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en:wiki:funktionen:td-functions:phase_extraction [29/06/2016 23:01] hamishen:wiki:funktionen:td-functions:phase_extraction [30/06/2016 09:32] uli
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 **Options: ** **Options: **
  
-   * //Minimum phase: //This option is used to calculate the unique impulse which has the specifed frequency response and is Minimum Phase. A minimum phase filter has the smallest possible group delay at any given frequency (consistent with producing the required overall frequency response), and hence the smallest signal  latency, but the group delay may not be constant at all relevant frequencies. The minimum phase impulse has no pre-ringing but may have post-ringing. The location of the pulse peak is usually at or near the first sample of the filter. +   * //Minimum phase: //This option is used to calculate the unique impulse which has the specifed frequency response and is Minimum Phase. A minimum phase filter has the smallest possible group delay at any given frequency (consistent with producing the required overall frequency response), and hence the smallest signal latency, but the group delay may not be constant at all relevant frequencies. The minimum phase impulse has no pre-ringing but may have post-ringing. The location of the pulse peak is usually at or near the first sample of the filter. 
-  * //Linear phase: //This option is used to calculate a filter (impulse) which has the specified frequency response with a constant group delay at all frequencies.  The price for a constant group delay is that the absolute group delay is large.  The large constant delay may not matter if listening to audio, but needs to be taken into account if audio is live or needs to be synchronized with video. The impulse is symmetric, with the pulse peak located in the center of the filter. Accordingly, the delay is equal to half the filter length / sampling rate (for a filter 65536 samples long at 44100 Hz, the delay is 32768/44100 or about 0.74 seconds). The build up of the pulse before the pulse peak is called "pre-ringing" which can cause audible artifacts. +  * //Linear phase: //This option is used to calculate a filter (impulse) which has the specified frequency response with a constant group delay at all frequencies. The price for a constant group delay is that the absolute group delay is large. The large constant delay may not matter if listening to audio, but needs to be taken into account if audio is live or needs to be synchronized with video. The impulse is symmetric, with the pulse peak located in the center of the filter. Accordingly, the delay is equal to half the filter length / sampling rate (for a filter 65536 samples long at 44100 Hz, the delay is 32768/44100 or about 0.74 seconds). The build up of the pulse before the pulse peak is called "pre-ringing" which in some cases may cause audible artifacts. 
-  * //Excess phase: //This option calculates a filter with constant 0db frequency response but which incorporates any group delay in the input curve in excess of that which the minimum phase component of that curve would produce. Because of the flat amplitude response, such a filter is often called an "All Pass" filter.  A typical impulse response can be separated into the minimum phase component and the excess phase component. Convolving the two components reproduces the original pulse.+  * //Excess phase: //This option calculates a filter with constant 0 dB frequency response but which incorporates any group delay in the input curve in excess of that which the minimum phase component of that curve would produce. Because of the flat amplitude response, such a filter is often called an "All Pass" filter. A typical impulse response can be separated into the minimum phase component and the excess phase component. Convolving the two components reproduces the original pulse.
   * //Advanced mixed phase: //With this option, while maintaining the frequency response, any phase curve between minimum phase (0), Linear Phase (1) and maximum phase (2) are calculated. The parameter can also be in between, for example, 0.159. Possible application: a steep low pass filter can be created between minimum phase (only post-ringing) and linear phase (same pre- and post-oscillation) can be set, with some pre-ringing but more post-ringing. (There is no free lunch: zero ringing is unachievable with systems of finite frequency response!)   * //Advanced mixed phase: //With this option, while maintaining the frequency response, any phase curve between minimum phase (0), Linear Phase (1) and maximum phase (2) are calculated. The parameter can also be in between, for example, 0.159. Possible application: a steep low pass filter can be created between minimum phase (only post-ringing) and linear phase (same pre- and post-oscillation) can be set, with some pre-ringing but more post-ringing. (There is no free lunch: zero ringing is unachievable with systems of finite frequency response!)
- \\ **Use:** \\ The choice of correction filter for a mixed-pulse response may be due principally due to the amount of acceptabe delay. For pure audio reproduction this usually does not matter. In the case of video or live music, a large delay may not be acceptable. Here the shortest lead time is achieved with a minimum phase correction filter. But but no excess phase is corrected.+ \\ **Use:** \\ The choice of correction filter for a mixed-pulse response may be due principally due to the amount of acceptabe delay. For pure audio reproduction this usually does not matter. In the case of video or live music, a large delay may not be acceptable. Here the shortest lead time is achieved with a minimum phase correction filter. But no excess phase is corrected.
  
 A limited band of frequency response can be used for a correction with this function with the frequency response information discarded above or below the selected spectrum (that is, the frequency response output of this funtion is a horizontal straight line below the lower limit and above the higher limit). This function is required, for example to use a near-field recording of frequency response of a speaker driver [[:wiki:anhang:anleitungen:linearisierung_von_frequenzweichen|to linearize digital crossovers]]. It allows the creation of a correction filter to correct the frequency response of the speaker driver only within a desired frequency band. A limited band of frequency response can be used for a correction with this function with the frequency response information discarded above or below the selected spectrum (that is, the frequency response output of this funtion is a horizontal straight line below the lower limit and above the higher limit). This function is required, for example to use a near-field recording of frequency response of a speaker driver [[:wiki:anhang:anleitungen:linearisierung_von_frequenzweichen|to linearize digital crossovers]]. It allows the creation of a correction filter to correct the frequency response of the speaker driver only within a desired frequency band.
en/wiki/funktionen/td-functions/phase_extraction.txt · Last modified: 30/06/2016 18:22 by hamish

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