Method readBufferIntern

Source/Amiga/paula.cpp:195–287

template<bool stereo>
int Paula::readBufferIntern(int16 *buffer, const int numSamples) {
	int samples = stereo ? numSamples / 2 : numSamples;
	while (samples > 0) {

		// Handle 'interrupts'. This gives subclasses the chance to adjust the channel data
		// (e.g. insert new samples, do pitch bending, whatever).
		if (_curInt == 0) {
			_curInt = _intFreq;
			interrupt();
		}

		// Compute how many samples to generate: at most the requested number of samples,
		// of course, but we may stop earlier when an 'interrupt' is expected.
		const unsigned int nSamples = std::min((unsigned int)samples, _curInt);

		// Loop over the four channels of the emulated Paula chip
		for (int voice = 0; voice < NUM_VOICES; voice++) {
			// No data, or paused -> skip channel
			if (!_voice[voice].data || (_voice[voice].period <= 0))
				continue;

			// The Paula chip apparently run at 7.0937892 MHz in the PAL
			// version and at 7.1590905 MHz in the NTSC version. We divide this
			// by the requested the requested output sampling rate _rate
			// (typically 44.1 kHz or 22.05 kHz) obtaining the value _periodScale.
			// This is then divided by the "period" of the channel we are
			// processing, to obtain the correct output 'rate'.
			frac_t rate = doubleToFrac(_periodScale / _voice[voice].period);
			// Cap the volume
			_voice[voice].volume = std::min((uint8) volumeMax[voice], _voice[voice].volume);


			Channel &ch = _voice[voice];
			int16 *p = buffer;
			int neededSamples = nSamples;

			// NOTE: A Protracker (or other module format) player might actually
			// push the offset past the sample length in its interrupt(), in which
			// case the first mixBuffer() call should not mix anything, and the loop
			// should be triggered.
			// Thus, doing an assert(ch.offset.int_off < ch.length) here is wrong.
			// An example where this happens is a certain Protracker module played
			// by the OS/2 version of Hopkins FBI.

			// Mix the generated samples into the output buffer
			neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning, _filterState, voice);

			// Wrap around if necessary
			if (ch.offset.int_off >= ch.length) {
				// Important: Wrap around the offset *before* updating the voice length.
				// Otherwise, if length != lengthRepeat we would wrap incorrectly.
				// Note: If offset >= 2*len ever occurs, the following would be wrong;
				// instead of subtracting, we then should compute the modulus using "%=".
				// Since that requires a division and is slow, and shouldn't be necessary
				// in practice anyway, we only use subtraction.
				ch.offset.int_off -= ch.length;
				ch.dmaCount++;