Digital Audio

DEEPAK JOSHI DY.
DIRECTOR STI(T)
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Digital Audio
Overview The principles which underlie almost all digital audio applications and devices, be they digital signal processing , digital recording , CD,HDBRS iPod playback ,Internet broadcasting ,Digital Broadcasting are all based on the basic concepts which follow here in. New forms of playback, file formats, compression and storage of data are all changing on a seemingly daily basis, but the underlying mechanisms for converting real-world sound into digital values, manipulating those data and finally converting them back into real-world sound has not varied much . Analog to digital (A/D) conversion Sampling Aliasing Nyquist criteria Sample rates Quantization Bit Rates Dither Encoding PCM or any other algorithm based..like MP3,AAC etc
DEEPAK JOSHI DY.DIRECTOR STI(T)
A to D / D to A conversion Digital recording converts the analog wave into a stream of numbers and records the numbers instead of the wave. The conversion is done by a device called an analog-todigital converter (ADC). To play back the music, the stream of numbers is converted back to an analog wave by a digital-toanalog converter (DAC). The analog wave produced by the DAC is amplified and fed to the speakers to produce the sound.

When you sample the wave with an analog-to-digital converter, you have control over two variables: The sampling frequency - Controls how many samples are taken per second . It tells sample rate which defines frequency resolution The sampling precision - Controls how many different gradations (quantization levels) are possible or required when taking the sample. Which helps in defining Bit depth or Bit resolution. Bit depth defines amplitude resolution. Higher Bit depth means greater dynamic range, lower noise floor and high fidelity.
Sampling
To convert an analog signal to a digital format, the voltage is sampled at regular intervals, thousands of times per second. The value of each sample is rounded to the nearest integer on a scale that varies according to the resolution of the signal. The integers are then converted to binary numbers. The sampling frequency is how many times per second the voltage of the analog signal is measured or The sampling rate is the number of times your audio is measured (sampled) per second.

In the adjoining figure, let's assume that the sampling rate is 1,000 per second and the precision is 10. The green rectangles represent samples. Every one-thousandth of a second. . When the DAC recreates the wave from these numbers, you get the blue line shown You can see that the blue line lost quite a bit of the detail originally found in the red line, and that means the fidelity of the reproduced wave is not very good. This is the sampling error.

Now In these figure, both the rate and the precision have been improved by a factor of 2 (20 gradations at a rate of 2,000 samples per second): Further the rate and the precision have been doubled again (40 gradations at 4,000 samples per second): You can see that as the rate and precision increase, the fidelity (the similarity between the original wave and the DAC's output) improves. Higher the sampling frequency the closer the shape of Digital wave form to that of original analogue signal


What should be the Ideal sampling frequency
No straight reply how ever Nyquist criteria comes to HELP

Nyquist criteria :
where: fs = 1/Ts is the sampling frequency & V(t) = value of signal at arbitrary time t. V( nTs) at discrete times tn = nTs where n = ... -1, 0 , 1 , 2 , 3 ... The Nyquist theorem states that if a signal V(t) does not contain frequencies higher than fs/2 (where fs = 1/Ts), then it can be fully recovered from its sampled values
The sampling rate must be at least twice as high as the highest frequency to be reproduced. In other words to reproduce a given frequency the sampling rate must be at least twice that frequency. Nyquist theory suggests that the actual upper threshold of a piece of digital audio will top out at half the sample rate. So if you are recording at 44.1, the highest frequencies generated will be around 22kHz. That is 2khz higher than the typical human with excellent hearing can hear. So at the red book standard for CDs, the sample rate is 44.1 kHz or 44,100 slices every second. Hence CD quality audio is sampled at 44.1 kHz (Nyquist frequency = 22.05 kHz)

Current standards of sampling full-quality (i.e. not compressed) digital audio rates are: Rate 32K 44.1K 48K 96K 192K Use(s) Older DATs, voice quality CD, DAT, digital recording software/hardware DAT, digital recording software/hardware Digital recording software/hardware Digital recording software/hardware
Why did the CD standard settle on 44.1K rather than say 48K? Rumor has it that video equipment already had clocks that ran at 44.1K that could be integrated into the first CD players. Its also heard that Herbert von Karajan complained to Sony that Beethoven's 9th would not fit on the early CD specifications. By lowering the rate to 44.1K, 74 minutes could be recorded onto a CD using 16-bit samples, enough to do the trick.
Aliasing
If the audio signal contains components which are above half the sample frequency aliasing will occur. First consider the example below which shows the frequency bands created when a 20 kHz audio band is sampled at 48 kHz.
The sampling process produces a signal at the sampling frequency and a band of frequencies which spread above and below this by an amount equal to the audio bandwidth. A further set of frequencies are created at double the sample frequency and at three times and so on
If the sample frequency is reduced to 32 kHz but the audio band remains at 20 kHz, the 20 kHz bands either side of the sample frequency will overlap with the audio band. Aliasing will occur which has the effect of generating additional side bands which can be extremely audible. Inter-modulation can also produce further frequencies and the sound can become highly distorted.

Solution to Aliasing problem:

There are two solutions to the issue of aliasing from frequencies above the Nyquist frequency in reproducing high-quality audio. The first is to band-limit the frequencies allowed to enter the system. The second is to have a higher sampling rate, thereby increasing the frequency range before aliasing occurs. In reality, both of these have been addressed by current digital audio systems.

Anti Aliasing Filter

To prevent aliasing the audio band must pass through a filter with a very rapid roll off above 20 kHz. These are called aliasing filters .Such filters can introduce irregularities to the frequency response below 20 kHz and many designs can also create undesirable phase shifts.

Anti Aliasing Filter Contd

Anti-aliasing filters are placed in front of any analog-to-digital converters before sampling to eliminate frequencies above the Nyquist frequency from ever being sampled. So far, though steep, existing technology has not afforded us a true 'brick wall' filter that suddenly eliminates unwanted frequencies without having any effect on those in the 'legal' range.
The diagram above indicates that even at a sampling rate of 44.1K (the CD rate), some audible frequencies are attenuated by the filter. Attenuation is determined by the quality of the filter and it steepness factor (or 'Q'). So note that The higher the sampling rate used, the less noticeable impact of the filter rolloff on audible frequencies become as more and more of the rolloff is above audio rate.
Over-sampling
Using a sample frequency which is double the one finally required moves the sides bands further away from the audio band and allows simpler filters to be used. Commercial designs can use sampling which is many times the required rate.

Quantization
When a wave form is digitised the amplitude is converted to a numeric value. Samples taken are assigned numeric values that the computer or
digital circuit can use or store in a process called quantization. Quantization is the process of selecting whole numbers to represent the voltage level of each sample. The A/D converter must select a whole number that is closest to the signal level at the instant its sampled. This produces small rounding errors that cause distortion. The number of available values is determined by the number of bits (0's and 1's) used for each sample, also called bit depth or bit resolution

Bit Depth refers to the number of bits you have to use to capture audio. The easiest way to envision this is as a series of levels, that audio energy can be sliced at any given moment in time. It is also called Bit Resolution . The smaller the number of bits used per sample, the greater the distances the analog values need to be rounded off to.

What should be the Ideal
Bit Depth.

Dynamic range of an audio

Frequently used terms in audio:
1) Head room: This is the audio level difference between nominal value of audio and maximum permissible value of audio. For example Headroom of a analog console is 24 db for a analog tape recorder 8 12 db, for FM radio and Vinyl disc it is 6 db. 2) Noise floor: The reason below the maximum value of noise level is known as Noise floor. 3) S/N: The difference of level of nominal audio level and noise floor is known as S/N. Audio source S/N db Vinyl disc 50 CD player 80 Cassette player 50 AM Radio 30 FM Radio 50


As per the pioneering work done by (Nakajima, 1983; Mieszkowski, 1987 )The maximum Signal-to-Noise Ratio (S/N) in decibels and The maximum representable signal amplitude to the maximum quantization error for of an ideal ADC or DSP-based digital system is actually calculated as:
1.76 dB is based on sinusoidal waveform statistics and would vary for other waveforms, while n represents the data word length in bits of the converter or the digital signal processor . In undithered DSP-based systems, the SNR definition above is not directly applicable since there is no noise present when there is no signal . Therefore, when referring to SNR or Dynamic range in terms of DSP data word size and quantization errors"both terms mean the same thing. So Note that the "6-dB-Per-Bit-Rule" is an approximation to calculating the actual dynamic range for a given word width.
As a rule of thumb, each bit of precision used in quantization adds 6 dB to the SQNR and dynamic range. Thus, converter resolutions of 8, 12,16, and 20 bits would allow a 48, 72, 96, and 120 dB S/N ratio or dynamic range respectively Sound sampled with 16-bit precision ("CD-quality") has a SQNR of 96 dB, which is quite good--much better than traditional tape recording. Why does bit depth matter ? It's a matter of the human ear, which is only accustomed to hearing and resolving frequencies from 2020,000 hz., but can resolve about 130 dB of dynamic range. The dynamic range of CDs and 16 bit digital audio is, at best, 90 dB; the dynamic range of 24 bit digital audio is 109-120 dB, depending on the quality of the converters. You see, the DVD audio standard allows a wider variety of amplitudes, or volumes, than CDs (which is its main improvement)
Dynamic Range of Hearing

In addition to its remarkable sensitivity, the human ear is capable of responding to the widest range of stimuli of any of the senses. The practical dynamic range could be said to be from the threshold of hearing to the threshold of pain:
Threshold of Hearing I0 0 decibels
Threshold of Pain 1013I0 = 10,000,000,000,000 I0 130 decibels

Various sounds in dB Figures

Sounds Rocket Launching Jet Engine Thunderclap, Air Raid Siren 1 Meter Jet takeoff (200 ft) Rock Concert, Discotheque Firecrackers, Subway Train Heavy Truck (15 Meter), City Traffic Alarm Clock (1 Meter), Hair Dryer Noisy Restaurant, Business Office Light Traffic (50 Meter), Average Home Living Room, Quiet Office Library, Soft Whisper (5 Meter) Broadcasting Studio, Rustling Leaves Hearing Threshold
dB 180 140 130 120 110 100 90 80 70 50 40 30 20 0
Air Conditioning Unit, Conversational Speech 60
Dynamic range of Various Audio systems :

Audio Device/Application AM Radio Analog Broadcast TV FM Radio Analog Cassette Player Video Camcorder ADI SoundPort Codecs 16-bit Audio Converters Digital Broadcast TV Mini-Disk Player CD Player 18-bit Audio Converters Digital Audio Tape (DAT) 20-bit Audio Converters 24-bit Audio Converters Analog Microphone Dynamic Range 48 dB 60 dB 70 dB 73 dB 75 dB 80 dB 90 to 95 dB 85 dB 90 dB 92 to 96 dB 104 dB 110 dB 110 dB 110 to 120 dB 120 dB
CD quality recordings are done with a sample size of 16 bits. Some newer audio devices like DVDs are capable of working with the sample size of 24 bits. One of the advantages of using digital recorders with more bits is the ability to directly record uncontrolled microphone signals.
Do we need to go for further higher sampling frequencies and Bit depths ?

Bit-rates RAW vs Compressed

The term bit-rate refers to how many bits (1s and 0s) are used each second to represent the signal. The bit rate tells how many bits per unit of time are needed to store the sound The bit-rate for digital audio is expressed in thousands of bits per second (kbps) and correlates directly to the file size and sound quality. Lower bit-rates result in smaller file sizes but poorer sound quality, and higher bit-rates result in better quality but larger files.
The bit-rate of uncompressed audio can be calculated by multiplying the sampling rate by the resolution (8-bit, 16-bit, etc.) and the number of channels. For example, CD Audio (or a WAV file extracted from a CD) has a sampling rate of 44,100 times per second, a resolution of 16 bits and two channels. The bit-rate would be approximately 1.4 million bits per second (1,411 kbps).
For compressed sound its lower than the above product and the difference
can be viewed as a degree of compression.

Calculating Bit-rates e.g CD stereo format

Sampling Rate 44,100 Hz x x Resolution 16 bit x x # of Channels 2 = = Bit-rate 1,411,200
Digital Storage Capacity

A CD can store up to 74 minutes of music, so the total amount of digital data that must be stored on a CD is: 44,100 samples/(channel*second) X 2 bytes/sample X 2 channels X 74 minutes X 60 seconds/minute = 783,216,000 bytes 747 MB
Space required for of stereo digital audio

File Size of one stereo minut e 10.1 mega bytes 11.0 mega bytes 33.0 mega bytes 0.94 mega bytes File size of a three minute song 30.3 mega bytes 33 mega bytes 99 mega bytes 2.82 mega bytes
Bit Depth
Sample Rate
Bit Rate
16
44,100
1.35 Mbit/sec 1.46 Mbit/sec 4.39 Mbit/sec 0.13 Mbit/Sec
16
48,000
24
96,000
128 mp3 file k/bit rate

Hard disk requirements for a multi-track 3 minute song

Bit depth/ sample rate 16/44.1 16/48 24/96 16/44.1 16/48 24/96 Songs per 20 gigabyte hard disk 164 150 50 82 74 24 Songs per 200 gigabyte hard disk 1640 1500 500 820 740 240
Number Size Size per of mono per mono song tracks track 8 8 8 16 16 16 15.1 Mb 16.5 Mb 49.5 Mb 15.1 Mb 16.5 Mb 49.5 Mb 121 Mb 132 Mb 396 Mb 242 Mb 264 Mb 792 Mb

File Size vs. Bit-rate

File Size (4-min. song) 41.3MB MB per Minute Compres Hours sion per Ratio GB None 1.7 4-min. Songs per GB 25
Bit-rate 1,411 kbps (CD Audio) 80 kbps 128 kbps 160 kbps 192 kbps 256 kbps 320 kbps
10.3
2.3MB 3.8MB 4.7MB 5.6MB 7.5MB 9.4MB
0.6 0.9 1.2 1.4 1.9 2.3
7.6 = 1 11.0 = 1 8.8 = 1 7.3 = 1 5.5 = 1 4.4 = 1
29.1 18.2 14.6 12.1 9.1 7.3
437 273 218 182 137 109

Question arises which is the purest digital audio ? Can we afford to record and store it in that format ?
What about the bandwidth aspect and transferability convenience ? What is the solution ?
Compression ..?

Table Bit-rates for High Quality Sound
Format
CD MPEG Layer-I MPEG Layer-II MPEG Layer-III (MP3) MPEG Layer-III (MP3) MPEG AAC
Bit-rate
1.4Mbps 384 kbps 256 kbps 192 kbps VBR Normal/High 128 kbps
Compression
None 3.6=1 5.5=1 7.3=1 7=1 to 10=1 11=1

Comparison Chart of Audio Formats
Format
DSP TrueSpeech
Attributes
8.0kHz,1 bit, mono 11.5kHz, 16kBit/s mono
Megabyte per hour (approximate)

4 4.5 25 56 82 600
Characteristics of source sound

Low-quality voice recording High-quality voice recording Low or middle quality music Near high-quality music High-quality music High-quality (CD quality) recordings DVD Audio, Super Audio CD recordings High-quality voice recording High-quality recordings High-quality recordings
MP3*
22.05kHz, 56kBit/s, stereo 44.1kHz, 128kBit/s, stereo 44.1kHz, 192kBit/s, stereo
PCM PCM (High Quality) WMA voice WMA lossless Flac lossless
44.1kHz,16 bit, stereo 96kHz, 24 bit, stereo (available for Professional and Developer Edition users only) 20kBit/s, 22.05kHz, mono VBR Quality 100, 44 kHz, 2 channel 16 bit 96kHz, 16 bit, stereo
1978
9 150 240

Digital audio file formats

Digital audio files, which are the storable and editable collection of samples organized in a standard form, can be stored on computer drives, transferred to other computers or samplers, shared on the Internet to be downloaded or played-back in real-time. They are different from audio CD or DAT tracks, which mostly contain only the raw sample data or items useless to a computer That is why a CD track must be 'extracted' or 'ripped' to an audio file format to be usable by a computer application. A standard 16-bit, 44.1K stereo file eats up about 10 Mb of disk space per minute of sound. Audio files come in a variety of types, which can influence their bit depth, multi-channel organization, compression scheme, sampling rate.
Audio Formats
Audio Formats can be broken down into three main categories :- Uncompressed formats , Lossless compression formats & Lossy compression formats.
Uncompressed audio formats (often referred to as PCM formats) are just as the name suggests formats that use no compression. This means all the data is available, at the risk of large file sizes. A WAV audio file is an example of an uncompressed audio file. Lossless compression format applies compression to an uncompressed audio file, but it doesn;t lose information or degrade the quality of the digital audio file. The WMA audio file format uses lossless compression . Lossy compression format will result in some loss of data as the compression algorithm eliminates redundant or unnecessary information basically it tosses what it sees as irrelevant information. Lossy compression has become popular online because of its small file size, it is easier to transmit over the Internet. MP3 and Real Audio files uses a lossy compression
Some common Formats are :
.aif or AIFF (Audio Interchange File Format), Apple Computer developed the Audio Interchange File Format (AIFF) audio file format .It is the gold standard of 16-bit audio, travels well between almost all computers and software . Also capable of 24- bit and 32-bit resolution .. Can be used to store high-quality sampled audio and musical instrument information. it uses much more disk space -- about 10MB for one minute of stereo audio. Extension icon: .aif compressed version of AIFF was thought to be candidate to supercede AIFF as it non lossy compression format,Compression obtained by manipulating bitrate. or BWF (Broadcast Wave Format) is a standard audio format created by the European Broadcasting Union as a successor to WAV. BWF allows metadata to be stored in the file. This format is the primary recording format used in many professional Audio Workstations used in the Television and Film industry. Extension icon: .bwf
.aifc or AIFC or AIFF-C

.bwf
WAV (.wav) WAV is the format used for storing sound in files developed jointly by Microsoft and IBM. Support for WAV files was built into Windows 95 making it the de facto standard for sound on PCs. Windows uses the Wave Form Audio (WAV) file format to store sounds as waveforms. One minute of Pulse Code Modulation (PCM)-encoded sound can occupy as little as 644 kilobytes (KB) or as much as 27 megabytes (MB) of storage. This size of the storage space depends on the sampling frequency, the type of sound (mono or stereo), and the number of bits that are used for the sample. WAV sound files end with a .wav extension and can be played by nearly all Windows applications that support sound. Extension icon: .wav WMA - Windows Media Audio (.wma) Windows Media Audio (.wma) Short for Windows Media Audio is a Microsoft file format for encoding digital audio files similar to MP3 though can compress files at a higher rate than MP3. Systems Format (.asf) files include audio that is compressed with the Windows Media Audio (WMA) codec. WMA files, which use the ".wma" file extension, can be of any size compressed to match many different connection speeds, or bandwidths. Extension icon: .wma
MP3 (.mp3) MP3 is the name of the file extension and also the name of the type of file for MPEG, audio layer 3. Layer 3 is one of three coding schemes (layer 1, layer 2 and layer 3) for the compression of audio signals. Layer 3 uses perceptual audio coding and psychoacoustic compression to remove all superfluous information (more specifically, the redundant and irrelevant parts of a sound signal. The stuff the human ear doesn't hear anyway). It also adds a MDCT (Modified Discrete Cosine Transform) that implements a filter bank, increasing the frequency resolution 18 times higher than that of layer 2. The result in real terms is layer 3 shrinks the original sound data from a CD quality sound (with a bit rate of 1411.2 kilobits per one second of stereo music) by a factor of 12 (down to 112-128kbps) with almost the same fidelity . Extension icon: .mp3 Real Audio (.ra .ram .rm) Real Audio is a proprietary format, and is used for streaming audio on line that enables you to play digital audio files in real-time. To use this type of file you must have RealPlayer (for Windows or Mac), which you can download for free. Real Audio was developed by RealNetworks Extension icon: .ra .ram .rm

Comparison Analogue v/s Digital Table I summarizes the comparison of studio quality reel-to-reel analog tape recorder with 16 bit digital recorder. These data are derived from specifications by various manufacturers of analog and digital audio products. This table implies that the digital recorder has many advantages over its analog counterpart Performance.
Parameter
S/N Ratio Total harmonic distortion Wow and Flutter Frequency Response Loss of S/N during copying
Reel to Reel Tape Recorder

65 dB (linear system) 0.2% 0.03% 30-20,000 Hz ( +/- 3 dB) 3 dB
16 Bit Digital Recorder

93 dB (linear system) 0.005% UNMEASURABLE (Quartz Accuracy) 0-20,000 Hz (+/- 0.5 dB) NONE (PERFECT COPY) - as long as done in digital domain NONE (as long as within limits of correctability of the system)
Deterioration over time
YES


Digital Audio Standards

AES/EBU SDIF (Sony digital interface Format) S/P DIF (Sony Philips digital Interface Format) MADI( Multi Channel Audio Digital Interface) MPEG Standards

What is AES/EBU ?
Is the most popular audio standards Is a bit serial communication protocol for transmitting digital audio It provides up-to two channels of 24 bit per sample Provides both professional and consumer modes

AES/EBU
Originally published in 1985 Ratified by EBU with provision of transformer coupling Later on IEC combined professional and consumer standards
It provides two channels of audio data, a method for communication control and status information and some error detection capabilities. AES/EBU is a bit-serial communications protocol for transmitting digital audio data through a single transmission line. Clocking information is derived from the AES/EBU bit stream, and is thus controlled by the transmitter. The standard mandates use of 32 kHz,44.1 kHz, or 48 kHz sample rates, but some interfaces can be made to work at other sample rates.
Audio Standard AES-3 1992

Format Physical XLR I/O XLR wiring Electrical Serial transmission of two channels of sampled and linearly encoded data Shielded twisted-pair (100 meters max.) Output male pins with female shell Input female pins with male shell Pin 1 shield, earth, ground Pin 2 signal Pin 3 signal 110 ohm source 2 V to 7 Vp-p across 110 ohm (balanced) Rise/Fall time 5-30 ns Intrinsic Jitter 0.025UI 32kHz, 38kHZ, 44.1kHz, 48kHz, 96kHz, 192kHz
Sampling frequencies

B loc k 1
B lo ck 2
B lo ck 3
B loc k 4
F ram e 1 91
F ram e 0
F ram e 1
S ub F ram e 1 0 B it N u m be r 3 4 A ux iliary bits 7 8 S am ple D ata 20 b its 3 2 B its 27
S u b F ram e 2 28 V U 29 C 30 P 31
X , Y , Z P re am ble
V U C P
= = = =
V alidity U ser D a ta C h an ne l S tatu s P arity

Sub-frame
Y Subframe
Frame
32 Bits/subframe 4 Preamble Bits, 24 Data Bits, 1 Validity, 1 User, 1Channel Status & 1 Parity

Highlights
LSB is sent first than MSB for simple arithmetic Even parity is employed for each sub frame Two sub frames make one frame First take L channel and the second sub frame take R channel

Highlights Contd..
At 48kHz the bit rate will be 3.072 MHz Two separate synchronizing patterns are used for easy separation L & R Channels When the resolution required is only 20 bits, then we can use 4 bits for talk back

Frame Sequence
One Block = 192 Frames One Frame = 2 Subframes Start of Block indicated by Z Preamble
Sub-frame Structure
Preamble
4 bit synchronizing word X (Channel1) Y (Channel 2) Z (Channel 1 & 192 Frame Identifier)
Aux
4 bit used for auxiliary data or 4 LSB of 24 bit audio word
Audio Data
24 bit audio or 20 bit audio
Validity Bit (V)

Value of 1 indicates audio word suitable for decoding
Parity (P)
Produces even parity in values 4-31. Simple error detection
Auxiliary data
Used for studio talk balk communication Samples at 1/3rd of sampling frequency 12 bit sample word Divided into 4 bit nibbles 192X4 = 768 bits per channel/block 64X12=768 talkback samples

USER BIT
Has flexible frame length User bit management is in byte 1 of Channel status Many professional eqpts. Do not use this. In such cases it is set to ZEROs


SPDIF
Sony-Philips Digital Interface format Refers to AES/EBU standard operated in consumer mode Un balanced RCA connector is used Not used for embedding in Digital video Used for CD and R Dat (44.14 and 48kHz)

Digital System - The benefits

It is highly flexible and versatile. No quality degradation with multiple usage. Highly reliable, efficient and nondestructive editing Complete automation is possible. Excellent signal to Noise Ratio - better than 90 dB. Dynamic Range - better than 90 dB Cross Talk - better than 90 dB Distortion - 0.004% Wow & Flutter - Nil Quality depends only on the quality of conversion process. It is possible to use compression technique. This results in bandwidth saving. Tremendous benefits in the areas of post-production, archiving and multi- generation. Since the signal is in binary form, it is immune from noise. Hard disk recording facilitates random access. Centralized storage system can be provided.


Digital Audio

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DEEPAK JOSHI DY.

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

What should be the Ideal sampling frequency

No straight reply how ever Nyquist criteria comes to HELP

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

Solution to Aliasing problem:

DEEPAK JOSHI DY.DIRECTOR STI(T)

Anti Aliasing Filter

DEEPAK JOSHI DY.DIRECTOR STI(T)

Anti Aliasing Filter Contd

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

What should be the Ideal

DEEPAK JOSHI DY.DIRECTOR STI(T)

Dynamic range of an audio

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

Dynamic Range of Hearing

Threshold of Hearing I0 0 decibels

Threshold of Pain 1013I0 = 10,000,000,000,000 I0 130 decibels

DEEPAK JOSHI DY.DIRECTOR STI(T)

Various sounds in dB Figures

dB 180 140 130 120 110 100 90 80 70 50 40 30 20 0

Air Conditioning Unit, Conversational Speech 60

Dynamic range of Various Audio systems :

Do we need to go for further higher sampling frequencies and Bit depths ?

DEEPAK JOSHI DY.DIRECTOR STI(T)

Bit-rates RAW vs Compressed

can be viewed as a degree of compression.

Calculating Bit-rates e.g CD stereo format

Digital Storage Capacity

Space required for of stereo digital audio

1.35 Mbit/sec 1.46 Mbit/sec 4.39 Mbit/sec 0.13 Mbit/Sec

128 mp3 file k/bit rate

DEEPAK JOSHI DY.DIRECTOR STI(T)

Hard disk requirements for a multi-track 3 minute song

DEEPAK JOSHI DY.DIRECTOR STI(T)

File Size vs. Bit-rate

2.3MB 3.8MB 4.7MB 5.6MB 7.5MB 9.4MB

0.6 0.9 1.2 1.4 1.9 2.3

7.6 = 1 11.0 = 1 8.8 = 1 7.3 = 1 5.5 = 1 4.4 = 1

29.1 18.2 14.6 12.1 9.1 7.3

437 273 218 182 137 109

DEEPAK JOSHI DY.DIRECTOR STI(T)

DEEPAK JOSHI DY.DIRECTOR STI(T)

Table Bit-rates for High Quality Sound

DEEPAK JOSHI DY.DIRECTOR STI(T)

Comparison Chart of Audio Formats

Megabyte per hour (approximate)

Characteristics of source sound

22.05kHz, 56kBit/s, stereo 44.1kHz, 128kBit/s, stereo 44.1kHz, 192kBit/s, stereo

DEEPAK JOSHI DY.DIRECTOR STI(T)

Digital audio file formats

Some common Formats are :

.aifc or AIFC or AIFF-C

DEEPAK JOSHI DY.DIRECTOR STI(T)

Reel to Reel Tape Recorder