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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:
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 */
1.1.1.19! root 39: /* have sound.c independent of psg.c (to ease replacement of */
1.1.1.12 root 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: *
1.1.1.19! root 470: * The output of the YM2149 is modelled well as pullup resistors.
1.1.1.18 root 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]
1.1.1.19! root 584: * Level=16383 and DoCenter=false -> [0,16383] (to avoid overflow with DMA sound on STe)
1.1.1.12 root 585: */
586:
587: static void YM2149_Normalise_5bit_Table(ymu16 *in_5bit , yms16 *out_5bit, unsigned int Level, bool DoCenter)
588: {
589: if ( Level )
1.1.1.11 root 590: {
1.1.1.14 root 591: int h;
1.1.1.12 root 592: int Max = in_5bit[0x7fff];
1.1.1.19! root 593: int Center = (Level+1)>>1;
1.1.1.14 root 594: //fprintf ( stderr , "level %d max %d center %d\n" , Level, Max, Center );
595:
1.1.1.12 root 596: /* Change the amplitude of the signal to 'level' : [0,max] -> [0,level] */
597: /* Then optionally center the signal around Level/2 */
598: /* This means we go from sthg like [0,65535] to [-32768, 32767] if Level=65535 and DoCenter=TRUE */
599: for (h=0; h<32*32*32; h++)
600: {
601: int tmp = in_5bit[h], res;
602: res = tmp * Level / Max;
1.1.1.14 root 603:
1.1.1.12 root 604: if ( DoCenter )
605: res -= Center;
606:
607: out_5bit[h] = res;
1.1.1.14 root 608: //fprintf ( stderr , "h %d in %d out %d\n" , h , tmp , res );
1.1.1.12 root 609: }
1.1.1.11 root 610: }
1.1 root 611: }
612:
1.1.1.5 root 613:
1.1.1.12 root 614:
615:
1.1.1.2 root 616: /*-----------------------------------------------------------------------*/
1.1.1.11 root 617: /**
1.1.1.12 root 618: * Precompute all 16 possible envelopes.
619: * Each envelope is made of 3 blocks of 32 volumes.
1.1.1.11 root 620: */
1.1.1.12 root 621:
622: static void YM2149_EnvBuild ( void )
1.1 root 623: {
1.1.1.12 root 624: int env;
625: int block;
626: int vol=0 , inc=0;
627: int i;
1.1 root 628:
1.1.1.12 root 629:
630: for ( env=0 ; env<16 ; env++ ) /* 16 possible envelopes */
631: for ( block=0 ; block<3 ; block++ ) /* 3 blocks to define an envelope */
632: {
1.1.1.14 root 633: switch ( YmEnvDef[ env ][ block ] )
634: {
1.1.1.12 root 635: case ENV_GODOWN : vol=31 ; inc=-1 ; break;
636: case ENV_GOUP : vol=0 ; inc=1 ; break;
637: case ENV_DOWN : vol=0 ; inc=0 ; break;
638: case ENV_UP : vol=31 ; inc=0 ; break;
1.1.1.14 root 639: }
640:
1.1.1.12 root 641: for ( i=0 ; i<32 ; i++ ) /* 32 volumes per block */
642: {
643: YmEnvWaves[ env ][ block*32 + i ] = YM_MERGE_VOICE ( vol , vol , vol );
644: vol += inc;
645: }
646: }
647: }
648:
649:
650:
651: /*-----------------------------------------------------------------------*/
652: /**
1.1.1.16 root 653: * Depending on the YM mixing method, build the table used to convert
654: * the 3 YM volumes into a single sample.
1.1.1.12 root 655: */
656:
1.1.1.16 root 657: static void Ym2149_BuildVolumeTable(void)
1.1.1.12 root 658: {
659: /* Depending on the volume mixing method, we use a table based on real measures */
660: /* or a table based on a linear volume mixing. */
1.1.1.17 root 661: if ( YmVolumeMixing == YM_MODEL_MIXING )
662: YM2149_BuildModelVolumeTable(ymout5_u16); /* create 32*32*32 circuit analysed model of the volume table */
663: else if ( YmVolumeMixing == YM_TABLE_MIXING )
1.1.1.16 root 664: interpolate_volumetable(ymout5_u16); /* expand the 16*16*16 values in volumetable_original to 32*32*32 */
1.1.1.12 root 665: else
666: YM2149_BuildLinearVolumeTable(ymout5_u16); /* combine the 32 possible volumes */
667:
668: /* Normalise/center the values (convert from u16 to s16) */
1.1.1.19! root 669: /* On STE/TT, we use YM_OUTPUT_LEVEL>>1 to avoid overflow with DMA sound */
! 670: if ( (ConfigureParams.System.nMachineType == MACHINE_STE) || (ConfigureParams.System.nMachineType == MACHINE_MEGA_STE)
! 671: || (ConfigureParams.System.nMachineType == MACHINE_TT) )
! 672: YM2149_Normalise_5bit_Table ( ymout5_u16[0][0] , ymout5 , (YM_OUTPUT_LEVEL>>1) , YM_OUTPUT_CENTERED );
! 673: else
! 674: YM2149_Normalise_5bit_Table ( ymout5_u16[0][0] , ymout5 , YM_OUTPUT_LEVEL , YM_OUTPUT_CENTERED );
1.1.1.16 root 675: }
676:
677:
678:
679: /*-----------------------------------------------------------------------*/
680: /**
681: * Init some internal tables for faster results (env, volume)
682: * and reset the internal states.
683: */
684:
685: static void Ym2149_Init(void)
686: {
687: /* Build the 16 envelope shapes */
688: YM2149_EnvBuild();
689:
690: /* Build the volume conversion table */
691: Ym2149_BuildVolumeTable();
1.1.1.12 root 692:
693: /* Reset YM2149 internal states */
694: Ym2149_Reset();
695: }
696:
697:
698:
699: /*-----------------------------------------------------------------------*/
700: /**
1.1.1.16 root 701: * Reset all ym registers as well as the internal variables
1.1.1.12 root 702: */
703:
704: static void Ym2149_Reset(void)
705: {
706: int i;
1.1.1.14 root 707:
1.1.1.12 root 708: for ( i=0 ; i<14 ; i++ )
709: Sound_WriteReg ( i , 0 );
710:
711: Sound_WriteReg ( 7 , 0xff );
712:
1.1.1.16 root 713: posA = 0;
714: posB = 0;
715: posC = 0;
716:
1.1.1.12 root 717: currentNoise = 0xffff;
1.1.1.14 root 718:
1.1.1.12 root 719: RndRack = 1;
1.1.1.14 root 720:
1.1.1.12 root 721: envShape = 0;
722: envPos = 0;
723: }
724:
725:
726:
727: /*-----------------------------------------------------------------------*/
728: /**
729: * Returns a pseudo random value, used to generate white noise.
1.1.1.19! root 730: * As measured by David Savinkoff, the YM2149 uses a 17 stage LSFR with
! 731: * 2 taps (17,14)
1.1.1.12 root 732: */
733:
734: static ymu32 YM2149_RndCompute(void)
735: {
1.1.1.19! root 736: /* 17 stage, 2 taps (17, 14) LFSR */
! 737: if (RndRack & 1)
! 738: {
! 739: RndRack = RndRack>>1 ^ 0x12000; /* bits 17 and 14 are ones */
! 740: return 0xffff;
! 741: }
! 742: else
! 743: { RndRack >>= 1;
! 744: return 0;
! 745: }
1.1.1.12 root 746: }
747:
748:
749:
750: /*-----------------------------------------------------------------------*/
751: /**
1.1.1.16 root 752: * Compute tone's step based on the input period.
753: * Although for tone we should have the same result when per==0 and per==1,
754: * this gives some very sharp and unpleasant sounds in the emulation.
755: * To get a better sound, we consider all per<=5 to give step=0, which will
756: * produce a constant output at value '1'. This should be handled with some
757: * proper filters to remove high frequencies as on a real ST (where per<=9
758: * gives nearly no audible sound).
759: * A common replay freq of 44.1 kHz will also not be high enough to correctly
760: * render possible tone's freq of 125 or 62.5 kHz (when per==1 or per==2)
1.1.1.12 root 761: */
762:
1.1.1.16 root 763: #define NEWSTEP
764: #ifndef NEWSTEP
1.1.1.12 root 765: static ymu32 Ym2149_ToneStepCompute(ymu8 rHigh , ymu8 rLow)
766: {
767: int per;
1.1.1.14 root 768: yms64 step;
1.1.1.12 root 769:
770: per = rHigh&15;
771: per = (per<<8)+rLow;
1.1.1.18 root 772:
773: if (per <= (int)(YM_ATARI_CLOCK/(YM_REPLAY_FREQ*7)) )
1.1.1.12 root 774: return 0;
775:
1.1.1.14 root 776: step = YM_ATARI_CLOCK;
1.1.1.12 root 777: step <<= (15+16-3);
778: step /= (per * YM_REPLAY_FREQ);
779:
780: return step;
781: }
1.1.1.16 root 782: #else
783: static ymu32 Ym2149_ToneStepCompute(ymu8 rHigh , ymu8 rLow)
784: {
785: int per;
786: yms64 step;
787:
788: per = rHigh&15;
789: per = (per<<8)+rLow;
790:
791: #if 0 /* need some high freq filters for this to work correctly */
792: if ( per == 0 )
793: per = 1; /* result for Per=0 is the same as for Per=1 */
794: #else
1.1.1.18 root 795: if (per <= (int)(YM_ATARI_CLOCK/(YM_REPLAY_FREQ*7)) )
796: return 0; /* discard frequencies higher than 80% of nyquist rate. */
1.1.1.16 root 797: #endif
798:
799: step = YM_ATARI_CLOCK;
800: step <<= 24;
801:
802: step /= (per * 8 * YM_REPLAY_FREQ); /* 0x5ab9 < step < 0x5ab3f46 at 44.1 kHz */
803:
804: return step;
805: }
806: #endif
1.1.1.12 root 807:
1.1.1.16 root 808: /*-----------------------------------------------------------------------*/
809: /**
810: * Compute noise's step based on the input period.
811: * On a real STF, we get the same result when per==0 and per==1.
812: * A common replay freq of 44.1 kHz will not be high enough to correctly
813: * render possible noise's freq of 125 or 62.5 kHz (when per==1 or per==2).
814: * With a random wave such as noise, this means that with a replay freq
815: * of 44.1 kHz, per==1 and per==2 (as well as per==3) will sound the same :
816: * per==1 step=0x2d59fa3 freq=125 kHz
817: * per==2 step=0x16acfd1 freq=62.5 kHz
818: * per==3 step=0x0f1dfe1 freq=41.7 kHz
819: */
1.1.1.12 root 820:
1.1.1.16 root 821: #ifndef NEWSTEP
1.1.1.12 root 822: static ymu32 Ym2149_NoiseStepCompute(ymu8 rNoise)
823: {
824: int per;
1.1.1.14 root 825: yms64 step;
1.1.1.12 root 826:
827: per = (rNoise&0x1f);
828: if (per<3)
829: return 0;
830:
1.1.1.14 root 831: step = YM_ATARI_CLOCK;
1.1.1.12 root 832: step <<= (16-1-3);
833: step /= (per * YM_REPLAY_FREQ);
834:
835: return step;
1.1 root 836: }
1.1.1.16 root 837: #else
838: static ymu32 Ym2149_NoiseStepCompute(ymu8 rNoise)
839: {
840: int per;
841: yms64 step;
842:
843: per = (rNoise&0x1f);
1.1 root 844:
1.1.1.16 root 845: if ( per == 0 )
846: per = 1; /* result for Per=0 is the same as for Per=1 */
847:
848: step = YM_ATARI_CLOCK;
849: step <<= 24;
850:
851: step /= (per * 16 * YM_REPLAY_FREQ); /* 0x17683f < step < 0x2d59fa3 at 44.1 kHz */
852:
853: return step;
854: }
855: #endif
1.1.1.5 root 856:
1.1.1.2 root 857: /*-----------------------------------------------------------------------*/
1.1.1.11 root 858: /**
1.1.1.12 root 859: * Compute envelope's step. The envelope is made of different patterns
860: * of 32 volumes. In each pattern, the volume is changed at frequency
861: * Fe = MasterClock / ( 8 * EnvPer ).
862: * In our case, we use a lower replay freq ; between 2 consecutive calls
863: * to envelope's generation, the internal counter will advance 'step'
864: * units, where step = MasterClock / ( 8 * EnvPer * YM_REPLAY_FREQ )
865: * As 'step' requires floating point to be stored, we use left shifting
866: * to multiply 'step' by a fixed amount. All operations are made with
867: * shifted values ; to get the final value, we must right shift the
868: * result. We use '<<24', which gives 8 bits for the integer part, and
869: * the equivalent of 24 bits for the fractional part.
870: * Since we're using large numbers, we temporarily use 64 bits integer
871: * to avoid overflow and keep largest precision possible.
1.1.1.16 root 872: * On a real STF, we get the same result when per==0 and per==1.
1.1.1.11 root 873: */
1.1.1.12 root 874:
875: static ymu32 Ym2149_EnvStepCompute(ymu8 rHigh , ymu8 rLow)
1.1 root 876: {
1.1.1.12 root 877: yms64 per;
1.1.1.14 root 878: yms64 step;
1.1 root 879:
1.1.1.12 root 880: per = rHigh;
881: per = (per<<8)+rLow;
882:
1.1.1.14 root 883: step = YM_ATARI_CLOCK;
1.1.1.12 root 884: step <<= 24;
1.1.1.16 root 885:
886: if ( per == 0 )
887: per = 1; /* result for Per=0 is the same as for Per=1 */
888:
889: step /= (8 * per * YM_REPLAY_FREQ); /* 0x5ab < step < 0x5ab3f46 at 44.1 kHz */
1.1.1.12 root 890:
891: return step;
892: }
893:
894:
895:
896: /*-----------------------------------------------------------------------*/
897: /**
898: * Main function : compute the value of the next sample.
899: * Mixes all 3 voices with tone+noise+env and apply low pass
900: * filter if needed.
1.1.1.16 root 901: * All operations are done with integer math, using <<24 to simulate
902: * floating point precision : upper 8 bits are the integer part, lower 24
903: * are the fractional part.
904: * Tone is a square wave with 2 states 0 or 1. If integer part of posX is
905: * even (bit24=0) we consider output is 0, else (bit24=1) we consider
906: * output is 1. This gives the value of bt for one voice after extending it
907: * to all 0 bits or all 1 bits using a '-'
1.1.1.12 root 908: */
909:
1.1.1.16 root 910: #ifndef NEWSTEP
1.1.1.12 root 911: static ymsample YM2149_NextSample(void)
912: {
913: ymsample sample;
914: int bt;
915: ymu32 bn;
916: ymu16 Env3Voices;
917: ymu16 Tone3Voices;
918:
919:
920: /* Noise value : 0 or 0xffff */
921: if ( noisePos&0xffff0000 )
922: {
1.1.1.19! root 923: currentNoise = YM2149_RndCompute();
1.1.1.12 root 924: noisePos &= 0xffff;
925: }
926: bn = currentNoise; /* 0 or 0xffff */
927:
928: /* Get the 5 bits volume corresponding to the current envelope's position */
929: Env3Voices = YmEnvWaves[ envShape ][ envPos>>24 ]; /* integer part of envPos is in bits 24-31 */
930: Env3Voices &= EnvMask3Voices; /* only keep volumes for voices using envelope */
931:
932: //fprintf ( stderr , "env %x %x %x\n" , Env3Voices , envStep , envPos );
933:
934: /* Tone3Voices will contain the output state of each voice : 0 or 0x1f */
935: bt = ((((yms32)posA)>>31) | mixerTA) & (bn | mixerNA); /* 0 or 0xffff */
936: Tone3Voices = bt & YM_MASK_1VOICE; /* 0 or 0x1f */
937: bt = ((((yms32)posB)>>31) | mixerTB) & (bn | mixerNB);
938: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 5;
939: bt = ((((yms32)posC)>>31) | mixerTC) & (bn | mixerNC);
940: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 10;
941:
942: /* Combine fixed volumes and envelope volumes and keep the resulting */
943: /* volumes depending on the output state of each voice (0 or 0x1f) */
944: Tone3Voices &= ( Env3Voices | Vol3Voices );
945:
1.1.1.18 root 946: /* When a step period is 0, the represented frequency was filtered from the */
947: /* ouput of the YM2149. Thus, use the transient DC component of the sample. */
948: /* Note that the "-1" table offset is a "good fit" for the DC component. */
949:
950: if (stepA == 0 && (Tone3Voices & YM_MASK_A) > 1)
951: Tone3Voices -= 1; /* Voice A AC component removed; Transient DC component remains */
952:
953: if (stepB == 0 && (Tone3Voices & YM_MASK_B) > 1<<5)
954: Tone3Voices -= 1<<5; /* Voice B AC component removed; Transient DC component remains */
955:
956: if (stepC == 0 && (Tone3Voices & YM_MASK_C) > 1<<10)
957: Tone3Voices -= 1<<10; /* Voice C AC component removed; Transient DC component remains */
958:
1.1.1.12 root 959: /* D/A conversion of the 3 volumes into a sample using a precomputed conversion table */
1.1.1.18 root 960:
1.1.1.12 root 961: sample = ymout5[ Tone3Voices ]; /* 16 bits signed value */
962:
963:
964: /* Increment positions */
965: posA += stepA;
966: posB += stepB;
967: posC += stepC;
968: noisePos += noiseStep;
1.1.1.14 root 969:
1.1.1.12 root 970: envPos += envStep;
971: if ( envPos >= (3*32) << 24 ) /* blocks 0, 1 and 2 were used (envPos 0 to 95) */
972: envPos -= (2*32) << 24; /* replay/loop blocks 1 and 2 (envPos 32 to 95) */
973:
974: /* Apply low pass filter ? */
975: if ( UseLowPassFilter )
1.1.1.18 root 976: return LowPassFilter(sample);
977: else
978: return PWMaliasFilter(sample);
1.1 root 979: }
1.1.1.16 root 980: #else
981: static ymsample YM2149_NextSample(void)
982: {
983: ymsample sample;
984: ymu32 bt;
985: ymu32 bn;
986: ymu16 Env3Voices; /* 0x00CCBBAA */
987: ymu16 Tone3Voices; /* 0x00CCBBAA */
1.1 root 988:
989:
1.1.1.16 root 990: /* Noise value : 0 or 0xffff */
991: if ( noisePos&0xff000000 ) /* integer part > 0 */
992: {
1.1.1.19! root 993: currentNoise = YM2149_RndCompute();
1.1.1.16 root 994: noisePos &= 0xffffff; /* keep fractional part of noisePos */
995: }
996: bn = currentNoise; /* 0 or 0xffff */
997:
998: /* Get the 5 bits volume corresponding to the current envelope's position */
999: Env3Voices = YmEnvWaves[ envShape ][ envPos>>24 ]; /* integer part of envPos is in bits 24-31 */
1000: Env3Voices &= EnvMask3Voices; /* only keep volumes for voices using envelope */
1001:
1002: //fprintf ( stderr , "env %x %x %x\n" , Env3Voices , envStep , envPos );
1003:
1004: /* Tone3Voices will contain the output state of each voice : 0 or 0x1f */
1005: bt = -( (posA>>24) & 1); /* 0 if bit24=0 or 0xffffffff if bit24=1 */
1006: bt = (bt | mixerTA) & (bn | mixerNA); /* 0 or 0xffff */
1007: Tone3Voices = bt & YM_MASK_1VOICE; /* 0 or 0x1f */
1008: bt = -( (posB>>24) & 1);
1009: bt = (bt | mixerTB) & (bn | mixerNB);
1010: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 5;
1011: bt = -( (posC>>24) & 1);
1012: bt = (bt | mixerTC) & (bn | mixerNC);
1013: Tone3Voices |= ( bt & YM_MASK_1VOICE ) << 10;
1014:
1015: /* Combine fixed volumes and envelope volumes and keep the resulting */
1016: /* volumes depending on the output state of each voice (0 or 0x1f) */
1017: Tone3Voices &= ( Env3Voices | Vol3Voices );
1018:
1019: /* D/A conversion of the 3 volumes into a sample using a precomputed conversion table */
1.1.1.18 root 1020:
1021: if (stepA == 0 && (Tone3Voices & YM_MASK_A) > 1)
1022: Tone3Voices -= 1; /* Voice A AC component removed; Transient DC component remains */
1023:
1024: if (stepB == 0 && (Tone3Voices & YM_MASK_B) > 1<<5)
1025: Tone3Voices -= 1<<5; /* Voice B AC component removed; Transient DC component remains */
1026:
1027: if (stepC == 0 && (Tone3Voices & YM_MASK_C) > 1<<10)
1028: Tone3Voices -= 1<<10; /* Voice C AC component removed; Transient DC component remains */
1029:
1.1.1.16 root 1030: sample = ymout5[ Tone3Voices ]; /* 16 bits signed value */
1031:
1032:
1033: /* Increment positions */
1034: posA += stepA;
1035: posB += stepB;
1036: posC += stepC;
1037: noisePos += noiseStep;
1038:
1039: envPos += envStep;
1040: if ( envPos >= (3*32) << 24 ) /* blocks 0, 1 and 2 were used (envPos 0 to 95) */
1041: envPos -= (2*32) << 24; /* replay/loop blocks 1 and 2 (envPos 32 to 95) */
1042:
1043: /* Apply low pass filter ? */
1044: if ( UseLowPassFilter )
1.1.1.18 root 1045: return LowPassFilter(sample);
1046: else
1047: return PWMaliasFilter(sample);
1.1.1.16 root 1048: }
1049: #endif
1050:
1.1.1.12 root 1051:
1.1.1.2 root 1052: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1053: /**
1.1.1.12 root 1054: * Update internal variables (steps, volume masks, ...) each
1055: * time an YM register is changed.
1.1.1.11 root 1056: */
1.1.1.16 root 1057: #ifndef NEWSTEP
1058: #define BIT_SHIFT 31
1059: #else
1060: #define BIT_SHIFT 24
1061: #endif
1.1.1.12 root 1062: void Sound_WriteReg( int reg , Uint8 data )
1.1.1.7 root 1063: {
1.1.1.12 root 1064: switch (reg)
1065: {
1066: case 0:
1067: SoundRegs[0] = data;
1068: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 1069: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1070: break;
1071:
1072: case 1:
1073: SoundRegs[1] = data & 0x0f;
1074: stepA = Ym2149_ToneStepCompute ( SoundRegs[1] , SoundRegs[0] );
1.1.1.16 root 1075: if (!stepA) posA = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1076: break;
1077:
1078: case 2:
1079: SoundRegs[2] = data;
1080: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 1081: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1082: break;
1.1.1.7 root 1083:
1.1.1.12 root 1084: case 3:
1085: SoundRegs[3] = data & 0x0f;
1086: stepB = Ym2149_ToneStepCompute ( SoundRegs[3] , SoundRegs[2] );
1.1.1.16 root 1087: if (!stepB) posB = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1088: break;
1089:
1090: case 4:
1091: SoundRegs[4] = data;
1092: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 1093: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1094: break;
1095:
1096: case 5:
1097: SoundRegs[5] = data & 0x0f;
1098: stepC = Ym2149_ToneStepCompute ( SoundRegs[5] , SoundRegs[4] );
1.1.1.16 root 1099: if (!stepC) posC = 1u<<BIT_SHIFT; // Assume output always 1 if 0 period (for Digi-sample)
1.1.1.12 root 1100: break;
1101:
1102: case 6:
1103: SoundRegs[6] = data & 0x1f;
1104: noiseStep = Ym2149_NoiseStepCompute ( SoundRegs[6] );
1105: if (!noiseStep)
1106: {
1107: noisePos = 0;
1108: currentNoise = 0xffff;
1109: }
1110: break;
1111:
1112: case 7:
1113: SoundRegs[7] = data & 0x3f; /* ignore bits 6 and 7 */
1114: mixerTA = (data&(1<<0)) ? 0xffff : 0;
1115: mixerTB = (data&(1<<1)) ? 0xffff : 0;
1116: mixerTC = (data&(1<<2)) ? 0xffff : 0;
1117: mixerNA = (data&(1<<3)) ? 0xffff : 0;
1118: mixerNB = (data&(1<<4)) ? 0xffff : 0;
1119: mixerNC = (data&(1<<5)) ? 0xffff : 0;
1120: break;
1121:
1122: case 8:
1123: SoundRegs[8] = data & 0x1f;
1124: if ( data & 0x10 )
1125: {
1126: EnvMask3Voices |= YM_MASK_A; /* env ON */
1127: Vol3Voices &= ~YM_MASK_A; /* fixed vol OFF */
1128: }
1129: else
1130: {
1131: EnvMask3Voices &= ~YM_MASK_A; /* env OFF */
1132: Vol3Voices &= ~YM_MASK_A; /* clear previous vol */
1133: Vol3Voices |= YmVolume4to5[ SoundRegs[8] ]; /* fixed vol ON */
1134: }
1135: break;
1.1.1.14 root 1136:
1.1.1.12 root 1137: case 9:
1138: SoundRegs[9] = data & 0x1f;
1139: if ( data & 0x10 )
1140: {
1141: EnvMask3Voices |= YM_MASK_B; /* env ON */
1142: Vol3Voices &= ~YM_MASK_B; /* fixed vol OFF */
1143: }
1144: else
1145: {
1146: EnvMask3Voices &= ~YM_MASK_B; /* env OFF */
1147: Vol3Voices &= ~YM_MASK_B; /* clear previous vol */
1148: Vol3Voices |= ( YmVolume4to5[ SoundRegs[9] ] ) << 5; /* fixed vol ON */
1149: }
1150: break;
1.1.1.14 root 1151:
1.1.1.12 root 1152: case 10:
1153: SoundRegs[10] = data & 0x1f;
1154: if ( data & 0x10 )
1155: {
1156: EnvMask3Voices |= YM_MASK_C; /* env ON */
1157: Vol3Voices &= ~YM_MASK_C; /* fixed vol OFF */
1158: }
1159: else
1160: {
1161: EnvMask3Voices &= ~YM_MASK_C; /* env OFF */
1162: Vol3Voices &= ~YM_MASK_C; /* clear previous vol */
1163: Vol3Voices |= ( YmVolume4to5[ SoundRegs[10] ] ) << 10; /* fixed vol ON */
1164: }
1165: break;
1166:
1167: case 11:
1168: SoundRegs[11] = data;
1169: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1170: break;
1171:
1172: case 12:
1173: SoundRegs[12] = data;
1174: envStep = Ym2149_EnvStepCompute ( SoundRegs[12] , SoundRegs[11] );
1175: break;
1176:
1177: case 13:
1178: SoundRegs[13] = data & 0xf;
1179: envPos = 0; /* when writing to EnvShape, we must reset the EnvPos */
1180: envShape = SoundRegs[13];
1.1.1.14 root 1181: bEnvelopeFreqFlag = true; /* used for YmFormat saving */
1.1.1.12 root 1182: break;
1183:
1184: }
1185: }
1186:
1187:
1188:
1189: /*-----------------------------------------------------------------------*/
1190: /**
1191: * Init random generator, sound tables and envelopes
1192: * (called only once when Hatari starts)
1193: */
1194: void Sound_Init(void)
1195: {
1196: /* Build volume/env tables, ... */
1197: Ym2149_Init();
1.1.1.14 root 1198:
1.1.1.11 root 1199: Sound_Reset();
1.1.1.7 root 1200: }
1201:
1202:
1203: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1204: /**
1.1.1.12 root 1205: * Reset the sound emulation (called from Reset_ST() in reset.c)
1.1.1.11 root 1206: */
1.1.1.7 root 1207: void Sound_Reset(void)
1208: {
1.1.1.11 root 1209: /* Lock audio system before accessing variables which are used by the
1210: * callback function, too! */
1211: Audio_Lock();
1.1.1.9 root 1212:
1.1.1.11 root 1213: /* Clear sound mixing buffer: */
1.1.1.12 root 1214: memset(MixBuffer, 0, sizeof(MixBuffer));
1.1.1.7 root 1215:
1.1.1.11 root 1216: /* Clear cycle counts, buffer index and register '13' flags */
1217: Cycles_SetCounter(CYCLES_COUNTER_SOUND, 0);
1.1.1.14 root 1218: bEnvelopeFreqFlag = false;
1219:
1.1.1.11 root 1220: CompleteSndBufIdx = 0;
1221: /* We do not start with 0 here to fake some initial samples: */
1222: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1223: ActiveSndBufIdx = nGeneratedSamples % MIXBUFFER_SIZE;
1.1.1.16 root 1224: SamplesPerFrame = SAMPLES_PER_FRAME;
1225: CurrentSamplesNb = 0;
1226: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1227: //fprintf ( stderr , "Sound_Reset SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1228: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.7 root 1229:
1.1.1.12 root 1230: Ym2149_Reset();
1.1.1.9 root 1231:
1.1.1.11 root 1232: Audio_Unlock();
1.1.1.7 root 1233: }
1234:
1235:
1236: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1237: /**
1238: * Reset the sound buffer index variables.
1.1.1.16 root 1239: * Very important : this function should only be called by setting
1240: * Sound_BufferIndexNeedReset=true ; sound buffer index should be reset
1241: * only after the sound for the whole VBL was updated (CurrentSamplesNb returns to 0)
1242: * else it will alter the value of DMA Frame Count ($ff8909/0b/0d) and
1243: * could cause crashes in some programs.
1.1.1.11 root 1244: */
1.1.1.9 root 1245: void Sound_ResetBufferIndex(void)
1.1.1.7 root 1246: {
1.1.1.11 root 1247: Audio_Lock();
1248: nGeneratedSamples = SoundBufferSize + SAMPLES_PER_FRAME;
1249: ActiveSndBufIdx = (CompleteSndBufIdx + nGeneratedSamples) % MIXBUFFER_SIZE;
1.1.1.16 root 1250: SamplesPerFrame = SAMPLES_PER_FRAME;
1251: CurrentSamplesNb = 0;
1252: ActiveSndBufIdxAvi = ActiveSndBufIdx;
1.1.1.15 root 1253: //fprintf ( stderr , "Sound_ResetBufferIndex SoundBufferSize %d SAMPLES_PER_FRAME %d nGeneratedSamples %d , ActiveSndBufIdx %d\n" ,
1254: // SoundBufferSize , SAMPLES_PER_FRAME, nGeneratedSamples , ActiveSndBufIdx );
1.1.1.11 root 1255: Audio_Unlock();
1.1.1.7 root 1256: }
1257:
1258:
1259: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1260: /**
1261: * Save/Restore snapshot of local variables('MemorySnapShot_Store' handles type)
1262: */
1.1.1.12 root 1263: void Sound_MemorySnapShot_Capture(bool bSave)
1.1.1.7 root 1264: {
1.1.1.11 root 1265: /* Save/Restore details */
1.1.1.12 root 1266: MemorySnapShot_Store(&stepA, sizeof(stepA));
1267: MemorySnapShot_Store(&stepB, sizeof(stepB));
1268: MemorySnapShot_Store(&stepC, sizeof(stepC));
1269: MemorySnapShot_Store(&posA, sizeof(posA));
1270: MemorySnapShot_Store(&posB, sizeof(posB));
1271: MemorySnapShot_Store(&posC, sizeof(posC));
1272:
1273: MemorySnapShot_Store(&mixerTA, sizeof(mixerTA));
1274: MemorySnapShot_Store(&mixerTB, sizeof(mixerTB));
1275: MemorySnapShot_Store(&mixerTC, sizeof(mixerTC));
1276: MemorySnapShot_Store(&mixerNA, sizeof(mixerNA));
1277: MemorySnapShot_Store(&mixerNB, sizeof(mixerNB));
1278: MemorySnapShot_Store(&mixerNC, sizeof(mixerNC));
1279:
1280: MemorySnapShot_Store(&noiseStep, sizeof(noiseStep));
1281: MemorySnapShot_Store(&noisePos, sizeof(noisePos));
1282: MemorySnapShot_Store(¤tNoise, sizeof(currentNoise));
1283: MemorySnapShot_Store(&RndRack, sizeof(RndRack));
1284:
1285: MemorySnapShot_Store(&envStep, sizeof(envStep));
1286: MemorySnapShot_Store(&envPos, sizeof(envPos));
1287: MemorySnapShot_Store(&envShape, sizeof(envShape));
1.1.1.14 root 1288:
1.1.1.12 root 1289: MemorySnapShot_Store(&EnvMask3Voices, sizeof(EnvMask3Voices));
1290: MemorySnapShot_Store(&Vol3Voices, sizeof(Vol3Voices));
1.1.1.14 root 1291:
1.1.1.12 root 1292: MemorySnapShot_Store(SoundRegs, sizeof(SoundRegs));
1293:
1.1.1.14 root 1294: // MemorySnapShot_Store(&YmVolumeMixing, sizeof(YmVolumeMixing));
1295: // MemorySnapShot_Store(&UseLowPassFilter, sizeof(UseLowPassFilter));
1.1.1.7 root 1296: }
1297:
1298:
1299: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1300: /**
1301: * Find how many samples to generate and store in 'nSamplesToGenerate'
1302: * Also update sound cycles counter to store how many we actually did
1303: * so generates set amount each frame.
1.1.1.16 root 1304: * If FillFrame is true, this means we reach the end of the VBL and me must
1305: * add as many samples as necessary to get a total of SamplesPerFrame
1306: * for this VBL.
1.1.1.11 root 1307: */
1.1.1.16 root 1308: static int Sound_SetSamplesPassed(bool FillFrame)
1.1 root 1309: {
1.1.1.11 root 1310: int nSoundCycles;
1.1.1.16 root 1311: int SamplesToGenerate; /* How many samples are needed for this time-frame */
1312:
1313: nSoundCycles = Cycles_GetCounter(CYCLES_COUNTER_VIDEO);
1.1.1.11 root 1314:
1.1.1.16 root 1315: /* example : 160256 cycles per VBL, 44Khz = 882 samples per VBL at 50 Hz */
1316: /* 882/160256 samples per cpu clock cycle */
1.1.1.11 root 1317:
1.1.1.16 root 1318: /* Total number of samples that we should have at this point of the VBL */
1319: SamplesToGenerate = nSoundCycles * SamplesPerFrame
1320: / ClocksTimings_GetCyclesPerVBL ( ConfigureParams.System.nMachineType , nScreenRefreshRate );
1.1.1.11 root 1321:
1.1.1.16 root 1322: //if (SamplesToGenerate > SamplesPerFrame )
1323: //fprintf ( stderr , "over run %d %d\n" , SamplesPerFrame , SamplesToGenerate );
1.1.1.11 root 1324:
1.1.1.16 root 1325: if (SamplesToGenerate > SamplesPerFrame)
1326: SamplesToGenerate = SamplesPerFrame;
1327:
1328: SamplesToGenerate -= CurrentSamplesNb; /* don't count samples that were already generated up to now */
1329: if ( SamplesToGenerate < 0 )
1330: SamplesToGenerate = 0;
1331:
1332:
1333: /* If we're called from the VBL interrupt (FillFrame==true), we must ensure we have */
1334: /* an exact total of SamplesPerFrame samples during a full VBL (we take into account */
1335: /* the samples that were already generated during this VBL) */
1336: if ( FillFrame )
1337: {
1338: SamplesToGenerate = SamplesPerFrame - CurrentSamplesNb; /* how many samples are missing to reach SamplesPerFrame */
1339: if ( SamplesToGenerate < 0 )
1340: SamplesToGenerate = 0;
1341: }
1.1.1.11 root 1342:
1.1.1.16 root 1343: /* Check we don't fill the sound's ring buffer before it's played by Audio_Callback() */
1344: /* This should never happen, except if the system suffers major slowdown due to other */
1345: /* processes or if we run in fast forward mode. */
1346: /* In the case of slowdown, we set Sound_BufferIndexNeedReset to "resync" the working */
1347: /* buffer's index ActiveSndBufIdx with the system buffer's index CompleteSndBufIdx. */
1348: /* In the case of fast forward, we do nothing here, Sound_BufferIndexNeedReset will be */
1349: /* set when the user exits fast forward mode. */
1350: if ( ( SamplesToGenerate > MIXBUFFER_SIZE - nGeneratedSamples ) && ( ConfigureParams.System.bFastForward == false )
1351: && ( ConfigureParams.Sound.bEnableSound == true ) )
1.1.1.11 root 1352: {
1.1.1.16 root 1353: Log_Printf ( LOG_WARN , "Your system is too slow, some sound samples were not correctly emulated\n" );
1354: Sound_BufferIndexNeedReset = true;
1.1.1.11 root 1355: }
1.1.1.16 root 1356:
1357: //fprintf ( stderr , "vbl %d hbl %d samp_gen %d / %d frac %lx\n" , nVBLs , nHBL , SamplesToGenerate , SamplesPerFrame , (long int)SamplesPerFrame_unrounded );
1358:
1359: return SamplesToGenerate;
1.1 root 1360: }
1361:
1.1.1.5 root 1362:
1.1.1.2 root 1363: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1364: /**
1365: * Generate samples for all channels during this time-frame
1366: */
1.1.1.16 root 1367: static void Sound_GenerateSamples(int SamplesToGenerate)
1.1 root 1368: {
1.1.1.12 root 1369: int i;
1370: int idx;
1.1.1.14 root 1371:
1.1.1.16 root 1372: if (SamplesToGenerate <= 0)
1.1.1.12 root 1373: return;
1.1.1.14 root 1374:
1.1.1.15 root 1375: if (ConfigureParams.System.nMachineType == MACHINE_FALCON)
1.1.1.11 root 1376: {
1.1.1.16 root 1377: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1378: {
1379: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1380: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1381: }
1382: /* If Falcon emulation, crossbar does the job */
1.1.1.16 root 1383: Crossbar_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.11 root 1384: }
1.1.1.15 root 1385: else if (ConfigureParams.System.nMachineType != MACHINE_ST)
1386: {
1.1.1.16 root 1387: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1388: {
1389: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.18 root 1390: MixBuffer[idx][0] = MixBuffer[idx][1] = YM2149_NextSample();
1.1.1.15 root 1391: }
1.1.1.17 root 1392: /* If Ste or TT emulation, DmaSnd does mixing and filtering */
1.1.1.16 root 1393: DmaSnd_GenerateSamples(ActiveSndBufIdx, SamplesToGenerate);
1.1.1.15 root 1394: }
1395: else if (ConfigureParams.System.nMachineType == MACHINE_ST)
1396: {
1.1.1.16 root 1397: for (i = 0; i < SamplesToGenerate; i++)
1.1.1.15 root 1398: {
1399: idx = (ActiveSndBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.17 root 1400: MixBuffer[idx][0] = MixBuffer[idx][1] = Subsonic_IIR_HPF_Left( YM2149_NextSample() );
1.1.1.15 root 1401: }
1402: }
1.1.1.12 root 1403:
1.1.1.16 root 1404: ActiveSndBufIdx = (ActiveSndBufIdx + SamplesToGenerate) % MIXBUFFER_SIZE;
1405: nGeneratedSamples += SamplesToGenerate;
1406: CurrentSamplesNb += SamplesToGenerate; /* number of samples generated for current VBL */
1.1 root 1407: }
1408:
1.1.1.5 root 1409:
1.1.1.2 root 1410: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1411: /**
1412: * This is called to built samples up until this clock cycle
1.1.1.16 root 1413: * Sound_Update can be called several times during a VBL ; we must ensure
1414: * that we generate exactly SamplesPerFrame samples between 2 calls
1415: * to Sound_Update_VBL.
1.1.1.11 root 1416: */
1.1.1.16 root 1417: void Sound_Update(bool FillFrame)
1.1 root 1418: {
1.1.1.11 root 1419: int OldSndBufIdx = ActiveSndBufIdx;
1.1.1.16 root 1420: int SamplesToGenerate;
1.1.1.5 root 1421:
1.1.1.11 root 1422: /* Make sure that we don't interfere with the audio callback function */
1423: Audio_Lock();
1.1.1.6 root 1424:
1.1.1.16 root 1425: /* Find how many samples to generate */
1426: SamplesToGenerate = Sound_SetSamplesPassed( FillFrame );
1427:
1.1.1.11 root 1428: /* And generate */
1.1.1.16 root 1429: Sound_GenerateSamples( SamplesToGenerate );
1.1 root 1430:
1.1.1.11 root 1431: /* Allow audio callback function to occur again */
1432: Audio_Unlock();
1.1.1.6 root 1433:
1.1.1.11 root 1434: /* Save to WAV file, if open */
1435: if (bRecordingWav)
1.1.1.16 root 1436: WAVFormat_Update(MixBuffer, OldSndBufIdx, SamplesToGenerate);
1.1 root 1437: }
1438:
1.1.1.5 root 1439:
1.1.1.2 root 1440: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1441: /**
1.1.1.16 root 1442: * On the end of each VBL, complete audio buffer to reach SamplesPerFrame samples.
1443: * As Sound_Update(false) could be called several times during the VBL, the audio
1444: * buffer might be already partially filled.
1445: * We must first complete the buffer using the same value of SamplesPerFrame
1446: * by calling Sound_Update(true) ; then we can compute a new value for
1447: * SamplesPerFrame that will be used for the next VBL to come.
1.1.1.11 root 1448: */
1.1.1.5 root 1449: void Sound_Update_VBL(void)
1.1 root 1450: {
1.1.1.16 root 1451: Sound_Update(true); /* generate as many samples as needed to fill this VBL */
1452: //fprintf ( stderr , "vbl done %d %d\n" , SamplesPerFrame , CurrentSamplesNb );
1.1.1.5 root 1453:
1.1.1.16 root 1454: CurrentSamplesNb = 0; /* VBL is complete, reset counter for next VBL */
1455:
1456: /*Compute a fractional equivalent of SamplesPerFrame for the next VBL, to avoid rounding propagation */
1457: SamplesPerFrame_unrounded += (yms64) ClocksTimings_GetSamplesPerVBL ( ConfigureParams.System.nMachineType ,
1458: nScreenRefreshRate , nAudioFrequency );
1459: SamplesPerFrame = SamplesPerFrame_unrounded >> 28; /* use integer part */
1460: SamplesPerFrame_unrounded &= 0x0fffffff; /* keep fractional part in the lower 28 bits */
1461:
1462: /* Reset sound buffer if needed (after pause, fast forward, slow system, ...) */
1463: if ( Sound_BufferIndexNeedReset )
1464: {
1465: Sound_ResetBufferIndex ();
1466: Sound_BufferIndexNeedReset = false;
1467: }
1468:
1469: /* Record AVI audio frame is necessary */
1.1.1.15 root 1470: if ( bRecordingAvi )
1471: {
1.1.1.16 root 1472: int Len;
1.1.1.15 root 1473:
1.1.1.16 root 1474: Len = ActiveSndBufIdx - ActiveSndBufIdxAvi; /* number of generated samples for this frame */
1475: if ( Len < 0 )
1476: Len += MIXBUFFER_SIZE; /* end of ring buffer was reached */
1.1.1.15 root 1477:
1.1.1.16 root 1478: Avi_RecordAudioStream ( MixBuffer , ActiveSndBufIdxAvi , Len );
1.1.1.15 root 1479: }
1480:
1.1.1.16 root 1481: ActiveSndBufIdxAvi = ActiveSndBufIdx; /* save new position for next AVI audio frame */
1482:
1.1.1.11 root 1483: /* Clear write to register '13', used for YM file saving */
1.1.1.14 root 1484: bEnvelopeFreqFlag = false;
1.1 root 1485: }
1486:
1487:
1.1.1.2 root 1488: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1489: /**
1490: * Start recording sound, as .YM or .WAV output
1491: */
1.1.1.12 root 1492: bool Sound_BeginRecording(char *pszCaptureFileName)
1.1 root 1493: {
1.1.1.12 root 1494: bool bRet;
1.1.1.7 root 1495:
1.1.1.11 root 1496: if (!pszCaptureFileName || strlen(pszCaptureFileName) <= 3)
1497: {
1498: Log_Printf(LOG_ERROR, "Illegal sound recording file name!\n");
1.1.1.14 root 1499: return false;
1.1.1.11 root 1500: }
1501:
1502: /* Did specify .YM or .WAV? If neither report error */
1503: if (File_DoesFileExtensionMatch(pszCaptureFileName,".ym"))
1504: bRet = YMFormat_BeginRecording(pszCaptureFileName);
1505: else if (File_DoesFileExtensionMatch(pszCaptureFileName,".wav"))
1506: bRet = WAVFormat_OpenFile(pszCaptureFileName);
1507: else
1508: {
1509: Log_AlertDlg(LOG_ERROR, "Unknown Sound Recording format.\n"
1510: "Please specify a .YM or .WAV output file.");
1.1.1.14 root 1511: bRet = false;
1.1.1.11 root 1512: }
1513:
1514: return bRet;
1.1 root 1515: }
1516:
1.1.1.5 root 1517:
1.1.1.2 root 1518: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1519: /**
1520: * End sound recording
1521: */
1.1.1.7 root 1522: void Sound_EndRecording(void)
1.1 root 1523: {
1.1.1.11 root 1524: /* Stop sound recording and close files */
1525: if (bRecordingYM)
1526: YMFormat_EndRecording();
1527: if (bRecordingWav)
1528: WAVFormat_CloseFile();
1.1 root 1529: }
1530:
1.1.1.6 root 1531:
1.1.1.2 root 1532: /*-----------------------------------------------------------------------*/
1.1.1.11 root 1533: /**
1534: * Are we recording sound data?
1535: */
1.1.1.12 root 1536: bool Sound_AreWeRecording(void)
1.1 root 1537: {
1.1.1.11 root 1538: return (bRecordingYM || bRecordingWav);
1.1 root 1539: }
1.1.1.12 root 1540:
1.1.1.16 root 1541:
1542: /*-----------------------------------------------------------------------*/
1543: /**
1544: * Rebuild volume conversion table
1545: */
1546: void Sound_SetYmVolumeMixing(void)
1547: {
1548: /* Build the volume conversion table */
1549: Ym2149_BuildVolumeTable();
1550: }
1551:
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