Editing Vocoder (section)

==Modern implementations==
{{See also|Speech codec|Audio codec}}

Even with the need to record several frequencies, and additional unvoiced sounds, the compression of vocoder systems is impressive. Standard speech-recording systems capture frequencies from about 500 to 3,400&nbsp;Hz, where most of the frequencies used in speech lie, typically using a sampling rate of 8&nbsp;kHz (slightly greater than the [[Nyquist rate]]). The sampling resolution is typically 8 or more bits per sample resolution, for a data rate in the range of {{nowrap|64 kbit/s}}, but a good vocoder can provide a reasonably good simulation of voice with as little as {{nowrap|5 kbit/s}} of data.

''Toll quality'' voice coders, such as ITU [[G.729]], are used in many telephone networks. G.729 in particular has a final data rate of {{nowrap|8 kbit/s}} with superb voice quality. G.723 achieves slightly worse quality at data rates of 5.3 and {{nowrap|6.4 kbit/s}}. Many voice vocoder systems use lower data rates, but below {{nowrap|5 kbit/s}} voice quality begins to drop rapidly.{{citation needed|date=January 2023}}

Several vocoder systems are used in [[NSA encryption systems]]:
* LPC-10, [[Federal Information Processing Standard|FIPS]] Pub 137, {{nowrap|2400 bit/s}}, which uses [[linear predictive coding]]
* [[Code-excited linear prediction]] (CELP), 2400 and {{nowrap|4800 bit/s}}, Federal Standard 1016, used in [[STU-III]]
* [[Continuously variable slope delta modulation]] (CVSD), {{nowrap|16 kbit/s}}, used in wide band encryptors such as the KY-57.
* [[Mixed-excitation linear prediction]] (MELP), MIL STD 3005, {{nowrap|2400 bit/s}}, used in the Future Narrowband Digital Terminal [[FNBDT]], [[NSA]]'s 21st century secure telephone.
* [[Adaptive Differential Pulse Code Modulation]] (ADPCM), former [[ITU-T]] G.721, {{nowrap|32 kbit/s}} used in [[Secure Terminal Equipment|STE]] secure telephone{{efn|ADPCM is not a proper vocoder but rather a waveform codec. [[ITU]] has gathered G.721 along with some other ADPCM codecs into G.726.}}

Modern vocoders that are used in communication equipment and in voice storage devices today are based on the following algorithms:
* [[Algebraic code-excited linear prediction]] (ACELP 4.7–24&nbsp;kbit/s)<ref>
 {{cite web
  | title     = Voice Age
  | url       = http://www.voiceage.com/codecs.php
  | format    = licensing
  | publisher = VoiceAge Corporation
 }}</ref>
* [[Mixed-excitation linear prediction]] (MELPe 2400, 1200 and {{nowrap|600 bit/s}})<ref>
 {{cite web
  | title     = <!-- Compandent -->MELPe – FAQ
  | url       = http://www.compandent.com/melpe_faq.htm
  | publisher = Compandent Inc
 }}</ref>
* [[Multi-band excitation]] (AMBE {{nowrap|2000 bit/s}}&nbsp;– {{nowrap|9600 bit/s}})<ref>{{cite web
  | title     = IMBE and AMBE
  | url     = http://dvsinc.com/papers/iambe.htm
  | publisher     = Digital Voice Systems, Inc.
  | access-date     = 2008-11-08
  | archive-date     = 2017-07-07
  | archive-url     = https://web.archive.org/web/20170707231905/http://www.dvsinc.com/papers/iambe.htm
  | url-status     = dead
  }}</ref>
* Sinusoidal-Pulsed Representation (SPR {{nowrap|600 bit/s}}&nbsp;– {{nowrap|4800 bit/s}})<ref>{{cite web
  | title     = SPR Vocoders
  | url     = https://dspini.com
  | publisher     = DSP Innovations Inc.
  | access-date     = 2008-11-08
  | archive-date     = 2016-04-09
  | archive-url     = https://dspini.com/vocoders/lowrate/spr-lowrate/spr600
  | url-status     = dead
  }}</ref>
* Robust Advanced Low-complexity Waveform Interpolation (RALCWI 2050, 2400 and {{nowrap|2750 bit/s}})<ref>{{cite web
  | title     = RALCWI Vocoder IC's
  | url     = http://www.cmlmicro.com/products/CMX608-618-638_RALCWI_Vocoder/
  | work     = CML Microcircuits
  | publisher     = CML Microsystems Plc
  | access-date     = 2013-05-17
  | archive-date     = 2018-03-15
  | archive-url     = https://web.archive.org/web/20180315114843/http://www.cmlmicro.com/products/CMX608-618-638_RALCWI_Vocoder/
  | url-status     = dead
  }}</ref>
* Tri-Wave Excited Linear Prediction (TWELP 300–9600&nbsp;bit/s)<ref>
 {{cite web
  | title     = TWELP Vocoder
  | url       = https://dspini.com
  | publisher = DSP Innovations Inc
 }}</ref>
* Noise Robust Vocoder (NRV 300 and {{nowrap|800 bit/s}})<ref>
 {{cite web
  | title     = Noise Rubust Vocoders<!-- 300-800bit/s from BBN -->
  | url       = http://www.bbn.com/technology/speech/nrv
  | publisher = Raytheon BBN Technologies
  | archive-url= https://web.archive.org/web/20140402124332/http://bbn.com/technology/speech/nrv
  | archive-date=2014-04-02
 }}</ref>

Vocoders are also currently used in [[psychophysics]], [[linguistics]], [[computational neuroscience]] and [[cochlear implant]] research.

===Linear prediction-based===
{{Further|Linear predictive coding}}

Since the late 1970s, most non-musical vocoders have been implemented using [[linear prediction]], whereby the target signal's spectral envelope (formant) is estimated by an all-pole [[Infinite impulse response|IIR]] [[digital filter|filter]]. In linear prediction coding, the all-pole filter replaces the bandpass filter bank of its predecessor and is used at the encoder to ''whiten'' the signal (i.e., flatten the spectrum) and again at the decoder to re-apply the spectral shape of the target speech signal.

One advantage of this type of filtering is that the location of the linear predictor's spectral peaks is entirely determined by the target signal, and can be as precise as allowed by the time period to be filtered. This is in contrast with vocoders realized using fixed-width filter banks, where the location of spectral peaks is constrained by the available fixed frequency bands. LP filtering also has disadvantages in that signals with a large number of constituent frequencies may exceed the number of frequencies that can be represented by the linear prediction filter. This restriction is the primary reason that LP coding is almost always used in tandem with other methods in high-compression voice coders.

===Waveform-interpolative===
Waveform-interpolative (WI) vocoder was developed at [[AT&T Bell Laboratories]] around 1995 by W.B. Kleijn, and subsequently, a low- complexity version was developed by AT&T for the DoD secure vocoder competition.  Notable enhancements to the WI coder were made at the [[University of California, Santa Barbara]].  AT&T holds the core patents related to WI and other institutes hold additional patents.<ref name="kleijn95">
{{cite book
 | last1     = Kleijn | first1 = W.B.
 | last2     = Haagen | first2 = J.
 | others    = (AT&T Bell Labs., Murray Hill, NJ)
 | chapter     = A speech coder based on decomposition of characteristic waveforms
 | doi       = 10.1109/ICASSP.1995.479640
| title = 1995 International Conference on Acoustics, Speech, and Signal Processing
 | volume = 1
 | pages = 508–511
 | year = 1995
 | isbn = 978-0-7803-2431-2
 | s2cid = 9105323
 }}</ref><ref name="kleijn96">
{{cite conference
 | last1      = Kleijn | first1 = W.B.
 | last2     = Shoham | first2 = Y.
 | last3     = Sen    | first3 = D.
 | last4     = Hagen  | first4 = R.
 | others    = (AT&T Bell Labs., Murray Hill, NJ)
 | chapter     = A low-complexity waveform interpolation coder
 | conference = [[International Conference on Acoustics, Speech, and Signal Processing]]
 | doi       = 10.1109/ICASSP.1996.540328
 | title = 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings
 | volume = 1
 | pages = 212–215
 | year = 1996
 | isbn = 978-0-7803-3192-1
 | s2cid = 44346744
 }}</ref><ref name="gottesman01">
{{cite journal
 | last1     = Gottesman | first1 = O.
 | last2     = Gersho    | first2 = A.
 | others    = (Dept. of Electr. & Comput. Eng., California Univ., Santa Barbara, CA)
 | title     = Enhanced waveform interpolative coding at low bit-rate
 | journal   = IEEE Transactions on Speech and Audio Processing
 | volume = 9
 | issue     = November 2001
 | pages = 786–798
 | doi       = 10.1109/89.966082
| year = 2001
 | s2cid = 17949435
 }}</ref>