![[BACK]](/cvsweb/icons/back.gif){kind=icon}
Return to sound.c CVS log ![[TXT]](/cvsweb/icons/text.gif){kind=icon}
![[DIR]](/cvsweb/icons/dir.gif){kind=icon}
Up to [HATARI the Atari ST Emulator] / hatari / src|
Return to sound.c CVS log | Up to [HATARI the Atari ST Emulator] / hatari / src |
1.1 root 1: /*
1.1.1.5 root 2: Hatari - sound.c
3:
1.1.1.19 root 4: This file is distributed under the GNU General Public License, version 2
5: or at 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:
1.1.1.20 root 21: The new version of the sound core uses/used 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)
1.1.1.20 root 23: (not used since Hatari 1.1.0)
1.1.1.12 root 24: - 5 bits volume table and 16*16*16 combinations of all volume are from Sc68 by Benjamin Gerard
25: - 4 bits to 5 bits volume interpolation from 16*16*16 to 32*32*32 from YM blep synthesis by Antti Lankila
26:
1.1.1.16 root 27: Special case for per==0 : as measured on a real STF, when tone/noise/env's per==0, we get
28: the same sound output as when per==1.
29:
30:
1.1 root 31: */
1.1.1.12 root 32:
33: /* 2008/05/05 [NP] Fix case where period is 0 for noise, sound or envelope. */
34: /* In that case, a real ST sounds as if period was in fact 1. */
35: /* (fix buggy sound replay in ESwat that set volume<0 and trigger */
36: /* a badly initialised envelope with envper=0). */
37: /* 2008/07/27 [NP] Better separation between accesses to the YM hardware registers */
38: /* and the sound rendering routines. Use Sound_WriteReg() to pass */
39: /* all writes to the sound rendering functions. This allows to */
1.1.1.19 root 40: /* have sound.c independent of psg.c (to ease replacement of */
1.1.1.12 root 41: /* sound.c by another rendering method). */
42: /* 2008/08/02 [NP] Initial convert of Ym2149Ex.cpp from C++ to C. */
43: /* Remove unused part of the code (StSound specific). */
44: /* 2008/08/09 [NP] Complete integration of StSound routines into sound.c */
45: /* Set EnvPer=3 if EnvPer<3 (ESwat buggy replay). */
46: /* 2008/08/13 [NP] StSound was generating samples in the range 0-32767, instead */
47: /* of really signed samples between -32768 and 32767, which could */
48: /* give incorrect results in many case. */
49: /* 2008/09/06 [NP] Use sc68 volumes table for a more accurate mixing of the voices */
50: /* All volumes are converted to 5 bits and the table contains */
51: /* 32*32*32 values. Samples are signed and centered to get the */
52: /* biggest amplitude possible. */
53: /* Faster mixing routines for tone+volume+envelope (don't use */
54: /* StSound's version anymore, it gave problem with some GCC). */
55: /* 2008/09/17 [NP] Add ym_normalise_5bit_table to normalise the 32*32*32 table and */
56: /* to optionally center 16 bit signed sample. */
57: /* Possibility to mix volumes using a table measured on ST or a */
58: /* linear mean of the 3 channels' volume. */
59: /* Default mixing set to YM_LINEAR_MIXING. */
60: /* 2008/10/14 [NP] Full support for 5 bits volumes : envelopes are generated with */
61: /* 32 volumes per pattern as on a real YM-2149. Fixed volumes */
62: /* on 4 bits are converted to their 5 bits equivalent. This should */
63: /* give the maximum accuracy possible when computing volumes. */
64: /* New version of Ym2149_EnvStepCompute to handle 5 bits volumes. */
65: /* Function YM2149_EnvBuild to compute the 96 volumes that define */
66: /* a single envelope (32 initial volumes, then 64 repeated values).*/
67: /* 2008/10/26 [NP] Correctly save/restore all necessary variables in */
68: /* Sound_MemorySnapShot_Capture. */
69: /* 2008/11/23 [NP] Clean source, remove old sound core. */
1.1.1.17 root 70: /* 2011/11/03 [DS] Stereo DC filtering which accounts for DMA sound. */
1.1.1.12 root 71:
72:
73:
1.1.1.14 root 74: const char Sound_fileid[] = "Hatari sound.c : " __DATE__ " " __TIME__;
1.1.1.5 root 75:
76: #include <SDL_types.h>
1.1 root 77:
78: #include "main.h"
79: #include "audio.h"
1.1.1.10 root 80: #include "cycles.h"
1.1.1.21! root 81: #include "m68000.h"
1.1.1.15 root 82: #include "configuration.h"
1.1.1.9 root 83: #include "dmaSnd.h"
1.1.1.15 root 84: #include "crossbar.h"
1.1 root 85: #include "file.h"
1.1.1.15 root 86: #include "cycInt.h"
1.1.1.8 root 87: #include "log.h"
1.1 root 88: #include "memorySnapShot.h"
89: #include "psg.h"
90: #include "sound.h"
1.1.1.16 root 91: #include "screen.h"
1.1 root 92: #include "video.h"
93: #include "wavFormat.h"
94: #include "ymFormat.h"
1.1.1.15 root 95: #include "avi_record.h"
1.1.1.16 root 96: #include "clocks_timings.h"
1.1 root 97:
98:
99:
1.1.1.12 root 100: /*--------------------------------------------------------------*/
101: /* Definition of the possible envelopes shapes (using 5 bits) */
102: /*--------------------------------------------------------------*/
103:
104: #define ENV_GODOWN 0 /* 31 -> 0 */
105: #define ENV_GOUP 1 /* 0 -> 31 */
106: #define ENV_DOWN 2 /* 0 -> 0 */
107: #define ENV_UP 3 /* 31 -> 31 */
108:
109: /* To generate an envelope, we first use block 0, then we repeat blocks 1 and 2 */
110: static const int YmEnvDef[ 16 ][ 3 ] = {
111: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 0 \___ */
112: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 1 \___ */
113: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 2 \___ */
114: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 3 \___ */
115: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 4 /___ */
116: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 5 /___ */
117: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 6 /___ */
118: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* 7 /___ */
119: { ENV_GODOWN, ENV_GODOWN, ENV_GODOWN } , /* 8 \\\\ */
120: { ENV_GODOWN, ENV_DOWN, ENV_DOWN } , /* 9 \___ */
121: { ENV_GODOWN, ENV_GOUP, ENV_GODOWN } , /* A \/\/ */
122: { ENV_GODOWN, ENV_UP, ENV_UP } , /* B \--- */
123: { ENV_GOUP, ENV_GOUP, ENV_GOUP } , /* C //// */
124: { ENV_GOUP, ENV_UP, ENV_UP } , /* D /--- */
125: { ENV_GOUP, ENV_GODOWN, ENV_GOUP } , /* E /\/\ */
126: { ENV_GOUP, ENV_DOWN, ENV_DOWN } , /* F /___ */
127: };
128:
129:
130: /* Buffer to store the 16 envelopes built from YmEnvDef */
131: static ymu16 YmEnvWaves[ 16 ][ 32 * 3 ]; /* 16 envelopes with 3 blocks of 32 volumes */
1.1.1.3 root 132:
1.1 root 133:
1.1.1.12 root 134:
135: /*--------------------------------------------------------------*/
136: /* Definition of the volumes tables (using 5 bits) and of the */
137: /* mixing parameters for the 3 voices. */
138: /*--------------------------------------------------------------*/
139:
140: /* 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) */
141: /* Vol 0 should be 310 when measuread as a voltage, but we set it to 0 in order to have a volume=0 matching */
142: /* the 0 level of a 16 bits unsigned sample (no sound output) */
143: static const ymu16 ymout1c5bit[ 32 ] =
144: {
145: 0 /*310*/, 369, 438, 521, 619, 735, 874, 1039,
146: 1234, 1467, 1744, 2072, 2463, 2927, 3479, 4135,
147: 4914, 5841, 6942, 8250, 9806,11654,13851,16462,
148: 19565,23253,27636,32845,39037,46395,55141,65535
1.1 root 149: };
150:
1.1.1.12 root 151: /* Convert a constant 4 bits volume to the internal 5 bits value : */
1.1.1.18 root 152: /* volume5=volume4*2+1, except for volumes 0 and 1 which remain 0 and 1, */
1.1.1.12 root 153: /* in order to map [0,15] into [0,31] (O must remain 0, and 15 must give 31) */
1.1.1.18 root 154: static const ymu16 YmVolume4to5[ 16 ] = { 0,1,5,7,9,11,13,15,17,19,21,23,25,27,29,31 };
1.1 root 155:
1.1.1.12 root 156: /* Table of unsigned 4 bit D/A output level for 3 channels as measured on a real ST */
1.1.1.18 root 157: static ymu16 volumetable_original[16][16][16] =
1.1.1.12 root 158: #include "ym2149_fixed_vol.h"
1.1.1.5 root 159:
1.1.1.12 root 160: /* Corresponding table interpolated to 5 bit D/A output level (16 bits unsigned) */
1.1.1.18 root 161: static ymu16 ymout5_u16[32][32][32];
1.1.1.12 root 162:
163: /* Same table, after conversion to signed results (same pointer, with different type) */
164: static yms16 *ymout5 = (yms16 *)ymout5_u16;
165:
166:
167:
168: /*--------------------------------------------------------------*/
169: /* Other constants / macros */
170: /*--------------------------------------------------------------*/
171:
172: /* Number of generated samples per frame (eg. 44Khz=882) */
1.1.1.16 root 173: #define SAMPLES_PER_FRAME (nAudioFrequency/nScreenRefreshRate)
1.1.1.12 root 174:
175: /* Current sound replay freq (usually 44100 Hz) */
1.1.1.14 root 176: #define YM_REPLAY_FREQ nAudioFrequency
1.1.1.12 root 177:
1.1.1.16 root 178: /* YM-2149 clock on all Atari models is 2 MHz */
179: #define YM_ATARI_CLOCK (MachineClocks.YM_Freq)
1.1.1.12 root 180:
181:
182: /* Merge/read the 3 volumes in a single integer (5 bits per volume) */
183: #define YM_MERGE_VOICE(C,B,A) ( (C)<<10 | (B)<<5 | A )
184: #define YM_MASK_1VOICE 0x1f
185: #define YM_MASK_A 0x1f
186: #define YM_MASK_B (0x1f<<5)
187: #define YM_MASK_C (0x1f<<10)
188:
189:
190: /* Constants for YM2149_Normalise_5bit_Table */
191: #define YM_OUTPUT_LEVEL 0x7fff /* amplitude of the final signal (0..65535 if centered, 0..32767 if not) */
1.1.1.14 root 192: #define YM_OUTPUT_CENTERED false
1.1.1.12 root 193:
194:
195:
196: /*--------------------------------------------------------------*/
197: /* Variables for the YM2149 emulator (need to be saved and */
198: /* restored in memory snapshots) */
199: /*--------------------------------------------------------------*/
200:
201: static ymu32 stepA , stepB , stepC;
202: static ymu32 posA , posB , posC;
203: static ymu32 mixerTA , mixerTB , mixerTC;
204: static ymu32 mixerNA , mixerNB , mixerNC;
205:
206: static ymu32 noiseStep;
207: static ymu32 noisePos;
208: static ymu32 currentNoise;
209: static ymu32 RndRack; /* current random seed */
210:
211: static ymu32 envStep;
212: static ymu32 envPos;
213: static int envShape;
214:
215: static ymu16 EnvMask3Voices = 0; /* mask is 0x1f for voices having an active envelope */
216: static ymu16 Vol3Voices = 0; /* volume 0-0x1f for voices having a constant volume */
217: /* volume is set to 0 if voice has an envelope in EnvMask3Voices */
218:
219:
220: /* Global variables that can be changed/read from other parts of Hatari */
221: Uint8 SoundRegs[ 14 ];
222:
1.1.1.16 root 223: int YmVolumeMixing = YM_TABLE_MIXING;
1.1.1.14 root 224: bool UseLowPassFilter = false;
1.1.1.12 root 225:
226: bool bEnvelopeFreqFlag; /* Cleared each frame for YM saving */
227:
228: Sint16 MixBuffer[MIXBUFFER_SIZE][2];
229: int nGeneratedSamples; /* Generated samples since audio buffer update */
230: static int ActiveSndBufIdx; /* Current working index into above mix buffer */
1.1.1.16 root 231: static int ActiveSndBufIdxAvi; /* Current working index to save an AVI audio frame */
232:
233: static yms64 SamplesPerFrame_unrounded = 0; /* Number of samples for the current VBL, with simulated fractional part */
234: static int SamplesPerFrame; /* Number of samples to generate for the current VBL */
235: static int CurrentSamplesNb = 0; /* Number of samples already generated for the current VBL */
1.1.1.12 root 236:
1.1.1.16 root 237: bool Sound_BufferIndexNeedReset = false;
1.1.1.12 root 238:
239:
240: /*--------------------------------------------------------------*/
241: /* Local functions prototypes */
242: /*--------------------------------------------------------------*/
243:
1.1.1.17 root 244: static ymsample LowPassFilter (ymsample x0);
1.1.1.18 root 245: static ymsample PWMaliasFilter (ymsample x0);
1.1.1.12 root 246:
1.1.1.18 root 247: static void interpolate_volumetable (ymu16 volumetable[32][32][32]);
1.1.1.12 root 248:
1.1.1.18 root 249: static void YM2149_BuildModelVolumeTable(ymu16 volumetable[32][32][32]);
250: static void YM2149_BuildLinearVolumeTable(ymu16 volumetable[32][32][32]);
1.1.1.12 root 251: static void YM2149_Normalise_5bit_Table(ymu16 *in_5bit , yms16 *out_5bit, unsigned int Level, bool DoCenter);
252:
253: static void YM2149_EnvBuild (void);
1.1.1.16 root 254: static void Ym2149_BuildVolumeTable (void);
1.1.1.12 root 255: static void Ym2149_Init (void);
256: static void Ym2149_Reset (void);
257:
258: static ymu32 YM2149_RndCompute (void);
259: static ymu32 Ym2149_ToneStepCompute (ymu8 rHigh , ymu8 rLow);
260: static ymu32 Ym2149_NoiseStepCompute (ymu8 rNoise);
261: static ymu32 Ym2149_EnvStepCompute (ymu8 rHigh , ymu8 rLow);
262: static ymsample YM2149_NextSample (void);
263:
1.1.1.16 root 264: static int Sound_SetSamplesPassed(bool FillFrame);
265: static void Sound_GenerateSamples(int SamplesToGenerate);
1.1.1.12 root 266:
267:
268:
269: /*--------------------------------------------------------------*/
1.1.1.17 root 270: /* DC Adjuster */
1.1.1.12 root 271: /*--------------------------------------------------------------*/
272:
1.1.1.17 root 273: /**
274: * 6dB/octave first order HPF fc = (1.0-0.998)*44100/(2.0*pi)
275: * Z pole = 0.99804 --> FS = 44100 Hz : fc=13.7 Hz (11 Hz meas)
276: * a = (int32_t)(32768.0*(1.0 - pole)) : a = 64 !!!
277: * Input range: -32768 to 32767 Maximum step: +65536 or -65472
278: */
279: ymsample Subsonic_IIR_HPF_Left(ymsample x0)
1.1.1.12 root 280: {
1.1.1.17 root 281: static yms32 x1 = 0, y1 = 0, y0 = 0;
1.1.1.14 root 282:
1.1.1.17 root 283: y1 += ((x0 - x1)<<15) - (y0<<6); /* 64*y0 */
284: y0 = y1>>15;
285: x1 = x0;
1.1.1.12 root 286:
1.1.1.17 root 287: return y0;
1.1.1.12 root 288: }
289:
290:
1.1.1.17 root 291: ymsample Subsonic_IIR_HPF_Right(ymsample x0)
1.1 root 292: {
1.1.1.17 root 293: static yms32 x1 = 0, y1 = 0, y0 = 0;
1.1.1.12 root 294:
1.1.1.17 root 295: y1 += ((x0 - x1)<<15) - (y0<<6); /* 64*y0 */
296: y0 = y1>>15;
297: x1 = x0;
1.1.1.12 root 298:
1.1.1.17 root 299: return y0;
1.1.1.12 root 300: }
301:
302:
1.1.1.17 root 303: /*--------------------------------------------------------------*/
304: /* Low Pass Filter routines. */
305: /*--------------------------------------------------------------*/
1.1.1.12 root 306:
1.1.1.17 root 307: /**
308: * Get coefficients for different Fs (C10 is in ST only):
309: * Wc = 2*M_PI*4895.1;
310: * Fs = 44100;
311: * warp = Wc/tanf((Wc/2)/Fs);
312: * b = Wc/(warp+Wc);
313: * a = (Wc-warp)/(warp+Wc);
314: *
315: * #define B_z (yms32)( 0.2667*(1<<15))
316: * #define A_z (yms32)(-0.4667*(1<<15))
317: *
318: * y0 = (B_z*(x0 + x1) - A_z*y0) >> 15;
319: * x1 = x0;
320: *
321: * The Lowpass Filter formed by C10 = 0.1 uF
322: * and
1.1.1.18 root 323: * R8=1k // 1k*(65119-46602)/65119 // R9=10k // R10=5.1k //
1.1.1.17 root 324: * (R12=470)*(100=Q1_HFE) = 206.865 ohms when YM2149 is High
325: * and
326: * R8=1k // R9=10k // R10=5.1k // (R12=470)*(100=Q1_HFE)
327: * = 759.1 ohms when YM2149 is Low
328: * High corner is 1/(2*pi*(0.1*10e-6)*206.865) fc = 7693.7 Hz
329: * Low corner is 1/(2*pi*(0.1*10e-6)*795.1) fc = 2096.6 Hz
330: * Notes:
331: * - using STF reference designators R8 R9 R10 C10 (from dec 1986 schematics)
332: * - using corresponding numbers from psgstrep and psgquart
333: * - 65119 is the largest value in Paulo's psgstrep table
334: * - 46602 is the largest value in Paulo's psgquart table
335: * - this low pass filter uses the highest cutoff frequency
336: * on the STf (a slightly lower frequency is reasonable).
337: *
338: * A first order lowpass filter with a high cutoff frequency
339: * is used when the YM2149 pulls high, and a lowpass filter
340: * with a low cutoff frequency is used when R8 pulls low.
341: */
342: static ymsample LowPassFilter(ymsample x0)
343: {
344: static yms32 y0 = 0, x1 = 0;
345:
346: if (x0 >= y0)
347: /* YM Pull up: fc = 7586.1 Hz (44.1 KHz), fc = 8257.0 Hz (48 KHz) */
348: y0 = (3*(x0 + x1) + (y0<<1)) >> 3;
349: else
350: /* R8 Pull down: fc = 1992.0 Hz (44.1 KHz), fc = 2168.0 Hz (48 KHz) */
351: y0 = ((x0 + x1) + (6*y0)) >> 3;
1.1.1.12 root 352:
1.1.1.17 root 353: x1 = x0;
354: return y0;
1.1.1.12 root 355: }
356:
1.1.1.18 root 357: /**
358: * This piecewise selective filter works by filtering the falling
359: * edge of a sampled pulse-wave differently from the rising edge.
360: *
361: * Piecewise selective filtering is effective because harmonics on
362: * one part of a wave partially define harmonics on other portions.
363: *
364: * Piecewise selective filtering can efficiently reduce aliasing
365: * with minimal harmonic removal.
366: *
367: * I disclose this information into the public domain so that it
368: * cannot be patented. May 23 2012 David Savinkoff.
369: */
370: static ymsample PWMaliasFilter(ymsample x0)
371: {
372: static yms32 y0 = 0, x1 = 0;
373:
374: if (x0 >= y0)
375: /* YM Pull up */
376: y0 = x0;
377: else
378: /* R8 Pull down */
379: y0 = (3*(x0 + x1) + (y0<<1)) >> 3;
380:
381: x1 = x0;
382: return y0;
383: }
384:
1.1.1.12 root 385:
386:
387: /*--------------------------------------------------------------*/
388: /* Build the volume conversion table used to simulate the */
389: /* behaviour of DAC used with the YM2149 in the atari ST. */
390: /* The final 32*32*32 table is built using a 16*16*16 table */
391: /* of all possible fixed volume combinations on a ST. */
392: /*--------------------------------------------------------------*/
393:
1.1.1.18 root 394: static void interpolate_volumetable(ymu16 volumetable[32][32][32])
1.1.1.12 root 395: {
1.1.1.18 root 396: int i, j, k;
1.1.1.12 root 397:
1.1.1.18 root 398: for (i = 1; i < 32; i += 2) { /* Copy 16 Panels to make a Block */
399: for (j = 1; j < 32; j += 2) { /* Copy 16 Rows to make a Panel */
400: for (k = 1; k < 32; k += 2) { /* Copy 16 Elements to make a Row */
401: volumetable[i][j][k] = volumetable_original[(i-1)/2][(j-1)/2][(k-1)/2];
1.1.1.12 root 402: }
1.1.1.18 root 403: volumetable[i][j][0] = volumetable[i][j][1]; /* Move 0th Element */
404: volumetable[i][j][1] = volumetable[i][j][3]; /* Move 1st Element */
405: /* Interpolate 3rd Element */
406: volumetable[i][j][3] = (ymu16)(0.5 + sqrt((double)volumetable[i][j][1] * volumetable[i][j][5]));
407: for (k = 2; k < 32; k += 2) /* Interpolate Even Elements */
408: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][j][k-1] * volumetable[i][j][k+1]));
1.1.1.12 root 409: }
1.1.1.18 root 410: for (k = 0; k < 32; k++) {
411: volumetable[i][0][k] = volumetable[i][1][k]; /* Move 0th Row */
412: volumetable[i][1][k] = volumetable[i][3][k]; /* Move 1st Row */
413: /* Interpolate 3rd Row */
414: volumetable[i][3][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][1][k] * volumetable[i][5][k]));
415: }
416: for (j = 2; j < 32; j += 2) /* Interpolate Even Rows */
417: for (k = 0; k < 32; k++)
418: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i][j-1][k] * volumetable[i][j+1][k]));
1.1.1.11 root 419: }
1.1.1.18 root 420: for (j = 0; j < 32; j++)
421: for (k = 0; k < 32; k++) {
422: volumetable[0][j][k] = volumetable[1][j][k]; /* Move 0th Panel */
423: volumetable[1][j][k] = volumetable[3][j][k]; /* Move 1st Panel */
424: /* Interpolate 3rd Panel */
425: volumetable[3][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[1][j][k] * volumetable[5][j][k]));
426: }
427: for (i = 2; i < 32; i += 2) /* Interpolate Even Panels */
428: for (j = 0; j < 32; j++) /* Interpolate Even Panels */
429: for (k = 0; k < 32; k++)
430: volumetable[i][j][k] = (ymu16)(0.5 + sqrt((double)volumetable[i-1][j][k] * volumetable[i+1][j][k]));
1.1 root 431: }
432:
1.1.1.5 root 433:
1.1.1.12 root 434:
435:
1.1.1.2 root 436: /*-----------------------------------------------------------------------*/
1.1.1.11 root 437: /**
1.1.1.12 root 438: * Build a linear version of the conversion table.
439: * We use the mean of the 3 volumes converted to 16 bit values
440: * (each value of ymout1c5bit is in [0,65535])
1.1.1.11 root 441: */
1.1.1.12 root 442:
1.1.1.18 root 443: static void YM2149_BuildLinearVolumeTable(ymu16 volumetable[32][32][32])
1.1 root 444: {
1.1.1.12 root 445: int i, j, k;
446:
447: for (i = 0; i < 32; i++)
448: for (j = 0; j < 32; j++)
449: for (k = 0; k < 32; k++)
1.1.1.18 root 450: volumetable[i][j][k] = (ymu16)( ((ymu32)ymout1c5bit[i] + ymout1c5bit[j] + ymout1c5bit[k]) / 3);
1.1.1.12 root 451: }
452:
453:
454:
1.1 root 455:
1.1.1.12 root 456: /*-----------------------------------------------------------------------*/
457: /**
1.1.1.17 root 458: * Build a circuit analysed version of the conversion table.
459: * David Savinkoff designed this algorithm by analysing data
460: * measured by Paulo Simoes and Benjamin Gerard.
461: * The numbers are arrived at by assuming a current steering
462: * resistor ladder network and using the voltage divider rule.
1.1.1.18 root 463: *
464: * If one looks at the ST schematic of the YM2149, one sees
465: * three sound pins tied together and attached to a 1000 ohm
466: * resistor (1k) that has the other end grounded.
467: * The 1k resistor is also in parallel with a 0.1 microfarad
468: * capacitor (on the Atari ST, not STE or others). The voltage
469: * developed across the 1K resistor is the output voltage which
470: * I call Vout.
471: *
1.1.1.19 root 472: * The output of the YM2149 is modelled well as pullup resistors.
1.1.1.18 root 473: * Thus, the three sound pins are seen as three parallel
474: * computer-controlled, adjustable pull-up resistors.
475: * To emulate the output of the YM2149, one must determine the
476: * resistance values of the YM2149 relative to the 1k resistor,
477: * which is done by the 'math model'.
478: *
479: * The AC + DC math model is:
480: *
481: * (MaxVol*WARP) / (1.0 + 1.0/(conductance_[i]+conductance_[j]+conductance_[k]))
482: * or
483: * (MaxVol*WARP) / (1.0 + 1.0/( 1/Ra +1/Rb +1/Rc )) , Ra = channel A resistance
484: *
485: * Note that the first 1.0 in the formula represents the
486: * normalized 1k resistor (1.0 * 1000 ohms = 1k).
487: *
488: * The YM2149 DC component model represents the output voltage
489: * filtered of high frequency AC component, but DC component
490: * remains.
491: * The YM2149 DC component mode treats the voltage exactly as if
492: * it were low pass filtered. This is more than what is required
493: * to make 'quartet mode sound'. Simplicity leads to Generality!
494: *
495: * The DC component model model is:
496: *
497: * (MaxVol*WARP) / (2.0 + 1.0/( 1/Ra + 1/Rb + 1/Rc))
498: * or
499: * (MaxVol*WARP*0.5) / (1.0 + 0.5/( 1/Ra + 1/Rb + 1/Rc))
500: *
501: * Note that the 1.0 represents the normalized 1k resistor.
502: * 0.5 represents 50% duty cycle for the parallel resistors
503: * being summed (this effectively doubles the pull-up resistance).
1.1.1.17 root 504: */
505:
1.1.1.18 root 506: static void YM2149_BuildModelVolumeTable(ymu16 volumetable[32][32][32])
1.1.1.17 root 507: {
1.1.1.18 root 508: #define MaxVol 65535.0 /* Normal Mode Maximum value in table */
509: #define FOURTH2 1.19 /* Fourth root of two from YM2149 */
510: #define WARP 1.666666666666666667 /* measured as 1.65932 from 46602 */
1.1.1.17 root 511:
512: double conductance;
513: double conductance_[32];
514: int i, j, k;
515:
516: /**
517: * YM2149 and R8=1k follows (2^-1/4)^(n-31) better when 2 voices are
518: * summed (A+B or B+C or C+A) rather than individually (A or B or C):
1.1.1.18 root 519: * conductance = 2.0/3.0/(1.0-1.0/WARP)-2.0/3.0;
1.1.1.17 root 520: * When taken into consideration with three voices.
521: *
522: * Note that the YM2149 does not use laser trimmed resistances, thus
523: * has offsets that are added and/or multiplied with (2^-1/4)^(n-31).
524: */
1.1.1.18 root 525: conductance = 2.0/3.0/(1.0-1.0/WARP)-2.0/3.0; /* conductance = 1.0 */
1.1.1.17 root 526:
527: /**
528: * Because the YM current output (voltage source with series resistances)
529: * is connected to a grounded resistor to develop the output voltage
530: * (instead of a current to voltage converter), the output transfer
531: * function is not linear. Thus:
532: * 2.0*conductance_[n] = 1.0/(1.0-1.0/FOURTH2/(1.0/conductance + 1.0))-1.0;
533: */
534: for (i = 31; i >= 1; i--)
535: {
536: conductance_[i] = conductance/2.0;
537: conductance = 1.0/(1.0-1.0/FOURTH2/(1.0/conductance + 1.0))-1.0;
538: }
1.1.1.18 root 539: conductance_[0] = 1.0e-8; /* Avoid divide by zero */
1.1.1.17 root 540:
1.1.1.18 root 541: /**
542: * YM2149 AC + DC components model:
543: * (Note that Maxvol is 65119 in Simoes' table, 65535 in Gerard's)
544: *
545: * Sum the conductances as a function of a voltage divider:
546: * Vout=Vin*Rout/(Rout+Rin)
547: */
1.1.1.17 root 548: for (i = 0; i < 32; i++)
549: for (j = 0; j < 32; j++)
550: for (k = 0; k < 32; k++)
551: {
1.1.1.18 root 552: volumetable[i][j][k] = (ymu16)(0.5+(MaxVol*WARP)/(1.0 +
1.1.1.17 root 553: 1.0/(conductance_[i]+conductance_[j]+conductance_[k])));
554: }
1.1.1.18 root 555:
556: /**
557: * YM2149 DC component model:
558: * R8=1k (pulldown) + YM//1K (pullup) with YM 50% duty PWM
559: * (Note that MaxVol is 46602 in Paulo Simoes Quartet mode table)
560: *
561: * for (i = 0; i < 32; i++)
562: * for (j = 0; j < 32; j++)
563: * for (k = 0; k < 32; k++)
564: * {
565: * volumetable[i][j][k] = (ymu16)(0.5+(MaxVol*WARP)/(1.0 +
566: * 2.0/(conductance_[i]+conductance_[j]+conductance_[k])));
567: * }
568: */
1.1.1.17 root 569: }
570:
571:
572:
573:
574: /*-----------------------------------------------------------------------*/
575: /**
1.1.1.12 root 576: * Normalise and optionally center the volume table used to
577: * convert the 3 volumes to a final signed 16 bit sample.
578: * This allows to adapt the amplitude/volume of the samples and
579: * to convert unsigned values to signed values.
580: * - in_5bit contains 32*32*32 unsigned values in the range
581: * [0,65535].
582: * - out_5bit will contain signed values
583: * Possible values are :
584: * Level=65535 and DoCenter=TRUE -> [-32768,32767]
1.1.1.14 root 585: * Level=32767 and DoCenter=false -> [0,32767]
1.1.1.19 root 586: * Level=16383 and DoCenter=false -> [0,16383] (to avoid overflow with DMA sound on STe)
1.1.1.12 root 587: */
588:
589: static void YM2149_Normalise_5bit_Table(ymu16 *in_5bit , yms16 *out_5bit, unsigned int Level, bool DoCenter)
590: {
591: if ( Level )
1.1.1.11 root 592: {
1.1.1.14 root 593: int h;
1.1.1.12 root 594: int Max = in_5bit[0x7fff];
1.1.1.19 root 595: int Center = (Level+1)>>1;
1.1.1.14 root 596: //fprintf ( stderr , "level %d max %d center %d\n" , Level, Max, Center );
597:
1.1.1.12 root 598: /* Change the amplitude of the signal to 'level' : [0,max] -> [0,level] */
599: /* Then optionally center the signal around Level/2 */
600: /* This means we go from sthg like [0,65535] to [-32768, 32767] if Level=65535 and DoCenter=TRUE */
601: for (h=0; h<32*32*32; h++)
602: {
603: int tmp = in_5bit[h], res;
604: res = tmp * Level / Max;
1.1.1.14 root 605:
1.1.1.12 root 606: if ( DoCenter )
607: res -= Center;
608:
609: out_5bit[h] = res;
1.1.1.14 root 610: //fprintf ( stderr , "h %d in %d out %d\n" , h , tmp , res );
1.1.1.12 root 611: }
1.1.1.11 root 612: }
1.1 root 613: }
614:
1.1.1.5 root 615:
1.1.1.12 root 616:
617:
1.1.1.2 root 618: /*-----------------------------------------------------------------------*/
1.1.1.11 root 619: /**
1.1.1.12 root 620: * Precompute all 16 possible envelopes.
621: * Each envelope is made of 3 blocks of 32 volumes.
1.1.1.11 root 622: */
1.1.1.12 root 623:
624: static void YM2149_EnvBuild ( void )
1.1 root 625: {
1.1.1.12 root 626: int env;
627: int block;
628: int vol=0 , inc=0;
629: int i;
1.1 root 630:
1.1.1.12 root 631:
632: for ( env=0 ; env<16 ; env++ ) /* 16 possible envelopes */
633: for ( block=0 ; block<3 ; block++ ) /* 3 blocks to define an envelope */
634: {
1.1.1.14 root 635: switch ( YmEnvDef[ env ][ block ] )
636: {
1.1.1.12 root 637: case ENV_GODOWN : vol=31 ; inc=-1 ; break;
638: case ENV_GOUP : vol=0 ; inc=1 ; break;
639: case ENV_DOWN : vol=0 ; inc=0 ; break;
640: case ENV_UP : vol=31 ; inc=0 ; break;
1.1.1.14 root 641: }
642:
1.1.1.12 root 643: for ( i=0 ; i<32 ; i++ ) /* 32 volumes per block */
644: {
645: YmEnvWaves[ env ][ block*32 + i ] = YM_MERGE_VOICE ( vol , vol , vol );
646: vol += inc;
647: }
648: }
649: }
650:
651:
652:
653: /*-----------------------------------------------------------------------*/
654: /**
1.1.1.16 root 655: * Depending on the YM mixing method, build the table used to convert
656: * the 3 YM volumes into a single sample.
1.1.1.12 root 657: */
658:
1.1.1.16 root 659: static void Ym2149_BuildVolumeTable(void)
1.1.1.12 root 660: {
661: /* Depending on the volume mixing method, we use a table based on real measures */
662: /* or a table based on a linear volume mixing. */
1.1.1.17 root 663: if ( YmVolumeMixing == YM_MODEL_MIXING )
664: YM2149_BuildModelVolumeTable(ymout5_u16); /* create 32*32*32 circuit analysed model of the volume table */
665: else if ( YmVolumeMixing == YM_TABLE_MIXING )
1.1.1.16 root 666: interpolate_volumetable(ymout5_u16); /* expand the 16*16*16 values in volumetable_original to 32*32*32 */
1.1.1.12 root 667: else
668: YM2149_BuildLinearVolumeTable(ymout5_u16); /* combine the 32 possible volumes */
669:
670: /* Normalise/center the values (convert from u16 to s16) */
1.1.1.19 root 671: /* On STE/TT, we use YM_OUTPUT_LEVEL>>1 to avoid overflow with DMA sound */
1.1.1.21! root 672: if (Config_IsMachineSTE() || Config_IsMachineTT())
1.1.1.19 root 673: YM2149_Normalise_5bit_Table ( ymout5_u16[0][0] , ymout5 , (YM_OUTPUT_LEVEL>>1) , YM_OUTPUT_CENTERED );
674: else
675: YM2149_Normalise_5bit_Table ( ymout5_u16[0][0] , ymout5 , YM_OUTPUT_LEVEL , YM_OUTPUT_CENTERED );
1.1.1.16 root 676: }
677:
678:
679:
680: /*-----------------------------------------------------------------------*/
681: /**
682: * Init some internal tables for faster results (env, volume)
683: * and reset the internal states.
684: */
685:
686: static void Ym2149_Init(void)
687: {
688: /* Build the 16 envelope shapes */
689: YM2149_EnvBuild();
690:
691: /* Build the volume conversion table */
692: Ym2149_BuildVolumeTable();
1.1.1.12 root 693:
694: /* Reset YM2149 internal states */
695: Ym2149_Reset();
696: }
697:
698:
699:
700: /*-----------------------------------------------------------------------*/
701: /**
1.1.1.16 root 702: * Reset all ym registers as well as the internal variables
1.1.1.12 root 703: */
704:
705: static void Ym2149_Reset(void)
706: {
707: int i;
1.1.1.14 root 708:
1.1.1.12 root 709: for ( i=0 ; i<14 ; i++ )
710: Sound_WriteReg ( i , 0 );
711:
712: Sound_WriteReg ( 7 , 0xff );
713:
1.1.1.16 root 714: posA = 0;
715: posB = 0;
716: posC = 0;
717:
1.1.1.12 root 718: currentNoise = 0xffff;
1.1.1.14 root 719:
1.1.1.12 root 720: RndRack = 1;
1.1.1.14 root 721:
1.1.1.12 root 722: envShape = 0;
723: envPos = 0;
724: }
725:
726:
727:
728: /*-----------------------------------------------------------------------*/
729: /**
730: * Returns a pseudo random value, used to generate white noise.
1.1.1.19 root 731: * As measured by David Savinkoff, the YM2149 uses a 17 stage LSFR with
732: * 2 taps (17,14)
1.1.1.12 root 733: */
734:
735: static ymu32 YM2149_RndCompute(void)
736: {
1.1.1.19 root 737: /* 17 stage, 2 taps (17, 14) LFSR */
738: if (RndRack & 1)
739: {
740: RndRack = RndRack>>1 ^ 0x12000; /* bits 17 and 14 are ones */
741: return 0xffff;
742: }
743: else
744: { RndRack >>= 1;
745: return 0;
746: }
1.1.1.12 root 747: }
748:
749:
750:
751: /*-----------------------------------------------------------------------*/
752: /**
1.1.1.16 root 753: * Compute tone's step based on the input period.
754: * Although for tone we should have the same result when per==0 and per==1,
755: * this gives some very sharp and unpleasant sounds in the emulation.
756: * To get a better sound, we consider all per<=5 to give step=0, which will
757: * produce a constant output at value '1'. This should be handled with some
758: * proper filters to remove high frequencies as on a real ST (where per<=9
759: * gives nearly no audible sound).
760: * A common replay freq of 44.1 kHz will also not be high enough to correctly
761: * render possible tone's freq of 125 or 62.5 kHz (when per==1 or per==2)
1.1.1.12 root 762: */
763:
1.1.1.16 root 764: static ymu32 Ym2149_ToneStepCompute(ymu8 rHigh , ymu8 rLow)
765: {
766: int per;
767: yms64 step;
768:
769: per = rHigh&15;
770: per = (per<<8)+rLow;
771:
772: #if 0 /* need some high freq filters for this to work correctly */
773: if ( per == 0 )
774: per = 1; /* result for Per=0 is the same as for Per=1 */
775: #else
1.1.1.18 root 776: if (per <= (int)(YM_ATARI_CLOCK/(YM_REPLAY_FREQ*7)) )
777: return 0; /* discard frequencies higher than 80% of nyquist rate. */
1.1.1.16 root 778: #endif
779:
780: step = YM_ATARI_CLOCK;
781: step <<= 24;
782:
783: step /= (per * 8 * YM_REPLAY_FREQ); /* 0x5ab9 < step < 0x5ab3f46 at 44.1 kHz */
784:
785: return step;
786: }
1.1.1.21! root 787:
! 788:
1.1.1.12 root 789:
1.1.1.16 root 790: /*-----------------------------------------------------------------------*/
791: /**
792: * Compute noise's step based on the input period.
793: * On a real STF, we get the same result when per==0 and per==1.
794: * A common replay freq of 44.1 kHz will not be high enough to correctly
795: * render possible noise's freq of 125 or 62.5 kHz (when per==1 or per==2).
796: * With a random wave such as noise, this means that with a replay freq
797: * of 44.1 kHz, per==1 and per==2 (as well as per==3) will sound the same :
798: * per==1 step=0x2d59fa3 freq=125 kHz
799: * per==2 step=0x16acfd1 freq=62.5 kHz
800: * per==3 step=0x0f1dfe1 freq=41.7 kHz
801: */
1.1.1.12 root 802:
1.1.1.16 root 803: static ymu32 Ym2149_NoiseStepCompute(ymu8 rNoise)
804: {
805: int per;
806: yms64 step;
807:
808: per = (rNoise&0x1f);
1.1 root 809:
1.1.1.16 root 810: if ( per == 0 )
811: per = 1; /* result for Per=0 is the same as for Per=1 */
812:
813: step = YM_ATARI_CLOCK;
814: step <<= 24;
815:
816: step /= (per * 16 * YM_REPLAY_FREQ); /* 0x17683f < step < 0x2d59fa3 at 44.1 kHz */
817:
818: return step;
819: }
1.1.1.21! root 820:
! 821:
1.1.1.5 root 822:
1.1.1.2 root 823: /*-----------------------------------------------------------------------*/
1.1.1.11 root 824: /**
1.1.1.12 root 825: * Compute envelope's step. The envelope is made of different patterns
826: * of 32 volumes. In each pattern, the volume is changed at frequency
827: * Fe = MasterClock / ( 8 * EnvPer ).
828: * In our case, we use a lower replay freq ; between 2 consecutive calls
829: * to envelope's generation, the internal counter will advance 'step'
830: * units, where step = MasterClock / ( 8 * EnvPer * YM_REPLAY_FREQ )
831: * As 'step' requires floating point to be stored, we use left shifting
832: * to multiply 'step' by a fixed amount. All operations are made with
833: * shifted values ; to get the final value, we must right shift the
834: * result. We use '<<24', which gives 8 bits for the integer part, and
835: * the equivalent of 24 bits for the fractional part.
836: * Since we're using large numbers, we temporarily use 64 bits integer
837: * to avoid overflow and keep largest precision possible.
1.1.1.16 root 838: * On a real STF, we get the same result when per==0 and per==1.
1.1.1.11 root 839: */
1.1.1.12 root 840:
841: static ymu32 Ym2149_EnvStepCompute(ymu8 rHigh , ymu8 rLow)
1.1 root 842: {
1.1.1.12 root 843: yms64 per;
1.1.1.14 root 844: yms64 step;
1.1 root 845:
1.1.1.12 root 846: per = rHigh;
847: per = (per<<8)+rLow;
848:
1.1.1.14 root 849: step = YM_ATARI_CLOCK;
1.1.1.12 root 850: step <<= 24;
1.1.1.16 root 851:
852: if ( per == 0 )
853: per = 1; /* result for Per=0 is the same as for Per=1 */
854:
855: step /= (8 * per * YM_REPLAY_FREQ); /* 0x5ab < step < 0x5ab3f46 at 44.1 kHz */
1.1.1.12 root 856:
857: return step;
858: }
859:
860:
861:
862: /*-----------------------------------------------------------------------*/
863: /**
864: * Main function : compute the value of the next sample.
865: * Mixes all 3 voices with tone+noise+env and apply low pass
866: * filter if needed.
1.1.1.16 root 867: * All operations are done with integer math, using <<24 to simulate
868: * floating point precision : upper 8 bits are the integer part, lower 24
869: * are the fractional part.
870: * Tone is a square wave with 2 states 0 or 1. If integer part of posX is
871: * even (bit24=0) we consider output is 0, else (bit24=1) we consider
872: * output is 1. This gives the value of bt for one voice after extending it
873: * to all 0 bits or all 1 bits using a '-'
1.1.1.12 root 874: */
875:
1.1.1.16 root 876: static ymsample YM2149_NextSample(void)
877: {
878: ymsample sample;
879: ymu32 bt;
880: ymu32 bn;
881: ymu16 Env3Voices; /* 0x00CCBBAA */
882: ymu16 Tone3Voices; /* 0x00CCBBAA */
1.1 root 883:
884:
1.1.1.16 root 885: /* Noise value : 0 or 0xffff */
886: if ( noisePos&0xff000000 ) /* integer part > 0 */
887: {
1.1.1.19 root 888: currentNoise = YM2149_RndCompute();
1.1.1.16 root 889: noisePos &= 0xffffff; /* keep fractional part of noisePos */
890: }
891: bn = currentNoise; /* 0 or 0xffff */
892:
893: /* Get the 5 bits volume corresponding to the current envelope's position */
894: Env3Voices = YmEnvWaves[ envShape ][ envPos>>24 ]; /* integer part of envPos is in bits 24-31 */
895: Env3Voices &= EnvMask3Voices; /* only keep volumes for voices using envelope */
896:
897: //fprintf ( stderr , "env %x %x %x\n" , Env3Voices , envStep , envPos );
898:
899: /* Tone3Voices will contain the output state of each voice : 0 or 0x1f */
900: bt = -( (posA>>24) & 1); /* 0 if bit24=0 or 0xffffffff if bit24=1 */
901: bt = (bt | mixerTA) & (bn | mixerNA); /* 0 or 0xffff */
902: Tone3Voices = bt & YM_MASK_1VOICE; /* 0 or 0x1f */
903: bt = -( (posB>>24) & 1);
904: bt = (bt | mixerTB) & (bn | mixerNB);
905: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 5;
906: bt = -( (posC>>24) & 1);
907: bt = (bt | mixerTC) & (bn | mixerNC);
908: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 10;
909:
910: /* Combine fixed volumes and envelope volumes and keep the resulting */
911: /* volumes depending on the output state of each voice (0 or 0x1f) */
912: Tone3Voices &= ( Env3Voices | Vol3Voices );
913:
914: /* D/A conversion of the 3 volumes into a sample using a precomputed conversion table */
1.1.1.18 root 915:
916: if (stepA == 0 && (Tone3Voices & YM_MASK_A) > 1)
917: Tone3Voices -= 1; /* Voice A AC component removed; Transient DC component remains */
918:
919: if (stepB == 0 && (Tone3Voices & YM_MASK_B) > 1<<5)
920: Tone3Voices -= 1<<5; /* Voice B AC component removed; Transient DC component remains */
921:
922: if (stepC == 0 && (Tone3Voices & YM_MASK_C) > 1<<10)
923: Tone3Voices -= 1<<10; /* Voice C AC component removed; Transient DC component remains */
924:
1.1.1.16 root 925: sample = ymout5[ Tone3Voices ]; /* 16 bits signed value */
926:
927:
928: /* Increment positions */
929: posA += stepA;
930: posB += stepB;
931: posC += stepC;
932: noisePos += noiseStep;
933:
934: envPos += envStep;
935: if ( envPos >= (3*32) << 24 ) /* blocks 0, 1 and 2 were used (envPos 0 to 95) */
936: envPos -= (2*32) << 24; /* replay/loop blocks 1 and 2 (envPos 32 to 95) */
937:
938: /* Apply low pass filter ? */
939: if ( UseLowPassFilter )
1.1.1.18 root 940: return LowPassFilter(sample);
941: else
942: return PWMaliasFilter(sample);
1.1.1.16 root 943: }
1.1.1.21! root 944:
1.1.1.16 root 945:
1.1.1.12 root 946:
1.1.1.2 root 947: /*-----------------------------------------------------------------------*/
1.1.1.11 root 948: /**
1.1.1.12 root 949: * Update internal variables (steps, volume masks, ...) each
950: * time an YM register is changed.
1.1.1.11 root 951: */
1.1.1.16 root 952: #define BIT_SHIFT 24
1.1.1.12 root 953: void Sound_WriteReg( int reg , Uint8 data )
1.1.1.7 root 954: {
1.1.1.12 root 955: switch (reg)
956: {
957: case 0:
958: SoundRegs[0] = data;
959: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 960: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 961: break;
962:
963: case 1:
964: SoundRegs[1] = data & 0x0f;
965: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 966: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 967: break;
968:
969: case 2:
970: SoundRegs[2] = data;
971: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 972: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 973: break;
1.1.1.7 root 974:
1.1.1.12 root 975: case 3:
976: SoundRegs[3] = data & 0x0f;
977: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 978: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 979: break;
980:
981: case 4:
982: SoundRegs[4] = data;
983: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 984: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 985: break;
986:
987: case 5:
988: SoundRegs[5] = data & 0x0f;
989: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 990: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 991: break;
992:
993: case 6:
994: SoundRegs[6] = data & 0x1f;
995: noiseStep = Ym2149_NoiseStepCompute ( SoundRegs[6] );
996: if (!noiseStep)
997: {
998: noisePos = 0;
999: currentNoise = 0xffff;
1000: }
1001: break;
1002:
1003: case 7:
1004: SoundRegs[7] = data & 0x3f; /* ignore bits 6 and 7 */
1005: mixerTA = (data&(1<<0)) ? 0xffff : 0;
1006: mixerTB = (data&(1<<1)) ? 0xffff : 0;
1007: mixerTC = (data&(1<<2)) ? 0xffff : 0;
1008: mixerNA = (data&(1<<3)) ? 0xffff : 0;
1009: mixerNB = (data&(1<<4)) ? 0xffff : 0;
1010: mixerNC = (data&(1<<5)) ? 0xffff : 0;
1011: break;
1012:
1013: case 8:
1014: SoundRegs[8] = data & 0x1f;
1015: if ( data & 0x10 )
1016: {
1017: EnvMask3Voices |= YM_MASK_A; /* env ON */
1018: Vol3Voices &= ~YM_MASK_A; /* fixed vol OFF */
1019: }
1020: else
1021: {
1022: EnvMask3Voices &= ~YM_MASK_A; /* env OFF */
1023: Vol3Voices &= ~YM_MASK_A; /* clear previous vol */
1024: Vol3Voices |= YmVolume4to5[ SoundRegs[8] ]; /* fixed vol ON */
1025: }
1026: break;
1.1.1.14 root 1027:
1.1.1.12 root 1028: case 9:
1029: SoundRegs[9] = data & 0x1f;
1030: if ( data & 0x10 )
1031: {
1032: EnvMask3Voices |= YM_MASK_B; /* env ON */
1033: Vol3Voices &= ~YM_MASK_B; /* fixed vol OFF */
1034: }
1035: else
1036: {
1037: EnvMask3Voices &= ~YM_MASK_B; /* env OFF */
1038: Vol3Voices &= ~YM_MASK_B; /* clear previous vol */
1039: Vol3Voices |= ( YmVolume4to5[ SoundRegs[9] ] ) << 5; /* fixed vol ON */
1040: }
1041: break;
1.1.1.14 root 1042:
1.1.1.12 root 1043: case 10:
1044: SoundRegs[10] = data & 0x1f;
1045: if ( data & 0x10 )
1046: {
1047: EnvMask3Voices |= YM_MASK_C; /* env ON */
1048: Vol3Voices &= ~YM_MASK_C; /* fixed vol OFF */
1049: }
1050: else
1051: {
1052: EnvMask3Voices &= ~YM_MASK_C; /* env OFF */
1053: Vol3Voices &= ~YM_MASK_C; /* clear previous vol */
1054: Vol3Voices |= ( YmVolume4to5[ SoundRegs[10] ] ) << 10; /* fixed vol ON */
1055: }
1056: break;
1057:
1058: case 11:
1059: SoundRegs[11] = data;
1060: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1061: break;
1062:
1063: case 12:
1064: SoundRegs[12] = data;
1065: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1066: break;
1067:
1068: case 13:
1069: SoundRegs[13] = data & 0xf;
1070: envPos = 0; /* when writing to EnvShape, we must reset the EnvPos */
1071: envShape = SoundRegs[13];
1.1.1.14 root 1072: bEnvelopeFreqFlag = true; /* used for YmFormat saving */
1.1.1.12 root 1073: break;
1074:
1075: }
1076: }
1077:
1078:
1079:
1080: /*-----------------------------------------------------------------------*/
1081: /**
1082: * Init random generator, sound tables and envelopes
1083: * (called only once when Hatari starts)
1084: */
1085: void Sound_Init(void)
1086: {
1087: /* Build volume/env tables, ... */
1088: Ym2149_Init();
1.1.1.14 root 1089:
1.1.1.11 root 1090: Sound_Reset();
1.1.1.7 root 1091: }
1092:
1093:
1094: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1095: /**
1.1.1.12 root 1096: * Reset the sound emulation (called from Reset_ST() in reset.c)
1.1.1.11 root 1097: */
1.1.1.7 root 1098: void Sound_Reset(void)
1099: {
1.1.1.11 root 1100: /* Lock audio system before accessing variables which are used by the
1101: * callback function, too! */
1102: Audio_Lock();
1.1.1.9 root 1103:
1.1.1.11 root 1104: /* Clear sound mixing buffer: */
1.1.1.12 root 1105: memset(MixBuffer, 0, sizeof(MixBuffer));
1.1.1.7 root 1106:
1.1.1.11 root 1107: /* Clear cycle counts, buffer index and register '13' flags */
1108: Cycles_SetCounter(CYCLES_COUNTER_SOUND, 0);
1.1.1.14 root 1109: bEnvelopeFreqFlag = false;
1110:
1.1.1.11 root 1111: CompleteSndBufIdx = 0;
1112: /* We do not start with 0 here to fake some initial samples: */
1113: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1114: ActiveSndBufIdx = nGeneratedSamples % MIXBUFFER_SIZE;
1.1.1.16 root 1115: SamplesPerFrame = SAMPLES_PER_FRAME;
1116: CurrentSamplesNb = 0;
1117: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1118: //fprintf ( stderr , "Sound_Reset SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1119: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.7 root 1120:
1.1.1.12 root 1121: Ym2149_Reset();
1.1.1.9 root 1122:
1.1.1.11 root 1123: Audio_Unlock();
1.1.1.7 root 1124: }
1125:
1126:
1127: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1128: /**
1129: * Reset the sound buffer index variables.
1.1.1.16 root 1130: * Very important : this function should only be called by setting
1131: * Sound_BufferIndexNeedReset=true ; sound buffer index should be reset
1132: * only after the sound for the whole VBL was updated (CurrentSamplesNb returns to 0)
1133: * else it will alter the value of DMA Frame Count ($ff8909/0b/0d) and
1134: * could cause crashes in some programs.
1.1.1.11 root 1135: */
1.1.1.9 root 1136: void Sound_ResetBufferIndex(void)
1.1.1.7 root 1137: {
1.1.1.11 root 1138: Audio_Lock();
1139: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1140: ActiveSndBufIdx = (CompleteSndBufIdx + nGeneratedSamples) % MIXBUFFER_SIZE;
1.1.1.16 root 1141: SamplesPerFrame = SAMPLES_PER_FRAME;
1142: CurrentSamplesNb = 0;
1143: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1144: //fprintf ( stderr , "Sound_ResetBufferIndex SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1145: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.11 root 1146: Audio_Unlock();
1.1.1.7 root 1147: }
1148:
1149:
1150: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1151: /**
1152: * Save/Restore snapshot of local variables('MemorySnapShot_Store' handles type)
1153: */
1.1.1.12 root 1154: void Sound_MemorySnapShot_Capture(bool bSave)
1.1.1.7 root 1155: {
1.1.1.11 root 1156: /* Save/Restore details */
1.1.1.12 root 1157: MemorySnapShot_Store(&stepA, sizeof(stepA));
1158: MemorySnapShot_Store(&stepB, sizeof(stepB));
1159: MemorySnapShot_Store(&stepC, sizeof(stepC));
1160: MemorySnapShot_Store(&posA, sizeof(posA));
1161: MemorySnapShot_Store(&posB, sizeof(posB));
1162: MemorySnapShot_Store(&posC, sizeof(posC));
1163:
1164: MemorySnapShot_Store(&mixerTA, sizeof(mixerTA));
1165: MemorySnapShot_Store(&mixerTB, sizeof(mixerTB));
1166: MemorySnapShot_Store(&mixerTC, sizeof(mixerTC));
1167: MemorySnapShot_Store(&mixerNA, sizeof(mixerNA));
1168: MemorySnapShot_Store(&mixerNB, sizeof(mixerNB));
1169: MemorySnapShot_Store(&mixerNC, sizeof(mixerNC));
1170:
1171: MemorySnapShot_Store(&noiseStep, sizeof(noiseStep));
1172: MemorySnapShot_Store(&noisePos, sizeof(noisePos));
1173: MemorySnapShot_Store(¤tNoise, sizeof(currentNoise));
1174: MemorySnapShot_Store(&RndRack, sizeof(RndRack));
1175:
1176: MemorySnapShot_Store(&envStep, sizeof(envStep));
1177: MemorySnapShot_Store(&envPos, sizeof(envPos));
1178: MemorySnapShot_Store(&envShape, sizeof(envShape));
1.1.1.14 root 1179:
1.1.1.12 root 1180: MemorySnapShot_Store(&EnvMask3Voices, sizeof(EnvMask3Voices));
1181: MemorySnapShot_Store(&Vol3Voices, sizeof(Vol3Voices));
1.1.1.14 root 1182:
1.1.1.12 root 1183: MemorySnapShot_Store(SoundRegs, sizeof(SoundRegs));
1184:
1.1.1.14 root 1185: // MemorySnapShot_Store(&YmVolumeMixing, sizeof(YmVolumeMixing));
1186: // MemorySnapShot_Store(&UseLowPassFilter, sizeof(UseLowPassFilter));
1.1.1.7 root 1187: }
1188:
1189:
1190: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1191: /**
1192: * Find how many samples to generate and store in 'nSamplesToGenerate'
1193: * Also update sound cycles counter to store how many we actually did
1194: * so generates set amount each frame.
1.1.1.16 root 1195: * If FillFrame is true, this means we reach the end of the VBL and me must
1196: * add as many samples as necessary to get a total of SamplesPerFrame
1197: * for this VBL.
1.1.1.11 root 1198: */
1.1.1.16 root 1199: static int Sound_SetSamplesPassed(bool FillFrame)
1.1 root 1200: {
1.1.1.11 root 1201: int nSoundCycles;
1.1.1.16 root 1202: int SamplesToGenerate; /* How many samples are needed for this time-frame */
1203:
1204: nSoundCycles = Cycles_GetCounter(CYCLES_COUNTER_VIDEO);
1.1.1.11 root 1205:
1.1.1.21! root 1206: #ifndef OLD_CPU_SHIFT
! 1207: nSoundCycles >>= nCpuFreqShift;
! 1208: #endif
! 1209:
1.1.1.16 root 1210: /* example : 160256 cycles per VBL, 44Khz = 882 samples per VBL at 50 Hz */
1211: /* 882/160256 samples per cpu clock cycle */
1.1.1.11 root 1212:
1.1.1.16 root 1213: /* Total number of samples that we should have at this point of the VBL */
1214: SamplesToGenerate = nSoundCycles * SamplesPerFrame
1215: / ClocksTimings_GetCyclesPerVBL ( ConfigureParams.System.nMachineType , nScreenRefreshRate );
1.1.1.11 root 1216:
1.1.1.16 root 1217: //if (SamplesToGenerate > SamplesPerFrame )
1218: //fprintf ( stderr , "over run %d %d\n" , SamplesPerFrame , SamplesToGenerate );
1.1.1.11 root 1219:
1.1.1.16 root 1220: if (SamplesToGenerate > SamplesPerFrame)
1221: SamplesToGenerate = SamplesPerFrame;
1222:
1223: SamplesToGenerate -= CurrentSamplesNb; /* don't count samples that were already generated up to now */
1224: if ( SamplesToGenerate < 0 )
1225: SamplesToGenerate = 0;
1226:
1227:
1228: /* If we're called from the VBL interrupt (FillFrame==true), we must ensure we have */
1229: /* an exact total of SamplesPerFrame samples during a full VBL (we take into account */
1230: /* the samples that were already generated during this VBL) */
1231: if ( FillFrame )
1232: {
1233: SamplesToGenerate = SamplesPerFrame - CurrentSamplesNb; /* how many samples are missing to reach SamplesPerFrame */
1234: if ( SamplesToGenerate < 0 )
1235: SamplesToGenerate = 0;
1236: }
1.1.1.11 root 1237:
1.1.1.16 root 1238: /* Check we don't fill the sound's ring buffer before it's played by Audio_Callback() */
1239: /* This should never happen, except if the system suffers major slowdown due to other */
1240: /* processes or if we run in fast forward mode. */
1241: /* In the case of slowdown, we set Sound_BufferIndexNeedReset to "resync" the working */
1242: /* buffer's index ActiveSndBufIdx with the system buffer's index CompleteSndBufIdx. */
1243: /* In the case of fast forward, we do nothing here, Sound_BufferIndexNeedReset will be */
1244: /* set when the user exits fast forward mode. */
1245: if ( ( SamplesToGenerate > MIXBUFFER_SIZE - nGeneratedSamples ) && ( ConfigureParams.System.bFastForward == false )
1246: && ( ConfigureParams.Sound.bEnableSound == true ) )
1.1.1.11 root 1247: {
1.1.1.21! root 1248: static int logcnt = 0;
! 1249: if (logcnt++ < 50)
! 1250: {
! 1251: Log_Printf(LOG_WARN, "Your system is too slow, "
! 1252: "some sound samples were not correctly emulated\n");
! 1253: }
1.1.1.16 root 1254: Sound_BufferIndexNeedReset = true;
1.1.1.11 root 1255: }
1.1.1.16 root 1256:
1257: //fprintf ( stderr , "vbl %d hbl %d samp_gen %d / %d frac %lx\n" , nVBLs , nHBL , SamplesToGenerate , SamplesPerFrame , (long int)SamplesPerFrame_unrounded );
1258:
1259: return SamplesToGenerate;
1.1 root 1260: }
1261:
1.1.1.5 root 1262:
1.1.1.2 root 1263: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1264: /**
1265: * Generate samples for all channels during this time-frame
1266: */
1.1.1.16 root 1267: static void Sound_GenerateSamples(int SamplesToGenerate)
1.1 root 1268: {
1.1.1.12 root 1269: int i;
1270: int idx;
1.1.1.14 root 1271:
1.1.1.16 root 1272: if (SamplesToGenerate <= 0)
1.1.1.12 root 1273: return;
1.1.1.14 root 1274:
1.1.1.21! root 1275: if (Config_IsMachineFalcon())
1.1.1.11 root 1276: {
1.1.1.16 root 1277: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1278: {
1279: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1280: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1281: }
1282: /* If Falcon emulation, crossbar does the job */
1.1.1.16 root 1283: Crossbar_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.11 root 1284: }
1.1.1.21! root 1285: else if (!Config_IsMachineST())
1.1.1.15 root 1286: {
1.1.1.16 root 1287: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1288: {
1289: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.18 root 1290: MixBuffer[idx][0] = MixBuffer[idx][1] = YM2149_NextSample();
1.1.1.15 root 1291: }
1.1.1.17 root 1292: /* If Ste or TT emulation, DmaSnd does mixing and filtering */
1.1.1.16 root 1293: DmaSnd_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.15 root 1294: }
1.1.1.21! root 1295: else
1.1.1.15 root 1296: {
1.1.1.16 root 1297: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1298: {
1299: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1300: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1301: }
1302: }
1.1.1.12 root 1303:
1.1.1.16 root 1304: ActiveSndBufIdx = (ActiveSndBufIdx + SamplesToGenerate) % MIXBUFFER_SIZE;
1305: nGeneratedSamples += SamplesToGenerate;
1306: CurrentSamplesNb += SamplesToGenerate; /* number of samples generated for current VBL */
1.1 root 1307: }
1308:
1.1.1.5 root 1309:
1.1.1.2 root 1310: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1311: /**
1312: * This is called to built samples up until this clock cycle
1.1.1.16 root 1313: * Sound_Update can be called several times during a VBL ; we must ensure
1314: * that we generate exactly SamplesPerFrame samples between 2 calls
1315: * to Sound_Update_VBL.
1.1.1.11 root 1316: */
1.1.1.16 root 1317: void Sound_Update(bool FillFrame)
1.1 root 1318: {
1.1.1.11 root 1319: int OldSndBufIdx = ActiveSndBufIdx;
1.1.1.16 root 1320: int SamplesToGenerate;
1.1.1.5 root 1321:
1.1.1.11 root 1322: /* Make sure that we don't interfere with the audio callback function */
1323: Audio_Lock();
1.1.1.6 root 1324:
1.1.1.16 root 1325: /* Find how many samples to generate */
1326: SamplesToGenerate = Sound_SetSamplesPassed( FillFrame );
1327:
1.1.1.11 root 1328: /* And generate */
1.1.1.16 root 1329: Sound_GenerateSamples( SamplesToGenerate );
1.1 root 1330:
1.1.1.11 root 1331: /* Allow audio callback function to occur again */
1332: Audio_Unlock();
1.1.1.6 root 1333:
1.1.1.11 root 1334: /* Save to WAV file, if open */
1335: if (bRecordingWav)
1.1.1.16 root 1336: WAVFormat_Update(MixBuffer, OldSndBufIdx, SamplesToGenerate);
1.1 root 1337: }
1338:
1.1.1.5 root 1339:
1.1.1.2 root 1340: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1341: /**
1.1.1.16 root 1342: * On the end of each VBL, complete audio buffer to reach SamplesPerFrame samples.
1343: * As Sound_Update(false) could be called several times during the VBL, the audio
1344: * buffer might be already partially filled.
1345: * We must first complete the buffer using the same value of SamplesPerFrame
1346: * by calling Sound_Update(true) ; then we can compute a new value for
1347: * SamplesPerFrame that will be used for the next VBL to come.
1.1.1.11 root 1348: */
1.1.1.5 root 1349: void Sound_Update_VBL(void)
1.1 root 1350: {
1.1.1.16 root 1351: Sound_Update(true); /* generate as many samples as needed to fill this VBL */
1352: //fprintf ( stderr , "vbl done %d %d\n" , SamplesPerFrame , CurrentSamplesNb );
1.1.1.5 root 1353:
1.1.1.16 root 1354: CurrentSamplesNb = 0; /* VBL is complete, reset counter for next VBL */
1355:
1356: /*Compute a fractional equivalent of SamplesPerFrame for the next VBL, to avoid rounding propagation */
1357: SamplesPerFrame_unrounded += (yms64) ClocksTimings_GetSamplesPerVBL ( ConfigureParams.System.nMachineType ,
1358: nScreenRefreshRate , nAudioFrequency );
1359: SamplesPerFrame = SamplesPerFrame_unrounded >> 28; /* use integer part */
1360: SamplesPerFrame_unrounded &= 0x0fffffff; /* keep fractional part in the lower 28 bits */
1361:
1362: /* Reset sound buffer if needed (after pause, fast forward, slow system, ...) */
1363: if ( Sound_BufferIndexNeedReset )
1364: {
1365: Sound_ResetBufferIndex ();
1366: Sound_BufferIndexNeedReset = false;
1367: }
1368:
1369: /* Record AVI audio frame is necessary */
1.1.1.15 root 1370: if ( bRecordingAvi )
1371: {
1.1.1.16 root 1372: int Len;
1.1.1.15 root 1373:
1.1.1.16 root 1374: Len = ActiveSndBufIdx - ActiveSndBufIdxAvi; /* number of generated samples for this frame */
1375: if ( Len < 0 )
1376: Len += MIXBUFFER_SIZE; /* end of ring buffer was reached */
1.1.1.15 root 1377:
1.1.1.16 root 1378: Avi_RecordAudioStream ( MixBuffer , ActiveSndBufIdxAvi , Len );
1.1.1.15 root 1379: }
1380:
1.1.1.16 root 1381: ActiveSndBufIdxAvi = ActiveSndBufIdx; /* save new position for next AVI audio frame */
1382:
1.1.1.11 root 1383: /* Clear write to register '13', used for YM file saving */
1.1.1.14 root 1384: bEnvelopeFreqFlag = false;
1.1 root 1385: }
1386:
1387:
1.1.1.2 root 1388: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1389: /**
1390: * Start recording sound, as .YM or .WAV output
1391: */
1.1.1.12 root 1392: bool Sound_BeginRecording(char *pszCaptureFileName)
1.1 root 1393: {
1.1.1.12 root 1394: bool bRet;
1.1.1.7 root 1395:
1.1.1.11 root 1396: if (!pszCaptureFileName || strlen(pszCaptureFileName) <= 3)
1397: {
1398: Log_Printf(LOG_ERROR, "Illegal sound recording file name!\n");
1.1.1.14 root 1399: return false;
1.1.1.11 root 1400: }
1401:
1402: /* Did specify .YM or .WAV? If neither report error */
1403: if (File_DoesFileExtensionMatch(pszCaptureFileName,".ym"))
1404: bRet = YMFormat_BeginRecording(pszCaptureFileName);
1405: else if (File_DoesFileExtensionMatch(pszCaptureFileName,".wav"))
1406: bRet = WAVFormat_OpenFile(pszCaptureFileName);
1407: else
1408: {
1409: Log_AlertDlg(LOG_ERROR, "Unknown Sound Recording format.\n"
1410: "Please specify a .YM or .WAV output file.");
1.1.1.14 root 1411: bRet = false;
1.1.1.11 root 1412: }
1413:
1414: return bRet;
1.1 root 1415: }
1416:
1.1.1.5 root 1417:
1.1.1.2 root 1418: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1419: /**
1420: * End sound recording
1421: */
1.1.1.7 root 1422: void Sound_EndRecording(void)
1.1 root 1423: {
1.1.1.11 root 1424: /* Stop sound recording and close files */
1425: if (bRecordingYM)
1426: YMFormat_EndRecording();
1427: if (bRecordingWav)
1428: WAVFormat_CloseFile();
1.1 root 1429: }
1430:
1.1.1.6 root 1431:
1.1.1.2 root 1432: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1433: /**
1434: * Are we recording sound data?
1435: */
1.1.1.12 root 1436: bool Sound_AreWeRecording(void)
1.1 root 1437: {
1.1.1.11 root 1438: return (bRecordingYM || bRecordingWav);
1.1 root 1439: }
1.1.1.12 root 1440:
1.1.1.16 root 1441:
1442: /*-----------------------------------------------------------------------*/
1443: /**
1444: * Rebuild volume conversion table
1445: */
1446: void Sound_SetYmVolumeMixing(void)
1447: {
1448: /* Build the volume conversion table */
1449: Ym2149_BuildVolumeTable();
1450: }
1451:
This archive runs on limited infrastructure. Preserving old code on modern bandwidth. Automated agents are requested to crawl responsibly.