What exactly is digital recording? The definition is, "An electronic format that is designed to duplicate sound, while affording extremely accurate control over any changes you might wish to make in the recording" (Mclan & Wichman,1988). In simple terms it means that the digital circuitry samples the signal and then reproduces what it has seen. The quality of the recording depends on the sampling rate of the machine. The sampled signal is then encoded to the tape or disc in 1's and 0's, just like a computer disk drive would encode information.
Instruments nowadays are generally considered easier to play than their predecessors hundreds of years ago (due to quality manufactured parts, and standardisation of music notation). Computer technology has not affected traditional human instrumental sound production. Only its recording and reproduction. One of the most important features of the introduction of computers in music and audio is the way in which sound is recorded digitally, as opposed to analogue. Digital sampling allows for recorded sounds to be reproduced almost exactly as they were Also once a sound has been sampled it's sound wave can then be displayed on the computer screen.
AIM The main aim of this project is to demonstrate the pulse code digitization and companding on a signal and to observe the effects of sampling depths and companding on the signal to noise ratio (SQR). INTRODUCTION PCM- In the pulse code modulation (PCM), the signal is a digitally represented analog signal in which the signal magnitude is sampled with uniform intervals. Each sample is quantized to the closest value of the digital signal. In the pulse code modulation, the signal is binary. The two possible states represented in PCM are logic 1(high) and logic 0 (low).
If the audio file consists of only one channel, Left only, or Right only, copy the channel and paste it in the project. Join the two channels so that both are selected when you double click on either one of the channels. Assign the copied and pasted channel Left or Right as required so that the audio now has both a left and a right channel. Normalise the track. This will ensure that the loudness of the track will be comparable to that of other tracks when played back.
Auditory masking and filtering is used in order to compress the sound file without compromising quality. The beauty of digital audio is mainly reliant on the processing of waves and frequencies. Through further study, more and more compression techniques and higher quality file formats will be attained in the future. Truly, the music experience will get better and better as sound technology becomes more and more sophisticated. References 1.
To interact analog systems with digital systems or digital systems with analog systems conversion is needed. There are 2 types of conversions: (1) Analog to digital converters (ADCs), (2) Digital to analog converters (DACs). FIGURE1.Analog to Digital converter In the analog to digital converter the input vary from minimum to maximum value of volts or amperes. The output is a digital number that represents the input value. In the digital to analog converter the input that specifies an output whose value changes in steps.
Testing and validation are indispensable steps in the development of software platforms designed to emulate hardware components. Since modern sound level meters rely heavily on digital signal processing for sound analysis, it seems reasonable to expect that results of equal or greater accuracy can be realized on computer platforms. Requirements for precision measuring instruments are specified in the international standard IEC 61672-2003 – Electroacoustics – Sound Level Meters. The Standard is applicable to selfcontained or multi-component analogue and digital hardware systems, only briefly mentioning computer software as a provisional part of the instrument for displaying results and limited data manipulation. On the other hand, modern digital multi-channel systems provide means of storing recorded waveforms on a PC hard disk, thus being fully dependent on dedicated software for all subsequent processing and analysis.
The six analog signals from the strain gauges are associated into one signal by the help of a multiplexer. After that, the logic multiplexer will send the signal to the ADC converter which then converts the signal into digital format for the PC to interpret. This process is shows in figure 6 Figure 6: Interfacing Six channels with the support of Multiplexer D5. The minimum “ Sampling rate” use in this system The sampling rate to be used depends on the speed at which the ADC is able to collect samples into an on-board buffer. In other words, it is the speed at which the digitizer’s ADC converts the input signal, after the signal has passed through the analogue input path to digitalised values.
When the laser and input data are correctly timed, the modulator is either fully ‘on’ or ‘off’ when optical pulses passes through it Before the multiplexing operation is performed the incoming bit streams are temporally offset from one another by delay for four channels i.e. 25ps, 30ps, 35ps, 40ps. 4*1 multiplexer is used in this paper. And it assembles the higher bit stream from the baseband signal. It also reduces crosstalk.
Comparing Analog and Digital Recording In the present time we are always coming up with better electronics, because that is what we expect. So, in the recording industry we have moved from analog to digital recording. Musicians want the best recording gear to give them the fastest, easiest, and best sounding recordings for their music. Digital recording is the newer way of recording music since analog recording, but is it always better? There are obviously definite positives of digital recording or we wouldn’t be using it, but does it fall short to analog recording in some areas?