Training session No.

Digital Audio
AES/EBU AES3 Interface

BY: R.Naithani

AUDIO
Longitudinal Pressure variation Audible Bandwidth 20KHZ Dynamic Range of Human Ear 140 db Dynamic Range of Music orchestra 120 db Perception of sound to year depends on a: Loudness b: Pitch c: quality

Analogue Audio
Analog Audio signal still the most familiar form of communication. a:AM Radio Broadcasting .carrier frequency range 535-1605 KHZ .Bandwidth 10KHZ b:

Comparison of Digital and Analogue Audio

Analogue Audio(LP Record) Frequency Response 30HZ-20 KHZ +3db Dynamic Range 70 db Signal to Noise Ratio 60 db Harmonic distortion 1-2% Durability: high frequency response degrades with playing Digital Audio(CD System) Frequency response 20HZ-20KHZ +0.5/-1 db Dynamic Range>90db Signal to Noise Ratio>90db Harmonic Distortion 0.005% Durability :Permanent

Sampling Theorem and its Importance

Sampling Theorem: A band limited signal can be reconstructed exactly if it is sampled at a rate at least twice the maximum frequency component in it." Figure 1 shows a signal g(t) that is band limited.

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Quantisation
In order to encode a continuous-time signal, we must first quantise it into a finite number of discrete amplitude signals. The quantisation depends on the desired number of different signal levels (the resolution), and the maximum variation of the signal to be represented

Fig: Binary Waveforms used for encoding Digital data's

a:Unipolar NRZ b:Polar NRZ c:Polar RZ d:Split Phase or Manchester e:Alternate mark inversion(AMI)

Biphase mark code

The Biphase mark code (also called FM1 code or Biphase encoding) is a type of encoding for binary data stream. binary data stream is sent without modification via a channel, there can be long series of logical ones or zeros without any transitions which makes clock recovery and synchronization difficult. Biphase mark code makes synchronization easier by ensuring that there is at least one transition on the channel between every data bit; in this way it behaves much like the Manchester code scheme. When encoding, the symbol rate must be twice the bitrate of the original signal. Every bit of the original data is represented as two logical states which, together, form a bit. Every logical 1 in the input is represented as two different bits (10 or 01) in the output. The input logical 0 is represented as two equal bits (00 or 11) in the output. Every logical level at the start of a cell is inversion of the level at the end of the previous cell. In BMC output the logical 1 and 0 are represented with the same voltage amplitude but opposite polarities, BMC coding provides a better synchronization since there is a change in the polarity at least every two bits. It is not necessary to know the polarity of the sent signal since the information is not kept in the actual values of the voltage but in their change: in other words it does not matter whether a logical 1 or 0 is received, but only whether the polarity is the same or is different from the previous value; this makes synchronization even easier. Finally, BMC coded signals have zero average DC voltage, thus reducing the necessary transmitting power and minimizing the amount of electromagnetic noise produced by the transmission line. All these positive aspects are achieved at the expense of doubling clock frequency

Quantization
In order to encode a continuous-time signal, we must first quantize it into a finite number of discrete amplitude signals. The quantisation depends on the desired number of different signal levels (the resolution), and the maximum variation of the signal to be represented (the range). The quantising operation can be viewed as a a function operating on each sample value. In the case of uniform quantising there is a uniform spacing between the quantisation levels

AES3-1992 (ANSI S4.40-1992): Broadcast Digital Audio Format

Developed by AES together with EBU A twisted pair of wire with wide band capabilities is the transmission medium it allows for bit serial transmission of digital audio data. The interface is primarily designed to carry monophonic or stereophonic signals in a studio environment at a 48kHz sampling frequency and with a resolution of 20 or 24 bits per sample The bit-parallel data words are serialized by sending the least significant bits (LSB) first. Word clock data is added to the bit stream to identify the start of each sample in the decoding process.

The AES/EBU digital audio data structure

Conceptual block diagram of an AES/EBU channel encoder.

channel for producer talkback or studio-to-studio communication. Alternately, they can be used to augment the audio word-length to 24 bits. Time slots 8 to 27: These time slots carry 20 bits of audio information starting with LSB and ending with MSB. If the source provides fewer than 20 bits, the unused LSBs will be set to the logical 0.

and it stipulated that up to four receivers could be connected in parallel across the audio cable. However, it gave no guidance on precautions needed to be taken by the user or systems integrator. This resulted in difficulties with reflections and standing waves, as the performance of the distribution link was unpredictable and depended on the wide variety of installation conditions encountered in practice. The unpredictability is compounded by the loose specification of the output signal amplitude, which puts an additional stress on the receiver. The standard was revised and reissued as AES3-1992. This second version specifies a receiver input impedance of 110 and warns against the use of more than one receiver across the feeding cable. The AES3id-1996 standard defines the unbalanced 75 impedance interface. This version recognizes the need to narrowly specify impedance tolerances in terms of return loss and transmitter output signal levels and, if properly implemented, results in a more predictable performance as it is based on well-known SDTV video signal distribution

Figure 2 shows a conceptual block diagram of an AES/EBU encoder. Figure 3 shows how the BPM-encoded signal waveform is obtained from an NRZ data stream. The NRZ is characterized by ones having a determined high value and zeros having a determined low value. This means that long strings of zeros and ones have no transitions and result in difficult clock recovery in the receiver. BPM alleviates this condition by introducing transitions in the middle of each one bit interval. At a 48kHz sampling rate, the total data rate is 32 2 48000 = 3.072Mb/s. The BPM encoding doubles the data stream rate to 6.144Mb/s. Figure 4 shows the respective spectra. Figure 5 on page 28 shows a conceptual block diagram of an AES/EBU decoder.

Your computer has a soundcard - it could be a separate card, like a SoundBlaster, or it could be built-in to your computer. Either way, your soundcard comes with an Analog-to-Digital Converter (ADC) for recording, and a Digital-toAnalog Converter (DAC) for playing audio. Your operating system (Windows, Mac OS X, Linux, etc.) talks to the sound card to actually handle the recording and playback, and Audacity talks to your operating system so that you can capture sounds to a file, edit them, and mix multiple tracks while playing.

Standard file formats for PCM audio

There are two main types of audio files on a computer: PCM stands for Pulse Code Modulation. This is just a fancy name for the technique described above, where each number in the digital audio file represents exactly one sample in the waveform. Common examples of PCM files are WAV files, AIFF files, and Sound Designer II files. Audacity supports WAV, AIFF, and many other PCM files. The other type is compressed files. Earlier formats used logarithmic encodings to squeeze more dynamic range out of fewer bits for each sample, like the u-law or alaw encoding in the Sun AU format. Modern compressed audio files use sophisticated psychoacoustics algorithms to represent the essential frequencies of the audio signal in far less space. Examples include MP3 (MPEG I, layer 3), Ogg Vorbis, and WMA (Windows Media Audio). Audacity supports MP3 and Ogg Vorbis, but not the proprietary WMA format or the MPEG4 format (AAC) used by Apple's iTunes. For details on the audio formats Audacity can import from and export to, please check out the Fileformats page of this documentation. Please remember that MP3 does not store uncompressed PCM audio data. When you create an MP3 file, you are deliberately losing some quality in order to use less

of up to 24 times can be achieved with near(but not) CD-quality. The beauty of MP3 is it's size vs. perceived quality, also its ability to be downloaded and then loaded into the flash memory of MP3 players. It can also be streamed to MP3 client software, recognized most Web browser audio helper applications. Files are encoded at certain bit-rates for target download speeds; for example, very good quality can be attained with 160 kbps encoding. Would you want to master all your music on MP3--no, but at least you can listen to it while you're jogging. Click here for massive information overload. .ra or .ram (Real Audiocan be streamed on the Internet from a Real Audio server, so sound starts playing before file fully downloaded. They can be encoded at multiple sampling rates to accommodate different user download speeds (modem, DSL, T1 lines, etc.) which range from 8 kbps to 1.5 Mbps (don't try 1.5 Mbps on your grandmother's 28.8 modem). Can also be combined with video for Real Media streaming. It spreads compression