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J-Himmy Productions
During chapter three I had learned quite a bit about Digital Audio. One of the things I learned was that that the job of a mic is to transduce the variations in air pressure into an analogous change in electric voltage. Once amplified, this continuously changing voltage can be sampled periodiclly through a process known as Sample and Hold. I had also learned about the different types of digital recorders such as Stereo recorders and Multi-Track recorders. The time i spent in the studio was a great experiance as well. My mentor had taken me into the B studio witch is one of my dream set ups so of course I had a great time. After going over the chapter, Luise went along to anser some questions I had and also showed me a few new projects in the making while giving me tricks/ tips to improve my own work at the same time. I feel that I may still need to study this chapter a little more because it takes me longer to understand things,but I do know that when i get it down 100% i will be able to use this info in the real world as an engineer.
-Section 1:
- A movie is made by taking a series of still photos in rapid sequence, at a constant rate of at least twenty-four frames per second.
- The job of a microphone is to transduce the variations in air pressure into an analogous change in electrical voltage.
- Sample and hold = Will instantly capture the incoming analog signal voltage, and hold its value constant until the next sample is taken.
- If sample rate is 48kHz then there are 48,000 sample points every second.
- The more sample points, the more accurate the soundwave.
- ADC (Analog-to-digital converter) = Receives the discrete voltages from the sample and hold circuit and assigns a numerical value to each amplitude.
- Quantization = Converting these voltages to numbers.
- DAC (Digital-to-analog converter) = Converts each number (From binary) to a voltage and feeds those voltages to an amplifier to increase the amplitude of the
voltage.
- PCM (Pulse code modulation) = Most widely used to represent sound digitally, uses two primary components to define the audio signal: Sample rate and Bit depth.
- Sample rate = Number of samples taken per second
- Bit depth = The number of bits used to define the amplitude of each individual sample which determines the resolution with which we can measure the signal.
- DCD (Direct stream digital)
- SACD (Super audio compact discs
- ΔΣ (Sigma Delta Modulation)
- Nyquist theorem = The sample rate or frequency at which samples are taken must be at least two times the highest frequency being recorded, in order to accurately
capture and encode all frequencies in the signal.
- Nyquist frequency must be filtered out before the sampling process takes place. This is accomplished by sending the electrical signal through a low-pass filter
which removes any frequencies above a certain point. If these frequencies are not filtered out prior to being sampled, they will lead to aliasing.
- Aliasing is the result of frequencies being misidentified and their values being applied to the incorrect frequencies.
- 1 Byte = 8 Bits
- Quantization error = White noise = added into the digital signal by the imprecise nature of quantization (Quantization noise)
- Quantization error is unavoidable, but can be reduced to an acceptable level by using more bits to represent the total range of amplitude.
- SQNR (Signal-to-quantization-noise-ratio) = Ratio of the maximum signal amplitude to the maxium quantization error.
- SQNR is equal to 2 to the n power, where n is the number of bits used to quantize each sample. (Expressed in Decibels)
- The more bits used define each sample, the better the potentail ratio of signal to noise.
- Each bit of precision used in quantization adds 6 dB to the SQNR.
- A signal that exceeds maximum amplitude will be clipped when it is quantized
- The amplitude of the electrical signal should not exceed the maximum expected by the ADC
- DBFS = Decibels relative to full scale.
- Any signal above 0 DBFS will clip
- 0 DBFS is assigned to the highest possible value that can be represented by its bit depth.
Review Questions:
- The Nyquist Theorem states that: The highest frequency able to be sampled is half the sample rate
- The "Sample Rate" refers to the amount of "Pictures" taken of a waveforms amplitude over 1 second
- The "Bit Depth" refers to the amount of possible amplitude values present in the digital audio signal
-Section 2 Review Questions:
- Sample rate is:
- The amount of samples taken over 1 second
- The rate at which samples are played back
- 44,100 on a standard CD
- The answer would be "All of the above" but these are what is included in all of the above
- PCM stands for: Pulse Code Modulation
- Bit Depth
- Determines the amount of amplitude values avaliable for the audio signal
- Is similar to the resolution of a digital picture
- Is 16 bit on a standard CD
- The answer would be "All of the above" but these are what is included in all of the above
Section 3 Review questions:
- All audio should be recorded to your computer's internal hard drive. FALSE
- At 16 bit, one minute of audio takes approximately "10 MB" of space on your computer
- Recording 24 tacks simultaneously at 32 bit/192k will use about "1 GB" every minute
- An audio CD that holds 700 MB of information, can hold about "70" minutes of audio
Section 4:
- Samplers = Playback recorded sounds
- Digital Synthesizers = Used to generate sounds
Section 4 Review questions:
- In 1976 the fairlight CMI was released. The Fairlight CMI was the first: Polyphonic Digital Sampler
- "Sampler" work by storing recorded sounds in memory, allowing the individual sounds to be played back as they are triggered.
- A "Sequencer" was one of the first digital audio devices. It is a device that keeps track of the order trigger events are played in.
Section 5:
- Multiplication is equivalent to "Audio amplification
- Addition is equivalent to "Audio mixing"
- DSP (Digital signal processing) = Concerned with the effects of digital filters.
Lossy Compressed Formats:
- AAc (Advanced Audio Coding) = Mostly used by apple.
- ATRAC3 (OMA, OMG) = Open Magic Gate = Used by sony, not compatible with anything but sony.
- LAT, LQT, LSL (Liquid Audio) = Not really used anymore.
- MP3 (MP3) = Mostly used/Preferred over most formats.
- MP3PRO (MP3) = Better version of MP3, has same file extension as regular MP3 files
- OGG Vorbis (OGG) = Designed as a subsitute for MP3 and WMA. Uses variable bitrate compression to produce better quality when needed.
- Quicktime Audio (MOV) = Uses the same MPEG-4 technology as the AAC format
- RealAudio Media (RA, RM, RMA) = Designed particularly for real-time streaming audio feeds
- Windows Media Audio (WMA) = Microsofts digital audio format, typically is slightly better than MP3
- Lossy = Don't sound quite as good as the originals.
Lossless Compressed formats:
- Apple Lossless audio codec (ALAC, M4A) = For apple
- Free lossless codec (FLAC) = Embraced by many consumer electronics manufacturers
- Windows media audio lossless (WMA) = Best option for lossless compression today, for windows
Non-compressed formats:
- AU (Audio) = Not widely used today
- Audio interchange format (AIF, AIFF) - Mac system sounds
- Compact disc digital audio (CDA) = Format on pretty much all CDs you get from the store
- SDII (Sound designer two) = use with Digidesign pro tools and mac only
- Waveform sound files (Wav) = Produces exact copy of original recording, with zero compression.
Section 5 review questions:
- The most popular digital music format used today is ".mp3"
- mp3 is a "lossy" file format
- Although .mp3 is the most popular digital music format, it is not a professsional audio format: TRUE
