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Kiley Cudmore
Tuesday, Feb 18, 2014 Lesson time; 9:00am-11:00am Chapter 3 notes
Microphone- used to translate the variations in the air pressure into an analogous change in electrical voltage/ once amplifired this changing voltage can be sampled periically through a process known as sample and hold- and capture incoming signal voltage. Ex; 48kHz = 48,000 samples per every second. Analog to Digital Converter(ADC)- receives the voltages from the dame and hold circuit and assigns a numerical value to each amplitude through quantization. Each numerical value is represented with a binary number system and is stored in memory where it is delivered to the DAC when played back. Digital to Analog Converter(DAC)- converts each number to a voltage and feeds those voltages to an amplifier to increase the amplitude of the voltage. Pulse Code Modulation(PCM)- the method most widely used to represent sound digitally-using two primary components to define the audio signal, sample rate, and bit depth. Sample Rate- number of samples taken per second; expressed as Hz or kHz-kiloHertz. Bit Depth- number of bits used to define the amplitude of each indiviual sample which determines the resolution with which we can measure the signal and how accurately the waveform can be reproduced. The Binary # System- is used by all computing devices for representing numerical values using bits in 4 possible values; 00, 01, 10, or 11. Direct Stream Digital(DSD)- 1 bit audio format which uses a sample rate of 2.8224 mHz -used by song and phillips for super audio compact discs using Sigma Delta Modulation- voltage is converted into a pulse, the frequency of the pulse is ued to communicate the signal. A comparator- analyzes the difference between the actual voltage and the expected voltage and assigns a 1 bit value to express if the actual voltage is < or > than the expected value. Noise shaping- pushes noise into a rage above human hearing so that quantization errors are less apparent. Protools and DAWS don't support DSD, ot SACD. Harold NyQuist- the sample rate must be at least two times the highest frequency being recorded(computer can accurately represent frequencies up to half the sampling rate). Aliasing- the result of frequencies being misidentified and their values being applied to the incorrect frequencies. GuardBand- the range of frequencies between where the LPF is first applied and the Nyquist frequency. Standard sample rate for cds; 44.1 kHz. Digital Signal- the stream of binary digits representing the quantized samples. Quantization Error- any sample will be off by a random amount/white noise/ not present in orginal sigal. Quantization Noise- added into the digital signal through imprecise quantization. Signal to Quantization noise rate(SQNR)- the ratio of the max signal amplitude to the max quantization error. Clipping- an electrical signal that exceeds max amplitude expected by ADC, it will be clipped when its quantized. Clipping also occurs when you produce numbers in a computer that exceed the max expected by the DAC, causing the sound to be a clipped version of the original. Digital Audio Signals- measured on a scale called DBFS for dB relative to full scale. 0 dbfs is assigned for the highest possible value that can be represented by its bit depth -half the highest value= -6dbfs -any signal above 0dbfs will clip. The standard sample rate for compact discs is 44.1k samples per second for each channel of audio. Standard depth for cd is 16 bit, so every 44,100th of a second a 16 bit sample is taken. One minute of audio is roughly 10 MBS. A compact disc(700MB) has about 70 minutes of sterio audio. Recording 24 tracks at 32 bit/192k will use about a Gigabyte per minute. Samplers- storing recorded sounds or samples, allowing individual sounds to be played back when triggered. Digital Synthesizers- used to generate sounds/ not play back. Digital audio synthesis measures sounds in a list of numerical values, for a list of numbers to be audible as sound, and fluctuate up and down at a frequency within the range of human hearing. The numbers are then sent to the DAC where they are converted into voltages. Digital Signal Processing(DSP)- effects of digital filters- formulae for modifying digital signals by delay, multiplication, addition, and other operations. mp3- cd quality sound, smaller file, most commonly used lossy format. Compression includes; Lossy-loses some information to create a smaller size. Lossless- doesn't effect original sound quality. Playback on computer; mp3, AAC, WMA. Playback on audio system; FLAC, WMA lossless. Lossless encoder- uses complex algorithms to determine what sounds a human is able to hear, based on psycho-acoustic models, and chops off sound outside this range.
