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"



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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

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

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

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

The camera that changed the world

Film Link: Camera That Changed the World
Only available for the next 5 days.


This is a quick post about a documentary availible on the BBC iPlayer that shows the evolution of the portable camera and how important it was to the documentary genre. At around the 45 minuite mark there is a short section on the importance of the Nagra and sync sound. So if you can spare an hour over the next 5 days it's worth a watch.


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

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

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

Q+A: AA Batteries...

AA batteries and mAh ratings
mAh stands for Milliamp an hour. This is commonly used to describe the total energy a battery can store over time. the higher mAh rating means when fully charged, the battery can power a device that consumes more power and/or for a longer amount of time before depleting and needing a recharge.

For example, a battery rated at 1500 mAh can power a device drawing 100 milliamps for 15 hours, or a device drawing 150 milliamps for 10 hours.


Thanks,


- Matt Price

Q+A: What is the difference between balanced and unbalanced cables?

Cables are essencial for most of the work in sound recording because they are very reliable and durable compared to wireless systems. The most common cable is the XLR cable for connecting microphones to external devices and recorders, however what is the difference between an often more expensive balanced XLR cable compared to an unbalanced one?

The easiest way to understand the difference is the signal separation that takes place inside the cable. Common cables like plugs for a kettle have 3 separate cables inside one for ground, one for positive and one for negative. A balanced cable would use all 3 and since each one is doing equal work is balanced. However unbalanced have or use 2 of these wires which means the negative wire is also doing the work of the ground wire.

Balanced cables mean they are less susceptible to extraneous noise. Their XLR connectors often have a locking mechanism that makes the connections more reliable and more difficult to accidentally unplug. Unbalanced phone, mini and RCA connectors are not hearty and can be easily damaged and dislodged from their inputs. Balanced cables often also include a tight metal mesh wrapper that acts as an electronic shield to protect them from interference. Electricity can generate hum through a broken cable or a damaged connector. It can even arise simply by laying a power cable parallel to an audio cable. The upshot to all this is that hum in a cable can ruin an audio recording

Unbalanced cables and high impedance accessories are fine when work with very short cable runs. Also, an inexpensive microphone and unbalanced cable is almost always better than just using the on-camera microphone. In a professional situation, you want to isolate the sound source, and eliminate any potential for hum. Balanced cables eliminate hum. Using an external microphone of low impedance is best in an interview situation or when it is not possible to get close to your subject.

Feel free to ask any questions and leave comments,

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