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

Production Sound Mixer Video Interviews

Underneath all this typing it's sometimes nice to have a few videos to just watch and listen. Some of the rarest to find in the masses and masses of rubbish on youtube are some interviews with sound mixers (there the guys with boom operators) and sound recordists (they do both)


Desktop Computer Users and Mobile users just use the player below and further down is help to better your experience! If you like the video collection then please "Like" the playlist so more can find it, Thanks Again.

Below is a screenshot for the best user experience on your PC, Mac or mobile.

The Blue Circle Is For Full Screen Then Click The Red Circle For The Full Playlist

Heres another great one from Vimeo,

Heinrich Rudolf Hertz - 155th Birthday

Well would be but sadly died at the age of 36! But didn't take him long to see what everyone else couldn't.

Im sure you have guess the young German electronic engineer and physicist who clarified and expanded the electromagnetic theory of light introduced by James Maxwell publishing his theory in 1865. Maxwell was also phenomenal is creating a theory from seemingly unrelated experiments and studies of electricity, magnetism and optics and combined them. He was also known for the first durable colour photograph earlier in 1861 and even did foundation work on rigidity for bridges, and is considered one of the greatest physicists of all time.

Back to our friend Heinrich Rudolf Hertz. He was the first to prove with immense certainty the presents of electromagnetic waves by designing and engineering instruments to transmit and receive radio pulses using procedures to discredit other sources of wireless phenomenon. As such his name was given to the unit of frequency known as, you guessed it Hertz (Hz)

Just So I Don't Breech Copyright :)

See a better image of him from Wikipedia

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

5 More Pre Production Location Sound Recording Basics (Part 2)

As I am a generious guy I thought I would bash out part 2 of pre production basics (meaning essencials and at the very least)
read part one of 5 pre production basics part one

6. Double system 99% of the time
Unless this is a run and gun documentry or live broadcast location news then you probably have the freedom to easily slate and thus easily sync sound from a seperate sound recording device. The double system gets its name because you have one system for visuals and one for sound (according to my understanding)
This has many advantages such as easier capture for wildtracks, better quality recording due to higher quality specific sound recording electronics and wiring, more flexability to be out of the cameras way and less interference from longer cables, balanced or otherwise.

7. Order of priority
Equipment wise, your only as good as your weakest link and many other metephors a side, the equipment is of this importance as it works in a chain, obviously if anything in this chain is terrible then bring that up to the level of the others.
Your first link is the microphone/s its self, this will determine the amount of electronic current that needs amplifaction, the quality of the frequency response and how much other sound outside the polar pattern will be discrete (blocked).
Next is the mixer of some sort (please not just your camera!) Where it will be able to amplify the microphones electric voltage differences to acoustic energy and the better the pre amps and limiters the better the signal to noise ratio and the less your likley to peak and distort any unwanted, un expexted loud sounds.
Finally the recording device which is last because thankfully you have a good mixer doing most the leg work, dont you :-), this allows the freedome of the double system mentioned earlier.
Also dont forget to get balanced cables (refer to glossary) and the fewer cable to cable connections the lesser the chance of interference and noise being amplified with any recording. along with a good support for your microphone on a boom pole so you limit handling noise.

8. Information is only potencial power
Grab as much information on storyboards and floor plans and sizes of rooms and whats near by the location referring to the script. This could save you loads of time on the day in regards to descision making and mike placement, we both know its never exactly like the plan but better to have a guide!

9. Be ahead of the game
Referring to the edit, before hand you can draw up lists for wildtracks and sound effects which could be benificial to the troops in the edit. On another small note: REMEMBER ROOM TONE/ATMOSPHERE RECORDINGS FOR EVERY LOCATION. It's kind of useful, do at least 15 clean! seconds

Finally Number 10. Fail to plan, plan to fail
Not all these elements might be relevant to the situation but it does save you time in the long run and sets a good solid foundation for the edit. After we are part of a bigger creative team to achieve a great end result and just by reading articles like this you know it's more than a job but a lifestyle especially in independant low budget projects. :-) happy days.

Thanks.and comment or share below

Matt Price

5 Basics For Location Sound Pre Production Part 1

Hello all,
It has been a while so I shall treat you to this blog covering 5 very basic things to cover in PREPRODUCTION for your location shoot.
This is for both sound peeps and production!

1. Location scouting
Always ask questions about sound at different times of day/night in any location and preferably try and go at peak hours of traffic to assess the worse case senario. if your there at 3am on a sunday but your shoot is 5pm on a friday its going to sound very different

2. Carpet and cloth blankets
echo (revereration time) from 'live' (hard, reflective surfaces) needs to be covered or as I like to say killed to create a 'dead' (very low revereration time) room or space even if you like it that way! !
Remember lots of gaffer for air conditioning vents etc... The carpet can be for crew to walk on as they never stay rigidly still.

3. Have an arsnel of microphones
With not just cardiod boom mikes (ntg2) but hypercardiod (senn 416) and even hire in mikes on certain days that are needed. Remember directors and production people: location sound is the best investment in terms of its value of return.

4. Sound report sheets
I may post some examples later but use a sensible sound report sheet for not just your own skin but for the poor editors who can easily see in the edit whats good, whats bad and why, even write when directors over turned any desicions and even leave feedback at the end of sound takes to help other sound editors. Slate everything and try to always push to clap at the front and not the end of a take.

5. Get a boom op...
This will help everyone out from setting up rooms for sound to miking actors and allow the recordist to focus on the quality of the sound! They arn't just a fancy mike stand showing off their muscles and eating all your food but good ones are usually sound recordists as well, two heads are better than one... Or at least 4 hands are better than 2 :-)
Continue to 5 more preproduction basics for location sound.
This concludes the first 5 basics to do in PREPRODUCTION, dont be shy to share this article if you know the production can benefit from these 5, CAUSE THEY CAN :-)
Thanks,
Matt Price

Lossless compared with lossy data compression

When i first looked into this i was like wtf? too... but this is all to do with the data compression.

Lossless data compression is a class of data compression algorithms that allows the exact original data to be reconstructed from the compression data. This is the opposite to lossy data compression, which allows an approximation of the original data to be reconstructed, in exchange fore better compressions rates.

What's the advantages?

One advantage of the lossy methods over lossless methods is that in some cases the lossy method can be compressed smaller than its counterpart, importantly while still meeting the requirements of the application.

Lossy methods are most often used by data are intended for human interpretation where the mind can fill in the blanks or see past minor errors. Ideally lossy compression is transparent or imperceptible, otherwise you would notice it sounded bad and not use it. If you do notice an anomaly it is called a compression artifacts.

Audio can be compressed at 10:1 with imperceptible loss of quality. The compression rate in lossy compression is 5-6%, where as in lossless compression it is about 50-60% of the actual file.

Cheers Guys,

Matt Price

List of Lossy Formats (links to wikipedia)
AAC
ADPCM
ATRAC
Dolby AC-3
MP2
MP3
Musepack (based on Musicam)
Ogg Vorbis (noted for its lack of patent restrictions)
WMA

List of Lossless formats (links to wikipedia)
Free Lossless Audio Codec – FLAC
Apple Lossless – ALAC (Apple Lossless Audio Codec)
apt-X – Lossless
Adaptive Transform Acoustic Coding – ATRAC
Audio Lossless Coding – also known as MPEG-4 ALS
MPEG-4 SLS – also known as HD-AAC
Direct Stream Transfer – DST
Dolby TrueHD
DTS-HD Master Audio
Meridian Lossless Packing – MLP
Monkey's Audio – Monkey's Audio APE
OptimFROG
RealPlayer – RealAudio Lossless
Shorten – SHN
TTA – True Audio Lossless
WavPack – WavPack lossless
WMA Lossless – Windows Media Lossless

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

PCM - Pulse Code Modulation

PCM or pulse-code modulation is a method to digitally represent sampled analog signals. This is the standard system for digital audio in computers and various DVD, Blu-ray formats and in digital telephone systems.


Each sample is quantized to the nearest value within a range of digital steps. The two properties of PCM are the sample rate and the bit depth, this determines the number of digital values in the range.


See article on Bit depth and Sample Rate for more info.


Cheers,


Matt Price

What would you like to see?

I have been working a lot lately and being based all over the UK means I dont really have proper time or a desktop computer to neatly research and explain topics... So this is just a quick post along for any questions our anything you would like to see in this blog, obviously excluding current content. Anyone can comment on this post without signing in so get cracking :-)

Thanks,

Matt Price

Phase and Interference...

What is wave interference?
when two or more waves from different sources are present at the same time in the same space to create a new wave. The more useful part of this question is what happens and what types there are...
Constructive Interference
when the compressions and the rarefactions match they create a wave of higher intensity.
Destructive Interference
When the waves are out of phase, as in the opposite to constructive interference but the sound is louder in some places and softer in other places. Often leading to pulses and beats of sound.
What is a Phase?
Phase in waves is the fraction of a wave cycle which has elapsed relative to an arbitrary point.
What is Phasing?
The relationship between the waves to form the new wave, often used to describe the resulting sound, commonly constructive or destructive.
Dead Spots?
This is where the compression of one wave matches or is in phase with the rarefaction of another and so the sound is cancelled out and nothing is heard.
What else to consider?
Sound waves also change speed due to rarefaction from entering different mediums such as air to water, thus changing angle and bending the wave. this leads onto:
The critical angle..
When a waves entering angle reaches a certain point it's called the critical angle. The rarefaction is parallel to the dividing line between the mediums. The greater the difference in speed from the sound in the two mediums, the greater the critical angle.
What does this mean?
If the sound hits the new medium with any angle smaller then the critical angle it will not be able to enter and so be reflected from the dividing line. Even if it enters the medium some will be reflected because of waves rarely propagate in straight lines only.
Example...
A wave travelling through the air hits a building at 20 degrees, which for this example is less than the critical angle, then it is all reflected at 160 degrees as the total angle is 180 degrees of the wall.
If the wave is above the critical angle then most will enter the brick and speed up because the molecules are closer and so is refracted. The rest bouncing off because it won't all be entering with in the critical angle. 

Hope this all flowed nicely,
Thanks,
Matt Price

Quick Note About Digital Microphones...

As we know, or should by now, analogue microphones are based on the same concept as our ears, this is the most effective way to actually hear because sound waves have no formula when it comes to multiple sources and are all added together depending on so many different circumstances that it is impossible to hear digitally, unless we get a chip in our brains to convert it some how it is a long long way off....

So what are digital microphones?

These beauties like the Shoeps SuperCMIT are really the future. They still are not completely digital for reasons above but have a digital transducer instead of going all the way down analogue cabling to be converted. This allows the signal effectively at source to be effectively set up so then your mixer/recorder can do even more with it in terms of collecting that sweet sweet sound.

So keep a look out for them because they will slowly become the norm, after all your not recording onto DAT tape i bet.

Thanks,

Matt

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

Transducer

A transducer is a device that converts one type of energy to another. Energy types include electrical, mechanical, electromagnetic (including light), chemical, acoustic or thermal energy and many more. While the term transducer commonly implies the use of a sensor/detector, any device which converts energy can be considered a transducer.


So a microphone is considered a transducer because it turns acoustic energy into electrical energy.


Thanks,

Matt Price

Ohm ohm ohm... amp?

Ohm's or Ω is the SI or system of international unit of electrical resistance and gets it's name from Georg Simon Ohm. Ohm determined that there is a direct proportionality between the potential difference (voltage) applied across a conductor and the resultant electric current. This relationship is now known as Ohm's law.

So an Ohm is a unit of resistance between two points of a conductor when, for instance, a constant 1 volt difference is applied to each point and it produces 1 ampere.

Ampere  It is named after André-Marie Ampère (1775–1836), French mathematician and physicist, considered the father of electrodynamics. In practice, its name is often shortened to amp. An amp is a SI for basically a rate of flow in a wire. 

So an ohm is just a way of measuring how easily electricity flows along a certain path. Electrical resistance shares some conceptual parallels with the mechanical notion of friction.

Leave comments if still confused =]

Thanks,

Matt Price

The famous Wilhelm scream

The Wilhelm Scream originates from the film Distant Drums in 1951. The scene in the video below soliders are wading through a swamp and one (Wilhelm) gets bitted and dragged underwater by an aligator, the scream was re-recorded in a single take, along with 5 others. the 4th 5th and 6th are also used earlier in the movie. 4-6 are all recognisable as the Wilhelm scream by many sound editors.

Below is the scene with the original Wilhelm Scream.

Below is a compalation of Wilhelm screams

Hope you enjoyed this,
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