Talk:Upsampling

message: Something seems wrong here.  The page says
:"Upsampling is the process of increasing the sampling rate of a signal. This is usually done to increase the bandwidth of a signal."
I think the second sentence is not correct.  Ideally, upsampling should not increase bandwidth, only sample rate.
Quoting further:
:"The upsampling factor (commonly denoted by L) is usually an integer or a rational fraction greater than unity. This factor multiplies the sampling rate or, equivalently, divides the sampling period. For example, if compact disc audio was upsampled by a factor of 5/4 then the resulting sampling rate goes from 44,100 Hz to 55,125 Hz, ... The range of valid frequencies (i.e., those that satisfy the Nyquist-Shannon sampling theorem) has gone from 22,050 Hz to 27,562.5 (an increase in 5,512.5 Hz."
While it is true that at the 55,125 Hz sampling rate, frequencies up to 27,562 Hz could be represented, producing such frequency content is not the purpose of the upsampling process.  In fact, any frequencies that are above 22,050 Hz after the upsampling must be undesirable artifacts, as they were not represented in the original (lower sample rate) signal - and the purpose of the low-pass filter is to eliminate any such increase in bandwidth.  After applying the low-pass interpolation filter, the bandwidth of the original signal and the bandwidth of the upsampled signal should be approximately the same if the filter has done a good job.  Thus the purpose of upsampling is not to increase the bandwidth of the signal. --- SudoMonas
:Note, please sign your posts with ~~~~
Certainly upsampling by 16 will increase your bandwidth by 16, but your original signal won't fill the extra bandwidth.  Regardless, increasing your sampling rate gives you more bandwidth. Cburnett 05:34, 9 Mar 2005 (UTC)
replies: text: The filter is the same, only the purpose is different. With upsampling, there is no problem with aliasing. Mirror Vax 15:36, 31 March 2006 (UTC)
thread_title:  Bandwidth versus sampling rate 

message: Towards the end of the section on 'Upsampling by integer factor', I would think that the following text might be useful
--- start text--
When zeros are inserted into the signal for upsampling, the spectrum of the upsampled sequence will have aliases at integer multiples of the original sampling frequency. These aliases can be removed to a reasonable extent by a finite impulse response low pass filter. The presence of zeros in the sequence which is passed through the filter can be used to reduce the complexity of the filter implementation. The original filter can be split to  L  subfilters and the output of each of these subfilters is sequentially tapped to obtain the filtered output sequence  Polyphase filters for interpolation  .
---stop text----
What say? 
Beetelbug 14:54, 12 May 2007 (UTC)
thread_title:  Additional text to upsampling  

message: Maybe we can understand upsampling like this: First the sampling rate is Fs, and we want to upsampling the rate to positive infinite. (also it seems the situation is going to change from Digital world to Analog world~. ) Note that when the signal is presented by the rate of Fs, in fact the signal is periodic in spectrum whose period is Fs, the reason we can't see it is just because the sampling rate Fs gives no ability to handle. So when the sampling rate comes to really big, there will be a lot of signals repeated on the spectrum with period of Fs. Now abviously we need a interpolation filter after the upsampling process.
Terrence007 (talk) 03:47, 6 November 2011 (UTC)
replies: text: interesting.  Dicklyon (talk) 03:55, 6 November 2011 (UTC)
thread_title:  an interesting point of understanding upsampling