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In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.
SBC, or low-complexity subband codec, is an audio subband codec specified by the Bluetooth Special Interest Group (SIG) for the Advanced Audio Distribution Profile (A2DP). [1] SBC is a digital audio encoder and decoder used to transfer data to Bluetooth audio output devices like headphones or loudspeakers. It can also be used on the Internet. [2]
Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream. [1]
The coding of temporal information in the auditory nerve can be disrupted by two main mechanisms: reduced synchrony and loss of synapses and/or auditory nerve fibers. [186] The impact of disrupted temporal coding on human auditory perception has been explored using physiologically inspired signal-processing tools.
G.722 [4] is an ITU-T standard wideband speech codec operating at 48, 56 and 64 kbit/s, based on subband coding with two channels and ADPCM coding of each. [5] Before the digitization process, it catches the analog signal and divides it in frequency bands with quadrature mirror filters (QMF) to get two subbands of the signal.
For very low-power systems, such as mobile phones, signal strength is usually expressed in dB-microvolts per metre (dBμV/m) or in decibels above a reference level of one milliwatt . In broadcasting terminology, 1 mV/m is 1000 μV/m or 60 dBμ (often written dBu). Examples. 100 dBμ or 100 mV/m: blanketing interference may occur on some receivers
Time-domain harmonic scaling (TDHS) is a method for time-scale modification of speech (or other audio signals), [1] allowing the apparent rate of speech articulation to be changed without affecting the pitch-contour and the time-evolution of the formant structure. [2]
The codec also incorporates an alternate coding mode, with a minimum bit rate of 12.65 kbit/s, which is bitstream interoperable with ITU-T Recommendation G.722.2, 3GPP AMR-WB and 3GPP2 VMR-WB mobile wideband speech coding standards. This option replaces Layer 1 and Layer 2, and the layers 3-5 are similar to the default option with the exception ...