Add Dsp::GetSamples and cleanup Apu
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scawful
@@ -7,282 +7,21 @@ namespace app {
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namespace emu {
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namespace audio {
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void DigitalSignalProcessor::Reset() {}
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void Dsp::Reset() {}
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uint8_t DigitalSignalProcessor::ReadVoiceReg(uint8_t voice, uint8_t reg) const {
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voice %= kNumVoices;
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switch (reg % kNumVoiceRegs) {
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case 0:
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return voices_[voice].vol_left;
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case 1:
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return voices_[voice].vol_right;
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case 2:
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return voices_[voice].pitch_low;
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case 3:
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return voices_[voice].pitch_high;
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case 4:
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return voices_[voice].source_number;
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case 5:
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return voices_[voice].adsr1;
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case 6:
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return voices_[voice].adsr2;
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case 7:
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return voices_[voice].gain;
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case 8:
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return voices_[voice].envx;
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case 9:
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return voices_[voice].outx;
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default:
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return 0; // This shouldn't happen, but it's good to have a default
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// case
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void Dsp::GetSamples(int16_t* sample_data, int samples_per_frame,
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bool pal_timing) {
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// resample from 534 / 641 samples per frame to wanted value
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float wantedSamples = (pal_timing ? 641.0 : 534.0);
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double adder = wantedSamples / samples_per_frame;
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double location = sample_offset_ - wantedSamples;
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for (int i = 0; i < samples_per_frame; i++) {
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sample_data[i * 2] = sample_buffer_[(((int)location) & 0x3ff) * 2];
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sample_data[i * 2 + 1] = sample_buffer_[(((int)location) & 0x3ff) * 2 + 1];
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location += adder;
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}
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}
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void DigitalSignalProcessor::WriteVoiceReg(uint8_t voice, uint8_t reg,
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uint8_t value) {
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voice %= kNumVoices;
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switch (reg % kNumVoiceRegs) {
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case 0:
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voices_[voice].vol_left = static_cast<int8_t>(value);
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break;
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case 1:
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voices_[voice].vol_right = static_cast<int8_t>(value);
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break;
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case 2:
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voices_[voice].pitch_low = value;
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break;
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case 3:
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voices_[voice].pitch_high = value;
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break;
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case 4:
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voices_[voice].source_number = value;
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break;
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case 5:
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voices_[voice].adsr1 = value;
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break;
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case 6:
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voices_[voice].adsr2 = value;
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break;
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case 7:
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voices_[voice].gain = value;
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break;
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// Note: envx and outx are read-only, so they don't have cases here
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}
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}
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// Set the callbacks
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void DigitalSignalProcessor::SetSampleFetcher(SampleFetcher fetcher) {
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sample_fetcher_ = fetcher;
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}
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void DigitalSignalProcessor::SetSamplePusher(SamplePusher pusher) {
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sample_pusher_ = pusher;
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}
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int16_t DigitalSignalProcessor::DecodeSample(uint8_t voice_num) {
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Voice const& voice = voices_[voice_num];
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uint16_t sample_address = voice.source_number;
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// Use the callback to fetch the sample
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int16_t sample = static_cast<int16_t>(sample_fetcher_(sample_address) << 8);
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return sample;
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}
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int16_t DigitalSignalProcessor::ProcessSample(uint8_t voice_num,
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int16_t sample) {
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Voice const& voice = voices_[voice_num];
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// Adjust the pitch (for simplicity, we're just adjusting the sample value)
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sample += voice.pitch_low + (voice.pitch_high << 8);
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// Apply volume (separate for left and right for stereo sound)
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int16_t left_sample = (sample * voice.vol_left) / 255;
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int16_t right_sample = (sample * voice.vol_right) / 255;
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// Combine stereo samples into a single 16-bit value
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return (left_sample + right_sample) / 2;
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}
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void DigitalSignalProcessor::MixSamples() {
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int16_t mixed_sample = 0;
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for (uint8_t i = 0; i < kNumVoices; i++) {
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int16_t decoded_sample = DecodeSample(i);
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int16_t processed_sample = ProcessSample(i, decoded_sample);
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mixed_sample += processed_sample;
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}
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// Clamp the mixed sample to 16-bit range
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if (mixed_sample > 32767) {
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mixed_sample = 32767;
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} else if (mixed_sample < -32768) {
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mixed_sample = -32768;
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}
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// Use the callback to push the mixed sample
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sample_pusher_(mixed_sample);
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}
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void DigitalSignalProcessor::UpdateEnvelope(uint8_t voice) {
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uint8_t adsr1 = ReadVoiceReg(voice, 0x05);
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uint8_t adsr2 = ReadVoiceReg(voice, 0x06);
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uint8_t gain = ReadVoiceReg(voice, 0x07);
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uint8_t enableADSR = (adsr1 & 0x80) >> 7;
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if (enableADSR) {
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// Handle ADSR envelope
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Voice& voice_obj = voices_[voice];
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switch (voice_obj.state) {
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case VoiceState::ATTACK:
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// Update amplitude based on attack rate
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voice_obj.current_amplitude += AttackRate(adsr1);
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if (voice_obj.current_amplitude >= ENVELOPE_MAX) {
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voice_obj.current_amplitude = ENVELOPE_MAX;
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voice_obj.state = VoiceState::DECAY;
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}
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break;
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case VoiceState::DECAY:
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// Update amplitude based on decay rate
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voice_obj.current_amplitude -= DecayRate(adsr2);
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if (voice_obj.current_amplitude <= voice_obj.decay_level) {
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voice_obj.current_amplitude = voice_obj.decay_level;
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voice_obj.state = VoiceState::SUSTAIN;
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}
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break;
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case VoiceState::SUSTAIN:
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// Keep amplitude at the calculated decay level
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voice_obj.current_amplitude = voice_obj.decay_level;
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break;
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case VoiceState::RELEASE:
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// Update amplitude based on release rate
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voice_obj.current_amplitude -= ReleaseRate(adsr2);
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if (voice_obj.current_amplitude <= 0) {
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voice_obj.current_amplitude = 0;
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voice_obj.state = VoiceState::OFF;
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}
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break;
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default:
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break;
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}
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} else {
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// Handle Gain envelope
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// Extract mode from the gain byte
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uint8_t mode = (gain & 0xE0) >> 5;
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uint8_t rate = gain & 0x1F;
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Voice& voice_obj = voices_[voice];
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switch (mode) {
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case 0: // Direct Designation
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case 1:
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case 2:
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case 3:
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voice_obj.current_amplitude =
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rate << 3; // Multiplying by 8 to scale to 0-255
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break;
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case 6: // Increase Mode (Linear)
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voice_obj.current_amplitude += gainTimings[0][rate];
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if (voice_obj.current_amplitude > ENVELOPE_MAX) {
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voice_obj.current_amplitude = ENVELOPE_MAX;
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}
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break;
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case 7: // Increase Mode (Bent Line)
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// Hypothetical behavior: Increase linearly at first, then increase
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// more slowly You'll likely need to adjust this based on your
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// specific requirements
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if (voice_obj.current_amplitude < (ENVELOPE_MAX / 2)) {
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voice_obj.current_amplitude += gainTimings[1][rate];
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} else {
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voice_obj.current_amplitude += gainTimings[1][rate] / 2;
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}
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if (voice_obj.current_amplitude > ENVELOPE_MAX) {
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voice_obj.current_amplitude = ENVELOPE_MAX;
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}
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break;
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case 4: // Decrease Mode (Linear)
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if (voice_obj.current_amplitude < gainTimings[2][rate]) {
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voice_obj.current_amplitude = 0;
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} else {
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voice_obj.current_amplitude -= gainTimings[2][rate];
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}
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break;
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case 5: // Decrease Mode (Exponential)
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voice_obj.current_amplitude -=
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(voice_obj.current_amplitude * gainTimings[3][rate]) / ENVELOPE_MAX;
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break;
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default:
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// Default behavior can be handled here if necessary
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break;
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}
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}
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}
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void DigitalSignalProcessor::update_voice_state(uint8_t voice_num) {
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if (voice_num >= kNumVoices) return;
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Voice& voice = voices_[voice_num];
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switch (voice.state) {
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case VoiceState::OFF:
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// Reset current amplitude
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voice.current_amplitude = 0;
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break;
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case VoiceState::ATTACK:
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// Increase the current amplitude at a rate defined by the ATTACK
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// setting
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voice.current_amplitude += AttackRate(voice.adsr1);
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if (voice.current_amplitude >= ENVELOPE_MAX) {
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voice.current_amplitude = ENVELOPE_MAX;
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voice.state = VoiceState::DECAY;
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voice.decay_level = CalculateDecayLevel(voice.adsr2);
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}
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break;
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case VoiceState::DECAY:
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// Decrease the current amplitude at a rate defined by the DECAY setting
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voice.current_amplitude -= DecayRate(voice.adsr2);
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if (voice.current_amplitude <= voice.decay_level) {
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voice.current_amplitude = voice.decay_level;
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voice.state = VoiceState::SUSTAIN;
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}
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break;
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case VoiceState::SUSTAIN:
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// Keep the current amplitude at the decay level
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break;
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case VoiceState::RELEASE:
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// Decrease the current amplitude at a rate defined by the RELEASE
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// setting
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voice.current_amplitude -= ReleaseRate(voice.adsr2);
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if (voice.current_amplitude == 0) {
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voice.state = VoiceState::OFF;
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}
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break;
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}
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}
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void DigitalSignalProcessor::process_envelope(uint8_t voice_num) {
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if (voice_num >= kNumVoices) return;
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Voice& voice = voices_[voice_num];
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// Update the voice state first (based on keys, etc.)
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update_voice_state(voice_num);
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// Calculate the envelope value based on the current amplitude
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voice.envx = calculate_envelope_value(voice.current_amplitude);
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// Apply the envelope value to the audio output
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apply_envelope_to_output(voice_num);
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}
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} // namespace audio
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} // namespace emu
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} // namespace app
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