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Staging: echo: fix up remaining checkpatch.pl issues

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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>
This commit is contained in:
Greg Kroah-Hartman 2009-08-10 10:45:25 -07:00
parent 817bb33433
commit 49bb9e6d75
2 changed files with 96 additions and 70 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
drivers/staging/echo/TODO
View File

@ -1,5 +1,4 @@
TODO:
- checkpatch.pl cleanups
- handle bit_operations.h (merge in or make part of common code?)
- remove proc interface, only use echo.h interface (proc interface is
racy and not correct.)

165
drivers/staging/echo/echo.c
View File

@ -82,9 +82,9 @@
[2] The classic, very useful paper that tells you how to
actually build a real world echo canceller:
Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
Echo Canceller with a TMS320020,
http://www.rowetel.com/images/echo/spra129.pdf
Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
Echo Canceller with a TMS320020,
http://www.rowetel.com/images/echo/spra129.pdf
[3] I have written a series of blog posts on this work, here is
Part 1: http://www.rowetel.com/blog/?p=18
@ -92,7 +92,7 @@
[4] The source code http://svn.rowetel.com/software/oslec/
[5] A nice reference on LMS filters:
http://en.wikipedia.org/wiki/Least_mean_squares_filter
http://en.wikipedia.org/wiki/Least_mean_squares_filter
Credits:
@ -102,21 +102,18 @@
Mark, Pawel, and Pavel.
*/
#include <linux/kernel.h> /* We're doing kernel work */
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/slab.h>
#include "bit_operations.h"
#include "echo.h"
#define MIN_TX_POWER_FOR_ADAPTION 64
#define MIN_RX_POWER_FOR_ADAPTION 64
#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
#define MIN_TX_POWER_FOR_ADAPTION 64
#define MIN_RX_POWER_FOR_ADAPTION 64
#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
/*-----------------------------------------------------------------------*\
FUNCTIONS
\*-----------------------------------------------------------------------*/
/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
@ -328,7 +325,7 @@ void oslec_snapshot(struct oslec_state *ec)
}
EXPORT_SYMBOL_GPL(oslec_snapshot);
/* Dual Path Echo Canceller ------------------------------------------------*/
/* Dual Path Echo Canceller */
int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
{
@ -336,9 +333,11 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
int clean_bg;
int tmp, tmp1;
/* Input scaling was found be required to prevent problems when tx
starts clipping. Another possible way to handle this would be the
filter coefficent scaling. */
/*
* Input scaling was found be required to prevent problems when tx
* starts clipping. Another possible way to handle this would be the
* filter coefficent scaling.
*/
ec->tx = tx;
ec->rx = rx;
@ -346,33 +345,40 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
rx >>= 1;
/*
Filter DC, 3dB point is 160Hz (I think), note 32 bit precision required
otherwise values do not track down to 0. Zero at DC, Pole at (1-Beta)
only real axis. Some chip sets (like Si labs) don't need
this, but something like a $10 X100P card does. Any DC really slows
down convergence.
Note: removes some low frequency from the signal, this reduces
the speech quality when listening to samples through headphones
but may not be obvious through a telephone handset.
Note that the 3dB frequency in radians is approx Beta, e.g. for
Beta = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
* Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
* required otherwise values do not track down to 0. Zero at DC, Pole
* at (1-Beta) only real axis. Some chip sets (like Si labs) don't
* need this, but something like a $10 X100P card does. Any DC really
* slows down convergence.
*
* Note: removes some low frequency from the signal, this reduces the
* speech quality when listening to samples through headphones but may
* not be obvious through a telephone handset.
*
* Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
* = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
*/
if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
tmp = rx << 15;
#if 1
/* Make sure the gain of the HPF is 1.0. This can still saturate a little under
impulse conditions, and it might roll to 32768 and need clipping on sustained peak
level signals. However, the scale of such clipping is small, and the error due to
any saturation should not markedly affect the downstream processing. */
/*
* Make sure the gain of the HPF is 1.0. This can still
* saturate a little under impulse conditions, and it might
* roll to 32768 and need clipping on sustained peak level
* signals. However, the scale of such clipping is small, and
* the error due to any saturation should not markedly affect
* the downstream processing.
*/
tmp -= (tmp >> 4);
#endif
ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
/* hard limit filter to prevent clipping. Note that at this stage
rx should be limited to +/- 16383 due to right shift above */
/*
* hard limit filter to prevent clipping. Note that at this
* stage rx should be limited to +/- 16383 due to right shift
* above
*/
tmp1 = ec->rx_1 >> 15;
if (tmp1 > 16383)
tmp1 = 16383;
@ -407,7 +413,7 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
ec->Lrxacc += abs(rx) - ec->Lrx;
ec->Lrx = (ec->Lrxacc + (1 << 4)) >> 5;
/* Foreground filter --------------------------------------------------- */
/* Foreground filter */
ec->fir_state.coeffs = ec->fir_taps16[0];
echo_value = fir16(&ec->fir_state, tx);
@ -415,14 +421,14 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
ec->Lcleanacc += abs(ec->clean) - ec->Lclean;
ec->Lclean = (ec->Lcleanacc + (1 << 4)) >> 5;
/* Background filter --------------------------------------------------- */
/* Background filter */
echo_value = fir16(&ec->fir_state_bg, tx);
clean_bg = rx - echo_value;
ec->Lclean_bgacc += abs(clean_bg) - ec->Lclean_bg;
ec->Lclean_bg = (ec->Lclean_bgacc + (1 << 4)) >> 5;
/* Background Filter adaption ----------------------------------------- */
/* Background Filter adaption */
/* Almost always adap bg filter, just simple DT and energy
detection to minimise adaption in cases of strong double talk.
@ -483,7 +489,7 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
if (ec->nonupdate_dwell)
ec->nonupdate_dwell--;
/* Transfer logic ------------------------------------------------------ */
/* Transfer logic */
/* These conditions are from the dual path paper [1], I messed with
them a bit to improve performance. */
@ -495,7 +501,10 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
/* (ec->Lclean_bg < 0.125*ec->Ltx) */
(8 * ec->Lclean_bg < ec->Ltx)) {
if (ec->cond_met == 6) {
/* BG filter has had better results for 6 consecutive samples */
/*
* BG filter has had better results for 6 consecutive
* samples
*/
ec->adapt = 1;
memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
ec->taps * sizeof(int16_t));
@ -504,25 +513,34 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
} else
ec->cond_met = 0;
/* Non-Linear Processing --------------------------------------------------- */
/* Non-Linear Processing */
ec->clean_nlp = ec->clean;
if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
/* Non-linear processor - a fancy way to say "zap small signals, to avoid
residual echo due to (uLaw/ALaw) non-linearity in the channel.". */
/*
* Non-linear processor - a fancy way to say "zap small
* signals, to avoid residual echo due to (uLaw/ALaw)
* non-linearity in the channel.".
*/
if ((16 * ec->Lclean < ec->Ltx)) {
/* Our e/c has improved echo by at least 24 dB (each factor of 2 is 6dB,
so 2*2*2*2=16 is the same as 6+6+6+6=24dB) */
/*
* Our e/c has improved echo by at least 24 dB (each
* factor of 2 is 6dB, so 2*2*2*2=16 is the same as
* 6+6+6+6=24dB)
*/
if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
ec->cng_level = ec->Lbgn;
/* Very elementary comfort noise generation. Just random
numbers rolled off very vaguely Hoth-like. DR: This
noise doesn't sound quite right to me - I suspect there
are some overlfow issues in the filtering as it's too
"crackly". TODO: debug this, maybe just play noise at
high level or look at spectrum.
/*
* Very elementary comfort noise generation.
* Just random numbers rolled off very vaguely
* Hoth-like. DR: This noise doesn't sound
* quite right to me - I suspect there are some
* overlfow issues in the filtering as it's too
* "crackly".
* TODO: debug this, maybe just play noise at
* high level or look at spectrum.
*/
ec->cng_rndnum =
@ -540,18 +558,22 @@ int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
if (ec->clean_nlp < -ec->Lbgn)
ec->clean_nlp = -ec->Lbgn;
} else {
/* just mute the residual, doesn't sound very good, used mainly
in G168 tests */
/*
* just mute the residual, doesn't sound very
* good, used mainly in G168 tests
*/
ec->clean_nlp = 0;
}
} else {
/* Background noise estimator. I tried a few algorithms
here without much luck. This very simple one seems to
work best, we just average the level using a slow (1 sec
time const) filter if the current level is less than a
(experimentally derived) constant. This means we dont
include high level signals like near end speech. When
combined with CNG or especially CLIP seems to work OK.
/*
* Background noise estimator. I tried a few
* algorithms here without much luck. This very simple
* one seems to work best, we just average the level
* using a slow (1 sec time const) filter if the
* current level is less than a (experimentally
* derived) constant. This means we dont include high
* level signals like near end speech. When combined
* with CNG or especially CLIP seems to work OK.
*/
if (ec->Lclean < 40) {
ec->Lbgn_acc += abs(ec->clean) - ec->Lbgn;
@ -587,12 +609,13 @@ EXPORT_SYMBOL_GPL(oslec_update);
It can also help by removing and DC in the tx signal. DC is bad
for LMS algorithms.
This is one of the classic DC removal filters, adjusted to provide sufficient
bass rolloff to meet the above requirement to protect hybrids from things that
upset them. The difference between successive samples produces a lousy HPF, and
then a suitably placed pole flattens things out. The final result is a nicely
rolled off bass end. The filtering is implemented with extended fractional
precision, which noise shapes things, giving very clean DC removal.
This is one of the classic DC removal filters, adjusted to provide
sufficient bass rolloff to meet the above requirement to protect hybrids
from things that upset them. The difference between successive samples
produces a lousy HPF, and then a suitably placed pole flattens things out.
The final result is a nicely rolled off bass end. The filtering is
implemented with extended fractional precision, which noise shapes things,
giving very clean DC removal.
*/
int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
@ -602,10 +625,14 @@ int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
tmp = tx << 15;
#if 1
/* Make sure the gain of the HPF is 1.0. The first can still saturate a little under
impulse conditions, and it might roll to 32768 and need clipping on sustained peak
level signals. However, the scale of such clipping is small, and the error due to
any saturation should not markedly affect the downstream processing. */
/*
* Make sure the gain of the HPF is 1.0. The first can still
* saturate a little under impulse conditions, and it might
* roll to 32768 and need clipping on sustained peak level
* signals. However, the scale of such clipping is small, and
* the error due to any saturation should not markedly affect
* the downstream processing.
*/
tmp -= (tmp >> 4);
#endif
ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;