User:Vinayr rao/Wideband Codec

Wideband Codecs in Digital telephony refers to the use of higher sampling rates than Narrowband Codecs or the utilization of embedded sub-band coding information to effectively increase the bandwidth of the baseband voice, from the traditional 200 Hz to 3.5 kHz used in Narrowband Codecs, to 50 Hz at the low end and anywhere from 7kHz to 22 kHz. at the high end, depending on the type of codec used. This results in a significant improvement in voice quality.

Introduction:
Traditional analog telephone systems were designed to use a bandwidth of approximately 200 Hz to 3300 Hz as this was considered adequate for transmitting human voice intelligibly over a telephone system. When digital telephone systems replaced the earlier analog telephone systems, voice was sampled at a rate of 8 kHz with an 8-bit depth. According to the Nyquist-Shannon sampling theorem, choosing an 8 kHz sampling rate, would be sufficient to perfectly reproduce baseband information of up to 4 kHz. Hence an 8 kHz sampling rate was deemed sufficient to digitize voice for telephony.

The ubiquitous PCM codec, the venerable ITU-T G.711 samples at the rate of 8 kHz and after companding (A-Law or u-law) encodes at 8-bits to produce a bit stream of 64 kbps. The use of 8-bit depth would normally yield a theoretical Signal to Quantization Noise Ratio (SQNR) of 49.93 dB for a pure sine-wave. However, with the use of companding (A-Law or u-law) it is possible to encode the 13-bit or 14-bit signed linear PCM samples respectively into logarithmic 8-bit samples. This achieves a better SQNR albeit at the expense of higher Total Harmonic Distortion (THD). Even so the voice quality of the G.711 PCM codec was deemed to be "toll quality" and was considered to be highest quality in telephone systems. During the early days of internet, the 64 kbps bit rate was often too high to use and the available bandwidths necessitated the use of compression techniques which further degraded voice quality.

Modern digital telecommunications systems are no longer limited by the constraints of the older systems. The advent of Broadband Access Networks and end-to-end Digital Networks such as 2G and 3G Wireless Networks have allowed the development of technologies for the next-generation of telephony. One such development is the wideband codec. Wideband Codecs typically use one or more of the following methods to encode so called "Wideband voice".

Increased Sampling Rate

Doubling the sampling rate to 16 kHz, allows sampling of signals as high as 8 kHz. This allows the wideband codec to transmit consonants, sibilants and other subtleties of the human voice formerly lost or clipped by narrowband codecs and significantly adds to the intelligibility and quality of the speech signal.

Sub-Band coding

Sub-Band Coding breaks the baseband signal into two or more frequency bands and encodes each one independently. Many wideband codecs make use of this method as it becomes easier to preserve backward compatibility.

Increased Bit-depth (Quantization)

With the advent of Broadband Access Networks, it is possible to carry high bandwidth data. It is now possible to use 16-bit depth in wideband codecs as the resulting high bitrates can still be carried comfortably over a broadband link. The use of 16-bit depth increases the theoretical SQNR for a pure sine-wave to 98.09 dB. Audio CD's are also typically recorded with 16-bit depth and so wideband codecs may sometimes use the term "CD-quality Voice", to describe their speech quality although they use a much lower sampling rate than CD's.

Voice Quality
Voice Quality is highly subjective and dependent upon a number of factors. consonant

sibilant

Dynamic Range

companding

A-Law

μ-law

Sampling rate

Quantization

Intelligibility

Wireline
ITU-T G.711.1 ITU-T G.722 ITU-T G.722.2

Wireless
Cellular Mobile Wireless

AMR-WB

VMR-WB

EVRC-C

Cordless Phone

DECT/CAT-iq 711 HD Siren 7

Testing Wideband Voice
PESQ Extensions to test Wideband Telephone Networks and Codecs ITU-T P.862.2