the mu-law method of quantization that is used in
the digital telecommunication systems of North America and
Japan. This method of quantization was adopted as the a-law
algorithm for use in Europe and much of the rest of the world.
Following the idea of allowing smaller step functions at lower
amplitudes rather than at higher amplitudes, the mu-law and
a-law provide a quasi-logarithmic scale. The voltage range has
16 segments (0 to 7 positive and 0 to 7 negative). Each segment
has 16 steps for a total of 256 points on the scale. Starting
with segment 0, which is closest to zero amplitude, the
segments grow bigger toward the maximum amplitudes and the size
of the steps increases. Within a segment, however, the size of
the steps is linear. The result of using mu-law and a-law is a
more accurate value for smaller amplitudes and a uniform
signal-to-noise quantization ratio (SQR) across the input
range. The ITU Telecommunication Standardization Sector (ITU-T)
standards for companding include the a-law and mu-law in
recommendation G.711. Note
By convention, when the
PSTN communicates between a mu-law country and an a-law
country, the mu-law country must change its signaling to
accommodate the a-law country.
Content 2.2
Digitizing and Packetizing Voice 2.2.7 Common
Voice Codec Characteristics Data compression squeezes data
so that the data requires less bandwidth on data transmission
channels. Most compression schemes take advantage of the fact
that datastreams have a lot of repetition. For example, while a
7-bit ASCII code represents alphanumeric characters, a
compression scheme can use a 3-bit code to represent the eight
most common letters. In voice, there are long stretches of
silence that can be replaced by a value that indicates how much
silence there is, or how long the silence exists. Similarly, in
graphic compression techniques, a value can replace white
spaces in an image by indicating the amount of white space that
is replaced. Early Plain Old Telephone Service (POTS) worked
wholly on an analog infrastructure. Long distance calling was
challenging primarily because of signal attenuation and line
noise. Periodic amplification solved problems to some extent
but also amplified the noise. When telephone companies
converted their trunk lines to digital and used pulse code
modulation (PCM) to digitize the signals, these problems
virtually disappeared. The basic PCM technique is to use a
coder-decoder (codec) to sample the amplitude of a voice signal
8000 times per second and then store the amplitude value as 8
bits of data. There is a formula to this storage procedure:
8000 samples/second × 8 bits/sample = 64,000 bits/sec The
result is the basis for the entire telephone system digital
hierarchy. Differential (or Delta) pulse-code modulation (DPCM)
encodes the PCM values as differences between the current and
the previous value. For audio, this type of encoding reduces
the number of bits required per sample by about 25 percent
compared to PCM. Adaptive DPCM (ADPCM) is a variant of DPCM
that varies the size of the quantization step to allow further
reduction of the required bandwidth for a given signal-to-noise
ratio. To a large extent, ADPCM has replaced PCM. ADPCM uses a
special encoding technique that reduces the data that is
required to store each sample, transmitting only the difference
between one sample and the next. An adaptive predictor
algorithm predicts in advance how the signal will change. The
prediction is usually very accurate. As samples vary, the
predictor rapidly adapts to the changes. ADPCM provides 48
voice channels on a T1 line, which benefits customers who use
such lines to interconnect their remote offices or connect
their internal phone systems to the phone company switches. The
table in Figure shows the most popular coding techniques by
their bit rates that are standardized for telephony by the
ITU-T in its G-series recommendations: - G.711:
Describes the 64-kbit/sec PCM voice coding technique. In G.711,
encoded voice is already in the correct format for digital
voice delivery in the PSTN or through PBXs. The ITU-T
standardized the earlier implementation of the mu-law and a-law
under this one recommendation.
- G.726:
Describes ADPCM coding at 40, 32, 24, and 16 kbps.
ADPCM-encoded voice can be interchanged between packet voice,
PSTN, and PBX networks if the PBX networks are configured to
support ADPCM.
- G.728: Describes a 16-kbps
low-delay variation of code excited linear prediction (CELP)
voice compression. CELP voice coding must be translated into a
public telephony format for delivery to or through the
PSTN.
- G.729: Describes CELP compression that
codes voice into an 8-kbps streams, or conjugate structure
algebraic code excited linear prediction (CS-ACELP).
-
G.729A: Describes audio data compression algorithm for
voice that compresses voice audio in sections of 10 ms. G.729A
is compatible with G.729 (which also uses CS-ACELP), but G.729A
requires less computation. This lower complexity does have the
disadvantage of marginally worsening speech quality. G.729 and
G.729 Annex A differ mainly in computational complexity; both
provide speech quality similar to 32-kbps ADPCM.
Content 2.2 Digitizing and Packetizing
Voice 2.2.8 Selecting a Codec Using the Mean
Opinion Score The most important characteristics of a codec
are the required bandwidth for the procedure and the quality
degradation caused by the analog-to-digital conversion and
compression. The mean opinion score (MOS) is a system of
grading the voice quality of the telephone connection that uses
codec. The MOS is a statistical measurement of voice quality
that is based on the judgment of several subscribers. Because
MOS is graded by humans, the score is subjective. MOS uses a
scale of 1 (bad) to 5 (excellent). An MOS of 5 represents
direct conversation. The table in Figure provides a description
of the five ratings. The table in Figure shows the MOS for the
most popular coding techniques. Note
A newer, more
objective measurement is quickly overtaking MOS scores as the
industry quality measurement of choice for coding algorithms.
Perceptual Speech Quality Measurement (PSQM), as per ITU
standard P.861, provides a rating on a scale of 0 to 6.5, where
0 is best and 6.5 is worst. Many vendors implement PSQM in test
equipment and monitoring systems. Some PSQM test equipment
converts the 0-to-6.5 scale to a 0-to-5 scale to correlate to
MOS. PSQM works by comparing the transmitted speech to the
original input and yielding a score. Test equipment from
various vendors is now capable of providing a PSQM score for a
test voice call over a particular packet network.
Content
2.2 Digitizing and Packetizing Voice
2.2.9 A Closer Look at a DSP Figure shows a typical
DSP module that might be used in a Cisco voice-enabled router.
A digital signal processor (DSP) is a specialized processor
used for telephony applications: - Voice
termination: DSPs terminate calls to the gateway that have
been received from or placed to traditional voice interfaces.
These interfaces can be digital or analog interfaces. For
example, when an analog telephone places a call to the PSTN
(over a digital trunk) or to a VoIP device, a DSP resource is
used to accommodate this call. The DSP converts the analog
signal to digital (and vice versa) and provides echo
cancellation, compression, voice activity detection (VAD),
comfort noise generation (CNG), jitter removal, and other
similar functions.
- Conferencing: In audio
conferencing, DSPs mix voice streams from multiple participants
into a single conference call stream. All participants send
their audio to the conference bridge (that is, the DSP), where
the streams are mixed and then played back to all
participants.
- Transcoding: DSP takes a voice
stream of one codec type and converts it to another codec type.
For example, transcoding takes a voice stream from a G.711
codec and transcodes the stream in real time to a G.729 codec