27-02-2015, 02:28 AM
I am looking for matlab code on adaptive differrential pulse code modlation.
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matlab code for adaptive differential pulse code modulation
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I am looking for matlab code on adaptive differrential pulse code modlation.
04-04-2017, 11:49 AM
Adaptive Differential Pulse Code (ADPCM) modulation is a variant of differential pulse code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the data bandwidth required for a Signal to noise ratio. Typically, the adaptation to signal statistics in ADPCM is simply an adaptive scaling factor before quantifying the difference in the DPCM encoder. ADPCM was developed in the early 1970s at Bell Labs for voice coding, by P. Cummiskey, N. Jayant and James L. Flanagan.
In telephony, a standard audio signal for a single telephone call is coded as 8000 analog samples per second, 8 bits each, giving a digital signal of 64 kbit / s known as DS0. The default compression encoding for a DS0 is μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems in which a 13 or 14-bit linear PCM sample number is assigned to an 8-bit value. This system is described by the international standard G.711. When the costs of the circuit are high and the loss of voice quality is acceptable, sometimes it makes sense to compress the voice signal even more. An ADPCM algorithm was used to map a series of 8-bit μ-law (or a law) PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard.
Adaptive differential pulse code modulation is a very efficient digital coding of waveforms that was developed by Bell Labs in the 1970s for the purpose of voice coding. ADPCM was also used in the early 1990s by the Interactive Multimedia Association (IMA) for the development of the legacy audio codec - also known as ADPCM DVI, IMA ADPCM or DVI4. Some ADPCM methods are also used in VoIP communications. The concept of ADPCM is to use the past behavior of a signal to forecast it in the future. The resulting signal will represent the error of the prediction, which does not matter. Therefore, the signal must be decoded to reconstruct a more significant original waveform.
The ADPCM technique is used to send sound signals over long distance fiber optic lines. This is especially useful for organizations that establish digital lines between remote sites to transmit both voice and data. Voice signals are digitized before being broadcast. In the field of telecommunications, the ADPCM technique is mainly used in speech compression because the method makes it possible to reduce the bit stream without compromising quality. The ADPCM method can be applied to all waveforms, high-quality audio, images and other modern data.
In telephony, a standard audio signal for a single telephone call is coded as 8000 analog samples per second, 8 bits each, giving a digital signal of 64 kbit / s known as DS0. The default compression encoding for a DS0 is μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems in which a 13 or 14-bit linear PCM sample number is assigned to an 8-bit value. This system is described by the international standard G.711. When the costs of the circuit are high and the loss of voice quality is acceptable, sometimes it makes sense to compress the voice signal even more. An ADPCM algorithm was used to map a series of 8-bit μ-law (or a law) PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard.
Adaptive differential pulse code modulation is a very efficient digital coding of waveforms that was developed by Bell Labs in the 1970s for the purpose of voice coding. ADPCM was also used in the early 1990s by the Interactive Multimedia Association (IMA) for the development of the legacy audio codec - also known as ADPCM DVI, IMA ADPCM or DVI4. Some ADPCM methods are also used in VoIP communications. The concept of ADPCM is to use the past behavior of a signal to forecast it in the future. The resulting signal will represent the error of the prediction, which does not matter. Therefore, the signal must be decoded to reconstruct a more significant original waveform.
The ADPCM technique is used to send sound signals over long distance fiber optic lines. This is especially useful for organizations that establish digital lines between remote sites to transmit both voice and data. Voice signals are digitized before being broadcast. In the field of telecommunications, the ADPCM technique is mainly used in speech compression because the method makes it possible to reduce the bit stream without compromising quality. The ADPCM method can be applied to all waveforms, high-quality audio, images and other modern data.
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