Noise shapingNoise shaping is a technique typically used in digital audio, , and video processing, usually in combination with dithering, as part of the process of quantization or bit-depth reduction of a digital signal. Its purpose is to increase the apparent signal-to-noise ratio of the resultant signal. It does this by altering the spectral shape of the error that is introduced by dithering and quantization; such that the noise power is at a lower level in frequency bands at which noise is considered to be less desirable and at a correspondingly higher level in bands where it is considered to be more desirable.
OversamplingIn signal processing, oversampling is the process of sampling a signal at a sampling frequency significantly higher than the Nyquist rate. Theoretically, a bandwidth-limited signal can be perfectly reconstructed if sampled at the Nyquist rate or above it. The Nyquist rate is defined as twice the bandwidth of the signal. Oversampling is capable of improving resolution and signal-to-noise ratio, and can be helpful in avoiding aliasing and phase distortion by relaxing anti-aliasing filter performance requirements.
Reconstruction filterIn a mixed-signal system (analog and digital), a reconstruction filter, sometimes called an anti-imaging filter, is used to construct a smooth analog signal from a digital input, as in the case of a digital to analog converter (DAC) or other sampled data output device. The sampling theorem describes why the input of an ADC requires a low-pass analog electronic filter, called the anti-aliasing filter: the sampled input signal must be bandlimited to prevent aliasing (here meaning waves of higher frequency being recorded as a lower frequency).
Pulse-code modulationPulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform.
Audio system measurementsAudio system measurements are a means of quantifying system performance. These measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing.
Noise floorIn signal theory, the noise floor is the measure of the signal created from the sum of all the noise sources and unwanted signals within a measurement system, where noise is defined as any signal other than the one being monitored. In radio communication and electronics, this may include thermal noise, black body, cosmic noise as well as atmospheric noise from distant thunderstorms and similar and any other unwanted man-made signals, sometimes referred to as incidental noise.
MIDIMIDI (ˈmɪdi; Musical Instrument Digital Interface) is a technical standard that describes a communications protocol, digital interface, and electrical connectors that connect a wide variety of electronic musical instruments, computers, and related audio devices for playing, editing, and recording music. A single MIDI cable can carry up to sixteen channels of MIDI data, each of which can be routed to a separate device. Each interaction with a key, button, knob or slider is converted into a MIDI event, which specifies musical instructions, such as a note's pitch, timing and loudness.
WAVWaveform Audio File Format (WAVE, or WAV due to its ; pronounced "wave" or "wæv" ) is an standard, developed by IBM and Microsoft, for storing an audio bitstream on personal computers. It is the main format used on Microsoft Windows systems for uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format. WAV is an application of the (RIFF) bitstream format method for storing data in chunks, and thus is similar to the 8SVX and the (AIFF) format used on Amiga and Macintosh computers, respectively.
Ableton LiveAbleton Live, also known as Live or sometimes colloquially as "Ableton", is a digital audio workstation for macOS and Windows developed by the German company Ableton. In contrast to many other software sequencers, Live is designed to be an instrument for live performances as well as a tool for composing, recording, arranging, mixing, and mastering. It is also used by DJs, as it offers a suite of controls for beatmatching, crossfading, and other different effects used by turntablists, and was one of the first music applications to automatically beatmatch songs.
FLACFLAC (flæk; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of the original audio data.
