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https://github.com/TomHarte/CLK.git
synced 2024-11-22 12:33:29 +00:00
Makes some attempt at stereo support, with the Amstrad CPC being the test case.
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@ -126,6 +126,7 @@ class AYDeferrer {
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/// Constructs a new AY instance and sets its clock rate.
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AYDeferrer() : ay_(GI::AY38910::Personality::AY38910, audio_queue_), speaker_(ay_) {
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speaker_.set_input_rate(1000000);
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ay_.set_output_mixing(true, 0.0, 0.5, 1.0, 1.0, 0.5, 0.0);
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}
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~AYDeferrer() {
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@ -160,7 +161,7 @@ class AYDeferrer {
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private:
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Concurrency::DeferringAsyncTaskQueue audio_queue_;
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GI::AY38910::AY38910 ay_;
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Outputs::Speaker::LowpassSpeaker<GI::AY38910::AY38910, false> speaker_;
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Outputs::Speaker::LowpassSpeaker<GI::AY38910::AY38910, true> speaker_;
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HalfCycles cycles_since_update_;
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};
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@ -66,7 +66,7 @@ template <typename SampleSource, bool is_stereo> class LowpassSpeaker: public Sp
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filter_parameters_.output_cycles_per_second = cycles_per_second;
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filter_parameters_.parameters_are_dirty = true;
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output_buffer_.resize(std::size_t(buffer_size));
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output_buffer_.resize(std::size_t(buffer_size) * (is_stereo ? 2 : 1);
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}
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bool get_is_stereo() final {
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@ -144,12 +144,11 @@ template <typename SampleSource, bool is_stereo> class LowpassSpeaker: public Sp
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switch(conversion_) {
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case Conversion::Copy:
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while(cycles_remaining) {
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const auto cycles_to_read = std::min(output_buffer_.size() - output_buffer_pointer_, cycles_remaining);
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const auto cycles_to_read = std::min((output_buffer_.size() - output_buffer_pointer_) / (is_stereo ? 2 : 1), cycles_remaining);
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sample_source_.get_samples(cycles_to_read, &output_buffer_[output_buffer_pointer_ ]);
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output_buffer_pointer_ += cycles_to_read * (is_stereo ? 2 : 1);
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sample_source_.get_samples(cycles_to_read, &output_buffer_[output_buffer_pointer_]);
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output_buffer_pointer_ += cycles_to_read;
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// announce to delegate if full
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// Announce to delegate if full.
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if(output_buffer_pointer_ == output_buffer_.size()) {
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output_buffer_pointer_ = 0;
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did_complete_samples(this, output_buffer_);
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@ -161,14 +160,16 @@ template <typename SampleSource, bool is_stereo> class LowpassSpeaker: public Sp
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case Conversion::ResampleSmaller:
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while(cycles_remaining) {
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const auto cycles_to_read = std::min(cycles_remaining, input_buffer_.size() - input_buffer_depth_);
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const auto cycles_to_read = std::min((input_buffer_.size() - input_buffer_depth_) / (is_stereo ? 2 : 1), cycles_remaining);
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sample_source_.get_samples(cycles_to_read, &input_buffer_[input_buffer_depth_]);
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cycles_remaining -= cycles_to_read;
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input_buffer_depth_ += cycles_to_read;
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input_buffer_depth_ += cycles_to_read * (is_stereo ? 2 : 1);
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if(input_buffer_depth_ == input_buffer_.size()) {
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resample_input_buffer();
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}
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cycles_remaining -= cycles_to_read;
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}
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break;
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@ -243,24 +244,31 @@ template <typename SampleSource, bool is_stereo> class LowpassSpeaker: public Sp
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// that means nothing to do.
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default: break;
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case Conversion::ResampleSmaller:
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case Conversion::ResampleSmaller: {
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// Reize the input buffer only if absolutely necessary; if sizing downward
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// such that a sample would otherwise be lost then output it now. Keep anything
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// currently in the input buffer that hasn't yet been processed.
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if(input_buffer_.size() != size_t(number_of_taps)) {
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if(input_buffer_depth_ >= size_t(number_of_taps)) {
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const size_t required_buffer_size = size_t(number_of_taps) * (is_stereo ? 2 : 1);
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if(input_buffer_.size() != required_buffer_size) {
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if(input_buffer_depth_ >= required_buffer_size) {
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resample_input_buffer();
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input_buffer_depth_ %= size_t(number_of_taps);
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input_buffer_depth_ %= required_buffer_size;
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}
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input_buffer_.resize(size_t(number_of_taps));
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input_buffer_.resize(required_buffer_size);
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}
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break;
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} break;
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}
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}
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inline void resample_input_buffer() {
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output_buffer_[output_buffer_pointer_] = filter_->apply(input_buffer_.data());
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output_buffer_pointer_++;
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if constexpr (is_stereo) {
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output_buffer_[output_buffer_pointer_ + 0] = filter_->apply(input_buffer_.data(), 2);
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output_buffer_[output_buffer_pointer_ + 1] = filter_->apply(input_buffer_.data() + 1, 2);
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output_buffer_pointer_+= 2;
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} else {
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output_buffer_[output_buffer_pointer_] = filter_->apply(input_buffer_.data());
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output_buffer_pointer_++;
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}
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// Announce to delegate if full.
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if(output_buffer_pointer_ == output_buffer_.size()) {
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@ -279,8 +287,9 @@ template <typename SampleSource, bool is_stereo> class LowpassSpeaker: public Sp
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sizeof(int16_t) * (input_buffer_.size() - steps));
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input_buffer_depth_ -= steps;
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} else {
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if(steps > input_buffer_.size())
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sample_source_.skip_samples(steps - input_buffer_.size());
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if(steps > input_buffer_.size()) {
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sample_source_.skip_samples((steps - input_buffer_.size()) / (is_stereo ? 2 : 1));
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}
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input_buffer_depth_ = 0;
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}
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}
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@ -49,15 +49,15 @@ class FIRFilter {
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@param src The source buffer to apply the filter to.
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@returns The result of applying the filter.
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*/
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inline short apply(const short *src) const {
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inline short apply(const short *src, size_t stride = 1) const {
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#ifdef __APPLE__
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short result;
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vDSP_dotpr_s1_15(filter_coefficients_.data(), 1, src, 1, &result, filter_coefficients_.size());
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vDSP_dotpr_s1_15(filter_coefficients_.data(), 1, src, stride, &result, filter_coefficients_.size());
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return result;
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#else
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int outputValue = 0;
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for(std::size_t c = 0; c < filter_coefficients_.size(); ++c) {
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outputValue += filter_coefficients_[c] * src[c];
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outputValue += filter_coefficients_[c] * src[c * stride];
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}
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return static_cast<short>(outputValue >> FixedShift);
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#endif
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