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

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

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

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

Refraction

This is the bending of waves when they enter a medium where their speed is different. This is less important than refraction of light as it affects image formation of images by lenses and eyes etc... 

If we use the example illustrated above shows that the first medium of air the light travels straight but when it hits the second medium of the glass it bends depending on which side hits the second medium first, it will bend left if the left side hits the second medium and slows down and refracts. 


Not only the direction changes but separation of the waves decreases as the frequency of the waves does not change by its source but the shower speed must shorten the wavelength.


This is also interesting in sound because if the air above the earth is warmer than the surface, sound will bend back downwards towards the surface by refraction. If the air above the earth is warmer than that at the surface, sound will be bent back downward toward the surface by refraction.


Refraction also amplifies sound sometimes over cool lakes in the morning. The water keeps the air cool near the water but as the sun comes up it heats the air higher up creating a thermal inversion. The speed of sound is faster in the warmer air and so bends some sound back towards you.


also see: Diffraction


Thanks,


Matt Price

Diffraction

One of the interesting properties of sound waves is that they can diffract like the image below demonstrates.

The possibility of hearing around corners or barriers involves diffraction and reflection of sound. Diffraction is mostly associated with longer wave lengths as thus implies that you hear low frequencies than higher ones. This is also explained by the fact air absorbs higher frequencies better than lower ones. A lightning strike for instance has a high crackle with a longer low rumble.

Sound proofing rooms takes this into account and so the room has to be fully sealed because diffraction can disrupt a lot from the smallest gaps. This is a similar reason for loud speakers to be sealed so they maximise output. 

An interesting characteristic involving imaging. When a wavelength is longer than an obstacle, for instance a pillar means you can't see the obstacle. This is the same reason you cant see a virus under a light microscope because the virus is smaller than the waves of light. This is due to the bigger wavelength diffracting round the obstacle and reconstructing past it. 

See also: Refraction

Feel free to comment,

Thanks,

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

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

Sound Pressure Level and Decibels

Sound Pressure Level is the amount of air pressure fluctuation a sound source creates. We perceive this pressure in terms of loudness. A simple example is a drum, when you hit it gently the surface doesn't move very far and so because the of this the pressure is lower than hitting it hard.
The pressure is also effected by where the listener is from the source and the environment they are in. A drum hit hard in a small bathroom will sound a lot louder than a drum in a cathedral because many of the discrete reflections will never reach the listener due to the expanse of the space. This is obviously even less in an open field where there is nothing to reflect off in the sky.


Sound pressure is usually expressed as pascals (Pa). Usual conversation pressure is 0.02 Pa and a petrol lawn mower is around 1 Pa and sound becomes painful at around 20 Pa. So a healthy range is around 0.00002 to 20 Pa. 


Pressure ranges obviously offer such a wide range of scale that we use the decibel scale. This is because it compresses the scale into a manageable range. This conversion from sound pressure to the decibel scale is called sound pressure level.


0dB is 0.00002 Pa and this is the start of the scale.

This is redrawn on paper due to copyright and is still an accurate reference.

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

Meta Data And Sound Library Organisation

Hello All,


This blog is on Meta data and keeping your tracks and libraries in order. There is nothing easier than being able to search your library and have a list of results that are exactly what you want.


Meta Data is basically data on data, so with meta data on your sound files you can say which project and which take it was and this can easily be used by editors and yourself to facilitate an efficient post production process. 


Many find their own way of organising their files and the easiest way is getting a client like Wave Agent from Sound Devices. Below is a screen shot of the Mac Version.

I recommend this application even if you dont have a sound devices recorder and i should point out this does a lot more if you do have a sound devices recorder like the 788t, Visit the sound devices site for more details.


Enjoy and thanks,


- Matt Price