![[BACK]](/cvsweb/icons/back.gif){kind=icon}
Return to dmaSnd.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 dmaSnd.c CVS log | Up to [HATARI the Atari ST Emulator] / hatari / src |
1.1 root 1: /*
2: Hatari - dmaSnd.c
3:
4: This file is distributed under the GNU Public License, version 2 or at
5: your option any later version. Read the file gpl.txt for details.
6:
7: STE DMA sound emulation. Does not seem to be very hard at first glance,
8: but since the DMA sound has to be mixed together with the PSG sound and
9: the output frequency of the host computer differs from the DMA sound
10: frequency, the copy function is a little bit complicated.
1.1.1.7 root 11: The update function also triggers the ST interrupts (Timer A and MFP-i7)
12: which are often used in ST programs for setting a new sound frame after
13: the old one has finished.
14:
1.1.1.8 root 15: To support programs that write into the frame buffer while it's played,
16: we should update dma sound on each video HBL.
17: This is also how it works on a real STE : bytes are read by the DMA
18: at the end of each HBL and stored in a small FIFO (8 bytes) that is sent
19: to the DAC depending on the chosen DMA output freq.
20:
1.1.1.7 root 21: Falcon sound emulation is all taken into account in crossbar.c
1.1 root 22:
23:
24: Hardware I/O registers:
25:
26: $FF8900 (word) : DMA sound control register
27: $FF8903 (byte) : Frame Start Hi
28: $FF8905 (byte) : Frame Start Mi
29: $FF8907 (byte) : Frame Start Lo
30: $FF8909 (byte) : Frame Count Hi
31: $FF890B (byte) : Frame Count Mi
32: $FF890D (byte) : Frame Count Lo
33: $FF890F (byte) : Frame End Hi
34: $FF8911 (byte) : Frame End Mi
35: $FF8913 (byte) : Frame End Lo
36: $FF8920 (word) : Sound Mode Control (frequency, mono/stereo)
37: $FF8922 (byte) : Microwire Data Register
38: $FF8924 (byte) : Microwire Mask Register
1.1.1.7 root 39:
40:
41: The Microwire and LMC 1992 commands :
42:
43: a command looks like: 10 CCC DDD DDD
44:
45: chipset address : 10
46: command :
47: 000 XXX XDD Mixing
48: 00 : DMA and (YM2149 - 12dB) mixing
49: 01 : DMA and YM2149 mixing
50: 10 : DMA only
51: 11 : Reserved
52:
53: 001 XXD DDD Bass
54: 0 000 : -12 dB
55: 0 110 : 0 dB
56: 1 100 : +12 dB
57:
58: 002 XXD DDD Treble
59: 0 000 : -12 dB
60: 0 110 : 0 dB
61: 1 100 : +12 dB
62:
63: 003 DDD DDD Master volume
64: 000 000 : -80 dB
65: 010 100 : -40 dB
66: 101 XXX : 0 dB
67:
68: 004 XDD DDD Right channel volume
69: 00 000 : -40 dB
70: 01 010 : -20 dB
71: 10 1XX : 0 dB
72:
73: 005 XDD DDD Left channel volume
74: 00 000 : -40 dB
75: 01 010 : -20 dB
76: 10 1XX : 0 dB
77:
78: Other : undefined
79:
80: LMC1992 IIR code Copyright by David Savinkoff 2010
81:
82: A first order bass filter is multiplied with a
83: first order treble filter to make a single
84: second order IIR shelf filter.
85:
86: Sound is stereo filtered by Boosting or Cutting
87: the Bass and Treble by +/-12dB in 2dB steps.
88:
89: This filter sounds exactly as the Atari TT or STE.
90: Sampling frequency = selectable
91: Bass turnover = 118.276Hz (8.2nF on LM1992 bass)
92: Treble turnover = 8438.756Hz (8.2nF on LM1992 treble)
1.1 root 93: */
1.1.1.7 root 94:
95:
1.1.1.5 root 96: const char DmaSnd_fileid[] = "Hatari dmaSnd.c : " __DATE__ " " __TIME__;
1.1 root 97:
98: #include "main.h"
99: #include "audio.h"
1.1.1.3 root 100: #include "configuration.h"
1.1 root 101: #include "dmaSnd.h"
1.1.1.7 root 102: #include "cycInt.h"
1.1 root 103: #include "ioMem.h"
1.1.1.6 root 104: #include "log.h"
1.1 root 105: #include "memorySnapShot.h"
106: #include "mfp.h"
107: #include "sound.h"
1.1.1.3 root 108: #include "stMemory.h"
1.1.1.7 root 109:
110: #define TONE_STEPS 13
111:
1.1.1.8 root 112: #define DMASND_FIFO_SIZE 8 /* 8 bytes : size of the DMA Audio's FIFO, filled on every HBL */
113: #define DMASND_FIFO_SIZE_MASK (DMASND_FIFO_SIZE-1) /* mask to keep FIFO_pos in 0-7 range */
114:
1.1.1.7 root 115:
116: /* Global variables that can be changed/read from other parts of Hatari */
117:
1.1.1.8 root 118: static void DmaSnd_Apply_LMC(int nMixBufIdx, int nSamplesToGenerate);
1.1.1.7 root 119: static void DmaSnd_Set_Tone_Level(int set_bass, int set_treb);
120: static float DmaSnd_IIRfilterL(float xn);
121: static float DmaSnd_IIRfilterR(float xn);
122: static struct first_order_s *DmaSnd_Treble_Shelf(float g, float fc, float Fs);
123: static struct first_order_s *DmaSnd_Bass_Shelf(float g, float fc, float Fs);
1.1.1.8 root 124: static Sint16 DmaSnd_LowPassFilterLeft(Sint16 in);
125: static Sint16 DmaSnd_LowPassFilterRight(Sint16 in);
126: static bool DmaSnd_LowPass;
1.1 root 127:
128:
129: Uint16 nDmaSoundControl; /* Sound control register */
130:
1.1.1.7 root 131: struct first_order_s { float a1, b0, b1; };
132: struct second_order_s { float a1, a2, b0, b1, b2; };
133:
134: struct dma_s {
135: Uint16 soundMode; /* Sound mode register */
136: Uint32 frameStartAddr; /* Sound frame start */
137: Uint32 frameEndAddr; /* Sound frame end */
1.1.1.8 root 138: Uint32 frameCounterAddr; /* Sound frame current address counter */
139:
140: /* Internal 8 byte FIFO */
141: Sint8 FIFO[ DMASND_FIFO_SIZE ];
142: Uint16 FIFO_Pos; /* from 0 to DMASND_FIFO_SIZE-1 */
143: Uint16 FIFO_NbBytes; /* from 0 to DMASND_FIFO_SIZE */
144:
1.1.1.9 ! root 145: Sint16 FrameLeft; /* latest values read from the FIFO */
1.1.1.8 root 146: Sint16 FrameRight;
1.1.1.7 root 147: };
148:
1.1.1.8 root 149: Sint64 frameCounter_float = 0;
150: bool DmaInitSample = false;
151:
152:
1.1.1.7 root 153: struct microwire_s {
154: Uint16 data; /* Microwire Data register */
155: Uint16 mask; /* Microwire Mask register */
1.1.1.9 ! root 156: Uint16 mwTransferSteps; /* Microwire shifting counter */
! 157: Uint16 pendingCyclesOver; /* Number of delayed cycles for the interrupt */
1.1.1.7 root 158: Uint16 mixing; /* Mixing command */
159: Uint16 bass; /* Bass command */
160: Uint16 treble; /* Treble command */
161: Uint16 masterVolume; /* Master volume command */
162: Uint16 leftVolume; /* Left channel volume command */
163: Uint16 rightVolume; /* Right channel volume command */
164: };
165:
166: struct lmc1992_s {
167: struct first_order_s bass_table[TONE_STEPS];
168: struct first_order_s treb_table[TONE_STEPS];
1.1.1.8 root 169: float coef[5]; /* IIR coefficients */
170: float left_gain;
171: float right_gain;
1.1.1.7 root 172: };
173:
174: static struct dma_s dma;
175: static struct microwire_s microwire;
176: static struct lmc1992_s lmc1992;
177:
178: /* dB = 20log(gain) : gain = antilog(dB/20) */
179: /* Table gain values = (int)(powf(10.0, dB/20.0)*65536.0 + 0.5) 2dB steps */
180:
181: /* Values for LMC1992 Master volume control (*65536) */
182: static const Uint16 LMC1992_Master_Volume_Table[64] =
183: {
184: 7, 8, 10, 13, 16, 21, 26, 33, 41, 52, /* -80dB */
185: 66, 83, 104, 131, 165, 207, 261, 328, 414, 521, /* -60dB */
186: 655, 825, 1039, 1308, 1646, 2072, 2609, 3285, 4135, 5206, /* -40dB */
187: 6554, 8250, 10387, 13076, 16462, 20724, 26090, 32846, 41350, 52057, /* -20dB */
188: 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, /* 0dB */
189: 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, /* 0dB */
190: 65535, 65535, 65535, 65535 /* 0dB */
191: };
1.1.1.6 root 192:
1.1.1.7 root 193: /* Values for LMC1992 Left and right volume control (*65536) */
194: static const Uint16 LMC1992_LeftRight_Volume_Table[32] =
195: {
196: 655, 825, 1039, 1308, 1646, 2072, 2609, 3285, 4135, 5206, /* -40dB */
197: 6554, 8250, 10387, 13076, 16462, 20724, 26090, 32846, 41350, 52057, /* -20dB */
198: 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, /* 0dB */
199: 65535, 65535 /* 0dB */
1.1 root 200: };
201:
1.1.1.7 root 202: /* Values for LMC1992 BASS and TREBLE */
203: static const Sint16 LMC1992_Bass_Treble_Table[16] =
204: {
205: 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 12, 12, 12
206: };
1.1 root 207:
1.1.1.7 root 208: static const int DmaSndSampleRates[4] =
1.1.1.3 root 209: {
1.1.1.8 root 210: 6258, 12517, 25033, 50066
1.1.1.3 root 211: };
212:
213:
1.1.1.8 root 214:
215: /*--------------------------------------------------------------*/
216: /* Local functions prototypes */
217: /*--------------------------------------------------------------*/
218:
219: static void DmaSnd_FIFO_Refill(void);
220: static Sint8 DmaSnd_FIFO_PullByte(void);
221: static void DmaSnd_FIFO_SetStereo(void);
222:
223: static int DmaSnd_DetectSampleRate(void);
224: static void DmaSnd_StartNewFrame(void);
225: static inline int DmaSnd_EndOfFrameReached(void);
226:
227:
1.1 root 228: /*-----------------------------------------------------------------------*/
1.1.1.3 root 229: /**
1.1.1.7 root 230: * Init DMA sound variables.
231: */
232: void DmaSnd_Init(void)
233: {
234: DmaSnd_Reset(1);
235: }
236:
237: /**
1.1.1.3 root 238: * Reset DMA sound variables.
239: */
1.1.1.4 root 240: void DmaSnd_Reset(bool bCold)
1.1 root 241: {
242: nDmaSoundControl = 0;
1.1.1.8 root 243: dma.soundMode = 0;
244:
245: /* [NP] Set start/end to 0 even on warm reset ? (fix 'Brace' by Diamond Design) */
246: IoMem[0xff8903] = 0; /* frame start addr = 0 */
247: IoMem[0xff8905] = 0;
248: IoMem[0xff8907] = 0;
249: IoMem[0xff890f] = 0; /* frame end addr = 0 */
250: IoMem[0xff8911] = 0;
251: IoMem[0xff8913] = 0;
252:
253: dma.FIFO_Pos = 0;
254: dma.FIFO_NbBytes = 0;
255: dma.FrameLeft = 0;
256: dma.FrameRight = 0;
1.1 root 257:
1.1.1.8 root 258: if ( bCold )
1.1 root 259: {
1.1.1.8 root 260: /* Microwire has no reset signal, it will keep its values on warm reset */
261: microwire.masterVolume = 7; /* -80 dB ; TOS 1.62 will put 0x28 (ie 65535) = 0 dB (max volume) */
262: microwire.leftVolume = 655; /* -40 dB ; TOS 1.62 will put 0x14 (ie 65535) = 0 dB (max volume) */
263: microwire.rightVolume = 655; /* -40 db ; TOS 1.62 will put 0x14 (ie 65535) = 0 dB (max volume) */
1.1.1.7 root 264: microwire.mixing = 0;
1.1.1.8 root 265: microwire.bass = 6; /* 0 dB (flat) */
266: microwire.treble = 6; /* 0 dB (flat) */
1.1 root 267: }
1.1.1.4 root 268:
1.1.1.7 root 269: /* Initialise microwire LMC1992 IIR filter parameters */
270: DmaSnd_Init_Bass_and_Treble_Tables();
1.1.1.6 root 271:
1.1.1.7 root 272: microwire.mwTransferSteps = 0;
1.1.1.9 ! root 273: microwire.pendingCyclesOver = 8;
1.1 root 274: }
275:
276: /*-----------------------------------------------------------------------*/
1.1.1.3 root 277: /**
278: * Save/Restore snapshot of local variables ('MemorySnapShot_Store' handles type)
279: */
1.1.1.4 root 280: void DmaSnd_MemorySnapShot_Capture(bool bSave)
1.1 root 281: {
282: /* Save/Restore details */
283: MemorySnapShot_Store(&nDmaSoundControl, sizeof(nDmaSoundControl));
1.1.1.7 root 284: MemorySnapShot_Store(&dma, sizeof(dma));
285: MemorySnapShot_Store(µwire, sizeof(microwire));
286: MemorySnapShot_Store(&lmc1992, sizeof(lmc1992));
1.1 root 287: }
288:
289:
1.1.1.8 root 290: /*-----------------------------------------------------------------------*/
291: /**
292: * This function is called on every HBL to ensure the DMA Audio's FIFO
293: * is kept full.
294: * In Hatari, the FIFO is handled like a ring buffer (to avoid memcopying bytes
295: * inside the FIFO when a byte is pushed/pulled).
296: * Note that the DMA fetches words, not bytes, so we read new data only
297: * when 2 bytes or more are missing.
298: * When end of frame is reached, we continue with a new frame if loop mode
299: * is on, else we stop DMA Audio.
300: */
301: static void DmaSnd_FIFO_Refill(void)
302: {
303: /* If DMA sound is OFF, don't update the FIFO */
304: if ( ( nDmaSoundControl & DMASNDCTRL_PLAY ) == 0)
305: return;
306:
307: /* If End Address == Start Address, don't update the FIFO */
308: if (dma.frameEndAddr == dma.frameStartAddr)
309: {
310: DmaSnd_EndOfFrameReached(); /* Stop dma audio if loop mode is off */
311: return;
312: }
313:
314: /* Refill the whole FIFO */
315: while ( DMASND_FIFO_SIZE - dma.FIFO_NbBytes >= 2 )
316: {
317: /* Add one word to the FIFO */
318: LOG_TRACE(TRACE_DMASND, "DMA snd fifo refill adr=%x pos %d nb %d %x %x\n", dma.frameCounterAddr , dma.FIFO_Pos , dma.FIFO_NbBytes ,
319: STRam[ dma.frameCounterAddr ] , STRam[ dma.frameCounterAddr+1 ] );
320:
321: dma.FIFO[ ( dma.FIFO_Pos+dma.FIFO_NbBytes+0 ) & DMASND_FIFO_SIZE_MASK ] = (Sint8)STRam[ dma.frameCounterAddr ]; /* add upper byte of the word */
322: dma.FIFO[ ( dma.FIFO_Pos+dma.FIFO_NbBytes+1 ) & DMASND_FIFO_SIZE_MASK ] = (Sint8)STRam[ dma.frameCounterAddr+1 ]; /* add lower byte of the word */
323:
324: dma.FIFO_NbBytes += 2; /* One word more in the FIFO */
325:
326: /* Increase current frame address and check if we reached frame's end */
327: dma.frameCounterAddr += 2;
328: if ( dma.frameCounterAddr == dma.frameEndAddr ) /* end of frame reached, should we loop or stop dma ? */
329: {
330: if ( DmaSnd_EndOfFrameReached() )
331: break; /* Loop mode off, dma audio is now turned off */
332: }
333: }
334: }
335:
336:
337: /*-----------------------------------------------------------------------*/
338: /**
339: * Pull one sample/byte from the DMA Audio's FIFO and decrease the number of
340: * remaining bytes.
341: * If the FIFO is empty, return 0 (empty sample)
342: * Note : on a real STE, the 8 bytes FIFO is refilled on each HBL, which gives
343: * a total of 313*8=125326 bytes per sec read by the DMA. As the max freq
344: * is 50066 Hz, the STE can play 100132 bytes per sec in stereo ; so on a real STE
345: * the FIFO can never be empty while DMA is ON.
346: * But on Hatari, if the user chooses an audio's output frequency that is much
347: * lower than the current DMA freq, audio will be updated less frequently than
348: * on each HBL and it could require to process more than DMASND_FIFO_SIZE in one
349: * call to DmaSnd_GenerateSamples(). This is why we allow DmaSnd_FIFO_Refill()
350: * to be called if FIFO is empty but DMA sound is still ON.
351: * This way, sound remains correct even if the user uses very low output freq.
352: */
353: static Sint8 DmaSnd_FIFO_PullByte(void)
354: {
355: Sint8 sample;
356:
357: if ( dma.FIFO_NbBytes == 0 )
358: {
359: DmaSnd_FIFO_Refill();
360: if ( dma.FIFO_NbBytes == 0 ) /* Refill didn't add any new bytes */
361: {
362: LOG_TRACE(TRACE_DMASND, "DMA snd fifo empty for pull\n" );
363: return 0;
364: }
365: }
366:
367:
368: LOG_TRACE(TRACE_DMASND, "DMA snd fifo pull pos %d nb %d %02x\n", dma.FIFO_Pos , dma.FIFO_NbBytes , (Uint8)dma.FIFO[ dma.FIFO_Pos ] );
369:
370: sample = dma.FIFO[ dma.FIFO_Pos ]; /* Get oldest byte from the FIFO */
371: dma.FIFO_Pos = (dma.FIFO_Pos+1) & DMASND_FIFO_SIZE_MASK;/* Pos to be pulled on next call */
372: dma.FIFO_NbBytes--; /* One byte less in the FIFO */
373:
374: return sample;
375: }
376:
377:
378: /*-----------------------------------------------------------------------*/
379: /**
380: * In case a program switches from mono to stereo, we must ensure that
381: * FIFO_pos is on even boundary to keep Left/Right bytes in the correct
382: * order (Left byte should be on even addresses and Right byte on odd ones).
383: * If this is not the case, we skip one byte.
384: */
385: static void DmaSnd_FIFO_SetStereo(void)
386: {
387: Uint16 NewPos;
388:
389: if ( dma.FIFO_Pos & 1 )
390: {
391: NewPos = (dma.FIFO_Pos+1) & DMASND_FIFO_SIZE_MASK; /* skip the byte on odd address */
392:
393: if ( nDmaSoundControl & DMASNDCTRL_PLAY ) /* print a log if we change while playing */
394: { LOG_TRACE(TRACE_DMASND, "DMA snd switching to stereo mode while playing mono FIFO_pos %d->%d\n", dma.FIFO_Pos , NewPos ); }
395: else
396: { LOG_TRACE(TRACE_DMASND, "DMA snd switching to stereo mode FIFO_pos %d->%d\n", dma.FIFO_Pos , NewPos ); }
397:
398: dma.FIFO_Pos = NewPos;
399:
400: if ( dma.FIFO_NbBytes > 0 )
401: dma.FIFO_NbBytes--; /* remove one byte if FIFO was not already empty */
402: }
403:
404: }
405:
406:
407: /*-----------------------------------------------------------------------*/
408: /**
409: * Returns the frequency corresponding to the 2 lower bits of dma.soundMode
410: */
1.1.1.7 root 411: static int DmaSnd_DetectSampleRate(void)
1.1.1.3 root 412: {
1.1.1.7 root 413: return DmaSndSampleRates[dma.soundMode & 3];
1.1.1.3 root 414: }
415:
416:
1.1 root 417: /*-----------------------------------------------------------------------*/
1.1.1.3 root 418: /**
419: * This function is called when a new sound frame is started.
1.1.1.8 root 420: * It copies the start and end address from the I/O registers and set
421: * the frame counter addr to the start of this new frame.
1.1.1.3 root 422: */
1.1 root 423: static void DmaSnd_StartNewFrame(void)
424: {
1.1.1.7 root 425: dma.frameStartAddr = (IoMem[0xff8903] << 16) | (IoMem[0xff8905] << 8) | (IoMem[0xff8907] & ~1);
426: dma.frameEndAddr = (IoMem[0xff890f] << 16) | (IoMem[0xff8911] << 8) | (IoMem[0xff8913] & ~1);
1.1 root 427:
1.1.1.8 root 428: dma.frameCounterAddr = dma.frameStartAddr;
1.1.1.6 root 429:
1.1.1.8 root 430: LOG_TRACE(TRACE_DMASND, "DMA snd new frame start=%x end=%x\n", dma.frameStartAddr, dma.frameEndAddr);
1.1 root 431: }
432:
433:
434: /*-----------------------------------------------------------------------*/
1.1.1.3 root 435: /**
1.1.1.7 root 436: * End-of-frame has been reached. Raise interrupts if needed.
1.1.1.6 root 437: * Returns true if DMA sound processing should be stopped now and false
1.1.1.7 root 438: * if it continues (DMA PLAYLOOP mode).
1.1.1.3 root 439: */
1.1.1.7 root 440: static inline int DmaSnd_EndOfFrameReached(void)
1.1 root 441: {
1.1.1.8 root 442: LOG_TRACE(TRACE_DMASND, "DMA snd end of frame\n");
443:
1.1.1.7 root 444: /* Raise end-of-frame interrupts (MFP-i7 and Time-A) */
445: MFP_InputOnChannel(MFP_TIMER_GPIP7_BIT, MFP_IERA, &MFP_IPRA);
446: if (MFP_TACR == 0x08) /* Is timer A in Event Count mode? */
447: MFP_TimerA_EventCount_Interrupt();
1.1 root 448:
1.1.1.7 root 449: if (nDmaSoundControl & DMASNDCTRL_PLAYLOOP)
450: {
451: DmaSnd_StartNewFrame();
1.1 root 452: }
1.1.1.7 root 453: else
1.1.1.6 root 454: {
1.1.1.7 root 455: nDmaSoundControl &= ~DMASNDCTRL_PLAY;
456: return true;
1.1.1.6 root 457: }
458:
1.1.1.7 root 459: return false;
1.1 root 460: }
461:
462:
463: /*-----------------------------------------------------------------------*/
1.1.1.3 root 464: /**
465: * Mix DMA sound sample with the normal PSG sound samples.
1.1.1.6 root 466: * Note: We adjust the volume level of the 8-bit DMA samples to factor
1.1.1.7 root 467: * 0.75 compared to the PSG sound samples.
1.1.1.8 root 468: *
469: * The following formula: -((256*3/4)/4)/4
470: *
471: * Multiply by 256 to convert 8 to 16 bits;
472: * DMA sound is 3/4 level of YM sound;
473: * Divide by 4 to account for MixBuffer[];
474: * Divide by 4 to account for DmaSnd_LowPassFilter;
475: * Multiply DMA sound by -1 because the LMC1992 inverts the signal
476: * ( YM sign is +1 :: -1(op-amp) * -1(Lmc1992) ).
1.1.1.3 root 477: */
1.1.1.8 root 478:
479:
1.1 root 480: void DmaSnd_GenerateSamples(int nMixBufIdx, int nSamplesToGenerate)
481: {
1.1.1.8 root 482: int i;
1.1.1.7 root 483: int nBufIdx;
1.1.1.8 root 484: Sint8 MonoByte , LeftByte , RightByte;
1.1.1.7 root 485: unsigned n;
1.1.1.8 root 486: Sint64 FreqRatio;
487:
488:
489: /* DMA Audio OFF and FIFO empty : process YM2149's output */
490: if ( !(nDmaSoundControl & DMASNDCTRL_PLAY) && ( dma.FIFO_NbBytes == 0 ) )
491: {
492: for (i = 0; i < nSamplesToGenerate; i++)
493: {
494: nBufIdx = (nMixBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.6 root 495:
1.1.1.8 root 496: switch (microwire.mixing) {
497: case 1:
498: /* YM2149 */
1.1.1.9 ! root 499: MixBuffer[nBufIdx][1] = MixBuffer[nBufIdx][0] + dma.FrameRight * -((256*3/4)/4)/4;
! 500: MixBuffer[nBufIdx][0] += dma.FrameLeft * -((256*3/4)/4)/4;
1.1.1.8 root 501: break;
502: default:
503: /* YM2149 - 12dB */
504: MixBuffer[nBufIdx][0] /= 4;
1.1.1.9 ! root 505: MixBuffer[nBufIdx][1] = MixBuffer[nBufIdx][0] + dma.FrameRight * -((256*3/4)/4)/4;
! 506: MixBuffer[nBufIdx][0] += dma.FrameLeft * -((256*3/4)/4)/4;
1.1.1.8 root 507: break;
508: }
509: }
510:
511: /* Apply LMC1992 sound modifications (Bass and Treble) */
512: DmaSnd_Apply_LMC ( nMixBufIdx , nSamplesToGenerate );
513:
1.1 root 514: return;
1.1.1.8 root 515: }
516:
1.1 root 517:
1.1.1.8 root 518: /* DMA Audio ON or FIFO not empty yet */
1.1 root 519:
1.1.1.8 root 520: /* Compute ratio between DMA's sound frequency and host computer's sound frequency, */
521: /* use << 32 to simulate floating point precision */
522: FreqRatio = ( ((Sint64)DmaSnd_DetectSampleRate()) << 32 ) / nAudioFrequency;
1.1.1.7 root 523:
524: if (dma.soundMode & DMASNDMODE_MONO)
1.1.1.3 root 525: {
526: /* Mono 8-bit */
1.1 root 527: for (i = 0; i < nSamplesToGenerate; i++)
528: {
1.1.1.8 root 529: if ( DmaInitSample )
530: {
531: MonoByte = DmaSnd_FIFO_PullByte ();
1.1.1.9 ! root 532: dma.FrameLeft = DmaSnd_LowPassFilterLeft( (Sint16)MonoByte );
! 533: dma.FrameRight = DmaSnd_LowPassFilterRight( (Sint16)MonoByte );
1.1.1.8 root 534: DmaInitSample = false;
1.1.1.7 root 535: }
536:
1.1 root 537: nBufIdx = (nMixBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.7 root 538:
539: switch (microwire.mixing) {
540: case 1:
541: /* DMA and YM2149 mixing */
1.1.1.9 ! root 542: MixBuffer[nBufIdx][0] = MixBuffer[nBufIdx][0] + dma.FrameLeft * -((256*3/4)/4)/4;
1.1.1.7 root 543: break;
544: case 2:
545: /* DMA sound only */
1.1.1.9 ! root 546: MixBuffer[nBufIdx][0] = dma.FrameLeft * -((256*3/4)/4)/4;
1.1.1.7 root 547: break;
548: default:
549: /* DMA and (YM2149 - 12dB) mixing */
550: /* instead of 16462 (-12dB), we approximate by 16384 */
1.1.1.9 ! root 551: MixBuffer[nBufIdx][0] = (dma.FrameLeft * -((256*3/4)/4)/4) +
1.1.1.7 root 552: (((Sint32)MixBuffer[nBufIdx][0] * 16384)/65536);
553: break;
554: }
555:
1.1.1.8 root 556: MixBuffer[nBufIdx][1] = MixBuffer[nBufIdx][0]; /* right = left */
557:
558: /* Increase freq counter */
559: frameCounter_float += FreqRatio;
560: n = frameCounter_float >> 32; /* number of samples to skip */
561: while ( n > 0 ) /* pull as many bytes from the FIFO as needed */
562: {
563: MonoByte = DmaSnd_FIFO_PullByte ();
1.1.1.9 ! root 564: dma.FrameLeft = DmaSnd_LowPassFilterLeft( (Sint16)MonoByte );
! 565: dma.FrameRight = DmaSnd_LowPassFilterRight( (Sint16)MonoByte );
1.1.1.8 root 566: n--;
1.1.1.7 root 567: }
1.1.1.8 root 568: frameCounter_float &= 0xffffffff; /* only keep the fractional part */
1.1 root 569: }
570: }
571: else
572: {
1.1.1.3 root 573: /* Stereo 8-bit */
1.1 root 574: for (i = 0; i < nSamplesToGenerate; i++)
575: {
1.1.1.8 root 576: if ( DmaInitSample )
577: {
578: LeftByte = DmaSnd_FIFO_PullByte ();
579: RightByte = DmaSnd_FIFO_PullByte ();
580: dma.FrameLeft = DmaSnd_LowPassFilterLeft( (Sint16)LeftByte );
581: dma.FrameRight = DmaSnd_LowPassFilterRight( (Sint16)RightByte );
582: DmaInitSample = false;
1.1.1.7 root 583: }
584:
1.1 root 585: nBufIdx = (nMixBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.8 root 586:
1.1.1.7 root 587: switch (microwire.mixing) {
588: case 1:
589: /* DMA and YM2149 mixing */
1.1.1.8 root 590: MixBuffer[nBufIdx][0] = MixBuffer[nBufIdx][0] + dma.FrameLeft * -((256*3/4)/4)/4;
591: MixBuffer[nBufIdx][1] = MixBuffer[nBufIdx][1] + dma.FrameRight * -((256*3/4)/4)/4;
1.1.1.7 root 592: break;
593: case 2:
594: /* DMA sound only */
1.1.1.8 root 595: MixBuffer[nBufIdx][0] = dma.FrameLeft * -((256*3/4)/4)/4;
596: MixBuffer[nBufIdx][1] = dma.FrameRight * -((256*3/4)/4)/4;
1.1.1.7 root 597: break;
598: default:
599: /* DMA and (YM2149 - 12dB) mixing */
600: /* instead of 16462 (-12dB), we approximate by 16384 */
1.1.1.8 root 601: MixBuffer[nBufIdx][0] = (dma.FrameLeft * -((256*3/4)/4)/4) +
1.1.1.7 root 602: (((Sint32)MixBuffer[nBufIdx][0] * 16384)/65536);
1.1.1.8 root 603: MixBuffer[nBufIdx][1] = (dma.FrameRight * -((256*3/4)/4)/4) +
1.1.1.7 root 604: (((Sint32)MixBuffer[nBufIdx][1] * 16384)/65536);
605: break;
606: }
607:
1.1.1.8 root 608: /* Increase freq counter */
609: frameCounter_float += FreqRatio;
610: n = frameCounter_float >> 32; /* number of samples to skip */
611: while ( n > 0 ) /* pull as many bytes from the FIFO as needed */
612: {
613: LeftByte = DmaSnd_FIFO_PullByte ();
614: RightByte = DmaSnd_FIFO_PullByte ();
615: dma.FrameLeft = DmaSnd_LowPassFilterLeft( (Sint16)LeftByte );
616: dma.FrameRight = DmaSnd_LowPassFilterRight( (Sint16)RightByte );
617: n--;
1.1.1.7 root 618: }
1.1.1.8 root 619: frameCounter_float &= 0xffffffff; /* only keep the fractional part */
1.1 root 620: }
621: }
1.1.1.7 root 622:
1.1.1.8 root 623: /* Apply LMC1992 sound modifications (Bass and Treble) */
624: DmaSnd_Apply_LMC ( nMixBufIdx , nSamplesToGenerate );
625: }
626:
627:
628: /*-----------------------------------------------------------------------*/
629: /**
630: * Apply LMC1992 sound modifications (Bass and Treble)
631: * The Bass and Treble get samples at nAudioFrequency rate.
632: * The tone control's sampling frequency must be at least 22050 Hz to sound good.
633: */
634: static void DmaSnd_Apply_LMC(int nMixBufIdx, int nSamplesToGenerate)
635: {
636: int nBufIdx;
637: int i;
1.1.1.7 root 638:
639: /* Apply LMC1992 sound modifications (Left, Right and Master Volume) */
640: for (i = 0; i < nSamplesToGenerate; i++) {
641: nBufIdx = (nMixBufIdx + i) % MIXBUFFER_SIZE;
1.1.1.9 ! root 642: MixBuffer[nBufIdx][0] = DmaSnd_IIRfilterL( Subsonic_IIR_HPF_Left(MixBuffer[nBufIdx][0]) );
! 643: MixBuffer[nBufIdx][1] = DmaSnd_IIRfilterR( Subsonic_IIR_HPF_Right(MixBuffer[nBufIdx][1]) );
1.1.1.8 root 644: }
1.1 root 645: }
646:
647:
648: /*-----------------------------------------------------------------------*/
1.1.1.3 root 649: /**
1.1.1.8 root 650: * STE DMA sound is using an 8 bytes FIFO that is checked and filled on each HBL
651: * (at 50066 Hz 8 bit stereo, the DMA requires approx 6.5 new bytes per HBL)
652: * Calling Sound_Update on each HBL allows to emulate some programs that modify
653: * the data between FrameStart and FrameEnd while DMA sound is ON
654: * (eg the demo 'Mental Hangover' or the game 'Power Up Plus')
655: * We first check if the FIFO needs to be refilled, then we call Sound_Update.
656: * This function should be called from the HBL's handler (in video.c)
1.1.1.3 root 657: */
1.1.1.8 root 658: void DmaSnd_STE_HBL_Update(void)
1.1 root 659: {
1.1.1.8 root 660: if ( ( ConfigureParams.System.nMachineType != MACHINE_STE )
661: && ( ConfigureParams.System.nMachineType != MACHINE_MEGA_STE ) )
662: return;
1.1 root 663:
1.1.1.8 root 664:
665: /* The DMA starts refilling the FIFO when display is OFF (eg cycle 376 in low res 50 Hz) */
666: DmaSnd_FIFO_Refill ();
667:
668: /* If DMA sound is ON or FIFO is not empty, update sound */
669: if ( (nDmaSoundControl & DMASNDCTRL_PLAY) || ( dma.FIFO_NbBytes > 0 ) )
670: Sound_Update(false);
671:
672: /* As long as display is OFF, the DMA will refill the FIFO after playing some samples during the HBL */
673: DmaSnd_FIFO_Refill ();
1.1 root 674: }
675:
676:
677: /*-----------------------------------------------------------------------*/
1.1.1.3 root 678: /**
1.1.1.8 root 679: * Return current frame counter address (value is always even)
1.1.1.3 root 680: */
1.1 root 681: static Uint32 DmaSnd_GetFrameCount(void)
682: {
683: Uint32 nActCount;
684:
1.1.1.8 root 685: /* Update sound to get the current DMA frame address */
686: Sound_Update(false);
687:
1.1 root 688: if (nDmaSoundControl & DMASNDCTRL_PLAY)
1.1.1.8 root 689: nActCount = dma.frameCounterAddr;
1.1 root 690: else
1.1.1.8 root 691: nActCount = (IoMem[0xff8903] << 16) | (IoMem[0xff8905] << 8) | (IoMem[0xff8907] & ~1);
1.1 root 692:
693: return nActCount;
694: }
695:
696:
697: /*-----------------------------------------------------------------------*/
1.1.1.3 root 698: /**
699: * Read word from sound control register (0xff8900).
700: */
1.1 root 701: void DmaSnd_SoundControl_ReadWord(void)
702: {
703: IoMem_WriteWord(0xff8900, nDmaSoundControl);
1.1.1.6 root 704:
1.1.1.7 root 705: LOG_TRACE(TRACE_DMASND, "DMA snd control read: 0x%04x\n", nDmaSoundControl);
1.1 root 706: }
707:
708:
709: /*-----------------------------------------------------------------------*/
1.1.1.3 root 710: /**
1.1.1.8 root 711: * Write word to sound control register (0xff8900).
712: */
713: void DmaSnd_SoundControl_WriteWord(void)
714: {
715: Uint16 nNewSndCtrl;
716:
717: LOG_TRACE(TRACE_DMASND, "DMA snd control write: 0x%04x\n", IoMem_ReadWord(0xff8900));
718:
719: /* Before starting/stopping DMA sound, create samples up until this point with current values */
720: Sound_Update(false);
721:
722: nNewSndCtrl = IoMem_ReadWord(0xff8900) & 3;
723:
724: if (!(nDmaSoundControl & DMASNDCTRL_PLAY) && (nNewSndCtrl & DMASNDCTRL_PLAY))
725: {
726: LOG_TRACE(TRACE_DMASND, "DMA snd control write: starting dma sound output\n");
727: DmaInitSample = true;
728: frameCounter_float = 0;
729: DmaSnd_StartNewFrame();
730: }
731: else if ((nDmaSoundControl & DMASNDCTRL_PLAY) && !(nNewSndCtrl & DMASNDCTRL_PLAY))
732: {
733: LOG_TRACE(TRACE_DMASND, "DMA snd control write: stopping dma sound output\n");
734: }
735:
736: nDmaSoundControl = nNewSndCtrl;
737: }
738:
739:
740: /*-----------------------------------------------------------------------*/
741: /**
1.1.1.3 root 742: * Read word from sound frame count high register (0xff8909).
743: */
1.1 root 744: void DmaSnd_FrameCountHigh_ReadByte(void)
745: {
746: IoMem_WriteByte(0xff8909, DmaSnd_GetFrameCount() >> 16);
747: }
748:
749:
750: /*-----------------------------------------------------------------------*/
1.1.1.3 root 751: /**
752: * Read word from sound frame count medium register (0xff890b).
753: */
1.1 root 754: void DmaSnd_FrameCountMed_ReadByte(void)
755: {
756: IoMem_WriteByte(0xff890b, DmaSnd_GetFrameCount() >> 8);
757: }
758:
759:
760: /*-----------------------------------------------------------------------*/
1.1.1.3 root 761: /**
762: * Read word from sound frame count low register (0xff890d).
763: */
1.1 root 764: void DmaSnd_FrameCountLow_ReadByte(void)
765: {
766: IoMem_WriteByte(0xff890d, DmaSnd_GetFrameCount());
767: }
768:
769:
770: /*-----------------------------------------------------------------------*/
1.1.1.3 root 771: /**
1.1.1.8 root 772: * Write bytes to various registers with no action.
773: */
774: void DmaSnd_FrameStartHigh_WriteByte(void)
775: {
776: LOG_TRACE(TRACE_DMASND, "DMA snd frame start high: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8903) ,
777: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
778: }
779:
780: void DmaSnd_FrameStartMed_WriteByte(void)
781: {
782: LOG_TRACE(TRACE_DMASND, "DMA snd frame start med: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8905) ,
783: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
784: }
785:
786: void DmaSnd_FrameStartLow_WriteByte(void)
787: {
788: LOG_TRACE(TRACE_DMASND, "DMA snd frame start low: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8907) ,
789: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
790: }
791:
792: void DmaSnd_FrameCountHigh_WriteByte(void)
793: {
794: LOG_TRACE(TRACE_DMASND, "DMA snd frame count high: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8909) ,
795: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
796: }
797:
798: void DmaSnd_FrameCountMed_WriteByte(void)
799: {
800: LOG_TRACE(TRACE_DMASND, "DMA snd frame count med: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff890b) ,
801: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
802: }
803:
804: void DmaSnd_FrameCountLow_WriteByte(void)
805: {
806: LOG_TRACE(TRACE_DMASND, "DMA snd frame count low: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff890d) ,
807: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
808: }
809:
810: void DmaSnd_FrameEndHigh_WriteByte(void)
811: {
812: LOG_TRACE(TRACE_DMASND, "DMA snd frame end high: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff890f) ,
813: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
814: }
815:
816: void DmaSnd_FrameEndMed_WriteByte(void)
817: {
818: LOG_TRACE(TRACE_DMASND, "DMA snd frame end med: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8911) ,
819: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
820: }
821:
822: void DmaSnd_FrameEndLow_WriteByte(void)
823: {
824: LOG_TRACE(TRACE_DMASND, "DMA snd frame end low: 0x%02x at pos %d/%d\n", IoMem_ReadByte(0xff8913) ,
825: dma.frameCounterAddr - dma.frameStartAddr , dma.frameEndAddr - dma.frameStartAddr );
826: }
827:
828:
829: /*-----------------------------------------------------------------------*/
830: /**
1.1.1.7 root 831: * Read word from sound mode register (0xff8921).
1.1.1.3 root 832: */
1.1.1.7 root 833: void DmaSnd_SoundModeCtrl_ReadByte(void)
1.1 root 834: {
1.1.1.7 root 835: IoMem_WriteByte(0xff8921, dma.soundMode);
1.1.1.6 root 836:
1.1.1.7 root 837: LOG_TRACE(TRACE_DMASND, "DMA snd mode read: 0x%02x\n", dma.soundMode);
1.1 root 838: }
839:
840:
841: /*-----------------------------------------------------------------------*/
1.1.1.3 root 842: /**
1.1.1.7 root 843: * Write word to sound mode register (0xff8921).
1.1.1.3 root 844: */
1.1.1.7 root 845: void DmaSnd_SoundModeCtrl_WriteByte(void)
1.1 root 846: {
1.1.1.8 root 847: Uint16 SoundModeNew;
848:
849: SoundModeNew = IoMem_ReadByte(0xff8921);
850:
851: LOG_TRACE(TRACE_DMASND, "DMA snd mode write: 0x%02x mode=%s freq=%d\n", SoundModeNew,
852: SoundModeNew & DMASNDMODE_MONO ? "mono" : "stereo" , DmaSndSampleRates[ SoundModeNew & 3 ]);
853:
854: /* We maskout to only bits that exist on a real STE */
855: SoundModeNew &= 0x8f;
1.1.1.3 root 856:
1.1.1.8 root 857: /* Are we switching from mono to stereo ? */
858: if ( ( dma.soundMode & DMASNDMODE_MONO ) && ( ( SoundModeNew & DMASNDMODE_MONO ) == 0 ) )
859: DmaSnd_FIFO_SetStereo ();
860:
861: dma.soundMode = SoundModeNew;
862: /* We also write the masked value back into the emulated hw registers so we have a correct value there */
1.1.1.7 root 863: IoMem_WriteByte(0xff8921, dma.soundMode);
1.1 root 864: }
865:
1.1.1.8 root 866:
1.1.1.6 root 867: /* ---------------------- Microwire / LMC 1992 ---------------------- */
868:
1.1.1.3 root 869: /**
1.1.1.4 root 870: * Handle the shifting/rotating of the microwire registers
871: * The microwire regs should be done after 16 usec = 32 NOPs = 128 cycles.
872: * That means we have to shift 16 times with a delay of 8 cycles.
1.1.1.3 root 873: */
1.1.1.4 root 874: void DmaSnd_InterruptHandler_Microwire(void)
1.1 root 875: {
1.1.1.7 root 876: Uint8 i, bit;
877: Uint16 saveData;
878:
1.1.1.9 ! root 879: /* How many cycle was this sound interrupt delayed (>= 0) */
! 880: microwire.pendingCyclesOver += -INT_CONVERT_FROM_INTERNAL ( PendingInterruptCount , INT_CPU_CYCLE );
1.1.1.4 root 881: /* Remove this interrupt from list and re-order */
1.1.1.7 root 882: CycInt_AcknowledgeInterrupt();
1.1.1.4 root 883:
1.1.1.9 ! root 884: /* Shift the mask and data according to the number of cycles (8 cycles for a shift) */
! 885: do
1.1 root 886: {
1.1.1.9 ! root 887: --microwire.mwTransferSteps;
! 888: /* Shift data register until it becomes zero. */
1.1.1.7 root 889: IoMem_WriteWord(0xff8922, microwire.data<<(16-microwire.mwTransferSteps));
1.1.1.9 ! root 890: /* Rotate mask register */
! 891: IoMem_WriteWord(0xff8924, (microwire.mask<<(16-microwire.mwTransferSteps))
! 892: |(microwire.mask>>microwire.mwTransferSteps));
! 893: /* 8 cycles for 1 shift */
! 894: microwire.pendingCyclesOver -= 8;
1.1 root 895: }
1.1.1.9 ! root 896: while ((microwire.mwTransferSteps != 0) && (microwire.pendingCyclesOver >= 8) );
1.1.1.4 root 897:
1.1.1.9 ! root 898: /* Is the transfer finished ? */
1.1.1.7 root 899: if (microwire.mwTransferSteps > 0)
1.1.1.4 root 900: {
1.1.1.9 ! root 901: /* No ==> start a new internal interrupt to continue to tranfer the data */
! 902: microwire.pendingCyclesOver = 8 - microwire.pendingCyclesOver;
! 903: CycInt_AddRelativeInterrupt(microwire.pendingCyclesOver, INT_CPU_CYCLE, INTERRUPT_DMASOUND_MICROWIRE);
1.1.1.4 root 904: }
1.1.1.9 ! root 905: else
! 906: {
! 907: /* Yes : decode the address + command word according to the binary mask */
1.1.1.7 root 908: bit = 0;
909: saveData = microwire.data;
910: microwire.data = 0;
911: for (i=0; i<16; i++) {
912: if ((microwire.mask >> i) & 1) {
913: microwire.data += ((saveData >> i) & 1) << bit;
914: bit ++;
915: }
916: }
1.1.1.4 root 917:
1.1.1.7 root 918: /* The LMC 1992 address should be 10 xxx xxx xxx */
919: if ((microwire.data & 0x600) != 0x400)
920: return;
921:
922: /* Update the LMC 1992 commands */
923: switch ((microwire.data >> 6) & 0x7) {
924: case 0:
925: /* Mixing command */
1.1.1.8 root 926: LOG_TRACE ( TRACE_DMASND, "Microwire new mixing=0x%x\n", microwire.data & 0x3 );
1.1.1.7 root 927: microwire.mixing = microwire.data & 0x3;
928: break;
929: case 1:
930: /* Bass command */
1.1.1.8 root 931: LOG_TRACE ( TRACE_DMASND, "Microwire new bass=0x%x\n", microwire.data & 0xf );
1.1.1.7 root 932: microwire.bass = microwire.data & 0xf;
933: DmaSnd_Set_Tone_Level(LMC1992_Bass_Treble_Table[microwire.bass],
934: LMC1992_Bass_Treble_Table[microwire.treble]);
935: break;
936: case 2:
937: /* Treble command */
1.1.1.8 root 938: LOG_TRACE ( TRACE_DMASND, "Microwire new trebble=0x%x\n", microwire.data & 0xf );
1.1.1.7 root 939: microwire.treble = microwire.data & 0xf;
940: DmaSnd_Set_Tone_Level(LMC1992_Bass_Treble_Table[microwire.bass],
941: LMC1992_Bass_Treble_Table[microwire.treble]);
942: break;
943: case 3:
944: /* Master volume command */
1.1.1.8 root 945: LOG_TRACE ( TRACE_DMASND, "Microwire new master volume=0x%x\n", microwire.data & 0x3f );
1.1.1.7 root 946: microwire.masterVolume = LMC1992_Master_Volume_Table[microwire.data & 0x3f];
1.1.1.8 root 947: lmc1992.left_gain = (microwire.leftVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
948: lmc1992.right_gain = (microwire.rightVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
1.1.1.7 root 949: break;
950: case 4:
951: /* Right channel volume */
1.1.1.8 root 952: LOG_TRACE ( TRACE_DMASND, "Microwire new right volume=0x%x\n", microwire.data & 0x1f );
1.1.1.7 root 953: microwire.rightVolume = LMC1992_LeftRight_Volume_Table[microwire.data & 0x1f];
1.1.1.8 root 954: lmc1992.right_gain = (microwire.rightVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
1.1.1.7 root 955: break;
956: case 5:
957: /* Left channel volume */
1.1.1.8 root 958: LOG_TRACE ( TRACE_DMASND, "Microwire new left volume=0x%x\n", microwire.data & 0x1f );
1.1.1.7 root 959: microwire.leftVolume = LMC1992_LeftRight_Volume_Table[microwire.data & 0x1f];
1.1.1.8 root 960: lmc1992.left_gain = (microwire.leftVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
1.1.1.7 root 961: break;
962: default:
963: /* Do nothing */
964: break;
965: }
966: }
967: }
1.1.1.4 root 968:
969: /**
970: * Read word from microwire data register (0xff8922).
971: */
972: void DmaSnd_MicrowireData_ReadWord(void)
973: {
974: /* Shifting is done in DmaSnd_InterruptHandler_Microwire! */
1.1.1.6 root 975: LOG_TRACE(TRACE_DMASND, "Microwire data read: 0x%x\n", IoMem_ReadWord(0xff8922));
1.1 root 976: }
977:
978:
1.1.1.3 root 979: /**
980: * Write word to microwire data register (0xff8922).
981: */
1.1 root 982: void DmaSnd_MicrowireData_WriteWord(void)
983: {
1.1.1.4 root 984: /* Only update, if no shift is in progress */
1.1.1.7 root 985: if (!microwire.mwTransferSteps)
1.1.1.4 root 986: {
1.1.1.7 root 987: microwire.data = IoMem_ReadWord(0xff8922);
1.1.1.4 root 988: /* Start shifting events to simulate a microwire transfer */
1.1.1.7 root 989: microwire.mwTransferSteps = 16;
1.1.1.9 ! root 990: microwire.pendingCyclesOver = 8;
! 991: CycInt_AddRelativeInterrupt(microwire.pendingCyclesOver, INT_CPU_CYCLE, INTERRUPT_DMASOUND_MICROWIRE);
1.1.1.4 root 992: }
993:
1.1.1.6 root 994: LOG_TRACE(TRACE_DMASND, "Microwire data write: 0x%x\n", IoMem_ReadWord(0xff8922));
1.1 root 995: }
996:
997:
1.1.1.3 root 998: /**
999: * Read word from microwire mask register (0xff8924).
1000: */
1.1 root 1001: void DmaSnd_MicrowireMask_ReadWord(void)
1002: {
1.1.1.4 root 1003: /* Same as with data register, but mask is rotated, not shifted. */
1.1.1.6 root 1004: LOG_TRACE(TRACE_DMASND, "Microwire mask read: 0x%x\n", IoMem_ReadWord(0xff8924));
1.1 root 1005: }
1006:
1007:
1.1.1.3 root 1008: /**
1009: * Write word to microwire mask register (0xff8924).
1010: */
1.1 root 1011: void DmaSnd_MicrowireMask_WriteWord(void)
1012: {
1.1.1.4 root 1013: /* Only update, if no shift is in progress */
1.1.1.7 root 1014: if (!microwire.mwTransferSteps)
1.1.1.4 root 1015: {
1.1.1.7 root 1016: microwire.mask = IoMem_ReadWord(0xff8924);
1.1.1.4 root 1017: }
1018:
1.1.1.6 root 1019: LOG_TRACE(TRACE_DMASND, "Microwire mask write: 0x%x\n", IoMem_ReadWord(0xff8924));
1020: }
1021:
1022:
1.1.1.7 root 1023: /*-------------------Bass / Treble filter ---------------------------*/
1.1.1.6 root 1024:
1.1.1.7 root 1025: /**
1026: * Left voice Filter for Bass/Treble.
1027: */
1028: static float DmaSnd_IIRfilterL(float xn)
1029: {
1030: static float data[2] = { 0.0, 0.0 };
1031: float a, yn;
1.1.1.6 root 1032:
1.1.1.7 root 1033: /* Input coefficients */
1034: /* biquad1 Note: 'a' coefficients are subtracted */
1.1.1.8 root 1035: a = lmc1992.left_gain * xn; /* a=g*xn; */
1.1.1.7 root 1036: a -= lmc1992.coef[0] * data[0]; /* a1; wn-1 */
1037: a -= lmc1992.coef[1] * data[1]; /* a2; wn-2 */
1038: /* If coefficient scale */
1039: /* factor = 0.5 then */
1040: /* multiply by 2 */
1041: /* Output coefficients */
1042: yn = lmc1992.coef[2] * a; /* b0; */
1043: yn += lmc1992.coef[3] * data[0]; /* b1; */
1044: yn += lmc1992.coef[4] * data[1]; /* b2; */
1.1.1.6 root 1045:
1.1.1.7 root 1046: data[1] = data[0]; /* wn-1 -> wn-2; */
1047: data[0] = a; /* wn -> wn-1 */
1048: return yn;
1.1.1.6 root 1049: }
1050:
1051:
1052: /**
1.1.1.7 root 1053: * Right voice Filter for Bass/Treble.
1.1.1.6 root 1054: */
1.1.1.7 root 1055: static float DmaSnd_IIRfilterR(float xn)
1.1.1.6 root 1056: {
1.1.1.7 root 1057: static float data[2] = { 0.0, 0.0 };
1058: float a, yn;
1059:
1060: /* Input coefficients */
1061: /* biquad1 Note: 'a' coefficients are subtracted */
1.1.1.8 root 1062: a = lmc1992.right_gain * xn; /* a=g*xn; */
1.1.1.7 root 1063: a -= lmc1992.coef[0]*data[0]; /* a1; wn-1 */
1064: a -= lmc1992.coef[1]*data[1]; /* a2; wn-2 */
1065: /* If coefficient scale */
1066: /* factor = 0.5 then */
1067: /* multiply by 2 */
1068: /* Output coefficients */
1069: yn = lmc1992.coef[2]*a; /* b0; */
1070: yn += lmc1992.coef[3]*data[0]; /* b1; */
1071: yn += lmc1992.coef[4]*data[1]; /* b2; */
1072:
1073: data[1] = data[0]; /* wn-1 -> wn-2; */
1074: data[0] = a; /* wn -> wn-1 */
1075: return yn;
1.1.1.6 root 1076: }
1077:
1078: /**
1.1.1.7 root 1079: * LowPass Filter Left
1.1.1.6 root 1080: */
1.1.1.8 root 1081: static Sint16 DmaSnd_LowPassFilterLeft(Sint16 in)
1.1.1.6 root 1082: {
1.1.1.8 root 1083: static Sint16 lowPassFilter[2] = { 0, 0 };
1084: static Sint16 out = 0;
1.1.1.6 root 1085:
1.1.1.8 root 1086: if (DmaSnd_LowPass)
1087: {
1088: out = lowPassFilter[0] + (lowPassFilter[1]<<1) + in;
1089: lowPassFilter[0] = lowPassFilter[1];
1090: lowPassFilter[1] = in;
1.1.1.6 root 1091:
1.1.1.8 root 1092: return out; /* Filter Gain = 4 */
1093: }else
1094: {
1095: return in << 2;
1096: }
1.1.1.6 root 1097: }
1098:
1099: /**
1.1.1.7 root 1100: * LowPass Filter Right
1.1.1.6 root 1101: */
1.1.1.8 root 1102: static Sint16 DmaSnd_LowPassFilterRight(Sint16 in)
1.1.1.6 root 1103: {
1.1.1.8 root 1104: static Sint16 lowPassFilter[2] = { 0, 0 };
1105: static Sint16 out = 0;
1.1.1.6 root 1106:
1.1.1.8 root 1107: if (DmaSnd_LowPass)
1108: {
1109: out = lowPassFilter[0] + (lowPassFilter[1]<<1) + in;
1110: lowPassFilter[0] = lowPassFilter[1];
1111: lowPassFilter[1] = in;
1.1.1.6 root 1112:
1.1.1.8 root 1113: return out; /* Filter Gain = 4 */
1114: }else
1115: {
1116: return in << 2;
1117: }
1.1.1.6 root 1118: }
1119:
1120: /**
1.1.1.7 root 1121: * Set Bass and Treble tone level
1.1.1.6 root 1122: */
1.1.1.7 root 1123: static void DmaSnd_Set_Tone_Level(int set_bass, int set_treb)
1124: {
1125: /* 13 levels; 0 through 12 correspond with -12dB to 12dB in 2dB steps */
1126: lmc1992.coef[0] = lmc1992.treb_table[set_treb].a1 + lmc1992.bass_table[set_bass].a1;
1127: lmc1992.coef[1] = lmc1992.treb_table[set_treb].a1 * lmc1992.bass_table[set_bass].a1;
1128: lmc1992.coef[2] = lmc1992.treb_table[set_treb].b0 * lmc1992.bass_table[set_bass].b0;
1129: lmc1992.coef[3] = lmc1992.treb_table[set_treb].b0 * lmc1992.bass_table[set_bass].b1 +
1130: lmc1992.treb_table[set_treb].b1 * lmc1992.bass_table[set_bass].b0;
1131: lmc1992.coef[4] = lmc1992.treb_table[set_treb].b1 * lmc1992.bass_table[set_bass].b1;
1.1.1.6 root 1132: }
1133:
1134:
1135: /**
1.1.1.7 root 1136: * Compute the first order bass shelf
1.1.1.6 root 1137: */
1.1.1.7 root 1138: static struct first_order_s *DmaSnd_Bass_Shelf(float g, float fc, float Fs)
1.1.1.6 root 1139: {
1.1.1.7 root 1140: static struct first_order_s bass;
1141: float a1;
1.1.1.6 root 1142:
1.1.1.7 root 1143: /* g, fc, Fs must be positve real numbers > 0.0 */
1144: if (g < 1.0)
1145: bass.a1 = a1 = (tanf(M_PI*fc/Fs) - g ) / (tanf(M_PI*fc/Fs) + g );
1146: else
1147: bass.a1 = a1 = (tanf(M_PI*fc/Fs) - 1.0) / (tanf(M_PI*fc/Fs) + 1.0);
1.1.1.6 root 1148:
1.1.1.7 root 1149: bass.b0 = (1.0 + a1) * (g - 1.0) / 2.0 + 1.0;
1150: bass.b1 = (1.0 + a1) * (g - 1.0) / 2.0 + a1;
1.1.1.6 root 1151:
1.1.1.7 root 1152: return &bass;
1.1.1.6 root 1153: }
1154:
1155:
1156: /**
1.1.1.7 root 1157: * Compute the first order treble shelf
1.1.1.6 root 1158: */
1.1.1.7 root 1159: static struct first_order_s *DmaSnd_Treble_Shelf(float g, float fc, float Fs)
1.1.1.6 root 1160: {
1.1.1.7 root 1161: static struct first_order_s treb;
1162: float a1;
1.1.1.6 root 1163:
1.1.1.7 root 1164: /* g, fc, Fs must be positve real numbers > 0.0 */
1165: if (g < 1.0)
1166: treb.a1 = a1 = (g*tanf(M_PI*fc/Fs) - 1.0) / (g*tanf(M_PI*fc/Fs) + 1.0);
1167: else
1168: treb.a1 = a1 = (tanf(M_PI*fc/Fs) - 1.0) / (tanf(M_PI*fc/Fs) + 1.0);
1.1.1.6 root 1169:
1.1.1.7 root 1170: treb.b0 = 1.0 + (1.0 - a1) * (g - 1.0) / 2.0;
1171: treb.b1 = a1 + (a1 - 1.0) * (g - 1.0) / 2.0;
1172:
1173: return &treb;
1.1.1.6 root 1174: }
1175:
1.1.1.7 root 1176:
1.1.1.6 root 1177: /**
1.1.1.7 root 1178: * Compute the bass and treble tables (nAudioFrequency)
1.1.1.6 root 1179: */
1.1.1.7 root 1180: void DmaSnd_Init_Bass_and_Treble_Tables(void)
1.1.1.6 root 1181: {
1.1.1.7 root 1182: struct first_order_s *bass;
1183: struct first_order_s *treb;
1184:
1.1.1.8 root 1185: float dB_adjusted, dB, g, fc_bt, fc_tt, Fs;
1.1.1.7 root 1186: int n;
1187:
1188: fc_bt = 118.2763;
1189: fc_tt = 8438.756;
1190: Fs = (float)nAudioFrequency;
1191:
1192: if ((Fs < 8000.0) || (Fs > 96000.0))
1193: Fs = 44100.0;
1194:
1.1.1.8 root 1195: if (fc_tt > 0.5*0.8*Fs)
1196: {
1197: fc_tt = 0.5*0.8*Fs;
1198: dB_adjusted = 2.0 * 0.5*0.8*Fs/fc_tt;
1199: }else
1200: {
1201: dB_adjusted = 2.0;
1202: }
1203:
1204: for (dB = dB_adjusted*(TONE_STEPS-1)/2, n = TONE_STEPS; n--; dB -= dB_adjusted)
1205: {
1206: g = powf(10.0, dB/20.0); /* 12dB to -12dB */
1207:
1208: treb = DmaSnd_Treble_Shelf(g, fc_tt, Fs);
1209:
1210: lmc1992.treb_table[n].a1 = treb->a1;
1211: lmc1992.treb_table[n].b0 = treb->b0;
1212: lmc1992.treb_table[n].b1 = treb->b1;
1213: }
1214:
1.1.1.7 root 1215: for (dB = 12.0, n = TONE_STEPS; n--; dB -= 2.0)
1216: {
1217: g = powf(10.0, dB/20.0); /* 12dB to -12dB */
1218:
1219: bass = DmaSnd_Bass_Shelf(g, fc_bt, Fs);
1220:
1221: lmc1992.bass_table[n].a1 = bass->a1;
1222: lmc1992.bass_table[n].b0 = bass->b0;
1223: lmc1992.bass_table[n].b1 = bass->b1;
1224: }
1225:
1226: DmaSnd_Set_Tone_Level(LMC1992_Bass_Treble_Table[microwire.bass & 0xf],
1227: LMC1992_Bass_Treble_Table[microwire.treble & 0xf]);
1.1.1.8 root 1228:
1229: /* Initialize IIR Filter Gain and use as a Volume Control */
1230: lmc1992.left_gain = (microwire.leftVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
1231: lmc1992.right_gain = (microwire.rightVolume * (Uint32)microwire.masterVolume) * (1.0/(65536.0*65536.0));
1232:
1233: /* Anti-alias filter is not required when nAudioFrequency == 50066 Hz */
1234: if (nAudioFrequency>50000 && nAudioFrequency<50100)
1235: DmaSnd_LowPass = false;
1236: else
1237: DmaSnd_LowPass = true;
1.1 root 1238: }
This archive runs on limited infrastructure. Preserving old code on modern bandwidth. Automated agents are requested to crawl responsibly.