diff --git a/WaveGain/WaveGain.icproj b/WaveGain/WaveGain.icproj
new file mode 100644
index 0000000..2d6a88b
--- /dev/null
+++ b/WaveGain/WaveGain.icproj
@@ -0,0 +1,163 @@
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diff --git a/audio.c b/audio.c
index 8d93231..5e62e25 100644
--- a/audio.c
+++ b/audio.c
@@ -72,6 +72,20 @@
#define WRITE_D64 write_d64_le
#endif
+static unsigned char pcm_guid[16] =
+{
+ /* (00000001-0000-0010-8000-00aa00389b71) */
+
+ 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
+};
+
+static unsigned char ieee_float_guid[16] =
+{
+ /* (00000003-0000-0010-8000-00aa00389b71) */
+
+ 0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
+};
+
/* Define the supported formats here */
input_format formats[] = {
{wav_id, 12, wav_open, wav_close, "wav", N_("WAV file reader")},
@@ -587,7 +601,7 @@ int wav_id(unsigned char *buf, int len)
int wav_open(FILE *in, wavegain_opt *opt, unsigned char *oldbuf, int buflen)
{
- unsigned char buf[16];
+ unsigned char buf[81];
Int64_t len;
Int64_t current_pos;
int samplesize;
@@ -614,21 +628,20 @@ int wav_open(FILE *in, wavegain_opt *opt, unsigned char *oldbuf, int buflen)
* in size. This is incorrect, but not fatal, so we only warn about
* it instead of refusing to work with the file. Please, if you
* have a program that's creating format chunks of sizes other than
- * 16 or 18 bytes in size, report a bug to the author.
+ * 16, 18 or 40 bytes in size, report a bug to the author.
+ * (40 bytes accommodates WAVEFORMATEXTENSIBLE conforming files.)
*/
- if (len != 16 && len != 18)
+ if (len != 16 && len != 18 && len != 40)
fprintf(stderr, "Warning: INVALID format chunk in wav header.\n"
" Trying to read anyway (may not work)...\n");
- if (fread(buf, 1, 16, in) < 16) {
- fprintf(stderr, "Warning: Unexpected EOF in reading WAV header (2)\n");
- return 0;
- }
-
/* Deal with stupid broken apps. Don't use these programs.
*/
- if (len - 16 > 0 && !seek_forward(in, len - 16))
- return 0;
+
+ if (fread(buf,1,len,in) < len) {
+ fprintf(stderr, "Warning: Unexpected EOF in reading WAV header\n");
+ return 0;
+ }
format.format = READ_U16_LE(buf);
format.channels = READ_U16_LE(buf+2);
@@ -667,26 +680,58 @@ int wav_open(FILE *in, wavegain_opt *opt, unsigned char *oldbuf, int buflen)
FSEEK64(in, current_pos, SEEK_SET);
}
- if (format.format == 1) {
- samplesize = format.samplesize / 8;
+ if(format.format == WAVE_FORMAT_PCM) {
+ samplesize = format.samplesize/8;
opt->read_samples = wav_read;
/* works with current enum */
opt->format = samplesize;
}
- else if (format.format == 3) {
+ else if(format.format == WAVE_FORMAT_IEEE_FLOAT) {
samplesize = 4;
opt->read_samples = wav_ieee_read;
opt->endianness = LITTLE;
opt->format = WAV_FMT_FLOAT;
}
+ else if (format.format == WAVE_FORMAT_EXTENSIBLE) {
+ format.channel_mask = READ_U32_LE(buf+20);
+ if (format.channel_mask != SPEAKER_FRONT_LEFT || format.channel_mask != (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)) {
+ fprintf(stderr, "ERROR: Wav file is unsupported type (must be standard 1 or 2 channel PCM\n"
+ " or type 3 floating point PCM)(2)\n");
+ return 0;
+ }
+ if (memcmp(buf+24, pcm_guid, 16) == 0) {
+ samplesize = format.samplesize/8;
+ opt->read_samples = wav_read;
+ /* works with current enum */
+ opt->format = samplesize;
+ }
+ else if (memcmp(buf+24, ieee_float_guid, 16) == 0) {
+ samplesize = 4;
+ opt->read_samples = wav_ieee_read;
+ opt->endianness = LITTLE;
+ opt->format = WAV_FMT_FLOAT;
+ }
+ else {
+ fprintf(stderr, "ERROR: Wav file is unsupported type (must be standard PCM\n"
+ " or type 3 floating point PCM)(2)\n");
+ return 0;
+ }
+ }
else {
fprintf(stderr, "ERROR: Wav file is unsupported type (must be standard PCM\n"
- " or type 3 floating point PCM)\n");
+ " or type 3 floating point PCM\n");
return 0;
}
opt->samplesize = format.samplesize;
+ if(format.align != format.channels * samplesize) {
+ /* This is incorrect according to the spec. Warn loudly, then ignore
+ * this value.
+ */
+ fprintf(stderr, _("Warning: WAV 'block alignment' value is incorrect, ignoring.\n"
+ "The software that created this file is incorrect.\n"));
+ }
if ( format.align == format.channels * samplesize &&
format.samplesize == samplesize * 8 &&
(format.samplesize == 32 || format.samplesize == 24 ||
diff --git a/audio.h b/audio.h
index 4045c67..41d3207 100644
--- a/audio.h
+++ b/audio.h
@@ -32,6 +32,23 @@ extern "C" {
#include
#include "misc.h"
+/* In WIN32, the following bitmap can be found in sdk\inc\ksmedia.h and sdk\inc\mmreg.h: */
+
+#ifndef SPEAKER_FRONT_LEFT
+# define SPEAKER_FRONT_LEFT 0x1
+# define SPEAKER_FRONT_RIGHT 0x2
+#endif
+
+#ifndef WAVE_FORMAT_PCM
+# define WAVE_FORMAT_PCM 0x0001
+#endif
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+# define WAVE_FORMAT_IEEE_FLOAT 0x0003
+#endif
+#ifndef WAVE_FORMAT_EXTENSIBLE
+# define WAVE_FORMAT_EXTENSIBLE 0xfffe
+#endif
+
typedef long (*audio_read_func)(void *src,
double **buffer,
int samples,
@@ -77,12 +94,13 @@ typedef struct
typedef struct {
- short format;
- short channels;
- int samplerate;
- int bytespersec;
- short align;
- short samplesize;
+ unsigned short format;
+ unsigned short channels;
+ unsigned int channel_mask;
+ unsigned int samplerate;
+ unsigned int bytespersec;
+ unsigned short align;
+ unsigned short samplesize;
} wav_fmt;
typedef struct {
diff --git a/gain_analysis.c b/gain_analysis.c
index cbef014..e6e4e40 100644
--- a/gain_analysis.c
+++ b/gain_analysis.c
@@ -1,472 +1,481 @@
-/*
- * ReplayGainAnalysis - analyzes input samples and give the recommended dB change
- * Copyright (C) 2001 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.
- *
- * GetTitleGain()
- *
- * will return the recommended dB level change for all samples analyzed
- * SINCE THE LAST TIME you called GetTitleGain() OR InitGainAnalysis().
- *
- * GetAlbumGain()
- *
- * will return the recommended dB level change for all samples analyzed
- * since InitGainAnalysis() was called and finalized with GetTitleGain().
- *
- * Pseudo-code to process an album:
- *
- * Float_t l_samples [4096];
- * Float_t r_samples [4096];
- * size_t num_samples;
- * unsigned int num_songs;
- * unsigned int i;
- *
- * InitGainAnalysis ( 44100 );
- * for ( i = 1; i <= num_songs; i++ ) {
- * while ( ( num_samples = getSongSamples ( song[i], left_samples, right_samples ) ) > 0 )
- * AnalyzeSamples ( left_samples, right_samples, num_samples, 2 );
- * fprintf ("Recommended dB change for song %2d: %+6.2f dB\n", i, GetTitleGain() );
- * }
- * fprintf ("Recommended dB change for whole album: %+6.2f dB\n", GetAlbumGain() );
- */
-
-/*
- * 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 number of previous samples
- * AND up to 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
-#include
-#include
-#include
-
-#include "gain_analysis.h"
-
-typedef unsigned short Uint16_t;
-typedef signed short Int16_t;
-typedef unsigned int Uint32_t;
-typedef signed int Int32_t;
-
-#define YULE_ORDER 10
-#define BUTTER_ORDER 2
-#define YULE_FILTER filterYule
-#define BUTTER_FILTER filterButter
-#define RMS_PERCENTILE 0.95 // percentile which is louder than the proposed level
-#define MAX_SAMP_FREQ 48000. // maximum allowed sample frequency [Hz]
-#define RMS_WINDOW_TIME 0.050 // Time slice size [s]
-#define STEPS_per_dB 100. // Table entries per dB
-#define MAX_dB 120. // Table entries for 0...MAX_dB (normal max. values are 70...80 dB)
-
-#define MAX_ORDER (BUTTER_ORDER > YULE_ORDER ? BUTTER_ORDER : YULE_ORDER)
-#define MAX_SAMPLES_PER_WINDOW (size_t) (MAX_SAMP_FREQ * RMS_WINDOW_TIME) // max. Samples per Time slice
-#define PINK_REF 64.82 //298640883795 // calibration value
-
-Float_t linprebuf [MAX_ORDER * 2];
-Float_t* linpre; // left input samples, with pre-buffer
-Float_t lstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
-Float_t* lstep; // left "first step" (i.e. post first filter) samples
-Float_t loutbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
-Float_t* lout; // left "out" (i.e. post second filter) samples
-Float_t rinprebuf [MAX_ORDER * 2];
-Float_t* rinpre; // right input samples ...
-Float_t rstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
-Float_t* rstep;
-Float_t routbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
-Float_t* rout;
-long sampleWindow; // number of samples required to reach number of milliseconds required for RMS window
-long totsamp;
-double lsum;
-double rsum;
-int freqindex;
-int first;
-static Uint32_t A [(size_t)(STEPS_per_dB * MAX_dB)];
-static Uint32_t B [(size_t)(STEPS_per_dB * MAX_dB)];
-
-// 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
-
-#ifdef WIN32
-#ifndef __GNUC__
-#pragma warning ( disable : 4305 )
-#endif
-#endif
-
-static const Float_t ABYule[9][2*YULE_ORDER + 1] = {
- {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 }
-};
-
-static const Float_t ABButter[9][2*BUTTER_ORDER + 1] = {
- {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 }
-};
-
-
-#ifdef WIN32
-#ifndef __GNUC__
-#pragma warning ( default : 4305 )
-#endif
-#endif
-
-// When calling these filter procedures, make sure that ip[-order] and op[-order] point to real data!
-
-// If your compiler complains that "'operation on 'output' may be undefined", you can
-// either ignore the warnings or uncomment the three "y" lines (and comment out the indicated line)
-
-static void
-filterYule (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel)
-{
-
- 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;
- }
-}
-
-static void
-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;
- }
-}
-
-
-// returns a INIT_GAIN_ANALYSIS_OK if successful, INIT_GAIN_ANALYSIS_ERROR if not
-
-int
-ResetSampleFrequency ( long samplefreq ) {
- 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 48000: freqindex = 0; break;
- case 44100: freqindex = 1; break;
- case 32000: freqindex = 2; break;
- case 24000: freqindex = 3; break;
- case 22050: freqindex = 4; break;
- case 16000: freqindex = 5; break;
- case 12000: freqindex = 6; break;
- case 11025: freqindex = 7; break;
- case 8000: freqindex = 8; 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
-InitGainAnalysis ( 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) );
-
- return INIT_GAIN_ANALYSIS_OK;
-}
-
-// returns GAIN_ANALYSIS_OK if successful, GAIN_ANALYSIS_ERROR if not
-
-static __inline double fsqr(const double d)
-{ return d*d;
-}
-
-int
-AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels )
-{
- const Float_t* curleft;
- const Float_t* curright;
- long batchsamples;
- long cursamples;
- long cursamplepos;
- int i;
-
- if ( num_samples == 0 )
- return GAIN_ANALYSIS_OK;
-
- cursamplepos = 0;
- batchsamples = 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;
-}
-
-
-static Float_t
-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
-GetTitleGain ( void )
-{
- Float_t retval;
- int i;
-
- retval = analyzeResult ( A, sizeof(A)/sizeof(*A) );
-
- for ( i = 0; i < (int)(sizeof(A)/sizeof(*A)); i++ ) {
- B[i] += A[i];
- A[i] = 0;
- }
-
- for ( i = 0; i < MAX_ORDER; i++ )
- linprebuf[i] = lstepbuf[i] = loutbuf[i] = rinprebuf[i] = rstepbuf[i] = routbuf[i] = 0.f;
-
- totsamp = 0;
- lsum = rsum = 0.;
- return retval;
-}
-
-
-Float_t
-GetAlbumGain ( void )
-{
- return analyzeResult ( B, sizeof(B)/sizeof(*B) );
-}
-
-/* end of gain_analysis.c */
+/*
+ * 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.
+ *
+ * GetTitleGain()
+ *
+ * will return the recommended dB level change for all samples analyzed
+ * SINCE THE LAST TIME you called GetTitleGain() OR InitGainAnalysis().
+ *
+ * GetAlbumGain()
+ *
+ * will return the recommended dB level change for all samples analyzed
+ * since InitGainAnalysis() was called and finalized with GetTitleGain().
+ *
+ * Pseudo-code to process an album:
+ *
+ * Float_t l_samples [4096];
+ * Float_t r_samples [4096];
+ * size_t num_samples;
+ * unsigned int num_songs;
+ * unsigned int i;
+ *
+ * InitGainAnalysis ( 44100 );
+ * for ( i = 1; i <= num_songs; i++ ) {
+ * while ( ( num_samples = getSongSamples ( song[i], left_samples, right_samples ) ) > 0 )
+ * AnalyzeSamples ( left_samples, right_samples, num_samples, 2 );
+ * fprintf ("Recommended dB change for song %2d: %+6.2f dB\n", i, GetTitleGain() );
+ * }
+ * fprintf ("Recommended dB change for whole album: %+6.2f dB\n", GetAlbumGain() );
+ */
+
+/*
+ * 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 number of previous samples
+ * AND up to 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
+#include
+#include
+#include
+
+#include "gain_analysis.h"
+
+typedef unsigned short Uint16_t;
+typedef signed short Int16_t;
+typedef unsigned int Uint32_t;
+typedef signed int Int32_t;
+
+#define YULE_ORDER 10
+#define BUTTER_ORDER 2
+#define YULE_FILTER filterYule
+#define BUTTER_FILTER filterButter
+#define RMS_PERCENTILE 0.95 // percentile which is louder than the proposed level
+#define MAX_SAMP_FREQ 96000. // maximum allowed sample frequency [Hz]
+#define RMS_WINDOW_TIME 0.050 // Time slice size [s]
+#define STEPS_per_dB 100. // Table entries per dB
+#define MAX_dB 120. // Table entries for 0...MAX_dB (normal max. values are 70...80 dB)
+
+#define MAX_ORDER (BUTTER_ORDER > YULE_ORDER ? BUTTER_ORDER : YULE_ORDER)
+#define MAX_SAMPLES_PER_WINDOW (size_t) (MAX_SAMP_FREQ * RMS_WINDOW_TIME + 1) // max. Samples per Time slice
+#define PINK_REF 64.82 //298640883795 // calibration value
+
+Float_t linprebuf [MAX_ORDER * 2];
+Float_t* linpre; // left input samples, with pre-buffer
+Float_t lstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
+Float_t* lstep; // left "first step" (i.e. post first filter) samples
+Float_t loutbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
+Float_t* lout; // left "out" (i.e. post second filter) samples
+Float_t rinprebuf [MAX_ORDER * 2];
+Float_t* rinpre; // right input samples ...
+Float_t rstepbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
+Float_t* rstep;
+Float_t routbuf [MAX_SAMPLES_PER_WINDOW + MAX_ORDER];
+Float_t* rout;
+long sampleWindow; // number of samples required to reach number of milliseconds required for RMS window
+long totsamp;
+double lsum;
+double rsum;
+int freqindex;
+int first;
+static Uint32_t A [(size_t)(STEPS_per_dB * MAX_dB)];
+static Uint32_t B [(size_t)(STEPS_per_dB * MAX_dB)];
+
+// 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
+
+#ifdef WIN32
+#ifndef __GNUC__
+#pragma warning ( disable : 4305 )
+#endif
+#endif
+
+static const Float_t 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 }
+};
+
+static const Float_t 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 }
+};
+
+
+#ifdef WIN32
+#ifndef __GNUC__
+#pragma warning ( default : 4305 )
+#endif
+#endif
+
+// When calling these filter procedures, make sure that ip[-order] and op[-order] point to real data!
+
+// If your compiler complains that "'operation on 'output' may be undefined", you can
+// either ignore the warnings or uncomment the three "y" lines (and comment out the indicated line)
+
+static void
+filterYule (const Float_t* input, Float_t* output, size_t nSamples, const Float_t* kernel)
+{
+
+ 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;
+ }
+}
+
+static void
+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;
+ }
+}
+
+
+// returns a INIT_GAIN_ANALYSIS_OK if successful, INIT_GAIN_ANALYSIS_ERROR if not
+
+int
+ResetSampleFrequency ( long samplefreq ) {
+ 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
+InitGainAnalysis ( 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) );
+
+ return INIT_GAIN_ANALYSIS_OK;
+}
+
+// returns GAIN_ANALYSIS_OK if successful, GAIN_ANALYSIS_ERROR if not
+
+static __inline double fsqr(const double d)
+{ return d*d;
+}
+
+int
+AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels )
+{
+ const Float_t* curleft;
+ const Float_t* curright;
+ long batchsamples;
+ long cursamples;
+ long cursamplepos;
+ int i;
+
+ if ( num_samples == 0 )
+ return GAIN_ANALYSIS_OK;
+
+ 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;
+}
+
+
+static Float_t
+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
+GetTitleGain ( void )
+{
+ Float_t retval;
+ int i;
+
+ retval = analyzeResult ( A, sizeof(A)/sizeof(*A) );
+
+ for ( i = 0; i < (int)(sizeof(A)/sizeof(*A)); i++ ) {
+ B[i] += A[i];
+ A[i] = 0;
+ }
+
+ for ( i = 0; i < MAX_ORDER; i++ )
+ linprebuf[i] = lstepbuf[i] = loutbuf[i] = rinprebuf[i] = rstepbuf[i] = routbuf[i] = 0.f;
+
+ totsamp = 0;
+ lsum = rsum = 0.;
+ return retval;
+}
+
+
+Float_t
+GetAlbumGain ( void )
+{
+ return analyzeResult ( B, sizeof(B)/sizeof(*B) );
+}
+
+/* end of gain_analysis.c */
diff --git a/gain_analysis.h b/gain_analysis.h
index a66ab92..8e95568 100644
--- a/gain_analysis.h
+++ b/gain_analysis.h
@@ -1,56 +1,56 @@
-/*
- * ReplayGainAnalysis - analyzes input samples and give the recommended dB change
- * Copyright (C) 2001 David Robinson and Glen Sawyer
- *
- * 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
- * coding by Glen Sawyer (mp3gain@hotmail.com) 735 W 255 N, Orem, UT 84057-4505 USA
- * -- blame him if you think this runs too slowly, or the coding is otherwise flawed
- *
- * For an explanation of the concepts and the basic algorithms involved, go to:
- * http://www.replaygain.org/
- */
-
-#ifndef GAIN_ANALYSIS_H
-#define GAIN_ANALYSIS_H
-
-#include
-
-#define GAIN_NOT_ENOUGH_SAMPLES -24601
-#define GAIN_ANALYSIS_ERROR 0
-#define GAIN_ANALYSIS_OK 1
-
-#define INIT_GAIN_ANALYSIS_ERROR 0
-#define INIT_GAIN_ANALYSIS_OK 1
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-typedef double Float_t; // Type used for filtering
-
-int InitGainAnalysis ( long samplefreq );
-int AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels );
-int ResetSampleFrequency ( long samplefreq );
-Float_t GetTitleGain ( void );
-Float_t GetAlbumGain ( void );
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif /* GAIN_ANALYSIS_H */
+/*
+ * ReplayGainAnalysis - analyzes input samples and give the recommended dB change
+ * Copyright (C) 2001-2009 David Robinson and Glen Sawyer
+ *
+ * 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
+ * coding by Glen Sawyer (mp3gain@hotmail.com) 735 W 255 N, Orem, UT 84057-4505 USA
+ * -- blame him if you think this runs too slowly, or the coding is otherwise flawed
+ *
+ * For an explanation of the concepts and the basic algorithms involved, go to:
+ * http://www.replaygain.org/
+ */
+
+#ifndef GAIN_ANALYSIS_H
+#define GAIN_ANALYSIS_H
+
+#include
+
+#define GAIN_NOT_ENOUGH_SAMPLES -24601
+#define GAIN_ANALYSIS_ERROR 0
+#define GAIN_ANALYSIS_OK 1
+
+#define INIT_GAIN_ANALYSIS_ERROR 0
+#define INIT_GAIN_ANALYSIS_OK 1
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef double Float_t; // Type used for filtering
+
+int InitGainAnalysis ( long samplefreq );
+int AnalyzeSamples ( const Float_t* left_samples, const Float_t* right_samples, size_t num_samples, int num_channels );
+int ResetSampleFrequency ( long samplefreq );
+Float_t GetTitleGain ( void );
+Float_t GetAlbumGain ( void );
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* GAIN_ANALYSIS_H */
diff --git a/main.c b/main.c
index 0858a7b..9b77a98 100644
--- a/main.c
+++ b/main.c
@@ -162,7 +162,7 @@ int process_files(FILE_LIST* file_list, SETTINGS* settings, const char* dir)
{
FILE_LIST* file;
double factor_clip,
- audiophile_gain,
+ audiophile_gain = 0.,
Gain,
scale,
dB,
@@ -347,7 +347,7 @@ int process_files(FILE_LIST* file_list, SETTINGS* settings, const char* dir)
static void usage(void)
{
fprintf(stdout, _("WaveGain v" WAVEGAIN_VERSION " Compiled " __DATE__ ".\n"));
- fprintf(stdout, _("Copyright (c) 2002-2005 John Edwards \n"));
+ fprintf(stdout, _("Copyright (c) 2002-2010 John Edwards \n"));
fprintf(stdout, _("Additional code by Magnus Holmgren, Gian-Carlo Pascutto, and Tycho\n\n"));
#ifdef _WIN32
fprintf(stdout, " Usage: wavegain [options] input.wav [...] [-e cmd [args]]\n\n");
@@ -396,7 +396,7 @@ static void usage(void)
fprintf(stdout, " 4 for dither with Heavy Noise Shaping.\n");
fprintf(stdout, " -t, --limiter Apply 6dB Hard Limiter to output.\n");
fprintf(stdout, " -g, --gain X Apply additional Manual Gain adjustment in decibels, where\n");
- fprintf(stdout, " X = any floating point number between -12.0 and +12.0.\n");
+ fprintf(stdout, " X = any floating point number between -20.0 and +12.0.\n");
fprintf(stdout, " Clipping Prevention WILL be applied UNLESS '-n' is used.\n");
fprintf(stdout, " -s, --fast Calculates and prints gain settings - DOES NOT APPLY THEM.\n");
fprintf(stdout, " NOTE: This method does NOT process all samples, it only\n");
@@ -427,8 +427,9 @@ static void usage(void)
#endif
fprintf(stdout, " INPUT FILES\n");
fprintf(stdout, " WaveGain input files may be 8, 16, 24 or 32 bit integer, or floating point\n");
- fprintf(stdout, " wave files with 1 or 2 channels and a sample rate of 48000Hz, 44100Hz,\n");
- fprintf(stdout, " 32000Hz, 24000Hz, 22050Hz, 16000Hz, 12000Hz, 11025Hz or 8000Hz.\n");
+ fprintf(stdout, " wave files with 1 or 2 channels and a sample rate of 96000Hz, 88200Hz,\n");
+ fprintf(stdout, " 64000Hz, 48000Hz, 44100Hz, 32000Hz, 24000Hz, 22050Hz, 16000Hz, 12000Hz,\n");
+ fprintf(stdout, " 11025Hz or 8000Hz.\n");
fprintf(stdout, " 16 bit integer 'aiff' files are also supported.\n");
fprintf(stdout, " Wildcards (?, *) can be used in the filename, or '-' for stdin.\n");
@@ -599,15 +600,15 @@ int main(int argc, char** argv)
fprintf(stderr, "Warning: manual gain %s not recognised, ignoring\n", optarg);
break;
}
- if(settings.man_gain < -12.0) {
+ if(settings.man_gain < -20.0) {
fprintf(stderr, "Warning: manual gain %s is out of range, "
- "applying gain of -12.0dB\n", optarg);
- settings.man_gain = -12.0;
+ "applying gain of -20.0dB\n", optarg);
+ settings.man_gain = -20.0;
}
else if(settings.man_gain > 12.0) {
fprintf(stderr, "Warning: manual gain %s is out of range, "
"applying gain of +12.0dB\n", optarg);
- settings.man_gain = -12.0;
+ settings.man_gain = 12.0;
}
break;
case 'b':
diff --git a/main.h b/main.h
index f1e4491..f4cf090 100644
--- a/main.h
+++ b/main.h
@@ -2,7 +2,7 @@
#define MAIN_H
/* Some version numbers */
-#define WAVEGAIN_VERSION "1.2.8"
+#define WAVEGAIN_VERSION "1.3.0"
#define BUFFER_LEN 16384
#define TEMP_NAME "wavegain.tmp"
diff --git a/wavegain.c b/wavegain.c
index 66599c1..e7f1139 100644
--- a/wavegain.c
+++ b/wavegain.c
@@ -17,6 +17,7 @@
#include
#include
#include
+#include
#include
#ifdef _WIN32
@@ -183,7 +184,7 @@ int get_gain(const char *filename, double *track_peak, double *track_gain,
/* Only initialize gain analysis once in audiophile mode */
if (settings->first_file || !settings->audiophile) {
if (InitGainAnalysis(wg_opts->rate) != INIT_GAIN_ANALYSIS_OK) {
- fprintf(stderr, " Error Initializing Gain Analysis (nonstandard samplerate?)\n");
+ fprintf(stderr, " Error Initializing Gain Analysis (non-standard samplerate?)\n");
goto exit;
}
}
@@ -382,6 +383,9 @@ int write_gains(const char *filename, double radio_gain, double audiophile_gain,
double wrap_prev_neg;
void *sample_buffer;
input_format *format;
+ char tempName[24] = "";
+ unsigned int serial;
+ char tempSerial[7] = "";
memset(wg_opts, 0, sizeof(wavegain_opt));
@@ -512,7 +516,13 @@ int write_gains(const char *filename, double radio_gain, double audiophile_gain,
wg_opts->std_out = settings->std_out;
- aufile = open_output_audio_file(TEMP_NAME, wg_opts);
+ // Create temp file name
+ srand(time(NULL) ^ getpid());
+ serial = rand();
+ sprintf(tempSerial, "%d", serial);
+ strcpy(tempName, TEMP_NAME);
+ strcat(tempName, tempSerial);
+ aufile = open_output_audio_file(tempName, wg_opts);
if (aufile == NULL) {
fprintf (stderr, " Not able to open output file %s.\n", TEMP_NAME);
@@ -639,8 +649,8 @@ int write_gains(const char *filename, double radio_gain, double audiophile_gain,
goto exit;
}
- if (rename(TEMP_NAME, filename) != 0) {
- fprintf(stderr, " Error renaming '" TEMP_NAME "' to '%s' (uh-oh)\n", filename);
+ if (rename(tempName, filename) != 0) {
+ fprintf(stderr, " Error renaming '%s' to '%s' (uh-oh)\n", tempName, filename);
goto exit;
}
}