* for (i = 0; i < 16*16; i++) [
* largetbl[i] = ((ht[16].hlen[i]) << 16) + ht[24].hlen[i];
* ]
*
*
*/
Tables$1.largetbl = [
0x010004, 0x050005, 0x070007, 0x090008, 0x0a0009, 0x0a000a, 0x0b000a, 0x0b000b,
0x0c000b, 0x0c000c, 0x0c000c, 0x0d000c, 0x0d000c, 0x0d000c, 0x0e000d, 0x0a000a,
0x040005, 0x060006, 0x080007, 0x090008, 0x0a0009, 0x0b000a, 0x0b000a, 0x0b000b,
0x0c000b, 0x0c000b, 0x0c000c, 0x0d000c, 0x0e000c, 0x0d000c, 0x0e000c, 0x0a000a,
0x070007, 0x080007, 0x090008, 0x0a0009, 0x0b0009, 0x0b000a, 0x0c000a, 0x0c000b,
0x0d000b, 0x0c000b, 0x0d000b, 0x0d000c, 0x0d000c, 0x0e000c, 0x0e000d, 0x0b0009,
0x090008, 0x090008, 0x0a0009, 0x0b0009, 0x0b000a, 0x0c000a, 0x0c000a, 0x0c000b,
0x0d000b, 0x0d000b, 0x0e000b, 0x0e000c, 0x0e000c, 0x0f000c, 0x0f000c, 0x0c0009,
0x0a0009, 0x0a0009, 0x0b0009, 0x0b000a, 0x0c000a, 0x0c000a, 0x0d000a, 0x0d000b,
0x0d000b, 0x0e000b, 0x0e000c, 0x0e000c, 0x0f000c, 0x0f000c, 0x0f000d, 0x0b0009,
0x0a000a, 0x0a0009, 0x0b000a, 0x0b000a, 0x0c000a, 0x0d000a, 0x0d000b, 0x0e000b,
0x0d000b, 0x0e000b, 0x0e000c, 0x0f000c, 0x0f000c, 0x0f000c, 0x10000c, 0x0c0009,
0x0b000a, 0x0b000a, 0x0b000a, 0x0c000a, 0x0d000a, 0x0d000b, 0x0d000b, 0x0d000b,
0x0e000b, 0x0e000c, 0x0e000c, 0x0e000c, 0x0f000c, 0x0f000c, 0x10000d, 0x0c0009,
0x0b000b, 0x0b000a, 0x0c000a, 0x0c000a, 0x0d000b, 0x0d000b, 0x0d000b, 0x0e000b,
0x0e000c, 0x0f000c, 0x0f000c, 0x0f000c, 0x0f000c, 0x11000d, 0x11000d, 0x0c000a,
0x0b000b, 0x0c000b, 0x0c000b, 0x0d000b, 0x0d000b, 0x0d000b, 0x0e000b, 0x0e000b,
0x0f000b, 0x0f000c, 0x0f000c, 0x0f000c, 0x10000c, 0x10000d, 0x10000d, 0x0c000a,
0x0c000b, 0x0c000b, 0x0c000b, 0x0d000b, 0x0d000b, 0x0e000b, 0x0e000b, 0x0f000c,
0x0f000c, 0x0f000c, 0x0f000c, 0x10000c, 0x0f000d, 0x10000d, 0x0f000d, 0x0d000a,
0x0c000c, 0x0d000b, 0x0c000b, 0x0d000b, 0x0e000b, 0x0e000c, 0x0e000c, 0x0e000c,
0x0f000c, 0x10000c, 0x10000c, 0x10000d, 0x11000d, 0x11000d, 0x10000d, 0x0c000a,
0x0d000c, 0x0d000c, 0x0d000b, 0x0d000b, 0x0e000b, 0x0e000c, 0x0f000c, 0x10000c,
0x10000c, 0x10000c, 0x10000c, 0x10000d, 0x10000d, 0x0f000d, 0x10000d, 0x0d000a,
0x0d000c, 0x0e000c, 0x0e000c, 0x0e000c, 0x0e000c, 0x0f000c, 0x0f000c, 0x0f000c,
0x0f000c, 0x11000c, 0x10000d, 0x10000d, 0x10000d, 0x10000d, 0x12000d, 0x0d000a,
0x0f000c, 0x0e000c, 0x0e000c, 0x0e000c, 0x0f000c, 0x0f000c, 0x10000c, 0x10000c,
0x10000d, 0x12000d, 0x11000d, 0x11000d, 0x11000d, 0x13000d, 0x11000d, 0x0d000a,
0x0e000d, 0x0f000c, 0x0d000c, 0x0e000c, 0x10000c, 0x10000c, 0x0f000c, 0x10000d,
0x10000d, 0x11000d, 0x12000d, 0x11000d, 0x13000d, 0x11000d, 0x10000d, 0x0d000a,
0x0a0009, 0x0a0009, 0x0a0009, 0x0b0009, 0x0b0009, 0x0c0009, 0x0c0009, 0x0c0009,
0x0d0009, 0x0d0009, 0x0d0009, 0x0d000a, 0x0d000a, 0x0d000a, 0x0d000a, 0x0a0006
];
/**
*
* for (i = 0; i < 3*3; i++) [
* table23[i] = ((ht[2].hlen[i]) << 16) + ht[3].hlen[i];
* ]
*
*
*/
Tables$1.table23 = [
0x010002, 0x040003, 0x070007,
0x040004, 0x050004, 0x070007,
0x060006, 0x070007, 0x080008
];
/**
*
* for (i = 0; i < 4*4; i++) [
* table56[i] = ((ht[5].hlen[i]) << 16) + ht[6].hlen[i];
* ]
*
*
*/
Tables$1.table56 = [
0x010003, 0x040004, 0x070006, 0x080008, 0x040004, 0x050004, 0x080006, 0x090007,
0x070005, 0x080006, 0x090007, 0x0a0008, 0x080007, 0x080007, 0x090008, 0x0a0009
];
Tables$1.bitrate_table = [
[0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, -1], /* MPEG 2 */
[0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, -1], /* MPEG 1 */
[0, 8, 16, 24, 32, 40, 48, 56, 64, -1, -1, -1, -1, -1, -1, -1], /* MPEG 2.5 */
];
/**
* MPEG 2, MPEG 1, MPEG 2.5.
*/
Tables$1.samplerate_table = [
[22050, 24000, 16000, -1],
[44100, 48000, 32000, -1],
[11025, 12000, 8000, -1],
];
/**
* This is the scfsi_band table from 2.4.2.7 of the IS.
*/
Tables$1.scfsi_band = [0, 6, 11, 16, 21];
var Tables_1 = Tables$1;
var VbrMode$19 = common.VbrMode;
var Float$19 = common.Float;
var Util$19 = common.Util;
var new_float$19 = common.new_float;
var new_int$19 = common.new_int;
var assert$19 = common.assert;
QuantizePVT$1.Q_MAX = (256 + 1);
QuantizePVT$1.Q_MAX2 = 116;
QuantizePVT$1.LARGE_BITS = 100000;
QuantizePVT$1.IXMAX_VAL = 8206;
function QuantizePVT$1() {
var tak = null;
var rv = null;
var psy = null;
this.setModules = function (_tk, _rv, _psy) {
tak = _tk;
rv = _rv;
psy = _psy;
};
function POW20(x) {
assert$19(0 <= (x + QuantizePVT$1.Q_MAX2) && x < QuantizePVT$1.Q_MAX);
return pow20[x + QuantizePVT$1.Q_MAX2];
}
this.IPOW20 = function (x) {
assert$19(0 <= x && x < QuantizePVT$1.Q_MAX);
return ipow20[x];
};
/**
* smallest such that 1.0+DBL_EPSILON != 1.0
*/
var DBL_EPSILON = 2.2204460492503131e-016;
/**
* ix always <= 8191+15. see count_bits()
*/
var IXMAX_VAL = QuantizePVT$1.IXMAX_VAL;
var PRECALC_SIZE = (IXMAX_VAL + 2);
var Q_MAX = QuantizePVT$1.Q_MAX;
/**
*
* minimum possible number of
* -cod_info.global_gain + ((scalefac[] + (cod_info.preflag ? pretab[sfb] : 0))
* << (cod_info.scalefac_scale + 1)) + cod_info.subblock_gain[cod_info.window[sfb]] * 8;
*
* for long block, 0+((15+3)<<2) = 18*4 = 72
* for short block, 0+(15<<2)+7*8 = 15*4+56 = 116
*
*/
var Q_MAX2 = QuantizePVT$1.Q_MAX2;
var LARGE_BITS = QuantizePVT$1.LARGE_BITS;
/**
* Assuming dynamic range=96dB, this value should be 92
*/
var NSATHSCALE = 100;
/**
* The following table is used to implement the scalefactor partitioning for
* MPEG2 as described in section 2.4.3.2 of the IS. The indexing corresponds
* to the way the tables are presented in the IS:
*
* [table_number][row_in_table][column of nr_of_sfb]
*/
this.nr_of_sfb_block = [
[[6, 5, 5, 5], [9, 9, 9, 9], [6, 9, 9, 9]],
[[6, 5, 7, 3], [9, 9, 12, 6], [6, 9, 12, 6]],
[[11, 10, 0, 0], [18, 18, 0, 0], [15, 18, 0, 0]],
[[7, 7, 7, 0], [12, 12, 12, 0], [6, 15, 12, 0]],
[[6, 6, 6, 3], [12, 9, 9, 6], [6, 12, 9, 6]],
[[8, 8, 5, 0], [15, 12, 9, 0], [6, 18, 9, 0]]];
/**
* Table B.6: layer3 preemphasis
*/
var pretab = [0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
2, 2, 3, 3, 3, 2, 0];
this.pretab = pretab;
/**
* Here are MPEG1 Table B.8 and MPEG2 Table B.1 -- Layer III scalefactor
* bands. * Index into this using a method such as:
* idx = fr_ps.header.sampling_frequency + (fr_ps.header.version * 3) */ this.sfBandIndex = [ // Table B.2.b: 22.05 kHz new ScaleFac_1([0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576], [0, 4, 8, 12, 18, 24, 32, 42, 56, 74, 100, 132, 174, 192] , [0, 0, 0, 0, 0, 0, 0] // sfb21 pseudo sub bands , [0, 0, 0, 0, 0, 0, 0] // sfb12 pseudo sub bands ), /* Table B.2.c: 24 kHz */ /* docs: 332. mpg123(broken): 330 */ new ScaleFac_1([0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 114, 136, 162, 194, 232, 278, 332, 394, 464, 540, 576], [0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 136, 180, 192] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* Table B.2.a: 16 kHz */ new ScaleFac_1([0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576], [0, 4, 8, 12, 18, 26, 36, 48, 62, 80, 104, 134, 174, 192] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* Table B.8.b: 44.1 kHz */ new ScaleFac_1([0, 4, 8, 12, 16, 20, 24, 30, 36, 44, 52, 62, 74, 90, 110, 134, 162, 196, 238, 288, 342, 418, 576], [0, 4, 8, 12, 16, 22, 30, 40, 52, 66, 84, 106, 136, 192] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* Table B.8.c: 48 kHz */ new ScaleFac_1([0, 4, 8, 12, 16, 20, 24, 30, 36, 42, 50, 60, 72, 88, 106, 128, 156, 190, 230, 276, 330, 384, 576], [0, 4, 8, 12, 16, 22, 28, 38, 50, 64, 80, 100, 126, 192] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* Table B.8.a: 32 kHz */ new ScaleFac_1([0, 4, 8, 12, 16, 20, 24, 30, 36, 44, 54, 66, 82, 102, 126, 156, 194, 240, 296, 364, 448, 550, 576], [0, 4, 8, 12, 16, 22, 30, 42, 58, 78, 104, 138, 180, 192] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* MPEG-2.5 11.025 kHz */ new ScaleFac_1([0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576], [0 / 3, 12 / 3, 24 / 3, 36 / 3, 54 / 3, 78 / 3, 108 / 3, 144 / 3, 186 / 3, 240 / 3, 312 / 3, 402 / 3, 522 / 3, 576 / 3] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* MPEG-2.5 12 kHz */ new ScaleFac_1([0, 6, 12, 18, 24, 30, 36, 44, 54, 66, 80, 96, 116, 140, 168, 200, 238, 284, 336, 396, 464, 522, 576], [0 / 3, 12 / 3, 24 / 3, 36 / 3, 54 / 3, 78 / 3, 108 / 3, 144 / 3, 186 / 3, 240 / 3, 312 / 3, 402 / 3, 522 / 3, 576 / 3] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ), /* MPEG-2.5 8 kHz */ new ScaleFac_1([0, 12, 24, 36, 48, 60, 72, 88, 108, 132, 160, 192, 232, 280, 336, 400, 476, 566, 568, 570, 572, 574, 576], [0 / 3, 24 / 3, 48 / 3, 72 / 3, 108 / 3, 156 / 3, 216 / 3, 288 / 3, 372 / 3, 480 / 3, 486 / 3, 492 / 3, 498 / 3, 576 / 3] , [0, 0, 0, 0, 0, 0, 0] /* sfb21 pseudo sub bands */ , [0, 0, 0, 0, 0, 0, 0] /* sfb12 pseudo sub bands */ ) ]; var pow20 = new_float$19(Q_MAX + Q_MAX2 + 1); var ipow20 = new_float$19(Q_MAX); var pow43 = new_float$19(PRECALC_SIZE); var adj43 = new_float$19(PRECALC_SIZE); this.adj43 = adj43; /** *
* compute the ATH for each scalefactor band cd range: 0..96db
*
* Input: 3.3kHz signal 32767 amplitude (3.3kHz is where ATH is smallest =
* -5db) longblocks: sfb=12 en0/bw=-11db max_en0 = 1.3db shortblocks: sfb=5
* -9db 0db
*
* Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) longblocks: amp=1
* sfb=12 en0/bw=-103 db max_en0 = -92db amp=32767 sfb=12 -12 db -1.4db
*
* Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) shortblocks: amp=1
* sfb=5 en0/bw= -99 -86 amp=32767 sfb=5 -9 db 4db
*
*
* MAX energy of largest wave at 3.3kHz = 1db AVE energy of largest wave at
* 3.3kHz = -11db Let's take AVE: -11db = maximum signal in sfb=12. Dynamic
* range of CD: 96db. Therefor energy of smallest audible wave in sfb=12 =
* -11 - 96 = -107db = ATH at 3.3kHz.
*
* ATH formula for this wave: -5db. To adjust to LAME scaling, we need ATH =
* ATH_formula - 103 (db) ATH = ATH * 2.5e-10 (ener)
*
*/
function ATHmdct(gfp, f) {
var ath = psy.ATHformula(f, gfp);
ath -= NSATHSCALE;
/* modify the MDCT scaling for the ATH and convert to energy */
ath = Math.pow(10.0, ath / 10.0 + gfp.ATHlower);
return ath;
}
function compute_ath(gfp) {
var ATH_l = gfp.internal_flags.ATH.l;
var ATH_psfb21 = gfp.internal_flags.ATH.psfb21;
var ATH_s = gfp.internal_flags.ATH.s;
var ATH_psfb12 = gfp.internal_flags.ATH.psfb12;
var gfc = gfp.internal_flags;
var samp_freq = gfp.out_samplerate;
for (var sfb = 0; sfb < Encoder_1.SBMAX_l; sfb++) {
var start = gfc.scalefac_band.l[sfb];
var end = gfc.scalefac_band.l[sfb + 1];
ATH_l[sfb] = Float$19.MAX_VALUE;
for (var i = start; i < end; i++) {
var freq = i * samp_freq / (2 * 576);
var ATH_f = ATHmdct(gfp, freq);
/* freq in kHz */
ATH_l[sfb] = Math.min(ATH_l[sfb], ATH_f);
}
}
for (var sfb = 0; sfb < Encoder_1.PSFB21; sfb++) {
var start = gfc.scalefac_band.psfb21[sfb];
var end = gfc.scalefac_band.psfb21[sfb + 1];
ATH_psfb21[sfb] = Float$19.MAX_VALUE;
for (var i = start; i < end; i++) {
var freq = i * samp_freq / (2 * 576);
var ATH_f = ATHmdct(gfp, freq);
/* freq in kHz */
ATH_psfb21[sfb] = Math.min(ATH_psfb21[sfb], ATH_f);
}
}
for (var sfb = 0; sfb < Encoder_1.SBMAX_s; sfb++) {
var start = gfc.scalefac_band.s[sfb];
var end = gfc.scalefac_band.s[sfb + 1];
ATH_s[sfb] = Float$19.MAX_VALUE;
for (var i = start; i < end; i++) {
var freq = i * samp_freq / (2 * 192);
var ATH_f = ATHmdct(gfp, freq);
/* freq in kHz */
ATH_s[sfb] = Math.min(ATH_s[sfb], ATH_f);
}
ATH_s[sfb] *= (gfc.scalefac_band.s[sfb + 1] - gfc.scalefac_band.s[sfb]);
}
for (var sfb = 0; sfb < Encoder_1.PSFB12; sfb++) {
var start = gfc.scalefac_band.psfb12[sfb];
var end = gfc.scalefac_band.psfb12[sfb + 1];
ATH_psfb12[sfb] = Float$19.MAX_VALUE;
for (var i = start; i < end; i++) {
var freq = i * samp_freq / (2 * 192);
var ATH_f = ATHmdct(gfp, freq);
/* freq in kHz */
ATH_psfb12[sfb] = Math.min(ATH_psfb12[sfb], ATH_f);
}
/* not sure about the following */
ATH_psfb12[sfb] *= (gfc.scalefac_band.s[13] - gfc.scalefac_band.s[12]);
}
/*
* no-ATH mode: reduce ATH to -200 dB
*/
if (gfp.noATH) {
for (var sfb = 0; sfb < Encoder_1.SBMAX_l; sfb++) {
ATH_l[sfb] = 1E-20;
}
for (var sfb = 0; sfb < Encoder_1.PSFB21; sfb++) {
ATH_psfb21[sfb] = 1E-20;
}
for (var sfb = 0; sfb < Encoder_1.SBMAX_s; sfb++) {
ATH_s[sfb] = 1E-20;
}
for (var sfb = 0; sfb < Encoder_1.PSFB12; sfb++) {
ATH_psfb12[sfb] = 1E-20;
}
}
/*
* work in progress, don't rely on it too much
*/
gfc.ATH.floor = 10. * Math.log10(ATHmdct(gfp, -1.));
}
/**
* initialization for iteration_loop
*/
this.iteration_init = function (gfp) {
var gfc = gfp.internal_flags;
var l3_side = gfc.l3_side;
var i;
if (gfc.iteration_init_init == 0) {
gfc.iteration_init_init = 1;
l3_side.main_data_begin = 0;
compute_ath(gfp);
pow43[0] = 0.0;
for (i = 1; i < PRECALC_SIZE; i++)
pow43[i] = Math.pow(i, 4.0 / 3.0);
for (i = 0; i < PRECALC_SIZE - 1; i++)
adj43[i] = ((i + 1) - Math.pow(
0.5 * (pow43[i] + pow43[i + 1]), 0.75));
adj43[i] = 0.5;
for (i = 0; i < Q_MAX; i++)
ipow20[i] = Math.pow(2.0, (i - 210) * -0.1875);
for (i = 0; i <= Q_MAX + Q_MAX2; i++)
pow20[i] = Math.pow(2.0, (i - 210 - Q_MAX2) * 0.25);
tak.huffman_init(gfc);
{
var bass, alto, treble, sfb21;
i = (gfp.exp_nspsytune >> 2) & 63;
if (i >= 32)
i -= 64;
bass = Math.pow(10, i / 4.0 / 10.0);
i = (gfp.exp_nspsytune >> 8) & 63;
if (i >= 32)
i -= 64;
alto = Math.pow(10, i / 4.0 / 10.0);
i = (gfp.exp_nspsytune >> 14) & 63;
if (i >= 32)
i -= 64;
treble = Math.pow(10, i / 4.0 / 10.0);
/*
* to be compatible with Naoki's original code, the next 6 bits
* define only the amount of changing treble for sfb21
*/
i = (gfp.exp_nspsytune >> 20) & 63;
if (i >= 32)
i -= 64;
sfb21 = treble * Math.pow(10, i / 4.0 / 10.0);
for (i = 0; i < Encoder_1.SBMAX_l; i++) {
var f;
if (i <= 6)
f = bass;
else if (i <= 13)
f = alto;
else if (i <= 20)
f = treble;
else
f = sfb21;
gfc.nsPsy.longfact[i] = f;
}
for (i = 0; i < Encoder_1.SBMAX_s; i++) {
var f;
if (i <= 5)
f = bass;
else if (i <= 10)
f = alto;
else if (i <= 11)
f = treble;
else
f = sfb21;
gfc.nsPsy.shortfact[i] = f;
}
}
}
};
/**
* allocate bits among 2 channels based on PE* mt 6/99
* bugfixes rh 8/01: often allocated more than the allowed 4095 bits */ this.on_pe = function (gfp, pe, targ_bits, mean_bits, gr, cbr) { var gfc = gfp.internal_flags; var tbits = 0, bits; var add_bits = new_int$19(2); var ch; /* allocate targ_bits for granule */ var mb = new MeanBits_1(tbits); var extra_bits = rv.ResvMaxBits(gfp, mean_bits, mb, cbr); tbits = mb.bits; /* maximum allowed bits for this granule */ var max_bits = tbits + extra_bits; if (max_bits > LameInternalFlags_1.MAX_BITS_PER_GRANULE) { // hard limit per granule max_bits = LameInternalFlags_1.MAX_BITS_PER_GRANULE; } for (bits = 0, ch = 0; ch < gfc.channels_out; ++ch) { /****************************************************************** * allocate bits for each channel ******************************************************************/ targ_bits[ch] = Math.min(LameInternalFlags_1.MAX_BITS_PER_CHANNEL, tbits / gfc.channels_out); add_bits[ch] = 0 | (targ_bits[ch] * pe[gr][ch] / 700.0 - targ_bits[ch]); /* at most increase bits by 1.5*average */ if (add_bits[ch] > mean_bits * 3 / 4) add_bits[ch] = mean_bits * 3 / 4; if (add_bits[ch] < 0) add_bits[ch] = 0; if (add_bits[ch] + targ_bits[ch] > LameInternalFlags_1.MAX_BITS_PER_CHANNEL) add_bits[ch] = Math.max(0, LameInternalFlags_1.MAX_BITS_PER_CHANNEL - targ_bits[ch]); bits += add_bits[ch]; } if (bits > extra_bits) { for (ch = 0; ch < gfc.channels_out; ++ch) { add_bits[ch] = extra_bits * add_bits[ch] / bits; } } for (ch = 0; ch < gfc.channels_out; ++ch) { targ_bits[ch] += add_bits[ch]; extra_bits -= add_bits[ch]; } for (bits = 0, ch = 0; ch < gfc.channels_out; ++ch) { bits += targ_bits[ch]; } if (bits > LameInternalFlags_1.MAX_BITS_PER_GRANULE) { var sum = 0; for (ch = 0; ch < gfc.channels_out; ++ch) { targ_bits[ch] *= LameInternalFlags_1.MAX_BITS_PER_GRANULE; targ_bits[ch] /= bits; sum += targ_bits[ch]; } assert$19(sum <= LameInternalFlags_1.MAX_BITS_PER_GRANULE); } return max_bits; }; this.reduce_side = function (targ_bits, ms_ener_ratio, mean_bits, max_bits) { assert$19(max_bits <= LameInternalFlags_1.MAX_BITS_PER_GRANULE); assert$19(targ_bits[0] + targ_bits[1] <= LameInternalFlags_1.MAX_BITS_PER_GRANULE); /* * ms_ener_ratio = 0: allocate 66/33 mid/side fac=.33 ms_ener_ratio =.5: * allocate 50/50 mid/side fac= 0 */ /* 75/25 split is fac=.5 */ var fac = .33 * (.5 - ms_ener_ratio) / .5; if (fac < 0) fac = 0; if (fac > .5) fac = .5; /* number of bits to move from side channel to mid channel */ /* move_bits = fac*targ_bits[1]; */ var move_bits = 0 | (fac * .5 * (targ_bits[0] + targ_bits[1])); if (move_bits > LameInternalFlags_1.MAX_BITS_PER_CHANNEL - targ_bits[0]) { move_bits = LameInternalFlags_1.MAX_BITS_PER_CHANNEL - targ_bits[0]; } if (move_bits < 0) move_bits = 0; if (targ_bits[1] >= 125) { /* dont reduce side channel below 125 bits */ if (targ_bits[1] - move_bits > 125) { /* if mid channel already has 2x more than average, dont bother */ /* mean_bits = bits per granule (for both channels) */ if (targ_bits[0] < mean_bits) targ_bits[0] += move_bits; targ_bits[1] -= move_bits; } else { targ_bits[0] += targ_bits[1] - 125; targ_bits[1] = 125; } } move_bits = targ_bits[0] + targ_bits[1]; if (move_bits > max_bits) { targ_bits[0] = (max_bits * targ_bits[0]) / move_bits; targ_bits[1] = (max_bits * targ_bits[1]) / move_bits; } assert$19(targ_bits[0] <= LameInternalFlags_1.MAX_BITS_PER_CHANNEL); assert$19(targ_bits[1] <= LameInternalFlags_1.MAX_BITS_PER_CHANNEL); assert$19(targ_bits[0] + targ_bits[1] <= LameInternalFlags_1.MAX_BITS_PER_GRANULE); }; /** * Robert Hegemann 2001-04-27: * this adjusts the ATH, keeping the original noise floor * affects the higher frequencies more than the lower ones */ this.athAdjust = function (a, x, athFloor) { /* * work in progress */ var o = 90.30873362; var p = 94.82444863; var u = Util$19.FAST_LOG10_X(x, 10.0); var v = a * a; var w = 0.0; u -= athFloor; /* undo scaling */ if (v > 1E-20) w = 1. + Util$19.FAST_LOG10_X(v, 10.0 / o); if (w < 0) w = 0.; u *= w; u += athFloor + o - p; /* redo scaling */ return Math.pow(10., 0.1 * u); }; /** * Calculate the allowed distortion for each scalefactor band, as determined * by the psychoacoustic model. xmin(sb) = ratio(sb) * en(sb) / bw(sb) * * returns number of sfb's with energy > ATH */ this.calc_xmin = function (gfp, ratio, cod_info, pxmin) { var pxminPos = 0; var gfc = gfp.internal_flags; var gsfb, j = 0, ath_over = 0; var ATH = gfc.ATH; var xr = cod_info.xr; var enable_athaa_fix = (gfp.VBR == VbrMode$19.vbr_mtrh) ? 1 : 0; var masking_lower = gfc.masking_lower; if (gfp.VBR == VbrMode$19.vbr_mtrh || gfp.VBR == VbrMode$19.vbr_mt) { /* was already done in PSY-Model */ masking_lower = 1.0; } for (gsfb = 0; gsfb < cod_info.psy_lmax; gsfb++) { var en0, xmin; var rh1, rh2; var width, l; if (gfp.VBR == VbrMode$19.vbr_rh || gfp.VBR == VbrMode$19.vbr_mtrh) xmin = athAdjust(ATH.adjust, ATH.l[gsfb], ATH.floor); else xmin = ATH.adjust * ATH.l[gsfb]; width = cod_info.width[gsfb]; rh1 = xmin / width; rh2 = DBL_EPSILON; l = width >> 1; en0 = 0.0; do { var xa, xb; xa = xr[j] * xr[j]; en0 += xa; rh2 += (xa < rh1) ? xa : rh1; j++; xb = xr[j] * xr[j]; en0 += xb; rh2 += (xb < rh1) ? xb : rh1; j++; } while (--l > 0); if (en0 > xmin) ath_over++; if (gsfb == Encoder_1.SBPSY_l) { var x = xmin * gfc.nsPsy.longfact[gsfb]; if (rh2 < x) { rh2 = x; } } if (enable_athaa_fix != 0) { xmin = rh2; } if (!gfp.ATHonly) { var e = ratio.en.l[gsfb]; if (e > 0.0) { var x; x = en0 * ratio.thm.l[gsfb] * masking_lower / e; if (enable_athaa_fix != 0) x *= gfc.nsPsy.longfact[gsfb]; if (xmin < x) xmin = x; } } if (enable_athaa_fix != 0) pxmin[pxminPos++] = xmin; else pxmin[pxminPos++] = xmin * gfc.nsPsy.longfact[gsfb]; } /* end of long block loop */ /* use this function to determine the highest non-zero coeff */ var max_nonzero = 575; if (cod_info.block_type != Encoder_1.SHORT_TYPE) { // NORM, START or STOP type, but not SHORT var k = 576; while (k-- != 0 && BitStream.EQ(xr[k], 0)) { max_nonzero = k; } } cod_info.max_nonzero_coeff = max_nonzero; for (var sfb = cod_info.sfb_smin; gsfb < cod_info.psymax; sfb++, gsfb += 3) { var width, b; var tmpATH; if (gfp.VBR == VbrMode$19.vbr_rh || gfp.VBR == VbrMode$19.vbr_mtrh) tmpATH = athAdjust(ATH.adjust, ATH.s[sfb], ATH.floor); else tmpATH = ATH.adjust * ATH.s[sfb]; width = cod_info.width[gsfb]; for (b = 0; b < 3; b++) { var en0 = 0.0, xmin; var rh1, rh2; var l = width >> 1; rh1 = tmpATH / width; rh2 = DBL_EPSILON; do { var xa, xb; xa = xr[j] * xr[j]; en0 += xa; rh2 += (xa < rh1) ? xa : rh1; j++; xb = xr[j] * xr[j]; en0 += xb; rh2 += (xb < rh1) ? xb : rh1; j++; } while (--l > 0); if (en0 > tmpATH) ath_over++; if (sfb == Encoder_1.SBPSY_s) { var x = tmpATH * gfc.nsPsy.shortfact[sfb]; if (rh2 < x) { rh2 = x; } } if (enable_athaa_fix != 0) xmin = rh2; else xmin = tmpATH; if (!gfp.ATHonly && !gfp.ATHshort) { var e = ratio.en.s[sfb][b]; if (e > 0.0) { var x; x = en0 * ratio.thm.s[sfb][b] * masking_lower / e; if (enable_athaa_fix != 0) x *= gfc.nsPsy.shortfact[sfb]; if (xmin < x) xmin = x; } } if (enable_athaa_fix != 0) pxmin[pxminPos++] = xmin; else pxmin[pxminPos++] = xmin * gfc.nsPsy.shortfact[sfb]; } /* b */ if (gfp.useTemporal) { if (pxmin[pxminPos - 3] > pxmin[pxminPos - 3 + 1]) pxmin[pxminPos - 3 + 1] += (pxmin[pxminPos - 3] - pxmin[pxminPos - 3 + 1]) * gfc.decay; if (pxmin[pxminPos - 3 + 1] > pxmin[pxminPos - 3 + 2]) pxmin[pxminPos - 3 + 2] += (pxmin[pxminPos - 3 + 1] - pxmin[pxminPos - 3 + 2]) * gfc.decay; } } /* end of short block sfb loop */ return ath_over; }; function StartLine(j) { this.s = j; } this.calc_noise_core = function (cod_info, startline, l, step) { var noise = 0; var j = startline.s; var ix = cod_info.l3_enc; if (j > cod_info.count1) { while ((l--) != 0) { var temp; temp = cod_info.xr[j]; j++; noise += temp * temp; temp = cod_info.xr[j]; j++; noise += temp * temp; } } else if (j > cod_info.big_values) { var ix01 = new_float$19(2); ix01[0] = 0; ix01[1] = step; while ((l--) != 0) { var temp; temp = Math.abs(cod_info.xr[j]) - ix01[ix[j]]; j++; noise += temp * temp; temp = Math.abs(cod_info.xr[j]) - ix01[ix[j]]; j++; noise += temp * temp; } } else { while ((l--) != 0) { var temp; temp = Math.abs(cod_info.xr[j]) - pow43[ix[j]] * step; j++; noise += temp * temp; temp = Math.abs(cod_info.xr[j]) - pow43[ix[j]] * step; j++; noise += temp * temp; } } startline.s = j; return noise; }; /** *
* -oo dB => -1.00
* - 6 dB => -0.97
* - 3 dB => -0.80
* - 2 dB => -0.64
* - 1 dB => -0.38
* 0 dB => 0.00
* + 1 dB => +0.49
* + 2 dB => +1.06
* + 3 dB => +1.68
* + 6 dB => +3.69
* +10 dB => +6.45
*
*/
this.calc_noise = function (cod_info, l3_xmin, distort, res, prev_noise) {
var distortPos = 0;
var l3_xminPos = 0;
var sfb, l, over = 0;
var over_noise_db = 0;
/* 0 dB relative to masking */
var tot_noise_db = 0;
/* -200 dB relative to masking */
var max_noise = -20.0;
var j = 0;
var scalefac = cod_info.scalefac;
var scalefacPos = 0;
res.over_SSD = 0;
for (sfb = 0; sfb < cod_info.psymax; sfb++) {
var s = cod_info.global_gain
- (((scalefac[scalefacPos++]) + (cod_info.preflag != 0 ? pretab[sfb]
: 0)) << (cod_info.scalefac_scale + 1))
- cod_info.subblock_gain[cod_info.window[sfb]] * 8;
var noise = 0.0;
if (prev_noise != null && (prev_noise.step[sfb] == s)) {
/* use previously computed values */
noise = prev_noise.noise[sfb];
j += cod_info.width[sfb];
distort[distortPos++] = noise / l3_xmin[l3_xminPos++];
noise = prev_noise.noise_log[sfb];
} else {
var step = POW20(s);
l = cod_info.width[sfb] >> 1;
if ((j + cod_info.width[sfb]) > cod_info.max_nonzero_coeff) {
var usefullsize;
usefullsize = cod_info.max_nonzero_coeff - j + 1;
if (usefullsize > 0)
l = usefullsize >> 1;
else
l = 0;
}
var sl = new StartLine(j);
noise = this.calc_noise_core(cod_info, sl, l, step);
j = sl.s;
if (prev_noise != null) {
/* save noise values */
prev_noise.step[sfb] = s;
prev_noise.noise[sfb] = noise;
}
noise = distort[distortPos++] = noise / l3_xmin[l3_xminPos++];
/* multiplying here is adding in dB, but can overflow */
noise = Util$19.FAST_LOG10(Math.max(noise, 1E-20));
if (prev_noise != null) {
/* save noise values */
prev_noise.noise_log[sfb] = noise;
}
}
if (prev_noise != null) {
/* save noise values */
prev_noise.global_gain = cod_info.global_gain;
}
tot_noise_db += noise;
if (noise > 0.0) {
var tmp;
tmp = Math.max(0 | (noise * 10 + .5), 1);
res.over_SSD += tmp * tmp;
over++;
/* multiplying here is adding in dB -but can overflow */
/* over_noise *= noise; */
over_noise_db += noise;
}
max_noise = Math.max(max_noise, noise);
}
res.over_count = over;
res.tot_noise = tot_noise_db;
res.over_noise = over_noise_db;
res.max_noise = max_noise;
return over;
};
/**
* updates plotting data
*
* Mark Taylor 2000-??-??
*
* Robert Hegemann: moved noise/distortion calc into it
*/
this.set_pinfo = function (gfp, cod_info, ratio, gr, ch) {
var gfc = gfp.internal_flags;
var sfb, sfb2;
var l;
var en0, en1;
var ifqstep = (cod_info.scalefac_scale == 0) ? .5 : 1.0;
var scalefac = cod_info.scalefac;
var l3_xmin = new_float$19(L3Side.SFBMAX);
var xfsf = new_float$19(L3Side.SFBMAX);
var noise = new CalcNoiseResult();
calc_xmin(gfp, ratio, cod_info, l3_xmin);
calc_noise(cod_info, l3_xmin, xfsf, noise, null);
var j = 0;
sfb2 = cod_info.sfb_lmax;
if (cod_info.block_type != Encoder_1.SHORT_TYPE
&& 0 == cod_info.mixed_block_flag)
sfb2 = 22;
for (sfb = 0; sfb < sfb2; sfb++) {
var start = gfc.scalefac_band.l[sfb];
var end = gfc.scalefac_band.l[sfb + 1];
var bw = end - start;
for (en0 = 0.0; j < end; j++)
en0 += cod_info.xr[j] * cod_info.xr[j];
en0 /= bw;
/* convert to MDCT units */
/* scaling so it shows up on FFT plot */
en1 = 1e15;
gfc.pinfo.en[gr][ch][sfb] = en1 * en0;
gfc.pinfo.xfsf[gr][ch][sfb] = en1 * l3_xmin[sfb] * xfsf[sfb] / bw;
if (ratio.en.l[sfb] > 0 && !gfp.ATHonly)
en0 = en0 / ratio.en.l[sfb];
else
en0 = 0.0;
gfc.pinfo.thr[gr][ch][sfb] = en1
* Math.max(en0 * ratio.thm.l[sfb], gfc.ATH.l[sfb]);
/* there is no scalefactor bands >= SBPSY_l */
gfc.pinfo.LAMEsfb[gr][ch][sfb] = 0;
if (cod_info.preflag != 0 && sfb >= 11)
gfc.pinfo.LAMEsfb[gr][ch][sfb] = -ifqstep * pretab[sfb];
if (sfb < Encoder_1.SBPSY_l) {
/* scfsi should be decoded by caller side */
assert$19(scalefac[sfb] >= 0);
gfc.pinfo.LAMEsfb[gr][ch][sfb] -= ifqstep * scalefac[sfb];
}
}
/* for sfb */
if (cod_info.block_type == Encoder_1.SHORT_TYPE) {
sfb2 = sfb;
for (sfb = cod_info.sfb_smin; sfb < Encoder_1.SBMAX_s; sfb++) {
var start = gfc.scalefac_band.s[sfb];
var end = gfc.scalefac_band.s[sfb + 1];
var bw = end - start;
for (var i = 0; i < 3; i++) {
for (en0 = 0.0, l = start; l < end; l++) {
en0 += cod_info.xr[j] * cod_info.xr[j];
j++;
}
en0 = Math.max(en0 / bw, 1e-20);
/* convert to MDCT units */
/* scaling so it shows up on FFT plot */
en1 = 1e15;
gfc.pinfo.en_s[gr][ch][3 * sfb + i] = en1 * en0;
gfc.pinfo.xfsf_s[gr][ch][3 * sfb + i] = en1 * l3_xmin[sfb2]
* xfsf[sfb2] / bw;
if (ratio.en.s[sfb][i] > 0)
en0 = en0 / ratio.en.s[sfb][i];
else
en0 = 0.0;
if (gfp.ATHonly || gfp.ATHshort)
en0 = 0;
gfc.pinfo.thr_s[gr][ch][3 * sfb + i] = en1
* Math.max(en0 * ratio.thm.s[sfb][i],
gfc.ATH.s[sfb]);
/* there is no scalefactor bands >= SBPSY_s */
gfc.pinfo.LAMEsfb_s[gr][ch][3 * sfb + i] = -2.0
* cod_info.subblock_gain[i];
if (sfb < Encoder_1.SBPSY_s) {
gfc.pinfo.LAMEsfb_s[gr][ch][3 * sfb + i] -= ifqstep
* scalefac[sfb2];
}
sfb2++;
}
}
}
/* block type short */
gfc.pinfo.LAMEqss[gr][ch] = cod_info.global_gain;
gfc.pinfo.LAMEmainbits[gr][ch] = cod_info.part2_3_length
+ cod_info.part2_length;
gfc.pinfo.LAMEsfbits[gr][ch] = cod_info.part2_length;
gfc.pinfo.over[gr][ch] = noise.over_count;
gfc.pinfo.max_noise[gr][ch] = noise.max_noise * 10.0;
gfc.pinfo.over_noise[gr][ch] = noise.over_noise * 10.0;
gfc.pinfo.tot_noise[gr][ch] = noise.tot_noise * 10.0;
gfc.pinfo.over_SSD[gr][ch] = noise.over_SSD;
};
/**
* updates plotting data for a whole frame
*
* Robert Hegemann 2000-10-21
*/
}
var QuantizePVT_1 = QuantizePVT$1;
/*
* MP3 huffman table selecting and bit counting
*
* Copyright (c) 1999-2005 Takehiro TOMINAGA
* Copyright (c) 2002-2005 Gabriel Bouvigne
*
* 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 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
* Library 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.
*/
/* $Id: Takehiro.java,v 1.26 2011/05/24 20:48:06 kenchis Exp $ */
//package mp3;
//import java.util.Arrays;
var System$18 = common.System;
var Arrays$18 = common.Arrays;
var new_int$18 = common.new_int;
var assert$18 = common.assert;
function Takehiro$1() {
var qupvt = null;
this.qupvt = null;
this.setModules = function (_qupvt) {
this.qupvt = _qupvt;
qupvt = _qupvt;
};
function Bits(b) {
this.bits = 0 | b;
}
var subdv_table = [[0, 0], /* 0 bands */
[0, 0], /* 1 bands */
[0, 0], /* 2 bands */
[0, 0], /* 3 bands */
[0, 0], /* 4 bands */
[0, 1], /* 5 bands */
[1, 1], /* 6 bands */
[1, 1], /* 7 bands */
[1, 2], /* 8 bands */
[2, 2], /* 9 bands */
[2, 3], /* 10 bands */
[2, 3], /* 11 bands */
[3, 4], /* 12 bands */
[3, 4], /* 13 bands */
[3, 4], /* 14 bands */
[4, 5], /* 15 bands */
[4, 5], /* 16 bands */
[4, 6], /* 17 bands */
[5, 6], /* 18 bands */
[5, 6], /* 19 bands */
[5, 7], /* 20 bands */
[6, 7], /* 21 bands */
[6, 7], /* 22 bands */
];
/**
* nonlinear quantization of xr More accurate formula than the ISO formula.
* Takes into account the fact that we are quantizing xr . ix, but we want
* ix^4/3 to be as close as possible to x^4/3. (taking the nearest int would
* mean ix is as close as possible to xr, which is different.)
*
* From Segher Boessenkool * if XRPOW_FTOI(x) = nearest_int(x), then QUANTFAC(x)=adj43asm[x]
* ROUNDFAC= -0.0946
* * if XRPOW_FTOI(x) = floor(x), then QUANTFAC(x)=asj43[x]
* ROUNDFAC=0.4054
* * Note: using floor() or 0| is extremely slow. On machines where the * TAKEHIRO_IEEE754_HACK code above does not work, it is worthwile to write * some ASM for XRPOW_FTOI(). */ function quantize_lines_xrpow(l, istep, xr, xrPos, ix, ixPos) { assert$18(l > 0); l = l >> 1; var remaining = l % 2; l = l >> 1; while (l-- != 0) { var x0, x1, x2, x3; var rx0, rx1, rx2, rx3; x0 = xr[xrPos++] * istep; x1 = xr[xrPos++] * istep; rx0 = 0 | x0; x2 = xr[xrPos++] * istep; rx1 = 0 | x1; x3 = xr[xrPos++] * istep; rx2 = 0 | x2; x0 += qupvt.adj43[rx0]; rx3 = 0 | x3; x1 += qupvt.adj43[rx1]; ix[ixPos++] = 0 | x0; x2 += qupvt.adj43[rx2]; ix[ixPos++] = 0 | x1; x3 += qupvt.adj43[rx3]; ix[ixPos++] = 0 | x2; ix[ixPos++] = 0 | x3; } if (remaining != 0) { var x0, x1; var rx0, rx1; x0 = xr[xrPos++] * istep; x1 = xr[xrPos++] * istep; rx0 = 0 | x0; rx1 = 0 | x1; x0 += qupvt.adj43[rx0]; x1 += qupvt.adj43[rx1]; ix[ixPos++] = 0 | x0; ix[ixPos++] = 0 | x1; } } /** * Quantization function This function will select which lines to quantize * and call the proper quantization function */ function quantize_xrpow(xp, pi, istep, codInfo, prevNoise) { /* quantize on xr^(3/4) instead of xr */ var sfb; var sfbmax; var j = 0; var prev_data_use; var accumulate = 0; var accumulate01 = 0; var xpPos = 0; var iData = pi; var iDataPos = 0; var acc_iData = iData; var acc_iDataPos = 0; var acc_xp = xp; var acc_xpPos = 0; /* * Reusing previously computed data does not seems to work if global * gain is changed. Finding why it behaves this way would allow to use a * cache of previously computed values (let's 10 cached values per sfb) * that would probably provide a noticeable speedup */ prev_data_use = (prevNoise != null && (codInfo.global_gain == prevNoise.global_gain)); if (codInfo.block_type == Encoder_1.SHORT_TYPE) sfbmax = 38; else sfbmax = 21; for (sfb = 0; sfb <= sfbmax; sfb++) { var step = -1; if (prev_data_use || codInfo.block_type == Encoder_1.NORM_TYPE) { step = codInfo.global_gain - ((codInfo.scalefac[sfb] + (codInfo.preflag != 0 ? qupvt.pretab[sfb] : 0)) << (codInfo.scalefac_scale + 1)) - codInfo.subblock_gain[codInfo.window[sfb]] * 8; } assert$18(codInfo.width[sfb] >= 0); if (prev_data_use && (prevNoise.step[sfb] == step)) { /* * do not recompute this part, but compute accumulated lines */ if (accumulate != 0) { quantize_lines_xrpow(accumulate, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate = 0; } if (accumulate01 != 0) { quantize_lines_xrpow_01(accumulate01, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate01 = 0; } } else { /* should compute this part */ var l = codInfo.width[sfb]; if ((j + codInfo.width[sfb]) > codInfo.max_nonzero_coeff) { /* do not compute upper zero part */ var usefullsize; usefullsize = codInfo.max_nonzero_coeff - j + 1; Arrays$18.fill(pi, codInfo.max_nonzero_coeff, 576, 0); l = usefullsize; if (l < 0) { l = 0; } /* no need to compute higher sfb values */ sfb = sfbmax + 1; } /* accumulate lines to quantize */ if (0 == accumulate && 0 == accumulate01) { acc_iData = iData; acc_iDataPos = iDataPos; acc_xp = xp; acc_xpPos = xpPos; } if (prevNoise != null && prevNoise.sfb_count1 > 0 && sfb >= prevNoise.sfb_count1 && prevNoise.step[sfb] > 0 && step >= prevNoise.step[sfb]) { if (accumulate != 0) { quantize_lines_xrpow(accumulate, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate = 0; acc_iData = iData; acc_iDataPos = iDataPos; acc_xp = xp; acc_xpPos = xpPos; } accumulate01 += l; } else { if (accumulate01 != 0) { quantize_lines_xrpow_01(accumulate01, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate01 = 0; acc_iData = iData; acc_iDataPos = iDataPos; acc_xp = xp; acc_xpPos = xpPos; } accumulate += l; } if (l <= 0) { /* * rh: 20040215 may happen due to "prev_data_use" * optimization */ if (accumulate01 != 0) { quantize_lines_xrpow_01(accumulate01, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate01 = 0; } if (accumulate != 0) { quantize_lines_xrpow(accumulate, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate = 0; } break; /* ends for-loop */ } } if (sfb <= sfbmax) { iDataPos += codInfo.width[sfb]; xpPos += codInfo.width[sfb]; j += codInfo.width[sfb]; } } if (accumulate != 0) { /* last data part */ quantize_lines_xrpow(accumulate, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate = 0; } if (accumulate01 != 0) { /* last data part */ quantize_lines_xrpow_01(accumulate01, istep, acc_xp, acc_xpPos, acc_iData, acc_iDataPos); accumulate01 = 0; } } /** * ix_max */ function ix_max(ix, ixPos, endPos) { var max1 = 0, max2 = 0; do { var x1 = ix[ixPos++]; var x2 = ix[ixPos++]; if (max1 < x1) max1 = x1; if (max2 < x2) max2 = x2; } while (ixPos < endPos); if (max1 < max2) max1 = max2; return max1; } function count_bit_ESC(ix, ixPos, end, t1, t2, s) { /* ESC-table is used */ var linbits = Tables_1.ht[t1].xlen * 65536 + Tables_1.ht[t2].xlen; var sum = 0, sum2; do { var x = ix[ixPos++]; var y = ix[ixPos++]; if (x != 0) { if (x > 14) { x = 15; sum += linbits; } x *= 16; } if (y != 0) { if (y > 14) { y = 15; sum += linbits; } x += y; } sum += Tables_1.largetbl[x]; } while (ixPos < end); sum2 = sum & 0xffff; sum >>= 16; if (sum > sum2) { sum = sum2; t1 = t2; } s.bits += sum; return t1; } function count_bit_noESC(ix, ixPos, end, s) { /* No ESC-words */ var sum1 = 0; var hlen1 = Tables_1.ht[1].hlen; do { var x = ix[ixPos + 0] * 2 + ix[ixPos + 1]; ixPos += 2; sum1 += hlen1[x]; } while (ixPos < end); s.bits += sum1; return 1; } function count_bit_noESC_from2(ix, ixPos, end, t1, s) { /* No ESC-words */ var sum = 0, sum2; var xlen = Tables_1.ht[t1].xlen; var hlen; if (t1 == 2) hlen = Tables_1.table23; else hlen = Tables_1.table56; do { var x = ix[ixPos + 0] * xlen + ix[ixPos + 1]; ixPos += 2; sum += hlen[x]; } while (ixPos < end); sum2 = sum & 0xffff; sum >>= 16; if (sum > sum2) { sum = sum2; t1++; } s.bits += sum; return t1; } function count_bit_noESC_from3(ix, ixPos, end, t1, s) { /* No ESC-words */ var sum1 = 0; var sum2 = 0; var sum3 = 0; var xlen = Tables_1.ht[t1].xlen; var hlen1 = Tables_1.ht[t1].hlen; var hlen2 = Tables_1.ht[t1 + 1].hlen; var hlen3 = Tables_1.ht[t1 + 2].hlen; do { var x = ix[ixPos + 0] * xlen + ix[ixPos + 1]; ixPos += 2; sum1 += hlen1[x]; sum2 += hlen2[x]; sum3 += hlen3[x]; } while (ixPos < end); var t = t1; if (sum1 > sum2) { sum1 = sum2; t++; } if (sum1 > sum3) { sum1 = sum3; t = t1 + 2; } s.bits += sum1; return t; } /*************************************************************************/ /* choose table */ /*************************************************************************/ var huf_tbl_noESC = [1, 2, 5, 7, 7, 10, 10, 13, 13, 13, 13, 13, 13, 13, 13]; /** * Choose the Huffman table that will encode ix[begin..end] with the fewest * bits. * * Note: This code contains knowledge about the sizes and characteristics of * the Huffman tables as defined in the IS (Table B.7), and will not work * with any arbitrary tables. */ function choose_table(ix, ixPos, endPos, s) { var max = ix_max(ix, ixPos, endPos); switch (max) { case 0: return max; case 1: return count_bit_noESC(ix, ixPos, endPos, s); case 2: case 3: return count_bit_noESC_from2(ix, ixPos, endPos, huf_tbl_noESC[max - 1], s); case 4: case 5: case 6: case 7: case 8: case 9: case 10: case 11: case 12: case 13: case 14: case 15: return count_bit_noESC_from3(ix, ixPos, endPos, huf_tbl_noESC[max - 1], s); default: /* try tables with linbits */ if (max > QuantizePVT_1.IXMAX_VAL) { s.bits = QuantizePVT_1.LARGE_BITS; return -1; } max -= 15; var choice2; for (choice2 = 24; choice2 < 32; choice2++) { if (Tables_1.ht[choice2].linmax >= max) { break; } } var choice; for (choice = choice2 - 8; choice < 24; choice++) { if (Tables_1.ht[choice].linmax >= max) { break; } } return count_bit_ESC(ix, ixPos, endPos, choice, choice2, s); } } /** * count_bit */ this.noquant_count_bits = function (gfc, gi, prev_noise) { var ix = gi.l3_enc; var i = Math.min(576, ((gi.max_nonzero_coeff + 2) >> 1) << 1); if (prev_noise != null) prev_noise.sfb_count1 = 0; /* Determine count1 region */ for (; i > 1; i -= 2) if ((ix[i - 1] | ix[i - 2]) != 0) break; gi.count1 = i; /* Determines the number of bits to encode the quadruples. */ var a1 = 0; var a2 = 0; for (; i > 3; i -= 4) { var p; /* hack to check if all values <= 1 */ //throw "TODO: HACK if ((((long) ix[i - 1] | (long) ix[i - 2] | (long) ix[i - 3] | (long) ix[i - 4]) & 0xffffffffL) > 1L " //if (true) { if (((ix[i - 1] | ix[i - 2] | ix[i - 3] | ix[i - 4]) & 0x7fffffff) > 1) { break; } p = ((ix[i - 4] * 2 + ix[i - 3]) * 2 + ix[i - 2]) * 2 + ix[i - 1]; a1 += Tables_1.t32l[p]; a2 += Tables_1.t33l[p]; } var bits = a1; gi.count1table_select = 0; if (a1 > a2) { bits = a2; gi.count1table_select = 1; } gi.count1bits = bits; gi.big_values = i; if (i == 0) return bits; if (gi.block_type == Encoder_1.SHORT_TYPE) { a1 = 3 * gfc.scalefac_band.s[3]; if (a1 > gi.big_values) a1 = gi.big_values; a2 = gi.big_values; } else if (gi.block_type == Encoder_1.NORM_TYPE) { assert$18(i <= 576); /* bv_scf has 576 entries (0..575) */ a1 = gi.region0_count = gfc.bv_scf[i - 2]; a2 = gi.region1_count = gfc.bv_scf[i - 1]; assert$18(a1 + a2 + 2 < Encoder_1.SBPSY_l); a2 = gfc.scalefac_band.l[a1 + a2 + 2]; a1 = gfc.scalefac_band.l[a1 + 1]; if (a2 < i) { var bi = new Bits(bits); gi.table_select[2] = choose_table(ix, a2, i, bi); bits = bi.bits; } } else { gi.region0_count = 7; /* gi.region1_count = SBPSY_l - 7 - 1; */ gi.region1_count = Encoder_1.SBMAX_l - 1 - 7 - 1; a1 = gfc.scalefac_band.l[7 + 1]; a2 = i; if (a1 > a2) { a1 = a2; } } /* have to allow for the case when bigvalues < region0 < region1 */ /* (and region0, region1 are ignored) */ a1 = Math.min(a1, i); a2 = Math.min(a2, i); assert$18(a1 >= 0); assert$18(a2 >= 0); /* Count the number of bits necessary to code the bigvalues region. */ if (0 < a1) { var bi = new Bits(bits); gi.table_select[0] = choose_table(ix, 0, a1, bi); bits = bi.bits; } if (a1 < a2) { var bi = new Bits(bits); gi.table_select[1] = choose_table(ix, a1, a2, bi); bits = bi.bits; } if (gfc.use_best_huffman == 2) { gi.part2_3_length = bits; best_huffman_divide(gfc, gi); bits = gi.part2_3_length; } if (prev_noise != null) { if (gi.block_type == Encoder_1.NORM_TYPE) { var sfb = 0; while (gfc.scalefac_band.l[sfb] < gi.big_values) { sfb++; } prev_noise.sfb_count1 = sfb; } } return bits; }; this.count_bits = function (gfc, xr, gi, prev_noise) { var ix = gi.l3_enc; /* since quantize_xrpow uses table lookup, we need to check this first: */ var w = (QuantizePVT_1.IXMAX_VAL) / qupvt.IPOW20(gi.global_gain); if (gi.xrpow_max > w) return QuantizePVT_1.LARGE_BITS; quantize_xrpow(xr, ix, qupvt.IPOW20(gi.global_gain), gi, prev_noise); if ((gfc.substep_shaping & 2) != 0) { var j = 0; /* 0.634521682242439 = 0.5946*2**(.5*0.1875) */ var gain = gi.global_gain + gi.scalefac_scale; var roundfac = 0.634521682242439 / qupvt.IPOW20(gain); for (var sfb = 0; sfb < gi.sfbmax; sfb++) { var width = gi.width[sfb]; assert$18(width >= 0); if (0 == gfc.pseudohalf[sfb]) { j += width; } else { var k; for (k = j, j += width; k < j; ++k) { ix[k] = (xr[k] >= roundfac) ? ix[k] : 0; } } } } return this.noquant_count_bits(gfc, gi, prev_noise); }; /** * re-calculate the best scalefac_compress using scfsi the saved bits are * kept in the bit reservoir. */ function recalc_divide_init(gfc, cod_info, ix, r01_bits, r01_div, r0_tbl, r1_tbl) { var bigv = cod_info.big_values; for (var r0 = 0; r0 <= 7 + 15; r0++) { r01_bits[r0] = QuantizePVT_1.LARGE_BITS; } for (var r0 = 0; r0 < 16; r0++) { var a1 = gfc.scalefac_band.l[r0 + 1]; if (a1 >= bigv) break; var r0bits = 0; var bi = new Bits(r0bits); var r0t = choose_table(ix, 0, a1, bi); r0bits = bi.bits; for (var r1 = 0; r1 < 8; r1++) { var a2 = gfc.scalefac_band.l[r0 + r1 + 2]; if (a2 >= bigv) break; var bits = r0bits; bi = new Bits(bits); var r1t = choose_table(ix, a1, a2, bi); bits = bi.bits; if (r01_bits[r0 + r1] > bits) { r01_bits[r0 + r1] = bits; r01_div[r0 + r1] = r0; r0_tbl[r0 + r1] = r0t; r1_tbl[r0 + r1] = r1t; } } } } function recalc_divide_sub(gfc, cod_info2, gi, ix, r01_bits, r01_div, r0_tbl, r1_tbl) { var bigv = cod_info2.big_values; for (var r2 = 2; r2 < Encoder_1.SBMAX_l + 1; r2++) { var a2 = gfc.scalefac_band.l[r2]; if (a2 >= bigv) break; var bits = r01_bits[r2 - 2] + cod_info2.count1bits; if (gi.part2_3_length <= bits) break; var bi = new Bits(bits); var r2t = choose_table(ix, a2, bigv, bi); bits = bi.bits; if (gi.part2_3_length <= bits) continue; gi.assign(cod_info2); gi.part2_3_length = bits; gi.region0_count = r01_div[r2 - 2]; gi.region1_count = r2 - 2 - r01_div[r2 - 2]; gi.table_select[0] = r0_tbl[r2 - 2]; gi.table_select[1] = r1_tbl[r2 - 2]; gi.table_select[2] = r2t; } } this.best_huffman_divide = function (gfc, gi) { var cod_info2 = new GrInfo_1(); var ix = gi.l3_enc; var r01_bits = new_int$18(7 + 15 + 1); var r01_div = new_int$18(7 + 15 + 1); var r0_tbl = new_int$18(7 + 15 + 1); var r1_tbl = new_int$18(7 + 15 + 1); /* SHORT BLOCK stuff fails for MPEG2 */ if (gi.block_type == Encoder_1.SHORT_TYPE && gfc.mode_gr == 1) return; cod_info2.assign(gi); if (gi.block_type == Encoder_1.NORM_TYPE) { recalc_divide_init(gfc, gi, ix, r01_bits, r01_div, r0_tbl, r1_tbl); recalc_divide_sub(gfc, cod_info2, gi, ix, r01_bits, r01_div, r0_tbl, r1_tbl); } var i = cod_info2.big_values; if (i == 0 || (ix[i - 2] | ix[i - 1]) > 1) return; i = gi.count1 + 2; if (i > 576) return; /* Determines the number of bits to encode the quadruples. */ cod_info2.assign(gi); cod_info2.count1 = i; var a1 = 0; var a2 = 0; assert$18(i <= 576); for (; i > cod_info2.big_values; i -= 4) { var p = ((ix[i - 4] * 2 + ix[i - 3]) * 2 + ix[i - 2]) * 2 + ix[i - 1]; a1 += Tables_1.t32l[p]; a2 += Tables_1.t33l[p]; } cod_info2.big_values = i; cod_info2.count1table_select = 0; if (a1 > a2) { a1 = a2; cod_info2.count1table_select = 1; } cod_info2.count1bits = a1; if (cod_info2.block_type == Encoder_1.NORM_TYPE) recalc_divide_sub(gfc, cod_info2, gi, ix, r01_bits, r01_div, r0_tbl, r1_tbl); else { /* Count the number of bits necessary to code the bigvalues region. */ cod_info2.part2_3_length = a1; a1 = gfc.scalefac_band.l[7 + 1]; if (a1 > i) { a1 = i; } if (a1 > 0) { var bi = new Bits(cod_info2.part2_3_length); cod_info2.table_select[0] = choose_table(ix, 0, a1, bi); cod_info2.part2_3_length = bi.bits; } if (i > a1) { var bi = new Bits(cod_info2.part2_3_length); cod_info2.table_select[1] = choose_table(ix, a1, i, bi); cod_info2.part2_3_length = bi.bits; } if (gi.part2_3_length > cod_info2.part2_3_length) gi.assign(cod_info2); } }; var slen1_n = [1, 1, 1, 1, 8, 2, 2, 2, 4, 4, 4, 8, 8, 8, 16, 16]; var slen2_n = [1, 2, 4, 8, 1, 2, 4, 8, 2, 4, 8, 2, 4, 8, 4, 8]; var slen1_tab = [0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4]; var slen2_tab = [0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3]; Takehiro$1.slen1_tab = slen1_tab; Takehiro$1.slen2_tab = slen2_tab; function scfsi_calc(ch, l3_side) { var sfb; var gi = l3_side.tt[1][ch]; var g0 = l3_side.tt[0][ch]; for (var i = 0; i < Tables_1.scfsi_band.length - 1; i++) { for (sfb = Tables_1.scfsi_band[i]; sfb < Tables_1.scfsi_band[i + 1]; sfb++) { if (g0.scalefac[sfb] != gi.scalefac[sfb] && gi.scalefac[sfb] >= 0) break; } if (sfb == Tables_1.scfsi_band[i + 1]) { for (sfb = Tables_1.scfsi_band[i]; sfb < Tables_1.scfsi_band[i + 1]; sfb++) { gi.scalefac[sfb] = -1; } l3_side.scfsi[ch][i] = 1; } } var s1 = 0; var c1 = 0; for (sfb = 0; sfb < 11; sfb++) { if (gi.scalefac[sfb] == -1) continue; c1++; if (s1 < gi.scalefac[sfb]) s1 = gi.scalefac[sfb]; } var s2 = 0; var c2 = 0; for (; sfb < Encoder_1.SBPSY_l; sfb++) { if (gi.scalefac[sfb] == -1) continue; c2++; if (s2 < gi.scalefac[sfb]) s2 = gi.scalefac[sfb]; } for (var i = 0; i < 16; i++) { if (s1 < slen1_n[i] && s2 < slen2_n[i]) { var c = slen1_tab[i] * c1 + slen2_tab[i] * c2; if (gi.part2_length > c) { gi.part2_length = c; gi.scalefac_compress = i; } } } } /** * Find the optimal way to store the scalefactors. Only call this routine * after final scalefactors have been chosen and the channel/granule will * not be re-encoded. */ this.best_scalefac_store = function (gfc, gr, ch, l3_side) { /* use scalefac_scale if we can */ var gi = l3_side.tt[gr][ch]; var sfb, i, j, l; var recalc = 0; /* * remove scalefacs from bands with ix=0. This idea comes from the AAC * ISO docs. added mt 3/00 */ /* check if l3_enc=0 */ j = 0; for (sfb = 0; sfb < gi.sfbmax; sfb++) { var width = gi.width[sfb]; assert$18(width >= 0); j += width; for (l = -width; l < 0; l++) { if (gi.l3_enc[l + j] != 0) break; } if (l == 0) gi.scalefac[sfb] = recalc = -2; /* anything goes. */ /* * only best_scalefac_store and calc_scfsi know--and only they * should know--about the magic number -2. */ } if (0 == gi.scalefac_scale && 0 == gi.preflag) { var s = 0; for (sfb = 0; sfb < gi.sfbmax; sfb++) if (gi.scalefac[sfb] > 0) s |= gi.scalefac[sfb]; if (0 == (s & 1) && s != 0) { for (sfb = 0; sfb < gi.sfbmax; sfb++) if (gi.scalefac[sfb] > 0) gi.scalefac[sfb] >>= 1; gi.scalefac_scale = recalc = 1; } } if (0 == gi.preflag && gi.block_type != Encoder_1.SHORT_TYPE && gfc.mode_gr == 2) { for (sfb = 11; sfb < Encoder_1.SBPSY_l; sfb++) if (gi.scalefac[sfb] < qupvt.pretab[sfb] && gi.scalefac[sfb] != -2) break; if (sfb == Encoder_1.SBPSY_l) { for (sfb = 11; sfb < Encoder_1.SBPSY_l; sfb++) if (gi.scalefac[sfb] > 0) gi.scalefac[sfb] -= qupvt.pretab[sfb]; gi.preflag = recalc = 1; } } for (i = 0; i < 4; i++) l3_side.scfsi[ch][i] = 0; if (gfc.mode_gr == 2 && gr == 1 && l3_side.tt[0][ch].block_type != Encoder_1.SHORT_TYPE && l3_side.tt[1][ch].block_type != Encoder_1.SHORT_TYPE) { scfsi_calc(ch, l3_side); recalc = 0; } for (sfb = 0; sfb < gi.sfbmax; sfb++) { if (gi.scalefac[sfb] == -2) { gi.scalefac[sfb] = 0; /* if anything goes, then 0 is a good choice */ } } if (recalc != 0) { if (gfc.mode_gr == 2) { this.scale_bitcount(gi); } else { this.scale_bitcount_lsf(gfc, gi); } } }; function all_scalefactors_not_negative(scalefac, n) { for (var i = 0; i < n; ++i) { if (scalefac[i] < 0) return false; } return true; } /** * number of bits used to encode scalefacs. * * 18*slen1_tab[i] + 18*slen2_tab[i] */ var scale_short = [0, 18, 36, 54, 54, 36, 54, 72, 54, 72, 90, 72, 90, 108, 108, 126]; /** * number of bits used to encode scalefacs. * * 17*slen1_tab[i] + 18*slen2_tab[i] */ var scale_mixed = [0, 18, 36, 54, 51, 35, 53, 71, 52, 70, 88, 69, 87, 105, 104, 122]; /** * number of bits used to encode scalefacs. * * 11*slen1_tab[i] + 10*slen2_tab[i] */ var scale_long = [0, 10, 20, 30, 33, 21, 31, 41, 32, 42, 52, 43, 53, 63, 64, 74]; /** * Also calculates the number of bits necessary to code the scalefactors. */ this.scale_bitcount = function (cod_info) { var k, sfb, max_slen1 = 0, max_slen2 = 0; /* maximum values */ var tab; var scalefac = cod_info.scalefac; assert$18(all_scalefactors_not_negative(scalefac, cod_info.sfbmax)); if (cod_info.block_type == Encoder_1.SHORT_TYPE) { tab = scale_short; if (cod_info.mixed_block_flag != 0) tab = scale_mixed; } else { /* block_type == 1,2,or 3 */ tab = scale_long; if (0 == cod_info.preflag) { for (sfb = 11; sfb < Encoder_1.SBPSY_l; sfb++) if (scalefac[sfb] < qupvt.pretab[sfb]) break; if (sfb == Encoder_1.SBPSY_l) { cod_info.preflag = 1; for (sfb = 11; sfb < Encoder_1.SBPSY_l; sfb++) scalefac[sfb] -= qupvt.pretab[sfb]; } } } for (sfb = 0; sfb < cod_info.sfbdivide; sfb++) if (max_slen1 < scalefac[sfb]) max_slen1 = scalefac[sfb]; for (; sfb < cod_info.sfbmax; sfb++) if (max_slen2 < scalefac[sfb]) max_slen2 = scalefac[sfb]; /* * from Takehiro TOMINAGA
* copy data out of the internal MP3 bit buffer into a user supplied
* unsigned char buffer.
*
* mp3data=0 indicates data in buffer is an id3tags and VBR tags
* mp3data=1 data is real mp3 frame data.
*
*/
this.copy_buffer = function (gfc, buffer, bufferPos, size, mp3data) {
var minimum = bufByteIdx + 1;
if (minimum <= 0)
return 0;
if (size != 0 && minimum > size) {
/* buffer is too small */
return -1;
}
System$17.arraycopy(buf, 0, buffer, bufferPos, minimum);
bufByteIdx = -1;
bufBitIdx = 0;
if (mp3data != 0) {
var crc = new_int$17(1);
crc[0] = gfc.nMusicCRC;
vbr.updateMusicCRC(crc, buffer, bufferPos, minimum);
gfc.nMusicCRC = crc[0];
/**
* sum number of bytes belonging to the mp3 stream this info will be
* written into the Xing/LAME header for seeking
*/
if (minimum > 0) {
gfc.VBR_seek_table.nBytesWritten += minimum;
}
if (gfc.decode_on_the_fly) { /* decode the frame */
var pcm_buf = new_float_n$17([2, 1152]);
var mp3_in = minimum;
var samples_out = -1;
var i;
/* re-synthesis to pcm. Repeat until we get a samples_out=0 */
while (samples_out != 0) {
samples_out = mpg.hip_decode1_unclipped(gfc.hip, buffer,
bufferPos, mp3_in, pcm_buf[0], pcm_buf[1]);
/*
* samples_out = 0: need more data to decode samples_out =
* -1: error. Lets assume 0 pcm output samples_out = number
* of samples output
*/
/*
* set the lenght of the mp3 input buffer to zero, so that
* in the next iteration of the loop we will be querying
* mpglib about buffered data
*/
mp3_in = 0;
if (samples_out == -1) {
/*
* error decoding. Not fatal, but might screw up the
* ReplayGain tag. What should we do? Ignore for now
*/
samples_out = 0;
}
if (samples_out > 0) {
/* process the PCM data */
/*
* this should not be possible, and indicates we have
* overflown the pcm_buf buffer
*/
assert$17(samples_out <= 1152);
if (gfc.findPeakSample) {
for (i = 0; i < samples_out; i++) {
if (pcm_buf[0][i] > gfc.PeakSample)
gfc.PeakSample = pcm_buf[0][i];
else if (-pcm_buf[0][i] > gfc.PeakSample)
gfc.PeakSample = -pcm_buf[0][i];
}
if (gfc.channels_out > 1)
for (i = 0; i < samples_out; i++) {
if (pcm_buf[1][i] > gfc.PeakSample)
gfc.PeakSample = pcm_buf[1][i];
else if (-pcm_buf[1][i] > gfc.PeakSample)
gfc.PeakSample = -pcm_buf[1][i];
}
}
if (gfc.findReplayGain)
if (ga.AnalyzeSamples(gfc.rgdata, pcm_buf[0], 0,
pcm_buf[1], 0, samples_out,
gfc.channels_out) == GainAnalysis.GAIN_ANALYSIS_ERROR)
return -6;
}
/* if (samples_out>0) */
}
/* while (samples_out!=0) */
}
/* if (gfc.decode_on_the_fly) */
}
/* if (mp3data) */
return minimum;
};
this.init_bit_stream_w = function (gfc) {
buf = new_byte$17(Lame.LAME_MAXMP3BUFFER);
gfc.h_ptr = gfc.w_ptr = 0;
gfc.header[gfc.h_ptr].write_timing = 0;
bufByteIdx = -1;
bufBitIdx = 0;
totbit = 0;
};
// From machine.h
}
var BitStream_1 = BitStream$1;
var System$2 = common.System;
var VbrMode$2 = common.VbrMode;
var ShortBlock$2 = common.ShortBlock;
var new_float$2 = common.new_float;
var new_int_n$2 = common.new_int_n;
var new_short_n$1 = common.new_short_n;
var assert$2 = common.assert;
function Lame$1() {
var self = this;
var LAME_MAXALBUMART = (128 * 1024);
Lame$1.V9 = 410;
Lame$1.V8 = 420;
Lame$1.V7 = 430;
Lame$1.V6 = 440;
Lame$1.V5 = 450;
Lame$1.V4 = 460;
Lame$1.V3 = 470;
Lame$1.V2 = 480;
Lame$1.V1 = 490;
Lame$1.V0 = 500;
/* still there for compatibility */
Lame$1.R3MIX = 1000;
Lame$1.STANDARD = 1001;
Lame$1.EXTREME = 1002;
Lame$1.INSANE = 1003;
Lame$1.STANDARD_FAST = 1004;
Lame$1.EXTREME_FAST = 1005;
Lame$1.MEDIUM = 1006;
Lame$1.MEDIUM_FAST = 1007;
/**
* maximum size of mp3buffer needed if you encode at most 1152 samples for
* each call to lame_encode_buffer. see lame_encode_buffer() below
* (LAME_MAXMP3BUFFER is now obsolete)
*/
var LAME_MAXMP3BUFFER = (16384 + LAME_MAXALBUMART);
Lame$1.LAME_MAXMP3BUFFER = LAME_MAXMP3BUFFER;
var ga;
var bs;
var p;
var qupvt;
var qu;
var psy = new PsyModel_1();
var vbr;
var ver;
var id3;
var mpglib;
this.enc = new Encoder_1();
this.setModules = function (_ga, _bs, _p, _qupvt, _qu, _vbr, _ver, _id3, _mpglib) {
ga = _ga;
bs = _bs;
p = _p;
qupvt = _qupvt;
qu = _qu;
vbr = _vbr;
ver = _ver;
id3 = _id3;
mpglib = _mpglib;
this.enc.setModules(bs, psy, qupvt, vbr);
};
/**
* PSY Model related stuff
*/
function PSY() {
/**
* The dbQ stuff.
*/
this.mask_adjust = 0.;
/**
* The dbQ stuff.
*/
this.mask_adjust_short = 0.;
/* at transition from one scalefactor band to next */
/**
* Band weight long scalefactor bands.
*/
this.bo_l_weight = new_float$2(Encoder_1.SBMAX_l);
/**
* Band weight short scalefactor bands.
*/
this.bo_s_weight = new_float$2(Encoder_1.SBMAX_s);
}
function LowPassHighPass() {
this.lowerlimit = 0.;
}
function BandPass(bitrate, lPass) {
this.lowpass = lPass;
}
var LAME_ID = 0xFFF88E3B;
function lame_init_old(gfp) {
var gfc;
gfp.class_id = LAME_ID;
gfc = gfp.internal_flags = new LameInternalFlags_1();
/* Global flags. set defaults here for non-zero values */
/* see lame.h for description */
/*
* set integer values to -1 to mean that LAME will compute the best
* value, UNLESS the calling program as set it (and the value is no
* longer -1)
*/
gfp.mode = MPEGMode.NOT_SET;
gfp.original = 1;
gfp.in_samplerate = 44100;
gfp.num_channels = 2;
gfp.num_samples = -1;
gfp.bWriteVbrTag = true;
gfp.quality = -1;
gfp.short_blocks = null;
gfc.subblock_gain = -1;
gfp.lowpassfreq = 0;
gfp.highpassfreq = 0;
gfp.lowpasswidth = -1;
gfp.highpasswidth = -1;
gfp.VBR = VbrMode$2.vbr_off;
gfp.VBR_q = 4;
gfp.ATHcurve = -1;
gfp.VBR_mean_bitrate_kbps = 128;
gfp.VBR_min_bitrate_kbps = 0;
gfp.VBR_max_bitrate_kbps = 0;
gfp.VBR_hard_min = 0;
gfc.VBR_min_bitrate = 1;
/* not 0 ????? */
gfc.VBR_max_bitrate = 13;
/* not 14 ????? */
gfp.quant_comp = -1;
gfp.quant_comp_short = -1;
gfp.msfix = -1;
gfc.resample_ratio = 1;
gfc.OldValue[0] = 180;
gfc.OldValue[1] = 180;
gfc.CurrentStep[0] = 4;
gfc.CurrentStep[1] = 4;
gfc.masking_lower = 1;
gfc.nsPsy.attackthre = -1;
gfc.nsPsy.attackthre_s = -1;
gfp.scale = -1;
gfp.athaa_type = -1;
gfp.ATHtype = -1;
/* default = -1 = set in lame_init_params */
gfp.athaa_loudapprox = -1;
/* 1 = flat loudness approx. (total energy) */
/* 2 = equal loudness curve */
gfp.athaa_sensitivity = 0.0;
/* no offset */
gfp.useTemporal = null;
gfp.interChRatio = -1;
/*
* The reason for int mf_samples_to_encode = ENCDELAY + POSTDELAY;
* ENCDELAY = internal encoder delay. And then we have to add
* POSTDELAY=288 because of the 50% MDCT overlap. A 576 MDCT granule
* decodes to 1152 samples. To synthesize the 576 samples centered under
* this granule we need the previous granule for the first 288 samples
* (no problem), and the next granule for the next 288 samples (not
* possible if this is last granule). So we need to pad with 288 samples
* to make sure we can encode the 576 samples we are interested in.
*/
gfc.mf_samples_to_encode = Encoder_1.ENCDELAY + Encoder_1.POSTDELAY;
gfp.encoder_padding = 0;
gfc.mf_size = Encoder_1.ENCDELAY - Encoder_1.MDCTDELAY;
/*
* we pad input with this many 0's
*/
gfp.findReplayGain = false;
gfp.decode_on_the_fly = false;
gfc.decode_on_the_fly = false;
gfc.findReplayGain = false;
gfc.findPeakSample = false;
gfc.RadioGain = 0;
gfc.AudiophileGain = 0;
gfc.noclipGainChange = 0;
gfc.noclipScale = -1.0;
gfp.preset = 0;
gfp.write_id3tag_automatic = true;
return 0;
}
this.lame_init = function () {
var gfp = new LameGlobalFlags_1();
var ret = lame_init_old(gfp);
if (ret != 0) {
return null;
}
gfp.lame_allocated_gfp = 1;
return gfp;
};
function filter_coef(x) {
if (x > 1.0)
return 0.0;
if (x <= 0.0)
return 1.0;
return Math.cos(Math.PI / 2 * x);
}
this.nearestBitrateFullIndex = function (bitrate) {
/* borrowed from DM abr presets */
var full_bitrate_table = [8, 16, 24, 32, 40, 48, 56, 64, 80,
96, 112, 128, 160, 192, 224, 256, 320];
var lower_range = 0, lower_range_kbps = 0, upper_range = 0, upper_range_kbps = 0;
/* We assume specified bitrate will be 320kbps */
upper_range_kbps = full_bitrate_table[16];
upper_range = 16;
lower_range_kbps = full_bitrate_table[16];
lower_range = 16;
/*
* Determine which significant bitrates the value specified falls
* between, if loop ends without breaking then we were correct above
* that the value was 320
*/
for (var b = 0; b < 16; b++) {
if ((Math.max(bitrate, full_bitrate_table[b + 1])) != bitrate) {
upper_range_kbps = full_bitrate_table[b + 1];
upper_range = b + 1;
lower_range_kbps = full_bitrate_table[b];
lower_range = (b);
break;
/* We found upper range */
}
}
/* Determine which range the value specified is closer to */
if ((upper_range_kbps - bitrate) > (bitrate - lower_range_kbps)) {
return lower_range;
}
return upper_range;
};
function optimum_samplefreq(lowpassfreq, input_samplefreq) {
/*
* Rules:
*
* - if possible, sfb21 should NOT be used
*/
var suggested_samplefreq = 44100;
if (input_samplefreq >= 48000)
suggested_samplefreq = 48000;
else if (input_samplefreq >= 44100)
suggested_samplefreq = 44100;
else if (input_samplefreq >= 32000)
suggested_samplefreq = 32000;
else if (input_samplefreq >= 24000)
suggested_samplefreq = 24000;
else if (input_samplefreq >= 22050)
suggested_samplefreq = 22050;
else if (input_samplefreq >= 16000)
suggested_samplefreq = 16000;
else if (input_samplefreq >= 12000)
suggested_samplefreq = 12000;
else if (input_samplefreq >= 11025)
suggested_samplefreq = 11025;
else if (input_samplefreq >= 8000)
suggested_samplefreq = 8000;
if (lowpassfreq == -1)
return suggested_samplefreq;
if (lowpassfreq <= 15960)
suggested_samplefreq = 44100;
if (lowpassfreq <= 15250)
suggested_samplefreq = 32000;
if (lowpassfreq <= 11220)
suggested_samplefreq = 24000;
if (lowpassfreq <= 9970)
suggested_samplefreq = 22050;
if (lowpassfreq <= 7230)
suggested_samplefreq = 16000;
if (lowpassfreq <= 5420)
suggested_samplefreq = 12000;
if (lowpassfreq <= 4510)
suggested_samplefreq = 11025;
if (lowpassfreq <= 3970)
suggested_samplefreq = 8000;
if (input_samplefreq < suggested_samplefreq) {
/*
* choose a valid MPEG sample frequency above the input sample
* frequency to avoid SFB21/12 bitrate bloat rh 061115
*/
if (input_samplefreq > 44100) {
return 48000;
}
if (input_samplefreq > 32000) {
return 44100;
}
if (input_samplefreq > 24000) {
return 32000;
}
if (input_samplefreq > 22050) {
return 24000;
}
if (input_samplefreq > 16000) {
return 22050;
}
if (input_samplefreq > 12000) {
return 16000;
}
if (input_samplefreq > 11025) {
return 12000;
}
if (input_samplefreq > 8000) {
return 11025;
}
return 8000;
}
return suggested_samplefreq;
}
/**
* convert samp freq in Hz to index
*/
function SmpFrqIndex(sample_freq, gpf) {
switch (sample_freq) {
case 44100:
gpf.version = 1;
return 0;
case 48000:
gpf.version = 1;
return 1;
case 32000:
gpf.version = 1;
return 2;
case 22050:
gpf.version = 0;
return 0;
case 24000:
gpf.version = 0;
return 1;
case 16000:
gpf.version = 0;
return 2;
case 11025:
gpf.version = 0;
return 0;
case 12000:
gpf.version = 0;
return 1;
case 8000:
gpf.version = 0;
return 2;
default:
gpf.version = 0;
return -1;
}
}
/**
* @param bRate
* legal rates from 8 to 320
*/
function FindNearestBitrate(bRate, version, samplerate) {
/* MPEG-1 or MPEG-2 LSF */
if (samplerate < 16000)
version = 2;
var bitrate = Tables_1.bitrate_table[version][1];
for (var i = 2; i <= 14; i++) {
if (Tables_1.bitrate_table[version][i] > 0) {
if (Math.abs(Tables_1.bitrate_table[version][i] - bRate) < Math
.abs(bitrate - bRate))
bitrate = Tables_1.bitrate_table[version][i];
}
}
return bitrate;
}
/**
* @param bRate
* legal rates from 32 to 448 kbps
* @param version
* MPEG-1 or MPEG-2/2.5 LSF
*/
function BitrateIndex(bRate, version, samplerate) {
/* convert bitrate in kbps to index */
if (samplerate < 16000)
version = 2;
for (var i = 0; i <= 14; i++) {
if (Tables_1.bitrate_table[version][i] > 0) {
if (Tables_1.bitrate_table[version][i] == bRate) {
return i;
}
}
}
return -1;
}
function optimum_bandwidth(lh, bitrate) {
/**
*
* Input:
* bitrate total bitrate in kbps
*
* Output:
* lowerlimit: best lowpass frequency limit for input filter in Hz
* upperlimit: best highpass frequency limit for input filter in Hz
*
*/
var freq_map = [new BandPass(8, 2000),
new BandPass(16, 3700), new BandPass(24, 3900),
new BandPass(32, 5500), new BandPass(40, 7000),
new BandPass(48, 7500), new BandPass(56, 10000),
new BandPass(64, 11000), new BandPass(80, 13500),
new BandPass(96, 15100), new BandPass(112, 15600),
new BandPass(128, 17000), new BandPass(160, 17500),
new BandPass(192, 18600), new BandPass(224, 19400),
new BandPass(256, 19700), new BandPass(320, 20500)];
var table_index = self.nearestBitrateFullIndex(bitrate);
lh.lowerlimit = freq_map[table_index].lowpass;
}
function lame_init_params_ppflt(gfp) {
var gfc = gfp.internal_flags;
/***************************************************************/
/* compute info needed for polyphase filter (filter type==0, default) */
/***************************************************************/
var lowpass_band = 32;
var highpass_band = -1;
if (gfc.lowpass1 > 0) {
var minband = 999;
for (var band = 0; band <= 31; band++) {
var freq = (band / 31.0);
/* this band and above will be zeroed: */
if (freq >= gfc.lowpass2) {
lowpass_band = Math.min(lowpass_band, band);
}
if (gfc.lowpass1 < freq && freq < gfc.lowpass2) {
minband = Math.min(minband, band);
}
}
/*
* compute the *actual* transition band implemented by the polyphase
* filter
*/
if (minband == 999) {
gfc.lowpass1 = (lowpass_band - .75) / 31.0;
} else {
gfc.lowpass1 = (minband - .75) / 31.0;
}
gfc.lowpass2 = lowpass_band / 31.0;
}
/*
* make sure highpass filter is within 90% of what the effective
* highpass frequency will be
*/
if (gfc.highpass2 > 0) {
if (gfc.highpass2 < .9 * (.75 / 31.0)) {
gfc.highpass1 = 0;
gfc.highpass2 = 0;
System$2.err.println("Warning: highpass filter disabled. "
+ "highpass frequency too small\n");
}
}
if (gfc.highpass2 > 0) {
var maxband = -1;
for (var band = 0; band <= 31; band++) {
var freq = band / 31.0;
/* this band and below will be zereod */
if (freq <= gfc.highpass1) {
highpass_band = Math.max(highpass_band, band);
}
if (gfc.highpass1 < freq && freq < gfc.highpass2) {
maxband = Math.max(maxband, band);
}
}
/*
* compute the *actual* transition band implemented by the polyphase
* filter
*/
gfc.highpass1 = highpass_band / 31.0;
if (maxband == -1) {
gfc.highpass2 = (highpass_band + .75) / 31.0;
} else {
gfc.highpass2 = (maxband + .75) / 31.0;
}
}
for (var band = 0; band < 32; band++) {
var fc1, fc2;
var freq = band / 31.0;
if (gfc.highpass2 > gfc.highpass1) {
fc1 = filter_coef((gfc.highpass2 - freq)
/ (gfc.highpass2 - gfc.highpass1 + 1e-20));
} else {
fc1 = 1.0;
}
if (gfc.lowpass2 > gfc.lowpass1) {
fc2 = filter_coef((freq - gfc.lowpass1)
/ (gfc.lowpass2 - gfc.lowpass1 + 1e-20));
} else {
fc2 = 1.0;
}
gfc.amp_filter[band] = (fc1 * fc2);
}
}
function lame_init_qval(gfp) {
var gfc = gfp.internal_flags;
switch (gfp.quality) {
default:
case 9: /* no psymodel, no noise shaping */
gfc.psymodel = 0;
gfc.noise_shaping = 0;
gfc.noise_shaping_amp = 0;
gfc.noise_shaping_stop = 0;
gfc.use_best_huffman = 0;
gfc.full_outer_loop = 0;
break;
case 8:
gfp.quality = 7;
//$FALL-THROUGH$
case 7:
/*
* use psymodel (for short block and m/s switching), but no noise
* shapping
*/
gfc.psymodel = 1;
gfc.noise_shaping = 0;
gfc.noise_shaping_amp = 0;
gfc.noise_shaping_stop = 0;
gfc.use_best_huffman = 0;
gfc.full_outer_loop = 0;
break;
case 6:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
gfc.noise_shaping_amp = 0;
gfc.noise_shaping_stop = 0;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 0;
gfc.full_outer_loop = 0;
break;
case 5:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
gfc.noise_shaping_amp = 0;
gfc.noise_shaping_stop = 0;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 0;
gfc.full_outer_loop = 0;
break;
case 4:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
gfc.noise_shaping_amp = 0;
gfc.noise_shaping_stop = 0;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 1;
gfc.full_outer_loop = 0;
break;
case 3:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
gfc.noise_shaping_amp = 1;
gfc.noise_shaping_stop = 1;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 1;
gfc.full_outer_loop = 0;
break;
case 2:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
if (gfc.substep_shaping == 0)
gfc.substep_shaping = 2;
gfc.noise_shaping_amp = 1;
gfc.noise_shaping_stop = 1;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 1;
/* inner loop */
gfc.full_outer_loop = 0;
break;
case 1:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
if (gfc.substep_shaping == 0)
gfc.substep_shaping = 2;
gfc.noise_shaping_amp = 2;
gfc.noise_shaping_stop = 1;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 1;
gfc.full_outer_loop = 0;
break;
case 0:
gfc.psymodel = 1;
if (gfc.noise_shaping == 0)
gfc.noise_shaping = 1;
if (gfc.substep_shaping == 0)
gfc.substep_shaping = 2;
gfc.noise_shaping_amp = 2;
gfc.noise_shaping_stop = 1;
if (gfc.subblock_gain == -1)
gfc.subblock_gain = 1;
gfc.use_best_huffman = 1;
/*
* type 2 disabled because of it slowness, in favor of full outer
* loop search
*/
gfc.full_outer_loop = 0;
/*
* full outer loop search disabled because of audible distortions it
* may generate rh 060629
*/
break;
}
}
function lame_init_bitstream(gfp) {
var gfc = gfp.internal_flags;
gfp.frameNum = 0;
if (gfp.write_id3tag_automatic) {
id3.id3tag_write_v2(gfp);
}
/* initialize histogram data optionally used by frontend */
gfc.bitrate_stereoMode_Hist = new_int_n$2([16, 4 + 1]);
gfc.bitrate_blockType_Hist = new_int_n$2([16, 4 + 1 + 1]);
gfc.PeakSample = 0.0;
/* Write initial VBR Header to bitstream and init VBR data */
if (gfp.bWriteVbrTag)
vbr.InitVbrTag(gfp);
}
/********************************************************************
* initialize internal params based on data in gf (globalflags struct filled
* in by calling program)
*
* OUTLINE:
*
* We first have some complex code to determine bitrate, output samplerate
* and mode. It is complicated by the fact that we allow the user to set
* some or all of these parameters, and need to determine best possible
* values for the rest of them:
*
* 1. set some CPU related flags 2. check if we are mono.mono, stereo.mono
* or stereo.stereo 3. compute bitrate and output samplerate: user may have
* set compression ratio user may have set a bitrate user may have set a
* output samplerate 4. set some options which depend on output samplerate
* 5. compute the actual compression ratio 6. set mode based on compression
* ratio
*
* The remaining code is much simpler - it just sets options based on the
* mode & compression ratio:
*
* set allow_diff_short based on mode select lowpass filter based on
* compression ratio & mode set the bitrate index, and min/max bitrates for
* VBR modes disable VBR tag if it is not appropriate initialize the
* bitstream initialize scalefac_band data set sideinfo_len (based on
* channels, CRC, out_samplerate) write an id3v2 tag into the bitstream
* write VBR tag into the bitstream set mpeg1/2 flag estimate the number of
* frames (based on a lot of data)
*
* now we set more flags: nspsytune: see code VBR modes see code CBR/ABR see
* code
*
* Finally, we set the algorithm flags based on the gfp.quality value
* lame_init_qval(gfp);
*
********************************************************************/
this.lame_init_params = function (gfp) {
var gfc = gfp.internal_flags;
gfc.Class_ID = 0;
if (gfc.ATH == null)
gfc.ATH = new ATH_1();
if (gfc.PSY == null)
gfc.PSY = new PSY();
if (gfc.rgdata == null)
gfc.rgdata = new ReplayGain_1();
gfc.channels_in = gfp.num_channels;
if (gfc.channels_in == 1)
gfp.mode = MPEGMode.MONO;
gfc.channels_out = (gfp.mode == MPEGMode.MONO) ? 1 : 2;
gfc.mode_ext = Encoder_1.MPG_MD_MS_LR;
if (gfp.mode == MPEGMode.MONO)
gfp.force_ms = false;
/*
* don't allow forced mid/side stereo for mono output
*/
if (gfp.VBR == VbrMode$2.vbr_off && gfp.VBR_mean_bitrate_kbps != 128
&& gfp.brate == 0)
gfp.brate = gfp.VBR_mean_bitrate_kbps;
if (gfp.VBR == VbrMode$2.vbr_off || gfp.VBR == VbrMode$2.vbr_mtrh
|| gfp.VBR == VbrMode$2.vbr_mt) {
/* these modes can handle free format condition */
} else {
gfp.free_format = false;
/* mode can't be mixed with free format */
}
if (gfp.VBR == VbrMode$2.vbr_off && gfp.brate == 0) {
/* no bitrate or compression ratio specified, use 11.025 */
if (BitStream_1.EQ(gfp.compression_ratio, 0))
gfp.compression_ratio = 11.025;
/*
* rate to compress a CD down to exactly 128000 bps
*/
}
/* find bitrate if user specify a compression ratio */
if (gfp.VBR == VbrMode$2.vbr_off && gfp.compression_ratio > 0) {
if (gfp.out_samplerate == 0)
gfp.out_samplerate = map2MP3Frequency((int)(0.97 * gfp.in_samplerate));
/*
* round up with a margin of 3 %
*/
/*
* choose a bitrate for the output samplerate which achieves
* specified compression ratio
*/
gfp.brate = 0 | (gfp.out_samplerate * 16 * gfc.channels_out / (1.e3 * gfp.compression_ratio));
/* we need the version for the bitrate table look up */
gfc.samplerate_index = SmpFrqIndex(gfp.out_samplerate, gfp);
if (!gfp.free_format) /*
* for non Free Format find the nearest allowed
* bitrate
*/
gfp.brate = FindNearestBitrate(gfp.brate, gfp.version,
gfp.out_samplerate);
}
if (gfp.out_samplerate != 0) {
if (gfp.out_samplerate < 16000) {
gfp.VBR_mean_bitrate_kbps = Math.max(gfp.VBR_mean_bitrate_kbps,
8);
gfp.VBR_mean_bitrate_kbps = Math.min(gfp.VBR_mean_bitrate_kbps,
64);
} else if (gfp.out_samplerate < 32000) {
gfp.VBR_mean_bitrate_kbps = Math.max(gfp.VBR_mean_bitrate_kbps,
8);
gfp.VBR_mean_bitrate_kbps = Math.min(gfp.VBR_mean_bitrate_kbps,
160);
} else {
gfp.VBR_mean_bitrate_kbps = Math.max(gfp.VBR_mean_bitrate_kbps,
32);
gfp.VBR_mean_bitrate_kbps = Math.min(gfp.VBR_mean_bitrate_kbps,
320);
}
}
/****************************************************************/
/* if a filter has not been enabled, see if we should add one: */
/****************************************************************/
if (gfp.lowpassfreq == 0) {
var lowpass = 16000.;
switch (gfp.VBR) {
case VbrMode$2.vbr_off:
{
var lh = new LowPassHighPass();
optimum_bandwidth(lh, gfp.brate);
lowpass = lh.lowerlimit;
break;
}
case VbrMode$2.vbr_abr:
{
var lh = new LowPassHighPass();
optimum_bandwidth(lh, gfp.VBR_mean_bitrate_kbps);
lowpass = lh.lowerlimit;
break;
}
case VbrMode$2.vbr_rh:
{
var x = [19500, 19000, 18600, 18000, 17500, 16000,
15600, 14900, 12500, 10000, 3950];
if (0 <= gfp.VBR_q && gfp.VBR_q <= 9) {
var a = x[gfp.VBR_q], b = x[gfp.VBR_q + 1], m = gfp.VBR_q_frac;
lowpass = linear_int(a, b, m);
} else {
lowpass = 19500;
}
break;
}
default:
{
var x = [19500, 19000, 18500, 18000, 17500, 16500,
15500, 14500, 12500, 9500, 3950];
if (0 <= gfp.VBR_q && gfp.VBR_q <= 9) {
var a = x[gfp.VBR_q], b = x[gfp.VBR_q + 1], m = gfp.VBR_q_frac;
lowpass = linear_int(a, b, m);
} else {
lowpass = 19500;
}
}
}
if (gfp.mode == MPEGMode.MONO
&& (gfp.VBR == VbrMode$2.vbr_off || gfp.VBR == VbrMode$2.vbr_abr))
lowpass *= 1.5;
gfp.lowpassfreq = lowpass | 0;
}
if (gfp.out_samplerate == 0) {
if (2 * gfp.lowpassfreq > gfp.in_samplerate) {
gfp.lowpassfreq = gfp.in_samplerate / 2;
}
gfp.out_samplerate = optimum_samplefreq(gfp.lowpassfreq | 0,
gfp.in_samplerate);
}
gfp.lowpassfreq = Math.min(20500, gfp.lowpassfreq);
gfp.lowpassfreq = Math.min(gfp.out_samplerate / 2, gfp.lowpassfreq);
if (gfp.VBR == VbrMode$2.vbr_off) {
gfp.compression_ratio = gfp.out_samplerate * 16 * gfc.channels_out
/ (1.e3 * gfp.brate);
}
if (gfp.VBR == VbrMode$2.vbr_abr) {
gfp.compression_ratio = gfp.out_samplerate * 16 * gfc.channels_out
/ (1.e3 * gfp.VBR_mean_bitrate_kbps);
}
/*
* do not compute ReplayGain values and do not find the peak sample if
* we can't store them
*/
if (!gfp.bWriteVbrTag) {
gfp.findReplayGain = false;
gfp.decode_on_the_fly = false;
gfc.findPeakSample = false;
}
gfc.findReplayGain = gfp.findReplayGain;
gfc.decode_on_the_fly = gfp.decode_on_the_fly;
if (gfc.decode_on_the_fly)
gfc.findPeakSample = true;
if (gfc.findReplayGain) {
if (ga.InitGainAnalysis(gfc.rgdata, gfp.out_samplerate) == GainAnalysis.INIT_GAIN_ANALYSIS_ERROR) {
gfp.internal_flags = null;
return -6;
}
}
if (gfc.decode_on_the_fly && !gfp.decode_only) {
if (gfc.hip != null) {
mpglib.hip_decode_exit(gfc.hip);
}
gfc.hip = mpglib.hip_decode_init();
}
gfc.mode_gr = gfp.out_samplerate <= 24000 ? 1 : 2;
/*
* Number of granules per frame
*/
gfp.framesize = 576 * gfc.mode_gr;
gfp.encoder_delay = Encoder_1.ENCDELAY;
gfc.resample_ratio = gfp.in_samplerate / gfp.out_samplerate;
/**
*
* sample freq bitrate compression ratio
* [kHz] [kbps/channel] for 16 bit input
* 44.1 56 12.6
* 44.1 64 11.025
* 44.1 80 8.82
* 22.05 24 14.7
* 22.05 32 11.025
* 22.05 40 8.82
* 16 16 16.0
* 16 24 10.667
*
*/
/**
*
* For VBR, take a guess at the compression_ratio.
* For example:
*
* VBR_q compression like
* - 4.4 320 kbps/44 kHz
* 0...1 5.5 256 kbps/44 kHz
* 2 7.3 192 kbps/44 kHz
* 4 8.8 160 kbps/44 kHz
* 6 11 128 kbps/44 kHz
* 9 14.7 96 kbps
*
* for lower bitrates, downsample with --resample
*
*/
switch (gfp.VBR) {
case VbrMode$2.vbr_mt:
case VbrMode$2.vbr_rh:
case VbrMode$2.vbr_mtrh:
{
/* numbers are a bit strange, but they determine the lowpass value */
var cmp = [5.7, 6.5, 7.3, 8.2, 10, 11.9, 13, 14,
15, 16.5];
gfp.compression_ratio = cmp[gfp.VBR_q];
}
break;
case VbrMode$2.vbr_abr:
gfp.compression_ratio = gfp.out_samplerate * 16 * gfc.channels_out
/ (1.e3 * gfp.VBR_mean_bitrate_kbps);
break;
default:
gfp.compression_ratio = gfp.out_samplerate * 16 * gfc.channels_out
/ (1.e3 * gfp.brate);
break;
}
/*
* mode = -1 (not set by user) or mode = MONO (because of only 1 input
* channel). If mode has not been set, then select J-STEREO
*/
if (gfp.mode == MPEGMode.NOT_SET) {
gfp.mode = MPEGMode.JOINT_STEREO;
}
/* apply user driven high pass filter */
if (gfp.highpassfreq > 0) {
gfc.highpass1 = 2. * gfp.highpassfreq;
if (gfp.highpasswidth >= 0)
gfc.highpass2 = 2. * (gfp.highpassfreq + gfp.highpasswidth);
else
/* 0% above on default */
gfc.highpass2 = (1 + 0.00) * 2. * gfp.highpassfreq;
gfc.highpass1 /= gfp.out_samplerate;
gfc.highpass2 /= gfp.out_samplerate;
} else {
gfc.highpass1 = 0;
gfc.highpass2 = 0;
}
/* apply user driven low pass filter */
if (gfp.lowpassfreq > 0) {
gfc.lowpass2 = 2. * gfp.lowpassfreq;
if (gfp.lowpasswidth >= 0) {
gfc.lowpass1 = 2. * (gfp.lowpassfreq - gfp.lowpasswidth);
if (gfc.lowpass1 < 0) /* has to be >= 0 */
gfc.lowpass1 = 0;
} else { /* 0% below on default */
gfc.lowpass1 = (1 - 0.00) * 2. * gfp.lowpassfreq;
}
gfc.lowpass1 /= gfp.out_samplerate;
gfc.lowpass2 /= gfp.out_samplerate;
} else {
gfc.lowpass1 = 0;
gfc.lowpass2 = 0;
}
/**********************************************************************/
/* compute info needed for polyphase filter (filter type==0, default) */
/**********************************************************************/
lame_init_params_ppflt(gfp);
/*******************************************************
* samplerate and bitrate index
*******************************************************/
gfc.samplerate_index = SmpFrqIndex(gfp.out_samplerate, gfp);
if (gfc.samplerate_index < 0) {
gfp.internal_flags = null;
return -1;
}
if (gfp.VBR == VbrMode$2.vbr_off) {
if (gfp.free_format) {
gfc.bitrate_index = 0;
} else {
gfp.brate = FindNearestBitrate(gfp.brate, gfp.version,
gfp.out_samplerate);
gfc.bitrate_index = BitrateIndex(gfp.brate, gfp.version,
gfp.out_samplerate);
if (gfc.bitrate_index <= 0) {
gfp.internal_flags = null;
return -1;
}
}
} else {
gfc.bitrate_index = 1;
}
/* for CBR, we will write an "info" tag. */
if (gfp.analysis)
gfp.bWriteVbrTag = false;
/* some file options not allowed if output is: not specified or stdout */
if (gfc.pinfo != null)
gfp.bWriteVbrTag = false;
/* disable Xing VBR tag */
bs.init_bit_stream_w(gfc);
var j = gfc.samplerate_index + (3 * gfp.version) + 6
* (gfp.out_samplerate < 16000 ? 1 : 0);
for (var i = 0; i < Encoder_1.SBMAX_l + 1; i++)
gfc.scalefac_band.l[i] = qupvt.sfBandIndex[j].l[i];
for (var i = 0; i < Encoder_1.PSFB21 + 1; i++) {
var size = (gfc.scalefac_band.l[22] - gfc.scalefac_band.l[21])
/ Encoder_1.PSFB21;
var start = gfc.scalefac_band.l[21] + i * size;
gfc.scalefac_band.psfb21[i] = start;
}
gfc.scalefac_band.psfb21[Encoder_1.PSFB21] = 576;
for (var i = 0; i < Encoder_1.SBMAX_s + 1; i++)
gfc.scalefac_band.s[i] = qupvt.sfBandIndex[j].s[i];
for (var i = 0; i < Encoder_1.PSFB12 + 1; i++) {
var size = (gfc.scalefac_band.s[13] - gfc.scalefac_band.s[12])
/ Encoder_1.PSFB12;
var start = gfc.scalefac_band.s[12] + i * size;
gfc.scalefac_band.psfb12[i] = start;
}
gfc.scalefac_band.psfb12[Encoder_1.PSFB12] = 192;
/* determine the mean bitrate for main data */
if (gfp.version == 1) /* MPEG 1 */
gfc.sideinfo_len = (gfc.channels_out == 1) ? 4 + 17 : 4 + 32;
else
/* MPEG 2 */
gfc.sideinfo_len = (gfc.channels_out == 1) ? 4 + 9 : 4 + 17;
if (gfp.error_protection)
gfc.sideinfo_len += 2;
lame_init_bitstream(gfp);
gfc.Class_ID = LAME_ID;
{
var k;
for (k = 0; k < 19; k++)
gfc.nsPsy.pefirbuf[k] = 700 * gfc.mode_gr * gfc.channels_out;
if (gfp.ATHtype == -1)
gfp.ATHtype = 4;
}
assert$2(gfp.VBR_q <= 9);
assert$2(gfp.VBR_q >= 0);
switch (gfp.VBR) {
case VbrMode$2.vbr_mt:
gfp.VBR = VbrMode$2.vbr_mtrh;
//$FALL-THROUGH$
case VbrMode$2.vbr_mtrh:
{
if (gfp.useTemporal == null) {
gfp.useTemporal = false;
/* off by default for this VBR mode */
}
p.apply_preset(gfp, 500 - (gfp.VBR_q * 10), 0);
/**
*
* The newer VBR code supports only a limited
* subset of quality levels:
* 9-5=5 are the same, uses x^3/4 quantization
* 4-0=0 are the same 5 plus best huffman divide code
*
*/
if (gfp.quality < 0)
gfp.quality = LAME_DEFAULT_QUALITY;
if (gfp.quality < 5)
gfp.quality = 0;
if (gfp.quality > 5)
gfp.quality = 5;
gfc.PSY.mask_adjust = gfp.maskingadjust;
gfc.PSY.mask_adjust_short = gfp.maskingadjust_short;
/*
* sfb21 extra only with MPEG-1 at higher sampling rates
*/
if (gfp.experimentalY)
gfc.sfb21_extra = false;
else
gfc.sfb21_extra = (gfp.out_samplerate > 44000);
gfc.iteration_loop = new VBRNewIterationLoop(qu);
break;
}
case VbrMode$2.vbr_rh:
{
p.apply_preset(gfp, 500 - (gfp.VBR_q * 10), 0);
gfc.PSY.mask_adjust = gfp.maskingadjust;
gfc.PSY.mask_adjust_short = gfp.maskingadjust_short;
/*
* sfb21 extra only with MPEG-1 at higher sampling rates
*/
if (gfp.experimentalY)
gfc.sfb21_extra = false;
else
gfc.sfb21_extra = (gfp.out_samplerate > 44000);
/*
* VBR needs at least the output of GPSYCHO, so we have to garantee
* that by setting a minimum quality level, actually level 6 does
* it. down to level 6
*/
if (gfp.quality > 6)
gfp.quality = 6;
if (gfp.quality < 0)
gfp.quality = LAME_DEFAULT_QUALITY;
gfc.iteration_loop = new VBROldIterationLoop(qu);
break;
}
default: /* cbr/abr */
{
var vbrmode;
/*
* no sfb21 extra with CBR code
*/
gfc.sfb21_extra = false;
if (gfp.quality < 0)
gfp.quality = LAME_DEFAULT_QUALITY;
vbrmode = gfp.VBR;
if (vbrmode == VbrMode$2.vbr_off)
gfp.VBR_mean_bitrate_kbps = gfp.brate;
/* second, set parameters depending on bitrate */
p.apply_preset(gfp, gfp.VBR_mean_bitrate_kbps, 0);
gfp.VBR = vbrmode;
gfc.PSY.mask_adjust = gfp.maskingadjust;
gfc.PSY.mask_adjust_short = gfp.maskingadjust_short;
if (vbrmode == VbrMode$2.vbr_off) {
gfc.iteration_loop = new CBRNewIterationLoop_1(qu);
} else {
gfc.iteration_loop = new ABRIterationLoop(qu);
}
break;
}
}
assert$2(gfp.scale >= 0);
/* initialize default values common for all modes */
if (gfp.VBR != VbrMode$2.vbr_off) { /* choose a min/max bitrate for VBR */
/* if the user didn't specify VBR_max_bitrate: */
gfc.VBR_min_bitrate = 1;
/*
* default: allow 8 kbps (MPEG-2) or 32 kbps (MPEG-1)
*/
gfc.VBR_max_bitrate = 14;
/*
* default: allow 160 kbps (MPEG-2) or 320 kbps (MPEG-1)
*/
if (gfp.out_samplerate < 16000)
gfc.VBR_max_bitrate = 8;
/* default: allow 64 kbps (MPEG-2.5) */
if (gfp.VBR_min_bitrate_kbps != 0) {
gfp.VBR_min_bitrate_kbps = FindNearestBitrate(
gfp.VBR_min_bitrate_kbps, gfp.version,
gfp.out_samplerate);
gfc.VBR_min_bitrate = BitrateIndex(gfp.VBR_min_bitrate_kbps,
gfp.version, gfp.out_samplerate);
if (gfc.VBR_min_bitrate < 0)
return -1;
}
if (gfp.VBR_max_bitrate_kbps != 0) {
gfp.VBR_max_bitrate_kbps = FindNearestBitrate(
gfp.VBR_max_bitrate_kbps, gfp.version,
gfp.out_samplerate);
gfc.VBR_max_bitrate = BitrateIndex(gfp.VBR_max_bitrate_kbps,
gfp.version, gfp.out_samplerate);
if (gfc.VBR_max_bitrate < 0)
return -1;
}
gfp.VBR_min_bitrate_kbps = Tables_1.bitrate_table[gfp.version][gfc.VBR_min_bitrate];
gfp.VBR_max_bitrate_kbps = Tables_1.bitrate_table[gfp.version][gfc.VBR_max_bitrate];
gfp.VBR_mean_bitrate_kbps = Math.min(
Tables_1.bitrate_table[gfp.version][gfc.VBR_max_bitrate],
gfp.VBR_mean_bitrate_kbps);
gfp.VBR_mean_bitrate_kbps = Math.max(
Tables_1.bitrate_table[gfp.version][gfc.VBR_min_bitrate],
gfp.VBR_mean_bitrate_kbps);
}
/* just another daily changing developer switch */
if (gfp.tune) {
gfc.PSY.mask_adjust += gfp.tune_value_a;
gfc.PSY.mask_adjust_short += gfp.tune_value_a;
}
/* initialize internal qval settings */
lame_init_qval(gfp);
assert$2(gfp.scale >= 0);
/*
* automatic ATH adjustment on
*/
if (gfp.athaa_type < 0)
gfc.ATH.useAdjust = 3;
else
gfc.ATH.useAdjust = gfp.athaa_type;
/* initialize internal adaptive ATH settings -jd */
gfc.ATH.aaSensitivityP = Math.pow(10.0, gfp.athaa_sensitivity
/ -10.0);
if (gfp.short_blocks == null) {
gfp.short_blocks = ShortBlock$2.short_block_allowed;
}
/*
* Note Jan/2003: Many hardware decoders cannot handle short blocks in
* regular stereo mode unless they are coupled (same type in both
* channels) it is a rare event (1 frame per min. or so) that LAME would
* use uncoupled short blocks, so lets turn them off until we decide how
* to handle this. No other encoders allow uncoupled short blocks, even
* though it is in the standard.
*/
/*
* rh 20040217: coupling makes no sense for mono and dual-mono streams
*/
if (gfp.short_blocks == ShortBlock$2.short_block_allowed
&& (gfp.mode == MPEGMode.JOINT_STEREO || gfp.mode == MPEGMode.STEREO)) {
gfp.short_blocks = ShortBlock$2.short_block_coupled;
}
if (gfp.quant_comp < 0)
gfp.quant_comp = 1;
if (gfp.quant_comp_short < 0)
gfp.quant_comp_short = 0;
if (gfp.msfix < 0)
gfp.msfix = 0;
/* select psychoacoustic model */
gfp.exp_nspsytune = gfp.exp_nspsytune | 1;
if (gfp.internal_flags.nsPsy.attackthre < 0)
gfp.internal_flags.nsPsy.attackthre = PsyModel_1.NSATTACKTHRE;
if (gfp.internal_flags.nsPsy.attackthre_s < 0)
gfp.internal_flags.nsPsy.attackthre_s = PsyModel_1.NSATTACKTHRE_S;
assert$2(gfp.scale >= 0);
if (gfp.scale < 0)
gfp.scale = 1;
if (gfp.ATHtype < 0)
gfp.ATHtype = 4;
if (gfp.ATHcurve < 0)
gfp.ATHcurve = 4;
if (gfp.athaa_loudapprox < 0)
gfp.athaa_loudapprox = 2;
if (gfp.interChRatio < 0)
gfp.interChRatio = 0;
if (gfp.useTemporal == null)
gfp.useTemporal = true;
/* on by default */
/*
* padding method as described in
* "MPEG-Layer3 / Bitstream Syntax and Decoding" by Martin Sieler, Ralph
* Sperschneider
*
* note: there is no padding for the very first frame
*
* Robert Hegemann 2000-06-22
*/
gfc.slot_lag = gfc.frac_SpF = 0;
if (gfp.VBR == VbrMode$2.vbr_off)
gfc.slot_lag = gfc.frac_SpF = (((gfp.version + 1) * 72000 * gfp.brate) % gfp.out_samplerate) | 0;
qupvt.iteration_init(gfp);
psy.psymodel_init(gfp);
assert$2(gfp.scale >= 0);
return 0;
};
function update_inbuffer_size(gfc, nsamples) {
if (gfc.in_buffer_0 == null || gfc.in_buffer_nsamples < nsamples) {
gfc.in_buffer_0 = new_float$2(nsamples);
gfc.in_buffer_1 = new_float$2(nsamples);
gfc.in_buffer_nsamples = nsamples;
}
}
this.lame_encode_flush = function (gfp, mp3buffer, mp3bufferPos, mp3buffer_size) {
var gfc = gfp.internal_flags;
var buffer = new_short_n$1([2, 1152]);
var imp3 = 0, mp3count, mp3buffer_size_remaining;
/*
* we always add POSTDELAY=288 padding to make sure granule with real
* data can be complety decoded (because of 50% overlap with next
* granule
*/
var end_padding;
var frames_left;
var samples_to_encode = gfc.mf_samples_to_encode - Encoder_1.POSTDELAY;
var mf_needed = calcNeeded(gfp);
/* Was flush already called? */
if (gfc.mf_samples_to_encode < 1) {
return 0;
}
mp3count = 0;
if (gfp.in_samplerate != gfp.out_samplerate) {
/*
* delay due to resampling; needs to be fixed, if resampling code
* gets changed
*/
samples_to_encode += 16. * gfp.out_samplerate / gfp.in_samplerate;
}
end_padding = gfp.framesize - (samples_to_encode % gfp.framesize);
if (end_padding < 576)
end_padding += gfp.framesize;
gfp.encoder_padding = end_padding;
frames_left = (samples_to_encode + end_padding) / gfp.framesize;
/*
* send in a frame of 0 padding until all internal sample buffers are
* flushed
*/
while (frames_left > 0 && imp3 >= 0) {
var bunch = mf_needed - gfc.mf_size;
var frame_num = gfp.frameNum;
bunch *= gfp.in_samplerate;
bunch /= gfp.out_samplerate;
if (bunch > 1152)
bunch = 1152;
if (bunch < 1)
bunch = 1;
mp3buffer_size_remaining = mp3buffer_size - mp3count;
/* if user specifed buffer size = 0, dont check size */
if (mp3buffer_size == 0)
mp3buffer_size_remaining = 0;
imp3 = this.lame_encode_buffer(gfp, buffer[0], buffer[1], bunch,
mp3buffer, mp3bufferPos, mp3buffer_size_remaining);
mp3bufferPos += imp3;
mp3count += imp3;
frames_left -= (frame_num != gfp.frameNum) ? 1 : 0;
}
/*
* Set gfc.mf_samples_to_encode to 0, so we may detect and break loops
* calling it more than once in a row.
*/
gfc.mf_samples_to_encode = 0;
if (imp3 < 0) {
/* some type of fatal error */
return imp3;
}
mp3buffer_size_remaining = mp3buffer_size - mp3count;
/* if user specifed buffer size = 0, dont check size */
if (mp3buffer_size == 0)
mp3buffer_size_remaining = 0;
/* mp3 related stuff. bit buffer might still contain some mp3 data */
bs.flush_bitstream(gfp);
imp3 = bs.copy_buffer(gfc, mp3buffer, mp3bufferPos,
mp3buffer_size_remaining, 1);
if (imp3 < 0) {
/* some type of fatal error */
return imp3;
}
mp3bufferPos += imp3;
mp3count += imp3;
mp3buffer_size_remaining = mp3buffer_size - mp3count;
/* if user specifed buffer size = 0, dont check size */
if (mp3buffer_size == 0)
mp3buffer_size_remaining = 0;
if (gfp.write_id3tag_automatic) {
/* write a id3 tag to the bitstream */
id3.id3tag_write_v1(gfp);
imp3 = bs.copy_buffer(gfc, mp3buffer, mp3bufferPos,
mp3buffer_size_remaining, 0);
if (imp3 < 0) {
return imp3;
}
mp3count += imp3;
}
return mp3count;
};
this.lame_encode_buffer = function (gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3bufPos, mp3buf_size) {
var gfc = gfp.internal_flags;
var in_buffer = [null, null];
if (gfc.Class_ID != LAME_ID)
return -3;
if (nsamples == 0)
return 0;
update_inbuffer_size(gfc, nsamples);
in_buffer[0] = gfc.in_buffer_0;
in_buffer[1] = gfc.in_buffer_1;
/* make a copy of input buffer, changing type to sample_t */
for (var i = 0; i < nsamples; i++) {
in_buffer[0][i] = buffer_l[i];
if (gfc.channels_in > 1)
in_buffer[1][i] = buffer_r[i];
}
return lame_encode_buffer_sample(gfp, in_buffer[0], in_buffer[1],
nsamples, mp3buf, mp3bufPos, mp3buf_size);
};
function calcNeeded(gfp) {
var mf_needed = Encoder_1.BLKSIZE + gfp.framesize - Encoder_1.FFTOFFSET;
/*
* amount needed for FFT
*/
mf_needed = Math.max(mf_needed, 512 + gfp.framesize - 32);
assert$2(LameInternalFlags_1.MFSIZE >= mf_needed);
return mf_needed;
}
function lame_encode_buffer_sample(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3bufPos, mp3buf_size) {
var gfc = gfp.internal_flags;
var mp3size = 0, ret, i, ch, mf_needed;
var mp3out;
var mfbuf = [null, null];
var in_buffer = [null, null];
if (gfc.Class_ID != LAME_ID)
return -3;
if (nsamples == 0)
return 0;
/* copy out any tags that may have been written into bitstream */
mp3out = bs.copy_buffer(gfc, mp3buf, mp3bufPos, mp3buf_size, 0);
if (mp3out < 0)
return mp3out;
/* not enough buffer space */
mp3bufPos += mp3out;
mp3size += mp3out;
in_buffer[0] = buffer_l;
in_buffer[1] = buffer_r;
/* Apply user defined re-scaling */
/* user selected scaling of the samples */
if (BitStream_1.NEQ(gfp.scale, 0) && BitStream_1.NEQ(gfp.scale, 1.0)) {
for (i = 0; i < nsamples; ++i) {
in_buffer[0][i] *= gfp.scale;
if (gfc.channels_out == 2)
in_buffer[1][i] *= gfp.scale;
}
}
/* user selected scaling of the channel 0 (left) samples */
if (BitStream_1.NEQ(gfp.scale_left, 0)
&& BitStream_1.NEQ(gfp.scale_left, 1.0)) {
for (i = 0; i < nsamples; ++i) {
in_buffer[0][i] *= gfp.scale_left;
}
}
/* user selected scaling of the channel 1 (right) samples */
if (BitStream_1.NEQ(gfp.scale_right, 0)
&& BitStream_1.NEQ(gfp.scale_right, 1.0)) {
for (i = 0; i < nsamples; ++i) {
in_buffer[1][i] *= gfp.scale_right;
}
}
/* Downsample to Mono if 2 channels in and 1 channel out */
if (gfp.num_channels == 2 && gfc.channels_out == 1) {
for (i = 0; i < nsamples; ++i) {
in_buffer[0][i] = 0.5 * ( in_buffer[0][i] + in_buffer[1][i]);
in_buffer[1][i] = 0.0;
}
}
mf_needed = calcNeeded(gfp);
mfbuf[0] = gfc.mfbuf[0];
mfbuf[1] = gfc.mfbuf[1];
var in_bufferPos = 0;
while (nsamples > 0) {
var in_buffer_ptr = [null, null];
var n_in = 0;
/* number of input samples processed with fill_buffer */
var n_out = 0;
/* number of samples output with fill_buffer */
/* n_in <> n_out if we are resampling */
in_buffer_ptr[0] = in_buffer[0];
in_buffer_ptr[1] = in_buffer[1];
/* copy in new samples into mfbuf, with resampling */
var inOut = new InOut();
fill_buffer(gfp, mfbuf, in_buffer_ptr, in_bufferPos, nsamples,
inOut);
n_in = inOut.n_in;
n_out = inOut.n_out;
/* compute ReplayGain of resampled input if requested */
if (gfc.findReplayGain && !gfc.decode_on_the_fly)
if (ga.AnalyzeSamples(gfc.rgdata, mfbuf[0], gfc.mf_size,
mfbuf[1], gfc.mf_size, n_out, gfc.channels_out) == GainAnalysis.GAIN_ANALYSIS_ERROR)
return -6;
/* update in_buffer counters */
nsamples -= n_in;
in_bufferPos += n_in;
if (gfc.channels_out == 2)
;// in_bufferPos += n_in;
/* update mfbuf[] counters */
gfc.mf_size += n_out;
assert$2(gfc.mf_size <= LameInternalFlags_1.MFSIZE);
/*
* lame_encode_flush may have set gfc.mf_sample_to_encode to 0 so we
* have to reinitialize it here when that happened.
*/
if (gfc.mf_samples_to_encode < 1) {
gfc.mf_samples_to_encode = Encoder_1.ENCDELAY + Encoder_1.POSTDELAY;
}
gfc.mf_samples_to_encode += n_out;
if (gfc.mf_size >= mf_needed) {
/* encode the frame. */
/* mp3buf = pointer to current location in buffer */
/* mp3buf_size = size of original mp3 output buffer */
/* = 0 if we should not worry about the */
/* buffer size because calling program is */
/* to lazy to compute it */
/* mp3size = size of data written to buffer so far */
/* mp3buf_size-mp3size = amount of space avalable */
var buf_size = mp3buf_size - mp3size;
if (mp3buf_size == 0)
buf_size = 0;
ret = lame_encode_frame(gfp, mfbuf[0], mfbuf[1], mp3buf,
mp3bufPos, buf_size);
if (ret < 0)
return ret;
mp3bufPos += ret;
mp3size += ret;
/* shift out old samples */
gfc.mf_size -= gfp.framesize;
gfc.mf_samples_to_encode -= gfp.framesize;
for (ch = 0; ch < gfc.channels_out; ch++)
for (i = 0; i < gfc.mf_size; i++)
mfbuf[ch][i] = mfbuf[ch][i + gfp.framesize];
}
}
assert$2(nsamples == 0);
return mp3size;
}
function lame_encode_frame(gfp, inbuf_l, inbuf_r, mp3buf, mp3bufPos, mp3buf_size) {
var ret = self.enc.lame_encode_mp3_frame(gfp, inbuf_l, inbuf_r, mp3buf,
mp3bufPos, mp3buf_size);
gfp.frameNum++;
return ret;
}
function InOut() {
this.n_in = 0;
this.n_out = 0;
}
function NumUsed() {
this.num_used = 0;
}
/**
* Greatest common divisor.
* * Joint work of Euclid and M. Hendry */ function gcd(i, j) { return j != 0 ? gcd(j, i % j) : i; } /** * Resampling via FIR filter, blackman window. */ function blackman(x, fcn, l) { /* * This algorithm from: SIGNAL PROCESSING ALGORITHMS IN FORTRAN AND C * S.D. Stearns and R.A. David, Prentice-Hall, 1992 */ var wcn = (Math.PI * fcn); x /= l; if (x < 0) x = 0; if (x > 1) x = 1; var x2 = x - .5; var bkwn = 0.42 - 0.5 * Math.cos(2 * x * Math.PI) + 0.08 * Math.cos(4 * x * Math.PI); if (Math.abs(x2) < 1e-9) return (wcn / Math.PI); else return (bkwn * Math.sin(l * wcn * x2) / (Math.PI * l * x2)); } function fill_buffer_resample(gfp, outbuf, outbufPos, desired_len, inbuf, in_bufferPos, len, num_used, ch) { var gfc = gfp.internal_flags; var i, j = 0, k; /* number of convolution functions to pre-compute */ var bpc = gfp.out_samplerate / gcd(gfp.out_samplerate, gfp.in_samplerate); if (bpc > LameInternalFlags_1.BPC) bpc = LameInternalFlags_1.BPC; var intratio = (Math.abs(gfc.resample_ratio - Math.floor(.5 + gfc.resample_ratio)) < .0001) ? 1 : 0; var fcn = 1.00 / gfc.resample_ratio; if (fcn > 1.00) fcn = 1.00; var filter_l = 31; if (0 == filter_l % 2) --filter_l; /* must be odd */ filter_l += intratio; /* unless resample_ratio=int, it must be even */ var BLACKSIZE = filter_l + 1; /* size of data needed for FIR */ if (gfc.fill_buffer_resample_init == 0) { gfc.inbuf_old[0] = new_float$2(BLACKSIZE); gfc.inbuf_old[1] = new_float$2(BLACKSIZE); for (i = 0; i <= 2 * bpc; ++i) gfc.blackfilt[i] = new_float$2(BLACKSIZE); gfc.itime[0] = 0; gfc.itime[1] = 0; /* precompute blackman filter coefficients */ for (j = 0; j <= 2 * bpc; j++) { var sum = 0.; var offset = (j - bpc) / (2. * bpc); for (i = 0; i <= filter_l; i++) sum += gfc.blackfilt[j][i] = blackman(i - offset, fcn, filter_l); for (i = 0; i <= filter_l; i++) gfc.blackfilt[j][i] /= sum; } gfc.fill_buffer_resample_init = 1; } var inbuf_old = gfc.inbuf_old[ch]; /* time of j'th element in inbuf = itime + j/ifreq; */ /* time of k'th element in outbuf = j/ofreq */ for (k = 0; k < desired_len; k++) { var time0; var joff; time0 = k * gfc.resample_ratio; /* time of k'th output sample */ j = 0 | Math.floor(time0 - gfc.itime[ch]); /* check if we need more input data */ if ((filter_l + j - filter_l / 2) >= len) break; /* blackman filter. by default, window centered at j+.5(filter_l%2) */ /* but we want a window centered at time0. */ var offset = (time0 - gfc.itime[ch] - (j + .5 * (filter_l % 2))); assert$2(Math.abs(offset) <= .501); /* find the closest precomputed window for this offset: */ joff = 0 | Math.floor((offset * 2 * bpc) + bpc + .5); var xvalue = 0.; for (i = 0; i <= filter_l; ++i) { var j2 = i + j - filter_l / 2; var y; assert$2(j2 < len); assert$2(j2 + BLACKSIZE >= 0); y = (j2 < 0) ? inbuf_old[BLACKSIZE + j2] : inbuf[in_bufferPos + j2]; xvalue += y * gfc.blackfilt[joff][i]; } outbuf[outbufPos + k] = xvalue; } /* k = number of samples added to outbuf */ /* last k sample used data from [j-filter_l/2,j+filter_l-filter_l/2] */ /* how many samples of input data were used: */ num_used.num_used = Math.min(len, filter_l + j - filter_l / 2); /* * adjust our input time counter. Incriment by the number of samples * used, then normalize so that next output sample is at time 0, next * input buffer is at time itime[ch] */ gfc.itime[ch] += num_used.num_used - k * gfc.resample_ratio; /* save the last BLACKSIZE samples into the inbuf_old buffer */ if (num_used.num_used >= BLACKSIZE) { for (i = 0; i < BLACKSIZE; i++) inbuf_old[i] = inbuf[in_bufferPos + num_used.num_used + i - BLACKSIZE]; } else { /* shift in num_used.num_used samples into inbuf_old */ var n_shift = BLACKSIZE - num_used.num_used; /* * number of samples to * shift */ /* * shift n_shift samples by num_used.num_used, to make room for the * num_used new samples */ for (i = 0; i < n_shift; ++i) inbuf_old[i] = inbuf_old[i + num_used.num_used]; /* shift in the num_used.num_used samples */ for (j = 0; i < BLACKSIZE; ++i, ++j) inbuf_old[i] = inbuf[in_bufferPos + j]; assert$2(j == num_used.num_used); } return k; /* return the number samples created at the new samplerate */ } function fill_buffer(gfp, mfbuf, in_buffer, in_bufferPos, nsamples, io) { var gfc = gfp.internal_flags; /* copy in new samples into mfbuf, with resampling if necessary */ if ((gfc.resample_ratio < .9999) || (gfc.resample_ratio > 1.0001)) { for (var ch = 0; ch < gfc.channels_out; ch++) { var numUsed = new NumUsed(); io.n_out = fill_buffer_resample(gfp, mfbuf[ch], gfc.mf_size, gfp.framesize, in_buffer[ch], in_bufferPos, nsamples, numUsed, ch); io.n_in = numUsed.num_used; } } else { io.n_out = Math.min(gfp.framesize, nsamples); io.n_in = io.n_out; for (var i = 0; i < io.n_out; ++i) { mfbuf[0][gfc.mf_size + i] = in_buffer[0][in_bufferPos + i]; if (gfc.channels_out == 2) mfbuf[1][gfc.mf_size + i] = in_buffer[1][in_bufferPos + i]; } } } } var Lame_1 = Lame$1; var VbrMode$20 = common.VbrMode; function Presets$1() { function VBRPresets(qual, comp, compS, y, shThreshold, shThresholdS, adj, adjShort, lower, curve, sens, inter, joint, mod, fix) { this.vbr_q = qual; this.quant_comp = comp; this.quant_comp_s = compS; this.expY = y; this.st_lrm = shThreshold; this.st_s = shThresholdS; this.masking_adj = adj; this.masking_adj_short = adjShort; this.ath_lower = lower; this.ath_curve = curve; this.ath_sensitivity = sens; this.interch = inter; this.safejoint = joint; this.sfb21mod = mod; this.msfix = fix; } function ABRPresets(kbps, comp, compS, joint, fix, shThreshold, shThresholdS, bass, sc, mask, lower, curve, interCh, sfScale) { this.quant_comp = comp; this.quant_comp_s = compS; this.safejoint = joint; this.nsmsfix = fix; this.st_lrm = shThreshold; this.st_s = shThresholdS; this.nsbass = bass; this.scale = sc; this.masking_adj = mask; this.ath_lower = lower; this.ath_curve = curve; this.interch = interCh; this.sfscale = sfScale; } var lame; this.setModules = function (_lame) { lame = _lame; }; /** *
* Switch mappings for VBR mode VBR_RH
* vbr_q qcomp_l qcomp_s expY st_lrm st_s mask adj_l adj_s ath_lower ath_curve ath_sens interChR safejoint sfb21mod msfix
*
*/
var vbr_old_switch_map = [
new VBRPresets(0, 9, 9, 0, 5.20, 125.0, -4.2, -6.3, 4.8, 1, 0, 0, 2, 21, 0.97),
new VBRPresets(1, 9, 9, 0, 5.30, 125.0, -3.6, -5.6, 4.5, 1.5, 0, 0, 2, 21, 1.35),
new VBRPresets(2, 9, 9, 0, 5.60, 125.0, -2.2, -3.5, 2.8, 2, 0, 0, 2, 21, 1.49),
new VBRPresets(3, 9, 9, 1, 5.80, 130.0, -1.8, -2.8, 2.6, 3, -4, 0, 2, 20, 1.64),
new VBRPresets(4, 9, 9, 1, 6.00, 135.0, -0.7, -1.1, 1.1, 3.5, -8, 0, 2, 0, 1.79),
new VBRPresets(5, 9, 9, 1, 6.40, 140.0, 0.5, 0.4, -7.5, 4, -12, 0.0002, 0, 0, 1.95),
new VBRPresets(6, 9, 9, 1, 6.60, 145.0, 0.67, 0.65, -14.7, 6.5, -19, 0.0004, 0, 0, 2.30),
new VBRPresets(7, 9, 9, 1, 6.60, 145.0, 0.8, 0.75, -19.7, 8, -22, 0.0006, 0, 0, 2.70),
new VBRPresets(8, 9, 9, 1, 6.60, 145.0, 1.2, 1.15, -27.5, 10, -23, 0.0007, 0, 0, 0),
new VBRPresets(9, 9, 9, 1, 6.60, 145.0, 1.6, 1.6, -36, 11, -25, 0.0008, 0, 0, 0),
new VBRPresets(10, 9, 9, 1, 6.60, 145.0, 2.0, 2.0, -36, 12, -25, 0.0008, 0, 0, 0)
];
/**
*
* vbr_q qcomp_l qcomp_s expY st_lrm st_s mask adj_l adj_s ath_lower ath_curve ath_sens interChR safejoint sfb21mod msfix
*
*/
var vbr_psy_switch_map = [
new VBRPresets(0, 9, 9, 0, 4.20, 25.0, -7.0, -4.0, 7.5, 1, 0, 0, 2, 26, 0.97),
new VBRPresets(1, 9, 9, 0, 4.20, 25.0, -5.6, -3.6, 4.5, 1.5, 0, 0, 2, 21, 1.35),
new VBRPresets(2, 9, 9, 0, 4.20, 25.0, -4.4, -1.8, 2, 2, 0, 0, 2, 18, 1.49),
new VBRPresets(3, 9, 9, 1, 4.20, 25.0, -3.4, -1.25, 1.1, 3, -4, 0, 2, 15, 1.64),
new VBRPresets(4, 9, 9, 1, 4.20, 25.0, -2.2, 0.1, 0, 3.5, -8, 0, 2, 0, 1.79),
new VBRPresets(5, 9, 9, 1, 4.20, 25.0, -1.0, 1.65, -7.7, 4, -12, 0.0002, 0, 0, 1.95),
new VBRPresets(6, 9, 9, 1, 4.20, 25.0, -0.0, 2.47, -7.7, 6.5, -19, 0.0004, 0, 0, 2),
new VBRPresets(7, 9, 9, 1, 4.20, 25.0, 0.5, 2.0, -14.5, 8, -22, 0.0006, 0, 0, 2),
new VBRPresets(8, 9, 9, 1, 4.20, 25.0, 1.0, 2.4, -22.0, 10, -23, 0.0007, 0, 0, 2),
new VBRPresets(9, 9, 9, 1, 4.20, 25.0, 1.5, 2.95, -30.0, 11, -25, 0.0008, 0, 0, 2),
new VBRPresets(10, 9, 9, 1, 4.20, 25.0, 2.0, 2.95, -36.0, 12, -30, 0.0008, 0, 0, 2)
];
function apply_vbr_preset(gfp, a, enforce) {
var vbr_preset = gfp.VBR == VbrMode$20.vbr_rh ? vbr_old_switch_map
: vbr_psy_switch_map;
var x = gfp.VBR_q_frac;
var p = vbr_preset[a];
var q = vbr_preset[a + 1];
var set = p;
// NOOP(vbr_q);
// NOOP(quant_comp);
// NOOP(quant_comp_s);
// NOOP(expY);
p.st_lrm = p.st_lrm + x * (q.st_lrm - p.st_lrm);
// LERP(st_lrm);
p.st_s = p.st_s + x * (q.st_s - p.st_s);
// LERP(st_s);
p.masking_adj = p.masking_adj + x * (q.masking_adj - p.masking_adj);
// LERP(masking_adj);
p.masking_adj_short = p.masking_adj_short + x
* (q.masking_adj_short - p.masking_adj_short);
// LERP(masking_adj_short);
p.ath_lower = p.ath_lower + x * (q.ath_lower - p.ath_lower);
// LERP(ath_lower);
p.ath_curve = p.ath_curve + x * (q.ath_curve - p.ath_curve);
// LERP(ath_curve);
p.ath_sensitivity = p.ath_sensitivity + x
* (q.ath_sensitivity - p.ath_sensitivity);
// LERP(ath_sensitivity);
p.interch = p.interch + x * (q.interch - p.interch);
// LERP(interch);
// NOOP(safejoint);
// NOOP(sfb21mod);
p.msfix = p.msfix + x * (q.msfix - p.msfix);
// LERP(msfix);
lame_set_VBR_q(gfp, set.vbr_q);
if (enforce != 0)
gfp.quant_comp = set.quant_comp;
else if (!(Math.abs(gfp.quant_comp - -1) > 0))
gfp.quant_comp = set.quant_comp;
// SET_OPTION(quant_comp, set.quant_comp, -1);
if (enforce != 0)
gfp.quant_comp_short = set.quant_comp_s;
else if (!(Math.abs(gfp.quant_comp_short - -1) > 0))
gfp.quant_comp_short = set.quant_comp_s;
// SET_OPTION(quant_comp_short, set.quant_comp_s, -1);
if (set.expY != 0) {
gfp.experimentalY = set.expY != 0;
}
if (enforce != 0)
gfp.internal_flags.nsPsy.attackthre = set.st_lrm;
else if (!(Math.abs(gfp.internal_flags.nsPsy.attackthre - -1) > 0))
gfp.internal_flags.nsPsy.attackthre = set.st_lrm;
// SET_OPTION(short_threshold_lrm, set.st_lrm, -1);
if (enforce != 0)
gfp.internal_flags.nsPsy.attackthre_s = set.st_s;
else if (!(Math.abs(gfp.internal_flags.nsPsy.attackthre_s - -1) > 0))
gfp.internal_flags.nsPsy.attackthre_s = set.st_s;
// SET_OPTION(short_threshold_s, set.st_s, -1);
if (enforce != 0)
gfp.maskingadjust = set.masking_adj;
else if (!(Math.abs(gfp.maskingadjust - 0) > 0))
gfp.maskingadjust = set.masking_adj;
// SET_OPTION(maskingadjust, set.masking_adj, 0);
if (enforce != 0)
gfp.maskingadjust_short = set.masking_adj_short;
else if (!(Math.abs(gfp.maskingadjust_short - 0) > 0))
gfp.maskingadjust_short = set.masking_adj_short;
// SET_OPTION(maskingadjust_short, set.masking_adj_short, 0);
if (enforce != 0)
gfp.ATHlower = -set.ath_lower / 10.0;
else if (!(Math.abs((-gfp.ATHlower * 10.0) - 0) > 0))
gfp.ATHlower = -set.ath_lower / 10.0;
// SET_OPTION(ATHlower, set.ath_lower, 0);
if (enforce != 0)
gfp.ATHcurve = set.ath_curve;
else if (!(Math.abs(gfp.ATHcurve - -1) > 0))
gfp.ATHcurve = set.ath_curve;
// SET_OPTION(ATHcurve, set.ath_curve, -1);
if (enforce != 0)
gfp.athaa_sensitivity = set.ath_sensitivity;
else if (!(Math.abs(gfp.athaa_sensitivity - -1) > 0))
gfp.athaa_sensitivity = set.ath_sensitivity;
// SET_OPTION(athaa_sensitivity, set.ath_sensitivity, 0);
if (set.interch > 0) {
if (enforce != 0)
gfp.interChRatio = set.interch;
else if (!(Math.abs(gfp.interChRatio - -1) > 0))
gfp.interChRatio = set.interch;
// SET_OPTION(interChRatio, set.interch, -1);
}
/* parameters for which there is no proper set/get interface */
if (set.safejoint > 0) {
gfp.exp_nspsytune = gfp.exp_nspsytune | set.safejoint;
}
if (set.sfb21mod > 0) {
gfp.exp_nspsytune = gfp.exp_nspsytune | (set.sfb21mod << 20);
}
if (enforce != 0)
gfp.msfix = set.msfix;
else if (!(Math.abs(gfp.msfix - -1) > 0))
gfp.msfix = set.msfix;
// SET_OPTION(msfix, set.msfix, -1);
if (enforce == 0) {
gfp.VBR_q = a;
gfp.VBR_q_frac = x;
}
}
/**
*
* Switch mappings for ABR mode
*
* kbps quant q_s safejoint nsmsfix st_lrm st_s ns-bass scale msk ath_lwr ath_curve interch , sfscale
*
*/
var abr_switch_map = [
new ABRPresets(8, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -30.0, 11, 0.0012, 1), /* 8, impossible to use in stereo */
new ABRPresets(16, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -25.0, 11, 0.0010, 1), /* 16 */
new ABRPresets(24, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -20.0, 11, 0.0010, 1), /* 24 */
new ABRPresets(32, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -15.0, 11, 0.0010, 1), /* 32 */
new ABRPresets(40, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -10.0, 11, 0.0009, 1), /* 40 */
new ABRPresets(48, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -10.0, 11, 0.0009, 1), /* 48 */
new ABRPresets(56, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -6.0, 11, 0.0008, 1), /* 56 */
new ABRPresets(64, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, -2.0, 11, 0.0008, 1), /* 64 */
new ABRPresets(80, 9, 9, 0, 0, 6.60, 145, 0, 0.95, 0, .0, 8, 0.0007, 1), /* 80 */
new ABRPresets(96, 9, 9, 0, 2.50, 6.60, 145, 0, 0.95, 0, 1.0, 5.5, 0.0006, 1), /* 96 */
new ABRPresets(112, 9, 9, 0, 2.25, 6.60, 145, 0, 0.95, 0, 2.0, 4.5, 0.0005, 1), /* 112 */
new ABRPresets(128, 9, 9, 0, 1.95, 6.40, 140, 0, 0.95, 0, 3.0, 4, 0.0002, 1), /* 128 */
new ABRPresets(160, 9, 9, 1, 1.79, 6.00, 135, 0, 0.95, -2, 5.0, 3.5, 0, 1), /* 160 */
new ABRPresets(192, 9, 9, 1, 1.49, 5.60, 125, 0, 0.97, -4, 7.0, 3, 0, 0), /* 192 */
new ABRPresets(224, 9, 9, 1, 1.25, 5.20, 125, 0, 0.98, -6, 9.0, 2, 0, 0), /* 224 */
new ABRPresets(256, 9, 9, 1, 0.97, 5.20, 125, 0, 1.00, -8, 10.0, 1, 0, 0), /* 256 */
new ABRPresets(320, 9, 9, 1, 0.90, 5.20, 125, 0, 1.00, -10, 12.0, 0, 0, 0) /* 320 */
];
function apply_abr_preset(gfp, preset, enforce) {
/* Variables for the ABR stuff */
var actual_bitrate = preset;
var r = lame.nearestBitrateFullIndex(preset);
gfp.VBR = VbrMode$20.vbr_abr;
gfp.VBR_mean_bitrate_kbps = actual_bitrate;
gfp.VBR_mean_bitrate_kbps = Math.min(gfp.VBR_mean_bitrate_kbps, 320);
gfp.VBR_mean_bitrate_kbps = Math.max(gfp.VBR_mean_bitrate_kbps, 8);
gfp.brate = gfp.VBR_mean_bitrate_kbps;
if (gfp.VBR_mean_bitrate_kbps > 320) {
gfp.disable_reservoir = true;
}
/* parameters for which there is no proper set/get interface */
if (abr_switch_map[r].safejoint > 0)
gfp.exp_nspsytune = gfp.exp_nspsytune | 2;
/* safejoint */
if (abr_switch_map[r].sfscale > 0) {
gfp.internal_flags.noise_shaping = 2;
}
/* ns-bass tweaks */
if (Math.abs(abr_switch_map[r].nsbass) > 0) {
var k = (int)(abr_switch_map[r].nsbass * 4);
if (k < 0)
k += 64;
gfp.exp_nspsytune = gfp.exp_nspsytune | (k << 2);
}
if (enforce != 0)
gfp.quant_comp = abr_switch_map[r].quant_comp;
else if (!(Math.abs(gfp.quant_comp - -1) > 0))
gfp.quant_comp = abr_switch_map[r].quant_comp;
// SET_OPTION(quant_comp, abr_switch_map[r].quant_comp, -1);
if (enforce != 0)
gfp.quant_comp_short = abr_switch_map[r].quant_comp_s;
else if (!(Math.abs(gfp.quant_comp_short - -1) > 0))
gfp.quant_comp_short = abr_switch_map[r].quant_comp_s;
// SET_OPTION(quant_comp_short, abr_switch_map[r].quant_comp_s, -1);
if (enforce != 0)
gfp.msfix = abr_switch_map[r].nsmsfix;
else if (!(Math.abs(gfp.msfix - -1) > 0))
gfp.msfix = abr_switch_map[r].nsmsfix;
// SET_OPTION(msfix, abr_switch_map[r].nsmsfix, -1);
if (enforce != 0)
gfp.internal_flags.nsPsy.attackthre = abr_switch_map[r].st_lrm;
else if (!(Math.abs(gfp.internal_flags.nsPsy.attackthre - -1) > 0))
gfp.internal_flags.nsPsy.attackthre = abr_switch_map[r].st_lrm;
// SET_OPTION(short_threshold_lrm, abr_switch_map[r].st_lrm, -1);
if (enforce != 0)
gfp.internal_flags.nsPsy.attackthre_s = abr_switch_map[r].st_s;
else if (!(Math.abs(gfp.internal_flags.nsPsy.attackthre_s - -1) > 0))
gfp.internal_flags.nsPsy.attackthre_s = abr_switch_map[r].st_s;
// SET_OPTION(short_threshold_s, abr_switch_map[r].st_s, -1);
/*
* ABR seems to have big problems with clipping, especially at low
* bitrates
*/
/*
* so we compensate for that here by using a scale value depending on
* bitrate
*/
if (enforce != 0)
gfp.scale = abr_switch_map[r].scale;
else if (!(Math.abs(gfp.scale - -1) > 0))
gfp.scale = abr_switch_map[r].scale;
// SET_OPTION(scale, abr_switch_map[r].scale, -1);
if (enforce != 0)
gfp.maskingadjust = abr_switch_map[r].masking_adj;
else if (!(Math.abs(gfp.maskingadjust - 0) > 0))
gfp.maskingadjust = abr_switch_map[r].masking_adj;
// SET_OPTION(maskingadjust, abr_switch_map[r].masking_adj, 0);
if (abr_switch_map[r].masking_adj > 0) {
if (enforce != 0)
gfp.maskingadjust_short = (abr_switch_map[r].masking_adj * .9);
else if (!(Math.abs(gfp.maskingadjust_short - 0) > 0))
gfp.maskingadjust_short = (abr_switch_map[r].masking_adj * .9);
// SET_OPTION(maskingadjust_short, abr_switch_map[r].masking_adj *
// .9, 0);
} else {
if (enforce != 0)
gfp.maskingadjust_short = (abr_switch_map[r].masking_adj * 1.1);
else if (!(Math.abs(gfp.maskingadjust_short - 0) > 0))
gfp.maskingadjust_short = (abr_switch_map[r].masking_adj * 1.1);
// SET_OPTION(maskingadjust_short, abr_switch_map[r].masking_adj *
// 1.1, 0);
}
if (enforce != 0)
gfp.ATHlower = -abr_switch_map[r].ath_lower / 10.;
else if (!(Math.abs((-gfp.ATHlower * 10.) - 0) > 0))
gfp.ATHlower = -abr_switch_map[r].ath_lower / 10.;
// SET_OPTION(ATHlower, abr_switch_map[r].ath_lower, 0);
if (enforce != 0)
gfp.ATHcurve = abr_switch_map[r].ath_curve;
else if (!(Math.abs(gfp.ATHcurve - -1) > 0))
gfp.ATHcurve = abr_switch_map[r].ath_curve;
// SET_OPTION(ATHcurve, abr_switch_map[r].ath_curve, -1);
if (enforce != 0)
gfp.interChRatio = abr_switch_map[r].interch;
else if (!(Math.abs(gfp.interChRatio - -1) > 0))
gfp.interChRatio = abr_switch_map[r].interch;
// SET_OPTION(interChRatio, abr_switch_map[r].interch, -1);
return preset;
}
this.apply_preset = function(gfp, preset, enforce) {
/* translate legacy presets */
switch (preset) {
case Lame.R3MIX:
{
preset = Lame.V3;
gfp.VBR = VbrMode$20.vbr_mtrh;
break;
}
case Lame.MEDIUM:
{
preset = Lame.V4;
gfp.VBR = VbrMode$20.vbr_rh;
break;
}
case Lame.MEDIUM_FAST:
{
preset = Lame.V4;
gfp.VBR = VbrMode$20.vbr_mtrh;
break;
}
case Lame.STANDARD:
{
preset = Lame.V2;
gfp.VBR = VbrMode$20.vbr_rh;
break;
}
case Lame.STANDARD_FAST:
{
preset = Lame.V2;
gfp.VBR = VbrMode$20.vbr_mtrh;
break;
}
case Lame.EXTREME:
{
preset = Lame.V0;
gfp.VBR = VbrMode$20.vbr_rh;
break;
}
case Lame.EXTREME_FAST:
{
preset = Lame.V0;
gfp.VBR = VbrMode$20.vbr_mtrh;
break;
}
case Lame.INSANE:
{
preset = 320;
gfp.preset = preset;
apply_abr_preset(gfp, preset, enforce);
gfp.VBR = VbrMode$20.vbr_off;
return preset;
}
}
gfp.preset = preset;
{
switch (preset) {
case Lame.V9:
apply_vbr_preset(gfp, 9, enforce);
return preset;
case Lame.V8:
apply_vbr_preset(gfp, 8, enforce);
return preset;
case Lame.V7:
apply_vbr_preset(gfp, 7, enforce);
return preset;
case Lame.V6:
apply_vbr_preset(gfp, 6, enforce);
return preset;
case Lame.V5:
apply_vbr_preset(gfp, 5, enforce);
return preset;
case Lame.V4:
apply_vbr_preset(gfp, 4, enforce);
return preset;
case Lame.V3:
apply_vbr_preset(gfp, 3, enforce);
return preset;
case Lame.V2:
apply_vbr_preset(gfp, 2, enforce);
return preset;
case Lame.V1:
apply_vbr_preset(gfp, 1, enforce);
return preset;
case Lame.V0:
apply_vbr_preset(gfp, 0, enforce);
return preset;
default:
break;
}
}
if (8 <= preset && preset <= 320) {
return apply_abr_preset(gfp, preset, enforce);
}
/* no corresponding preset found */
gfp.preset = 0;
return preset;
};
// Rest from getset.c:
/**
* VBR quality level.* 0 = highest
* 9 = lowest */ function lame_set_VBR_q(gfp, VBR_q) { var ret = 0; if (0 > VBR_q) { /* Unknown VBR quality level! */ ret = -1; VBR_q = 0; } if (9 < VBR_q) { ret = -1; VBR_q = 9; } gfp.VBR_q = VBR_q; gfp.VBR_q_frac = 0; return ret; } } var Presets_1 = Presets$1; function VBRQuantize() { var qupvt; var tak; this.setModules = function (_qupvt, _tk) { qupvt = _qupvt; tak = _tk; }; //TODO } var VBRQuantize_1 = VBRQuantize; //package mp3; function CalcNoiseResult$1() { /** * sum of quantization noise > masking */ this.over_noise = 0.; /** * sum of all quantization noise */ this.tot_noise = 0.; /** * max quantization noise */ this.max_noise = 0.; /** * number of quantization noise > masking */ this.over_count = 0; /** * SSD-like cost of distorted bands */ this.over_SSD = 0; this.bits = 0; } var CalcNoiseResult_1 = CalcNoiseResult$1; var new_float$22 = common.new_float; var new_int$22 = common.new_int; function CalcNoiseData() { this.global_gain = 0; this.sfb_count1 = 0; this.step = new_int$22(39); this.noise = new_float$22(39); this.noise_log = new_float$22(39); } var CalcNoiseData_1 = CalcNoiseData; /* * MP3 quantization * * Copyright (c) 1999-2000 Mark Taylor * Copyright (c) 1999-2003 Takehiro Tominaga * Copyright (c) 2000-2007 Robert Hegemann * Copyright (c) 2001-2005 Gabriel Bouvigne * * 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 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 * Library 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. */ /* $Id: Quantize.java,v 1.24 2011/05/24 20:48:06 kenchis Exp $ */ //package mp3; //import java.util.Arrays; var System$21 = common.System; var VbrMode$21 = common.VbrMode; var Util$21 = common.Util; var Arrays$21 = common.Arrays; var new_float$21 = common.new_float; var assert$21 = common.assert; function Quantize$1() { var bs; this.rv = null; var rv; this.qupvt = null; var qupvt; var vbr = new VBRQuantize_1(); var tk; this.setModules = function (_bs, _rv, _qupvt, _tk) { bs = _bs; rv = _rv; this.rv = _rv; qupvt = _qupvt; this.qupvt = _qupvt; tk = _tk; vbr.setModules(qupvt, tk); }; /** * convert from L/R <. Mid/Side */ this.ms_convert = function (l3_side, gr) { for (var i = 0; i < 576; ++i) { var l = l3_side.tt[gr][0].xr[i]; var r = l3_side.tt[gr][1].xr[i]; l3_side.tt[gr][0].xr[i] = (l + r) * (Util$21.SQRT2 * 0.5); l3_side.tt[gr][1].xr[i] = (l - r) * (Util$21.SQRT2 * 0.5); } }; /** * mt 6/99 * * initializes cod_info, scalefac and xrpow * * returns 0 if all energies in xr are zero, else 1 */ function init_xrpow_core(cod_info, xrpow, upper, sum) { sum = 0; for (var i = 0; i <= upper; ++i) { var tmp = Math.abs(cod_info.xr[i]); sum += tmp; xrpow[i] = Math.sqrt(tmp * Math.sqrt(tmp)); if (xrpow[i] > cod_info.xrpow_max) cod_info.xrpow_max = xrpow[i]; } return sum; } this.init_xrpow = function (gfc, cod_info, xrpow) { var sum = 0; var upper = 0 | cod_info.max_nonzero_coeff; assert$21(xrpow != null); cod_info.xrpow_max = 0; /* * check if there is some energy we have to quantize and calculate xrpow * matching our fresh scalefactors */ assert$21(0 <= upper && upper <= 575); Arrays$21.fill(xrpow, upper, 576, 0); sum = init_xrpow_core(cod_info, xrpow, upper, sum); /* * return 1 if we have something to quantize, else 0 */ if (sum > 1E-20) { var j = 0; if ((gfc.substep_shaping & 2) != 0) j = 1; for (var i = 0; i < cod_info.psymax; i++) gfc.pseudohalf[i] = j; return true; } Arrays$21.fill(cod_info.l3_enc, 0, 576, 0); return false; }; /** * Gabriel Bouvigne feb/apr 2003
* Analog silence detection in partitionned sfb21 or sfb12 for short blocks * * From top to bottom of sfb, changes to 0 coeffs which are below ath. It * stops on the first coeff higher than ath. */ function psfb21_analogsilence(gfc, cod_info) { var ath = gfc.ATH; var xr = cod_info.xr; if (cod_info.block_type != Encoder_1.SHORT_TYPE) { /* NORM, START or STOP type, but not SHORT blocks */ var stop = false; for (var gsfb = Encoder_1.PSFB21 - 1; gsfb >= 0 && !stop; gsfb--) { var start = gfc.scalefac_band.psfb21[gsfb]; var end = gfc.scalefac_band.psfb21[gsfb + 1]; var ath21 = qupvt.athAdjust(ath.adjust, ath.psfb21[gsfb], ath.floor); if (gfc.nsPsy.longfact[21] > 1e-12) ath21 *= gfc.nsPsy.longfact[21]; for (var j = end - 1; j >= start; j--) { if (Math.abs(xr[j]) < ath21) xr[j] = 0; else { stop = true; break; } } } } else { /* note: short blocks coeffs are reordered */ for (var block = 0; block < 3; block++) { var stop = false; for (var gsfb = Encoder_1.PSFB12 - 1; gsfb >= 0 && !stop; gsfb--) { var start = gfc.scalefac_band.s[12] * 3 + (gfc.scalefac_band.s[13] - gfc.scalefac_band.s[12]) * block + (gfc.scalefac_band.psfb12[gsfb] - gfc.scalefac_band.psfb12[0]); var end = start + (gfc.scalefac_band.psfb12[gsfb + 1] - gfc.scalefac_band.psfb12[gsfb]); var ath12 = qupvt.athAdjust(ath.adjust, ath.psfb12[gsfb], ath.floor); if (gfc.nsPsy.shortfact[12] > 1e-12) ath12 *= gfc.nsPsy.shortfact[12]; for (var j = end - 1; j >= start; j--) { if (Math.abs(xr[j]) < ath12) xr[j] = 0; else { stop = true; break; } } } } } } this.init_outer_loop = function (gfc, cod_info) { /* * initialize fresh cod_info */ cod_info.part2_3_length = 0; cod_info.big_values = 0; cod_info.count1 = 0; cod_info.global_gain = 210; cod_info.scalefac_compress = 0; /* mixed_block_flag, block_type was set in psymodel.c */ cod_info.table_select[0] = 0; cod_info.table_select[1] = 0; cod_info.table_select[2] = 0; cod_info.subblock_gain[0] = 0; cod_info.subblock_gain[1] = 0; cod_info.subblock_gain[2] = 0; cod_info.subblock_gain[3] = 0; /* this one is always 0 */ cod_info.region0_count = 0; cod_info.region1_count = 0; cod_info.preflag = 0; cod_info.scalefac_scale = 0; cod_info.count1table_select = 0; cod_info.part2_length = 0; cod_info.sfb_lmax = Encoder_1.SBPSY_l; cod_info.sfb_smin = Encoder_1.SBPSY_s; cod_info.psy_lmax = gfc.sfb21_extra ? Encoder_1.SBMAX_l : Encoder_1.SBPSY_l; cod_info.psymax = cod_info.psy_lmax; cod_info.sfbmax = cod_info.sfb_lmax; cod_info.sfbdivide = 11; for (var sfb = 0; sfb < Encoder_1.SBMAX_l; sfb++) { cod_info.width[sfb] = gfc.scalefac_band.l[sfb + 1] - gfc.scalefac_band.l[sfb]; /* which is always 0. */ cod_info.window[sfb] = 3; } if (cod_info.block_type == Encoder_1.SHORT_TYPE) { var ixwork = new_float$21(576); cod_info.sfb_smin = 0; cod_info.sfb_lmax = 0; if (cod_info.mixed_block_flag != 0) { /* * MPEG-1: sfbs 0-7 long block, 3-12 short blocks MPEG-2(.5): * sfbs 0-5 long block, 3-12 short blocks */ cod_info.sfb_smin = 3; cod_info.sfb_lmax = gfc.mode_gr * 2 + 4; } cod_info.psymax = cod_info.sfb_lmax + 3 * ((gfc.sfb21_extra ? Encoder_1.SBMAX_s : Encoder_1.SBPSY_s) - cod_info.sfb_smin); cod_info.sfbmax = cod_info.sfb_lmax + 3 * (Encoder_1.SBPSY_s - cod_info.sfb_smin); cod_info.sfbdivide = cod_info.sfbmax - 18; cod_info.psy_lmax = cod_info.sfb_lmax; /* re-order the short blocks, for more efficient encoding below */ /* By Takehiro TOMINAGA */ /* * Within each scalefactor band, data is given for successive time * windows, beginning with window 0 and ending with window 2. Within * each window, the quantized values are then arranged in order of * increasing frequency... */ var ix = gfc.scalefac_band.l[cod_info.sfb_lmax]; System$21.arraycopy(cod_info.xr, 0, ixwork, 0, 576); for (var sfb = cod_info.sfb_smin; sfb < Encoder_1.SBMAX_s; sfb++) { var start = gfc.scalefac_band.s[sfb]; var end = gfc.scalefac_band.s[sfb + 1]; for (var window = 0; window < 3; window++) { for (var l = start; l < end; l++) { cod_info.xr[ix++] = ixwork[3 * l + window]; } } } var j = cod_info.sfb_lmax; for (var sfb = cod_info.sfb_smin; sfb < Encoder_1.SBMAX_s; sfb++) { cod_info.width[j] = cod_info.width[j + 1] = cod_info.width[j + 2] = gfc.scalefac_band.s[sfb + 1] - gfc.scalefac_band.s[sfb]; cod_info.window[j] = 0; cod_info.window[j + 1] = 1; cod_info.window[j + 2] = 2; j += 3; } } cod_info.count1bits = 0; cod_info.sfb_partition_table = qupvt.nr_of_sfb_block[0][0]; cod_info.slen[0] = 0; cod_info.slen[1] = 0; cod_info.slen[2] = 0; cod_info.slen[3] = 0; cod_info.max_nonzero_coeff = 575; /* * fresh scalefactors are all zero */ Arrays$21.fill(cod_info.scalefac, 0); psfb21_analogsilence(gfc, cod_info); }; function BinSearchDirection(ordinal) { this.ordinal = ordinal; } BinSearchDirection.BINSEARCH_NONE = new BinSearchDirection(0); BinSearchDirection.BINSEARCH_UP = new BinSearchDirection(1); BinSearchDirection.BINSEARCH_DOWN = new BinSearchDirection(2); /** * author/date?? * * binary step size search used by outer_loop to get a quantizer step size * to start with */ function bin_search_StepSize(gfc, cod_info, desired_rate, ch, xrpow) { var nBits; var CurrentStep = gfc.CurrentStep[ch]; var flagGoneOver = false; var start = gfc.OldValue[ch]; var Direction = BinSearchDirection.BINSEARCH_NONE; cod_info.global_gain = start; desired_rate -= cod_info.part2_length; assert$21(CurrentStep != 0); for (; ;) { var step; nBits = tk.count_bits(gfc, xrpow, cod_info, null); if (CurrentStep == 1 || nBits == desired_rate) break; /* nothing to adjust anymore */ if (nBits > desired_rate) { /* increase Quantize_StepSize */ if (Direction == BinSearchDirection.BINSEARCH_DOWN) flagGoneOver = true; if (flagGoneOver) CurrentStep /= 2; Direction = BinSearchDirection.BINSEARCH_UP; step = CurrentStep; } else { /* decrease Quantize_StepSize */ if (Direction == BinSearchDirection.BINSEARCH_UP) flagGoneOver = true; if (flagGoneOver) CurrentStep /= 2; Direction = BinSearchDirection.BINSEARCH_DOWN; step = -CurrentStep; } cod_info.global_gain += step; if (cod_info.global_gain < 0) { cod_info.global_gain = 0; flagGoneOver = true; } if (cod_info.global_gain > 255) { cod_info.global_gain = 255; flagGoneOver = true; } } assert$21(cod_info.global_gain >= 0); assert$21(cod_info.global_gain < 256); while (nBits > desired_rate && cod_info.global_gain < 255) { cod_info.global_gain++; nBits = tk.count_bits(gfc, xrpow, cod_info, null); } gfc.CurrentStep[ch] = (start - cod_info.global_gain >= 4) ? 4 : 2; gfc.OldValue[ch] = cod_info.global_gain; cod_info.part2_3_length = nBits; return nBits; } this.trancate_smallspectrums = function (gfc, gi, l3_xmin, work) { var distort = new_float$21(L3Side_1.SFBMAX); if ((0 == (gfc.substep_shaping & 4) && gi.block_type == Encoder_1.SHORT_TYPE) || (gfc.substep_shaping & 0x80) != 0) return; qupvt.calc_noise(gi, l3_xmin, distort, new CalcNoiseResult_1(), null); for (var j = 0; j < 576; j++) { var xr = 0.0; if (gi.l3_enc[j] != 0) xr = Math.abs(gi.xr[j]); work[j] = xr; } var j = 0; var sfb = 8; if (gi.block_type == Encoder_1.SHORT_TYPE) sfb = 6; do { var allowedNoise, trancateThreshold; var nsame, start; var width = gi.width[sfb]; j += width; if (distort[sfb] >= 1.0) continue; Arrays$21.sort(work, j - width, width); if (BitStream.EQ(work[j - 1], 0.0)) continue; /* all zero sfb */ allowedNoise = (1.0 - distort[sfb]) * l3_xmin[sfb]; trancateThreshold = 0.0; start = 0; do { var noise; for (nsame = 1; start + nsame < width; nsame++) if (BitStream.NEQ(work[start + j - width], work[start + j + nsame - width])) break; noise = work[start + j - width] * work[start + j - width] * nsame; if (allowedNoise < noise) { if (start != 0) trancateThreshold = work[start + j - width - 1]; break; } allowedNoise -= noise; start += nsame; } while (start < width); if (BitStream.EQ(trancateThreshold, 0.0)) continue; do { if (Math.abs(gi.xr[j - width]) <= trancateThreshold) gi.l3_enc[j - width] = 0; } while (--width > 0); } while (++sfb < gi.psymax); gi.part2_3_length = tk.noquant_count_bits(gfc, gi, null); }; /** * author/date?? * * Function: Returns zero if there is a scalefac which has not been * amplified. Otherwise it returns one. */ function loop_break(cod_info) { for (var sfb = 0; sfb < cod_info.sfbmax; sfb++) if (cod_info.scalefac[sfb] + cod_info.subblock_gain[cod_info.window[sfb]] == 0) return false; return true; } /* mt 5/99: Function: Improved calc_noise for a single channel */ function penalties(noise) { return Util$21.FAST_LOG10((0.368 + 0.632 * noise * noise * noise)); } /** * author/date?? * * several different codes to decide which quantization is better */ function get_klemm_noise(distort, gi) { var klemm_noise = 1E-37; for (var sfb = 0; sfb < gi.psymax; sfb++) klemm_noise += penalties(distort[sfb]); return Math.max(1e-20, klemm_noise); } function quant_compare(quant_comp, best, calc, gi, distort) { /** * noise is given in decibels (dB) relative to masking thesholds.
* * over_noise: ??? (the previous comment is fully wrong)
* tot_noise: ??? (the previous comment is fully wrong)
* max_noise: max quantization noise */ var better; switch (quant_comp) { default: case 9: { if (best.over_count > 0) { /* there are distorted sfb */ better = calc.over_SSD <= best.over_SSD; if (calc.over_SSD == best.over_SSD) better = calc.bits < best.bits; } else { /* no distorted sfb */ better = ((calc.max_noise < 0) && ((calc.max_noise * 10 + calc.bits) <= (best.max_noise * 10 + best.bits))); } break; } case 0: better = calc.over_count < best.over_count || (calc.over_count == best.over_count && calc.over_noise < best.over_noise) || (calc.over_count == best.over_count && BitStream.EQ(calc.over_noise, best.over_noise) && calc.tot_noise < best.tot_noise); break; case 8: calc.max_noise = get_klemm_noise(distort, gi); //$FALL-THROUGH$ case 1: better = calc.max_noise < best.max_noise; break; case 2: better = calc.tot_noise < best.tot_noise; break; case 3: better = (calc.tot_noise < best.tot_noise) && (calc.max_noise < best.max_noise); break; case 4: better = (calc.max_noise <= 0.0 && best.max_noise > 0.2) || (calc.max_noise <= 0.0 && best.max_noise < 0.0 && best.max_noise > calc.max_noise - 0.2 && calc.tot_noise < best.tot_noise) || (calc.max_noise <= 0.0 && best.max_noise > 0.0 && best.max_noise > calc.max_noise - 0.2 && calc.tot_noise < best.tot_noise + best.over_noise) || (calc.max_noise > 0.0 && best.max_noise > -0.05 && best.max_noise > calc.max_noise - 0.1 && calc.tot_noise + calc.over_noise < best.tot_noise + best.over_noise) || (calc.max_noise > 0.0 && best.max_noise > -0.1 && best.max_noise > calc.max_noise - 0.15 && calc.tot_noise + calc.over_noise + calc.over_noise < best.tot_noise + best.over_noise + best.over_noise); break; case 5: better = calc.over_noise < best.over_noise || (BitStream.EQ(calc.over_noise, best.over_noise) && calc.tot_noise < best.tot_noise); break; case 6: better = calc.over_noise < best.over_noise || (BitStream.EQ(calc.over_noise, best.over_noise) && (calc.max_noise < best.max_noise || (BitStream .EQ(calc.max_noise, best.max_noise) && calc.tot_noise <= best.tot_noise))); break; case 7: better = calc.over_count < best.over_count || calc.over_noise < best.over_noise; break; } if (best.over_count == 0) { /* * If no distorted bands, only use this quantization if it is * better, and if it uses less bits. Unfortunately, part2_3_length * is sometimes a poor estimator of the final size at low bitrates. */ better = better && calc.bits < best.bits; } return better; } /** * author/date?? * *
* Amplify the scalefactor bands that violate the masking threshold.
* See ISO 11172-3 Section C.1.5.4.3.5
*
* distort[] = noise/masking
* distort[] > 1 ==> noise is not masked
* distort[] < 1 ==> noise is masked
* max_dist = maximum value of distort[]
*
* Three algorithms:
* noise_shaping_amp
* 0 Amplify all bands with distort[]>1.
*
* 1 Amplify all bands with distort[] >= max_dist^(.5);
* ( 50% in the db scale)
*
* 2 Amplify first band with distort[] >= max_dist;
*
*
* For algorithms 0 and 1, if max_dist < 1, then amplify all bands
* with distort[] >= .95*max_dist. This is to make sure we always
* amplify at least one band.
*
*/
function amp_scalefac_bands(gfp, cod_info, distort, xrpow, bRefine) {
var gfc = gfp.internal_flags;
var ifqstep34;
if (cod_info.scalefac_scale == 0) {
ifqstep34 = 1.29683955465100964055;
/* 2**(.75*.5) */
} else {
ifqstep34 = 1.68179283050742922612;
/* 2**(.75*1) */
}
/* compute maximum value of distort[] */
var trigger = 0;
for (var sfb = 0; sfb < cod_info.sfbmax; sfb++) {
if (trigger < distort[sfb])
trigger = distort[sfb];
}
var noise_shaping_amp = gfc.noise_shaping_amp;
if (noise_shaping_amp == 3) {
if (bRefine)
noise_shaping_amp = 2;
else
noise_shaping_amp = 1;
}
switch (noise_shaping_amp) {
case 2:
/* amplify exactly 1 band */
break;
case 1:
/* amplify bands within 50% of max (on db scale) */
if (trigger > 1.0)
trigger = Math.pow(trigger, .5);
else
trigger *= .95;
break;
case 0:
default:
/* ISO algorithm. amplify all bands with distort>1 */
if (trigger > 1.0)
trigger = 1.0;
else
trigger *= .95;
break;
}
var j = 0;
for (var sfb = 0; sfb < cod_info.sfbmax; sfb++) {
var width = cod_info.width[sfb];
var l;
j += width;
if (distort[sfb] < trigger)
continue;
if ((gfc.substep_shaping & 2) != 0) {
gfc.pseudohalf[sfb] = (0 == gfc.pseudohalf[sfb]) ? 1 : 0;
if (0 == gfc.pseudohalf[sfb] && gfc.noise_shaping_amp == 2)
return;
}
cod_info.scalefac[sfb]++;
for (l = -width; l < 0; l++) {
xrpow[j + l] *= ifqstep34;
if (xrpow[j + l] > cod_info.xrpow_max)
cod_info.xrpow_max = xrpow[j + l];
}
if (gfc.noise_shaping_amp == 2)
return;
}
}
/**
* Takehiro Tominaga 2000-xx-xx
*
* turns on scalefac scale and adjusts scalefactors
*/
function inc_scalefac_scale(cod_info, xrpow) {
var ifqstep34 = 1.29683955465100964055;
var j = 0;
for (var sfb = 0; sfb < cod_info.sfbmax; sfb++) {
var width = cod_info.width[sfb];
var s = cod_info.scalefac[sfb];
if (cod_info.preflag != 0)
s += qupvt.pretab[sfb];
j += width;
if ((s & 1) != 0) {
s++;
for (var l = -width; l < 0; l++) {
xrpow[j + l] *= ifqstep34;
if (xrpow[j + l] > cod_info.xrpow_max)
cod_info.xrpow_max = xrpow[j + l];
}
}
cod_info.scalefac[sfb] = s >> 1;
}
cod_info.preflag = 0;
cod_info.scalefac_scale = 1;
}
/**
* Takehiro Tominaga 2000-xx-xx
*
* increases the subblock gain and adjusts scalefactors
*/
function inc_subblock_gain(gfc, cod_info, xrpow) {
var sfb;
var scalefac = cod_info.scalefac;
/* subbloc_gain can't do anything in the long block region */
for (sfb = 0; sfb < cod_info.sfb_lmax; sfb++) {
if (scalefac[sfb] >= 16)
return true;
}
for (var window = 0; window < 3; window++) {
var s1 = 0;
var s2 = 0;
for (sfb = cod_info.sfb_lmax + window; sfb < cod_info.sfbdivide; sfb += 3) {
if (s1 < scalefac[sfb])
s1 = scalefac[sfb];
}
for (; sfb < cod_info.sfbmax; sfb += 3) {
if (s2 < scalefac[sfb])
s2 = scalefac[sfb];
}
if (s1 < 16 && s2 < 8)
continue;
if (cod_info.subblock_gain[window] >= 7)
return true;
/*
* even though there is no scalefactor for sfb12 subblock gain
* affects upper frequencies too, that's why we have to go up to
* SBMAX_s
*/
cod_info.subblock_gain[window]++;
var j = gfc.scalefac_band.l[cod_info.sfb_lmax];
for (sfb = cod_info.sfb_lmax + window; sfb < cod_info.sfbmax; sfb += 3) {
var amp;
var width = cod_info.width[sfb];
var s = scalefac[sfb];
assert$21(s >= 0);
s = s - (4 >> cod_info.scalefac_scale);
if (s >= 0) {
scalefac[sfb] = s;
j += width * 3;
continue;
}
scalefac[sfb] = 0;
{
var gain = 210 + (s << (cod_info.scalefac_scale + 1));
amp = qupvt.IPOW20(gain);
}
j += width * (window + 1);
for (var l = -width; l < 0; l++) {
xrpow[j + l] *= amp;
if (xrpow[j + l] > cod_info.xrpow_max)
cod_info.xrpow_max = xrpow[j + l];
}
j += width * (3 - window - 1);
}
{
var amp = qupvt.IPOW20(202);
j += cod_info.width[sfb] * (window + 1);
for (var l = -cod_info.width[sfb]; l < 0; l++) {
xrpow[j + l] *= amp;
if (xrpow[j + l] > cod_info.xrpow_max)
cod_info.xrpow_max = xrpow[j + l];
}
}
}
return false;
}
/**
*
* Takehiro Tominaga /date??
* Robert Hegemann 2000-09-06: made a function of it
*
* amplifies scalefactor bands,
* - if all are already amplified returns 0
* - if some bands are amplified too much:
* * try to increase scalefac_scale
* * if already scalefac_scale was set
* try on short blocks to increase subblock gain
*
*/
function balance_noise(gfp, cod_info, distort, xrpow, bRefine) {
var gfc = gfp.internal_flags;
amp_scalefac_bands(gfp, cod_info, distort, xrpow, bRefine);
/*
* check to make sure we have not amplified too much loop_break returns
* 0 if there is an unamplified scalefac scale_bitcount returns 0 if no
* scalefactors are too large
*/
var status = loop_break(cod_info);
if (status)
return false;
/* all bands amplified */
/*
* not all scalefactors have been amplified. so these scalefacs are
* possibly valid. encode them:
*/
if (gfc.mode_gr == 2)
status = tk.scale_bitcount(cod_info);
else
status = tk.scale_bitcount_lsf(gfc, cod_info);
if (!status)
return true;
/* amplified some bands not exceeding limits */
/*
* some scalefactors are too large. lets try setting scalefac_scale=1
*/
if (gfc.noise_shaping > 1) {
Arrays$21.fill(gfc.pseudohalf, 0);
if (0 == cod_info.scalefac_scale) {
inc_scalefac_scale(cod_info, xrpow);
status = false;
} else {
if (cod_info.block_type == Encoder_1.SHORT_TYPE
&& gfc.subblock_gain > 0) {
status = (inc_subblock_gain(gfc, cod_info, xrpow) || loop_break(cod_info));
}
}
}
if (!status) {
if (gfc.mode_gr == 2)
status = tk.scale_bitcount(cod_info);
else
status = tk.scale_bitcount_lsf(gfc, cod_info);
}
return !status;
}
/**
*
* Function: The outer iteration loop controls the masking conditions
* of all scalefactorbands. It computes the best scalefac and
* global gain. This module calls the inner iteration loop
*
* mt 5/99 completely rewritten to allow for bit reservoir control,
* mid/side channels with L/R or mid/side masking thresholds,
* and chooses best quantization instead of last quantization when
* no distortion free quantization can be found.
*
* added VBR support mt 5/99
*
* some code shuffle rh 9/00
*
*
* @param l3_xmin
* allowed distortion
* @param xrpow
* coloured magnitudes of spectral
* @param targ_bits
* maximum allowed bits
*/
this.outer_loop = function (gfp, cod_info, l3_xmin, xrpow, ch, targ_bits) {
var gfc = gfp.internal_flags;
var cod_info_w = new GrInfo_1();
var save_xrpow = new_float$21(576);
var distort = new_float$21(L3Side_1.SFBMAX);
var best_noise_info = new CalcNoiseResult_1();
var better;
var prev_noise = new CalcNoiseData_1();
var best_part2_3_length = 9999999;
var bEndOfSearch = false;
var bRefine = false;
var best_ggain_pass1 = 0;
bin_search_StepSize(gfc, cod_info, targ_bits, ch, xrpow);
if (0 == gfc.noise_shaping)
/* fast mode, no noise shaping, we are ready */
return 100;
/* default noise_info.over_count */
/* compute the distortion in this quantization */
/* coefficients and thresholds both l/r (or both mid/side) */
qupvt.calc_noise(cod_info, l3_xmin, distort, best_noise_info,
prev_noise);
best_noise_info.bits = cod_info.part2_3_length;
cod_info_w.assign(cod_info);
var age = 0;
System$21.arraycopy(xrpow, 0, save_xrpow, 0, 576);
while (!bEndOfSearch) {
/* BEGIN MAIN LOOP */
do {
var noise_info = new CalcNoiseResult_1();
var search_limit;
var maxggain = 255;
/*
* When quantization with no distorted bands is found, allow up
* to X new unsuccesful tries in serial. This gives us more
* possibilities for different quant_compare modes. Much more
* than 3 makes not a big difference, it is only slower.
*/
if ((gfc.substep_shaping & 2) != 0) {
search_limit = 20;
} else {
search_limit = 3;
}
/*
* Check if the last scalefactor band is distorted. in VBR mode
* we can't get rid of the distortion, so quit now and VBR mode
* will try again with more bits. (makes a 10% speed increase,
* the files I tested were binary identical, 2000/05/20 Robert
* Hegemann) distort[] > 1 means noise > allowed noise
*/
if (gfc.sfb21_extra) {
if (distort[cod_info_w.sfbmax] > 1.0)
break;
if (cod_info_w.block_type == Encoder_1.SHORT_TYPE
&& (distort[cod_info_w.sfbmax + 1] > 1.0 || distort[cod_info_w.sfbmax + 2] > 1.0))
break;
}
/* try a new scalefactor conbination on cod_info_w */
if (!balance_noise(gfp, cod_info_w, distort, xrpow, bRefine))
break;
if (cod_info_w.scalefac_scale != 0)
maxggain = 254;
/*
* inner_loop starts with the initial quantization step computed
* above and slowly increases until the bits < huff_bits. Thus
* it is important not to start with too large of an inital
* quantization step. Too small is ok, but inner_loop will take
* longer
*/
var huff_bits = targ_bits - cod_info_w.part2_length;
if (huff_bits <= 0)
break;
/*
* increase quantizer stepsize until needed bits are below
* maximum
*/
while ((cod_info_w.part2_3_length = tk.count_bits(gfc, xrpow,
cod_info_w, prev_noise)) > huff_bits
&& cod_info_w.global_gain <= maxggain)
cod_info_w.global_gain++;
if (cod_info_w.global_gain > maxggain)
break;
if (best_noise_info.over_count == 0) {
while ((cod_info_w.part2_3_length = tk.count_bits(gfc,
xrpow, cod_info_w, prev_noise)) > best_part2_3_length
&& cod_info_w.global_gain <= maxggain)
cod_info_w.global_gain++;
if (cod_info_w.global_gain > maxggain)
break;
}
/* compute the distortion in this quantization */
qupvt.calc_noise(cod_info_w, l3_xmin, distort, noise_info,
prev_noise);
noise_info.bits = cod_info_w.part2_3_length;
/*
* check if this quantization is better than our saved
* quantization
*/
if (cod_info.block_type != Encoder_1.SHORT_TYPE) {
// NORM, START or STOP type
better = gfp.quant_comp;
} else
better = gfp.quant_comp_short;
better = quant_compare(better, best_noise_info, noise_info,
cod_info_w, distort) ? 1 : 0;
/* save data so we can restore this quantization later */
if (better != 0) {
best_part2_3_length = cod_info.part2_3_length;
best_noise_info = noise_info;
cod_info.assign(cod_info_w);
age = 0;
/* save data so we can restore this quantization later */
/* store for later reuse */
System$21.arraycopy(xrpow, 0, save_xrpow, 0, 576);
} else {
/* early stop? */
if (gfc.full_outer_loop == 0) {
if (++age > search_limit
&& best_noise_info.over_count == 0)
break;
if ((gfc.noise_shaping_amp == 3) && bRefine && age > 30)
break;
if ((gfc.noise_shaping_amp == 3)
&& bRefine
&& (cod_info_w.global_gain - best_ggain_pass1) > 15)
break;
}
}
} while ((cod_info_w.global_gain + cod_info_w.scalefac_scale) < 255);
if (gfc.noise_shaping_amp == 3) {
if (!bRefine) {
/* refine search */
cod_info_w.assign(cod_info);
System$21.arraycopy(save_xrpow, 0, xrpow, 0, 576);
age = 0;
best_ggain_pass1 = cod_info_w.global_gain;
bRefine = true;
} else {
/* search already refined, stop */
bEndOfSearch = true;
}
} else {
bEndOfSearch = true;
}
}
assert$21((cod_info.global_gain + cod_info.scalefac_scale) <= 255);
/*
* finish up
*/
if (gfp.VBR == VbrMode$21.vbr_rh || gfp.VBR == VbrMode$21.vbr_mtrh)
/* restore for reuse on next try */
System$21.arraycopy(save_xrpow, 0, xrpow, 0, 576);
/*
* do the 'substep shaping'
*/
else if ((gfc.substep_shaping & 1) != 0)
trancate_smallspectrums(gfc, cod_info, l3_xmin, xrpow);
return best_noise_info.over_count;
};
/**
* Robert Hegemann 2000-09-06
*
* update reservoir status after FINAL quantization/bitrate
*/
this.iteration_finish_one = function (gfc, gr, ch) {
var l3_side = gfc.l3_side;
var cod_info = l3_side.tt[gr][ch];
/*
* try some better scalefac storage
*/
tk.best_scalefac_store(gfc, gr, ch, l3_side);
/*
* best huffman_divide may save some bits too
*/
if (gfc.use_best_huffman == 1)
tk.best_huffman_divide(gfc, cod_info);
/*
* update reservoir status after FINAL quantization/bitrate
*/
rv.ResvAdjust(gfc, cod_info);
};
/**
*
* 2000-09-04 Robert Hegemann
*
* @param l3_xmin
* allowed distortion of the scalefactor
* @param xrpow
* coloured magnitudes of spectral values
*/
this.VBR_encode_granule = function (gfp, cod_info, l3_xmin, xrpow, ch, min_bits, max_bits) {
var gfc = gfp.internal_flags;
var bst_cod_info = new GrInfo_1();
var bst_xrpow = new_float$21(576);
var Max_bits = max_bits;
var real_bits = max_bits + 1;
var this_bits = (max_bits + min_bits) / 2;
var dbits, over, found = 0;
var sfb21_extra = gfc.sfb21_extra;
assert$21(Max_bits <= LameInternalFlags.MAX_BITS_PER_CHANNEL);
Arrays$21.fill(bst_cod_info.l3_enc, 0);
/*
* search within round about 40 bits of optimal
*/
do {
assert$21(this_bits >= min_bits);
assert$21(this_bits <= max_bits);
assert$21(min_bits <= max_bits);
if (this_bits > Max_bits - 42)
gfc.sfb21_extra = false;
else
gfc.sfb21_extra = sfb21_extra;
over = outer_loop(gfp, cod_info, l3_xmin, xrpow, ch, this_bits);
/*
* is quantization as good as we are looking for ? in this case: is
* no scalefactor band distorted?
*/
if (over <= 0) {
found = 1;
/*
* now we know it can be done with "real_bits" and maybe we can
* skip some iterations
*/
real_bits = cod_info.part2_3_length;
/*
* store best quantization so far
*/
bst_cod_info.assign(cod_info);
System$21.arraycopy(xrpow, 0, bst_xrpow, 0, 576);
/*
* try with fewer bits
*/
max_bits = real_bits - 32;
dbits = max_bits - min_bits;
this_bits = (max_bits + min_bits) / 2;
} else {
/*
* try with more bits
*/
min_bits = this_bits + 32;
dbits = max_bits - min_bits;
this_bits = (max_bits + min_bits) / 2;
if (found != 0) {
found = 2;
/*
* start again with best quantization so far
*/
cod_info.assign(bst_cod_info);
System$21.arraycopy(bst_xrpow, 0, xrpow, 0, 576);
}
}
} while (dbits > 12);
gfc.sfb21_extra = sfb21_extra;
/*
* found=0 => nothing found, use last one found=1 => we just found the
* best and left the loop found=2 => we restored a good one and have now
* l3_enc to restore too
*/
if (found == 2) {
System$21.arraycopy(bst_cod_info.l3_enc, 0, cod_info.l3_enc, 0, 576);
}
assert$21(cod_info.part2_3_length <= Max_bits);
};
/**
* Robert Hegemann 2000-09-05
*
* calculates * how many bits are available for analog silent granules * how
* many bits to use for the lowest allowed bitrate * how many bits each
* bitrate would provide
*/
this.get_framebits = function (gfp, frameBits) {
var gfc = gfp.internal_flags;
/*
* always use at least this many bits per granule per channel unless we
* detect analog silence, see below
*/
gfc.bitrate_index = gfc.VBR_min_bitrate;
var bitsPerFrame = bs.getframebits(gfp);
/*
* bits for analog silence
*/
gfc.bitrate_index = 1;
bitsPerFrame = bs.getframebits(gfp);
for (var i = 1; i <= gfc.VBR_max_bitrate; i++) {
gfc.bitrate_index = i;
var mb = new MeanBits(bitsPerFrame);
frameBits[i] = rv.ResvFrameBegin(gfp, mb);
bitsPerFrame = mb.bits;
}
};
/* RH: this one needs to be overhauled sometime */
/**
*
* 2000-09-04 Robert Hegemann
*
* * converts LR to MS coding when necessary
* * calculates allowed/adjusted quantization noise amounts
* * detects analog silent frames
*
* some remarks:
* - lower masking depending on Quality setting
* - quality control together with adjusted ATH MDCT scaling
* on lower quality setting allocate more noise from
* ATH masking, and on higher quality setting allocate
* less noise from ATH masking.
* - experiments show that going more than 2dB over GPSYCHO's
* limits ends up in very annoying artefacts
*
*/
this.VBR_old_prepare = function (gfp, pe, ms_ener_ratio, ratio, l3_xmin, frameBits, min_bits,
max_bits, bands) {
var gfc = gfp.internal_flags;
var masking_lower_db, adjust = 0.0;
var analog_silence = 1;
var bits = 0;
gfc.bitrate_index = gfc.VBR_max_bitrate;
var avg = rv.ResvFrameBegin(gfp, new MeanBits(0)) / gfc.mode_gr;
get_framebits(gfp, frameBits);
for (var gr = 0; gr < gfc.mode_gr; gr++) {
var mxb = qupvt.on_pe(gfp, pe, max_bits[gr], avg, gr, 0);
if (gfc.mode_ext == Encoder_1.MPG_MD_MS_LR) {
ms_convert(gfc.l3_side, gr);
qupvt.reduce_side(max_bits[gr], ms_ener_ratio[gr], avg, mxb);
}
for (var ch = 0; ch < gfc.channels_out; ++ch) {
var cod_info = gfc.l3_side.tt[gr][ch];
if (cod_info.block_type != Encoder_1.SHORT_TYPE) {
// NORM, START or STOP type
adjust = 1.28 / (1 + Math
.exp(3.5 - pe[gr][ch] / 300.)) - 0.05;
masking_lower_db = gfc.PSY.mask_adjust - adjust;
} else {
adjust = 2.56 / (1 + Math
.exp(3.5 - pe[gr][ch] / 300.)) - 0.14;
masking_lower_db = gfc.PSY.mask_adjust_short - adjust;
}
gfc.masking_lower = Math.pow(10.0,
masking_lower_db * 0.1);
init_outer_loop(gfc, cod_info);
bands[gr][ch] = qupvt.calc_xmin(gfp, ratio[gr][ch], cod_info,
l3_xmin[gr][ch]);
if (bands[gr][ch] != 0)
analog_silence = 0;
min_bits[gr][ch] = 126;
bits += max_bits[gr][ch];
}
}
for (var gr = 0; gr < gfc.mode_gr; gr++) {
for (var ch = 0; ch < gfc.channels_out; ch++) {
if (bits > frameBits[gfc.VBR_max_bitrate]) {
max_bits[gr][ch] *= frameBits[gfc.VBR_max_bitrate];
max_bits[gr][ch] /= bits;
}
if (min_bits[gr][ch] > max_bits[gr][ch])
min_bits[gr][ch] = max_bits[gr][ch];
}
/* for ch */
}
/* for gr */
return analog_silence;
};
this.bitpressure_strategy = function (gfc, l3_xmin, min_bits, max_bits) {
for (var gr = 0; gr < gfc.mode_gr; gr++) {
for (var ch = 0; ch < gfc.channels_out; ch++) {
var gi = gfc.l3_side.tt[gr][ch];
var pxmin = l3_xmin[gr][ch];
var pxminPos = 0;
for (var sfb = 0; sfb < gi.psy_lmax; sfb++)
pxmin[pxminPos++] *= 1. + .029 * sfb * sfb
/ Encoder_1.SBMAX_l / Encoder_1.SBMAX_l;
if (gi.block_type == Encoder_1.SHORT_TYPE) {
for (var sfb = gi.sfb_smin; sfb < Encoder_1.SBMAX_s; sfb++) {
pxmin[pxminPos++] *= 1. + .029 * sfb * sfb
/ Encoder_1.SBMAX_s / Encoder_1.SBMAX_s;
pxmin[pxminPos++] *= 1. + .029 * sfb * sfb
/ Encoder_1.SBMAX_s / Encoder_1.SBMAX_s;
pxmin[pxminPos++] *= 1. + .029 * sfb * sfb
/ Encoder_1.SBMAX_s / Encoder_1.SBMAX_s;
}
}
max_bits[gr][ch] = 0 | Math.max(min_bits[gr][ch],
0.9 * max_bits[gr][ch]);
}
}
};
this.VBR_new_prepare = function (gfp, pe, ratio, l3_xmin, frameBits, max_bits) {
var gfc = gfp.internal_flags;
var analog_silence = 1;
var avg = 0, bits = 0;
var maximum_framebits;
if (!gfp.free_format) {
gfc.bitrate_index = gfc.VBR_max_bitrate;
var mb = new MeanBits(avg);
rv.ResvFrameBegin(gfp, mb);
avg = mb.bits;
get_framebits(gfp, frameBits);
maximum_framebits = frameBits[gfc.VBR_max_bitrate];
} else {
gfc.bitrate_index = 0;
var mb = new MeanBits(avg);
maximum_framebits = rv.ResvFrameBegin(gfp, mb);
avg = mb.bits;
frameBits[0] = maximum_framebits;
}
for (var gr = 0; gr < gfc.mode_gr; gr++) {
qupvt.on_pe(gfp, pe, max_bits[gr], avg, gr, 0);
if (gfc.mode_ext == Encoder_1.MPG_MD_MS_LR) {
ms_convert(gfc.l3_side, gr);
}
for (var ch = 0; ch < gfc.channels_out; ++ch) {
var cod_info = gfc.l3_side.tt[gr][ch];
gfc.masking_lower = Math.pow(10.0,
gfc.PSY.mask_adjust * 0.1);
init_outer_loop(gfc, cod_info);
if (0 != qupvt.calc_xmin(gfp, ratio[gr][ch], cod_info,
l3_xmin[gr][ch]))
analog_silence = 0;
bits += max_bits[gr][ch];
}
}
for (var gr = 0; gr < gfc.mode_gr; gr++) {
for (var ch = 0; ch < gfc.channels_out; ch++) {
if (bits > maximum_framebits) {
max_bits[gr][ch] *= maximum_framebits;
max_bits[gr][ch] /= bits;
}
}
/* for ch */
}
/* for gr */
return analog_silence;
};
/**
* calculates target bits for ABR encoding
*
* mt 2000/05/31
*/
this.calc_target_bits = function (gfp, pe, ms_ener_ratio, targ_bits, analog_silence_bits, max_frame_bits) {
var gfc = gfp.internal_flags;
var l3_side = gfc.l3_side;
var res_factor;
var gr, ch, totbits, mean_bits = 0;
gfc.bitrate_index = gfc.VBR_max_bitrate;
var mb = new MeanBits(mean_bits);
max_frame_bits[0] = rv.ResvFrameBegin(gfp, mb);
mean_bits = mb.bits;
gfc.bitrate_index = 1;
mean_bits = bs.getframebits(gfp) - gfc.sideinfo_len * 8;
analog_silence_bits[0] = mean_bits / (gfc.mode_gr * gfc.channels_out);
mean_bits = gfp.VBR_mean_bitrate_kbps * gfp.framesize * 1000;
if ((gfc.substep_shaping & 1) != 0)
mean_bits *= 1.09;
mean_bits /= gfp.out_samplerate;
mean_bits -= gfc.sideinfo_len * 8;
mean_bits /= (gfc.mode_gr * gfc.channels_out);
/**
*
* res_factor is the percentage of the target bitrate that should
* be used on average. the remaining bits are added to the
* bitreservoir and used for difficult to encode frames.
*
* Since we are tracking the average bitrate, we should adjust
* res_factor "on the fly", increasing it if the average bitrate
* is greater than the requested bitrate, and decreasing it
* otherwise. Reasonable ranges are from .9 to 1.0
*
* Until we get the above suggestion working, we use the following
* tuning:
* compression ratio res_factor
* 5.5 (256kbps) 1.0 no need for bitreservoir
* 11 (128kbps) .93 7% held for reservoir
*
* with linear interpolation for other values.
*
*/
res_factor = .93 + .07 * (11.0 - gfp.compression_ratio)
/ (11.0 - 5.5);
if (res_factor < .90)
res_factor = .90;
if (res_factor > 1.00)
res_factor = 1.00;
for (gr = 0; gr < gfc.mode_gr; gr++) {
var sum = 0;
for (ch = 0; ch < gfc.channels_out; ch++) {
targ_bits[gr][ch] = (int)(res_factor * mean_bits);
if (pe[gr][ch] > 700) {
var add_bits = (int)((pe[gr][ch] - 700) / 1.4);
var cod_info = l3_side.tt[gr][ch];
targ_bits[gr][ch] = (int)(res_factor * mean_bits);
/* short blocks use a little extra, no matter what the pe */
if (cod_info.block_type == Encoder_1.SHORT_TYPE) {
if (add_bits < mean_bits / 2)
add_bits = mean_bits / 2;
}
/* at most increase bits by 1.5*average */
if (add_bits > mean_bits * 3 / 2)
add_bits = mean_bits * 3 / 2;
else if (add_bits < 0)
add_bits = 0;
targ_bits[gr][ch] += add_bits;
}
if (targ_bits[gr][ch] > LameInternalFlags.MAX_BITS_PER_CHANNEL) {
targ_bits[gr][ch] = LameInternalFlags.MAX_BITS_PER_CHANNEL;
}
sum += targ_bits[gr][ch];
}
/* for ch */
if (sum > LameInternalFlags.MAX_BITS_PER_GRANULE) {
for (ch = 0; ch < gfc.channels_out; ++ch) {
targ_bits[gr][ch] *= LameInternalFlags.MAX_BITS_PER_GRANULE;
targ_bits[gr][ch] /= sum;
}
}
}
/* for gr */
if (gfc.mode_ext == Encoder_1.MPG_MD_MS_LR)
for (gr = 0; gr < gfc.mode_gr; gr++) {
qupvt.reduce_side(targ_bits[gr], ms_ener_ratio[gr], mean_bits
* gfc.channels_out,
LameInternalFlags.MAX_BITS_PER_GRANULE);
}
/*
* sum target bits
*/
totbits = 0;
for (gr = 0; gr < gfc.mode_gr; gr++) {
for (ch = 0; ch < gfc.channels_out; ch++) {
if (targ_bits[gr][ch] > LameInternalFlags.MAX_BITS_PER_CHANNEL)
targ_bits[gr][ch] = LameInternalFlags.MAX_BITS_PER_CHANNEL;
totbits += targ_bits[gr][ch];
}
}
/*
* repartion target bits if needed
*/
if (totbits > max_frame_bits[0]) {
for (gr = 0; gr < gfc.mode_gr; gr++) {
for (ch = 0; ch < gfc.channels_out; ch++) {
targ_bits[gr][ch] *= max_frame_bits[0];
targ_bits[gr][ch] /= totbits;
}
}
}
};
}
var Quantize_1 = Quantize$1;
/*
* bit reservoir source file
*
* Copyright (c) 1999-2000 Mark Taylor
*
* 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 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
* Library 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.
*/
/* $Id: Reservoir.java,v 1.9 2011/05/24 20:48:06 kenchis Exp $ */
//package mp3;
/**
* ResvFrameBegin:* Called (repeatedly) at the beginning of a frame. Updates the maximum size of * the reservoir, and checks to make sure main_data_begin was set properly by * the formatter
* Background information: * * This is the original text from the ISO standard. Because of sooo many bugs * and irritations correcting comments are added in brackets []. A '^W' means * you should remove the last word. * *
* 1. The following rule can be used to calculate the maximum * number of bits used for one granule [^W frame]:*/ var assert$23 = common.assert; function Reservoir$1() { var bs; this.setModules = function(_bs) { bs = _bs; }; this.ResvFrameBegin = function(gfp, mean_bits) { var gfc = gfp.internal_flags; var maxmp3buf; var l3_side = gfc.l3_side; var frameLength = bs.getframebits(gfp); mean_bits.bits = (frameLength - gfc.sideinfo_len * 8) / gfc.mode_gr; /** *
* At the highest possible bitrate of Layer III (320 kbps * per stereo signal [^W^W^W], 48 kHz) the frames must be of * [^W^W^W are designed to have] constant length, i.e. * one buffer [^W^W the frame] length is:
* * 320 kbps * 1152/48 kHz = 7680 bit = 960 byte * * This value is used as the maximum buffer per channel [^W^W] at * lower bitrates [than 320 kbps]. At 64 kbps mono or 128 kbps * stereo the main granule length is 64 kbps * 576/48 kHz = 768 bit * [per granule and channel] at 48 kHz sampling frequency. * This means that there is a maximum deviation (short time buffer * [= reservoir]) of 7680 - 2*2*768 = 4608 bits is allowed at 64 kbps. * The actual deviation is equal to the number of bytes [with the * meaning of octets] denoted by the main_data_end offset pointer. * The actual maximum deviation is (2^9-1)*8 bit = 4088 bits * [for MPEG-1 and (2^8-1)*8 bit for MPEG-2, both are hard limits]. * ... The xchange of buffer bits between the left and right channel * is allowed without restrictions [exception: dual channel]. * Because of the [constructed] constraint on the buffer size * main_data_end is always set to 0 in the case of bit_rate_index==14, * i.e. data rate 320 kbps per stereo signal [^W^W^W]. In this case * all data are allocated between adjacent header [^W sync] words * [, i.e. there is no buffering at all]. *
* Meaning of the variables: * resvLimit: (0, 8, ..., 8*255 (MPEG-2), 8*511 (MPEG-1)) * Number of bits can be stored in previous frame(s) due to * counter size constaints * maxmp3buf: ( ??? ... 8*1951 (MPEG-1 and 2), 8*2047 (MPEG-2.5)) * Number of bits allowed to encode one frame (you can take 8*511 bit * from the bit reservoir and at most 8*1440 bit from the current * frame (320 kbps, 32 kHz), so 8*1951 bit is the largest possible * value for MPEG-1 and -2) * * maximum allowed granule/channel size times 4 = 8*2047 bits., * so this is the absolute maximum supported by the format. * * * fullFrameBits: maximum number of bits available for encoding * the current frame. * * mean_bits: target number of bits per granule. * * frameLength: * * gfc.ResvMax: maximum allowed reservoir * * gfc.ResvSize: current reservoir size * * l3_side.resvDrain_pre: * ancillary data to be added to previous frame: * (only usefull in VBR modes if it is possible to have * maxmp3buf < fullFrameBits)). Currently disabled, * see #define NEW_DRAIN * 2010-02-13: RH now enabled, it seems to be needed for CBR too, * as there exists one example, where the FhG decoder * can't decode a -b320 CBR file anymore. * * l3_side.resvDrain_post: * ancillary data to be added to this frame: * **/ /* main_data_begin has 9 bits in MPEG-1, 8 bits MPEG-2 */ var resvLimit = (8 * 256) * gfc.mode_gr - 8; /* * maximum allowed frame size. dont use more than this number of bits, * even if the frame has the space for them: */ if (gfp.brate > 320) { /* in freeformat the buffer is constant */ maxmp3buf = 8 * ((int) ((gfp.brate * 1000) / (gfp.out_samplerate / 1152) / 8 + .5)); } else { /* * all mp3 decoders should have enough buffer to handle this value: * size of a 320kbps 32kHz frame */ maxmp3buf = 8 * 1440; /* * Bouvigne suggests this more lax interpretation of the ISO doc * instead of using 8*960. */ if (gfp.strict_ISO) { maxmp3buf = 8 * ((int) (320000 / (gfp.out_samplerate / 1152) / 8 + .5)); } } gfc.ResvMax = maxmp3buf - frameLength; if (gfc.ResvMax > resvLimit) gfc.ResvMax = resvLimit; if (gfc.ResvMax < 0 || gfp.disable_reservoir) gfc.ResvMax = 0; var fullFrameBits = mean_bits.bits * gfc.mode_gr + Math.min(gfc.ResvSize, gfc.ResvMax); if (fullFrameBits > maxmp3buf) fullFrameBits = maxmp3buf; assert$23 (0 == gfc.ResvMax % 8); assert$23 (gfc.ResvMax >= 0); l3_side.resvDrain_pre = 0; // frame analyzer code if (gfc.pinfo != null) { /* * expected bits per channel per granule [is this also right for * mono/stereo, MPEG-1/2 ?] */ gfc.pinfo.mean_bits = mean_bits.bits / 2; gfc.pinfo.resvsize = gfc.ResvSize; } return fullFrameBits; }; /** * returns targ_bits: target number of bits to use for 1 granule
* extra_bits: amount extra available from reservoir
* Mark Taylor 4/99 */ this.ResvMaxBits = function(gfp, mean_bits, targ_bits, cbr) { var gfc = gfp.internal_flags; var add_bits; var ResvSize = gfc.ResvSize, ResvMax = gfc.ResvMax; /* compensate the saved bits used in the 1st granule */ if (cbr != 0) ResvSize += mean_bits; if ((gfc.substep_shaping & 1) != 0) ResvMax *= 0.9; targ_bits.bits = mean_bits; /* extra bits if the reservoir is almost full */ if (ResvSize * 10 > ResvMax * 9) { add_bits = ResvSize - (ResvMax * 9) / 10; targ_bits.bits += add_bits; gfc.substep_shaping |= 0x80; } else { add_bits = 0; gfc.substep_shaping &= 0x7f; /* * build up reservoir. this builds the reservoir a little slower * than FhG. It could simple be mean_bits/15, but this was rigged to * always produce 100 (the old value) at 128kbs */ if (!gfp.disable_reservoir && 0 == (gfc.substep_shaping & 1)) targ_bits.bits -= .1 * mean_bits; } /* amount from the reservoir we are allowed to use. ISO says 6/10 */ var extra_bits = (ResvSize < (gfc.ResvMax * 6) / 10 ? ResvSize : (gfc.ResvMax * 6) / 10); extra_bits -= add_bits; if (extra_bits < 0) extra_bits = 0; return extra_bits; }; /** * Called after a granule's bit allocation. Readjusts the size of the * reservoir to reflect the granule's usage. */ this.ResvAdjust = function(gfc, gi) { gfc.ResvSize -= gi.part2_3_length + gi.part2_length; }; /** * Called after all granules in a frame have been allocated. Makes sure that * the reservoir size is within limits, possibly by adding stuffing bits. */ this.ResvFrameEnd = function(gfc, mean_bits) { var over_bits; var l3_side = gfc.l3_side; gfc.ResvSize += mean_bits * gfc.mode_gr; var stuffingBits = 0; l3_side.resvDrain_post = 0; l3_side.resvDrain_pre = 0; /* we must be byte aligned */ if ((over_bits = gfc.ResvSize % 8) != 0) stuffingBits += over_bits; over_bits = (gfc.ResvSize - stuffingBits) - gfc.ResvMax; if (over_bits > 0) { assert$23 (0 == over_bits % 8); assert$23 (over_bits >= 0); stuffingBits += over_bits; } /* * NOTE: enabling the NEW_DRAIN code fixes some problems with FhG * decoder shipped with MS Windows operating systems. Using this, it is * even possible to use Gabriel's lax buffer consideration again, which * assumes, any decoder should have a buffer large enough for a 320 kbps * frame at 32 kHz sample rate. * * old drain code: lame -b320 BlackBird.wav --. does not play with * GraphEdit.exe using FhG decoder V1.5 Build 50 * * new drain code: lame -b320 BlackBird.wav --. plays fine with * GraphEdit.exe using FhG decoder V1.5 Build 50 * * Robert Hegemann, 2010-02-13. */ /* * drain as many bits as possible into previous frame ancillary data In * particular, in VBR mode ResvMax may have changed, and we have to make * sure main_data_begin does not create a reservoir bigger than ResvMax * mt 4/00 */ { var mdb_bytes = Math.min(l3_side.main_data_begin * 8, stuffingBits) / 8; l3_side.resvDrain_pre += 8 * mdb_bytes; stuffingBits -= 8 * mdb_bytes; gfc.ResvSize -= 8 * mdb_bytes; l3_side.main_data_begin -= mdb_bytes; } /* drain the rest into this frames ancillary data */ l3_side.resvDrain_post += stuffingBits; gfc.ResvSize -= stuffingBits; }; } var Reservoir_1 = Reservoir$1; function Version() { /** * URL for the LAME website. */ var LAME_URL = "http://www.mp3dev.org/"; /** * Major version number. */ var LAME_MAJOR_VERSION = 3; /** * Minor version number. */ var LAME_MINOR_VERSION = 98; /** * Patch level. */ var LAME_PATCH_VERSION = 4; /** * Major version number. */ var PSY_MAJOR_VERSION = 0; /** * Minor version number. */ var PSY_MINOR_VERSION = 93; /** * A string which describes the version of LAME. * * @return string which describes the version of LAME */ this.getLameVersion = function () { // primary to write screen reports return (LAME_MAJOR_VERSION + "." + LAME_MINOR_VERSION + "." + LAME_PATCH_VERSION); }; /** * The short version of the LAME version string. * * @return short version of the LAME version string */ this.getLameShortVersion = function () { // Adding date and time to version string makes it harder for output // validation return (LAME_MAJOR_VERSION + "." + LAME_MINOR_VERSION + "." + LAME_PATCH_VERSION); }; /** * The shortest version of the LAME version string. * * @return shortest version of the LAME version string */ this.getLameVeryShortVersion = function () { // Adding date and time to version string makes it harder for output return ("LAME" + LAME_MAJOR_VERSION + "." + LAME_MINOR_VERSION + "r"); }; /** * String which describes the version of GPSYCHO * * @return string which describes the version of GPSYCHO */ this.getPsyVersion = function () { return (PSY_MAJOR_VERSION + "." + PSY_MINOR_VERSION); }; /** * String which is a URL for the LAME website. * * @return string which is a URL for the LAME website */ this.getLameUrl = function () { return LAME_URL; }; /** * Quite useless for a java version, however we are compatible ;-) * * @return "32bits" */ this.getLameOsBitness = function () { return "32bits"; }; } var Version_1 = Version; var System$22 = common.System; var VbrMode$22 = common.VbrMode; var ShortBlock$22 = common.ShortBlock; var Arrays$22 = common.Arrays; var new_byte$22 = common.new_byte; var assert$24 = common.assert; /** * A Vbr header may be present in the ancillary data field of the first frame of * an mp3 bitstream
* The Vbr header (optionally) contains *
-
*
- frames total number of audio frames in the bitstream *
- bytes total number of bytes in the bitstream *
- toc table of contents *
* The ith entry determines the seek point for i-percent duration.
* seek point in bytes = (toc[i]/256.0) * total_bitstream_bytes
* e.g. half duration seek point = (toc[50]/256.0) * total_bitstream_bytes */ VBRTag.NUMTOCENTRIES = 100; VBRTag.MAXFRAMESIZE = 2880; function VBRTag() { var lame; var bs; var v; this.setModules = function (_lame, _bs, _v) { lame = _lame; bs = _bs; v = _v; }; var FRAMES_FLAG = 0x0001; var BYTES_FLAG = 0x0002; var TOC_FLAG = 0x0004; var VBR_SCALE_FLAG = 0x0008; var NUMTOCENTRIES = VBRTag.NUMTOCENTRIES; /** * (0xB40) the max freeformat 640 32kHz framesize. */ var MAXFRAMESIZE = VBRTag.MAXFRAMESIZE; /** *
* 4 bytes for Header Tag
* 4 bytes for Header Flags
* 100 bytes for entry (toc)
* 4 bytes for frame size
* 4 bytes for stream size
* 4 bytes for VBR scale. a VBR quality indicator: 0=best 100=worst
* 20 bytes for LAME tag. for example, "LAME3.12 (beta 6)"
* ___________
* 140 bytes
*
*/
var VBRHEADERSIZE = (NUMTOCENTRIES + 4 + 4 + 4 + 4 + 4);
var LAMEHEADERSIZE = (VBRHEADERSIZE + 9 + 1 + 1 + 8
+ 1 + 1 + 3 + 1 + 1 + 2 + 4 + 2 + 2);
/**
* The size of the Xing header MPEG-1, bit rate in kbps.
*/
var XING_BITRATE1 = 128;
/**
* The size of the Xing header MPEG-2, bit rate in kbps.
*/
var XING_BITRATE2 = 64;
/**
* The size of the Xing header MPEG-2.5, bit rate in kbps.
*/
var XING_BITRATE25 = 32;
/**
* ISO-8859-1 charset for byte to string operations.
*/
var ISO_8859_1 = null; //Charset.forName("ISO-8859-1");
/**
* VBR header magic string.
*/
var VBRTag0 = "Xing";
/**
* VBR header magic string (VBR == VBRMode.vbr_off).
*/
var VBRTag1 = "Info";
/**
* Lookup table for fast CRC-16 computation. Uses the polynomial
* x^16+x^15+x^2+1
*/
var crc16Lookup = [0x0000, 0xC0C1, 0xC181, 0x0140,
0xC301, 0x03C0, 0x0280, 0xC241, 0xC601, 0x06C0, 0x0780, 0xC741,
0x0500, 0xC5C1, 0xC481, 0x0440, 0xCC01, 0x0CC0, 0x0D80, 0xCD41,
0x0F00, 0xCFC1, 0xCE81, 0x0E40, 0x0A00, 0xCAC1, 0xCB81, 0x0B40,
0xC901, 0x09C0, 0x0880, 0xC841, 0xD801, 0x18C0, 0x1980, 0xD941,
0x1B00, 0xDBC1, 0xDA81, 0x1A40, 0x1E00, 0xDEC1, 0xDF81, 0x1F40,
0xDD01, 0x1DC0, 0x1C80, 0xDC41, 0x1400, 0xD4C1, 0xD581, 0x1540,
0xD701, 0x17C0, 0x1680, 0xD641, 0xD201, 0x12C0, 0x1380, 0xD341,
0x1100, 0xD1C1, 0xD081, 0x1040, 0xF001, 0x30C0, 0x3180, 0xF141,
0x3300, 0xF3C1, 0xF281, 0x3240, 0x3600, 0xF6C1, 0xF781, 0x3740,
0xF501, 0x35C0, 0x3480, 0xF441, 0x3C00, 0xFCC1, 0xFD81, 0x3D40,
0xFF01, 0x3FC0, 0x3E80, 0xFE41, 0xFA01, 0x3AC0, 0x3B80, 0xFB41,
0x3900, 0xF9C1, 0xF881, 0x3840, 0x2800, 0xE8C1, 0xE981, 0x2940,
0xEB01, 0x2BC0, 0x2A80, 0xEA41, 0xEE01, 0x2EC0, 0x2F80, 0xEF41,
0x2D00, 0xEDC1, 0xEC81, 0x2C40, 0xE401, 0x24C0, 0x2580, 0xE541,
0x2700, 0xE7C1, 0xE681, 0x2640, 0x2200, 0xE2C1, 0xE381, 0x2340,
0xE101, 0x21C0, 0x2080, 0xE041, 0xA001, 0x60C0, 0x6180, 0xA141,
0x6300, 0xA3C1, 0xA281, 0x6240, 0x6600, 0xA6C1, 0xA781, 0x6740,
0xA501, 0x65C0, 0x6480, 0xA441, 0x6C00, 0xACC1, 0xAD81, 0x6D40,
0xAF01, 0x6FC0, 0x6E80, 0xAE41, 0xAA01, 0x6AC0, 0x6B80, 0xAB41,
0x6900, 0xA9C1, 0xA881, 0x6840, 0x7800, 0xB8C1, 0xB981, 0x7940,
0xBB01, 0x7BC0, 0x7A80, 0xBA41, 0xBE01, 0x7EC0, 0x7F80, 0xBF41,
0x7D00, 0xBDC1, 0xBC81, 0x7C40, 0xB401, 0x74C0, 0x7580, 0xB541,
0x7700, 0xB7C1, 0xB681, 0x7640, 0x7200, 0xB2C1, 0xB381, 0x7340,
0xB101, 0x71C0, 0x7080, 0xB041, 0x5000, 0x90C1, 0x9181, 0x5140,
0x9301, 0x53C0, 0x5280, 0x9241, 0x9601, 0x56C0, 0x5780, 0x9741,
0x5500, 0x95C1, 0x9481, 0x5440, 0x9C01, 0x5CC0, 0x5D80, 0x9D41,
0x5F00, 0x9FC1, 0x9E81, 0x5E40, 0x5A00, 0x9AC1, 0x9B81, 0x5B40,
0x9901, 0x59C0, 0x5880, 0x9841, 0x8801, 0x48C0, 0x4980, 0x8941,
0x4B00, 0x8BC1, 0x8A81, 0x4A40, 0x4E00, 0x8EC1, 0x8F81, 0x4F40,
0x8D01, 0x4DC0, 0x4C80, 0x8C41, 0x4400, 0x84C1, 0x8581, 0x4540,
0x8701, 0x47C0, 0x4680, 0x8641, 0x8201, 0x42C0, 0x4380, 0x8341,
0x4100, 0x81C1, 0x8081, 0x4040];
/***********************************************************************
* Robert Hegemann 2001-01-17
***********************************************************************/
function addVbr(v, bitrate) {
v.nVbrNumFrames++;
v.sum += bitrate;
v.seen++;
if (v.seen < v.want) {
return;
}
if (v.pos < v.size) {
v.bag[v.pos] = v.sum;
v.pos++;
v.seen = 0;
}
if (v.pos == v.size) {
for (var i = 1; i < v.size; i += 2) {
v.bag[i / 2] = v.bag[i];
}
v.want *= 2;
v.pos /= 2;
}
}
function xingSeekTable(v, t) {
if (v.pos <= 0)
return;
for (var i = 1; i < NUMTOCENTRIES; ++i) {
var j = i / NUMTOCENTRIES, act, sum;
var indx = 0 | (Math.floor(j * v.pos));
if (indx > v.pos - 1)
indx = v.pos - 1;
act = v.bag[indx];
sum = v.sum;
var seek_point = 0 | (256. * act / sum);
if (seek_point > 255)
seek_point = 255;
t[i] = 0xff & seek_point;
}
}
/**
* Add VBR entry, used to fill the VBR TOC entries.
*
* @param gfp
* global flags
*/
this.addVbrFrame = function (gfp) {
var gfc = gfp.internal_flags;
var kbps = Tables.bitrate_table[gfp.version][gfc.bitrate_index];
assert$24(gfc.VBR_seek_table.bag != null);
addVbr(gfc.VBR_seek_table, kbps);
};
/**
* Read big endian integer (4-bytes) from header.
*
* @param buf
* header containing the integer
* @param bufPos
* offset into the header
* @return extracted integer
*/
function extractInteger(buf, bufPos) {
var x = buf[bufPos + 0] & 0xff;
x <<= 8;
x |= buf[bufPos + 1] & 0xff;
x <<= 8;
x |= buf[bufPos + 2] & 0xff;
x <<= 8;
x |= buf[bufPos + 3] & 0xff;
return x;
}
/**
* Write big endian integer (4-bytes) in the header.
*
* @param buf
* header to write the integer into
* @param bufPos
* offset into the header
* @param value
* integer value to write
*/
function createInteger(buf, bufPos, value) {
buf[bufPos + 0] = 0xff & ((value >> 24) & 0xff);
buf[bufPos + 1] = 0xff & ((value >> 16) & 0xff);
buf[bufPos + 2] = 0xff & ((value >> 8) & 0xff);
buf[bufPos + 3] = 0xff & (value & 0xff);
}
/**
* Write big endian short (2-bytes) in the header.
*
* @param buf
* header to write the integer into
* @param bufPos
* offset into the header
* @param value
* integer value to write
*/
function createShort(buf, bufPos, value) {
buf[bufPos + 0] = 0xff & ((value >> 8) & 0xff);
buf[bufPos + 1] = 0xff & (value & 0xff);
}
/**
* Check for magic strings (Xing/Info).
*
* @param buf
* header to check
* @param bufPos
* header offset to check
* @return magic string found
*/
function isVbrTag(buf, bufPos) {
return new String(buf, bufPos, VBRTag0.length(), ISO_8859_1)
.equals(VBRTag0)
|| new String(buf, bufPos, VBRTag1.length(), ISO_8859_1)
.equals(VBRTag1);
}
function shiftInBitsValue(x, n, v) {
return 0xff & ((x << n) | (v & ~(-1 << n)));
}
/**
* Construct the MP3 header using the settings of the global flags.
*
* 
*
* @param gfp
* global flags
* @param buffer
* header
*/
function setLameTagFrameHeader(gfp, buffer) {
var gfc = gfp.internal_flags;
// MP3 Sync Word
buffer[0] = shiftInBitsValue(buffer[0], 8, 0xff);
buffer[1] = shiftInBitsValue(buffer[1], 3, 7);
buffer[1] = shiftInBitsValue(buffer[1], 1,
(gfp.out_samplerate < 16000) ? 0 : 1);
// Version
buffer[1] = shiftInBitsValue(buffer[1], 1, gfp.version);
// 01 == Layer 3
buffer[1] = shiftInBitsValue(buffer[1], 2, 4 - 3);
// Error protection
buffer[1] = shiftInBitsValue(buffer[1], 1, (!gfp.error_protection) ? 1
: 0);
// Bit rate
buffer[2] = shiftInBitsValue(buffer[2], 4, gfc.bitrate_index);
// Frequency
buffer[2] = shiftInBitsValue(buffer[2], 2, gfc.samplerate_index);
// Pad. Bit
buffer[2] = shiftInBitsValue(buffer[2], 1, 0);
// Priv. Bit
buffer[2] = shiftInBitsValue(buffer[2], 1, gfp.extension);
// Mode
buffer[3] = shiftInBitsValue(buffer[3], 2, gfp.mode.ordinal());
// Mode extension (Used with Joint Stereo)
buffer[3] = shiftInBitsValue(buffer[3], 2, gfc.mode_ext);
// Copy
buffer[3] = shiftInBitsValue(buffer[3], 1, gfp.copyright);
// Original
buffer[3] = shiftInBitsValue(buffer[3], 1, gfp.original);
// Emphasis
buffer[3] = shiftInBitsValue(buffer[3], 2, gfp.emphasis);
/* the default VBR header. 48 kbps layer III, no padding, no crc */
/* but sampling freq, mode and copyright/copy protection taken */
/* from first valid frame */
buffer[0] = 0xff;
var abyte = 0xff & (buffer[1] & 0xf1);
var bitrate;
if (1 == gfp.version) {
bitrate = XING_BITRATE1;
} else {
if (gfp.out_samplerate < 16000)
bitrate = XING_BITRATE25;
else
bitrate = XING_BITRATE2;
}
if (gfp.VBR == VbrMode$22.vbr_off)
bitrate = gfp.brate;
var bbyte;
if (gfp.free_format)
bbyte = 0x00;
else
bbyte = 0xff & (16 * lame.BitrateIndex(bitrate, gfp.version,
gfp.out_samplerate));
/*
* Use as much of the info from the real frames in the Xing header:
* samplerate, channels, crc, etc...
*/
if (gfp.version == 1) {
/* MPEG1 */
buffer[1] = 0xff & (abyte | 0x0a);
/* was 0x0b; */
abyte = 0xff & (buffer[2] & 0x0d);
/* AF keep also private bit */
buffer[2] = 0xff & (bbyte | abyte);
/* 64kbs MPEG1 frame */
} else {
/* MPEG2 */
buffer[1] = 0xff & (abyte | 0x02);
/* was 0x03; */
abyte = 0xff & (buffer[2] & 0x0d);
/* AF keep also private bit */
buffer[2] = 0xff & (bbyte | abyte);
/* 64kbs MPEG2 frame */
}
}
/**
* Get VBR tag information
*
* @param buf
* header to analyze
* @param bufPos
* offset into the header
* @return VBR tag data
*/
this.getVbrTag = function (buf) {
var pTagData = new VBRTagData();
var bufPos = 0;
/* get Vbr header data */
pTagData.flags = 0;
/* get selected MPEG header data */
var hId = (buf[bufPos + 1] >> 3) & 1;
var hSrIndex = (buf[bufPos + 2] >> 2) & 3;
var hMode = (buf[bufPos + 3] >> 6) & 3;
var hBitrate = ((buf[bufPos + 2] >> 4) & 0xf);
hBitrate = Tables.bitrate_table[hId][hBitrate];
/* check for FFE syncword */
if ((buf[bufPos + 1] >> 4) == 0xE)
pTagData.samprate = Tables.samplerate_table[2][hSrIndex];
else
pTagData.samprate = Tables.samplerate_table[hId][hSrIndex];
/* determine offset of header */
if (hId != 0) {
/* mpeg1 */
if (hMode != 3)
bufPos += (32 + 4);
else
bufPos += (17 + 4);
} else {
/* mpeg2 */
if (hMode != 3)
bufPos += (17 + 4);
else
bufPos += (9 + 4);
}
if (!isVbrTag(buf, bufPos))
return null;
bufPos += 4;
pTagData.hId = hId;
/* get flags */
var head_flags = pTagData.flags = extractInteger(buf, bufPos);
bufPos += 4;
if ((head_flags & FRAMES_FLAG) != 0) {
pTagData.frames = extractInteger(buf, bufPos);
bufPos += 4;
}
if ((head_flags & BYTES_FLAG) != 0) {
pTagData.bytes = extractInteger(buf, bufPos);
bufPos += 4;
}
if ((head_flags & TOC_FLAG) != 0) {
if (pTagData.toc != null) {
for (var i = 0; i < NUMTOCENTRIES; i++)
pTagData.toc[i] = buf[bufPos + i];
}
bufPos += NUMTOCENTRIES;
}
pTagData.vbrScale = -1;
if ((head_flags & VBR_SCALE_FLAG) != 0) {
pTagData.vbrScale = extractInteger(buf, bufPos);
bufPos += 4;
}
pTagData.headersize = ((hId + 1) * 72000 * hBitrate)
/ pTagData.samprate;
bufPos += 21;
var encDelay = buf[bufPos + 0] << 4;
encDelay += buf[bufPos + 1] >> 4;
var encPadding = (buf[bufPos + 1] & 0x0F) << 8;
encPadding += buf[bufPos + 2] & 0xff;
/* check for reasonable values (this may be an old Xing header, */
/* not a INFO tag) */
if (encDelay < 0 || encDelay > 3000)
encDelay = -1;
if (encPadding < 0 || encPadding > 3000)
encPadding = -1;
pTagData.encDelay = encDelay;
pTagData.encPadding = encPadding;
/* success */
return pTagData;
};
/**
* Initializes the header
*
* @param gfp
* global flags
*/
this.InitVbrTag = function (gfp) {
var gfc = gfp.internal_flags;
/**
*
* Xing VBR pretends to be a 48kbs layer III frame. (at 44.1kHz).
* (at 48kHz they use 56kbs since 48kbs frame not big enough for
* table of contents)
* let's always embed Xing header inside a 64kbs layer III frame.
* this gives us enough room for a LAME version string too.
* size determined by sampling frequency (MPEG1)
* 32kHz: 216 bytes@48kbs 288bytes@ 64kbs
* 44.1kHz: 156 bytes 208bytes@64kbs (+1 if padding = 1)
* 48kHz: 144 bytes 192
*
* MPEG 2 values are the same since the framesize and samplerate
* are each reduced by a factor of 2.
*
*/
var kbps_header;
if (1 == gfp.version) {
kbps_header = XING_BITRATE1;
} else {
if (gfp.out_samplerate < 16000)
kbps_header = XING_BITRATE25;
else
kbps_header = XING_BITRATE2;
}
if (gfp.VBR == VbrMode$22.vbr_off)
kbps_header = gfp.brate;
// make sure LAME Header fits into Frame
var totalFrameSize = ((gfp.version + 1) * 72000 * kbps_header)
/ gfp.out_samplerate;
var headerSize = (gfc.sideinfo_len + LAMEHEADERSIZE);
gfc.VBR_seek_table.TotalFrameSize = totalFrameSize;
if (totalFrameSize < headerSize || totalFrameSize > MAXFRAMESIZE) {
/* disable tag, it wont fit */
gfp.bWriteVbrTag = false;
return;
}
gfc.VBR_seek_table.nVbrNumFrames = 0;
gfc.VBR_seek_table.nBytesWritten = 0;
gfc.VBR_seek_table.sum = 0;
gfc.VBR_seek_table.seen = 0;
gfc.VBR_seek_table.want = 1;
gfc.VBR_seek_table.pos = 0;
if (gfc.VBR_seek_table.bag == null) {
gfc.VBR_seek_table.bag = new int[400];
gfc.VBR_seek_table.size = 400;
}
// write dummy VBR tag of all 0's into bitstream
var buffer = new_byte$22(MAXFRAMESIZE);
setLameTagFrameHeader(gfp, buffer);
var n = gfc.VBR_seek_table.TotalFrameSize;
for (var i = 0; i < n; ++i) {
bs.add_dummy_byte(gfp, buffer[i] & 0xff, 1);
}
};
/**
* Fast CRC-16 computation (uses table crc16Lookup).
*
* @param value
* @param crc
* @return
*/
function crcUpdateLookup(value, crc) {
var tmp = crc ^ value;
crc = (crc >> 8) ^ crc16Lookup[tmp & 0xff];
return crc;
}
this.updateMusicCRC = function (crc, buffer, bufferPos, size) {
for (var i = 0; i < size; ++i)
crc[0] = crcUpdateLookup(buffer[bufferPos + i], crc[0]);
};
/**
* Write LAME info: mini version + info on various switches used (Jonathan
* Dee 2001/08/31).
*
* @param gfp
* global flags
* @param musicLength
* music length
* @param streamBuffer
* pointer to output buffer
* @param streamBufferPos
* offset into the output buffer
* @param crc
* computation of CRC-16 of Lame Tag so far (starting at frame
* sync)
* @return number of bytes written to the stream
*/
function putLameVBR(gfp, musicLength, streamBuffer, streamBufferPos, crc) {
var gfc = gfp.internal_flags;
var bytesWritten = 0;
/* encoder delay */
var encDelay = gfp.encoder_delay;
/* encoder padding */
var encPadding = gfp.encoder_padding;
/* recall: gfp.VBR_q is for example set by the switch -V */
/* gfp.quality by -q, -h, -f, etc */
var quality = (100 - 10 * gfp.VBR_q - gfp.quality);
var version = v.getLameVeryShortVersion();
var vbr;
var revision = 0x00;
var revMethod;
// numbering different in vbr_mode vs. Lame tag
var vbrTypeTranslator = [1, 5, 3, 2, 4, 0, 3];
var lowpass = 0 | (((gfp.lowpassfreq / 100.0) + .5) > 255 ? 255
: (gfp.lowpassfreq / 100.0) + .5);
var peakSignalAmplitude = 0;
var radioReplayGain = 0;
var audiophileReplayGain = 0;
var noiseShaping = gfp.internal_flags.noise_shaping;
var stereoMode = 0;
var nonOptimal = 0;
var sourceFreq = 0;
var misc = 0;
var musicCRC = 0;
// psy model type: Gpsycho or NsPsytune
var expNPsyTune = (gfp.exp_nspsytune & 1) != 0;
var safeJoint = (gfp.exp_nspsytune & 2) != 0;
var noGapMore = false;
var noGapPrevious = false;
var noGapCount = gfp.internal_flags.nogap_total;
var noGapCurr = gfp.internal_flags.nogap_current;
// 4 bits
var athType = gfp.ATHtype;
var flags = 0;
// vbr modes
var abrBitrate;
switch (gfp.VBR) {
case vbr_abr:
abrBitrate = gfp.VBR_mean_bitrate_kbps;
break;
case vbr_off:
abrBitrate = gfp.brate;
break;
default:
abrBitrate = gfp.VBR_min_bitrate_kbps;
}
// revision and vbr method
if (gfp.VBR.ordinal() < vbrTypeTranslator.length)
vbr = vbrTypeTranslator[gfp.VBR.ordinal()];
else
vbr = 0x00; // unknown
revMethod = 0x10 * revision + vbr;
// ReplayGain
if (gfc.findReplayGain) {
if (gfc.RadioGain > 0x1FE)
gfc.RadioGain = 0x1FE;
if (gfc.RadioGain < -0x1FE)
gfc.RadioGain = -0x1FE;
// set name code
radioReplayGain = 0x2000;
// set originator code to `determined automatically'
radioReplayGain |= 0xC00;
if (gfc.RadioGain >= 0) {
// set gain adjustment
radioReplayGain |= gfc.RadioGain;
} else {
// set the sign bit
radioReplayGain |= 0x200;
// set gain adjustment
radioReplayGain |= -gfc.RadioGain;
}
}
// peak sample
if (gfc.findPeakSample)
peakSignalAmplitude = Math
.abs(0 | ((( gfc.PeakSample) / 32767.0) * Math.pow(2, 23) + .5));
// nogap
if (noGapCount != -1) {
if (noGapCurr > 0)
noGapPrevious = true;
if (noGapCurr < noGapCount - 1)
noGapMore = true;
}
// flags
flags = athType + ((expNPsyTune ? 1 : 0) << 4)
+ ((safeJoint ? 1 : 0) << 5) + ((noGapMore ? 1 : 0) << 6)
+ ((noGapPrevious ? 1 : 0) << 7);
if (quality < 0)
quality = 0;
// stereo mode field (Intensity stereo is not implemented)
switch (gfp.mode) {
case MONO:
stereoMode = 0;
break;
case STEREO:
stereoMode = 1;
break;
case DUAL_CHANNEL:
stereoMode = 2;
break;
case JOINT_STEREO:
if (gfp.force_ms)
stereoMode = 4;
else
stereoMode = 3;
break;
case NOT_SET:
//$FALL-THROUGH$
default:
stereoMode = 7;
break;
}
if (gfp.in_samplerate <= 32000)
sourceFreq = 0x00;
else if (gfp.in_samplerate == 48000)
sourceFreq = 0x02;
else if (gfp.in_samplerate > 48000)
sourceFreq = 0x03;
else {
// default is 44100Hz
sourceFreq = 0x01;
}
// Check if the user overrided the default LAME behavior with some
// nasty options
if (gfp.short_blocks == ShortBlock$22.short_block_forced
|| gfp.short_blocks == ShortBlock$22.short_block_dispensed
|| ((gfp.lowpassfreq == -1) && (gfp.highpassfreq == -1)) || /* "-k" */
(gfp.scale_left < gfp.scale_right)
|| (gfp.scale_left > gfp.scale_right)
|| (gfp.disable_reservoir && gfp.brate < 320) || gfp.noATH
|| gfp.ATHonly || (athType == 0) || gfp.in_samplerate <= 32000)
nonOptimal = 1;
misc = noiseShaping + (stereoMode << 2) + (nonOptimal << 5)
+ (sourceFreq << 6);
musicCRC = gfc.nMusicCRC;
// Write all this information into the stream
createInteger(streamBuffer, streamBufferPos + bytesWritten, quality);
bytesWritten += 4;
for (var j = 0; j < 9; j++) {
streamBuffer[streamBufferPos + bytesWritten + j] = 0xff & version .charAt(j);
}
bytesWritten += 9;
streamBuffer[streamBufferPos + bytesWritten] = 0xff & revMethod;
bytesWritten++;
streamBuffer[streamBufferPos + bytesWritten] = 0xff & lowpass;
bytesWritten++;
createInteger(streamBuffer, streamBufferPos + bytesWritten,
peakSignalAmplitude);
bytesWritten += 4;
createShort(streamBuffer, streamBufferPos + bytesWritten,
radioReplayGain);
bytesWritten += 2;
createShort(streamBuffer, streamBufferPos + bytesWritten,
audiophileReplayGain);
bytesWritten += 2;
streamBuffer[streamBufferPos + bytesWritten] = 0xff & flags;
bytesWritten++;
if (abrBitrate >= 255)
streamBuffer[streamBufferPos + bytesWritten] = 0xFF;
else
streamBuffer[streamBufferPos + bytesWritten] = 0xff & abrBitrate;
bytesWritten++;
streamBuffer[streamBufferPos + bytesWritten] = 0xff & (encDelay >> 4);
streamBuffer[streamBufferPos + bytesWritten + 1] = 0xff & ((encDelay << 4) + (encPadding >> 8));
streamBuffer[streamBufferPos + bytesWritten + 2] = 0xff & encPadding;
bytesWritten += 3;
streamBuffer[streamBufferPos + bytesWritten] = 0xff & misc;
bytesWritten++;
// unused in rev0
streamBuffer[streamBufferPos + bytesWritten++] = 0;
createShort(streamBuffer, streamBufferPos + bytesWritten, gfp.preset);
bytesWritten += 2;
createInteger(streamBuffer, streamBufferPos + bytesWritten, musicLength);
bytesWritten += 4;
createShort(streamBuffer, streamBufferPos + bytesWritten, musicCRC);
bytesWritten += 2;
// Calculate tag CRC.... must be done here, since it includes previous
// information
for (var i = 0; i < bytesWritten; i++)
crc = crcUpdateLookup(streamBuffer[streamBufferPos + i], crc);
createShort(streamBuffer, streamBufferPos + bytesWritten, crc);
bytesWritten += 2;
return bytesWritten;
}
function skipId3v2(fpStream) {
// seek to the beginning of the stream
fpStream.seek(0);
// read 10 bytes in case there's an ID3 version 2 header here
var id3v2Header = new_byte$22(10);
fpStream.readFully(id3v2Header);
/* does the stream begin with the ID3 version 2 file identifier? */
var id3v2TagSize;
if (!new String(id3v2Header, "ISO-8859-1").startsWith("ID3")) {
/*
* the tag size (minus the 10-byte header) is encoded into four
* bytes where the most significant bit is clear in each byte
*/
id3v2TagSize = (((id3v2Header[6] & 0x7f) << 21)
| ((id3v2Header[7] & 0x7f) << 14)
| ((id3v2Header[8] & 0x7f) << 7) | (id3v2Header[9] & 0x7f))
+ id3v2Header.length;
} else {
/* no ID3 version 2 tag in this stream */
id3v2TagSize = 0;
}
return id3v2TagSize;
}
this.getLameTagFrame = function (gfp, buffer) {
var gfc = gfp.internal_flags;
if (!gfp.bWriteVbrTag) {
return 0;
}
if (gfc.Class_ID != Lame.LAME_ID) {
return 0;
}
if (gfc.VBR_seek_table.pos <= 0) {
return 0;
}
if (buffer.length < gfc.VBR_seek_table.TotalFrameSize) {
return gfc.VBR_seek_table.TotalFrameSize;
}
Arrays$22.fill(buffer, 0, gfc.VBR_seek_table.TotalFrameSize, 0);
// 4 bytes frame header
setLameTagFrameHeader(gfp, buffer);
// Create TOC entries
var toc = new_byte$22(NUMTOCENTRIES);
if (gfp.free_format) {
for (var i = 1; i < NUMTOCENTRIES; ++i)
toc[i] = 0xff & (255 * i / 100);
} else {
xingSeekTable(gfc.VBR_seek_table, toc);
}
// Start writing the tag after the zero frame
var streamIndex = gfc.sideinfo_len;
/**
* Note: Xing header specifies that Xing data goes in the ancillary data
* with NO ERROR PROTECTION. If error protecton in enabled, the Xing
* data still starts at the same offset, and now it is in sideinfo data
* block, and thus will not decode correctly by non-Xing tag aware
* players
*/
if (gfp.error_protection)
streamIndex -= 2;
// Put Vbr tag
if (gfp.VBR == VbrMode$22.vbr_off) {
buffer[streamIndex++] = 0xff & VBRTag1.charAt(0);
buffer[streamIndex++] = 0xff & VBRTag1.charAt(1);
buffer[streamIndex++] = 0xff & VBRTag1.charAt(2);
buffer[streamIndex++] = 0xff & VBRTag1.charAt(3);
} else {
buffer[streamIndex++] = 0xff & VBRTag0.charAt(0);
buffer[streamIndex++] = 0xff & VBRTag0.charAt(1);
buffer[streamIndex++] = 0xff & VBRTag0.charAt(2);
buffer[streamIndex++] = 0xff & VBRTag0.charAt(3);
}
// Put header flags
createInteger(buffer, streamIndex, FRAMES_FLAG + BYTES_FLAG + TOC_FLAG
+ VBR_SCALE_FLAG);
streamIndex += 4;
// Put Total Number of frames
createInteger(buffer, streamIndex, gfc.VBR_seek_table.nVbrNumFrames);
streamIndex += 4;
// Put total audio stream size, including Xing/LAME Header
var streamSize = (gfc.VBR_seek_table.nBytesWritten + gfc.VBR_seek_table.TotalFrameSize);
createInteger(buffer, streamIndex, 0 | streamSize);
streamIndex += 4;
/* Put TOC */
System$22.arraycopy(toc, 0, buffer, streamIndex, toc.length);
streamIndex += toc.length;
if (gfp.error_protection) {
// (jo) error_protection: add crc16 information to header
bs.CRC_writeheader(gfc, buffer);
}
// work out CRC so far: initially crc = 0
var crc = 0x00;
for (var i = 0; i < streamIndex; i++)
crc = crcUpdateLookup(buffer[i], crc);
// Put LAME VBR info
streamIndex += putLameVBR(gfp, streamSize, buffer, streamIndex, crc);
return gfc.VBR_seek_table.TotalFrameSize;
};
/**
* Write final VBR tag to the file.
*
* @param gfp
* global flags
* @param stream
* stream to add the VBR tag to
* @return 0 (OK), -1 else
* @throws IOException
* I/O error
*/
this.putVbrTag = function (gfp, stream) {
var gfc = gfp.internal_flags;
if (gfc.VBR_seek_table.pos <= 0)
return -1;
// Seek to end of file
stream.seek(stream.length());
// Get file size, abort if file has zero length.
if (stream.length() == 0)
return -1;
// The VBR tag may NOT be located at the beginning of the stream. If an
// ID3 version 2 tag was added, then it must be skipped to write the VBR
// tag data.
var id3v2TagSize = skipId3v2(stream);
// Seek to the beginning of the stream
stream.seek(id3v2TagSize);
var buffer = new_byte$22(MAXFRAMESIZE);
var bytes = getLameTagFrame(gfp, buffer);
if (bytes > buffer.length) {
return -1;
}
if (bytes < 1) {
return 0;
}
// Put it all to disk again
stream.write(buffer, 0, bytes);
// success
return 0;
};
}
var VBRTag_1 = VBRTag;
var new_byte = common.new_byte;
var assert = common.assert;
Lame = Lame_1;
Presets = Presets_1;
GainAnalysis = GainAnalysis_1;
QuantizePVT = QuantizePVT_1;
Quantize = Quantize_1;
Takehiro = Takehiro_1;
Reservoir = Reservoir_1;
MPEGMode = MPEGMode_1;
BitStream = BitStream_1;
function GetAudio() {
var parse;
var mpg;
this.setModules = function (parse2, mpg2) {
parse = parse2;
mpg = mpg2;
};
}
function Parse() {
var ver;
var id3;
var pre;
this.setModules = function (ver2, id32, pre2) {
ver = ver2;
id3 = id32;
pre = pre2;
};
}
function MPGLib() {
}
function ID3Tag() {
var bits;
var ver;
this.setModules = function (_bits, _ver) {
bits = _bits;
ver = _ver;
};
}
function Mp3Encoder(channels, samplerate, kbps) {
if (arguments.length != 3) {
console.error('WARN: Mp3Encoder(channels, samplerate, kbps) not specified');
channels = 1;
samplerate = 44100;
kbps = 128;
}
var lame = new Lame();
var gaud = new GetAudio();
var ga = new GainAnalysis();
var bs = new BitStream();
var p = new Presets();
var qupvt = new QuantizePVT();
var qu = new Quantize();
var vbr = new VBRTag_1();
var ver = new Version_1();
var id3 = new ID3Tag();
var rv = new Reservoir();
var tak = new Takehiro();
var parse = new Parse();
var mpg = new MPGLib();
lame.setModules(ga, bs, p, qupvt, qu, vbr, ver, id3, mpg);
bs.setModules(ga, mpg, ver, vbr);
id3.setModules(bs, ver);
p.setModules(lame);
qu.setModules(bs, rv, qupvt, tak);
qupvt.setModules(tak, rv, lame.enc.psy);
rv.setModules(bs);
tak.setModules(qupvt);
vbr.setModules(lame, bs, ver);
gaud.setModules(parse, mpg);
parse.setModules(ver, id3, p);
var gfp = lame.lame_init();
gfp.num_channels = channels;
gfp.in_samplerate = samplerate;
gfp.brate = kbps;
gfp.mode = MPEGMode.STEREO;
gfp.quality = 3;
gfp.bWriteVbrTag = false;
gfp.disable_reservoir = true;
gfp.write_id3tag_automatic = false;
var retcode = lame.lame_init_params(gfp);
assert(0 == retcode);
var maxSamples = 1152;
var mp3buf_size = 0 | (1.25 * maxSamples + 7200);
var mp3buf = new_byte(mp3buf_size);
this.encodeBuffer = function (left, right) {
if (channels == 1) {
right = left;
}
assert(left.length == right.length);
if (left.length > maxSamples) {
maxSamples = left.length;
mp3buf_size = 0 | (1.25 * maxSamples + 7200);
mp3buf = new_byte(mp3buf_size);
}
var _sz = lame.lame_encode_buffer(gfp, left, right, left.length, mp3buf, 0, mp3buf_size);
return new Int8Array(mp3buf.subarray(0, _sz));
};
this.flush = function () {
var _sz = lame.lame_encode_flush(gfp, mp3buf, 0, mp3buf_size);
return new Int8Array(mp3buf.subarray(0, _sz));
};
}
function WavHeader() {
this.dataOffset = 0;
this.dataLen = 0;
this.channels = 0;
this.sampleRate = 0;
}
function fourccToInt(fourcc) {
return fourcc.charCodeAt(0) << 24 | fourcc.charCodeAt(1) << 16 | fourcc.charCodeAt(2) << 8 | fourcc.charCodeAt(3);
}
WavHeader.RIFF = fourccToInt("RIFF");
WavHeader.WAVE = fourccToInt("WAVE");
WavHeader.fmt_ = fourccToInt("fmt ");
WavHeader.data = fourccToInt("data");
WavHeader.readHeader = function (dataView) {
var w = new WavHeader();
var header = dataView.getUint32(0, false);
if (WavHeader.RIFF != header) {
return;
}
var fileLen = dataView.getUint32(4, true);
if (WavHeader.WAVE != dataView.getUint32(8, false)) {
return;
}
if (WavHeader.fmt_ != dataView.getUint32(12, false)) {
return;
}
var fmtLen = dataView.getUint32(16, true);
var pos = 16 + 4;
switch (fmtLen) {
case 16:
case 18:
w.channels = dataView.getUint16(pos + 2, true);
w.sampleRate = dataView.getUint32(pos + 4, true);
break;
default:
throw 'extended fmt chunk not implemented';
}
pos += fmtLen;
var data = WavHeader.data;
var len = 0;
while (data != header) {
header = dataView.getUint32(pos, false);
len = dataView.getUint32(pos + 4, true);
if (data == header) {
break;
}
pos += (len + 8);
}
w.dataLen = len;
w.dataOffset = pos + 8;
return w;
};
var Mp3Encoder_1 = Mp3Encoder;
var classCallCheck = function (instance, Constructor) {
if (!(instance instanceof Constructor)) {
throw new TypeError("Cannot call a class as a function");
}
};
var createClass = function () {
function defineProperties(target, props) {
for (var i = 0; i < props.length; i++) {
var descriptor = props[i];
descriptor.enumerable = descriptor.enumerable || false;
descriptor.configurable = true;
if ("value" in descriptor) descriptor.writable = true;
Object.defineProperty(target, descriptor.key, descriptor);
}
}
return function (Constructor, protoProps, staticProps) {
if (protoProps) defineProperties(Constructor.prototype, protoProps);
if (staticProps) defineProperties(Constructor, staticProps);
return Constructor;
};
}();
var Encoder = function () {
function Encoder(config) {
classCallCheck(this, Encoder);
this.config = {
sampleRate: 44100,
bitRate: 128
};
Object.assign(this.config, config);
this.mp3Encoder = new Mp3Encoder_1(1, this.config.sampleRate, this.config.bitRate);
// Audio is processed by frames of 1152 samples per audio channel
// http://lame.sourceforge.net/tech-FAQ.txt
this.maxSamples = 1152;
this.samplesMono = null;
this.clearBuffer();
}
/**
* Clear active buffer
*/
createClass(Encoder, [{
key: 'clearBuffer',
value: function clearBuffer() {
this.dataBuffer = [];
}
/**
* Append new audio buffer to current active buffer
* @param {Buffer} buffer
*/
}, {
key: 'appendToBuffer',
value: function appendToBuffer(buffer) {
this.dataBuffer.push(new Int8Array(buffer));
}
/**
* Float current data to 16 bits PCM
* @param {Float32Array} input
* @param {Int16Array} output
*/
}, {
key: 'floatTo16BitPCM',
value: function floatTo16BitPCM(input, output) {
for (var i = 0; i < input.length; i++) {
var s = Math.max(-1, Math.min(1, input[i]));
output[i] = s < 0 ? s * 0x8000 : s * 0x7FFF;
}
}
/**
* Convert buffer to proper format
* @param {Array} arrayBuffer
*/
}, {
key: 'convertBuffer',
value: function convertBuffer(arrayBuffer) {
var data = new Float32Array(arrayBuffer);
var out = new Int16Array(arrayBuffer.length);
this.floatTo16BitPCM(data, out);
return out;
}
/**
* Encode and append current buffer to dataBuffer
* @param {Array} arrayBuffer
*/
}, {
key: 'encode',
value: function encode(arrayBuffer) {
this.samplesMono = this.convertBuffer(arrayBuffer);
var remaining = this.samplesMono.length;
for (var i = 0; remaining >= 0; i += this.maxSamples) {
var left = this.samplesMono.subarray(i, i + this.maxSamples);
var mp3buffer = this.mp3Encoder.encodeBuffer(left);
this.appendToBuffer(mp3buffer);
remaining -= this.maxSamples;
}
}
/**
* Return full dataBuffer
*/
}, {
key: 'finish',
value: function finish() {
this.appendToBuffer(this.mp3Encoder.flush());
return this.dataBuffer;
}
}]);
return Encoder;
}();
var MicRecorder = function () {
function MicRecorder(config) {
classCallCheck(this, MicRecorder);
this.config = {
// 128 or 160 kbit/s – mid-range bitrate quality
bitRate: 128,
// There is a known issue with some macOS machines, where the recording
// will sometimes have a loud 'pop' or 'pop-click' sound. This flag
// prevents getting audio from the microphone a few milliseconds after
// the begining of the recording. It also helps to remove the mouse
// "click" sound from the output mp3 file.
startRecordingAt: 300,
deviceId: null
};
this.activeStream = null;
this.context = null;
this.microphone = null;
this.processor = null;
this.startTime = 0;
Object.assign(this.config, config);
}
/**
* Starts to listen for the microphone sound
* @param {MediaStream} stream
*/
createClass(MicRecorder, [{
key: 'addMicrophoneListener',
value: function addMicrophoneListener(stream) {
var _this = this;
this.activeStream = stream;
// This prevents the weird noise once you start listening to the microphone
this.timerToStart = setTimeout(function () {
delete _this.timerToStart;
}, this.config.startRecordingAt);
// Set up Web Audio API to process data from the media stream (microphone).
this.microphone = this.context.createMediaStreamSource(stream);
// Settings a bufferSize of 0 instructs the browser to choose the best bufferSize
this.processor = this.context.createScriptProcessor(0, 1, 1);
// Add all buffers from LAME into an array.
this.processor.onaudioprocess = function (event) {
if (_this.timerToStart) {
return;
}
// Send microphone data to LAME for MP3 encoding while recording.
_this.lameEncoder.encode(event.inputBuffer.getChannelData(0));
};
// Begin retrieving microphone data.
this.microphone.connect(this.processor);
this.processor.connect(this.context.destination);
}
}, {
key: 'stop',
/**
* Disconnect microphone, processor and remove activeStream
*/
value: function stop() {
if (this.processor && this.microphone) {
// Clean up the Web Audio API resources.
this.microphone.disconnect();
this.processor.disconnect();
// If all references using this.context are destroyed, context is closed
// automatically. DOMException is fired when trying to close again
if (this.context && this.context.state !== 'closed') {
this.context.close();
}
this.processor.onaudioprocess = null;
// Stop all audio tracks. Also, removes recording icon from chrome tab
this.activeStream.getAudioTracks().forEach(function (track) {
return track.stop();
});
}
return this;
}
}, {
key: 'start',
/**
* Requests access to the microphone and start recording
* @return Promise
*/
value: function start() {
var _this2 = this;
var AudioContext = window.AudioContext || window.webkitAudioContext;
this.context = new AudioContext();
this.config.sampleRate = this.context.sampleRate;
this.lameEncoder = new Encoder(this.config);
var audio = this.config.deviceId ? { deviceId: { exact: this.config.deviceId } } : true;
return new Promise(function (resolve, reject) {
navigator.mediaDevices.getUserMedia({ audio: audio }).then(function (stream) {
_this2.addMicrophoneListener(stream);
resolve(stream);
}).catch(function (err) {
reject(err);
});
});
}
}, {
key: 'getMp3',
/**
* Return Mp3 Buffer and Blob with type mp3
* @return {Promise}
*/
value: function getMp3() {
var _this3 = this;
var finalBuffer = this.lameEncoder.finish();
return new Promise(function (resolve, reject) {
if (finalBuffer.length === 0) {
reject(new Error('No buffer to send'));
} else {
resolve([finalBuffer, new Blob(finalBuffer, { type: 'audio/mp3' })]);
_this3.lameEncoder.clearBuffer();
}
});
}
}]);
return MicRecorder;
}();
return MicRecorder;
})));
//# sourceMappingURL=index.js.map