/* ScummVM - Graphic Adventure Engine * * ScummVM is the legal property of its developers, whose names * are too numerous to list here. Please refer to the COPYRIGHT * file distributed with this source distribution. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. * * $URL$ * $Id$ * */ #include "common/stdafx.h" #include "sound/mixer.h" #include "agi/agi.h" #include "agi/sound.h" namespace Agi { #define USE_INTERPOLATION #define USE_CHORUS /* TODO: add support for variable sampling rate in the output device */ #ifdef USE_IIGS_SOUND struct IIgsEnvelopeSegment { uint8 bp; uint16 inc; ///< 8b.8b fixed point, big endian? }; #define ENVELOPE_SEGMENT_COUNT 8 struct IIgsEnvelope { IIgsEnvelopeSegment seg[ENVELOPE_SEGMENT_COUNT]; }; // 2**(1/12) i.e. the 12th root of 2 #define SEMITONE 1.059463094359295 struct IIgsWaveInfo { uint8 top; uint8 addr; uint8 size; // Oscillator channel (Bits 4-7 of mode-byte). Simplified to use only stereo here. #define MASK_OSC_CHANNEL (1 << 4) #define OSC_CHANNEL_LEFT (1 << 4) #define OSC_CHANNEL_RIGHT (0 << 4) // Oscillator halt bit (Bit 0 of mode-byte) #define MASK_OSC_HALT (1 << 0) #define OSC_HALT (1 << 0) // Oscillator mode (Bits 1 and 2 of mode-byte) #define MASK_OSC_MODE (3 << 1) #define OSC_MODE_LOOP (0 << 1) #define OSC_MODE_ONESHOT (1 << 1) #define OSC_MODE_SYNC_AM (2 << 1) #define OSC_MODE_SWAP (3 << 1) uint8 mode; uint16 relPitch; ///< 8b.8b fixed point, big endian? }; #define MAX_WAVE_COUNT 8 struct IIgsInstrumentHeader { IIgsEnvelope env; uint8 relseg; uint8 priority; uint8 bendrange; uint8 vibdepth; uint8 vibspeed; uint8 spare; uint8 wac; uint8 wbc; IIgsWaveInfo wal[MAX_WAVE_COUNT]; IIgsWaveInfo wbl[MAX_WAVE_COUNT]; }; struct IIgsSampleHeader { uint16 type; uint8 pitch; ///< Logarithmic, base is 2**(1/12), unknown multiplier (Possibly in range 1040-1080) uint8 unknownByte_Ofs3; // 0x7F in Gold Rush's sound resource 60, 0 in all others. uint8 volume; ///< Current guess: Logarithmic in 6 dB steps uint8 unknownByte_Ofs5; ///< 0 in all tested samples. uint16 instrumentSize; ///< Little endian. 44 in all tested samples. A guess. uint16 sampleSize; ///< Little endian. Accurate in all tested samples excluding Manhunter I's sound resource 16. IIgsInstrumentHeader instrument; }; #if 0 static SoundInstrument *instruments; static int numInstruments; static uint8 *wave; #endif bool readIIgsEnvelope(IIgsEnvelope &envelope, Common::SeekableReadStream &stream) { for (int segNum = 0; segNum < ENVELOPE_SEGMENT_COUNT; segNum++) { envelope.seg[segNum].bp = stream.readByte(); envelope.seg[segNum].inc = stream.readUint16BE(); } return !stream.ioFailed(); } bool readIIgsWaveInfo(IIgsWaveInfo &waveInfo, Common::SeekableReadStream &stream) { waveInfo.top = stream.readByte(); waveInfo.addr = stream.readByte(); waveInfo.size = stream.readByte(); waveInfo.mode = stream.readByte(); waveInfo.relPitch = stream.readUint16BE(); return !stream.ioFailed(); } /** * Read an Apple IIGS instrument header from the given stream. * @param header The header to which to write the data. * @param stream The source stream from which to read the data. * @return True if successful, false otherwise. */ bool readIIgsInstrumentHeader(IIgsInstrumentHeader &header, Common::SeekableReadStream &stream) { readIIgsEnvelope(header.env, stream); header.relseg = stream.readByte(); header.priority = stream.readByte(); header.bendrange = stream.readByte(); header.vibdepth = stream.readByte(); header.vibspeed = stream.readByte(); header.spare = stream.readByte(); header.wac = stream.readByte(); header.wbc = stream.readByte(); for (int waveA = 0; waveA < header.wac; waveA++) // Read A wave lists readIIgsWaveInfo(header.wal[waveA], stream); for (int waveB = 0; waveB < header.wbc; waveB++) // Read B wave lists readIIgsWaveInfo(header.wbl[waveB], stream); return !stream.ioFailed(); } /** * Read an Apple IIGS AGI sample header from the given stream. * @param header The header to which to write the data. * @param stream The source stream from which to read the data. * @return True if successful, false otherwise. */ bool readIIgsSampleHeader(IIgsSampleHeader &header, Common::SeekableReadStream &stream) { header.type = stream.readUint16LE(); header.pitch = stream.readByte(); header.unknownByte_Ofs3 = stream.readByte(); header.volume = stream.readByte(); header.unknownByte_Ofs5 = stream.readByte(); header.instrumentSize = stream.readUint16LE(); header.sampleSize = stream.readUint16LE(); return readIIgsInstrumentHeader(header.instrument, stream); } /** * Load an Apple IIGS AGI sample resource from the given stream and * create an AudioStream out of it. * * @param stream The source stream. * @param resnum Sound resource number. Optional. Used for error messages. * @return A non-null AudioStream pointer if successful, NULL otherwise. * @note In case of failure (i.e. NULL is returned), stream is reset back * to its original position and its I/O failed -status is cleared. * TODO: Add better handling of invalid resource number when printing error messages. * TODO: Add support for looping sounds. * FIXME: Fix sample rate calculation, it's probably not accurate at the moment. */ Audio::AudioStream *makeIIgsSampleStream(Common::SeekableReadStream &stream, int resnum = -1) { const uint32 startPos = stream.pos(); IIgsSampleHeader header; Audio::AudioStream *result = NULL; bool readHeaderOk = readIIgsSampleHeader(header, stream); // Check that the header was read ok and that it's of the correct type // and that there's room for the sample data in the stream. if (readHeaderOk && header.type == AGI_SOUND_SAMPLE) { // An Apple IIGS AGI sample resource uint32 tailLen = stream.size() - stream.pos(); if (tailLen < header.sampleSize) { // Check if there's no room for the sample data in the stream // Apple IIGS Manhunter I: Sound resource 16 has only 16074 bytes // of sample data although header says it should have 16384 bytes. warning("Apple IIGS sample (%d) too short (%d bytes. Should be %d bytes). Using the part that's left", resnum, tailLen, header.sampleSize); header.sampleSize = (uint16) tailLen; // Use the part that's left } if (header.pitch > 0x7F) { // Check if the pitch is invalid warning("Apple IIGS sample (%d) has too high pitch (0x%02x)", resnum, header.pitch); header.pitch &= 0x7F; // Apple IIGS AGI probably did it this way too } // Allocate memory for the sample data and read it in byte *sampleData = (byte *) malloc(header.sampleSize); uint32 readBytes = stream.read(sampleData, header.sampleSize); if (readBytes == header.sampleSize) { // Check that we got all the data we requested // Make an audio stream from the mono, 8 bit, unsigned input data byte flags = Audio::Mixer::FLAG_AUTOFREE | Audio::Mixer::FLAG_UNSIGNED; int rate = (int) (1076 * pow(SEMITONE, header.pitch)); result = Audio::makeLinearInputStream(sampleData, header.sampleSize, rate, flags, 0, 0); } } // If couldn't make a sample out of the input stream for any reason then // rewind back to stream's starting position and clear I/O failed -status. if (result == NULL) { stream.seek(startPos); stream.clearIOFailed(); } return result; } #endif static int playing; static ChannelInfo chn[NUM_CHANNELS]; static int endflag = -1; static int playingSound = -1; static uint8 *song; static uint8 env; static int16 *sndBuffer; static int16 *waveform; static int16 waveformRamp[WAVEFORM_SIZE] = { 0, 8, 16, 24, 32, 40, 48, 56, 64, 72, 80, 88, 96, 104, 112, 120, 128, 136, 144, 152, 160, 168, 176, 184, 192, 200, 208, 216, 224, 232, 240, 255, 0, -248, -240, -232, -224, -216, -208, -200, -192, -184, -176, -168, -160, -152, -144, -136, -128, -120, -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8 /* Ramp up */ }; static int16 waveformSquare[WAVEFORM_SIZE] = { 255, 230, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 220, 110, -255, -230, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -220, -110, 0, 0, 0, 0 /* Square */ }; static int16 waveformMac[WAVEFORM_SIZE] = { 45, 110, 135, 161, 167, 173, 175, 176, 156, 137, 123, 110, 91, 72, 35, -2, -60, -118, -142, -165, -170, -176, -177, -179, -177, -176, -164, -152, -117, -82, -17, 47, 92, 137, 151, 166, 170, 173, 171, 169, 151, 133, 116, 100, 72, 43, -7, -57, -99, -141, -156, -170, -174, -177, -178, -179, -175, -172, -165, -159, -137, -114, -67, -19 }; #ifdef USE_IIGS_SOUND static uint16 period[] = { 1024, 1085, 1149, 1218, 1290, 1367, 1448, 1534, 1625, 1722, 1825, 1933 }; static struct AgiNote playSample[] = { {0xff, 0x7f, 0x18, 0x00, 0x7f}, {0xff, 0xff, 0x00, 0x00, 0x00}, {0xff, 0xff, 0x00, 0x00, 0x00}, {0xff, 0xff, 0x00, 0x00, 0x00} }; static int noteToPeriod(int note) { return 10 * (period[note % 12] >> (note / 12 - 3)); } #endif /* USE_IIGS_SOUND */ void SoundMgr::unloadSound(int resnum) { if (_vm->_game.dirSound[resnum].flags & RES_LOADED) { if (_vm->_game.sounds[resnum].flags & SOUND_PLAYING) /* FIXME: Stop playing */ ; /* Release RAW data for sound */ free(_vm->_game.sounds[resnum].rdata); _vm->_game.sounds[resnum].rdata = NULL; _vm->_game.dirSound[resnum].flags &= ~RES_LOADED; } } void SoundMgr::decodeSound(int resnum) { #if 0 int type, size; int16 *buf; uint8 *src; struct SoundIIgsSample *smp; debugC(3, kDebugLevelSound, "(%d)", resnum); type = READ_LE_UINT16(_vm->_game.sounds[resnum].rdata); if (type == AGI_SOUND_SAMPLE) { /* Convert sample data to 16 bit signed format */ smp = (struct SoundIIgsSample *)_vm->_game.sounds[resnum].rdata; size = ((int)smp->sizeHi << 8) + smp->sizeLo; src = (uint8 *)_vm->_game.sounds[resnum].rdata; buf = (int16 *)calloc(1, 54 + (size << 1) + 100); /* FIXME */ memcpy(buf, src, 54); for (; size--; buf[size + 54] = ((int16)src[size + 54] - 0x80) << 4); /* FIXME */ _vm->_game.sounds[resnum].rdata = (uint8 *) buf; free(src); } #endif } void SoundMgr::startSound(int resnum, int flag) { int i, type; #if 0 struct SoundIIgsSample *smp; #endif if (_vm->_game.sounds[resnum].flags & SOUND_PLAYING) return; stopSound(); if (_vm->_game.sounds[resnum].rdata == NULL) return; type = READ_LE_UINT16(_vm->_game.sounds[resnum].rdata); if (type != AGI_SOUND_SAMPLE && type != AGI_SOUND_MIDI && type != AGI_SOUND_4CHN) return; _vm->_game.sounds[resnum].flags |= SOUND_PLAYING; _vm->_game.sounds[resnum].type = type; playingSound = resnum; song = (uint8 *)_vm->_game.sounds[resnum].rdata; switch (type) { #if 0 case AGI_SOUND_SAMPLE: debugC(3, kDebugLevelSound, "IIGS sample"); smp = (struct SoundIIgsSample *)_vm->_game.sounds[resnum].rdata; for (i = 0; i < NUM_CHANNELS; i++) { chn[i].type = type; chn[i].flags = 0; chn[i].ins = (int16 *)&_vm->_game.sounds[resnum].rdata[54]; chn[i].size = ((int)smp->sizeHi << 8) + smp->sizeLo; chn[i].ptr = &playSample[i]; chn[i].timer = 0; chn[i].vol = 0; chn[i].end = 0; } break; case AGI_SOUND_MIDI: debugC(3, kDebugLevelSound, "IIGS MIDI sequence"); for (i = 0; i < NUM_CHANNELS; i++) { chn[i].type = type; chn[i].flags = AGI_SOUND_LOOP | AGI_SOUND_ENVELOPE; chn[i].ins = waveform; chn[i].size = WAVEFORM_SIZE; chn[i].vol = 0; chn[i].end = 0; } chn[0].timer = *(song + 2); chn[0].ptr = (struct AgiNote *)(song + 3); break; #endif case AGI_SOUND_4CHN: /* Initialize channel info */ for (i = 0; i < NUM_CHANNELS; i++) { chn[i].type = type; chn[i].flags = AGI_SOUND_LOOP; if (env) { chn[i].flags |= AGI_SOUND_ENVELOPE; chn[i].adsr = AGI_SOUND_ENV_ATTACK; } chn[i].ins = waveform; chn[i].size = WAVEFORM_SIZE; chn[i].ptr = (struct AgiNote *)(song + (song[i << 1] | (song[(i << 1) + 1] << 8))); chn[i].timer = 0; chn[i].vol = 0; chn[i].end = 0; } break; } memset(sndBuffer, 0, BUFFER_SIZE << 1); endflag = flag; /* Nat Budin reports that the flag should be reset when sound starts */ _vm->setflag(endflag, false); } void SoundMgr::stopSound() { int i; endflag = -1; for (i = 0; i < NUM_CHANNELS; i++) stopNote(i); if (playingSound != -1) { _vm->_game.sounds[playingSound].flags &= ~SOUND_PLAYING; playingSound = -1; } } static int16 *buffer; int SoundMgr::initSound() { int r = -1; buffer = sndBuffer = (int16 *)calloc(2, BUFFER_SIZE); env = false; switch (_vm->_soundemu) { case SOUND_EMU_NONE: waveform = waveformRamp; env = true; break; case SOUND_EMU_AMIGA: case SOUND_EMU_PC: waveform = waveformSquare; break; case SOUND_EMU_MAC: waveform = waveformMac; break; } report("Initializing sound:\n"); report("sound: envelopes "); if (env) { report("enabled (decay=%d, sustain=%d)\n", ENV_DECAY, ENV_SUSTAIN); } else { report("disabled\n"); } #ifdef USE_IIGS_SOUND /*loadInstruments("demo.sys"); */ #endif _mixer->playInputStream(Audio::Mixer::kPlainSoundType, &_soundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, false, true); return r; } void SoundMgr::deinitSound() { debugC(3, kDebugLevelSound, "()"); _mixer->stopHandle(_soundHandle); free(sndBuffer); } void SoundMgr::stopNote(int i) { chn[i].adsr = AGI_SOUND_ENV_RELEASE; #ifdef USE_CHORUS /* Stop chorus ;) */ if (chn[i].type == AGI_SOUND_4CHN && _vm->_soundemu == SOUND_EMU_NONE && i < 3) { stopNote(i + 4); } #endif } void SoundMgr::playNote(int i, int freq, int vol) { if (!_vm->getflag(fSoundOn)) vol = 0; else if (vol && _vm->_soundemu == SOUND_EMU_PC) vol = 160; chn[i].phase = 0; chn[i].freq = freq; chn[i].vol = vol; chn[i].env = 0x10000; chn[i].adsr = AGI_SOUND_ENV_ATTACK; #ifdef USE_CHORUS /* Add chorus ;) */ if (chn[i].type == AGI_SOUND_4CHN && _vm->_soundemu == SOUND_EMU_NONE && i < 3) { int newfreq = freq * 1007 / 1000; if (freq == newfreq) newfreq++; playNote(i + 4, newfreq, vol * 2 / 3); } #endif } #ifdef USE_IIGS_SOUND void SoundMgr::playMidiSound() { uint8 *p; uint8 parm1, parm2; static uint8 cmd, ch; playing = 1; if (chn[0].timer > 0) { chn[0].timer -= 2; return; } p = (uint8 *)chn[0].ptr; if (*p & 0x80) { cmd = *p++; ch = cmd & 0x0f; cmd >>= 4; } switch (cmd) { case 0x08: parm1 = *p++; parm2 = *p++; if (ch < NUM_CHANNELS) stopNote(ch); break; case 0x09: parm1 = *p++; parm2 = *p++; if (ch < NUM_CHANNELS) playNote(ch, noteToPeriod(parm1), 127); break; case 0x0b: parm1 = *p++; parm2 = *p++; debugC(3, kDebugLevelSound, "controller %02x, ch %02x, val %02x", parm1, ch, parm2); break; case 0x0c: parm1 = *p++; #if 0 if (ch < NUM_CHANNELS) { chn[ch].ins = (uint16 *)&wave[waveaddr[parm1]]; chn[ch].size = wavesize[parm1]; } debugC(3, kDebugLevelSound, "set patch %02x (%d,%d), ch %02x", parm1, waveaddr[parm1], wavesize[parm1], ch); #endif break; } chn[0].timer = *p++; chn[0].ptr = (struct AgiNote *)p; if (*p >= 0xfc) { debugC(3, kDebugLevelSound, "end of sequence"); playing = 0; return; } } void SoundMgr::playSampleSound() { playNote(0, 11025 * 10, 200); playing = 1; } #endif /* USE_IIGS_SOUND */ void SoundMgr::playAgiSound() { int i, freq; for (playing = i = 0; i < (_vm->_soundemu == SOUND_EMU_PC ? 1 : 4); i++) { playing |= !chn[i].end; if (chn[i].end) continue; if ((--chn[i].timer) <= 0) { stopNote(i); freq = ((chn[i].ptr->frq0 & 0x3f) << 4) | (int)(chn[i].ptr->frq1 & 0x0f); if (freq) { uint8 v = chn[i].ptr->vol & 0x0f; playNote(i, freq * 10, v == 0xf ? 0 : 0xff - (v << 1)); } chn[i].timer = ((int)chn[i].ptr->durHi << 8) | chn[i].ptr->durLo; if (chn[i].timer == 0xffff) { chn[i].end = 1; chn[i].vol = 0; chn[i].env = 0; #ifdef USE_CHORUS /* chorus */ if (chn[i].type == AGI_SOUND_4CHN && _vm->_soundemu == SOUND_EMU_NONE && i < 3) { chn[i + 4].vol = 0; chn[i + 4].env = 0; } #endif } chn[i].ptr++; } } } void SoundMgr::playSound() { int i; if (endflag == -1) return; #ifdef USE_IIGS_SOUND if (chn[0].type == AGI_SOUND_MIDI) { /* play_midi_sound (); */ playing = 0; } else if (chn[0].type == AGI_SOUND_SAMPLE) { playSampleSound(); } else #endif playAgiSound(); if (!playing) { for (i = 0; i < NUM_CHANNELS; chn[i++].vol = 0); if (endflag != -1) _vm->setflag(endflag, true); if (playingSound != -1) _vm->_game.sounds[playingSound].flags &= ~SOUND_PLAYING; playingSound = -1; endflag = -1; } } uint32 SoundMgr::mixSound(void) { register int i, p; int16 *src; int c, b, m; memset(sndBuffer, 0, BUFFER_SIZE << 1); for (c = 0; c < NUM_CHANNELS; c++) { if (!chn[c].vol) continue; m = chn[c].flags & AGI_SOUND_ENVELOPE ? chn[c].vol * chn[c].env >> 16 : chn[c].vol; if (chn[c].type != AGI_SOUND_4CHN || c != 3) { src = chn[c].ins; p = chn[c].phase; for (i = 0; i < BUFFER_SIZE; i++) { b = src[p >> 8]; #ifdef USE_INTERPOLATION b += ((src[((p >> 8) + 1) % chn[c].size] - src[p >> 8]) * (p & 0xff)) >> 8; #endif sndBuffer[i] += (b * m) >> 4; p += (uint32) 118600 *4 / chn[c].freq; /* FIXME */ if (chn[c].flags & AGI_SOUND_LOOP) { p %= chn[c].size << 8; } else { if (p >= chn[c].size << 8) { p = chn[c].vol = 0; chn[c].end = 1; break; } } } chn[c].phase = p; } else { /* Add white noise */ for (i = 0; i < BUFFER_SIZE; i++) { b = _vm->_rnd->getRandomNumber(255) - 128; sndBuffer[i] += (b * m) >> 4; } } switch (chn[c].adsr) { case AGI_SOUND_ENV_ATTACK: /* not implemented */ chn[c].adsr = AGI_SOUND_ENV_DECAY; break; case AGI_SOUND_ENV_DECAY: if (chn[c].env > chn[c].vol * ENV_SUSTAIN + ENV_DECAY) { chn[c].env -= ENV_DECAY; } else { chn[c].env = chn[c].vol * ENV_SUSTAIN; chn[c].adsr = AGI_SOUND_ENV_SUSTAIN; } break; case AGI_SOUND_ENV_SUSTAIN: break; case AGI_SOUND_ENV_RELEASE: if (chn[c].env >= ENV_RELEASE) { chn[c].env -= ENV_RELEASE; } else { chn[c].env = 0; } } } return BUFFER_SIZE; } #ifdef USE_IIGS_SOUND #if 0 int SoundMgr::loadInstruments(char *fname) { Common::File fp; int i, j, k; struct SoundInstrument ai; int numWav; char *path; path = "sierrast"; if (!fp.open(path)) return errBadFileOpen; report("Loading samples: %s\n", path); if ((wave = malloc(0x10000 * 2)) == NULL) return errNotEnoughMemory; fp.read(wave, 0x10000); fp.close(); for (i = 0x10000; i--;) { ((int16 *)wave)[i] = 2 * ((int16)wave[i] - 128); } fp = fopen("bla", "w"); fwrite(wave, 2, 0x10000, fp); fclose(fp); report("Loading instruments: %s\n", path); if ((fp = fopen(path, "rb")) == NULL) return errBadFileOpen; fseek(fp, 0x8469, SEEK_SET); for (numWav = j = 0; j < 40; j++) { fread(&ai, 1, 32, fp); if (ai.env[0].bp > 0x7f) break; #if 0 printf("Instrument %d loaded ----------------\n", j); printf("Envelope:\n"); for (i = 0; i < 8; i++) printf("[seg %d]: BP %02x Inc %04x\n", i, ai.env[i].bp, ((int)ai.env[i].inc_hi << 8) | ai.env[i].inc_lo); printf("rel seg: %d, pri inc: %d, bend range: %d, vib dep: %d, " "vib spd: %d\n", ai.relseg, ai.priority, ai.bendrange, ai.vibdepth, ai.vibspeed); printf("A wave count: %d, B wave count: %d\n", ai.wac, ai.wbc); #endif for (k = 0; k < ai.wac; k++, num_wav++) { fread(&ai.wal[k], 1, 6, fp); #if 0 printf("[A %d of %d] top: %02x, wave address: %02x, " "size: %02x, mode: %02x, relPitch: %04x\n", k + 1, ai.wac, ai.wal[k].top, ai.wal[k].addr, ai.wal[k].size, ai.wal[k].mode, ((int)ai.wal[k].rel_hi << 8) | ai.wal[k].rel_lo); #endif } for (k = 0; k < ai.wbc; k++, num_wav++) { fread(&ai.wbl[k], 1, 6, fp); #if 0 printf("[B %d of %d] top: %02x, wave address: %02x, " "size: %02x, mode: %02x, relPitch: %04x\n", k + 1, ai.wbc, ai.wbl[k].top, ai.wbl[k].addr, ai.wbl[k].size, ai.wbl[k].mode, ((int)ai.wbl[k].rel_hi << 8) | ai.wbl[k].rel_lo); #endif } waveaddr[j] = 256 * ai.wal[0].addr; wavesize[j] = 256 * (1 << ((ai.wal[0].size) & 0x07)); #if 1 printf("%d addr = %d\n", j, waveaddr[j]); printf(" size = %d\n", wavesize[j]); #endif } numInstruments = j; printf("%d Ensoniq 5503 instruments loaded. (%d waveforms)\n", num_instruments, num_wav); fclose(fp); return errOK; } void Sound::unloadInstruments() { free(instruments); } #endif #endif /* USE_IIGS_SOUND */ static void fillAudio(void *udata, int16 *stream, uint len) { SoundMgr *soundMgr = (SoundMgr *)udata; uint32 p = 0; static uint32 n = 0, s = 0; len <<= 2; debugC(5, kDebugLevelSound, "(%p, %p, %d)", (void *)udata, (void *)stream, len); memcpy(stream, (uint8 *)buffer + s, p = n); for (n = 0, len -= p; n < len; p += n, len -= n) { soundMgr->playSound(); n = soundMgr->mixSound() << 1; if (len < n) { memcpy((uint8 *)stream + p, buffer, len); s = len; n -= s; return; } else { memcpy((uint8 *)stream + p, buffer, n); } } soundMgr->playSound(); n = soundMgr->mixSound() << 1; memcpy((uint8 *)stream + p, buffer, s = len); n -= s; } SoundMgr::SoundMgr(AgiEngine *agi, Audio::Mixer *pMixer) { _vm = agi; _mixer = pMixer; _sampleRate = pMixer->getOutputRate(); } void SoundMgr::premixerCall(int16 *data, uint len) { fillAudio(this, data, len); } void SoundMgr::setVolume(uint8 volume) { // TODO } SoundMgr::~SoundMgr() { } } // End of namespace Agi