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Coder-decoder
- this page is a stub
A CODEC is a device or computer program which encodes and decodes an analogue
signal into a digital data stream and vice versa.
The expression CODEC is a portmanteau 1 of coder/decoder.
The coder part is typically used to convert sound or images from the analogue
world to the digital domain, and is also known as a digitizer. In the case of
speech, it is also known as a VOCODER. 2
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A combination of two (parts of) words into one new word.
➤ Wikipedia
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VOCODER = voice-coder.
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There are several techniques for digitising audio signals, for example:
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- Pulse Code Modulation (PCM)
PCM is a generic expression for digitizing an analog signal. A PCM
stream is the numerical or digital representation of the analog
signal. Sending PCM data typically requires twice the bandwidth of the
original analog signal, but the quality is unsurpassed.
➤ More
- Delta modulation (DM)
This technique uses only 1 bit/sample, as it only registers the
difference between the current sample and the previous one
(1 = higher, 0 = lower). This means that the signal can only
increment or decrement in single discrete steps.
A higher sample rate will result in a closer approximation of the original
sound wave and, hence, a better quality.
➤ More
- Continuous Variable Slope Delta modulation (CVSD)
This is an improved version of Delta Modulation (DM) that gives a better
approximation of the original sound waves at lower bit rates.
It has a typical sample rate between 8 and 16 kHz, which results in
8 to 16 kbps data. Examples of equipment that uses CVSD are the
Philips Spendex 10,
the Spendex 50 and
the American KY-68.
➤ More
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Before speech can be encrypted, is has to be converted from the analogue to the
digital domain, by means of a sampler, or digitizer, or vocoder.
Generally speaking, a digital signal needs more bandwidth than its analogue
equivalent (typically twice the bandwidth), but methods have been developed
to reduce this, by analysing the properties of speech and
sending these as parameters to the other end, where they are used to
reconstruct, or synthesize, the original voice again.
This method is known as a Vocoder and is not always good enough to recognise a person's voice. The first vocoder,
named VODER, was developed
at Bell Labs in 1939. Its principle was first used during WWII on the
transatlantic
SIGSALY cryptographic telephone.
A speech analyser/synthesizer is also known as a CODEC (coder-decoder).
Here are some examples of speech digitisers:
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- Formant analysis
This method is based on the way the human voice is generated in the
vocal tract (throat and mouth) and is therefore only suitable for
speech encoding.
It analyses the properties of speech – such as pitch, voiced
sounds, voiceless sounds (hiss) and formants – and stores these properties
as parameters that can be used by the decoder to reconstruct (synthesize)
an approximation of the original sound.
- LPC - Linear Predictive Coding
Early vocoder for narrow bandwidth connections. LPC-10 has a sampling
rate of 8 kHz and a coding rate of 2.4 kbps. Developed by the NSA and
used in the
first generation secure terminal units (STU).
Later versions have a coding rate of 800 bps.
➤ More
- RELP - Residual-Excited Linear Prediction
Improved (but now obsolete) variant of LPC, and predecessor of CELP.
➤ Wikipedia
- CELP - Code-Excited Linear Prediction
Improved variant of LPC and RELP that provides better speech quality at
lower bitrates. CELP exists in many variants and is also used in MPEG-4
audio coding. It is the most widely used speech coding algorithm today.
➤ Wikipedia
- MELP - Mixed-Excitation Linear Prediction
Medium quality vocoder, mainly used by the US Department of Defense for
secure communication via satellites and and military radios. It has a sampling
rate of 8 kHz and a coding rate of 2.4 kbps (2400 bps).
➤ Wikipedia
- MELPe - Enhanced Mixed-Excitation Linear Prediction
High-quality low-bitrate enhanced version of MELP with a sampling rate
of 8 kHz and a coding rate of 2400, 1200 or 600 bps.
➤ Wikipedia
- MRELP - Modified Residually-Excited Linear Prediction
Improved variant of MELP and MELPe that produces better results at higher
bitrates, for example at 9600 bps.
- MBE
Vocoder which uses the multi-band representation of speech, patented in 1997
by Digital Voice Systems, Inc. (DVSI) and based on the pioneering work of
Osamu Fujimura at MIT (1967) [4]. The patent has meanwhile expired.
- IMBE
Improved version of the MBE codec (see above), developed by
Digital Voice Systems, Inc. (DVSI).
Used in the Motorola XTL and XTS series Phase 1 radios.
The patent has meanwhile expired.
- AMBE
Advanced version of MBE, featuring a
high-quality and low-data-rate codebook-based vocoder, developed by
Digital Voice Systems, Inc. (DVSI). Operates at bitrates from 2 kb/s to
9.6 kb/s and sampling rates of 8 kHz in 20 ms frames. It allows
Forward Error Correction (FEC) with up to 7 b/s of correction data.
The total RF bandwidth is 2250 Hz (vs 2700-3000 Hz for an equivalent
analogue SSB channel).
- AMBE+
Improved version of AMBE with better speech quality.
Used, for example, in Icom's D-Star standard.
- AMBE+2
More advanced iteration of AMBE+, developed by DVSI and used in a
variety of products like APCO Project 25 (P25), DMR, dPMR and
Yaesu Fusion. It features drastically improved speech quality and
background noise suppression. Backward compatible with AMBE(+) and IMBE.
AMBE+2 technology can be licenced from DVSI in the form of DSP libraries,
CPU libraries and dedicated chips.
➤ Developer
- Codec2
Open-source initiative for a high quality codec that uses half the
bandwidth of AMBE to achieve similar speech quality, created by
David Rowe and Bruce Perens.
➤ Wikipedia
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© Crypto Museum. Created: Thursday 09 June 2022. Last changed: Sunday, 07 June 2026 - 09:20 CET.
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