Showing posts with label Audio Video Engineering. Show all posts
Showing posts with label Audio Video Engineering. Show all posts

Monday, February 11, 2013

Advanced Digital Audio Codecs


DTS standards
DTS (Digital Theater Sound) is a digital sound coding standard created by Universal. Compared with the Dolby Digital standard, DTS uses four times less compression and digitises sound at 20 bits instead of 16. Therefore, DTS's sound quality is theoretically higher, at the cost of a higher bit rate. To be able to play DTS-encoded media, you need a certified DTS decoder.

DTS falls into four different categories:

DTS 6, the most commonly used 5.1 standard, which can encode six-channel sound with less compression than the Dolby Digital standard. The first five channels are used for the satellite speakers, while the last is reserved for the subwoofer. These devices are normally identified by the presence of this logo:
DTS ES (Digital Theater Sound Extended Surround), 6.1 standard which uses an additional rear channel (rear central). DTS ES uses less compression than Dolby Digital EX.
The DTS ES standard has two variants:
o DTS ES Matrix, which has a seventh channel interpolated with the primary channels. This is called "virtualisation".
o DTS ES Discrete has an seventh independent channel.

DTS 24/96 represents an audio format used for storing high-definition music with several channels. This format is primarily used in DVD Audio, or audio tracks which accompany video DVDs. The name comes from the fact that the tracks are recorded in 24 bits at 96 kHz. It may be in either stereo or 5.1.
DTS Neo:6 is a format for upmixing (virtualising) from a stereo sound source.
Image Courtesy: http://electronics.howstuffworks.com http://www.logotypes101.com

Dolby Digital
Dolby Digital and DTS are six-channel digital surround sound systems and are currently the standard in major motion pictures, music, and digital television.

They both use the 5.1 speaker format The format consists of three speakers across the front and two speakers in the rear. The .1 is a sixth channel called an LFE that is sent to a subwoofer.

Dolby Digital uses the AC-3 file format, which any Dolby Digital Decoder can decoder to produce 5.1 audio. Dolby Digitalis the technical name for Dolby's multi-channel digital sound coding technique, more commonly referred to as Dolby 5.1.

A six-channel sound coding process (one channel each for front, left, center, right surround, left surround and a sub-woofer) originally created by Dolby for theaters, AC-3 was subsequently adapted for home use and is now steadily becoming the most common sound format for DVD.



The difference between Dolby Digital (AC-3) and DTS is:
Both systems are great but statistics for reference only..

• DTS seems to provide a deeper and tighter low frequency presence
• DTS allows the sound to breath - transparency
• AC-3 seems to leave the impression that something is missing from the mix.
• At lower bit-rates AC-3 starts to sound like MP3's encoded at 96kbps (artifacts)

Sunday, February 10, 2013

An Introduction to Digital Audio

What is sound? 
Sound is vibrations in the air; that is, a series of rising and falling pressures in the air, deviating from the average, which is represented by atmospheric pressure. The simplest way to create a sound is to make an object vibrate.

In this manner, a violin makes a sound when the bow makes its strikes vibrate, and a piano sounds a note when a key is struck, because a hammer struck a string and made it vibrate.

Speakers are generally used to reproduce these sounds. They are a membrane connected to an electromagnet; as an electrical current travels in front of and behind the magnet very rapidly, it causes vibrations in the air in front of it, and that vibration is sound! This is how sound waves are produced; they can be represented in a diagram as changes in air pressure (or in the electricity level of the magnet) as a function of time.

A sonogram, on the other hand, depicts sound frequencies as a function of time. It should be noted that a sonogram shows fundamental frequency, on top of which higher frequencies, called harmonics, are superimposed. This is what allows us to distinguish between different sources of sound: low notes have low frequencies, while high notes have higher frequencies.
 
Sound as an input and output to the Computer
Sound sampling To play sound on a computer, it must be converted into a digital format, as this is the only kind of information computers can work with. 
 
 
 
A computer program intersperses small samples of the sound (which amount to differences in pressure) at specific intervals of time. This is called sampling or digitising sound. 
 
The period of time between two samples is called the sampling rate. As reproducing audio which sounds continuous to the ear requires samples at least once every few 100,000ths of a second, it is more practical to go by the number of samples per second, expressed in Hertz (Hz). 
 
Here are a few examples of common sampling rates, and what sound quality they correspond to:
Sampling rate -Sound quality 
44,100 Hz -CD quality 
22,000 Hz -Radio quality 
8,000 Hz -Telephone quality 
 
The sampling rate of an audio CD, for example, is not arbitrary. In fact, it follows from Shannon's theorem. 
 
Sampling frequency must be high enough to preserve the form of the signal. 
 
The Nyquist-Shannon theorem stipulates that the sampling rate must be equal to or greater than twice the maximum frequency contained in the signal. 
 
 
Our ears can hear sounds up to about 20,000 Hz. Therefore, for a satisfactory level of sound quality, the sampling rate must be at least on the order of 40,000 Hz. 
 
There are several standardized sampling rates in use:
• 32 kHz: for digital FM radio (band-limited to 15 kHz)
• 44.1 kHz: for professional audio and compact discs
• 48 kHz: for professional digital multitrack recording, and consumer recording equipment (like DAT or MiniDisc)

A computer works with bits, so the number of possible values that the sample could have must be determined. This is done by setting the number of bits on which the sample values are encoded.

• With 8-bit coding, there are 28 (= 256) possible values.
• With 16-bit coding, there are 216 (= 65536) possible values. 
 
The second option clearly offers higher sound fidelity, but at the cost of using more computer memory. 
 
Finally, stereo sound requires two channels, with sound recorded individually on each one. One channel is fed into the left speaker, while the other is broadcast from the right speaker. 
 
In computer processing, a sound is therefore represented by several parameters:
• The sampling rate
• The number of bits in a sample
• The number of channels (one for mono, two for stereo, and four for quadrophonic sound
 
 
Memory required for storing a sound file 
 
It is easy to calculate what size an uncompressed audio sequence will be. By knowing how many bits are used to code the sample, you know its size (as the sample size is the number of bits) 
 
To find out the size of a channel, all you need to know is the sample rate, and thus the number of samples per second, and from that the amount of space taken up by one second of music.
 

Courtesy: http://en.kioskea.net  
Image Courtesy :http://freesoftwaremagazine.com

Friday, April 27, 2012

How Television screen works?

The Basic idea of "video" is a special property of the human eyes, the Persistence of vision.
Persistence of vision is the phenomenon of the eye by which an afterimage is thought to persist for approximately one twenty-fifth of a second on the retina.

Hence, the Cinema we are watching in the Theater is created by moving 24 still images successively before a projector in one second.

However, they found that it is creating some flickers in the video (ie, jump from one still image to another).
They resolved the issue by duplicating the images; ie, 48 images per second. Thus we have a smooth video motion.



(Image: This animated cartoon of a galloping horse is displayed at 12 drawings per second )















Thus, we have to create 48 images per second on the TV screen.
However, in PAL system, we are creating 50 frames per second. This is to make the system compatible with the electric power system india. (230V/50Hz)
This is done by moving an electron beam very fast over the TV screen.
When the beam hits different phosphor spots on the TV screen, it is illuminated.
The phosphor spots are arranged in the form of a matrix.
There are 625 phosphor spot lines arranged horizontally on the PAL Television System(India).
Thus, the electron beam should move from the first line to the 625th line within 1/48th of a second.

In Black and White TV, the TV signal contains only the brightness information at a particular spot.
However, in the colour TV, the TV signal contains both the colour and brightness information.

The TV screen has the RGB (Red, Green, Blue) spots arranged horizontally.
The combination of the three colours at various intensities can reproduce any colour on the TV screen.

(Image: How 3 Colour Electron Gun  Works)





















In order to create 50 frame per second, the electron beam should move in a very high speed and it will make the circuitry very complex.
Hence we can use interlaced scanning method, ie scanning alternate lines in one frame.
Successive scan TVs are also available in the market, which will provide a better image quality.

(Image: Interlaced Scanning in the TV)






























Thus the TV signal received in the TV contains a luminescence signal and a chrominescence signal.

Different Television broadcasting system differ on the basis of the number of lines, number of frames per second etc.
The TV System in US is NTSC and it is SECAM in Europe.

The modern High defenition LCD TVs differ slightly from the above explained operation.
It usually contains 1080 lines, successive scanning and 16:9 Aspect ratio.

NTSC
the NTSC system was developed as the first color TV standard. This basic color system works well, and after 40 years is still in use in North America and Japan. NTSC stands for: National Television System Committee, which was the organization that defined this color TV standard. NTSC is based on 525 picture lines and 60 Hz (60 scans across the screen in one second). The major problem with NTSC is that hue errors might occur. In order to correct this, all NTSC receivers are equipped with a special hue control.

PAL
PAL stands for: Phase Alternating Line and was introduced 15 years after NTSC. The system is based on 625 lines and 50 Hz. It has more picture detail than NTSC, because it uses 100 more lines and is able to write colors with greater accuracy. If there is a color distortion on one line, it will correct this on the next line by reversing the error. If, for instance, the intended green shifts to a yellow, on the next line it will correct this with a reverse error. The result is that the next line will be cyan (blue-green), which results in an optical green; the yellow and cyan will blend to green.
The major disadvantage of the PAL system compared with NTSC, is that it makes use of 50 scans per second rather than 60. This results in flicker and a more unstable picture. However, the problem of flicker is solved when the raster frequency is adjusted to 100 Hz (100 scans per second).

SECAM
SECAM or Sequence Couleur à Mémoire, (which translates as color sequence in memory) was developed around the same time as PAL. Whereas with both NTSC and PAL color errors still occur, with SECAM there are no color errors
The systems are not compatible because of the difference in the number of lines and the raster frequency, and the way they handle color reproduction. For those who want to watch movies recorded in a different standard, TVs and VCRs have been developed which are able to handle two or three systems. Conversion of TV, satellite or cassette programs to other systems results in lower quality.

Note: TVs, DVDs, VCRs etc purchased in different countries may not work in another country due to the various reasons discussed above.