Pulse Code PCM
Pulse Code PCM
PCM is used with T-1 and T-3 carrier systems. These carrier systems combine the PCM
signals from many lines and transmit them over a single cable or other medium.
PCM is also the usual digital method used for music audio playback of music CDs. While
supported by DVDs, DVDs have a greater volume so they use Linear PCM, which has a
higher sampling rate — up to 24-bit at a sampling rate of 96 kHz.
by Alec Reeves in 1937. It is the standard form for digital audio in computers and various Compact
representation of an analog signal, in where the magnitude of the analogue signal is sampled regularly at uniform
intervals, with each sample being quantized to the nearest value within a range of digital steps.
PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate,
which is the number of times per second that samples are taken; and the bit-depth, which determines the number of
[edit]Modulation
sampled at regular intervals, shown as ticks on the x-axis. For each sample, one of the available values (ticks on the
y-axis) is chosen by some algorithm (in this case, the floor function is used). This produces a fully discrete
representation of the input signal (shaded area) that can be easily encoded as digital data for storage or
manipulation. For the sine wave example at right, we can verify that the quantized values at the sampling moments
are 7, 9, 11, 12, 13, 14, 14, 15, 15, 15, 14, etc. Encoding these values as binary numbers would result in the following
set ofnibbles: 0111, 1001, 1011, 1100, 1101, 1110, 1110, 1111, 1111, 1111, 1110, etc. These digital values could
then be further processed or analyzed by a purpose-specific digital signal processor or general purpose DSP. Several
Pulse Code Modulation streams could also bemultiplexed into a larger aggregate data stream, generally for
transmission of multiple streams over a single physical link. One technique is called time-division multiplexing, or
TDM, and is widely used, notably in the modern public telephone system. Another technique is called Frequency-
division multiplexing, where the signal is assigned a frequency in a spectrum, and transmitted along with other signals
inside that spectrum. Currently, TDM is much more widely used than FDM because of its natural compatibility with
There are many ways to implement a real device that performs this task. In real systems, such a device is commonly
implemented on a single integrated circuit that lacks only the clock necessary for sampling, and is generally referred
to as an ADC (Analog-to-Digital converter). These devices will produce on their output a binary representation of the
input whenever they are triggered by a clock signal, which would then be read by a processor of some sort.
[edit]Demodulation
To produce output from the sampled data, the procedure of modulation is applied in reverse. After each sampling
period has passed, the next value is read and a signal is shifted to the new value. As a result of these transitions, the
signal will have a significant amount of high-frequency energy. To smooth out the signal and remove these
undesirable aliasing frequencies, the signal would be passed through analog filters that suppress energy outside the
expected frequency range (that is, greater than the Nyquist frequency fs / 2). Some systems usedigital filtering to
remove some of the aliasing, converting the signal from digital to analog at a higher sample rate such that the analog
filter required for anti-aliasing is much simpler. In some systems, no explicit filtering is done at all; as it's impossible
for any system to reproduce a signal with infinite bandwidth, inherent losses in the system compensate for the
artifacts — or the system simply does not require much precision. The sampling theorem suggests that practical PCM
devices, provided a sampling frequency that is sufficiently greater than that of the input signal, can operate without
The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for
generating the digital signal. These devices are DACs (digital-to-analog converters), and operate similarly to ADCs.
They produce on their output a voltage or current (depending on type) that represents the value presented on their
inputs. This output would then generally be filtered and amplified for use.
[edit]Limitations
Choosing a discrete value near the analog signal for each sample (quantization error)
The quantization error swings between to . In the ideal case (with a fully linear ADC) it is equally distributed
Between samples no measurement of the signal is made; due to the sampling theorem this results in any
frequency above or equal to (fs being the sampling frequency) being distorted or lost completely
As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or
decoding clock is not stable, its frequency drift will directly affect the output quality of the device. A slight difference
between the encoding and decoding clock frequencies is not generally a major concern; a small constant error is not
noticeable. Clock error does become a major issue if the clock is not stable, however. A drifting clock, even with a
relatively small error, will cause very obvious distortions in audio and video signals, for example.
Extra information: PCM data from a master with a clock frequency that can not be influenced requires an exact clock
at the decoding side to ensure that all the data is used in a continuous stream without buffer underrun or buffer
overflow. Any frequency difference will be audible at the output since the number of samples per time interval can not
be correct. The data speed in a compact disk can be steered by means of a servo that controls the rotation speed of
the disk; here the output clock is the master clock. For all "external master" systems like DAB the output stream must
be decoded with a regenerated and exact synchronous clock. When the wanted output sample rate differs from the
incoming data stream clock then a sample rate converter must be inserted in the chain to convert the samples to the
In conventional PCM, the analog signal may be processed (e.g. by amplitude compression) before being digitized.
Once the signal is digitized, the PCM signal is usually subjected to further processing (e.g. digital data compression).
in the analog domain as part of the A/D process; newer implementations do so in the digital domain. These simple
DPCM encodes the PCM values as differences between the current and the predicted value. An algorithm
predicts the next sample based on the previous samples, and the encoder stores only the difference between
this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same
information. For audio, this type of encoding reduces the number of bits required per sample by about 25%
compared to PCM.
Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further
Delta modulation is a form of DPCM which uses one bit per sample.
In telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of 8 bits
each, giving a 64 kbit/s digital signal known as DS0. The defaultsignal compression encoding on a DS0 is either 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 where a 12 or 13-bit linear PCM sample number is mapped into an 8-bit value. This
system is described by international standard G.711. An alternative proposal for a floating point representation, with
Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice
signal even further. An ADPCM algorithm is 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.
Later it was found that even further compression was possible and additional standards were published. Some of
these international standards describe systems and ideas which are covered by privately owned patents and thus use
synchronized using in-band information, there must not be long sequences of identical symbols, such as ones or
Ones-density is often controlled using precoding techniques such as Run Length Limited encoding, where the PCM
code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into the
channel. In other cases, extra framing bits are added into the stream which guarantee at least occasional symbol
transitions.
Another technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend
to turn the raw data stream into a stream that looks pseudo-random, but where the raw stream can be recovered
exactly by reversing the effect of the polynomial. In this case, long runs of zeroes or ones are still possible on the
output, but are considered unlikely enough to be within normal engineering tolerance.
In other cases, the long term DC value of the modulated signal is important, as building up a DC offset will tend to
bias detector circuits out of their operating range. In this case special measures are taken to keep a count of the
cumulative DC offset, and to modify the codes if necessary to make the DC offset always tend back to zero.
Many of these codes are bipolar codes, where the pulses can be positive, negative or absent. In the typical alternate
mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to
standard
Linear pulse code modulation (LPCM) is a method of encoding audio information digitally. The term also refers
collectively to formats using this method of encoding. The term Pulse-code modulation (PCM), though strictly more
Contents
[hide]
1 Description
2 Implementations
rates
4 DVD standards
5 See also
6 References
7 External links
[edit]Description
LPCM is a particular method of pulse code modulation which represents an audio waveform as a sequence of
LPCM represents sample amplitudes on a linear scale.[6] LPCM specifies that the values stored are proportional to
the amplitudes, rather than representing say the logarithm of the amplitude (e.g. companding - A-law/u-law), or being
related in some other manner (e.g. DPCM or ADPCM).[6] In practice these values will be quantized.
LPCM audio is coded using a combination of various parameters - such as resolution/sample size (e.g. 8, 16, 20, 24
bit, etc), frequency/sample rate (e.g. 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000 Hz / "samples per
second", etc), sign (signed or unsigned), number of channels (monaural, stereo, quadrophonic, etc) and interleaving
of channels, byte order (little endian, big endian).[7] If the sample is 16-bit signed, the sample range is from -32768 to
32767, with a centerpoint of 0.[8] (For example, signed LPCM data is used on Audio CD, DVD Video, 16-bit LPCM in
WAV, audio/L16, etc.) If the sample is 16-bit unsigned, the sample range is from 0 to 65535, with a centerpoint of
32768.[7]
[edit]Implementations
LPCM is the method of encoding generally used for uncompressed audio, although there are other methods such
1982).
On PCs, the term PCM and LPCM often refer to the format used in WAV (defined in 1991) and AIFF audio
container formats (defined in 1988). LPCM data may also be stored in other formats such as AU, raw audio
LPCM has been defined as a part of the DVD (since 1995) and Blu-ray (since 2006) standards.[9][10][11] It is
also defined as a part of various digital video and audio storage formats (e.g. DV since 1995[12], AVCHD since
2006[13]).
Linear pulse code modulation is used by HDMI (defined in 2002), a single-cable digital audio/video
RF64 container format (defined in 2007) uses LPCM and also allows non-PCM bitstream storage: various
compression formats contained in the RF64 file as data bursts (Dolby E, Dolby AC3, DTS, MPEG-1/MPEG-2
LPCM encodes a single sound channel. Support for multichannel audio depends on file format and relies on
interweaving or synchronization of LPCM streams.[16][17] While two channels (stereo) is the most common format,
Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in compact discs.
Sampling frequencies of 96 kHz or 192 kHz can be used on some newer equipment, with the higher value equating to
6.144 megabit per second for two channels at 16-bit per sample value. The bitrate limit for LPCM audio on DVD-
Video is also 6.144 Mbit/s, allowing 8 channels (7.1 surround) x 48 kHz x 16-bit per sample = 6144 kbit/s.
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Pulse-Code Modulation
Figure 2-49 shows the relationship between decimal numbers, binary numbers,
and a pulse-code waveform that represents the numbers. The table is for a 16-
level code; that is, 16 standard values of a quantized wave could be
represented by these pulse groups. Only the presence or absence of the pulses
are important. The next step up would be a 32-level code, with each decimal
number represented by a series of five binary digits, rather than the four
digits of figure 2-49. Six-digit groups would provide a 64-level code, seven
digits a 128-level code, and so forth. Figure 2-50 shows the application of
pulse-coded groups to the standard values of a quantized wave.
The system does, of course, have some distortion introduced by quantizing the
signal. Both the standard values selected and the sampling interval tend to
make the reconstructed wave depart from the original. This distortion, called
QUANTIZING NOISE, is initially introduced at the quantizing and coding
modulator and remains fixed throughout the transmission and retransmission
processes. Its magnitude can be reduced by making the standard quantizing
levels closer together. The relationship of the quantizing noise to the number
of digits in the binary code is given by the following standard relationship:
Where:
Thus, with the 4-digit code of figure 2-50 and 2-51, the quantizing noise will
be about 35 dB weaker than the peak signal which the channel will accommodate.
Older DVD players only support 48 kHz/16-bit capability. Recent players have built-in 96 kHz/24-bit capabilities.