Q+A: What Do Microphone Specs Mean?

There are a lot of microphones out in the marketplace and all the spec sheets are not identical in the information they provide or how they provide it. Firstly we need to understand what it all means and what you should look out for in a really good quality microphone that will last you till the end of your career!
IMPEDANCE
This isn't how good your microphone is at reproducing with other microphones but effectively how good the microphone signal is at travelling down long cables. Obviously using low-impedance microphones means less hum picked up along the way (300 Ohms or less is good) you can always help this statistic by using as shorter cable as needed. Alternatively get a digital microphone, which is the same as a condenser microphone but with an analog/digital (A/D) converter built into the microphone and so the signal in the cable is digital and picks up no hum.
MAXIMUM SPL
This refers to the maximum level of sound your microphone can pick up without distortion. 120db being good and above 140db being excellent. 120db the threshold of pain for human ears to give you a reference
SELF NOISE
This refers to how noisy the microphone is. A <25dba is good, <20dba very good and less than 15dba is excellent. The lower the better, especially for quieter scenes or applications. For louder applications like super cars or loud music 30dba is usually fine due to the signal being so much louder then the microphone its self.
[The "a" in dba stands for a scale of frequency response that resembles how the ear picks up sound]
SIGNAL TO NOISE RATIO
This is another measurement for how noisy a microphone is but more relevant to the strength of the signal to the inherent noise. 69dba is good, 74dba is very good and 79dba+ being excellent. Again, this is important of quieter applications as louder sounds already produce louder signals.
SENSITIVITY
This is basically telling you how powerful a signal is being generated by the sound hitting the microphone. Condenser mics have the highest sensitivity. Generally above 6mv/pa is very sensitive and (sensing a pattern yet? ) the higher the sensitivity the stronger the signal and the lower the pick up of hum and other noise.
Hope that clears up any issues, otherwise just comment, like, tweet or +1 below
Thanks

7.1 will become the new standard

Speaker systems for your home theatre are usually your normal 5.1 setup with either Dolby or DTS that runs it all. (find out more about Dolby vs DTS here)

You have your 5 surround speakers (center bar, front L+R and rear L+R with the .1 being the subwoofer for those deep "end of the world" disasters etc..)

This is great for being able to follow that transformer running across screen or that slow motion bullet passing by but thats still all based around the front to enhance the visual actions or presence on screen if you're in that sweet spot in the middle of the square speaker setup.

Now with 7.1 its a more "real" world environment with 2 extra speakers (making 7.1) at the rear sides  Left + Right. This giving you a full surround sound experience as there is much better coverage from the sides. In 7.1 that helicopter swooping round to destroy that enemy will really feel like its not just jumping from front to the rear but thundering all the way round!

In the interest of only using my own work you will have to put up with my terrible drawing.

If you want more info then watch this DTS-HD presentation on youtube

Thanks for viewing and please tweet, like or +1 to share the info and resources in the blog, Matt Price.

Oldest Known Recording Unearthed From 1860

Previously the oldest recorded sound was thought to be Edison's phonograph recording of a children's nursery 

rhyme "Mary had a little lamb.." in 1877. So what did they use before the phonograph?

The "phonautograph", is used by etching paper covered in soot. US scientists used a virtual stylus to read the lines. The recording was found by audio historian David Giovannoni who said to the Associated Press "When I first heard the recording as you hear it ... it was magical, so ethereal,"

"The fact is it's recorded in smoke. The voice is coming out from behind this screen of aural smoke."

The phonautograph was made by a Parisian inventor, Edouard-Leon Scott de Martinville. The new recording will be presented on 28 March at a conference of the Association for Recorded Sound Collections at Stanford University in California.

The video below is the 10 second clip of "Au Clair de la Lune".




This is the video of Edison's recording of "Mary had a little Lamb"



Thanks

Recording Meters and Levels

Hello There,

When i started out in Sound Recording i was confused by the different meters and how you can record at -dB, surely that was impossible? Why record at -12dB on a digital system but aim for 6 on a VU meter? So here is a run down of what it all means.

Digital Systems compared to analogue systems

Digital is obviously here to stay for many reasons but an important note in terms of levels are how the different systems deal with very loud signals.

Sound is always analogue at the source and how we hear it only in between can it be digital.

Analogue recording systems gracefully handle extremely loud signals and slowly get completely messed up. Digital on the other hand just cuts off and dies horribly!. This is due to Sampling theorem and the Dynamic Range of data that can be stored on a sample before it starts to distort. Thus it is very important to have headroom and understand your meter system.

Digital Meters

These range from around -60dB up to 0dB. The zero has no actual value other than the loudest sound that can be free of distortion. When measuring in just dB it is only a representative change like a percentage or ratio compared to another value.

Example: If i said his voice was 20dB higher then hers, it would be his voice is 10 times louder than hers. Just saying my voice is 10dB means nothing unless compared to something else, if i said my voice should be at 10dBu or dBA then it means something. (But it will get it`s own topic soon)

Meters!!

These are different standard for playback purposes and final mixes of program but easily applied to production recording devices.

PPM (Peak Programme Meter - BBC)

Range = 0 to 7
line-up: 4 (needle straight up)
Max level: 6
Each division: 4dBs
Average level: 5
Edit suite/CTA measurement


See a real version of a PPM meter system 

VU (Volume Units - USA)
line up:-4 (between 3 & 5 black)
Max level: +4 (Red notch right of 3)
Each division: 1dB
Average level: 0
Analogue VTRs

See real example of VU meter system

Digital
line up: -20
Max level: 0
Each division: Varies
Average level: -14
Digital VTRs, Digital Recorders

See real example of Digital Meter System






As always i hope this answers everything and if not drop a comment below and i shall amend it asap.

Cheers,

Matt

www.soundrolling.com

Don't Forget to +1 This Below

Dither and Quantization Error

Dither is an internationally applied form of noise used to randomize quantization error, preventing large scale patterns such as "banding" in images. 

Example of image "Banding"

Dither is routinely used to processing both digital audio and digital video data and usually one of the last stages in post production for compact discs. 

Quantization Error is the difference in the actual analog value and the quantized digital value. This is due to either rounding of digital views or truncation. The error is sometimes considered as an additional random signal called quantization noise because of its "stochastic" or non-deterministic behavior

Cheers Guys,

Matt Price

5.1 Surround Sound, Dolby or DTS?

This is section explaining what Dolby Digital 5.1 and DTS 5.1 are, along with a bit about why people think one is better than the other etc...

What is 5.1?
Refers to this speaker set up: (left front, left rear, right front, right rear, and center), plus a subwoofer channel (the .1 in 5.1)

What is Dolby Digital Surround Sound?
This is the most common format for surround sound on media such as movies. Its a discrete channel surround sound format because the output has been controlled to come from a variety of speakers, allowing a car to sound like it is moving across the screen etc....

What is DTS?
DTS (Digital Theater Systems) is a digital surround-sound system first introduced in theaters in 1993. DVDs encoded with a DTS soundtrack require a DVD player and stereo receiver equipped with DTS-processing capability. This is partly due to the DTS demands for more data space on a DVD (often sacrificing bonus features), but many believe the audio quality to be superior to that of Dolby Digital 5.1-channel surround sound.

What do people think?
I don't happen to be 'the people' but reading up and around the issue DTS has a higher data rate and so that roughly translates into 'better' sound. Where as many Dolby fans argue that low compression but higher data rate provides 'better sound'....

Ill need another 10 years to provide a better description of 'better sound' but many blogs start with the underdog against the giant, like in all fairy tales...

I should be pointed out that both Dolby Digital and DTS Digital Surround encoding schemes now have even higher sampling rate of 48 kHz at 20-bits per sample, thus yielding an even wider dynamic range between sound level extremes of approximately 120dB.

They both have to be compressed in some form to fit on the disk so that is always going to be an issue, raw data over efficiency is hard to prove when sound is subjective.

"Compression and bit-rate are not the only differences when comparing Dolby vs. DTS formats. For example, the added rear surround channel in Dolby's extended surround format 'Dolby Digital EX', is matrixed over the two left and right surrounds, rather than discrete; instead the DTS counterpart uses a discrete channel. This also explains why DTS ES (Extended Surround) can provide a more precise location for the rear-effects soundstage than the Dolby EX format." - Source here

"Both Dolby Digital and DTS audio are capable of achieving similar end results in delivering surround sound, even though the lower compression/higher bit-rate of DTS Digital Surround should theoretically yields apparent benefits in sound quality.

At the same time, one cannot ignore the fact that these two formats make use of different coding schemes and syntax to perceptually compress audio.

This means that efficiency in terms of data utilization between these two formats is different. Therefore, a Dolby vs. DTS direct comparison based solely on these formats raw bit rates cannot be taken as a measure of sound-quality.

Thus, while it is objectively possible to compare the resultant sound quality for the same audio format encoded at different bit rates, and therefore, to determine whether the same format in a moviehouse application sounds better or worse than in a consumer implementation in home entertainment, it is not so straightforward when dealing with different formats.

Rather, the reality is that for identically sourced audio content, it would be much easier for the listener during a Dolby vs. DTS 'blind' listening test to notice a change in sound quality when changing the playback equipment say between different brands, than when changing from a Dolby Digital to the DTS surround audio track." - Source here

So it is down to choice really and if it you set up your theatre, home cinema etc... then you might notice one suiting your needs better, Sadly this is going to be a never ending debate but at least you can argue both ways forever with your friends.

Thanks

Matt Price

Sampling theorem

The Nyquist–Shannon sampling theorem, after Harry Nyquist and Claude Shannon, is a fundamental result in the field of information theory, in particular telecommunications and signal processing. Sampling is the process of converting a signal(for example, a function of continuous time or space) into a numeric sequence (a function of discrete time or space).

Sampling theorem which in essence shows that a bandlimited analog signal can be reconstructed perfectly from an infinite sequence of samples if the sample rate exceeds 2B samples per second. B is the highest frequency in the original signal. If the signal contains components at exactly B hertz, then samples spaced 1/(2B) seconds do NOT completely determine the signal.

E.g. 20,000 Hertz into this forumla will not be reconstructed perfectly at anything less than 0.00003 seconds per sample. Recording at a sample rate of 48kHz is sampling at 0.00002 seconds per sample, hence for situations like recording dialogue where the highest frequency for normal conversation is below 20kHz (because we can't hear higher).

Possibly Useful Drawing of a 2hz sine wave

So in recording effects if you wanted to have flexibility to manipulate then you can record at 192kHz and then the spacing is 1 sample every 0.0000052083 of a second, this is useful for slowing down samples more effectively than less samples.

This is again like most perfect theories based on a perfect world and so is an approximate in our not so perfect world.

Hope this ended up making sense.

Thanks,

Matt Price

ADR, dubbing, looping and post sync

All the names in the title mean the same thing roughly. Though you should be aware that you can dub on music to a film. ADR stands for automated dialogue replacement, this is the post production process for replacing the lines, words, sentences of dialogue with a new copy or different lines. 

This is for various reasons which include but are not limited to: Poor recording on set, pops and clicks that can't be cleaned, wrong lines, poor performance and so on... This is usually hell for many directors, actors and crew because it is time consuming and so expensive, coupled with the fact the voice has to match the performance even though the time from the actual recording to the ADR session could be months or even years!. 

Foreign films obviously get dubbed in the language of the country they are being distributed to and so don't necessarily need to be in perfect sync. Though across Europe different countries have different rules, naturally. as seen on this map here.

Looping is also a name for dialogue replacement as the lines that need to be retaken will be done over several takes so the actor can get into the rhythm of the piece and deliver a better performance.

This is was a brief one as im in between filming,

Thanks for viewing, comments appreciated,

Matt Price

Radio Mics: Sennhieser EW112p ENG G3

This is a quick video i did for a demonstration on the radio mic set ew112p G3 ENG set from Sennhieser. They are around £420 a pair and come with:

1x Transmitter

1x Receiver

1x AF Jack to XLR (female)

1x AF Jack to 3.5mm Jack

1x ME2 Microphone

1x Camera mount for receiver 

1x Tie Clip

1x Mic Head

4x AA Batteries

The first part is a demonstration of performance in an ok area and a really noisy one. The really noisy one is compared with what my phone (Samsung Galaxy S2) picks up on the recording.

HD Version to replace this one soon!

I think for the price these radio mics perform really well and are easy to tune and change settings on the fly. I purchased mine from ProAv.co.uk which i highly recommend for their customer service, all questions i had where answered and delivery was prompt (within the 2-3 day limit)

Enjoy and Thanks,

Matt Price

Reverberation Time

A reverberant sound in any closed space like a room or concert hall diminishes as the sound energy is absorbed by multiple interactions with the surfaces of the room. Though the speed it takes to diminish depends on the variable of the rooms surfaces, for instance a reflective  or "live" room will take longer than a very absorbent or "dead" room. The time for the sound to completely die away will depend how loud the sound is at the start and how well the observer hears. 


The standard reverberation time has be established as the time it takes for a sound source to diminish by 60 dB below its original level. This is established at 60dB from testing in auditoriums where the loudest crescendo in an orchestra is 100dB and typical background noise for a music-making area being around 40dB. 60dB is also roughly the dynamic range for orchestral music.


In terms of time it depends on the rooms use. In a classroom you want the room to have low reverberation time to keep articulation clear and reduce build up from people talking. In an auditorium you probably want a bit more so you can really feel the music. As a general rule you wont get long reverberation time in a small room because of how fast the energy is bouncing off the surfaces and being absorbed a bit each time.


Reverberation time is just one part in creating pleasing environments for sound, you also need to consider the absorption in terms of frequencies.


Thanks,


Matt Price

Q+A: 6 Miscellaneous Sound Questions... Part 1

Compilation of general and random questions...

1. What does 94.1 mean on a radio dial?

This mean the station is emitting at 94.1 million Hertz or 94,100,000 waves per second.

2. If sound becomes louder, what wave characteristics increase? frequency, wavelength, amplitude or speed?

Only amplitude.

3. How much bigger is a sound at 40dB compared to 0dB? 

10,000 times.

4. How much bigger is a sound at 110dB compared to 50dB?

1,000,000 times.

5. What does supersonic mean? 

Faster than the speed of sound

6. General speeds according to (this site)

WATER:
Distilled water = 1489 m/sec
Sea Water = 1531 m/sec (higher because it is denser)
WOOD:
Ash, along the fiber = 4670 m/sec
Ash, across the rings = 1390 m/sec, about 3 times slower, and a little slower
than the speed of water!
Beech, along the fiber = 3340 m/sec
Elm, along the fiber = 4120 m/sec
Maple, along the fiber = 4110 m/sec
Oak, along the fiber = 3950 m/sec.


Thanks,


Matt Price

Inverse Square Law Of Sound

Hello All,


This blog is on the inverse square law of sound which is a very important law that not many really know about (we will get onto that later) Sound waves propagate spherically and so distributed over and ever increasing surface of diameter at the front surface of the wave. The inverse square law tells us that every doubling of the distance from the sound source in a free field situation the sound pressure level will diminish by 6 decibels.


Why many may not know this law is that it is hard to get a free field situation such as an explosion of a bomb in mid air. Though not perfect for real life situations it is a very important guide to sound sources and how they diminish over distance. 


As an example lets say a person shouts at 70dB at 3 feet away from you in a room, at 6 feet it will be 62dB and at 12 feet 56dB and at 24 feet 50dB. This example on works as a guide not taking into account reflections from any walls of a room and any other sound in the room at the time of shouting.


Thanks,


Matt Price

World-a-lies-ing

Instead of you having to read everything like back in the 1800's i thought i would indulge all your senses to a quick video interview with Walter Murch. I will write up another blog that will go into more detail about different types of reflections of sound off surfaces and all of that jazz later today as i have just a few more days till im back on several shoots. Enjoy!

Manipulating sound until it seemed to be something that existed in real space. This refers to playing back existing recordings through a speaker or speakers in real-world acoustic situations, and recording that playback with microphones so that the new recording takes on the acoustic characteristics of the place it was "re-recorded." [via filmsound.org]

Thanks,

Matt Price

Microphone Polar Patterns...

This outlines the types of polar patterns. Polar patterns are the field of focus that the microphone will pick up due to the position of the diaphragm and other factors.

0 degrees is straight infront of the microphone, 180 degrees is directly behind the microphone

Omni Directional
This encompasses a field of all directions, or nondirectional. In theory it should be a perfect sphere but because the microphone its self is not infinitely small it gets in its own way, just like if it was on a persons body. So the smaller the body the better the polar pattern. The wave length of 10Khz is 3.4cm so small microphones are effective at even high frequencies.


Cardiod

This is the most common unidirectional microphone. There is a dip at the back due to the microphone casing and the direction of the diaphragm. A Sub cardiod is similar to the omnidirectional polar pattern but flatter. and Super and Hyper cardiod have tighter patterns compared to the basic cardiod pattern, but they add more sensitivity to the cardiod because of there higher sensitivity.

Shotgun

These are the highest directional microphones. They have small fields of sensitivity to the left, right and rear. This is still significantly smaller than other directional microphones. This is the most common on film sets for booming over subjects.

See Also: Condenser Microphones, Dynamic Microphones and Carbon Microphones.


images are free to reuse via wikipedia

Thanks,

- Matt Price

Parabolic Refectors

A Parabolic microphone is a microphone with a parabolic reflector fitted to it. This collects and focuses sound on the microphones head like a TV dish picks up satellite signals. This has many advantages for picking up and amplifying distant sounds that you want to isolate from the rest of the environment. A classic example is bird song because parabolic reflectors are very good at picking up high frequencies. this is also used in sports broadcasting, eavesdropping and espionage. 

The high frequencies are picked up more due to the direct physical laws of sound waves. This is because they only focus waves with a wave length much smaller than the diameter of the parabola. To obtain hi-fidelity sounds including the lower end frequencies (down to 20Hz) you need a parabola around 17 metres. This is because if we say the speed of sound is 342 m/s through the air (speed of sound) 342 ms / 20 Hz = 17 metres. Parabolic reflectors will sacrifice the lower end frequencies to be more manageable sizes.

Representation of how a reflector works with microphone facing the central point  for maximum pickup.

Thanks,

- Matt Price

Q+A: Sample rates explained....

Sample rate is the number of samples per unit of time taken from a continuous signal to make a discrete signal, the unit of sample rates are Hertz. We record a discrete signal because we only need lots of little samples of the signal to track how loud it is etc... So the higher the samples the more of these measurements are taken per second. The standard for recording dialogue is 48kHz (48,000 hz) or 44.1kHz and unless your playing on really nice speakers you really wont be able to tell the difference.

For some sound effects recording people record at 192kHz which has advantages for slowing effects down and manipulating them. Also higher sample rates help prevent aliasing, which refers to an effect that causes different signals to become indistinguishable 


also see: Bit Depth

Thanks,

- Matt Price

Q+A: What is difference between room tone, ambience, wild sound?

Hey,

These definitions vary depending who you are so this is ONLY a guide =].

Room Tone: This is the natural sound that a room "makes" from the natural reflections in the room. This is important for being able to patch over areas of sound so make sure you GET IT AFTER THE SCENE and hold your microphones ideally in the exact same locations.

Ambience: This is similar to room tone as it is the natural environment. However what makes it more specific is can be illustrated with an example of a scene in a restaurant where people are in the background, you will want to collect the general ambience (atmos) to again be able to patch over any gaps in post production.

Wild Sound: This can be any sound that doesn't need to be recorded in sync and is usually an object such as a fridge in a kitchen.

Feel i have missed anything? Then just comment =]

Thanks,

- Matt Price

Q+A: How Do Carbon Microphones Work?

• The carbon microphone is seen as a simple device to turn sound into electronic signal. Some examples of their use where in telephones, radio broadcast systems and the popularity was at a peak around the 80's. These where eventually replaced by more powerful and less noisy microphones. Some are still used today due to their durability and working in low power environments.

  The tapping of microphones came from this microphone as the carbon atoms would get stuck together sometimes and so needed to be hit to separate them to ensure the microphone worked properly, this is still carried on today even though there is no advantage compared to just talking into it.

• With reference to the diagram below. Carbon is a resistor, which isn't very efficient at conducting electricity as it tends to resist, hence being a resistor. A current runs from the diaphragm which is the first plate, through the carbon to the other plate. The carbon molecules normally resist it somewhat, lowering the power flow. When a sound wave pushes down on the top plate, however, it squeezes the carbon molecules more tightly between the two plates. This increases their conductivity, creating more electric current. As the plate moves up and down with the sound wave, the current increases and decreases, creating an electric wave in the shape of the sound wave.
  
  
  

  

  Grey Circles = Carbon Granules - Diaphragm (flexible electrode) moves, changing signal output strength

  If you feel i have missed anything or want to comment feel free.

• • Thanks,

    - Matt Price

Q+A: How Do Dynamic Microphones Work?

With reference to the diagram below a dynamic microphone has a thin diaphragm attached to a coil of wire, which in turn is surrounding a magnet. When sound waves vibrate the diaphragm this vibrates the coil and the difference on the magnet translate to current in the wire and then goes to an external device or speaker. The speed of vibration determines the current and the resulting sound.

Dynamic microphones are generally used for live events such as concerts. This is due to the fact unlike the condenser microphone, it has a low sensitivity and can handle a lot louder environments.

Any questions or feel i have missed anything please comment below,

Thanks

-Matt Price

Q+A: How Do Condenser Microphones Work?

The word condenser actually means capacitor. A capacitor is a common electrical component which stores energy in the form of an electrostatic field. The capacitor allows acoustical energy to be converted to electrical energy.


The electrostatic field needs external or battery power to be created, the image below illustrates that this field is created between the two plates of the diaphragm and the back plate. As the diaphragm is moved from sound waves the change in the electrostatic field information is sent to an external recording device and then amplified as the initial voltage differences are so small.

The resulting audio signal is stronger signal than that from a dynamic. Condensers also tend to be more sensitive and responsive than dynamics, making them well-suited to capturing subtle nuances in a sound. They are not ideal for high-volume work, as their sensitivity makes them prone to distort.

Feel free to comment for more information to be added or made more clear.

Thanks,

Matt Price