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 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

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

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

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

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

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: Bit Depth and Bit Rate Explained....

This is a quick guide to bit depth, sample rate and recording volume.


Bit Depth
Bit depth describes the number of bits of information recorded for each sample. This directly corresponds to the resolution of each sample.


a "bit' is an abbreviation for a single binary digit, represented by a 0 or 1. A word in audio is a binary number with more than one digit. A 16-bit binary number for example could be 0111011010110010. The number of bits per word is simply how many digits there are in the corresponding number. Common bit depths are 8, 16 and 24 so the higher the quality is the higher the number. 


The bit depth in recording refers to how many steps the amplitude can be broken down into to be represented digitally. 
16 Bit = 65,536 Steps 
24 Bit = 16,777,216 Steps
You can clearly see that 24 Bit would be considered higher quality as it is able to plot smoother more accurate lines of amplitude.


The Bit Rate is how many of these bits are being transmitted or received per second. This is used for streaming purposes on videos as well. For playback, using the example of a standard CD with a datarate of 48kHz/24, meaning the sample rate was 48,000 times per second with bit depth of 24. The amount of audio data per second is worked out like this: 


48000 x 24 x 2 (for stereo recordings) = 2,304,000 bit/s or 2.3 Mbit/s.


The size of an audio file size can be calculated with a similar equation.The equation below is with the example of recording at 48khz at 24 bit depth, that plays out in stereo (so 2 channels, recording 5 you would x5) for 60 minutes which is 3,600 seconds and divide by 8, you then get this:


48000 x 24 x 2 x 3600 / 8 = 1036,800,000 Bytes or 1.036GB


So bit depth is your resolution for the information for each sample but many would argue there is no massive difference for recording dialogue between 16bit or 24bit but if you have the space why not.


See also: Sample Rate Explained


Thanks,


- Matt Price