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1.1 root 1: /*
1.1.1.5 root 2: Hatari - sound.c
3:
4: This file is distributed under the GNU Public License, version 2 or at
5: your option any later version. Read the file gpl.txt for details.
1.1 root 6:
1.1.1.11 root 7: This is where we emulate the YM2149. To obtain cycle-accurate timing we store
8: the current cycle time and this is incremented during each instruction.
9: When a write occurs in the PSG registers we take the difference in time and
10: generate this many samples using the previous register data.
11: Now we begin again from this point. To make sure we always have 1/50th of
12: samples we update the buffer generation every 1/50th second, just in case no
13: write took place on the PSG.
14: NOTE: If the emulator runs slower than 50fps it cannot update the buffers,
15: but the sound thread still needs some data to play to prevent a 'pop'. The
16: ONLY feasible solution is to play the same buffer again. I have tried all
17: kinds of methods to play the sound 'slower', but this produces un-even timing
18: in the sound and it simply doesn't work. If the emulator cannot keep the
19: speed, users will have to turn off the sound - that's it.
1.1.1.12 root 20:
21: The new version of the sound core uses some code/ideas from the following GPL projects :
1.1.1.15 root 22: - tone and noise steps computations are from StSound 1.2 by Arnaud Carré (Leonard/Oxygene)
1.1.1.12 root 23: - 5 bits volume table and 16*16*16 combinations of all volume are from Sc68 by Benjamin Gerard
24: - 4 bits to 5 bits volume interpolation from 16*16*16 to 32*32*32 from YM blep synthesis by Antti Lankila
25:
1.1.1.16 root 26: Special case for per==0 : as measured on a real STF, when tone/noise/env's per==0, we get
27: the same sound output as when per==1.
28:
29:
1.1 root 30: */
1.1.1.12 root 31:
32: /* 2008/05/05 [NP] Fix case where period is 0 for noise, sound or envelope. */
33: /* In that case, a real ST sounds as if period was in fact 1. */
34: /* (fix buggy sound replay in ESwat that set volume<0 and trigger */
35: /* a badly initialised envelope with envper=0). */
36: /* 2008/07/27 [NP] Better separation between accesses to the YM hardware registers */
37: /* and the sound rendering routines. Use Sound_WriteReg() to pass */
38: /* all writes to the sound rendering functions. This allows to */
39: /* have sound.c independant of psg.c (to ease replacement of */
40: /* sound.c by another rendering method). */
41: /* 2008/08/02 [NP] Initial convert of Ym2149Ex.cpp from C++ to C. */
42: /* Remove unused part of the code (StSound specific). */
43: /* 2008/08/09 [NP] Complete integration of StSound routines into sound.c */
44: /* Set EnvPer=3 if EnvPer<3 (ESwat buggy replay). */
45: /* 2008/08/13 [NP] StSound was generating samples in the range 0-32767, instead */
46: /* of really signed samples between -32768 and 32767, which could */
47: /* give incorrect results in many case. */
48: /* 2008/09/06 [NP] Use sc68 volumes table for a more accurate mixing of the voices */
49: /* All volumes are converted to 5 bits and the table contains */
50: /* 32*32*32 values. Samples are signed and centered to get the */
51: /* biggest amplitude possible. */
52: /* Faster mixing routines for tone+volume+envelope (don't use */
53: /* StSound's version anymore, it gave problem with some GCC). */
54: /* 2008/09/17 [NP] Add ym_normalise_5bit_table to normalise the 32*32*32 table and */
55: /* to optionally center 16 bit signed sample. */
56: /* Possibility to mix volumes using a table measured on ST or a */
57: /* linear mean of the 3 channels' volume. */
58: /* Default mixing set to YM_LINEAR_MIXING. */
59: /* 2008/10/14 [NP] Full support for 5 bits volumes : envelopes are generated with */
60: /* 32 volumes per pattern as on a real YM-2149. Fixed volumes */
61: /* on 4 bits are converted to their 5 bits equivalent. This should */
62: /* give the maximum accuracy possible when computing volumes. */
63: /* New version of Ym2149_EnvStepCompute to handle 5 bits volumes. */
64: /* Function YM2149_EnvBuild to compute the 96 volumes that define */
65: /* a single envelope (32 initial volumes, then 64 repeated values).*/
66: /* 2008/10/26 [NP] Correctly save/restore all necessary variables in */
67: /* Sound_MemorySnapShot_Capture. */
68: /* 2008/11/23 [NP] Clean source, remove old sound core. */
1.1.1.17 root 69: /* 2011/11/03 [DS] Stereo DC filtering which accounts for DMA sound. */
1.1.1.12 root 70:
71:
72:
1.1.1.14 root 73: const char Sound_fileid[] = "Hatari sound.c : " __DATE__ " " __TIME__;
1.1.1.5 root 74:
75: #include <SDL_types.h>
1.1 root 76:
77: #include "main.h"
78: #include "audio.h"
1.1.1.10 root 79: #include "cycles.h"
1.1.1.15 root 80: #include "configuration.h"
1.1.1.9 root 81: #include "dmaSnd.h"
1.1.1.15 root 82: #include "crossbar.h"
1.1 root 83: #include "file.h"
1.1.1.15 root 84: #include "cycInt.h"
1.1.1.8 root 85: #include "log.h"
1.1 root 86: #include "memorySnapShot.h"
87: #include "psg.h"
88: #include "sound.h"
1.1.1.16 root 89: #include "screen.h"
1.1 root 90: #include "video.h"
91: #include "wavFormat.h"
92: #include "ymFormat.h"
1.1.1.15 root 93: #include "avi_record.h"
1.1.1.16 root 94: #include "clocks_timings.h"
1.1 root 95:
96:
97:
1.1.1.12 root 98: /*--------------------------------------------------------------*/
99: /* Definition of the possible envelopes shapes (using 5 bits) */
100: /*--------------------------------------------------------------*/
101:
102: #define ENV_GODOWN 0 /* 31 -> 0 */
103: #define ENV_GOUP 1 /* 0 -> 31 */
104: #define ENV_DOWN 2 /* 0 -> 0 */
105: #define ENV_UP 3 /* 31 -> 31 */
106:
107: /* To generate an envelope, we first use block 0, then we repeat blocks 1 and 2 */
108: static const int YmEnvDef[ 16 ][ 3 ] = {
109: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 0 \___ */
110: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 1 \___ */
111: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 2 \___ */
112: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 3 \___ */
113: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 4 /___ */
114: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 5 /___ */
115: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 6 /___ */
116: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 7 /___ */
117: { ENV_GODOWN, ENV_GODOWN, ENV_GODOWN } , /* 8 \\\\ */
118: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 9 \___ */
119: { ENV_GODOWN, ENV_GOUP, ENV_GODOWN } , /* A \/\/ */
120: { ENV_GODOWN, ENV_UP, ENV_UP } , /* B \--- */
121: { ENV_GOUP, ENV_GOUP, ENV_GOUP } , /* C //// */
122: { ENV_GOUP, ENV_UP, ENV_UP } , /* D /--- */
123: { ENV_GOUP, ENV_GODOWN, ENV_GOUP } , /* E /\/\ */
124: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* F /___ */
125: };
126:
127:
128: /* Buffer to store the 16 envelopes built from YmEnvDef */
129: static ymu16 YmEnvWaves[ 16 ][ 32 * 3 ]; /* 16 envelopes with 3 blocks of 32 volumes */
1.1.1.3 root 130:
1.1 root 131:
1.1.1.12 root 132:
133: /*--------------------------------------------------------------*/
134: /* Definition of the volumes tables (using 5 bits) and of the */
135: /* mixing parameters for the 3 voices. */
136: /*--------------------------------------------------------------*/
137:
138: /* Table of unsigned 5 bit D/A output level for 1 channel as measured on a real ST (expanded from 4 bits to 5 bits) */
139: /* Vol 0 should be 310 when measuread as a voltage, but we set it to 0 in order to have a volume=0 matching */
140: /* the 0 level of a 16 bits unsigned sample (no sound output) */
141: static const ymu16 ymout1c5bit[ 32 ] =
142: {
143: 0 /*310*/, 369, 438, 521, 619, 735, 874, 1039,
144: 1234, 1467, 1744, 2072, 2463, 2927, 3479, 4135,
145: 4914, 5841, 6942, 8250, 9806,11654,13851,16462,
146: 19565,23253,27636,32845,39037,46395,55141,65535
1.1 root 147: };
148:
1.1.1.12 root 149: /* Convert a constant 4 bits volume to the internal 5 bits value : */
1.1.1.18! root 150: /* volume5=volume4*2+1, except for volumes 0 and 1 which remain 0 and 1, */
1.1.1.12 root 151: /* in order to map [0,15] into [0,31] (O must remain 0, and 15 must give 31) */
1.1.1.18! root 152: static const ymu16 YmVolume4to5[ 16 ] = { 0,1,5,7,9,11,13,15,17,19,21,23,25,27,29,31 };
1.1 root 153:
1.1.1.12 root 154: /* Table of unsigned 4 bit D/A output level for 3 channels as measured on a real ST */
1.1.1.18! root 155: static ymu16 volumetable_original[16][16][16] =
1.1.1.12 root 156: #include "ym2149_fixed_vol.h"
1.1.1.5 root 157:
1.1.1.12 root 158: /* Corresponding table interpolated to 5 bit D/A output level (16 bits unsigned) */
1.1.1.18! root 159: static ymu16 ymout5_u16[32][32][32];
1.1.1.12 root 160:
161: /* Same table, after conversion to signed results (same pointer, with different type) */
162: static yms16 *ymout5 = (yms16 *)ymout5_u16;
163:
164:
165:
166: /*--------------------------------------------------------------*/
167: /* Other constants / macros */
168: /*--------------------------------------------------------------*/
169:
170: /* Number of generated samples per frame (eg. 44Khz=882) */
1.1.1.16 root 171: #define SAMPLES_PER_FRAME (nAudioFrequency/nScreenRefreshRate)
1.1.1.12 root 172:
173: /* Current sound replay freq (usually 44100 Hz) */
1.1.1.14 root 174: #define YM_REPLAY_FREQ nAudioFrequency
1.1.1.12 root 175:
1.1.1.16 root 176: /* YM-2149 clock on all Atari models is 2 MHz */
177: #define YM_ATARI_CLOCK (MachineClocks.YM_Freq)
1.1.1.12 root 178:
179:
180: /* Merge/read the 3 volumes in a single integer (5 bits per volume) */
181: #define YM_MERGE_VOICE(C,B,A) ( (C)<<10 | (B)<<5 | A )
182: #define YM_MASK_1VOICE 0x1f
183: #define YM_MASK_A 0x1f
184: #define YM_MASK_B (0x1f<<5)
185: #define YM_MASK_C (0x1f<<10)
186:
187:
188: /* Constants for YM2149_Normalise_5bit_Table */
189: #define YM_OUTPUT_LEVEL 0x7fff /* amplitude of the final signal (0..65535 if centered, 0..32767 if not) */
1.1.1.14 root 190: #define YM_OUTPUT_CENTERED false
1.1.1.12 root 191:
192:
193:
194: /*--------------------------------------------------------------*/
195: /* Variables for the YM2149 emulator (need to be saved and */
196: /* restored in memory snapshots) */
197: /*--------------------------------------------------------------*/
198:
199: static ymu32 stepA , stepB , stepC;
200: static ymu32 posA , posB , posC;
201: static ymu32 mixerTA , mixerTB , mixerTC;
202: static ymu32 mixerNA , mixerNB , mixerNC;
203:
204: static ymu32 noiseStep;
205: static ymu32 noisePos;
206: static ymu32 currentNoise;
207: static ymu32 RndRack; /* current random seed */
208:
209: static ymu32 envStep;
210: static ymu32 envPos;
211: static int envShape;
212:
213: static ymu16 EnvMask3Voices = 0; /* mask is 0x1f for voices having an active envelope */
214: static ymu16 Vol3Voices = 0; /* volume 0-0x1f for voices having a constant volume */
215: /* volume is set to 0 if voice has an envelope in EnvMask3Voices */
216:
217:
218: /* Global variables that can be changed/read from other parts of Hatari */
219: Uint8 SoundRegs[ 14 ];
220:
1.1.1.16 root 221: int YmVolumeMixing = YM_TABLE_MIXING;
1.1.1.14 root 222: bool UseLowPassFilter = false;
1.1.1.12 root 223:
224: bool bEnvelopeFreqFlag; /* Cleared each frame for YM saving */
225:
226: Sint16 MixBuffer[MIXBUFFER_SIZE][2];
227: int nGeneratedSamples; /* Generated samples since audio buffer update */
228: static int ActiveSndBufIdx; /* Current working index into above mix buffer */
1.1.1.16 root 229: static int ActiveSndBufIdxAvi; /* Current working index to save an AVI audio frame */
230:
231: static yms64 SamplesPerFrame_unrounded = 0; /* Number of samples for the current VBL, with simulated fractional part */
232: static int SamplesPerFrame; /* Number of samples to generate for the current VBL */
233: static int CurrentSamplesNb = 0; /* Number of samples already generated for the current VBL */
1.1.1.12 root 234:
1.1.1.16 root 235: bool Sound_BufferIndexNeedReset = false;
1.1.1.12 root 236:
237:
238: /*--------------------------------------------------------------*/
239: /* Local functions prototypes */
240: /*--------------------------------------------------------------*/
241:
1.1.1.17 root 242: static ymsample LowPassFilter (ymsample x0);
1.1.1.18! root 243: static ymsample PWMaliasFilter (ymsample x0);
1.1.1.12 root 244:
1.1.1.18! root 245: static void interpolate_volumetable (ymu16 volumetable[32][32][32]);
1.1.1.12 root 246:
1.1.1.18! root 247: static void YM2149_BuildModelVolumeTable(ymu16 volumetable[32][32][32]);
! 248: static void YM2149_BuildLinearVolumeTable(ymu16 volumetable[32][32][32]);
1.1.1.12 root 249: static void YM2149_Normalise_5bit_Table(ymu16 *in_5bit , yms16 *out_5bit, unsigned int Level, bool DoCenter);
250:
251: static void YM2149_EnvBuild (void);
1.1.1.16 root 252: static void Ym2149_BuildVolumeTable (void);
1.1.1.12 root 253: static void Ym2149_Init (void);
254: static void Ym2149_Reset (void);
255:
256: static ymu32 YM2149_RndCompute (void);
257: static ymu32 Ym2149_ToneStepCompute (ymu8 rHigh , ymu8 rLow);
258: static ymu32 Ym2149_NoiseStepCompute (ymu8 rNoise);
259: static ymu32 Ym2149_EnvStepCompute (ymu8 rHigh , ymu8 rLow);
260: static ymsample YM2149_NextSample (void);
261:
1.1.1.16 root 262: static int Sound_SetSamplesPassed(bool FillFrame);
263: static void Sound_GenerateSamples(int SamplesToGenerate);
1.1.1.12 root 264:
265:
266:
267: /*--------------------------------------------------------------*/
1.1.1.17 root 268: /* DC Adjuster */
1.1.1.12 root 269: /*--------------------------------------------------------------*/
270:
1.1.1.17 root 271: /**
272: * 6dB/octave first order HPF fc = (1.0-0.998)*44100/(2.0*pi)
273: * Z pole = 0.99804 --> FS = 44100 Hz : fc=13.7 Hz (11 Hz meas)
274: * a = (int32_t)(32768.0*(1.0 - pole)) : a = 64 !!!
275: * Input range: -32768 to 32767 Maximum step: +65536 or -65472
276: */
277: ymsample Subsonic_IIR_HPF_Left(ymsample x0)
1.1.1.12 root 278: {
1.1.1.17 root 279: static yms32 x1 = 0, y1 = 0, y0 = 0;
1.1.1.14 root 280:
1.1.1.17 root 281: y1 += ((x0 - x1)<<15) - (y0<<6); /* 64*y0 */
282: y0 = y1>>15;
283: x1 = x0;
1.1.1.12 root 284:
1.1.1.17 root 285: return y0;
1.1.1.12 root 286: }
287:
288:
1.1.1.17 root 289: ymsample Subsonic_IIR_HPF_Right(ymsample x0)
1.1 root 290: {
1.1.1.17 root 291: static yms32 x1 = 0, y1 = 0, y0 = 0;
1.1.1.12 root 292:
1.1.1.17 root 293: y1 += ((x0 - x1)<<15) - (y0<<6); /* 64*y0 */
294: y0 = y1>>15;
295: x1 = x0;
1.1.1.12 root 296:
1.1.1.17 root 297: return y0;
1.1.1.12 root 298: }
299:
300:
1.1.1.17 root 301: /*--------------------------------------------------------------*/
302: /* Low Pass Filter routines. */
303: /*--------------------------------------------------------------*/
1.1.1.12 root 304:
1.1.1.17 root 305: /**
306: * Get coefficients for different Fs (C10 is in ST only):
307: * Wc = 2*M_PI*4895.1;
308: * Fs = 44100;
309: * warp = Wc/tanf((Wc/2)/Fs);
310: * b = Wc/(warp+Wc);
311: * a = (Wc-warp)/(warp+Wc);
312: *
313: * #define B_z (yms32)( 0.2667*(1<<15))
314: * #define A_z (yms32)(-0.4667*(1<<15))
315: *
316: * y0 = (B_z*(x0 + x1) - A_z*y0) >> 15;
317: * x1 = x0;
318: *
319: * The Lowpass Filter formed by C10 = 0.1 uF
320: * and
1.1.1.18! root 321: * R8=1k // 1k*(65119-46602)/65119 // R9=10k // R10=5.1k //
1.1.1.17 root 322: * (R12=470)*(100=Q1_HFE) = 206.865 ohms when YM2149 is High
323: * and
324: * R8=1k // R9=10k // R10=5.1k // (R12=470)*(100=Q1_HFE)
325: * = 759.1 ohms when YM2149 is Low
326: * High corner is 1/(2*pi*(0.1*10e-6)*206.865) fc = 7693.7 Hz
327: * Low corner is 1/(2*pi*(0.1*10e-6)*795.1) fc = 2096.6 Hz
328: * Notes:
329: * - using STF reference designators R8 R9 R10 C10 (from dec 1986 schematics)
330: * - using corresponding numbers from psgstrep and psgquart
331: * - 65119 is the largest value in Paulo's psgstrep table
332: * - 46602 is the largest value in Paulo's psgquart table
333: * - this low pass filter uses the highest cutoff frequency
334: * on the STf (a slightly lower frequency is reasonable).
335: *
336: * A first order lowpass filter with a high cutoff frequency
337: * is used when the YM2149 pulls high, and a lowpass filter
338: * with a low cutoff frequency is used when R8 pulls low.
339: */
340: static ymsample LowPassFilter(ymsample x0)
341: {
342: static yms32 y0 = 0, x1 = 0;
343:
344: if (x0 >= y0)
345: /* YM Pull up: fc = 7586.1 Hz (44.1 KHz), fc = 8257.0 Hz (48 KHz) */
346: y0 = (3*(x0 + x1) + (y0<<1)) >> 3;
347: else
348: /* R8 Pull down: fc = 1992.0 Hz (44.1 KHz), fc = 2168.0 Hz (48 KHz) */
349: y0 = ((x0 + x1) + (6*y0)) >> 3;
1.1.1.12 root 350:
1.1.1.17 root 351: x1 = x0;
352: return y0;
1.1.1.12 root 353: }
354:
1.1.1.18! root 355: /**
! 356: * This piecewise selective filter works by filtering the falling
! 357: * edge of a sampled pulse-wave differently from the rising edge.
! 358: *
! 359: * Piecewise selective filtering is effective because harmonics on
! 360: * one part of a wave partially define harmonics on other portions.
! 361: *
! 362: * Piecewise selective filtering can efficiently reduce aliasing
! 363: * with minimal harmonic removal.
! 364: *
! 365: * I disclose this information into the public domain so that it
! 366: * cannot be patented. May 23 2012 David Savinkoff.
! 367: */
! 368: static ymsample PWMaliasFilter(ymsample x0)
! 369: {
! 370: static yms32 y0 = 0, x1 = 0;
! 371:
! 372: if (x0 >= y0)
! 373: /* YM Pull up */
! 374: y0 = x0;
! 375: else
! 376: /* R8 Pull down */
! 377: y0 = (3*(x0 + x1) + (y0<<1)) >> 3;
! 378:
! 379: x1 = x0;
! 380: return y0;
! 381: }
! 382:
1.1.1.12 root 383:
384:
385: /*--------------------------------------------------------------*/
386: /* Build the volume conversion table used to simulate the */
387: /* behaviour of DAC used with the YM2149 in the atari ST. */
388: /* The final 32*32*32 table is built using a 16*16*16 table */
389: /* of all possible fixed volume combinations on a ST. */
390: /*--------------------------------------------------------------*/
391:
1.1.1.18! root 392: static void interpolate_volumetable(ymu16 volumetable[32][32][32])
1.1.1.12 root 393: {
1.1.1.18! root 394: int i, j, k;
1.1.1.12 root 395:
1.1.1.18! root 396: for (i = 1; i < 32; i += 2) { /* Copy 16 Panels to make a Block */
! 397: for (j = 1; j < 32; j += 2) { /* Copy 16 Rows to make a Panel */
! 398: for (k = 1; k < 32; k += 2) { /* Copy 16 Elements to make a Row */
! 399: volumetable[i][j][k] = volumetable_original[(i-1)/2][(j-1)/2][(k-1)/2];
1.1.1.12 root 400: }
1.1.1.18! root 401: volumetable[i][j][0] = volumetable[i][j][1]; /* Move 0th Element */
! 402: volumetable[i][j][1] = volumetable[i][j][3]; /* Move 1st Element */
! 403: /* Interpolate 3rd Element */
! 404: volumetable[i][j][3] = (ymu16)(0.5 + sqrt((double)volumetable[i][j][1] * volumetable[i][j][5]));
! 405: for (k = 2; k < 32; k += 2) /* Interpolate Even Elements */
! 406: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][j][k-1] * volumetable[i][j][k+1]));
1.1.1.12 root 407: }
1.1.1.18! root 408: for (k = 0; k < 32; k++) {
! 409: volumetable[i][0][k] = volumetable[i][1][k]; /* Move 0th Row */
! 410: volumetable[i][1][k] = volumetable[i][3][k]; /* Move 1st Row */
! 411: /* Interpolate 3rd Row */
! 412: volumetable[i][3][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][1][k] * volumetable[i][5][k]));
! 413: }
! 414: for (j = 2; j < 32; j += 2) /* Interpolate Even Rows */
! 415: for (k = 0; k < 32; k++)
! 416: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][j-1][k] * volumetable[i][j+1][k]));
1.1.1.11 root 417: }
1.1.1.18! root 418: for (j = 0; j < 32; j++)
! 419: for (k = 0; k < 32; k++) {
! 420: volumetable[0][j][k] = volumetable[1][j][k]; /* Move 0th Panel */
! 421: volumetable[1][j][k] = volumetable[3][j][k]; /* Move 1st Panel */
! 422: /* Interpolate 3rd Panel */
! 423: volumetable[3][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[1][j][k] * volumetable[5][j][k]));
! 424: }
! 425: for (i = 2; i < 32; i += 2) /* Interpolate Even Panels */
! 426: for (j = 0; j < 32; j++) /* Interpolate Even Panels */
! 427: for (k = 0; k < 32; k++)
! 428: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i-1][j][k] * volumetable[i+1][j][k]));
1.1 root 429: }
430:
1.1.1.5 root 431:
1.1.1.12 root 432:
433:
1.1.1.2 root 434: /*-----------------------------------------------------------------------*/
1.1.1.11 root 435: /**
1.1.1.12 root 436: * Build a linear version of the conversion table.
437: * We use the mean of the 3 volumes converted to 16 bit values
438: * (each value of ymout1c5bit is in [0,65535])
1.1.1.11 root 439: */
1.1.1.12 root 440:
1.1.1.18! root 441: static void YM2149_BuildLinearVolumeTable(ymu16 volumetable[32][32][32])
1.1 root 442: {
1.1.1.12 root 443: int i, j, k;
444:
445: for (i = 0; i < 32; i++)
446: for (j = 0; j < 32; j++)
447: for (k = 0; k < 32; k++)
1.1.1.18! root 448: volumetable[i][j][k] = (ymu16)( ((ymu32)ymout1c5bit[i] + ymout1c5bit[j] + ymout1c5bit[k]) / 3);
1.1.1.12 root 449: }
450:
451:
452:
1.1 root 453:
1.1.1.12 root 454: /*-----------------------------------------------------------------------*/
455: /**
1.1.1.17 root 456: * Build a circuit analysed version of the conversion table.
457: * David Savinkoff designed this algorithm by analysing data
458: * measured by Paulo Simoes and Benjamin Gerard.
459: * The numbers are arrived at by assuming a current steering
460: * resistor ladder network and using the voltage divider rule.
1.1.1.18! root 461: *
! 462: * If one looks at the ST schematic of the YM2149, one sees
! 463: * three sound pins tied together and attached to a 1000 ohm
! 464: * resistor (1k) that has the other end grounded.
! 465: * The 1k resistor is also in parallel with a 0.1 microfarad
! 466: * capacitor (on the Atari ST, not STE or others). The voltage
! 467: * developed across the 1K resistor is the output voltage which
! 468: * I call Vout.
! 469: *
! 470: * The output of the YM2149 is modeled well as pullup resistors.
! 471: * Thus, the three sound pins are seen as three parallel
! 472: * computer-controlled, adjustable pull-up resistors.
! 473: * To emulate the output of the YM2149, one must determine the
! 474: * resistance values of the YM2149 relative to the 1k resistor,
! 475: * which is done by the 'math model'.
! 476: *
! 477: * The AC + DC math model is:
! 478: *
! 479: * (MaxVol*WARP) / (1.0 + 1.0/(conductance_[i]+conductance_[j]+conductance_[k]))
! 480: * or
! 481: * (MaxVol*WARP) / (1.0 + 1.0/( 1/Ra +1/Rb +1/Rc )) , Ra = channel A resistance
! 482: *
! 483: * Note that the first 1.0 in the formula represents the
! 484: * normalized 1k resistor (1.0 * 1000 ohms = 1k).
! 485: *
! 486: * The YM2149 DC component model represents the output voltage
! 487: * filtered of high frequency AC component, but DC component
! 488: * remains.
! 489: * The YM2149 DC component mode treats the voltage exactly as if
! 490: * it were low pass filtered. This is more than what is required
! 491: * to make 'quartet mode sound'. Simplicity leads to Generality!
! 492: *
! 493: * The DC component model model is:
! 494: *
! 495: * (MaxVol*WARP) / (2.0 + 1.0/( 1/Ra + 1/Rb + 1/Rc))
! 496: * or
! 497: * (MaxVol*WARP*0.5) / (1.0 + 0.5/( 1/Ra + 1/Rb + 1/Rc))
! 498: *
! 499: * Note that the 1.0 represents the normalized 1k resistor.
! 500: * 0.5 represents 50% duty cycle for the parallel resistors
! 501: * being summed (this effectively doubles the pull-up resistance).
1.1.1.17 root 502: */
503:
1.1.1.18! root 504: static void YM2149_BuildModelVolumeTable(ymu16 volumetable[32][32][32])
1.1.1.17 root 505: {
1.1.1.18! root 506: #define MaxVol 65535.0 /* Normal Mode Maximum value in table */
! 507: #define FOURTH2 1.19 /* Fourth root of two from YM2149 */
! 508: #define WARP 1.666666666666666667 /* measured as 1.65932 from 46602 */
1.1.1.17 root 509:
510: double conductance;
511: double conductance_[32];
512: int i, j, k;
513:
514: /**
515: * YM2149 and R8=1k follows (2^-1/4)^(n-31) better when 2 voices are
516: * summed (A+B or B+C or C+A) rather than individually (A or B or C):
1.1.1.18! root 517: * conductance = 2.0/3.0/(1.0-1.0/WARP)-2.0/3.0;
1.1.1.17 root 518: * When taken into consideration with three voices.
519: *
520: * Note that the YM2149 does not use laser trimmed resistances, thus
521: * has offsets that are added and/or multiplied with (2^-1/4)^(n-31).
522: */
1.1.1.18! root 523: conductance = 2.0/3.0/(1.0-1.0/WARP)-2.0/3.0; /* conductance = 1.0 */
1.1.1.17 root 524:
525: /**
526: * Because the YM current output (voltage source with series resistances)
527: * is connected to a grounded resistor to develop the output voltage
528: * (instead of a current to voltage converter), the output transfer
529: * function is not linear. Thus:
530: * 2.0*conductance_[n] = 1.0/(1.0-1.0/FOURTH2/(1.0/conductance + 1.0))-1.0;
531: */
532: for (i = 31; i >= 1; i--)
533: {
534: conductance_[i] = conductance/2.0;
535: conductance = 1.0/(1.0-1.0/FOURTH2/(1.0/conductance + 1.0))-1.0;
536: }
1.1.1.18! root 537: conductance_[0] = 1.0e-8; /* Avoid divide by zero */
1.1.1.17 root 538:
1.1.1.18! root 539: /**
! 540: * YM2149 AC + DC components model:
! 541: * (Note that Maxvol is 65119 in Simoes' table, 65535 in Gerard's)
! 542: *
! 543: * Sum the conductances as a function of a voltage divider:
! 544: * Vout=Vin*Rout/(Rout+Rin)
! 545: */
1.1.1.17 root 546: for (i = 0; i < 32; i++)
547: for (j = 0; j < 32; j++)
548: for (k = 0; k < 32; k++)
549: {
1.1.1.18! root 550: volumetable[i][j][k] = (ymu16)(0.5+(MaxVol*WARP)/(1.0 +
1.1.1.17 root 551: 1.0/(conductance_[i]+conductance_[j]+conductance_[k])));
552: }
1.1.1.18! root 553:
! 554: /**
! 555: * YM2149 DC component model:
! 556: * R8=1k (pulldown) + YM//1K (pullup) with YM 50% duty PWM
! 557: * (Note that MaxVol is 46602 in Paulo Simoes Quartet mode table)
! 558: *
! 559: * for (i = 0; i < 32; i++)
! 560: * for (j = 0; j < 32; j++)
! 561: * for (k = 0; k < 32; k++)
! 562: * {
! 563: * volumetable[i][j][k] = (ymu16)(0.5+(MaxVol*WARP)/(1.0 +
! 564: * 2.0/(conductance_[i]+conductance_[j]+conductance_[k])));
! 565: * }
! 566: */
1.1.1.17 root 567: }
568:
569:
570:
571:
572: /*-----------------------------------------------------------------------*/
573: /**
1.1.1.12 root 574: * Normalise and optionally center the volume table used to
575: * convert the 3 volumes to a final signed 16 bit sample.
576: * This allows to adapt the amplitude/volume of the samples and
577: * to convert unsigned values to signed values.
578: * - in_5bit contains 32*32*32 unsigned values in the range
579: * [0,65535].
580: * - out_5bit will contain signed values
581: * Possible values are :
582: * Level=65535 and DoCenter=TRUE -> [-32768,32767]
1.1.1.14 root 583: * Level=32767 and DoCenter=false -> [0,32767]
1.1.1.12 root 584: */
585:
586: static void YM2149_Normalise_5bit_Table(ymu16 *in_5bit , yms16 *out_5bit, unsigned int Level, bool DoCenter)
587: {
588: if ( Level )
1.1.1.11 root 589: {
1.1.1.14 root 590: int h;
1.1.1.12 root 591: int Max = in_5bit[0x7fff];
592: int Center = Level>>1;
1.1.1.14 root 593: //fprintf ( stderr , "level %d max %d center %d\n" , Level, Max, Center );
594:
1.1.1.12 root 595: /* Change the amplitude of the signal to 'level' : [0,max] -> [0,level] */
596: /* Then optionally center the signal around Level/2 */
597: /* This means we go from sthg like [0,65535] to [-32768, 32767] if Level=65535 and DoCenter=TRUE */
598: for (h=0; h<32*32*32; h++)
599: {
600: int tmp = in_5bit[h], res;
601: res = tmp * Level / Max;
1.1.1.14 root 602:
1.1.1.12 root 603: if ( DoCenter )
604: res -= Center;
605:
606: out_5bit[h] = res;
1.1.1.14 root 607: //fprintf ( stderr , "h %d in %d out %d\n" , h , tmp , res );
1.1.1.12 root 608: }
1.1.1.11 root 609: }
1.1 root 610: }
611:
1.1.1.5 root 612:
1.1.1.12 root 613:
614:
1.1.1.2 root 615: /*-----------------------------------------------------------------------*/
1.1.1.11 root 616: /**
1.1.1.12 root 617: * Precompute all 16 possible envelopes.
618: * Each envelope is made of 3 blocks of 32 volumes.
1.1.1.11 root 619: */
1.1.1.12 root 620:
621: static void YM2149_EnvBuild ( void )
1.1 root 622: {
1.1.1.12 root 623: int env;
624: int block;
625: int vol=0 , inc=0;
626: int i;
1.1 root 627:
1.1.1.12 root 628:
629: for ( env=0 ; env<16 ; env++ ) /* 16 possible envelopes */
630: for ( block=0 ; block<3 ; block++ ) /* 3 blocks to define an envelope */
631: {
1.1.1.14 root 632: switch ( YmEnvDef[ env ][ block ] )
633: {
1.1.1.12 root 634: case ENV_GODOWN : vol=31 ; inc=-1 ; break;
635: case ENV_GOUP : vol=0 ; inc=1 ; break;
636: case ENV_DOWN : vol=0 ; inc=0 ; break;
637: case ENV_UP : vol=31 ; inc=0 ; break;
1.1.1.14 root 638: }
639:
1.1.1.12 root 640: for ( i=0 ; i<32 ; i++ ) /* 32 volumes per block */
641: {
642: YmEnvWaves[ env ][ block*32 + i ] = YM_MERGE_VOICE ( vol , vol , vol );
643: vol += inc;
644: }
645: }
646: }
647:
648:
649:
650: /*-----------------------------------------------------------------------*/
651: /**
1.1.1.16 root 652: * Depending on the YM mixing method, build the table used to convert
653: * the 3 YM volumes into a single sample.
1.1.1.12 root 654: */
655:
1.1.1.16 root 656: static void Ym2149_BuildVolumeTable(void)
1.1.1.12 root 657: {
658: /* Depending on the volume mixing method, we use a table based on real measures */
659: /* or a table based on a linear volume mixing. */
1.1.1.17 root 660: if ( YmVolumeMixing == YM_MODEL_MIXING )
661: YM2149_BuildModelVolumeTable(ymout5_u16); /* create 32*32*32 circuit analysed model of the volume table */
662: else if ( YmVolumeMixing == YM_TABLE_MIXING )
1.1.1.16 root 663: interpolate_volumetable(ymout5_u16); /* expand the 16*16*16 values in volumetable_original to 32*32*32 */
1.1.1.12 root 664: else
665: YM2149_BuildLinearVolumeTable(ymout5_u16); /* combine the 32 possible volumes */
666:
667: /* Normalise/center the values (convert from u16 to s16) */
1.1.1.18! root 668: YM2149_Normalise_5bit_Table ( ymout5_u16[0][0] , ymout5 , YM_OUTPUT_LEVEL , YM_OUTPUT_CENTERED );
1.1.1.16 root 669: }
670:
671:
672:
673: /*-----------------------------------------------------------------------*/
674: /**
675: * Init some internal tables for faster results (env, volume)
676: * and reset the internal states.
677: */
678:
679: static void Ym2149_Init(void)
680: {
681: /* Build the 16 envelope shapes */
682: YM2149_EnvBuild();
683:
684: /* Build the volume conversion table */
685: Ym2149_BuildVolumeTable();
1.1.1.12 root 686:
687: /* Reset YM2149 internal states */
688: Ym2149_Reset();
689: }
690:
691:
692:
693: /*-----------------------------------------------------------------------*/
694: /**
1.1.1.16 root 695: * Reset all ym registers as well as the internal variables
1.1.1.12 root 696: */
697:
698: static void Ym2149_Reset(void)
699: {
700: int i;
1.1.1.14 root 701:
1.1.1.12 root 702: for ( i=0 ; i<14 ; i++ )
703: Sound_WriteReg ( i , 0 );
704:
705: Sound_WriteReg ( 7 , 0xff );
706:
1.1.1.16 root 707: posA = 0;
708: posB = 0;
709: posC = 0;
710:
1.1.1.12 root 711: currentNoise = 0xffff;
1.1.1.14 root 712:
1.1.1.12 root 713: RndRack = 1;
1.1.1.14 root 714:
1.1.1.12 root 715: envShape = 0;
716: envPos = 0;
717: }
718:
719:
720:
721: /*-----------------------------------------------------------------------*/
722: /**
723: * Returns a pseudo random value, used to generate white noise.
724: */
725:
726: static ymu32 YM2149_RndCompute(void)
727: {
728: ymu32 bit;
1.1.1.14 root 729:
1.1.1.12 root 730: bit = (RndRack&1) ^ ((RndRack>>2)&1);
731: RndRack = (RndRack>>1) | (bit<<16);
732: return (bit ? 0 : 0xffff);
733: }
734:
735:
736:
737: /*-----------------------------------------------------------------------*/
738: /**
1.1.1.16 root 739: * Compute tone's step based on the input period.
740: * Although for tone we should have the same result when per==0 and per==1,
741: * this gives some very sharp and unpleasant sounds in the emulation.
742: * To get a better sound, we consider all per<=5 to give step=0, which will
743: * produce a constant output at value '1'. This should be handled with some
744: * proper filters to remove high frequencies as on a real ST (where per<=9
745: * gives nearly no audible sound).
746: * A common replay freq of 44.1 kHz will also not be high enough to correctly
747: * render possible tone's freq of 125 or 62.5 kHz (when per==1 or per==2)
1.1.1.12 root 748: */
749:
1.1.1.16 root 750: #define NEWSTEP
751: #ifndef NEWSTEP
1.1.1.12 root 752: static ymu32 Ym2149_ToneStepCompute(ymu8 rHigh , ymu8 rLow)
753: {
754: int per;
1.1.1.14 root 755: yms64 step;
1.1.1.12 root 756:
757: per = rHigh&15;
758: per = (per<<8)+rLow;
1.1.1.18! root 759:
! 760: if (per <= (int)(YM_ATARI_CLOCK/(YM_REPLAY_FREQ*7)) )
1.1.1.12 root 761: return 0;
762:
1.1.1.14 root 763: step = YM_ATARI_CLOCK;
1.1.1.12 root 764: step <<= (15+16-3);
765: step /= (per * YM_REPLAY_FREQ);
766:
767: return step;
768: }
1.1.1.16 root 769: #else
770: static ymu32 Ym2149_ToneStepCompute(ymu8 rHigh , ymu8 rLow)
771: {
772: int per;
773: yms64 step;
774:
775: per = rHigh&15;
776: per = (per<<8)+rLow;
777:
778: #if 0 /* need some high freq filters for this to work correctly */
779: if ( per == 0 )
780: per = 1; /* result for Per=0 is the same as for Per=1 */
781: #else
1.1.1.18! root 782: if (per <= (int)(YM_ATARI_CLOCK/(YM_REPLAY_FREQ*7)) )
! 783: return 0; /* discard frequencies higher than 80% of nyquist rate. */
1.1.1.16 root 784: #endif
785:
786: step = YM_ATARI_CLOCK;
787: step <<= 24;
788:
789: step /= (per * 8 * YM_REPLAY_FREQ); /* 0x5ab9 < step < 0x5ab3f46 at 44.1 kHz */
790:
791: return step;
792: }
793: #endif
1.1.1.12 root 794:
1.1.1.16 root 795: /*-----------------------------------------------------------------------*/
796: /**
797: * Compute noise's step based on the input period.
798: * On a real STF, we get the same result when per==0 and per==1.
799: * A common replay freq of 44.1 kHz will not be high enough to correctly
800: * render possible noise's freq of 125 or 62.5 kHz (when per==1 or per==2).
801: * With a random wave such as noise, this means that with a replay freq
802: * of 44.1 kHz, per==1 and per==2 (as well as per==3) will sound the same :
803: * per==1 step=0x2d59fa3 freq=125 kHz
804: * per==2 step=0x16acfd1 freq=62.5 kHz
805: * per==3 step=0x0f1dfe1 freq=41.7 kHz
806: */
1.1.1.12 root 807:
1.1.1.16 root 808: #ifndef NEWSTEP
1.1.1.12 root 809: static ymu32 Ym2149_NoiseStepCompute(ymu8 rNoise)
810: {
811: int per;
1.1.1.14 root 812: yms64 step;
1.1.1.12 root 813:
814: per = (rNoise&0x1f);
815: if (per<3)
816: return 0;
817:
1.1.1.14 root 818: step = YM_ATARI_CLOCK;
1.1.1.12 root 819: step <<= (16-1-3);
820: step /= (per * YM_REPLAY_FREQ);
821:
822: return step;
1.1 root 823: }
1.1.1.16 root 824: #else
825: static ymu32 Ym2149_NoiseStepCompute(ymu8 rNoise)
826: {
827: int per;
828: yms64 step;
829:
830: per = (rNoise&0x1f);
1.1 root 831:
1.1.1.16 root 832: if ( per == 0 )
833: per = 1; /* result for Per=0 is the same as for Per=1 */
834:
835: step = YM_ATARI_CLOCK;
836: step <<= 24;
837:
838: step /= (per * 16 * YM_REPLAY_FREQ); /* 0x17683f < step < 0x2d59fa3 at 44.1 kHz */
839:
840: return step;
841: }
842: #endif
1.1.1.5 root 843:
1.1.1.2 root 844: /*-----------------------------------------------------------------------*/
1.1.1.11 root 845: /**
1.1.1.12 root 846: * Compute envelope's step. The envelope is made of different patterns
847: * of 32 volumes. In each pattern, the volume is changed at frequency
848: * Fe = MasterClock / ( 8 * EnvPer ).
849: * In our case, we use a lower replay freq ; between 2 consecutive calls
850: * to envelope's generation, the internal counter will advance 'step'
851: * units, where step = MasterClock / ( 8 * EnvPer * YM_REPLAY_FREQ )
852: * As 'step' requires floating point to be stored, we use left shifting
853: * to multiply 'step' by a fixed amount. All operations are made with
854: * shifted values ; to get the final value, we must right shift the
855: * result. We use '<<24', which gives 8 bits for the integer part, and
856: * the equivalent of 24 bits for the fractional part.
857: * Since we're using large numbers, we temporarily use 64 bits integer
858: * to avoid overflow and keep largest precision possible.
1.1.1.16 root 859: * On a real STF, we get the same result when per==0 and per==1.
1.1.1.11 root 860: */
1.1.1.12 root 861:
862: static ymu32 Ym2149_EnvStepCompute(ymu8 rHigh , ymu8 rLow)
1.1 root 863: {
1.1.1.12 root 864: yms64 per;
1.1.1.14 root 865: yms64 step;
1.1 root 866:
1.1.1.12 root 867: per = rHigh;
868: per = (per<<8)+rLow;
869:
1.1.1.14 root 870: step = YM_ATARI_CLOCK;
1.1.1.12 root 871: step <<= 24;
1.1.1.16 root 872:
873: if ( per == 0 )
874: per = 1; /* result for Per=0 is the same as for Per=1 */
875:
876: step /= (8 * per * YM_REPLAY_FREQ); /* 0x5ab < step < 0x5ab3f46 at 44.1 kHz */
1.1.1.12 root 877:
878: return step;
879: }
880:
881:
882:
883: /*-----------------------------------------------------------------------*/
884: /**
885: * Main function : compute the value of the next sample.
886: * Mixes all 3 voices with tone+noise+env and apply low pass
887: * filter if needed.
1.1.1.16 root 888: * All operations are done with integer math, using <<24 to simulate
889: * floating point precision : upper 8 bits are the integer part, lower 24
890: * are the fractional part.
891: * Tone is a square wave with 2 states 0 or 1. If integer part of posX is
892: * even (bit24=0) we consider output is 0, else (bit24=1) we consider
893: * output is 1. This gives the value of bt for one voice after extending it
894: * to all 0 bits or all 1 bits using a '-'
1.1.1.12 root 895: */
896:
1.1.1.16 root 897: #ifndef NEWSTEP
1.1.1.12 root 898: static ymsample YM2149_NextSample(void)
899: {
900: ymsample sample;
901: int bt;
902: ymu32 bn;
903: ymu16 Env3Voices;
904: ymu16 Tone3Voices;
905:
906:
907: /* Noise value : 0 or 0xffff */
908: if ( noisePos&0xffff0000 )
909: {
910: currentNoise ^= YM2149_RndCompute();
911: noisePos &= 0xffff;
912: }
913: bn = currentNoise; /* 0 or 0xffff */
914:
915: /* Get the 5 bits volume corresponding to the current envelope's position */
916: Env3Voices = YmEnvWaves[ envShape ][ envPos>>24 ]; /* integer part of envPos is in bits 24-31 */
917: Env3Voices &= EnvMask3Voices; /* only keep volumes for voices using envelope */
918:
919: //fprintf ( stderr , "env %x %x %x\n" , Env3Voices , envStep , envPos );
920:
921: /* Tone3Voices will contain the output state of each voice : 0 or 0x1f */
922: bt = ((((yms32)posA)>>31) | mixerTA) & (bn | mixerNA); /* 0 or 0xffff */
923: Tone3Voices = bt & YM_MASK_1VOICE; /* 0 or 0x1f */
924: bt = ((((yms32)posB)>>31) | mixerTB) & (bn | mixerNB);
925: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 5;
926: bt = ((((yms32)posC)>>31) | mixerTC) & (bn | mixerNC);
927: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 10;
928:
929: /* Combine fixed volumes and envelope volumes and keep the resulting */
930: /* volumes depending on the output state of each voice (0 or 0x1f) */
931: Tone3Voices &= ( Env3Voices | Vol3Voices );
932:
1.1.1.18! root 933: /* When a step period is 0, the represented frequency was filtered from the */
! 934: /* ouput of the YM2149. Thus, use the transient DC component of the sample. */
! 935: /* Note that the "-1" table offset is a "good fit" for the DC component. */
! 936:
! 937: if (stepA == 0 && (Tone3Voices & YM_MASK_A) > 1)
! 938: Tone3Voices -= 1; /* Voice A AC component removed; Transient DC component remains */
! 939:
! 940: if (stepB == 0 && (Tone3Voices & YM_MASK_B) > 1<<5)
! 941: Tone3Voices -= 1<<5; /* Voice B AC component removed; Transient DC component remains */
! 942:
! 943: if (stepC == 0 && (Tone3Voices & YM_MASK_C) > 1<<10)
! 944: Tone3Voices -= 1<<10; /* Voice C AC component removed; Transient DC component remains */
! 945:
1.1.1.12 root 946: /* D/A conversion of the 3 volumes into a sample using a precomputed conversion table */
1.1.1.18! root 947:
1.1.1.12 root 948: sample = ymout5[ Tone3Voices ]; /* 16 bits signed value */
949:
950:
951: /* Increment positions */
952: posA += stepA;
953: posB += stepB;
954: posC += stepC;
955: noisePos += noiseStep;
1.1.1.14 root 956:
1.1.1.12 root 957: envPos += envStep;
958: if ( envPos >= (3*32) << 24 ) /* blocks 0, 1 and 2 were used (envPos 0 to 95) */
959: envPos -= (2*32) << 24; /* replay/loop blocks 1 and 2 (envPos 32 to 95) */
960:
961: /* Apply low pass filter ? */
962: if ( UseLowPassFilter )
1.1.1.18! root 963: return LowPassFilter(sample);
! 964: else
! 965: return PWMaliasFilter(sample);
1.1 root 966: }
1.1.1.16 root 967: #else
968: static ymsample YM2149_NextSample(void)
969: {
970: ymsample sample;
971: ymu32 bt;
972: ymu32 bn;
973: ymu16 Env3Voices; /* 0x00CCBBAA */
974: ymu16 Tone3Voices; /* 0x00CCBBAA */
1.1 root 975:
976:
1.1.1.16 root 977: /* Noise value : 0 or 0xffff */
978: if ( noisePos&0xff000000 ) /* integer part > 0 */
979: {
980: currentNoise ^= YM2149_RndCompute();
981: noisePos &= 0xffffff; /* keep fractional part of noisePos */
982: }
983: bn = currentNoise; /* 0 or 0xffff */
984:
985: /* Get the 5 bits volume corresponding to the current envelope's position */
986: Env3Voices = YmEnvWaves[ envShape ][ envPos>>24 ]; /* integer part of envPos is in bits 24-31 */
987: Env3Voices &= EnvMask3Voices; /* only keep volumes for voices using envelope */
988:
989: //fprintf ( stderr , "env %x %x %x\n" , Env3Voices , envStep , envPos );
990:
991: /* Tone3Voices will contain the output state of each voice : 0 or 0x1f */
992: bt = -( (posA>>24) & 1); /* 0 if bit24=0 or 0xffffffff if bit24=1 */
993: bt = (bt | mixerTA) & (bn | mixerNA); /* 0 or 0xffff */
994: Tone3Voices = bt & YM_MASK_1VOICE; /* 0 or 0x1f */
995: bt = -( (posB>>24) & 1);
996: bt = (bt | mixerTB) & (bn | mixerNB);
997: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 5;
998: bt = -( (posC>>24) & 1);
999: bt = (bt | mixerTC) & (bn | mixerNC);
1000: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 10;
1001:
1002: /* Combine fixed volumes and envelope volumes and keep the resulting */
1003: /* volumes depending on the output state of each voice (0 or 0x1f) */
1004: Tone3Voices &= ( Env3Voices | Vol3Voices );
1005:
1006: /* D/A conversion of the 3 volumes into a sample using a precomputed conversion table */
1.1.1.18! root 1007:
! 1008: if (stepA == 0 && (Tone3Voices & YM_MASK_A) > 1)
! 1009: Tone3Voices -= 1; /* Voice A AC component removed; Transient DC component remains */
! 1010:
! 1011: if (stepB == 0 && (Tone3Voices & YM_MASK_B) > 1<<5)
! 1012: Tone3Voices -= 1<<5; /* Voice B AC component removed; Transient DC component remains */
! 1013:
! 1014: if (stepC == 0 && (Tone3Voices & YM_MASK_C) > 1<<10)
! 1015: Tone3Voices -= 1<<10; /* Voice C AC component removed; Transient DC component remains */
! 1016:
1.1.1.16 root 1017: sample = ymout5[ Tone3Voices ]; /* 16 bits signed value */
1018:
1019:
1020: /* Increment positions */
1021: posA += stepA;
1022: posB += stepB;
1023: posC += stepC;
1024: noisePos += noiseStep;
1025:
1026: envPos += envStep;
1027: if ( envPos >= (3*32) << 24 ) /* blocks 0, 1 and 2 were used (envPos 0 to 95) */
1028: envPos -= (2*32) << 24; /* replay/loop blocks 1 and 2 (envPos 32 to 95) */
1029:
1030: /* Apply low pass filter ? */
1031: if ( UseLowPassFilter )
1.1.1.18! root 1032: return LowPassFilter(sample);
! 1033: else
! 1034: return PWMaliasFilter(sample);
1.1.1.16 root 1035: }
1036: #endif
1037:
1.1.1.12 root 1038:
1.1.1.2 root 1039: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1040: /**
1.1.1.12 root 1041: * Update internal variables (steps, volume masks, ...) each
1042: * time an YM register is changed.
1.1.1.11 root 1043: */
1.1.1.16 root 1044: #ifndef NEWSTEP
1045: #define BIT_SHIFT 31
1046: #else
1047: #define BIT_SHIFT 24
1048: #endif
1.1.1.12 root 1049: void Sound_WriteReg( int reg , Uint8 data )
1.1.1.7 root 1050: {
1.1.1.12 root 1051: switch (reg)
1052: {
1053: case 0:
1054: SoundRegs[0] = data;
1055: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 1056: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1057: break;
1058:
1059: case 1:
1060: SoundRegs[1] = data & 0x0f;
1061: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 1062: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1063: break;
1064:
1065: case 2:
1066: SoundRegs[2] = data;
1067: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 1068: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1069: break;
1.1.1.7 root 1070:
1.1.1.12 root 1071: case 3:
1072: SoundRegs[3] = data & 0x0f;
1073: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 1074: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1075: break;
1076:
1077: case 4:
1078: SoundRegs[4] = data;
1079: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 1080: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1081: break;
1082:
1083: case 5:
1084: SoundRegs[5] = data & 0x0f;
1085: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 1086: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1087: break;
1088:
1089: case 6:
1090: SoundRegs[6] = data & 0x1f;
1091: noiseStep = Ym2149_NoiseStepCompute ( SoundRegs[6] );
1092: if (!noiseStep)
1093: {
1094: noisePos = 0;
1095: currentNoise = 0xffff;
1096: }
1097: break;
1098:
1099: case 7:
1100: SoundRegs[7] = data & 0x3f; /* ignore bits 6 and 7 */
1101: mixerTA = (data&(1<<0)) ? 0xffff : 0;
1102: mixerTB = (data&(1<<1)) ? 0xffff : 0;
1103: mixerTC = (data&(1<<2)) ? 0xffff : 0;
1104: mixerNA = (data&(1<<3)) ? 0xffff : 0;
1105: mixerNB = (data&(1<<4)) ? 0xffff : 0;
1106: mixerNC = (data&(1<<5)) ? 0xffff : 0;
1107: break;
1108:
1109: case 8:
1110: SoundRegs[8] = data & 0x1f;
1111: if ( data & 0x10 )
1112: {
1113: EnvMask3Voices |= YM_MASK_A; /* env ON */
1114: Vol3Voices &= ~YM_MASK_A; /* fixed vol OFF */
1115: }
1116: else
1117: {
1118: EnvMask3Voices &= ~YM_MASK_A; /* env OFF */
1119: Vol3Voices &= ~YM_MASK_A; /* clear previous vol */
1120: Vol3Voices |= YmVolume4to5[ SoundRegs[8] ]; /* fixed vol ON */
1121: }
1122: break;
1.1.1.14 root 1123:
1.1.1.12 root 1124: case 9:
1125: SoundRegs[9] = data & 0x1f;
1126: if ( data & 0x10 )
1127: {
1128: EnvMask3Voices |= YM_MASK_B; /* env ON */
1129: Vol3Voices &= ~YM_MASK_B; /* fixed vol OFF */
1130: }
1131: else
1132: {
1133: EnvMask3Voices &= ~YM_MASK_B; /* env OFF */
1134: Vol3Voices &= ~YM_MASK_B; /* clear previous vol */
1135: Vol3Voices |= ( YmVolume4to5[ SoundRegs[9] ] ) << 5; /* fixed vol ON */
1136: }
1137: break;
1.1.1.14 root 1138:
1.1.1.12 root 1139: case 10:
1140: SoundRegs[10] = data & 0x1f;
1141: if ( data & 0x10 )
1142: {
1143: EnvMask3Voices |= YM_MASK_C; /* env ON */
1144: Vol3Voices &= ~YM_MASK_C; /* fixed vol OFF */
1145: }
1146: else
1147: {
1148: EnvMask3Voices &= ~YM_MASK_C; /* env OFF */
1149: Vol3Voices &= ~YM_MASK_C; /* clear previous vol */
1150: Vol3Voices |= ( YmVolume4to5[ SoundRegs[10] ] ) << 10; /* fixed vol ON */
1151: }
1152: break;
1153:
1154: case 11:
1155: SoundRegs[11] = data;
1156: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1157: break;
1158:
1159: case 12:
1160: SoundRegs[12] = data;
1161: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1162: break;
1163:
1164: case 13:
1165: SoundRegs[13] = data & 0xf;
1166: envPos = 0; /* when writing to EnvShape, we must reset the EnvPos */
1167: envShape = SoundRegs[13];
1.1.1.14 root 1168: bEnvelopeFreqFlag = true; /* used for YmFormat saving */
1.1.1.12 root 1169: break;
1170:
1171: }
1172: }
1173:
1174:
1175:
1176: /*-----------------------------------------------------------------------*/
1177: /**
1178: * Init random generator, sound tables and envelopes
1179: * (called only once when Hatari starts)
1180: */
1181: void Sound_Init(void)
1182: {
1183: /* Build volume/env tables, ... */
1184: Ym2149_Init();
1.1.1.14 root 1185:
1.1.1.11 root 1186: Sound_Reset();
1.1.1.7 root 1187: }
1188:
1189:
1190: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1191: /**
1.1.1.12 root 1192: * Reset the sound emulation (called from Reset_ST() in reset.c)
1.1.1.11 root 1193: */
1.1.1.7 root 1194: void Sound_Reset(void)
1195: {
1.1.1.11 root 1196: /* Lock audio system before accessing variables which are used by the
1197: * callback function, too! */
1198: Audio_Lock();
1.1.1.9 root 1199:
1.1.1.11 root 1200: /* Clear sound mixing buffer: */
1.1.1.12 root 1201: memset(MixBuffer, 0, sizeof(MixBuffer));
1.1.1.7 root 1202:
1.1.1.11 root 1203: /* Clear cycle counts, buffer index and register '13' flags */
1204: Cycles_SetCounter(CYCLES_COUNTER_SOUND, 0);
1.1.1.14 root 1205: bEnvelopeFreqFlag = false;
1206:
1.1.1.11 root 1207: CompleteSndBufIdx = 0;
1208: /* We do not start with 0 here to fake some initial samples: */
1209: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1210: ActiveSndBufIdx = nGeneratedSamples % MIXBUFFER_SIZE;
1.1.1.16 root 1211: SamplesPerFrame = SAMPLES_PER_FRAME;
1212: CurrentSamplesNb = 0;
1213: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1214: //fprintf ( stderr , "Sound_Reset SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1215: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.7 root 1216:
1.1.1.12 root 1217: Ym2149_Reset();
1.1.1.9 root 1218:
1.1.1.11 root 1219: Audio_Unlock();
1.1.1.7 root 1220: }
1221:
1222:
1223: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1224: /**
1225: * Reset the sound buffer index variables.
1.1.1.16 root 1226: * Very important : this function should only be called by setting
1227: * Sound_BufferIndexNeedReset=true ; sound buffer index should be reset
1228: * only after the sound for the whole VBL was updated (CurrentSamplesNb returns to 0)
1229: * else it will alter the value of DMA Frame Count ($ff8909/0b/0d) and
1230: * could cause crashes in some programs.
1.1.1.11 root 1231: */
1.1.1.9 root 1232: void Sound_ResetBufferIndex(void)
1.1.1.7 root 1233: {
1.1.1.11 root 1234: Audio_Lock();
1235: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1236: ActiveSndBufIdx = (CompleteSndBufIdx + nGeneratedSamples) % MIXBUFFER_SIZE;
1.1.1.16 root 1237: SamplesPerFrame = SAMPLES_PER_FRAME;
1238: CurrentSamplesNb = 0;
1239: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1240: //fprintf ( stderr , "Sound_ResetBufferIndex SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1241: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.11 root 1242: Audio_Unlock();
1.1.1.7 root 1243: }
1244:
1245:
1246: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1247: /**
1248: * Save/Restore snapshot of local variables('MemorySnapShot_Store' handles type)
1249: */
1.1.1.12 root 1250: void Sound_MemorySnapShot_Capture(bool bSave)
1.1.1.7 root 1251: {
1.1.1.11 root 1252: /* Save/Restore details */
1.1.1.12 root 1253: MemorySnapShot_Store(&stepA, sizeof(stepA));
1254: MemorySnapShot_Store(&stepB, sizeof(stepB));
1255: MemorySnapShot_Store(&stepC, sizeof(stepC));
1256: MemorySnapShot_Store(&posA, sizeof(posA));
1257: MemorySnapShot_Store(&posB, sizeof(posB));
1258: MemorySnapShot_Store(&posC, sizeof(posC));
1259:
1260: MemorySnapShot_Store(&mixerTA, sizeof(mixerTA));
1261: MemorySnapShot_Store(&mixerTB, sizeof(mixerTB));
1262: MemorySnapShot_Store(&mixerTC, sizeof(mixerTC));
1263: MemorySnapShot_Store(&mixerNA, sizeof(mixerNA));
1264: MemorySnapShot_Store(&mixerNB, sizeof(mixerNB));
1265: MemorySnapShot_Store(&mixerNC, sizeof(mixerNC));
1266:
1267: MemorySnapShot_Store(&noiseStep, sizeof(noiseStep));
1268: MemorySnapShot_Store(&noisePos, sizeof(noisePos));
1269: MemorySnapShot_Store(¤tNoise, sizeof(currentNoise));
1270: MemorySnapShot_Store(&RndRack, sizeof(RndRack));
1271:
1272: MemorySnapShot_Store(&envStep, sizeof(envStep));
1273: MemorySnapShot_Store(&envPos, sizeof(envPos));
1274: MemorySnapShot_Store(&envShape, sizeof(envShape));
1.1.1.14 root 1275:
1.1.1.12 root 1276: MemorySnapShot_Store(&EnvMask3Voices, sizeof(EnvMask3Voices));
1277: MemorySnapShot_Store(&Vol3Voices, sizeof(Vol3Voices));
1.1.1.14 root 1278:
1.1.1.12 root 1279: MemorySnapShot_Store(SoundRegs, sizeof(SoundRegs));
1280:
1.1.1.14 root 1281: // MemorySnapShot_Store(&YmVolumeMixing, sizeof(YmVolumeMixing));
1282: // MemorySnapShot_Store(&UseLowPassFilter, sizeof(UseLowPassFilter));
1.1.1.7 root 1283: }
1284:
1285:
1286: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1287: /**
1288: * Find how many samples to generate and store in 'nSamplesToGenerate'
1289: * Also update sound cycles counter to store how many we actually did
1290: * so generates set amount each frame.
1.1.1.16 root 1291: * If FillFrame is true, this means we reach the end of the VBL and me must
1292: * add as many samples as necessary to get a total of SamplesPerFrame
1293: * for this VBL.
1.1.1.11 root 1294: */
1.1.1.16 root 1295: static int Sound_SetSamplesPassed(bool FillFrame)
1.1 root 1296: {
1.1.1.11 root 1297: int nSoundCycles;
1.1.1.16 root 1298: int SamplesToGenerate; /* How many samples are needed for this time-frame */
1299:
1300: nSoundCycles = Cycles_GetCounter(CYCLES_COUNTER_VIDEO);
1.1.1.11 root 1301:
1.1.1.16 root 1302: /* example : 160256 cycles per VBL, 44Khz = 882 samples per VBL at 50 Hz */
1303: /* 882/160256 samples per cpu clock cycle */
1.1.1.11 root 1304:
1.1.1.16 root 1305: /* Total number of samples that we should have at this point of the VBL */
1306: SamplesToGenerate = nSoundCycles * SamplesPerFrame
1307: / ClocksTimings_GetCyclesPerVBL ( ConfigureParams.System.nMachineType , nScreenRefreshRate );
1.1.1.11 root 1308:
1.1.1.16 root 1309: //if (SamplesToGenerate > SamplesPerFrame )
1310: //fprintf ( stderr , "over run %d %d\n" , SamplesPerFrame , SamplesToGenerate );
1.1.1.11 root 1311:
1.1.1.16 root 1312: if (SamplesToGenerate > SamplesPerFrame)
1313: SamplesToGenerate = SamplesPerFrame;
1314:
1315: SamplesToGenerate -= CurrentSamplesNb; /* don't count samples that were already generated up to now */
1316: if ( SamplesToGenerate < 0 )
1317: SamplesToGenerate = 0;
1318:
1319:
1320: /* If we're called from the VBL interrupt (FillFrame==true), we must ensure we have */
1321: /* an exact total of SamplesPerFrame samples during a full VBL (we take into account */
1322: /* the samples that were already generated during this VBL) */
1323: if ( FillFrame )
1324: {
1325: SamplesToGenerate = SamplesPerFrame - CurrentSamplesNb; /* how many samples are missing to reach SamplesPerFrame */
1326: if ( SamplesToGenerate < 0 )
1327: SamplesToGenerate = 0;
1328: }
1.1.1.11 root 1329:
1.1.1.16 root 1330: /* Check we don't fill the sound's ring buffer before it's played by Audio_Callback() */
1331: /* This should never happen, except if the system suffers major slowdown due to other */
1332: /* processes or if we run in fast forward mode. */
1333: /* In the case of slowdown, we set Sound_BufferIndexNeedReset to "resync" the working */
1334: /* buffer's index ActiveSndBufIdx with the system buffer's index CompleteSndBufIdx. */
1335: /* In the case of fast forward, we do nothing here, Sound_BufferIndexNeedReset will be */
1336: /* set when the user exits fast forward mode. */
1337: if ( ( SamplesToGenerate > MIXBUFFER_SIZE - nGeneratedSamples ) && ( ConfigureParams.System.bFastForward == false )
1338: && ( ConfigureParams.Sound.bEnableSound == true ) )
1.1.1.11 root 1339: {
1.1.1.16 root 1340: Log_Printf ( LOG_WARN , "Your system is too slow, some sound samples were not correctly emulated\n" );
1341: Sound_BufferIndexNeedReset = true;
1.1.1.11 root 1342: }
1.1.1.16 root 1343:
1344: //fprintf ( stderr , "vbl %d hbl %d samp_gen %d / %d frac %lx\n" , nVBLs , nHBL , SamplesToGenerate , SamplesPerFrame , (long int)SamplesPerFrame_unrounded );
1345:
1346: return SamplesToGenerate;
1.1 root 1347: }
1348:
1.1.1.5 root 1349:
1.1.1.2 root 1350: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1351: /**
1352: * Generate samples for all channels during this time-frame
1353: */
1.1.1.16 root 1354: static void Sound_GenerateSamples(int SamplesToGenerate)
1.1 root 1355: {
1.1.1.12 root 1356: int i;
1357: int idx;
1.1.1.14 root 1358:
1.1.1.16 root 1359: if (SamplesToGenerate <= 0)
1.1.1.12 root 1360: return;
1.1.1.14 root 1361:
1.1.1.15 root 1362: if (ConfigureParams.System.nMachineType == MACHINE_FALCON)
1.1.1.11 root 1363: {
1.1.1.16 root 1364: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1365: {
1366: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1367: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1368: }
1369: /* If Falcon emulation, crossbar does the job */
1.1.1.16 root 1370: Crossbar_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.11 root 1371: }
1.1.1.15 root 1372: else if (ConfigureParams.System.nMachineType != MACHINE_ST)
1373: {
1.1.1.16 root 1374: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1375: {
1376: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.18! root 1377: MixBuffer[idx][0] = MixBuffer[idx][1] = YM2149_NextSample();
1.1.1.15 root 1378: }
1.1.1.17 root 1379: /* If Ste or TT emulation, DmaSnd does mixing and filtering */
1.1.1.16 root 1380: DmaSnd_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.15 root 1381: }
1382: else if (ConfigureParams.System.nMachineType == MACHINE_ST)
1383: {
1.1.1.16 root 1384: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1385: {
1386: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1387: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1388: }
1389: }
1.1.1.12 root 1390:
1.1.1.16 root 1391: ActiveSndBufIdx = (ActiveSndBufIdx + SamplesToGenerate) % MIXBUFFER_SIZE;
1392: nGeneratedSamples += SamplesToGenerate;
1393: CurrentSamplesNb += SamplesToGenerate; /* number of samples generated for current VBL */
1.1 root 1394: }
1395:
1.1.1.5 root 1396:
1.1.1.2 root 1397: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1398: /**
1399: * This is called to built samples up until this clock cycle
1.1.1.16 root 1400: * Sound_Update can be called several times during a VBL ; we must ensure
1401: * that we generate exactly SamplesPerFrame samples between 2 calls
1402: * to Sound_Update_VBL.
1.1.1.11 root 1403: */
1.1.1.16 root 1404: void Sound_Update(bool FillFrame)
1.1 root 1405: {
1.1.1.11 root 1406: int OldSndBufIdx = ActiveSndBufIdx;
1.1.1.16 root 1407: int SamplesToGenerate;
1.1.1.5 root 1408:
1.1.1.11 root 1409: /* Make sure that we don't interfere with the audio callback function */
1410: Audio_Lock();
1.1.1.6 root 1411:
1.1.1.16 root 1412: /* Find how many samples to generate */
1413: SamplesToGenerate = Sound_SetSamplesPassed( FillFrame );
1414:
1.1.1.11 root 1415: /* And generate */
1.1.1.16 root 1416: Sound_GenerateSamples( SamplesToGenerate );
1.1 root 1417:
1.1.1.11 root 1418: /* Allow audio callback function to occur again */
1419: Audio_Unlock();
1.1.1.6 root 1420:
1.1.1.11 root 1421: /* Save to WAV file, if open */
1422: if (bRecordingWav)
1.1.1.16 root 1423: WAVFormat_Update(MixBuffer, OldSndBufIdx, SamplesToGenerate);
1.1 root 1424: }
1425:
1.1.1.5 root 1426:
1.1.1.2 root 1427: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1428: /**
1.1.1.16 root 1429: * On the end of each VBL, complete audio buffer to reach SamplesPerFrame samples.
1430: * As Sound_Update(false) could be called several times during the VBL, the audio
1431: * buffer might be already partially filled.
1432: * We must first complete the buffer using the same value of SamplesPerFrame
1433: * by calling Sound_Update(true) ; then we can compute a new value for
1434: * SamplesPerFrame that will be used for the next VBL to come.
1.1.1.11 root 1435: */
1.1.1.5 root 1436: void Sound_Update_VBL(void)
1.1 root 1437: {
1.1.1.16 root 1438: Sound_Update(true); /* generate as many samples as needed to fill this VBL */
1439: //fprintf ( stderr , "vbl done %d %d\n" , SamplesPerFrame , CurrentSamplesNb );
1.1.1.5 root 1440:
1.1.1.16 root 1441: CurrentSamplesNb = 0; /* VBL is complete, reset counter for next VBL */
1442:
1443: /*Compute a fractional equivalent of SamplesPerFrame for the next VBL, to avoid rounding propagation */
1444: SamplesPerFrame_unrounded += (yms64) ClocksTimings_GetSamplesPerVBL ( ConfigureParams.System.nMachineType ,
1445: nScreenRefreshRate , nAudioFrequency );
1446: SamplesPerFrame = SamplesPerFrame_unrounded >> 28; /* use integer part */
1447: SamplesPerFrame_unrounded &= 0x0fffffff; /* keep fractional part in the lower 28 bits */
1448:
1449: /* Reset sound buffer if needed (after pause, fast forward, slow system, ...) */
1450: if ( Sound_BufferIndexNeedReset )
1451: {
1452: Sound_ResetBufferIndex ();
1453: Sound_BufferIndexNeedReset = false;
1454: }
1455:
1456: /* Record AVI audio frame is necessary */
1.1.1.15 root 1457: if ( bRecordingAvi )
1458: {
1.1.1.16 root 1459: int Len;
1.1.1.15 root 1460:
1.1.1.16 root 1461: Len = ActiveSndBufIdx - ActiveSndBufIdxAvi; /* number of generated samples for this frame */
1462: if ( Len < 0 )
1463: Len += MIXBUFFER_SIZE; /* end of ring buffer was reached */
1.1.1.15 root 1464:
1.1.1.16 root 1465: Avi_RecordAudioStream ( MixBuffer , ActiveSndBufIdxAvi , Len );
1.1.1.15 root 1466: }
1467:
1.1.1.16 root 1468: ActiveSndBufIdxAvi = ActiveSndBufIdx; /* save new position for next AVI audio frame */
1469:
1.1.1.11 root 1470: /* Clear write to register '13', used for YM file saving */
1.1.1.14 root 1471: bEnvelopeFreqFlag = false;
1.1 root 1472: }
1473:
1474:
1.1.1.2 root 1475: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1476: /**
1477: * Start recording sound, as .YM or .WAV output
1478: */
1.1.1.12 root 1479: bool Sound_BeginRecording(char *pszCaptureFileName)
1.1 root 1480: {
1.1.1.12 root 1481: bool bRet;
1.1.1.7 root 1482:
1.1.1.11 root 1483: if (!pszCaptureFileName || strlen(pszCaptureFileName) <= 3)
1484: {
1485: Log_Printf(LOG_ERROR, "Illegal sound recording file name!\n");
1.1.1.14 root 1486: return false;
1.1.1.11 root 1487: }
1488:
1489: /* Did specify .YM or .WAV? If neither report error */
1490: if (File_DoesFileExtensionMatch(pszCaptureFileName,".ym"))
1491: bRet = YMFormat_BeginRecording(pszCaptureFileName);
1492: else if (File_DoesFileExtensionMatch(pszCaptureFileName,".wav"))
1493: bRet = WAVFormat_OpenFile(pszCaptureFileName);
1494: else
1495: {
1496: Log_AlertDlg(LOG_ERROR, "Unknown Sound Recording format.\n"
1497: "Please specify a .YM or .WAV output file.");
1.1.1.14 root 1498: bRet = false;
1.1.1.11 root 1499: }
1500:
1501: return bRet;
1.1 root 1502: }
1503:
1.1.1.5 root 1504:
1.1.1.2 root 1505: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1506: /**
1507: * End sound recording
1508: */
1.1.1.7 root 1509: void Sound_EndRecording(void)
1.1 root 1510: {
1.1.1.11 root 1511: /* Stop sound recording and close files */
1512: if (bRecordingYM)
1513: YMFormat_EndRecording();
1514: if (bRecordingWav)
1515: WAVFormat_CloseFile();
1.1 root 1516: }
1517:
1.1.1.6 root 1518:
1.1.1.2 root 1519: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1520: /**
1521: * Are we recording sound data?
1522: */
1.1.1.12 root 1523: bool Sound_AreWeRecording(void)
1.1 root 1524: {
1.1.1.11 root 1525: return (bRecordingYM || bRecordingWav);
1.1 root 1526: }
1.1.1.12 root 1527:
1.1.1.16 root 1528:
1529: /*-----------------------------------------------------------------------*/
1530: /**
1531: * Rebuild volume conversion table
1532: */
1533: void Sound_SetYmVolumeMixing(void)
1534: {
1535: /* Build the volume conversion table */
1536: Ym2149_BuildVolumeTable();
1537: }
1538:
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