of each other, both on top of the loudspeaker. Because of its directionality, the ribbon microphones are insensitive to the sound waves emanating from below it, but are sensitive to the voices of the conference attendees around the table.

Bostwick‟s teleconference solution

Olson 1940

In 1940, Harry Olson filed a patent for an improved version of the 1933 unidirectional microphone . Olson discovered that he can design a unidirectional microphone with a single ribbon instead of the dual-ribbon method used previously. Furthermore, this microphone could be easily changed from unidirectional to bidirectional, to non-directional operation. The method of achieving this was to place a pipe structure (resemblance of a smoking pipe) behind a single ribbon. The pipe ends in a labyrinth that is filled with a soft fabric acting as acoustic resistance. The pipe has shutters that can be opened or closed to achieve the desired results, i.e. shutters open allows bidirectional operation, shutters partly closed enables unidirectional operation and shutters completely closed constrains it to non-directional operation. This patent illustrates once again the influence of the surroundings of the ribbon on its operation.  

Anderson 1942 

Leslie Anderson built upon Olson‟s 1933 patent by filing a patent in 1942 about the magnetic equalization of sensitivity in a unidirectional microphone. It is fundamentally an improvement on Olson and Weinberger‟s design by adding a mechanism to adjust the magnetic fields for the two ribbons and by changing the shape of the pole pieces in such a way that it is possible to vary the flux density of one air gap relative to the other. This patent indicates support for Olson and Weinberger‟s idea of modifying the baffle in order to change the operation of the microphone for a specific purpose.

Rogers 1942 

Ernest Rogers filed a patent in 1942 for a microphone with selective discrimination between sound sources . He claimed that the microphone can be used to determine the direction of a sound source. The microphone receives sound waves approaching the microphone straight-on, but attenuates sound waves approaching the microphone from an angular displaced direction. The construction of the microphone is basically four ribbon microphones assembled in an X-pattern. The ribbons are connected to a mixing circuit in such a way that the phase relationship between the ribbons can be adjusted. By adjusting the phase relationship, the microphone can be tuned so that the phases of sound waves from a certain direction cancel each other out, while the phases of sounds waves from another direction will add to each other. 

Olson 1946

The ribbon microphone inherently has a directivity pattern. Olson filed a patent in 1946 to combine two ribbon microphones in a perpendicularly fashion in order to get a microphone that has a 360° pattern on the horizontal plane. This microphone would be ideal for use in orchestras for example, recording all the instruments around it, but attenuating sound waves reflecting from the ceiling and from the floor. Part of the idea is borrowed from Bostwick‟s teleconferencing microphone discussed earlier. 

Anderson 1947 

Anderson‟s patent application in 1947 was filed as an improvement on Olson‟s 1931 design. According to Anderson, Olson‟s design displayed a considerable drop in output when the distance around baffle approaches one fourth of the sound wave‟s wavelength. Anderson‟s design claimed to have a more uniform response over the operating range of the microphone and also to have an enhanced high frequency response compared to the conventional design. This was accomplished by mounting one or more semi-circular bands behind the ribbon. The bands provide cavities that are resonant at the frequencies where the enhancements are deemed necessary.  

Nature of phaselock

A phaselock loop contains three basic components

1. A phase detector (PD).
2. A loop filter
3. A voltage-controlled oscillator (VCO) , whose frequency is controlled by an external voltage.

The phase detector compares the phase of a periodic input signal against the phase of the VCO; output of the PD is a measure of the phase diffrence between its two inputs. The diffrence voltage is then filtered by the loop filter and applied to the VCO, Control voltage on the VCO changes the frequency in a direction that reduces the phase diffrence between the input signal and the local oscillator.
When the loop is locked,the control voltage is such that the frequency of the VCO is exactly equal to the average frequency of the input signal.For each cycle of input there is one, and only one , cycle of oscillator output.
One obvious application of phaselock is in automatic frequency control (AFC). Perfect frequency control can be achieved by this method,whereas conventional AFC techniques necessarily entail some frequency error.
To maintain the control voltage needed for lock it is generally necessary to have a nonzero output from the phase detector.Consequently , the loop operates with some phase error present ; as a practical matter, however this error tends to be small in a well designed loop.

TIP #1 - Effect of sound level on the perception of pitch

The level of sound affects the perception of pitch. For low frequencies, the pitch goes down as the level of sound is increased. At high frequencies, the reverse takes place—the pitch increases with sound level. The following is an experiment within the reach of many readers that was suggested by Harvey Fletcher. Two audio oscillators are required, as well as a frequency counter. One oscillator is fed to the input of one channel of a high-fidelity system, the other oscillator to the other channel. After the oscillators have warmed up and stabilized, adjust the frequency of the left channel oscillator to 168 Hz and that of the right channel to 318 Hz. At low level these two tones are quite discordant. Increase the level until the pitches of the 168-Hz and 318-Hz tones decrease to the 150-Hz–300-Hz octave relationship, which gives a pleasant sound. This illustrates the decrease of pitch at the lower frequencies. An interesting follow-up would be to devise a similar test to show that the pitch of higher frequency tones increases with sound level.

TheAshkOne - Vitality [A Trance Live Jam]

A Trance Jam Produced Using My Handmade Synthesizer

Paper Recommendation #1

0.04-mm2103-dB-A Dynamic Range Second-Order VCO-Based AudioΣΔADC in 0.13-μm CMOS

Abstract:

This paper presents a compact-area, low-power, highly digital analog-to-digital converter (ADC) for audio applications. The proposed converter is implemented using only oscillators and digital circuitry, without operational amplifiers nor other highly linear circuits. The ADC consists of two twin secondorder ΣA modulators, which can work both individually or in a pseudodifferential configuration. The proposed system has been implemented in a 0.13-μm standard CMOS technology. The single-ended configuration occupies an active area of 0.02 mm 2 , is powered at 1.8 V with a current consumption of 155 μA, and achieves an A-weighted dynamic range (DR) of 98 dB-A. The pseudodifferential configuration achieves 103 dB-A of A-weighted DR at the expense of doubling the area and power consumption.

Published in: IEEE Journal of Solid-State Circuits ( Volume: 53 , Issue: 6 , June 2018 )

DOI: 10.1109/JSSC.2018.2799938

Op Amps with Data Converters (Part 2)

It is useful to examine a few general trends in data converters, to better understand any associated op amp requirements. Converter performance is first and foremost; maintaining that performance in a system application is extremely important.
In low frequency measurement applications (10 Hz bandwidth signals or lower), sigma-delta ADCs with resolutions up to 24 bits are now quite common. These converters generally have automatic or factory calibration features to maintain required gain and offset accuracy. In higher frequency signal processing, ADCs must have wide dynamic range (low distortion and noise), high sampling frequencies, and generally excellent ac specifications.
In addition to sheer performance, other characteristics such as low power, single-supply operation, low cost, and small surface-mount packages also drive the data conversion market. These requirements result in application problems because of reduced signal swings, increased sensitivity to noise, and so forth. In addition, many data converters are now produced on low-cost foundry CMOS processes which generally make on-chip amplifier design more difficult and therefore less likely to be incorporated on-chip.
Analog input to a CMOS ADC is usually connected directly to a switched-capacitor sample-and-hold (SHA), which generates transient currents that must be buffered from the signal source. On the other hand, data converters fabricated on Bi-CMOS or bipolar processes are more likely to have internal buffering, but generally have higher cost and power than their CMOS counterparts. It should be clear by now that selecting an appropriate op amp for a data converter application is highly dependent on the particular converter under consideration. Generalizations are difficult, but some meaningful guidelines can be followed.

• Higher sampling rates, higher resolution, higher ac performance

• Single supply operation (e.g., 5V, 3V)

• Lower power

• Smaller input/output signal swings

• Maximize usage of low cost foundry CMOS processes

• Smaller packages

• Surface-mount technology

The most obvious requirement for a data converter buffer amplifier is that it not degrade the dc or ac performance of the converter. One might assume that a careful reading of the op amp datasheets would assist in the selection process: simply lay the data converter and the op amp datasheets side by side, and compare each critical performance specification. It is true that this method will provide some degree of success; however, in order to perform an accurate comparison, the op amp must be specified under the exact operating conditions required by the data converter application. Such factors as gain, gain setting resistor values, source impedance, output load, input and output signal amplitude, input and output common-mode (CM) level, power supply voltage, and so forth, all affect op amp performance.