mistvan <miklos@dbresearch.co.uk> wrote: < Thanks everyone for taking the time to respond. < For each frame, I append (L-1) samples to the beginning of the current < frame from the previous frame. So my input frame is now 128 + (11-1)= 138 < samples. I convolve this with my filter, resulting in an ouput of 138 + (10 < -1) = 147 output samples. I think that should be 148, but it doesn't matter since you discard them anyway. What do you do with the first frame? I would say prepend 10 zeros, but you didn't say that. < I take my 128 output samples starting at sample number 11. The remaining < output samples at the end are discarded. I then save the end 10 samples < from the 'input' ready for the arrival of the next frame. < Is this the correct procedure? I think that sounds right. Are you sure that is actually what you are doing? -- glen
Linear Convolution of input seuence split into frames
Started by ●May 13, 2009
Reply by ●May 15, 20092009-05-15
Reply by ●May 15, 20092009-05-15
>mistvan wrote: >>> On May 13, 7:03=A0pm, "mistvan" <mik...@dbresearch.co.uk> wrote: >>> >>> Jerry sez: >>>> > ...You aren't using FFT convolution for a >>>>> mere 11-point convolution, are you? >>> ... >>>> No not using fft. >>>> >>>> I'm afraid it is not possible to concatenate the frames. It is real >> time >>>> 2d data. I apply the fir to each row of data within this 2d frame. >> Then >>>> wait for the next frame to process. I have plotted the input andoutput>> o= >>> f >>>> one of the rows,this shows that the filtering is not happening atthe>> fra= >>> me >>>> boundaries. It is ok before and after. >>>> >>>> I'm new to dsp, so I might be barking up the wrong tree, but Ithought>>>> this would work >>>> >>>> My procedure for a row within a frame is as follows >>>> a. filter the row (typically 128 samples, but this varys) >>>> b. save L-1 samples from end of current row for next row >>>> c. append the L-1 samples to the beginning of the next row >>>> d =A0goto (a) >>>> >>>> Any ideas? >>> well, this is the overlap-save technique which is what you would have >>> to do frame-by-frame, whether you were using the FFT or not. >>> >>> so each time you filter a row, how many samples do you get out? 128 >>> or 128-(L-1)? >>> >>> r b-j >>> >>> >> >> Hi >> >> Thanks everyone for taking the time to respond. >> >> For each frame, I append (L-1) samples to the beginning of the current >> frame from the previous frame. So my input frame is now 128 + (11-1)=138>> samples. I convolve this with my filter, resulting in an ouput of 138 +(10>> -1) = 147 output samples. > >Are the appended samples in any way related to the rest, or are they >arbitrary filler? > >> I take my 128 output samples starting at sample number 11. Theremaining>> output samples at the end are discarded. I then save the end 10samples>> from the 'input' ready for the arrival of the next frame. >> >> Is this the correct procedure? > >I don't think so. If the frames are independent, you can't avoid losing >the margins. > >Jerry >-- >Engineering is the art of making what you want from things you can get. >����������������������������������������������������������������������� >Yes , the frames are related. It is one long sequence. When you say 'you cant avoid losing the margins'. Could you elaborate. I thought that with an FIR filter the current output sample is the convolution sum of a number of previous input samples multiplied by the corresponding coefficients. If I save these previous input samples (in my case this would be 11 -1 samples) then why will I have problems at the margins. Thanks for your time, much appreciated
Reply by ●May 15, 20092009-05-15
>mistvan <miklos@dbresearch.co.uk> wrote: > >< Thanks everyone for taking the time to respond. > >< For each frame, I append (L-1) samples to the beginning of the current >< frame from the previous frame. So my input frame is now 128 + (11-1)=138>< samples. I convolve this with my filter, resulting in an ouput of 138 +(10>< -1) = 147 output samples. > >I think that should be 148, but it doesn't matter since you discard >them anyway. > >What do you do with the first frame? I would say prepend 10 zeros, >but you didn't say that. > >< I take my 128 output samples starting at sample number 11. Theremaining>< output samples at the end are discarded. I then save the end 10samples>< from the 'input' ready for the arrival of the next frame. > >< Is this the correct procedure? > >I think that sounds right. Are you sure that is actually >what you are doing? > >-- glen >Thanks glen, I do prepend 10 zeroes to the beginning and yes it should be 148. And no that is not what I was doing. Doh! I've got it working now. Thanks for everyone's advice. Everyone has been most helpful :)
Reply by ●May 15, 20092009-05-15
On May 13, 2:03�pm, "mistvan" <mik...@dbresearch.co.uk> wrote:> Hi > > I am trying to implement a linear convolution of FIR IR and a large input > sequence. The IR is symmetrical (L=11). I receive the data in frames of say > 128 samples (could be less or more than this). The output exhibits problems > at the frame boundaries. What is the best way of overcoming these boundary > effects. I have tried saving L-1 input samples from the end of the previous > frame, which I tag onto the front of the current frame, but the problems > still exist.Is it a symmetric FIR and you want the equivalent of "zero delay"? Then you may need to prepend 5 or 6 samples from the previous frame and also append to the end 5 or 6 samples from the next frame. Or you could offset your output "in time" and just use saved samples from the previous frame. You could also save state by keeping a triangle of fractional FIR calculations and using them to finish the first few FIR taps at the beginning of the next frame. IMHO. YMMV. -- rhn A.T nichlson d.0.t C-o-M http://www.nicholson.com/rhn/dsp.html
Reply by ●May 15, 20092009-05-15
mistvan wrote:>> mistvan wrote: >>>> On May 13, 7:03=A0pm, "mistvan" <mik...@dbresearch.co.uk> wrote: >>>> >>>> Jerry sez: >>>>> > ...You aren't using FFT convolution for a >>>>>> mere 11-point convolution, are you? >>>> ... >>>>> No not using fft. >>>>> >>>>> I'm afraid it is not possible to concatenate the frames. It is real >>> time >>>>> 2d data. I apply the fir to each row of data within this 2d frame. >>> Then >>>>> wait for the next frame to process. I have plotted the input and > output >>> o= >>>> f >>>>> one of the rows,this shows that the filtering is not happening at > the >>> fra= >>>> me >>>>> boundaries. It is ok before and after. >>>>> >>>>> I'm new to dsp, so I might be barking up the wrong tree, but I > thought >>>>> this would work >>>>> >>>>> My procedure for a row within a frame is as follows >>>>> a. filter the row (typically 128 samples, but this varys) >>>>> b. save L-1 samples from end of current row for next row >>>>> c. append the L-1 samples to the beginning of the next row >>>>> d =A0goto (a) >>>>> >>>>> Any ideas? >>>> well, this is the overlap-save technique which is what you would have >>>> to do frame-by-frame, whether you were using the FFT or not. >>>> >>>> so each time you filter a row, how many samples do you get out? 128 >>>> or 128-(L-1)? >>>> >>>> r b-j >>>> >>>> >>> Hi >>> >>> Thanks everyone for taking the time to respond. >>> >>> For each frame, I append (L-1) samples to the beginning of the current >>> frame from the previous frame. So my input frame is now 128 + (11-1)= > 138 >>> samples. I convolve this with my filter, resulting in an ouput of 138 + > (10 >>> -1) = 147 output samples. >> Are the appended samples in any way related to the rest, or are they >> arbitrary filler? >> >>> I take my 128 output samples starting at sample number 11. The > remaining >>> output samples at the end are discarded. I then save the end 10 > samples >>> from the 'input' ready for the arrival of the next frame. >>> >>> Is this the correct procedure? >> I don't think so. If the frames are independent, you can't avoid losing >> the margins. >> >> Jerry >> -- >> Engineering is the art of making what you want from things you can get. >> ����������������������������������������������������������������������� >> > > Yes , the frames are related. It is one long sequence. > > When you say 'you cant avoid losing the margins'. Could you elaborate. I > thought that with an FIR filter the current output sample is the > convolution sum of a number of previous input samples multiplied by the > corresponding coefficients. If I save these previous input samples (in my > case this would be 11 -1 samples) then why will I have problems at the > margins.How are they related? In time? If so, then why can't you concatenate the rows of to make a continuous data stream? Just do your saving in what amounts to the filter's shift register. If you use a circular buffer, make it 128 + 11 = 139 samples long. Load the new block where it won't overwrite the 11 samples that need saving. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
