From 49bb9e6d75346980acdc43f5198032c2a0a22c2c Mon Sep 17 00:00:00 2001 From: Greg Kroah-Hartman Date: Mon, 10 Aug 2009 10:45:25 -0700 Subject: [PATCH] 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 Cc: David Rowe Signed-off-by: Greg Kroah-Hartman --- drivers/staging/echo/TODO | 1 - drivers/staging/echo/echo.c | 165 ++++++++++++++++++++++++++------------------ 2 files changed, 96 insertions(+), 70 deletions(-) diff --git a/drivers/staging/echo/TODO b/drivers/staging/echo/TODO index f6d8580..18f7b4a 100644 --- a/drivers/staging/echo/TODO +++ b/drivers/staging/echo/TODO @@ -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.) diff --git a/drivers/staging/echo/echo.c b/drivers/staging/echo/echo.c index 79d15c6..d05642e 100644 --- a/drivers/staging/echo/echo.c +++ b/drivers/staging/echo/echo.c @@ -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 /* We're doing kernel work */ +#include #include #include #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; -- 2.7.4