Sound originates as an analogue signal this means that the sound is a continuous wave that varies in strength and quality when a sound is recorded into a sound recorder, the sound becomes a digital signal which is a direct pulse of electricity, the signal can either be on or off, this on or off transmits data in the form of binary data which means the data is 0 or 1 for example the 0 could be off and the 1 could be on this forms the digital sound. When recording sound, the sound starts as a vibration in the air and travels to the microphone, once in the microphone it converts it into a varying electrical current which travels down the microphone lead, when doing this it will pick up some noise from the cable this amount of noise can be reduced by using a balanced cable. Then this electrical signal goes into the input then a pre-amp where its voltage is boosted as the amount of electricity created by the sound vibrations is extremely small and wouldn’t work well without being boosted, the signal then goes to an analogue to digital converter where the varying analogue signal is converted into binary form (0 or1) which is a direct pulse of electricity rather than a continuous wave. This converted digital sound then goes to the medium where the converted file is stored. A medium is somewhere you store the files such as an SD card or Flash memory.
When replaying a sound, the same happens but in reverse so the sound starts at the storage medium this file then goes to the digital to analogue converter (DAC) where the digital sound is converted back to an analogue sound ready to be played out of a speaker or something similar this converts it back to a continuous wave rather than a direct pulse. Similar to when recording sound, the voltage is very low when outputting sound so the sound has to go through the pre-amp again and have the voltage of the sound boosted this makes it louder and will be a better volume to listen back to rather than the tiny voltage without the pre-amp which would make it almost impossible to listen back to it as it would be so quiet. This process is repeated in the power amp to give it the right voltage for output the sound then travels along a cable to a speaker whilst doing this the signal will pick up some noise from the cable this amount of noise can be significantly reduced by using balanced cables. When the electrical signals reach the speaker they cause the speaker to vibrate and will give out the sound that was recorded.
A codec is an encoder and a decoder put together, a codec is used to encode digital audio files this can compress the file and may be lossless compression or lossy compression and is also used to decode it so that the file can be played back. A file format is the container that the audio is in. A file format can be used to separate out different types of content such as audio, metadata and video.
A file format is also called a container which separates out different types of content such as Audio and video and metadata, this will have a file extension such as .mp3. When listening to an mp3 it will have an audio stream but will also have other parts in the file this is the metadata which is the information for the file so when you’re playing back the file it will say the song name and who it is by etc. Sometimes the format and the codec are the same for example .wma (windows media audio files). You also get container formats that can be used with a lot of different codecs like .ogg (OGG Vorbis). Sometimes a file format has a codec that has the same name but the codec can still be used with other file formats.
Pulse code modulation is the basic method for digitally representing sound. All uncompressed sound is basically a pulse code modulation stream. The sample rate is the amount of times per second that a sample is taken, whilst the bit depth is the number of possible digital values that can be used to represent each value.
Lossless compressed formats include
• WMA – windows media audio which is a codec and a format
• FLAC which is also a codec and a format
• ALAC – apple lossless audio codec often used with the .wma format.
What a lossless compressed format does is take your pulse code modulation and turns this into something smaller.
Lossy compressed formats such as
• MP3 (codec and format)
• AAC (Codec and occasionally format usually used with the .m4a format)
• Vorbis (codec generally used with the .ogg format)
Decibels (dB) is used for many different purposed not just audio, they measure the ratio of two quantities to compare their relative strength. Decibels use a logarithmic scale for example a sound of 20 decibels has 10 times more powerful than a sound that was 10 decibels and if you had a sound that was 30 decibels this is 100 times more powerful. When talking about acoustics decibels measure sound pressure relative to 0db whilst in sound recording it measures the input level relative to the maximum level that the recorder can take before it distorts the audio.
When recording it is important to keep an eye on recording levels as it determines how good your recording sounds. 0 Db is the very maximum level you can record on a recording device if your sound goes above this noise you will notice that your sound has been clipped and distorted. A common compromise for this clipping is to aim for -3 decibels and peaking at 0 decibels. -6b is half the ideal level whilst -10db is half the ideal level making -3db the common compromise.
Stereo sound is where you can hear the sound in both ears to do this you have to record two channels and use a coincident crossed pair this gives you true stereo audio. Mono audio is where there is only one channel of sound and thus the sound will only be played from one side.
Replay levels are relative to the power of the amplifier, the taste and the environment. When talking about replay levels db refers to
• Volume
• Gain (anything above 0db will amplify the signal)
• Sound pressure
There are many different types of microphones these are:
• Dynamic Microphones
• Condenser (Capacitor) Microphones
• Ribbon Microphones
• Piezo Microphones
• Parabolic Microphones
• Radio Microphones
Dynamic microphones function like this, a sound is produced and travels through the air, the sound is vibrations in the air and thus when it reaches the diaphragm in the dynamic microphone it causes this diaphragm to vibrate. When the diaphragm in the microphone vibrates it moves a coil which is inside a small magnetic field, this in turn generates an electrical current. Dynamic microphones such as the Shure SM57 and the SM58 are good at picking up low frequencies and are very robust. Dynamic microphones don’t need a power source, and tend to be larger microphones due to having the coil inside.
A condenser (capacitor) microphone such as the Behringer B2 Pro functions like this, there is two plates which have an electrical charge and when they vibrate it changes the voltage that can be stored between the two plates similar to a capacitor, this creates waves in electricity which is how the sound is produced from the microphone. This type of microphone needs an electrical input to make it work. They are more sensitive to higher frequencies whereas dynamic is not however they are much more fragile and thus would not be suitable for using it in a concert environment as it may get dropped and it would break instead you would be more likely to use a dynamic microphone such as the Shure SM58 as they can be dropped and have very little if any impact on performance. There are different types of condenser microphones there is DC which is a standard microphone, there is RF which stands for radio frequency these are rare and have a good wider frequency response, and a good tolerance towards damp so this would be a better choice of condenser microphone if it had to be used in an outdoor environment. Finally, there is electret microphones such as the Sennheiser M2 where a power supply to it is not essential and this can be a much smaller so it is typically found in cheap headsets however they are much worse quality than other condenser microphones. Other microphone types include Ribbon microphones which are similar to condenser microphones as they are extremely fragile, very sensitive and very expensive such as the Rhode NTR Ribbon Microphone. There are shotgun microphones which have the microphone placed at the bottom of a tube which allows the sound from the side to be cut out in the case of this the longer the shotgun tube the more sound from the side that it will cut out. Wave interference is either constructive or destructive. If you had 2 waves that were the same it would increase the volume of the sound but would be the same sound if the wave didn’t match it would be a destructive wave and would not be the original sound.
A microphone with a Unidirectional cardioid pattern will capture sound from one direction this type of pattern allows the sound to be picked up mostly from right in front of the microphone with very little sound being able to be recorded from any other direction, Whereas an omnidirectional microphone pattern allows sound to be captured from all directions which could be useful if you are wanting 360 degree sound or to record background noise in order to capture the most detail it wouldn’t however be useful for a concert environment as the microphone would pick up the noise from the crowd and the monitors as well as the sound you intended to capture. Bi-directional microphone patterns allow the sound to be captured from both sides