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Sampling Rates Sampling Rates The sampling rate, sample rate, or sampling frequency defines the number of samples per second (or per other unit) taken from a continuous signal (audio, video, etc.) to make a discrete signal. For time-domain signals, it can be measured in hertz (Hz). The inverse of the sampling frequency is the sampling period or sampling interval, which is the time between samples. The concept of sampling frequency can only be applied to samplers in which samples are taken periodically. Some samplers may sample at a non-periodic rate. The common notation for sampling frequency is fs which stands for frequency (subscript) sampled. COMMON SAMPLING FREQUENCIES (digital audio) 8,000 Hz - telephone, adequate for human speech 11,025 Hz 22,050 Hz - quarter and half the sampling rate of audio CDs (44,100 Hz, see below), used for lower-quality PCM and MPEG audio 32,000 Hz - some miniDV digital video camcorders, DAT (LP mode), Germany's Digitales Satellitenradio (German) 44,056 Hz - PCM adaptor using NTSC video tapes (245 lines by 3 samples by 59.94 frames per second), sometimes misused to play back audio streams sampled at 44,100 Hz (and vice versa) 44,100 Hz - audio CD, also most commonly used with MPEG-1 audio (VCD, SVCD, MP3), adopted from the PCM adaptor using PAL video tapes (294 lines by 3 samples by 50 frames per second) 47,250 Hz - world's first commercial PCM sound recorder by Nippon Columbia (Denon) 48,000 Hz - digital sound used for miniDV, digital TV, DVD, DAT, films and professional audio 50,000 Hz - first commercial digital audio recorders from the late 70's from 3M and Soundstream 50,400 Hz - sampling rate used by the Mitsubishi X-80 digital audio recorder 96,000 or 192,000 Hz - DVD-Audio, some LPCM DVD tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD-DVD (High-Definition DVD) audio tracks 2.8224 MHz - SACD, 1-bit sigma-delta modulation process known as Direct Stream Digital, co-developed by Sony and Philips Nyquist-Shannon The Nyquist–Shannon sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the bandwidth of the signal being sampled, or equivalently, that the Nyquist frequency (half the sample rate) exceeds the bandwidth of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal. For example, if a signal has a bandwidth of 100 Hz, to avoid aliasing the sampling frequency should be greater than 200 Hz. Oversampling In some cases, it is desirable to have a sampling frequency considerably more than twice the desired system bandwidth so that a digital filter can be used in exchange for a weaker analog anti-aliasing filter. This process is known as oversampling. Bitrate (or Bitdepth) In telecommunications and computing, bitrate (sometimes written bit rate, data rate or as a variable Rbit) is the number of bits that are conveyed or processed per unit of time. Bit rate is often used as synonym to the terms connection speed, transfer rate, channel capacity, maximum throughput and digital bandwidth capacity of a communication system. In digital multimedia, bitrate is the number of bits used per unit of time to represent a continuous medium such as audio or video after source coding (data compression). In this sense it corresponds to the term digital bandwidth consumption, or goodput (application level throughput). The bit rate is quantified using the 'bit per second' (bit/s or bps) unit, often in conjunction with a SI prefix such as kilo (kbit/s or kbps), Mega (Mbit/s or Mbps), Giga (Gbit/s or Gbps) or Tera (Tbit/s or Tbps). While often referred to as "speed", bitrate does not measure distance/time but quantity/time, and thus should be distinguished from the "propagation speed" (which depends on the transmission medium and has the usual physical meaning). Bitrates in multimedia In digital multimedia, bitrate represents the amount of information, or detail, that is stored per unit of time of a recording. The bitrate depends on several factors: • the original material may be sampled at different frequencies • the samples may use different numbers of bits • the data may be encoded by different schemes • the information may be digitally compressed by different algorithms or to different degrees Generally, choices are made about the above factors in order to achieve the desired trade-off between minimizing the bitrate and maximizing the quality of the material when it is played. If lossy data compression is used on audio or visual data, differences from the original signal will be introduced; if the compression is substantial, or lossy data is decompressed and recompressed, this may become noticeable in the form of compression artifacts. Whether these affect the perceived quality, and if so how much, depends on the compression scheme, encoder power, the characteristics of the input data, the listener’s perceptions, the listener's familiarity with artifacts, and the listening or viewing environment. Experts and audiophiles may detect artifacts in many cases in which the average listener would not. Some musicians enjoy the distinct artifacts of low bitrate (sub- FM quality) encoding and there is a growing scene of net labels distributing stylized low bitrate music. The bitrates in this section are approximately the minimum that the average listener in a typical listening or viewing environment, when using the best available compression, would perceive as not significantly worse than the reference standard Audio (MP3) 32 kbit/s — MW (AM) quality 96 kbit/s — FM quality 128 - 160 kbit/s - Decent quality, difference can sometimes be obvious 192 kbit/s — Good quality, difference can only be heard by a few 224 - 320 kbit/s — High quality, nearly lossless quality Other audio 4 kbit/s — minimum necessary for recognizable speech (using special- purpose speech codecs) 8 kbit/s — telephone quality (using speech codecs) 500 kbit/s–1 Mbit/s — lossless audio as used in formats such as FLAC, WavPack or Monkey's Audio 1411 kbit/s — PCM (WAV) sound format of Compact Disc Digital Audio Video (MPEG2) 16 kbit/s — videophone quality (minimum necessary for a consumer- acceptable "talking head" picture) 128 – 384 kbit/s — business-oriented videoconferencing system quality 1 Mbit/s — VHS quality 5 Mbit/s — DVD quality 15 Mbit/s — HDTV quality\ Notes For technical reasons (hardware/software protocols, overheads, encoding schemes, etc.) the actual bitrates used by some of the compared-to devices may be significantly higher than what is listed above. For example: Telephone circuits using µlaw or A-law companding (pulse code modulation) — 64 kbit/s CDs using CDDA — 1.4 Mbit/s REVIEW: When audio is digitized, an analog recording is played back through an electronic device, and the variations of the electric current generated by the device are sampled at very fast time intervals. The amplitude of the current, corresponding to the amplitude of the original sound wave, is recorded as a number at each sampling point. The quality and resolution of digitized audio is determined by two factors: 1. The number of times per second the amplitude of the wave is measured 2. The range of numbers used to record each measurement. The first factor, the "sampling rate" is described in kilohertz (kHz), or thousands of samples per second. Consumer audio CDs are recorded at a sampling rate of 44.1 kHz. That means that each second of audio is represented as 44,100 separate amplitude measurements as the wave flows past a point. Visual representation of two sample rates A. The top wave represents a low sample rate which does not accurately reproduce the shape of the original sound wave B. The bottom wave represents a high sample rate which more accurately reproduces the wave The second factor, the "bit depth" describes the range of numbers used to represent each amplitude measurement. For example, if each measurement were represented on a scale from 1-10, that would be a rougher measurement than a scale from 1-1000. Sample size is measured in bits. Eight-bit numbers range from 0-255; 16-bit numbers range from 0-65,535; and 24-bit numbers range from 0- 16,777,215. Since human ears are sensitive to the volume of sound, measured in decibals (dB), higher bit depths result in a “smoother” or more realistic representation of the audio source, or greater “dynamic range.” All audio CDs are 16 bit recordings. Please not that the discussion above discusses only factors related to digital audio reformatting. The quality of the original analog playback into the digital system will also greatly affect the final digital audio product. .
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