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Sample-rate conversion prevents changes in speed and pitch that would otherwise occur when transferring recorded material between such systems. More specific types of resampling include: upsampling or upscaling; downsampling, downscaling, or decimation; and interpolation. The term multi-rate digital signal processing is sometimes used to refer ...
The audio sample rate is fixed at 8 kHz. The reference implementation is open source and is freely available in a GitHub repository. [ 3 ] The source code is released under the terms of version 2.1 of the GNU Lesser General Public License (LGPL). [ 4 ]
G.729 is a royalty-free [1] narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP), and was introduced in 1996. [2]
A typical choice of characteristic frequency is the sampling rate that is used to create the digital signal from a continuous one. The normalized quantity, f ′ = f f s , {\displaystyle f'={\tfrac {f}{f_{s}}},} has the unit cycle per sample regardless of whether the original signal is a function of time or distance.
Users can customize the output by altering the bit rate, maximum file size or bit rate and sample rate via "constant quality". [9] HandBrake supports adaptive deinterlacing, scaling, detelecine, and cropping, both automatic and manual. [2]
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For most phonemes, almost all of the energy is contained in the 100 Hz – 4 kHz range, allowing a sampling rate of 8 kHz. This is the sampling rate used by nearly all telephony systems, which use the G.711 sampling and quantization specifications. [citation needed]
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