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|
/*
* ReplayGainAnalysis - analyzes input samples and give the recommended dB change
* Copyright (C) 2001-2009 David Robinson and Glen Sawyer
* Improvements and optimizations added by Frank Klemm, and by Marcel Müller
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* concept and filter values by David Robinson (David@Robinson.org)
* -- blame him if you think the idea is flawed
* original coding by Glen Sawyer (mp3gain@hotmail.com)
* -- blame him if you think this runs too slowly, or the coding is otherwise flawed
*
* lots of code improvements by Frank Klemm ( http://www.uni-jena.de/~pfk/mpp/ )
* -- credit him for all the _good_ programming ;)
*
*
* For an explanation of the concepts and the basic algorithms involved, go to:
* http://www.replaygain.org/
*/
/*
* Here's the deal. Call
*
* InitGainAnalysis ( long samplefreq );
*
* to initialize everything. Call
*
* AnalyzeSamples ( const Float_t* left_samples,
* const Float_t* right_samples,
* size_t num_samples,
* int num_channels );
*
* as many times as you want, with as many or as few samples as you want.
* If mono, pass the sample buffer in through left_samples, leave
* right_samples NULL, and make sure num_channels = 1.
*
* GetGain()
*
* will return the recommended dB level change for all samples analyzed
* SINCE THE LAST TIME you called InitGainAnalysis().
*
*/
/*
* So here's the main source of potential code confusion:
*
* The filters applied to the incoming samples are IIR filters,
* meaning they rely on up to <filter order> number of previous samples
* AND up to <filter order> number of previous filtered samples.
*
* I set up the AnalyzeSamples routine to minimize memory usage and interface
* complexity. The speed isn't compromised too much (I don't think), but the
* internal complexity is higher than it should be for such a relatively
* simple routine.
*
* Optimization/clarity suggestions are welcome.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "audio_gain_analysis.h"
const Float_t cMarkAdAudioGainAnalysis::ABYule[12][2*YULE_ORDER + 1] =
{
{0.006471345933032, -7.22103125152679, -0.02567678242161, 24.7034187975904, 0.049805860704367, -52.6825833623896, -0.05823001743528, 77.4825736677539, 0.040611847441914, -82.0074753444205, -0.010912036887501, 63.1566097101925, -0.00901635868667, -34.889569769245, 0.012448886238123, 13.2126852760198, -0.007206683749426, -3.09445623301669, 0.002167156433951, 0.340344741393305, -0.000261819276949},
{0.015415414474287, -7.19001570087017, -0.07691359399407, 24.4109412087159, 0.196677418516518, -51.6306373580801, -0.338855114128061, 75.3978476863163, 0.430094579594561, -79.4164552507386, -0.415015413747894, 61.0373661948115, 0.304942508151101, -33.7446462547014, -0.166191795926663, 12.8168791146274, 0.063198189938739, -3.01332198541437, -0.015003978694525, 0.223619893831468, 0.001748085184539},
{0.021776466467053, -5.74819833657784, -0.062376961003801, 16.246507961894, 0.107731165328514, -29.9691822642542, -0.150994515142316, 40.027597579378, 0.170334807313632, -40.3209196052655, -0.157984942890531, 30.8542077487718, 0.121639833268721, -17.5965138737281, -0.074094040816409, 7.10690214103873, 0.031282852041061, -1.82175564515191, -0.00755421235941, 0.223619893831468, 0.00117925454213 },
{0.03857599435200, -3.84664617118067, -0.02160367184185, 7.81501653005538, -0.00123395316851, -11.34170355132042, -0.00009291677959, 13.05504219327545, -0.01655260341619, -12.28759895145294, 0.02161526843274, 9.48293806319790, -0.02074045215285, -5.87257861775999, 0.00594298065125, 2.75465861874613, 0.00306428023191, -0.86984376593551, 0.00012025322027, 0.13919314567432, 0.00288463683916 },
{0.05418656406430, -3.47845948550071, -0.02911007808948, 6.36317777566148, -0.00848709379851, -8.54751527471874, -0.00851165645469, 9.47693607801280, -0.00834990904936, -8.81498681370155, 0.02245293253339, 6.85401540936998, -0.02596338512915, -4.39470996079559, 0.01624864962975, 2.19611684890774, -0.00240879051584, -0.75104302451432, 0.00674613682247, 0.13149317958808, -0.00187763777362 },
{0.15457299681924, -2.37898834973084, -0.09331049056315, 2.84868151156327, -0.06247880153653, -2.64577170229825, 0.02163541888798, 2.23697657451713, -0.05588393329856, -1.67148153367602, 0.04781476674921, 1.00595954808547, 0.00222312597743, -0.45953458054983, 0.03174092540049, 0.16378164858596, -0.01390589421898, -0.05032077717131, 0.00651420667831, 0.02347897407020, -0.00881362733839 },
{0.30296907319327, -1.61273165137247, -0.22613988682123, 1.07977492259970, -0.08587323730772, -0.25656257754070, 0.03282930172664, -0.16276719120440, -0.00915702933434, -0.22638893773906, -0.02364141202522, 0.39120800788284, -0.00584456039913, -0.22138138954925, 0.06276101321749, 0.04500235387352, -0.00000828086748, 0.02005851806501, 0.00205861885564, 0.00302439095741, -0.02950134983287 },
{0.33642304856132, -1.49858979367799, -0.25572241425570, 0.87350271418188, -0.11828570177555, 0.12205022308084, 0.11921148675203, -0.80774944671438, -0.07834489609479, 0.47854794562326, -0.00469977914380, -0.12453458140019, -0.00589500224440, -0.04067510197014, 0.05724228140351, 0.08333755284107, 0.00832043980773, -0.04237348025746, -0.01635381384540, 0.02977207319925, -0.01760176568150 },
{0.44915256608450, -0.62820619233671, -0.14351757464547, 0.29661783706366, -0.22784394429749, -0.37256372942400, -0.01419140100551, 0.00213767857124, 0.04078262797139, -0.42029820170918, -0.12398163381748, 0.22199650564824, 0.04097565135648, 0.00613424350682, 0.10478503600251, 0.06747620744683, -0.01863887810927, 0.05784820375801, -0.03193428438915, 0.03222754072173, 0.00541907748707 },
{0.56619470757641, -1.04800335126349, -0.75464456939302, 0.29156311971249, 0.16242137742230, -0.26806001042947, 0.16744243493672, 0.00819999645858, -0.18901604199609, 0.45054734505008, 0.30931782841830, -0.33032403314006, -0.27562961986224, 0.06739368333110, 0.00647310677246, -0.04784254229033, 0.08647503780351, 0.01639907836189, -0.03788984554840, 0.01807364323573, -0.00588215443421 },
{0.58100494960553, -0.51035327095184, -0.53174909058578, -0.31863563325245, -0.14289799034253, -0.20256413484477, 0.17520704835522, 0.14728154134330, 0.02377945217615, 0.38952639978999, 0.15558449135573, -0.23313271880868, -0.25344790059353, -0.05246019024463, 0.01628462406333, -0.02505961724053, 0.06920467763959, 0.02442357316099, -0.03721611395801, 0.01818801111503, -0.00749618797172 },
{0.53648789255105, -0.25049871956020, -0.42163034350696, -0.43193942311114, -0.00275953611929, -0.03424681017675, 0.04267842219415, -0.04678328784242, -0.10214864179676, 0.26408300200955, 0.14590772289388, 0.15113130533216, -0.02459864859345, -0.17556493366449, -0.11202315195388, -0.18823009262115, -0.04060034127000, 0.05477720428674, 0.04788665548180, 0.04704409688120, -0.02217936801134 }
};
const Float_t cMarkAdAudioGainAnalysis::ABButter[12][2*BUTTER_ORDER + 1] =
{
{0.99308203517541, -1.98611621154089, -1.98616407035082, 0.986211929160751, 0.99308203517541 },
{0.992472550461293,-1.98488843762334, -1.98494510092258, 0.979389350028798, 0.992472550461293},
{0.989641019334721,-1.97917472731008, -1.97928203866944, 0.979389350028798, 0.989641019334721},
{0.98621192462708, -1.97223372919527, -1.97242384925416, 0.97261396931306, 0.98621192462708 },
{0.98500175787242, -1.96977855582618, -1.97000351574484, 0.97022847566350, 0.98500175787242 },
{0.97938932735214, -1.95835380975398, -1.95877865470428, 0.95920349965459, 0.97938932735214 },
{0.97531843204928, -1.95002759149878, -1.95063686409857, 0.95124613669835, 0.97531843204928 },
{0.97316523498161, -1.94561023566527, -1.94633046996323, 0.94705070426118, 0.97316523498161 },
{0.96454515552826, -1.92783286977036, -1.92909031105652, 0.93034775234268, 0.96454515552826 },
{0.96009142950541, -1.91858953033784, -1.92018285901082, 0.92177618768381, 0.96009142950541 },
{0.95856916599601, -1.91542108074780, -1.91713833199203, 0.91885558323625, 0.95856916599601 },
{0.94597685600279, -1.88903307939452, -1.89195371200558, 0.89487434461664, 0.94597685600279 }
};
void cMarkAdAudioGainAnalysis::filterYule (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel)
{
// for each filter:
// [0] 48 kHz, [1] 44.1 kHz, [2] 32 kHz, [3] 24 kHz, [4] 22050 Hz,
// [5] 16 kHz, [6] 12 kHz, [7] is 11025 Hz, [8] 8 kHz
// When calling these filter procedures, make sure that ip[-order] and op[-order] point to real data!
while (nSamples--)
{
*output = 1e-10 /* 1e-10 is a hack to avoid slowdown because of denormals */
+ input [0] * kernel[0]
- output[-1] * kernel[1]
+ input [-1] * kernel[2]
- output[-2] * kernel[3]
+ input [-2] * kernel[4]
- output[-3] * kernel[5]
+ input [-3] * kernel[6]
- output[-4] * kernel[7]
+ input [-4] * kernel[8]
- output[-5] * kernel[9]
+ input [-5] * kernel[10]
- output[-6] * kernel[11]
+ input [-6] * kernel[12]
- output[-7] * kernel[13]
+ input [-7] * kernel[14]
- output[-8] * kernel[15]
+ input [-8] * kernel[16]
- output[-9] * kernel[17]
+ input [-9] * kernel[18]
- output[-10]* kernel[19]
+ input [-10]* kernel[20];
++output;
++input;
}
}
void cMarkAdAudioGainAnalysis::filterButter (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel)
{
while (nSamples--)
{
*output =
input [0] * kernel[0]
- output[-1] * kernel[1]
+ input [-1] * kernel[2]
- output[-2] * kernel[3]
+ input [-2] * kernel[4];
++output;
++input;
}
}
int cMarkAdAudioGainAnalysis::ResetSampleFrequency ( long samplefreq )
{
// returns a INIT_GAIN_ANALYSIS_OK if successful, INIT_GAIN_ANALYSIS_ERROR if not
int i;
// zero out initial values
for ( i = 0; i < MAX_ORDER; i++ )
linprebuf[i] = lstepbuf[i] = loutbuf[i] = rinprebuf[i] = rstepbuf[i] = routbuf[i] = 0.;
switch ( (int)(samplefreq) )
{
case 96000:
freqindex = 0;
break;
case 88200:
freqindex = 1;
break;
case 64000:
freqindex = 2;
break;
case 48000:
freqindex = 3;
break;
case 44100:
freqindex = 4;
break;
case 32000:
freqindex = 5;
break;
case 24000:
freqindex = 6;
break;
case 22050:
freqindex = 7;
break;
case 16000:
freqindex = 8;
break;
case 12000:
freqindex = 9;
break;
case 11025:
freqindex = 10;
break;
case 8000:
freqindex = 11;
break;
default:
return INIT_GAIN_ANALYSIS_ERROR;
}
sampleWindow = (int) ceil (samplefreq * RMS_WINDOW_TIME);
lsum = 0.;
rsum = 0.;
totsamp = 0;
memset ( A, 0, sizeof(A) );
return INIT_GAIN_ANALYSIS_OK;
}
int cMarkAdAudioGainAnalysis::Init(long samplefreq)
{
if (ResetSampleFrequency(samplefreq) != INIT_GAIN_ANALYSIS_OK)
{
return INIT_GAIN_ANALYSIS_ERROR;
}
linpre = linprebuf + MAX_ORDER;
rinpre = rinprebuf + MAX_ORDER;
lstep = lstepbuf + MAX_ORDER;
rstep = rstepbuf + MAX_ORDER;
lout = loutbuf + MAX_ORDER;
rout = routbuf + MAX_ORDER;
memset ( B, 0, sizeof(B) );
gnum_samples=0;
return INIT_GAIN_ANALYSIS_OK;
}
static __inline double fsqr(const double d)
{
return d*d;
}
int cMarkAdAudioGainAnalysis::AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels )
{
// returns GAIN_ANALYSIS_OK if successful, GAIN_ANALYSIS_ERROR if not
const Float_t* curleft;
const Float_t* curright;
long batchsamples;
long cursamples;
long cursamplepos;
int i;
if ( num_samples == 0 )
return GAIN_ANALYSIS_OK;
gnum_samples+=num_samples;
cursamplepos = 0;
batchsamples = (long)num_samples;
switch ( num_channels)
{
case 1:
right_samples = left_samples;
case 2:
break;
default:
return GAIN_ANALYSIS_ERROR;
}
if ( num_samples < MAX_ORDER )
{
memcpy ( linprebuf + MAX_ORDER, left_samples , num_samples * sizeof(Float_t) );
memcpy ( rinprebuf + MAX_ORDER, right_samples, num_samples * sizeof(Float_t) );
}
else
{
memcpy ( linprebuf + MAX_ORDER, left_samples, MAX_ORDER * sizeof(Float_t) );
memcpy ( rinprebuf + MAX_ORDER, right_samples, MAX_ORDER * sizeof(Float_t) );
}
while ( batchsamples > 0 )
{
cursamples = batchsamples > sampleWindow-totsamp ? sampleWindow - totsamp : batchsamples;
if ( cursamplepos < MAX_ORDER )
{
curleft = linpre+cursamplepos;
curright = rinpre+cursamplepos;
if (cursamples > MAX_ORDER - cursamplepos )
cursamples = MAX_ORDER - cursamplepos;
}
else
{
curleft = left_samples + cursamplepos;
curright = right_samples + cursamplepos;
}
YULE_FILTER ( curleft , lstep + totsamp, cursamples, ABYule[freqindex]);
YULE_FILTER ( curright, rstep + totsamp, cursamples, ABYule[freqindex]);
BUTTER_FILTER ( lstep + totsamp, lout + totsamp, cursamples, ABButter[freqindex]);
BUTTER_FILTER ( rstep + totsamp, rout + totsamp, cursamples, ABButter[freqindex]);
curleft = lout + totsamp; // Get the squared values
curright = rout + totsamp;
i = cursamples % 16;
while (i--)
{
lsum += fsqr(*curleft++);
rsum += fsqr(*curright++);
}
i = cursamples / 16;
while (i--)
{
lsum += fsqr(curleft[0])
+ fsqr(curleft[1])
+ fsqr(curleft[2])
+ fsqr(curleft[3])
+ fsqr(curleft[4])
+ fsqr(curleft[5])
+ fsqr(curleft[6])
+ fsqr(curleft[7])
+ fsqr(curleft[8])
+ fsqr(curleft[9])
+ fsqr(curleft[10])
+ fsqr(curleft[11])
+ fsqr(curleft[12])
+ fsqr(curleft[13])
+ fsqr(curleft[14])
+ fsqr(curleft[15]);
curleft += 16;
rsum += fsqr(curright[0])
+ fsqr(curright[1])
+ fsqr(curright[2])
+ fsqr(curright[3])
+ fsqr(curright[4])
+ fsqr(curright[5])
+ fsqr(curright[6])
+ fsqr(curright[7])
+ fsqr(curright[8])
+ fsqr(curright[9])
+ fsqr(curright[10])
+ fsqr(curright[11])
+ fsqr(curright[12])
+ fsqr(curright[13])
+ fsqr(curright[14])
+ fsqr(curright[15]);
curright += 16;
}
batchsamples -= cursamples;
cursamplepos += cursamples;
totsamp += cursamples;
if ( totsamp == sampleWindow ) // Get the Root Mean Square (RMS) for this set of samples
{
double val = STEPS_per_dB * 10. * log10 ( (lsum+rsum) / totsamp * 0.5 + 1.e-37 );
int ival = (int) val;
if ( ival < 0 ) ival = 0;
if ( ival >= (int)(sizeof(A)/sizeof(*A)) ) ival = sizeof(A)/sizeof(*A) - 1;
A [ival]++;
lsum = rsum = 0.;
memmove ( loutbuf , loutbuf + totsamp, MAX_ORDER * sizeof(Float_t) );
memmove ( routbuf , routbuf + totsamp, MAX_ORDER * sizeof(Float_t) );
memmove ( lstepbuf, lstepbuf + totsamp, MAX_ORDER * sizeof(Float_t) );
memmove ( rstepbuf, rstepbuf + totsamp, MAX_ORDER * sizeof(Float_t) );
totsamp = 0;
}
if ( totsamp > sampleWindow )
// somehow I really screwed up: Error in programming!
// Contact author about totsamp > sampleWindow
return GAIN_ANALYSIS_ERROR;
}
if ( num_samples < MAX_ORDER )
{
memmove ( linprebuf,linprebuf + num_samples, (MAX_ORDER-num_samples) * sizeof(Float_t) );
memmove ( rinprebuf,rinprebuf + num_samples, (MAX_ORDER-num_samples) * sizeof(Float_t) );
memcpy ( linprebuf + MAX_ORDER - num_samples, left_samples,num_samples * sizeof(Float_t) );
memcpy ( rinprebuf + MAX_ORDER - num_samples, right_samples,num_samples * sizeof(Float_t) );
}
else
{
memcpy ( linprebuf, left_samples + num_samples - MAX_ORDER, MAX_ORDER * sizeof(Float_t) );
memcpy ( rinprebuf, right_samples + num_samples - MAX_ORDER, MAX_ORDER * sizeof(Float_t) );
}
return GAIN_ANALYSIS_OK;
}
Float_t cMarkAdAudioGainAnalysis::analyzeResult ( Uint32_t* Array, size_t len )
{
Uint32_t elems;
Int32_t upper;
size_t i;
elems = 0;
for ( i = 0; i < len; i++ )
elems += Array[i];
if ( elems == 0 )
return GAIN_NOT_ENOUGH_SAMPLES;
upper = (Int32_t) ceil (elems * (1. - RMS_PERCENTILE));
for ( i = len; i-- > 0; )
{
if ( (upper -= Array[i]) <= 0 )
break;
}
return (Float_t) ((Float_t)PINK_REF - (Float_t)i / (Float_t)STEPS_per_dB);
}
Float_t cMarkAdAudioGainAnalysis::GetGain ( void )
{
return analyzeResult ( A, sizeof(A)/sizeof(*A) );
}
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