View source for G.711

{{short description|ITU-T recommendation}}
{{Infobox technology standard
| title             = G.711
| long_name         = Pulse code modulation (PCM) of voice frequencies    
| image             = Pcm.svg 
| caption           = 
| status            = In force
| year_started      = 1972
| version           = (02/00)
| version_date      = February 2000
| preview           = 
| preview_date      = 
| organization      = [[ITU-T]]
| committee         = 
| base_standards    = 
| related_standards = [[G.711.0]], [[G.711.1]]
| abbreviation      = 
| domain            = [[audio compression (data)|audio compression]]
| license           = 
| website           = https://www.itu.int/rec/T-REC-G.711
}}

'''G.711''' is an [[ITU-T]] standard (Recommendation) for audio [[companding]], titled '''Pulse code modulation (PCM) of voice frequencies''', it is a required standard in many technologies, such as in the [[H.320]] and [[H.323]] standards.<ref name=":0">{{Cite web|url=https://www.itu.int/rec/T-REC-G.711|title=G.711 : Pulse code modulation (PCM) of voice frequencies|last=|first=|date=|website=www.itu.int|url-status=live|archive-url=https://web.archive.org/web/20190617130328/https://www.itu.int/rec/T-REC-G.711|archive-date=2019-06-17|access-date=2019-11-11}}</ref> It was originally designed for use in [[telephony]] and was released for use in 1972.  It can also be used for [[fax]] communication over IP networks (as defined in [[T.38]] specification). G.711 is a [[narrowband]] audio codec that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400&nbsp;Hz and samples them at the rate of 8,000 samples per second, with the tolerance on that rate of 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. There are two slightly different versions: [[μ-law]], which is used primarily in North America and Japan, and [[A-law]], which is in use in most other countries outside North America.

==Enhancements==
Two enhancements to G.711 have been published: '''G.711.0''' utilizes [[lossless data compression]] to reduce the bandwidth usage and '''G.711.1''' increases audio quality by increasing bandwidth.

==Features==
* Sampling frequency 8&nbsp;kHz
* 64 kbit/s bitrate (8&nbsp;kHz sampling frequency × 8 bits per sample)
* Typical algorithmic delay is 0.125 ms, with no look-ahead delay
* G.711 is a [[waveform]] [[speech coder]]
* G.711 Appendix I defines a [[packet loss concealment]] (PLC) algorithm to help hide transmission losses in a packetized network
* G.711 Appendix II defines a [[discontinuous transmission]] (DTX) algorithm which uses [[voice activity detection]] (VAD) and [[Comfort noise|comfort noise generation]] (CNG) to reduce bandwidth usage during silence periods
* [[PSQM]] testing under ideal conditions yields [[mean opinion score]]s of 4.45 for G.711 μ-law, 4.45 for G.711 A-law
* [[PSQM]] testing under network stress yields [[mean opinion score]]s of 4.13 for G.711 μ-law, 4.11 for G.711 A-law

==Types==
G.711 defines two main [[companding]] algorithms, the [[μ-law algorithm]] and [[A-law algorithm]]. Both are [[logarithmic scale|logarithmic]], but A-law was specifically designed to be simpler for a computer to process. The standard also defines a sequence of repeating code values which defines the power level of 0 [[Decibel|dB]].

The μ-law and A-law algorithms encode 14-bit and 13-bit signed linear PCM samples (respectively) to logarithmic 8-bit samples. Thus, the G.711 [[encoder]] will create a 64 kbit/s bitstream for a signal sampled at 8&nbsp;kHz.<ref name=":0" />

G.711 μ-law tends to give more resolution to higher range signals while G.711 A-law provides more quantization levels at lower signal levels.

=== A-law ===
{{Main|A-law algorithm}}
A-law encoding thus takes a 13-bit signed linear audio sample as input and converts it to an 8 bit value as follows:

{| class="wikitable" style="text-align:left"
|-
! Linear input code<br><ref group=note>This value is produced by taking the [[two's complement]] representation of the input value, and inverting all bits after the sign bit if the value is negative.</ref>
! Compressed code<br>XOR 01010101
! Linear output code<br><ref group=note>[[Signed magnitude]] representation</ref>
|-
| <tt>s0000000abcdx</tt> || <tt>{{overline|s}}000abcd</tt> || <tt>s0000000abcd1</tt>
|-
| <tt>s0000001abcdx</tt> || <tt>{{overline|s}}001abcd</tt> || <tt>s0000001abcd1</tt>
|-
| <tt>s000001abcdxx</tt> || <tt>{{overline|s}}010abcd</tt> || <tt>s000001abcd10</tt>
|-
| <tt>s00001abcdxxx</tt> || <tt>{{overline|s}}011abcd</tt> || <tt>s00001abcd100</tt>
|-
| <tt>s0001abcdxxxx</tt> || <tt>{{overline|s}}100abcd</tt> || <tt>s0001abcd1000</tt>
|-
| <tt>s001abcdxxxxx</tt> || <tt>{{overline|s}}101abcd</tt> || <tt>s001abcd10000</tt>
|-
| <tt>s01abcdxxxxxx</tt> || <tt>{{overline|s}}110abcd</tt> || <tt>s01abcd100000</tt>
|-
| <tt>s1abcdxxxxxxx</tt> || <tt>{{overline|s}}111abcd</tt> || <tt>s1abcd1000000</tt>
|}
<references group=note/>

Where {{code|s}} is the sign bit, <code>{{overline|s}}</code> is its inverse (i.e. positive values are encoded with [[most significant bit|MSB]]&nbsp;=&nbsp;{{var|{{overline|s}}}}&nbsp;=&nbsp;1), and bits marked {{code|x}} are discarded. Note that the first column of the table uses different representation of negative values than the third column. So for example, input decimal value −21 is represented in binary after bit inversion as 1000000010100, which maps to 00001010 (according to the first row of the table). When decoding, this maps back to 1000000010101, which is interpreted as output value −21 in decimal. Input value +52 (0000000110100 in binary) maps to 10011010 (according to the second row), which maps back to 0000000110101 (+53 in decimal).

This can be seen as a [[Floating-point arithmetic|floating-point]] number with 4 bits of [[Significand|mantissa]] {{var|m}} (equivalent to a 5-bit precision), 3 bits of [[exponent]] {{var|e}} and 1 sign bit {{var|s}}, formatted as <code>{{overline|s}}eeemmmm</code> with the decoded linear value {{var|y}} given by formula
:<math>y = (-1)^s \cdot (16 \cdot \min \{ e, 1 \} + m + 0.5) \cdot 2^{\max \{ e, 1 \} },</math>
which is a 13-bit signed integer in the range ±1 to ±(2{{sup|12}}&nbsp;−&nbsp;2{{sup|6}}). Note that no compressed code decodes to zero due to the addition of 0.5 (half of a quantization step).

In addition, the standard specifies that all resulting even bits ([[least significant bit|LSB]] is even) are inverted before the octet is transmitted. This is to provide plenty of 0/1 transitions to facilitate the [[clock recovery]] process in the PCM receivers. Thus, a silent A-law encoded PCM channel has the 8 bit samples coded 0xD5 instead of 0x80 in the octets.

When data is sent over E0 ([[G.703]]), MSB (sign) is sent first and LSB is sent last.

ITU-T STL<ref>[http://www.itu.int/rec/T-REC-G.191-201003-I/en G.191 : Software tools for speech and audio coding standardization]. Function {{code|alaw_expand}} in file {{code|Software/stl2009/g711/g711.c}}. Itu.int. Retrieved on 2013-09-18.</ref> defines the algorithm for decoding as follows (it puts the decoded values in the 13 most significant bits of the 16-bit output data type).

<syntaxhighlight lang="c">
void            alaw_expand(lseg, logbuf, linbuf)
  long            lseg;
  short          *linbuf;
  short          *logbuf;
{
  short           ix, mant, iexp;
  long            n;

  for (n = 0; n < lseg; n++)
  {
    ix = logbuf[n] ^ (0x0055);	/* re-toggle toggled bits */

    ix &= (0x007F);		/* remove sign bit */
    iexp = ix >> 4;		/* extract exponent */
    mant = ix & (0x000F);	/* now get mantissa */
    if (iexp > 0)
      mant = mant + 16;		/* add leading '1', if exponent > 0 */

    mant = (mant << 4) + (0x0008);	/* now mantissa left justified and */
    /* 1/2 quantization step added */
    if (iexp > 1)		/* now left shift according exponent */
      mant = mant << (iexp - 1);

    linbuf[n] = logbuf[n] > 127	/* invert, if negative sample */
      ? mant
      : -mant;
  }
}
</syntaxhighlight>

See also "ITU-T Software Tool Library 2009 User's manual" that can be found at.<ref>[http://www.itu.int/rec/T-REC-G.191/recommendation.asp?lang=en&parent=T-REC-G.191-200911-I G.191 : ITU-T Software Tool Library 2009 User's manual]. Itu.int (2010-07-23). Retrieved on 2013-09-18.</ref>

=== μ-law ===
{{Main|μ-law algorithm}}
The μ-law (sometimes referred to as ulaw, G.711Mu, or G.711μ) encoding takes a 14-bit signed linear audio sample in [[two's complement]] representation as input, inverts all bits after the sign bit if the value is negative, adds 33 (binary 100001) and converts it to an 8 bit value as follows:

{| class="wikitable" style="text-align:left"
|-
! Linear input value<br><ref group=note>This value is produced by taking the [[two's complement]] representation of the input value, inverting all bits after the sign bit if the value is negative, and adding 33.</ref>
! Compressed code<br>XOR 11111111
! Linear output value<br><ref group=note>[[Signed magnitude]] representation. Final result is produced by decreasing the magnitude of this value by 33.</ref>
|-
| <tt>s00000001abcdx</tt> || <tt>s000abcd</tt> || <tt>s00000001abcd1</tt>
|-
| <tt>s0000001abcdxx</tt> || <tt>s001abcd</tt> || <tt>s0000001abcd10</tt>
|-
| <tt>s000001abcdxxx</tt> || <tt>s010abcd</tt> || <tt>s000001abcd100</tt>
|-
| <tt>s00001abcdxxxx</tt> || <tt>s011abcd</tt> || <tt>s00001abcd1000</tt>
|-
| <tt>s0001abcdxxxxx</tt> || <tt>s100abcd</tt> || <tt>s0001abcd10000</tt>
|-
| <tt>s001abcdxxxxxx</tt> || <tt>s101abcd</tt> || <tt>s001abcd100000</tt>
|-
| <tt>s01abcdxxxxxxx</tt> || <tt>s110abcd</tt> || <tt>s01abcd1000000</tt>
|-
| <tt>s1abcdxxxxxxxx</tt> || <tt>s111abcd</tt> || <tt>s1abcd10000000</tt>
|}
<references group=note/>

Where {{code|s}} is the sign bit, and bits marked {{code|x}} are discarded.

In addition, the standard specifies that all result bits are inverted before the octet is transmitted. Thus, a silent μ-law encoded PCM channel has the 8 bit samples coded 0xFF instead of 0x00 in the octets.

Adding 33 is necessary so that all values fall into a compression group and it is subtracted back when decoding. This addition means that an overflow would occur for values outside the ±8159 range, so such values are clipped during encoding to avoid it.

Breaking the encoded value formatted as <code>seeemmmm</code> into 4 bits of mantissa {{var|m}}, 3 bits of exponent {{var|e}} and 1 sign bit {{var|s}}, the decoded linear value {{var|y}} is given by formula
:<math>y = (-1)^s \cdot [(16.5 + m) \cdot 2^{e + 1} - 33],</math>
which is a 14-bit signed integer in the range ±0 to ±8031.

Note that 0 is encoded as 0xFF, and −1 is encoded as 0x7F, but when decoded back the result is 0 in both cases.

== G.711.0 ==
G.711.0, also known as G.711 LLC, utilizes [[lossless data compression]] to reduce the bandwidth usage by as much as 50 percent.<ref>{{cite web|url=http://www.itu.int/ITU-T/newslog/Voice+Codec+Gets+New+Lossless+Compression.aspx|title=ITU-T Newslog - Voice codec gets new lossless compression|author=ITU-T|first=|date=2009-07-17|website=|url-status=live|archive-url=https://web.archive.org/web/20160303224212/https://www.itu.int/ITU-T/newslog/Voice+Codec+Gets+New+Lossless+Compression.aspx|archive-date=2016-03-03|accessdate=2010-02-28}}</ref> The ''Lossless compression of G.711 pulse code modulation'' standard was approved by ITU-T in September 2009.<ref>{{cite web |url=http://www.itu.int/rec/T-REC-G.711.0-200909-P/en |title=G.711.0 : Lossless compression of G.711 pulse code modulation |author=ITU-T |accessdate=2010-02-28}}</ref><ref name="itu-t-devel">{{citation |url=http://www.eurasip.org/Proceedings/Eusipco/Eusipco2008/plenaries/lamblin.pdf |title=Recent Audio/Speech Coding Developments in ITU-T and future trends |date=August 2008 |accessdate=2010-02-28}}</ref>

== G.711.1 ==
G.711.1 is an extension to G.711, published as ITU-T Recommendation G.711.1 in March 2008. Its formal name is ''Wideband embedded extension for G.711 pulse code modulation''.<ref name="itu-t-devel" /><ref name="g711-4">ITU-T (2008) [http://www.itu.int/rec/T-REC-G.711.1/en G.711.1 : Wideband embedded extension for G.711 pulse code modulation] Retrieved on 2009-06-19</ref><ref>{{citation |url=http://www.eurasip.org/Proceedings/Eusipco/Eusipco2008/papers/1569104469.pdf |title=G.711.1: a wideband extension to ITU-T G.711 |author=Hiwasaki |date=2008-08-25 |accessdate=2015-06-13|display-authors=etal}}</ref>

G.711.1, allows the addition of [[narrowband]] and/or [[wideband]] (16000 samples/s) enhancements, each at 25% of the bitrate of the (included) base G.711 bitstream, leading to data rates of 64, 80 or 96 kbit/s.

G.711.1 is compatible with G.711 at 64 kbit/s,<ref>{{citation |url=http://www.eurasip.org/Proceedings/Eusipco/Eusipco2008/papers/1569105406.pdf |title=Noise shaping in an ITU-T G.711-Interoperable embedded codec |author=Lapierre |date=2008-08-25 |accessdate=2015-06-13|display-authors=etal}}</ref> hence an efficient deployment in existing G.711-based voice over IP ([[VoIP]]) infrastructures is foreseen. The G.711.1 coder can encode signals at 16&nbsp;kHz with a bandwidth of 50–7000&nbsp;Hz at 80 and 96 kbit/s, and for 8-kHz sampling the output may produce signals with a bandwidth ranging from 50 up to 4000&nbsp;Hz, operating at 64 and 80 kbit/s.<ref name="g711-4" />

The G.711.1 encoder creates an embedded bitstream structured in three layers corresponding to three available bit rates: 64, 80 and 96 kbit/s. The bitstream does not contain any information on which layers are contained, an implementation would require outband signalling on which layers are available. The three G.711.1 layers are: log companded pulse code modulation (PCM) of the lower band including noise feedback, embedded PCM extension with adaptive bit allocation for enhancing the quality of the base layer in the lower band, and weighted vector quantization coding of the higher band based on [[modified discrete cosine transform]]ation (MDCT).<ref name="g711-4" />

Two extensions for G.711.1 are planned in 2010: superwideband extension (bandwidth to 14000&nbsp;Hz) and lossless bitstream compression.<ref>{{citation |url=http://www.ficora.fi/attachments/suomiry/5lN8SYM14/Stand09-Hagqvist.pdf |title=Coding standards |author=Nokia Research Center |date=2009-04-06 |accessdate=2010-03-01}}</ref>

== Licensing ==
Since G.711 was released in 1972 its patents have long since expired, so it is freely available.<ref name=":0" />

== See also ==
*[[List of codecs]]
*[[Comparison of audio coding formats]]
*[[RTP audio video profile]]
*[[Au file format]]

== References ==
<references/>

== External links ==
* [https://www.itu.int/rec/T-REC-G.711 ITU-T Recommendation G.711]
* [http://www.itu.int/rec/T-REC-G.191 ITU-T G.191 software tools for speech and audio coding, including G.711 C code]
* [https://web.archive.org/web/20060720180425/http://www.codeproject.com/csharp/g711audio.asp Code Project C# implementation of G.711 with source code]
* [http://tools.ietf.org/html/rfc3551#page-28 RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control] - G.711 - PCMA and PCMU definition.
* [http://tools.ietf.org/html/rfc4856#page-21 RFC 4856 - Registration of Media Type audio/PCMA and audio/PCMU]
* RFC 5391 - RTP Payload Format for ITU-T Recommendation G.711.1 (PCMA-WB and PCMU-WB)

{{Compression formats}}

[[Category:Audio codecs]]
[[Category:Speech codecs]]
[[Category:ITU-T recommendations]]
[[Category:ITU-T G Series Recommendations]]
[[Category:Telecommunications-related introductions in 1972]]