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pfb_channelizer_ccf.h
Go to the documentation of this file.
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/* -*- c++ -*- */
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/*
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* Copyright 2009,2010,2012 Free Software Foundation, Inc.
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*
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* This file is part of GNU Radio
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*
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* GNU Radio is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 3, or (at your option)
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* any later version.
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*
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* GNU Radio is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with GNU Radio; see the file COPYING. If not, write to
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* the Free Software Foundation, Inc., 51 Franklin Street,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef INCLUDED_FILTER_PFB_CHANNELIZER_CCF_H
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#define INCLUDED_FILTER_PFB_CHANNELIZER_CCF_H
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#include <
gnuradio/filter/api.h
>
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#include <
gnuradio/block.h
>
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namespace
gr {
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namespace
filter {
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/*!
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* \brief Polyphase filterbank channelizer with
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* gr_complex input, gr_complex output and float taps
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* \ingroup channelizers_blk
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*
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* \details
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* This block takes in complex inputs and channelizes it to <EM>M</EM>
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* channels of equal bandwidth. Each of the resulting channels is
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* decimated to the new rate that is the input sampling rate
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* <EM>fs</EM> divided by the number of channels, <EM>M</EM>.
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*
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* The PFB channelizer code takes the taps generated above and builds
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* a set of filters. The set contains <EM>M</EM> number of filters
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* and each filter contains ceil(taps.size()/decim) number of taps.
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* Each tap from the filter prototype is sequentially inserted into
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* the next filter. When all of the input taps are used, the remaining
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* filters in the filterbank are filled out with 0's to make sure each
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* filter has the same number of taps.
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*
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* Each filter operates using the gr::blocks::fir_filter_XXX
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* classs of GNU Radio, which takes the input stream at <EM>i</EM>
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* and performs the inner product calculation to <EM>i+(n-1)</EM>
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* where <EM>n</EM> is the number of filter taps. To efficiently
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* handle this in the GNU Radio structure, each filter input must
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* come from its own input stream. So the channelizer must be
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* provided with <EM>M</EM> streams where the input stream has
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* been deinterleaved. This is most easily done using the
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* gr::blocks::stream_to_streams block.
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*
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* The output is then produced as a vector, where index <EM>i</EM>
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* in the vector is the next sample from the <EM>i</EM>th
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* channel. This is most easily handled by sending the output to a
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* gr::blocks::vector_to_streams block to handle the conversion
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* and passing <EM>M</EM> streams out.
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*
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* The input and output formatting is done using a hier_block2 called
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* pfb_channelizer_ccf. This can take in a single stream and outputs
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* <EM>M</EM> streams based on the behavior described above.
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*
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* The filter's taps should be based on the input sampling rate.
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*
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* For example, using the GNU Radio's firdes utility to building
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* filters, we build a low-pass filter with a sampling rate of
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* <EM>fs</EM>, a 3-dB bandwidth of <EM>BW</EM> and a transition
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* bandwidth of <EM>TB</EM>. We can also specify the out-of-band
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* attenuation to use, <EM>ATT</EM>, and the filter window
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* function (a Blackman-harris window in this case). The first input
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* is the gain of the filter, which we specify here as unity.
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*
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* <B><EM>self._taps = filter.firdes.low_pass_2(1, fs, BW, TB,
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* attenuation_dB=ATT, window=filter.firdes.WIN_BLACKMAN_hARRIS)</EM></B>
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*
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* The filter output can also be overs ampled. The over sampling rate
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* is the ratio of the the actual output sampling rate to the normal
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* output sampling rate. It must be rationally related to the number
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* of channels as N/i for i in [1,N], which gives an outputsample rate
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* of [fs/N, fs] where fs is the input sample rate and N is the number
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* of channels.
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*
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* For example, for 6 channels with fs = 6000 Hz, the normal rate is
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* 6000/6 = 1000 Hz. Allowable oversampling rates are 6/6, 6/5, 6/4,
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* 6/3, 6/2, and 6/1 where the output sample rate of a 6/1 oversample
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* ratio is 6000 Hz, or 6 times the normal 1000 Hz. A rate of 6/5 = 1.2,
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* so the output rate would be 1200 Hz.
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*
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* The theory behind this block can be found in Chapter 6 of
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* the following book.
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*
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* <B><EM>f. harris, "Multirate Signal Processing for Communication
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* Systems," Upper Saddle River, NJ: Prentice Hall, Inc. 2004.</EM></B>
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*
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* When dealing with oversampling, the above book is still a good
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* reference along with this paper:
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*
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* <B><EM>E. Venosa, X. Chen, and fred harris, “Polyphase analysis
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* filter bank down-converts unequal channel bandwidths with
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* arbitrary center frequencies - design I,” in SDR’10-WinnComm,
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* 2010.</EM></B>
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*/
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class
FILTER_API
pfb_channelizer_ccf
:
virtual
public
block
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{
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public
:
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// gr::filter::pfb_channelizer_ccf::sptr
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typedef
boost::shared_ptr<pfb_channelizer_ccf>
sptr
;
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/*!
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* Build the polyphase filterbank decimator.
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* \param numchans (unsigned integer) Specifies the number of
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* channels <EM>M</EM>
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* \param taps (vector/list of floats) The prototype filter to
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* populate the filterbank.
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* \param oversample_rate (float) The over sampling rate is the
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* ratio of the the actual output
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* sampling rate to the normal
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* output sampling rate. It must
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* be rationally related to the
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* number of channels as N/i for
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* i in [1,N], which gives an
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* outputsample rate of [fs/N,
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* fs] where fs is the input
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* sample rate and N is the
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* number of channels.
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*
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* For example, for 6 channels
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* with fs = 6000 Hz, the normal
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* rateis 6000/6 = 1000
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* Hz. Allowable oversampling
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* rates are 6/6, 6/5, 6/4, 6/3,
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* 6/2, and 6/1 where the output
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* sample rate of a 6/1
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* oversample ratio is 6000 Hz,
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* or 6 times the normal 1000 Hz.
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*/
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static
sptr
make(
unsigned
int
numchans,
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const
std::vector<float> &
taps
,
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float
oversample_rate);
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/*!
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* Resets the filterbank's filter taps with the new prototype filter
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* \param taps (vector/list of floats) The prototype filter to populate the filterbank.
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*/
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virtual
void
set_taps(
const
std::vector<float> &
taps
) = 0;
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/*!
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* Print all of the filterbank taps to screen.
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*/
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virtual
void
print_taps() = 0;
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/*!
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* Return a vector<vector<>> of the filterbank taps
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*/
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virtual
std::vector<std::vector<float> >
taps
()
const
= 0;
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/*!
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* Set the channel map. Channels are numbers as:
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*
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* N/2+1 | ... | N-1 | 0 | 1 | 2 | ... | N/2
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* <------------------- 0 -------------------->
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* freq
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*
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* So output stream 0 comes from channel 0, etc. Setting a new
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* channel map allows the user to specify which channel in frequency
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* he/she wants to got to which output stream.
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*
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* The map should have the same number of elements as the number
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* of output connections from the block. The minimum value of
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* the map is 0 (for the 0th channel) and the maximum number is
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* N-1 where N is the number of channels.
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*
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* We specify M as the number of output connections made where M
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* <= N, so only M out of N channels are driven to an output
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* stream. The number of items in the channel map should be at
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* least M long. If there are more channels specified, any value
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* in the map over M-1 will be ignored. If the size of the map
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* is less than M the behavior is unknown (we don't wish to
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* check every entry into the work function).
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*
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* This means that if the channelizer is splitting the signal up
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* into N channels but only M channels are specified in the map
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* (where M <= N), then M output streams must be connected and
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* the map and the channel numbers used must be less than
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* N-1. Output channel number can be reused, too. By default,
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* the map is [0...M-1] with M = N.
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*/
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virtual
void
set_channel_map(
const
std::vector<int> &
map
) = 0;
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/*!
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* Gets the current channel map.
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*/
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virtual
std::vector<int> channel_map()
const
= 0;
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};
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}
/* namespace filter */
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}
/* namespace gr */
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#endif
/* INCLUDED_FILTER_PFB_CHANNELIZER_CCF_H */
gr-filter
include
gnuradio
filter
pfb_channelizer_ccf.h
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