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Practical Audio Interfacing
Published in Francis Rumsey, John Watkinson, Digital Interface Handbook, 2013
Francis Rumsey, John Watkinson
Unlike analog audio, digital audio has a discrete-time structure, because it is a sampled signal in which the samples may be further grouped into frames and blocks having a certain time duration. If digital audio devices are to communicate with each other, or if digital signals are to be combined in any way, then they need to be synchronized to a common reference in order that the sampling frequencies of the devices are identical and do not drift with relation to each other. It is not enough for two devices to be running at nominally the same sampling frequency (say both at 44.1 kHz). Between the sampling clocks of professional audio equipment it is possible for differences in frequency of up to ±10 parts per million (ppm) to exist and even a very slow drift means that two devices are not truly synchronous. Consumer devices can exhibit an even greater range of sampling frequencies that are nominally the same.
The Mastering Engineer
Published in Evren Göknar, Major Label Mastering, 2020
Bit depth determines dynamic range in decibels (dB) of sampled digital audio at a resolution of 6dB per bit; 16bit digital audio has a dynamic range of 96dB, and 24bit digital audio has a dynamic range of 144dB. There is also a difference between 16bit and 24bit in other subjective audible qualities, namely depth, dimension, and richness of the audio image; 24bit represents the limit of DA conversion, so although many other bit depths are used in host DAW internal processing or storage (32bit and 64bit floating-point are common), the audio remains at 24bit resolution.
Digital Audio and Video
Published in Skip Pizzi, Graham A. Jones, A Broadcast Engineering Tutorial for Non-Engineers, 2014
Five different standard sampling rates are typically used for digital audio in studios: 32,000 samples per second, 44,100 samples per second, 48,000 samples per second, 96,000 samples per second, and 192,000 samples per second. Usually these rates are referred to simply as 32 kHz, 44.1 kHz, 48 kHz, 96 kHz, and 192 kHz, respectively. Audio compact discs (CDs) use a sampling rate of 44.1 kHz, while most broadcast digital audio studio equipment uses 48 kHz sampling or higher rates.
The Changing Face of Public Broadcasting in India
Published in IETE Journal of Education, 2023
DRM [1] is a revolutionary technique, which can provide high quality sound from standard AM channels, and can cope up with fading and ambient noise. India has started expanding its radio network with DRM. It uses coded orthogonal frequency division multiplexing (COFDM) with quadrature amplitude modulation to transmit digitized and data compressed audio signals. The process of data compression and coding is defined as per DRM30 standard, and the bandwidth requirement is substantially reduced. The digital audio signal modulates a subcarrier of 12 KHz. This modulated subcarrier is linearly added to the primary audio, which has been band limited to 5 KHz. This multiplex, in turn, is used to amplitude modulate the carrier frequency in the MW or SW wave band. Thus a single carrier at RF can transmit both AM double sideband (AM DSB) as well DRM subcarrier on a single channel [2]. The reception of DRM signals requires radio receivers specifically designed for decoding digitally modulated signals. At the receiving end, a 12 KHz band pass filter is used to separate the DRM signal, and a 5 KHz low pass filter to recover the regular AM signal, post demodulation. The DRM section would require a set of filters, and use digital signal processing (DSP). It has now become possible to implement DSP in a small space inside a receiver. Block diagram of a typical DRM receiver has been given in Figure 1.
Detectability of auditory warning signals in the ambient noise of Dutch train cabins
Published in Ergonomics, 2021
Hanneke E.M. van der Hoek-Snieders, Rolph Houben, Wouter A. Dreschler
The acoustical data were collected by sound recordings on digital tape. The on-site measurement set-up consisted of a calibrated sound level meter (B&K 2260 SLM with calibrator B&K 4230) connected to a portable Digital Audio Tape (DAT)-recorder (Tascam DAP). Prior, during, and after the on-site measurements, the recording system was calibrated and checked with a B&K Sound Calibrator Type 4231. The level of the calibration tone was recorded on the Tascam DAT recorder in the same way as the real measurements were made. This recorded calibration-tone was then used to determine the correct level of the DAT recordings during the off-line analysis. The acoustical data were digitally transferred to a computer that was connected to an Echo Gina 24/96 sound card. A-weighting and octave band filtering were applied in compliance with respectively IEC 61260 Class 1 and IEC 61260 (Couvreur C. Octave 1997). The DSD and ATP signals were measured in all trains. If adjustable, the volume setting of the warning signal was set at maximum. The DSD signal was measured in quiet. The ATP signal does not occur in quiet and was therefore measured at the lowest speed at which the signal occurs. Unlike the DSD, the ATP signal decays over time. The ATP recordings were therefore averaged over the first 200 milliseconds after onset. This duration roughly corresponds to the human integration time for tonal signals (Viemeister 1996).
On the relationship between land use and sound sources in the urban environment
Published in Journal of Urban Design, 2020
Efstathios Margaritis, Jian Kang, Francesco Aletta, Östen Axelsson
For the on-site audio recordings, the equipment included a stereo microphone kit (DPA 4060) connected to a digital audio recorder (R-44 Edirol), a mini microphone (Micw i436), and a sound calibrator (CAL21). The ‘Audiotool’ Android application, installed on a mobile phone with the Micw i436 attached, was used to record the sound pressure levels at each location. The selection of the equipment described above aimed at showing that reasonably representative physical measurements can nowadays be obtained with affordable hardware and software. However, the designer should not have to conduct empirical investigations of this sort. Rather the long-term objective of this line of research is to develop a more general model for soundscape planning and design that would guide design decisions.