Talk:Linear pulse-code modulation: Difference between revisions

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{{merged-to|Pulse-code modulation|date=2014-03-01}}
== Zero value in signed code==
The article sais ''If the sample is 16-bit signed, the sample range is from -32768 to 32767, with a centerpoint of 0''. If the entire range <math>[-2^{N-1}, 2^{N-1}-1]</math> is allowed, then the zero value actually is (assuming the zero value is the average of maximum and minimum)
 
<math>
\frac{-2^{N-1} + 2^{N-1}-1}{2} = 2^N\frac{-2^{-1} + 2^{-1}-2^{-N}}{2}=-2^N 2^{-N}/2=-1/2\neq 0
</math>
 
==Merge with Pulse Code Modulation==
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___ <span style="font-size: smaller;" class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/67.122.114.28|67.122.114.28]] ([[User talk:67.122.114.28|talk]]) 03:02, 19 August 2005 (UTC)</span><!-- Template:UnsignedIP2 -->
 
:I agree -- I'm after a standard, ISO something something -- very much related to computer audio (DLNA specifically) so it's computer-type information that I'm after. <span style="font-size: smaller;" class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/210.11.153.86|210.11.153.86]] ([[User talk:210.11.153.86|talk]]) 09:16, 31 August 2005 (UTC)</span><!-- Template:UnsignedIP2 -->
 
:I support a merge. I don't see much support for LPCM being a distinct file format. The references cited use the term "linear audio" but not "linear PCM". These are generic audio formats and the individule files already have their own articles, e.g. [[WAV]], [[AIFF]], [[Au file format]] -—[[user talk:Kvng|Kvng]] 13:47, 27 September 2012 (UTC)
 
{{done}} ~[[user talk:Kvng|KvnG]] 00:27, 2 March 2014 (UTC)
 
== Source of the LPCM stub ==
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The stub for LPCM was taken from an ad for a computer program from the company Cyberlink. This is the website:
 
http://www.cyberlink.com/english/dv-entertainment/articles/lpcm.jsp
 
<span style="font-size: smaller;" class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/65.87.26.127|65.87.26.127]] ([[User talk:65.87.26.127|talk]]) 17:31, 28 September 2006 (UTC)</span><!-- Template:UnsignedIP2 -->
 
== vs. PCM ==
 
I think this article should explain the difference between PCM and LPCM.
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It is unfortunate that the example used here is so similar to the max bitrate for PCM tracks on a DVD - 6144 kbps.
 
<span style="font-size: smaller;" class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/128.111.117.22|128.111.117.22]] ([[User talk:128.111.117.22|talk]]) 16:04, 13 May 2008 (UTC) </span><!-- Template:UnsignedIP2 -->
[[User:Dynamicimanyd|Dynamicimanyd]] ([[User talk:Dynamicimanyd|talk]]) 17:14, 6 January 2009 (UTC) 6144000 bps = 6.144 Mbps (megabit per second), but 5.859375 Mibps (mebibit per second). This is the usual use of Mbps in computing and engineering in data communications including discussions of audio bitrates (the exception being Microsoft software like Windows Media Player). It appears that 6.144 x 10<sup>6</sup> bit/s is the maximum total audio data rate in DVD-Video for ALL channels put together, implying that 192 kHz/16-bit stereo is at maximum bitrate for DVD-video (because 6144 kbps = 6.144 Mbps), but more usefully, 8 channels (7.1 surround) of 48 kHz/16-bit audio can be used, which is surely where the bitrate ceiling came from. It's tough to prove that greater bit-depth or sampling frequency could be useful in an 8-channel LPCM audio content delivery system for humans (though it's useful in the recording/mixing/mastering process that precedes it).
 
[[User:Dynamicimanyd|Dynamicimanyd]] ([[User talk:Dynamicimanyd|talk]]) 17:14, 6 January 2009 (UTC) 6144000 bps = 6.144 Mbps (megabit per second), but 5.859375 Mibps (mebibit per second). This is the usual use of Mbps in computing and engineering in data communications including discussions of audio bitrates (the exception being Microsoft software like Windows Media Player). It appears that 6.144 x 10<sup>6</sup> bit/s is the maximum total audio data rate in DVD-Video for ALL channels put together, implying that 192 kHz/16-bit stereo is at maximum bitrate for DVD-video (because 6144 kbps = 6.144 Mbps), but more usefully, 8 channels (7.1 surround) of 48 kHz/16-bit audio can be used, which is surely where the bitrate ceiling came from. It's tough to prove that greater bit-depth or sampling frequency could be useful in an 8-channel LPCM audio content delivery system for humans (though it's useful in the recording/mixing/mastering process that precedes it).
—[[User:Dynamicimanyd|Dynamicimanyd]] ([[User talk:Dynamicimanyd|talk]]) 17:14, 6 January 2009 (UTC)
 
== Incorrect Shannon Theory Claim ==
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</blockquote>
But this is wrong. The Nyquist Theorem says that the highest frequency of signal that CAN be recorded is 1/2 the sampling rate. It does not say that a sampling rate of R will accurately record a signal of frequency R/2. For example, to accurately record a frequency of, say, 22 kHz with a 44 kHz signal requires that the samples be taken at exactly the maximum of the peak and exactly the minimum of the valleys; if the 22 kHz signal is phase-shifted at all from the period of the sampling, then the 22 kHz will be aliased as a lower-frequency signal by the 44 kHz model. <small><span class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[User:MarkRLindsey|MarkRLindsey]] ([[User talk:MarkRLindsey|talk]] • [[Special:Contributions/MarkRLindsey|contribs]]) 01:06, 2 February 2009 (UTC)</span></small><!-- Template:Unsigned --> <!--Autosigned by SineBot-->
== Zero value in signed code==
The article sais ''If the sample is 16-bit signed, the sample range is from -32768 to 32767, with a centerpoint of 0''. If the entire range <math>[-2^{N-1}, 2^{N-1}-1]</math> is allowed, then the zero value actually is (assuming the zero value is the average of maximum and minimum)
 
<math>
\frac{-2^{N-1} + 2^{N-1}-1}{2} = 2^N\frac{-2^{-1} + 2^{-1}-2^{-N}}{2}=-2^N 2^{-N}/2=-1/2\neq 0
</math>
 
<span style="font-size: smaller;" class="autosigned">—Preceding [[Wikipedia:Signatures|unsigned]] comment added by [[Special:Contributions/90.229.142.67|90.229.142.67]] ([[User talk:90.229.142.67|talk]]) 14:09, 31 December 2010 (UTC)</span><!-- Template:UnsignedIP2 -->
 
==Strictly speaking, the term "linear quantization" is self-contradictory==
This article says that "LPCM is PCM with linear quantization". However, there is no such thing as linear quantization. Quantization is an inherently non-linear process. Linear processes are invertible (exceptordinarily) in the degenerate special case)invertible. Quantization is not invertible. No quantizer is linear. Some abuse of basic mathematical concepts is necessary to come up with such a term. This strange term "linear quantization" should be removed, or at least explained. The referenced document does not provide a definition of this self-contradictory term. —[[User:SudoMonas|SudoMonas]] ([[User talk:SudoMonas|talk]]) 16:51, 25 April 2011 (UTC)
 
:Linear here refers to the step size use for encoding. An example on non-linear PCM is something encoded using [[Mu-law]]. -—[[user talk:Kvng|Kvng]] 13:47, 27 September 2012 (UTC)
 
== Contributions needing work ==
 
{{u|Doorknob747}} contributed the following to the article. I'm pulling these contributions here for discussion before inclusion.
 
I'm not aware of 32-bit linear PCM in consumer applications. 32-bit floating point is a thing. Do you have a citation?
 
As for the second contribution, discussion of subjective sound quality doesn't usually lead anywhere productive. Especially so for uncited discussion. ~[[user talk:Kvng|KvnG]] 20:30, 21 December 2013 (UTC)
 
:<nowiki>==Linear 32bit PCM==
There is a L32 bit PCM, and there are many sound cards that support it.
 
===Similar effects of DTS-HD master audio (192kHz DTS)===
It is said that there is no difference at all that can be heard from a Linear 32 bit PCM at a 96&nbsp;kHz sample rate playback sound to a, high quality DTS-HD Master Audio (192&nbsp;kHz DTS). They both sound the same because, of the extremely high quality sound playback from these two types of codecs. But, in reality the 192&nbsp;kHz DTS sound file has actually 1.45 times better quality than a sound file of L32 at 96&nbsp;kHz.</nowiki>