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Staging: echo: fix up remaining checkpatch.pl issues
It's all just minor comment spacing issues. This patch fixes up the remaining ones and now the code is checkpatch.pl clean. Cc: Steve Underwood <steveu@coppice.org> Cc: David Rowe <david@rowetel.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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2 changed files with 96 additions and 70 deletions
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@ -1,5 +1,4 @@
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TODO:
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- checkpatch.pl cleanups
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- handle bit_operations.h (merge in or make part of common code?)
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- remove proc interface, only use echo.h interface (proc interface is
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racy and not correct.)
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@ -82,9 +82,9 @@
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[2] The classic, very useful paper that tells you how to
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actually build a real world echo canceller:
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Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
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Echo Canceller with a TMS320020,
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http://www.rowetel.com/images/echo/spra129.pdf
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Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
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Echo Canceller with a TMS320020,
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http://www.rowetel.com/images/echo/spra129.pdf
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[3] I have written a series of blog posts on this work, here is
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Part 1: http://www.rowetel.com/blog/?p=18
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@ -92,7 +92,7 @@
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[4] The source code http://svn.rowetel.com/software/oslec/
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[5] A nice reference on LMS filters:
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http://en.wikipedia.org/wiki/Least_mean_squares_filter
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http://en.wikipedia.org/wiki/Least_mean_squares_filter
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Credits:
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@ -102,21 +102,18 @@
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Mark, Pawel, and Pavel.
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*/
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#include <linux/kernel.h> /* We're doing kernel work */
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#include <linux/kernel.h>
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#include <linux/module.h>
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#include <linux/slab.h>
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#include "bit_operations.h"
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#include "echo.h"
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#define MIN_TX_POWER_FOR_ADAPTION 64
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#define MIN_RX_POWER_FOR_ADAPTION 64
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#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
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#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
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#define MIN_TX_POWER_FOR_ADAPTION 64
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#define MIN_RX_POWER_FOR_ADAPTION 64
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#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
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#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
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/*-----------------------------------------------------------------------*\
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FUNCTIONS
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\*-----------------------------------------------------------------------*/
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/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
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@ -328,7 +325,7 @@ void oslec_snapshot(struct oslec_state *ec)
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}
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EXPORT_SYMBOL_GPL(oslec_snapshot);
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/* Dual Path Echo Canceller ------------------------------------------------*/
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/* Dual Path Echo Canceller */
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int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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{
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@ -336,9 +333,11 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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int clean_bg;
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int tmp, tmp1;
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/* Input scaling was found be required to prevent problems when tx
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starts clipping. Another possible way to handle this would be the
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filter coefficent scaling. */
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/*
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* Input scaling was found be required to prevent problems when tx
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* starts clipping. Another possible way to handle this would be the
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* filter coefficent scaling.
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*/
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ec->tx = tx;
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ec->rx = rx;
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@ -346,33 +345,40 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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rx >>= 1;
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/*
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Filter DC, 3dB point is 160Hz (I think), note 32 bit precision required
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otherwise values do not track down to 0. Zero at DC, Pole at (1-Beta)
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only real axis. Some chip sets (like Si labs) don't need
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this, but something like a $10 X100P card does. Any DC really slows
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down convergence.
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Note: removes some low frequency from the signal, this reduces
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the speech quality when listening to samples through headphones
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but may not be obvious through a telephone handset.
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Note that the 3dB frequency in radians is approx Beta, e.g. for
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Beta = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
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* Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
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* required otherwise values do not track down to 0. Zero at DC, Pole
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* at (1-Beta) only real axis. Some chip sets (like Si labs) don't
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* need this, but something like a $10 X100P card does. Any DC really
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* slows down convergence.
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*
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* Note: removes some low frequency from the signal, this reduces the
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* speech quality when listening to samples through headphones but may
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* not be obvious through a telephone handset.
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*
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* Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
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* = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
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*/
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if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
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tmp = rx << 15;
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#if 1
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/* Make sure the gain of the HPF is 1.0. This can still saturate a little under
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impulse conditions, and it might roll to 32768 and need clipping on sustained peak
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level signals. However, the scale of such clipping is small, and the error due to
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any saturation should not markedly affect the downstream processing. */
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/*
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* Make sure the gain of the HPF is 1.0. This can still
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* saturate a little under impulse conditions, and it might
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* roll to 32768 and need clipping on sustained peak level
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* signals. However, the scale of such clipping is small, and
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* the error due to any saturation should not markedly affect
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* the downstream processing.
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*/
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tmp -= (tmp >> 4);
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#endif
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ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
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/* hard limit filter to prevent clipping. Note that at this stage
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rx should be limited to +/- 16383 due to right shift above */
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/*
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* hard limit filter to prevent clipping. Note that at this
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* stage rx should be limited to +/- 16383 due to right shift
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* above
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*/
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tmp1 = ec->rx_1 >> 15;
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if (tmp1 > 16383)
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tmp1 = 16383;
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@ -407,7 +413,7 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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ec->Lrxacc += abs(rx) - ec->Lrx;
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ec->Lrx = (ec->Lrxacc + (1 << 4)) >> 5;
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/* Foreground filter --------------------------------------------------- */
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/* Foreground filter */
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ec->fir_state.coeffs = ec->fir_taps16[0];
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echo_value = fir16(&ec->fir_state, tx);
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ec->Lcleanacc += abs(ec->clean) - ec->Lclean;
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ec->Lclean = (ec->Lcleanacc + (1 << 4)) >> 5;
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/* Background filter --------------------------------------------------- */
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/* Background filter */
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echo_value = fir16(&ec->fir_state_bg, tx);
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clean_bg = rx - echo_value;
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ec->Lclean_bgacc += abs(clean_bg) - ec->Lclean_bg;
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ec->Lclean_bg = (ec->Lclean_bgacc + (1 << 4)) >> 5;
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/* Background Filter adaption ----------------------------------------- */
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/* Background Filter adaption */
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/* Almost always adap bg filter, just simple DT and energy
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detection to minimise adaption in cases of strong double talk.
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@ -483,7 +489,7 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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if (ec->nonupdate_dwell)
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ec->nonupdate_dwell--;
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/* Transfer logic ------------------------------------------------------ */
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/* Transfer logic */
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/* These conditions are from the dual path paper [1], I messed with
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them a bit to improve performance. */
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/* (ec->Lclean_bg < 0.125*ec->Ltx) */
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(8 * ec->Lclean_bg < ec->Ltx)) {
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if (ec->cond_met == 6) {
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/* BG filter has had better results for 6 consecutive samples */
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/*
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* BG filter has had better results for 6 consecutive
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* samples
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*/
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ec->adapt = 1;
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memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
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ec->taps * sizeof(int16_t));
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} else
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ec->cond_met = 0;
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/* Non-Linear Processing --------------------------------------------------- */
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/* Non-Linear Processing */
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ec->clean_nlp = ec->clean;
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if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
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/* Non-linear processor - a fancy way to say "zap small signals, to avoid
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residual echo due to (uLaw/ALaw) non-linearity in the channel.". */
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/*
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* Non-linear processor - a fancy way to say "zap small
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* signals, to avoid residual echo due to (uLaw/ALaw)
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* non-linearity in the channel.".
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*/
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if ((16 * ec->Lclean < ec->Ltx)) {
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/* Our e/c has improved echo by at least 24 dB (each factor of 2 is 6dB,
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so 2*2*2*2=16 is the same as 6+6+6+6=24dB) */
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/*
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* Our e/c has improved echo by at least 24 dB (each
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* factor of 2 is 6dB, so 2*2*2*2=16 is the same as
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* 6+6+6+6=24dB)
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*/
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if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
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ec->cng_level = ec->Lbgn;
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/* Very elementary comfort noise generation. Just random
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numbers rolled off very vaguely Hoth-like. DR: This
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noise doesn't sound quite right to me - I suspect there
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are some overlfow issues in the filtering as it's too
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"crackly". TODO: debug this, maybe just play noise at
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high level or look at spectrum.
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/*
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* Very elementary comfort noise generation.
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* Just random numbers rolled off very vaguely
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* Hoth-like. DR: This noise doesn't sound
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* quite right to me - I suspect there are some
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* overlfow issues in the filtering as it's too
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* "crackly".
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* TODO: debug this, maybe just play noise at
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* high level or look at spectrum.
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*/
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ec->cng_rndnum =
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if (ec->clean_nlp < -ec->Lbgn)
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ec->clean_nlp = -ec->Lbgn;
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} else {
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/* just mute the residual, doesn't sound very good, used mainly
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in G168 tests */
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/*
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* just mute the residual, doesn't sound very
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* good, used mainly in G168 tests
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*/
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ec->clean_nlp = 0;
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}
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} else {
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/* Background noise estimator. I tried a few algorithms
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here without much luck. This very simple one seems to
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work best, we just average the level using a slow (1 sec
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time const) filter if the current level is less than a
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(experimentally derived) constant. This means we dont
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include high level signals like near end speech. When
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combined with CNG or especially CLIP seems to work OK.
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/*
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* Background noise estimator. I tried a few
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* algorithms here without much luck. This very simple
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* one seems to work best, we just average the level
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* using a slow (1 sec time const) filter if the
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* current level is less than a (experimentally
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* derived) constant. This means we dont include high
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* level signals like near end speech. When combined
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* with CNG or especially CLIP seems to work OK.
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*/
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if (ec->Lclean < 40) {
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ec->Lbgn_acc += abs(ec->clean) - ec->Lbgn;
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It can also help by removing and DC in the tx signal. DC is bad
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for LMS algorithms.
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This is one of the classic DC removal filters, adjusted to provide sufficient
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bass rolloff to meet the above requirement to protect hybrids from things that
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upset them. The difference between successive samples produces a lousy HPF, and
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then a suitably placed pole flattens things out. The final result is a nicely
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rolled off bass end. The filtering is implemented with extended fractional
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precision, which noise shapes things, giving very clean DC removal.
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This is one of the classic DC removal filters, adjusted to provide
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sufficient bass rolloff to meet the above requirement to protect hybrids
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from things that upset them. The difference between successive samples
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produces a lousy HPF, and then a suitably placed pole flattens things out.
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The final result is a nicely rolled off bass end. The filtering is
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implemented with extended fractional precision, which noise shapes things,
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giving very clean DC removal.
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*/
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int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
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if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
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tmp = tx << 15;
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#if 1
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/* Make sure the gain of the HPF is 1.0. The first can still saturate a little under
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impulse conditions, and it might roll to 32768 and need clipping on sustained peak
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level signals. However, the scale of such clipping is small, and the error due to
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any saturation should not markedly affect the downstream processing. */
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/*
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* Make sure the gain of the HPF is 1.0. The first can still
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* saturate a little under impulse conditions, and it might
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* roll to 32768 and need clipping on sustained peak level
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* signals. However, the scale of such clipping is small, and
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* the error due to any saturation should not markedly affect
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* the downstream processing.
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*/
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tmp -= (tmp >> 4);
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#endif
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ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;
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