And change all of that **** of tabs into space here.
This commit is contained in:
Godzil
parent
1f8f6330e5
commit
8b19769069
@ -97,9 +97,9 @@ VOID cdecl MixAudioData16SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice);
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static VOID QuantAudioData(LPVOID lpBuffer, LPLONG lpData, UINT nCount)
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{
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if (Synth.wFormat & AUDIO_FORMAT_16BITS)
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QuantAudioData16(lpBuffer, lpData, nCount);
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QuantAudioData16(lpBuffer, lpData, nCount);
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else
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QuantAudioData08(lpBuffer, lpData, nCount);
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QuantAudioData08(lpBuffer, lpData, nCount);
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}
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#else
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@ -114,26 +114,26 @@ static VOID QuantAudioData(LPVOID lpBuffer, LPLONG lpData, UINT nCount)
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LONG dwSample;
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if (Synth.wFormat & AUDIO_FORMAT_16BITS) {
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lpwBuffer = (LPSHORT) lpBuffer;
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while (nCount-- > 0) {
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dwSample = *lpData++;
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if (dwSample < -32768)
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dwSample = -32768;
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else if (dwSample > +32767)
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dwSample = +32767;
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*lpwBuffer++ = (SHORT) dwSample;
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}
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lpwBuffer = (LPSHORT) lpBuffer;
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while (nCount-- > 0) {
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dwSample = *lpData++;
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if (dwSample < -32768)
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dwSample = -32768;
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else if (dwSample > +32767)
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dwSample = +32767;
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*lpwBuffer++ = (SHORT) dwSample;
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}
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}
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else {
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lpbBuffer = (LPBYTE) lpBuffer;
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while (nCount-- > 0) {
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dwSample = *lpData++;
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if (dwSample < -32768)
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dwSample = -32768;
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else if (dwSample > +32767)
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dwSample = +32767;
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*lpbBuffer++ = (BYTE) ((dwSample >> 8) + 128);
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}
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lpbBuffer = (LPBYTE) lpBuffer;
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while (nCount-- > 0) {
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dwSample = *lpData++;
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if (dwSample < -32768)
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dwSample = -32768;
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else if (dwSample > +32767)
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dwSample = +32767;
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*lpbBuffer++ = (BYTE) ((dwSample >> 8) + 128);
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}
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}
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}
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@ -153,8 +153,8 @@ MixAudioData08M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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count = nCount;
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do {
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*buf++ += table[data[accum >> ACCURACY]];
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accum += delta;
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*buf++ += table[data[accum >> ACCURACY]];
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accum += delta;
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} while (--count != 0);
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lpVoice->dwAccum = accum;
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@ -179,11 +179,11 @@ MixAudioData08S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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buf = lpBuffer;
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do {
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a = data[accum >> ACCURACY];
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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a = data[accum >> ACCURACY];
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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} while (--nCount != 0);
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lpVoice->dwAccum = accum;
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@ -199,9 +199,9 @@ MixAudioData08MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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/* adaptive oversampling interpolation comparison */
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if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData08M(lpBuffer, nCount, lpVoice);
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return;
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData08M(lpBuffer, nCount, lpVoice);
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return;
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}
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accum = lpVoice->dwAccum;
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@ -215,20 +215,20 @@ MixAudioData08MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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frac = ((long) delta < 0 ? -delta : +delta) >> (ACCURACY - 5);
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ftable = lpFilterTable + (frac << 8);
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do {
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a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
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*buf++ += table[a];
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accum += delta;
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a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
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*buf++ += table[a];
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accum += delta;
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} while (--nCount != 0);
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lpVoice->bReserved = (BYTE) a;
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#else
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do {
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register INT b;
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a = (signed char) data[accum >> ACCURACY];
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b = (signed char) data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (BYTE)(a + ((frac * (b - a)) >> 5));
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*buf++ += table[a];
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accum += delta;
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register INT b;
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a = (signed char) data[accum >> ACCURACY];
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b = (signed char) data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (BYTE)(a + ((frac * (b - a)) >> 5));
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*buf++ += table[a];
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accum += delta;
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} while (--nCount != 0);
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#endif
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@ -245,9 +245,9 @@ MixAudioData08SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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/* adaptive oversampling interpolation comparison */
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if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData08S(lpBuffer, nCount, lpVoice);
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return;
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData08S(lpBuffer, nCount, lpVoice);
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return;
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}
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accum = lpVoice->dwAccum;
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@ -263,24 +263,24 @@ MixAudioData08SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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frac = ((long) delta < 0 ? -delta : +delta) >> (ACCURACY - 5);
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ftable = lpFilterTable + (frac << 8);
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do {
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a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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a = (BYTE)(a + ftable[data[accum >> ACCURACY]] - ftable[a]);
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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} while (--nCount != 0);
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lpVoice->bReserved = (BYTE) a;
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#else
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do {
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register INT b;
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a = (signed char) data[accum >> ACCURACY];
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b = (signed char) data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (BYTE)(a + ((frac * (b - a)) >> 5));
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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register INT b;
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a = (signed char) data[accum >> ACCURACY];
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b = (signed char) data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (BYTE)(a + ((frac * (b - a)) >> 5));
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buf[0] += ltable[a];
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buf[1] += rtable[a];
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accum += delta;
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buf += 2;
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} while (--nCount != 0);
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#endif
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@ -303,9 +303,9 @@ MixAudioData16M(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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count = nCount;
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do {
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a = data[accum >> ACCURACY];
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*buf++ += table[a >> 8];
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accum += delta;
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a = data[accum >> ACCURACY];
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*buf++ += table[a >> 8];
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accum += delta;
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} while (--count != 0);
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lpVoice->dwAccum = accum;
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@ -328,11 +328,11 @@ MixAudioData16S(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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buf = lpBuffer;
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do {
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a = data[accum >> ACCURACY];
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buf[0] += ltable[a >> 8];
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buf[1] += rtable[a >> 8];
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accum += delta;
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buf += 2;
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a = data[accum >> ACCURACY];
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buf[0] += ltable[a >> 8];
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buf[1] += rtable[a >> 8];
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accum += delta;
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buf += 2;
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} while (--nCount != 0);
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lpVoice->dwAccum = accum;
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@ -349,9 +349,9 @@ MixAudioData16MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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/* adaptive oversampling interpolation comparison */
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if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData16M(lpBuffer, nCount, lpVoice);
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return;
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData16M(lpBuffer, nCount, lpVoice);
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return;
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}
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accum = lpVoice->dwAccum;
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@ -362,12 +362,12 @@ MixAudioData16MI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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count = nCount;
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do {
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a = data[accum >> ACCURACY];
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b = data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (WORD)((SHORT)a + ((frac * ((SHORT)b - (SHORT)a)) >> 5));
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*buf++ += table[a >> 8];
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accum += delta;
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a = data[accum >> ACCURACY];
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b = data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (WORD)((SHORT)a + ((frac * ((SHORT)b - (SHORT)a)) >> 5));
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*buf++ += table[a >> 8];
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accum += delta;
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} while (--count != 0);
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lpVoice->dwAccum = accum;
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@ -383,9 +383,9 @@ MixAudioData16SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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/* adaptive oversampling interpolation comparison */
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if (lpVoice->dwFrequency < -(1 << ACCURACY) ||
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData16S(lpBuffer, nCount, lpVoice);
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return;
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lpVoice->dwFrequency > +(1 << ACCURACY)) {
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MixAudioData16S(lpBuffer, nCount, lpVoice);
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return;
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}
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accum = lpVoice->dwAccum;
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@ -397,14 +397,14 @@ MixAudioData16SI(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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buf = lpBuffer;
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do {
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a = data[accum >> ACCURACY];
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b = data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (WORD)((SHORT)a + ((frac * ((int)(SHORT)b - (int)(SHORT)a)) >> 5));
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buf[0] += ltable[a >> 8];
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buf[1] += rtable[a >> 8];
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accum += delta;
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buf += 2;
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a = data[accum >> ACCURACY];
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b = data[(accum >> ACCURACY) + 1];
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frac = (accum & ((1 << ACCURACY) - 1)) >> (ACCURACY - 5);
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a = (WORD)((SHORT)a + ((frac * ((int)(SHORT)b - (int)(SHORT)a)) >> 5));
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buf[0] += ltable[a >> 8];
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buf[1] += rtable[a >> 8];
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accum += delta;
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buf += 2;
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} while (--nCount != 0);
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lpVoice->dwAccum = accum;
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@ -417,83 +417,83 @@ static VOID MixAudioData(LPLONG lpBuffer, UINT nCount, LPVOICE lpVoice)
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UINT nSize;
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if (Synth.wFormat & AUDIO_FORMAT_STEREO)
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nCount >>= 1;
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nCount >>= 1;
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while (nCount > 0 && !(lpVoice->bControl & VOICE_STOP)) {
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/* check boundary conditions */
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if (lpVoice->bControl & VOICE_REVERSE) {
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if (lpVoice->dwAccum < lpVoice->dwLoopStart) {
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if (lpVoice->bControl & VOICE_BIDILOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopStart +
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(lpVoice->dwLoopStart - lpVoice->dwAccum);
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lpVoice->bControl ^= VOICE_REVERSE;
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continue;
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}
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else if (lpVoice->bControl & VOICE_LOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopEnd -
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(lpVoice->dwLoopStart - lpVoice->dwAccum);
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continue;
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}
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else {
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lpVoice->bControl |= VOICE_STOP;
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break;
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}
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}
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}
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else {
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if (lpVoice->dwAccum > lpVoice->dwLoopEnd) {
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if (lpVoice->bControl & VOICE_BIDILOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopEnd -
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(lpVoice->dwAccum - lpVoice->dwLoopEnd);
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lpVoice->bControl ^= VOICE_REVERSE;
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continue;
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}
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else if (lpVoice->bControl & VOICE_LOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopStart +
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(lpVoice->dwAccum - lpVoice->dwLoopEnd);
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continue;
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}
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else {
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lpVoice->bControl |= VOICE_STOP;
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break;
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}
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}
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}
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/* check boundary conditions */
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if (lpVoice->bControl & VOICE_REVERSE) {
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if (lpVoice->dwAccum < lpVoice->dwLoopStart) {
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if (lpVoice->bControl & VOICE_BIDILOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopStart +
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(lpVoice->dwLoopStart - lpVoice->dwAccum);
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lpVoice->bControl ^= VOICE_REVERSE;
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continue;
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}
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else if (lpVoice->bControl & VOICE_LOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopEnd -
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(lpVoice->dwLoopStart - lpVoice->dwAccum);
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continue;
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}
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else {
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lpVoice->bControl |= VOICE_STOP;
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break;
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}
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}
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}
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else {
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if (lpVoice->dwAccum > lpVoice->dwLoopEnd) {
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if (lpVoice->bControl & VOICE_BIDILOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopEnd -
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(lpVoice->dwAccum - lpVoice->dwLoopEnd);
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lpVoice->bControl ^= VOICE_REVERSE;
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continue;
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}
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else if (lpVoice->bControl & VOICE_LOOP) {
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lpVoice->dwAccum = lpVoice->dwLoopStart +
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(lpVoice->dwAccum - lpVoice->dwLoopEnd);
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continue;
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}
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else {
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lpVoice->bControl |= VOICE_STOP;
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break;
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}
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}
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}
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/* check for overflow and clip if necessary */
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nSize = nCount;
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if (lpVoice->bControl & VOICE_REVERSE) {
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if (nSize * lpVoice->dwFrequency >
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lpVoice->dwAccum - lpVoice->dwLoopStart)
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nSize = (lpVoice->dwAccum - lpVoice->dwLoopStart +
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lpVoice->dwFrequency) / lpVoice->dwFrequency;
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}
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else {
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if (nSize * lpVoice->dwFrequency >
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lpVoice->dwLoopEnd - lpVoice->dwAccum)
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nSize = (lpVoice->dwLoopEnd - lpVoice->dwAccum +
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lpVoice->dwFrequency) / lpVoice->dwFrequency;
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}
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/* check for overflow and clip if necessary */
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nSize = nCount;
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if (lpVoice->bControl & VOICE_REVERSE) {
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if (nSize * lpVoice->dwFrequency >
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lpVoice->dwAccum - lpVoice->dwLoopStart)
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nSize = (lpVoice->dwAccum - lpVoice->dwLoopStart +
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lpVoice->dwFrequency) / lpVoice->dwFrequency;
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}
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else {
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if (nSize * lpVoice->dwFrequency >
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lpVoice->dwLoopEnd - lpVoice->dwAccum)
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nSize = (lpVoice->dwLoopEnd - lpVoice->dwAccum +
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lpVoice->dwFrequency) / lpVoice->dwFrequency;
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}
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if (lpVoice->bControl & VOICE_REVERSE)
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lpVoice->dwFrequency = -lpVoice->dwFrequency;
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if (lpVoice->bControl & VOICE_REVERSE)
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lpVoice->dwFrequency = -lpVoice->dwFrequency;
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/* mixes chunk of data in a burst mode */
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if (lpVoice->bControl & VOICE_16BITS) {
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Synth.lpfnMixAudioProc[1] (lpBuffer, nSize, lpVoice);
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}
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else {
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Synth.lpfnMixAudioProc[0] (lpBuffer, nSize, lpVoice);
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}
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/* mixes chunk of data in a burst mode */
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if (lpVoice->bControl & VOICE_16BITS) {
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Synth.lpfnMixAudioProc[1] (lpBuffer, nSize, lpVoice);
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}
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else {
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Synth.lpfnMixAudioProc[0] (lpBuffer, nSize, lpVoice);
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}
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if (lpVoice->bControl & VOICE_REVERSE)
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lpVoice->dwFrequency = -lpVoice->dwFrequency;
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if (lpVoice->bControl & VOICE_REVERSE)
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lpVoice->dwFrequency = -lpVoice->dwFrequency;
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/* update mixing buffer address and counter */
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lpBuffer += nSize;
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if (Synth.wFormat & AUDIO_FORMAT_STEREO)
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lpBuffer += nSize;
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nCount -= nSize;
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/* update mixing buffer address and counter */
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lpBuffer += nSize;
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if (Synth.wFormat & AUDIO_FORMAT_STEREO)
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lpBuffer += nSize;
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nCount -= nSize;
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}
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}
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@ -502,21 +502,21 @@ static VOID MixAudioVoices(LPLONG lpBuffer, UINT nCount)
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UINT nVoice, nSize;
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while (nCount > 0) {
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nSize = nCount;
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if (Synth.lpfnAudioTimer != NULL) {
|
||||
nSize = (Synth.dwTimerRate - Synth.dwTimerAccum + 64L) & ~63L;
|
||||
if (nSize > nCount)
|
||||
nSize = nCount;
|
||||
if ((Synth.dwTimerAccum += nSize) >= Synth.dwTimerRate) {
|
||||
Synth.dwTimerAccum -= Synth.dwTimerRate;
|
||||
Synth.lpfnAudioTimer();
|
||||
}
|
||||
}
|
||||
for (nVoice = 0; nVoice < Synth.nVoices; nVoice++) {
|
||||
MixAudioData(lpBuffer, nSize, &Synth.aVoices[nVoice]);
|
||||
}
|
||||
lpBuffer += nSize;
|
||||
nCount -= nSize;
|
||||
nSize = nCount;
|
||||
if (Synth.lpfnAudioTimer != NULL) {
|
||||
nSize = (Synth.dwTimerRate - Synth.dwTimerAccum + 64L) & ~63L;
|
||||
if (nSize > nCount)
|
||||
nSize = nCount;
|
||||
if ((Synth.dwTimerAccum += nSize) >= Synth.dwTimerRate) {
|
||||
Synth.dwTimerAccum -= Synth.dwTimerRate;
|
||||
Synth.lpfnAudioTimer();
|
||||
}
|
||||
}
|
||||
for (nVoice = 0; nVoice < Synth.nVoices; nVoice++) {
|
||||
MixAudioData(lpBuffer, nSize, &Synth.aVoices[nVoice]);
|
||||
}
|
||||
lpBuffer += nSize;
|
||||
nCount -= nSize;
|
||||
}
|
||||
}
|
||||
|
||||
@ -541,30 +541,30 @@ static UINT AIAPI OpenAudio(LPAUDIOINFO lpInfo)
|
||||
Synth.wFormat = lpInfo->wFormat;
|
||||
Synth.nSampleRate = lpInfo->nSampleRate;
|
||||
if (Synth.wFormat & AUDIO_FORMAT_FILTER) {
|
||||
Synth.lpfnMixAudioProc[0] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData08SI : MixAudioData08MI;
|
||||
Synth.lpfnMixAudioProc[1] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData16SI : MixAudioData16MI;
|
||||
Synth.lpfnMixAudioProc[0] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData08SI : MixAudioData08MI;
|
||||
Synth.lpfnMixAudioProc[1] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData16SI : MixAudioData16MI;
|
||||
}
|
||||
else {
|
||||
Synth.lpfnMixAudioProc[0] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData08S : MixAudioData08M;
|
||||
Synth.lpfnMixAudioProc[1] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData16S : MixAudioData16M;
|
||||
Synth.lpfnMixAudioProc[0] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData08S : MixAudioData08M;
|
||||
Synth.lpfnMixAudioProc[1] = (Synth.wFormat & AUDIO_FORMAT_STEREO) ?
|
||||
MixAudioData16S : MixAudioData16M;
|
||||
}
|
||||
|
||||
/* allocate volume (0-64) and filter (0-31) table */
|
||||
Synth.lpMemory = malloc(sizeof(LONG) * 65 * 256 +
|
||||
sizeof(BYTE) * 32 * 256 + 1023);
|
||||
sizeof(BYTE) * 32 * 256 + 1023);
|
||||
if (Synth.lpMemory != NULL) {
|
||||
uint64_t tmp;
|
||||
tmp = ((uint64_t)(Synth.lpMemory) + 1023) & ~1023;
|
||||
lpVolumeTable = (LPLONG)(tmp);
|
||||
lpFilterTable = (LPBYTE) (lpVolumeTable + 65 * 256);
|
||||
ASetAudioMixerValue(AUDIO_MIXER_MASTER_VOLUME, 96);
|
||||
ASetAudioCallback(UpdateVoices);
|
||||
lpVolumeTable = (LPLONG)(tmp);
|
||||
lpFilterTable = (LPBYTE) (lpVolumeTable + 65 * 256);
|
||||
ASetAudioMixerValue(AUDIO_MIXER_MASTER_VOLUME, 96);
|
||||
ASetAudioCallback(UpdateVoices);
|
||||
|
||||
return AUDIO_ERROR_NONE;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_NOMEMORY;
|
||||
}
|
||||
@ -572,7 +572,7 @@ static UINT AIAPI OpenAudio(LPAUDIOINFO lpInfo)
|
||||
static UINT AIAPI CloseAudio(VOID)
|
||||
{
|
||||
if (Synth.lpMemory != NULL)
|
||||
free(Synth.lpMemory);
|
||||
free(Synth.lpMemory);
|
||||
memset(&Synth, 0, sizeof(Synth));
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
@ -585,38 +585,38 @@ static UINT AIAPI SetAudioMixerValue(UINT nChannel, UINT nMixerValue)
|
||||
LONG dwAccum, dwDelta;
|
||||
|
||||
if (Synth.lpMemory == NULL)
|
||||
return AUDIO_ERROR_NOTSUPPORTED;
|
||||
return AUDIO_ERROR_NOTSUPPORTED;
|
||||
|
||||
/* master volume must be less than 256 units */
|
||||
if (nChannel != AUDIO_MIXER_MASTER_VOLUME || nMixerValue > 256)
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
|
||||
/* half dynamic range for mono output */
|
||||
if (!(Synth.wFormat & AUDIO_FORMAT_STEREO))
|
||||
nMixerValue >>= 1;
|
||||
nMixerValue >>= 1;
|
||||
|
||||
/* build volume table (0-64) */
|
||||
lpVolume = lpVolumeTable;
|
||||
dwDelta = 0;
|
||||
for (nVolume = 0; nVolume <= 64; nVolume++, dwDelta += nMixerValue) {
|
||||
dwAccum = 0;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += dwDelta)
|
||||
*lpVolume++ = dwAccum >> 4;
|
||||
dwAccum = -dwAccum;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += dwDelta)
|
||||
*lpVolume++ = dwAccum >> 4;
|
||||
dwAccum = 0;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += dwDelta)
|
||||
*lpVolume++ = dwAccum >> 4;
|
||||
dwAccum = -dwAccum;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += dwDelta)
|
||||
*lpVolume++ = dwAccum >> 4;
|
||||
}
|
||||
|
||||
#ifdef __FILTER__
|
||||
/* build lowpass filter table (0-31) */
|
||||
lpFilter = lpFilterTable;
|
||||
for (nVolume = 0; nVolume < 32; nVolume++) {
|
||||
dwAccum = 0;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += nVolume)
|
||||
*lpFilter++ = dwAccum >> 5;
|
||||
dwAccum = -dwAccum;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += nVolume)
|
||||
*lpFilter++ = dwAccum >> 5;
|
||||
dwAccum = 0;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += nVolume)
|
||||
*lpFilter++ = dwAccum >> 5;
|
||||
dwAccum = -dwAccum;
|
||||
for (nSample = 0; nSample < 128; nSample++, dwAccum += nVolume)
|
||||
*lpFilter++ = dwAccum >> 5;
|
||||
}
|
||||
#endif
|
||||
|
||||
@ -631,10 +631,10 @@ static UINT AIAPI OpenVoices(UINT nVoices)
|
||||
* Initialize waveform synthesizer structure for playback
|
||||
*/
|
||||
if (nVoices >= 1 && nVoices <= AUDIO_MAX_VOICES) {
|
||||
Synth.nVoices = nVoices;
|
||||
for (nVoice = 0; nVoice < Synth.nVoices; nVoice++)
|
||||
Synth.aVoices[nVoice].bControl = VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
Synth.nVoices = nVoices;
|
||||
for (nVoice = 0; nVoice < Synth.nVoices; nVoice++)
|
||||
Synth.aVoices[nVoice].bControl = VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
@ -645,7 +645,7 @@ static UINT AIAPI CloseVoices(VOID)
|
||||
|
||||
memset(Synth.aVoices, 0, sizeof(Synth.aVoices));
|
||||
for (nVoice = 0; nVoice < AUDIO_MAX_VOICES; nVoice++)
|
||||
Synth.aVoices[nVoice].bControl = VOICE_STOP;
|
||||
Synth.aVoices[nVoice].bControl = VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
|
||||
@ -662,8 +662,8 @@ static LONG AIAPI GetAudioDataAvail(VOID)
|
||||
static UINT AIAPI CreateAudioData(LPAUDIOWAVE lpWave)
|
||||
{
|
||||
if (lpWave != NULL) {
|
||||
lpWave->dwHandle = (uintptr_t) lpWave->lpData;
|
||||
return AUDIO_ERROR_NONE;
|
||||
lpWave->dwHandle = (uintptr_t) lpWave->lpData;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -671,7 +671,7 @@ static UINT AIAPI CreateAudioData(LPAUDIOWAVE lpWave)
|
||||
static UINT AIAPI DestroyAudioData(LPAUDIOWAVE lpWave)
|
||||
{
|
||||
if (lpWave != NULL && lpWave->dwHandle != 0) {
|
||||
return AUDIO_ERROR_NONE;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -679,15 +679,15 @@ static UINT AIAPI DestroyAudioData(LPAUDIOWAVE lpWave)
|
||||
static UINT AIAPI WriteAudioData(LPAUDIOWAVE lpWave, DWORD dwOffset, UINT nCount)
|
||||
{
|
||||
if (lpWave != NULL && lpWave->dwHandle != 0) {
|
||||
/* anticlick removal work around */
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_LOOP) {
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLoopEnd) =
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLoopStart);
|
||||
}
|
||||
else if (dwOffset + nCount >= lpWave->dwLength) {
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLength) = 0;
|
||||
}
|
||||
return AUDIO_ERROR_NONE;
|
||||
/* anticlick removal work around */
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_LOOP) {
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLoopEnd) =
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLoopStart);
|
||||
}
|
||||
else if (dwOffset + nCount >= lpWave->dwLength) {
|
||||
*(LPDWORD) (lpWave->dwHandle + lpWave->dwLength) = 0;
|
||||
}
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -697,30 +697,30 @@ static UINT AIAPI PrimeVoice(UINT nVoice, LPAUDIOWAVE lpWave)
|
||||
LPVOICE lpVoice;
|
||||
|
||||
if (nVoice < Synth.nVoices && lpWave != NULL && lpWave->dwHandle != 0) {
|
||||
lpVoice = &Synth.aVoices[nVoice];
|
||||
lpVoice->lpData = (LPVOID) lpWave->dwHandle;
|
||||
lpVoice->bControl = VOICE_STOP;
|
||||
lpVoice->dwAccum = 0;
|
||||
if (lpWave->wFormat & (AUDIO_FORMAT_LOOP | AUDIO_FORMAT_BIDILOOP)) {
|
||||
lpVoice->dwLoopStart = lpWave->dwLoopStart;
|
||||
lpVoice->dwLoopEnd = lpWave->dwLoopEnd;
|
||||
lpVoice->bControl |= VOICE_LOOP;
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_BIDILOOP)
|
||||
lpVoice->bControl |= VOICE_BIDILOOP;
|
||||
}
|
||||
else {
|
||||
lpVoice->dwLoopStart = lpWave->dwLength;
|
||||
lpVoice->dwLoopEnd = lpWave->dwLength;
|
||||
}
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_16BITS) {
|
||||
lpVoice->dwLoopStart >>= 1;
|
||||
lpVoice->dwLoopEnd >>= 1;
|
||||
lpVoice->bControl |= VOICE_16BITS;
|
||||
}
|
||||
lpVoice->dwAccum <<= ACCURACY;
|
||||
lpVoice->dwLoopStart <<= ACCURACY;
|
||||
lpVoice->dwLoopEnd <<= ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
lpVoice = &Synth.aVoices[nVoice];
|
||||
lpVoice->lpData = (LPVOID) lpWave->dwHandle;
|
||||
lpVoice->bControl = VOICE_STOP;
|
||||
lpVoice->dwAccum = 0;
|
||||
if (lpWave->wFormat & (AUDIO_FORMAT_LOOP | AUDIO_FORMAT_BIDILOOP)) {
|
||||
lpVoice->dwLoopStart = lpWave->dwLoopStart;
|
||||
lpVoice->dwLoopEnd = lpWave->dwLoopEnd;
|
||||
lpVoice->bControl |= VOICE_LOOP;
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_BIDILOOP)
|
||||
lpVoice->bControl |= VOICE_BIDILOOP;
|
||||
}
|
||||
else {
|
||||
lpVoice->dwLoopStart = lpWave->dwLength;
|
||||
lpVoice->dwLoopEnd = lpWave->dwLength;
|
||||
}
|
||||
if (lpWave->wFormat & AUDIO_FORMAT_16BITS) {
|
||||
lpVoice->dwLoopStart >>= 1;
|
||||
lpVoice->dwLoopEnd >>= 1;
|
||||
lpVoice->bControl |= VOICE_16BITS;
|
||||
}
|
||||
lpVoice->dwAccum <<= ACCURACY;
|
||||
lpVoice->dwLoopStart <<= ACCURACY;
|
||||
lpVoice->dwLoopEnd <<= ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -728,8 +728,8 @@ static UINT AIAPI PrimeVoice(UINT nVoice, LPAUDIOWAVE lpWave)
|
||||
static UINT AIAPI StartVoice(UINT nVoice)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
Synth.aVoices[nVoice].bControl &= ~VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
Synth.aVoices[nVoice].bControl &= ~VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -737,8 +737,8 @@ static UINT AIAPI StartVoice(UINT nVoice)
|
||||
static UINT AIAPI StopVoice(UINT nVoice)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
Synth.aVoices[nVoice].bControl |= VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
Synth.aVoices[nVoice].bControl |= VOICE_STOP;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
|
||||
@ -747,12 +747,12 @@ static UINT AIAPI StopVoice(UINT nVoice)
|
||||
static UINT AIAPI SetVoicePosition(UINT nVoice, LONG dwPosition)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
dwPosition <<= ACCURACY;
|
||||
if (dwPosition >= 0 && dwPosition < Synth.aVoices[nVoice].dwLoopEnd) {
|
||||
Synth.aVoices[nVoice].dwAccum = dwPosition;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
dwPosition <<= ACCURACY;
|
||||
if (dwPosition >= 0 && dwPosition < Synth.aVoices[nVoice].dwLoopEnd) {
|
||||
Synth.aVoices[nVoice].dwAccum = dwPosition;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -760,13 +760,13 @@ static UINT AIAPI SetVoicePosition(UINT nVoice, LONG dwPosition)
|
||||
static UINT AIAPI SetVoiceFrequency(UINT nVoice, LONG dwFrequency)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (dwFrequency >= AUDIO_MIN_FREQUENCY &&
|
||||
dwFrequency <= AUDIO_MAX_FREQUENCY) {
|
||||
Synth.aVoices[nVoice].dwFrequency = ((dwFrequency << ACCURACY) +
|
||||
(Synth.nSampleRate >> 1)) / Synth.nSampleRate;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (dwFrequency >= AUDIO_MIN_FREQUENCY &&
|
||||
dwFrequency <= AUDIO_MAX_FREQUENCY) {
|
||||
Synth.aVoices[nVoice].dwFrequency = ((dwFrequency << ACCURACY) +
|
||||
(Synth.nSampleRate >> 1)) / Synth.nSampleRate;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -774,11 +774,11 @@ static UINT AIAPI SetVoiceFrequency(UINT nVoice, LONG dwFrequency)
|
||||
static UINT AIAPI SetVoiceVolume(UINT nVoice, UINT nVolume)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (nVolume <= AUDIO_MAX_VOLUME) {
|
||||
Synth.aVoices[nVoice].nVolume = nVolume >> 1;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (nVolume <= AUDIO_MAX_VOLUME) {
|
||||
Synth.aVoices[nVoice].nVolume = nVolume >> 1;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -786,11 +786,11 @@ static UINT AIAPI SetVoiceVolume(UINT nVoice, UINT nVolume)
|
||||
static UINT AIAPI SetVoicePanning(UINT nVoice, UINT nPanning)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (nPanning <= AUDIO_MAX_PANNING) {
|
||||
Synth.aVoices[nVoice].nPanning = nPanning;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (nPanning <= AUDIO_MAX_PANNING) {
|
||||
Synth.aVoices[nVoice].nPanning = nPanning;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -798,11 +798,11 @@ static UINT AIAPI SetVoicePanning(UINT nVoice, UINT nPanning)
|
||||
static UINT AIAPI GetVoicePosition(UINT nVoice, LPLONG lpdwPosition)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (lpdwPosition != NULL) {
|
||||
*lpdwPosition = Synth.aVoices[nVoice].dwAccum >> ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (lpdwPosition != NULL) {
|
||||
*lpdwPosition = Synth.aVoices[nVoice].dwAccum >> ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -810,12 +810,12 @@ static UINT AIAPI GetVoicePosition(UINT nVoice, LPLONG lpdwPosition)
|
||||
static UINT AIAPI GetVoiceFrequency(UINT nVoice, LPLONG lpdwFrequency)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (lpdwFrequency != NULL) {
|
||||
*lpdwFrequency = (Synth.aVoices[nVoice].dwFrequency *
|
||||
Synth.nSampleRate) >> ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (lpdwFrequency != NULL) {
|
||||
*lpdwFrequency = (Synth.aVoices[nVoice].dwFrequency *
|
||||
Synth.nSampleRate) >> ACCURACY;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -823,11 +823,11 @@ static UINT AIAPI GetVoiceFrequency(UINT nVoice, LPLONG lpdwFrequency)
|
||||
static UINT AIAPI GetVoiceVolume(UINT nVoice, LPUINT lpnVolume)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (lpnVolume != NULL) {
|
||||
*lpnVolume = Synth.aVoices[nVoice].nVolume << 1;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (lpnVolume != NULL) {
|
||||
*lpnVolume = Synth.aVoices[nVoice].nVolume << 1;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -835,11 +835,11 @@ static UINT AIAPI GetVoiceVolume(UINT nVoice, LPUINT lpnVolume)
|
||||
static UINT AIAPI GetVoicePanning(UINT nVoice, LPUINT lpnPanning)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (lpnPanning != NULL) {
|
||||
*lpnPanning = Synth.aVoices[nVoice].nPanning;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (lpnPanning != NULL) {
|
||||
*lpnPanning = Synth.aVoices[nVoice].nPanning;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -847,11 +847,11 @@ static UINT AIAPI GetVoicePanning(UINT nVoice, LPUINT lpnPanning)
|
||||
static UINT AIAPI GetVoiceStatus(UINT nVoice, LPBOOL lpnStatus)
|
||||
{
|
||||
if (nVoice < Synth.nVoices) {
|
||||
if (lpnStatus != NULL) {
|
||||
*lpnStatus = (Synth.aVoices[nVoice].bControl & VOICE_STOP) != 0;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
if (lpnStatus != NULL) {
|
||||
*lpnStatus = (Synth.aVoices[nVoice].bControl & VOICE_STOP) != 0;
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
return AUDIO_ERROR_INVALHANDLE;
|
||||
}
|
||||
@ -865,11 +865,11 @@ static UINT AIAPI SetAudioTimerProc(LPFNAUDIOTIMER lpfnAudioTimer)
|
||||
static UINT AIAPI SetAudioTimerRate(UINT nBPM)
|
||||
{
|
||||
if (nBPM >= 0x20 && nBPM <= 0xFF) {
|
||||
Synth.dwTimerRate = Synth.nSampleRate;
|
||||
if (Synth.wFormat & AUDIO_FORMAT_STEREO)
|
||||
Synth.dwTimerRate <<= 1;
|
||||
Synth.dwTimerRate = (5 * Synth.dwTimerRate) / (2 * nBPM);
|
||||
return AUDIO_ERROR_NONE;
|
||||
Synth.dwTimerRate = Synth.nSampleRate;
|
||||
if (Synth.wFormat & AUDIO_FORMAT_STEREO)
|
||||
Synth.dwTimerRate <<= 1;
|
||||
Synth.dwTimerRate = (5 * Synth.dwTimerRate) / (2 * nBPM);
|
||||
return AUDIO_ERROR_NONE;
|
||||
}
|
||||
return AUDIO_ERROR_INVALPARAM;
|
||||
}
|
||||
@ -880,15 +880,15 @@ static VOID AIAPI UpdateVoices(LPBYTE lpData, UINT nCount)
|
||||
UINT nSamples;
|
||||
|
||||
if (Synth.wFormat & AUDIO_FORMAT_16BITS)
|
||||
nCount >>= 1;
|
||||
nCount >>= 1;
|
||||
while (nCount > 0) {
|
||||
if ((nSamples = nCount) > BUFFERSIZE)
|
||||
nSamples = BUFFERSIZE;
|
||||
memset(aBuffer, 0, nSamples << 2);
|
||||
MixAudioVoices(aBuffer, nSamples);
|
||||
QuantAudioData(lpData, aBuffer, nSamples);
|
||||
lpData += nSamples << ((Synth.wFormat & AUDIO_FORMAT_16BITS) != 0);
|
||||
nCount -= nSamples;
|
||||
if ((nSamples = nCount) > BUFFERSIZE)
|
||||
nSamples = BUFFERSIZE;
|
||||
memset(aBuffer, 0, nSamples << 2);
|
||||
MixAudioVoices(aBuffer, nSamples);
|
||||
QuantAudioData(lpData, aBuffer, nSamples);
|
||||
lpData += nSamples << ((Synth.wFormat & AUDIO_FORMAT_16BITS) != 0);
|
||||
nCount -= nSamples;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user