1 files changed, 249 insertions, 0 deletions
diff --git a/servers/audio/effects/audio_effect_spectrum_analyzer.cpp b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp
new file mode 100644
index 0000000000..71a8904472
--- /dev/null
+++ b/servers/audio/effects/audio_effect_spectrum_analyzer.cpp
@@ -0,0 +1,249 @@
+#include "audio_effect_spectrum_analyzer.h"
+#include "servers/audio_server.h"
+
+static void smbFft(float *fftBuffer, long fftFrameSize, long sign)
+/*
+	FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
+	Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
+	time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
+	and returns the cosine and sine parts in an interleaved manner, ie.
+	fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
+	must be a power of 2. It expects a complex input signal (see footnote 2),
+	ie. when working with 'common' audio signals our input signal has to be
+	passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
+	of the frequencies of interest is in fftBuffer[0...fftFrameSize].
+*/
+{
+	float wr, wi, arg, *p1, *p2, temp;
+	float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
+	long i, bitm, j, le, le2, k;
+
+	for (i = 2; i < 2 * fftFrameSize - 2; i += 2) {
+		for (bitm = 2, j = 0; bitm < 2 * fftFrameSize; bitm <<= 1) {
+			if (i & bitm) j++;
+			j <<= 1;
+		}
+		if (i < j) {
+			p1 = fftBuffer + i;
+			p2 = fftBuffer + j;
+			temp = *p1;
+			*(p1++) = *p2;
+			*(p2++) = temp;
+			temp = *p1;
+			*p1 = *p2;
+			*p2 = temp;
+		}
+	}
+	for (k = 0, le = 2; k < (long)(log((double)fftFrameSize) / log(2.) + .5); k++) {
+		le <<= 1;
+		le2 = le >> 1;
+		ur = 1.0;
+		ui = 0.0;
+		arg = Math_PI / (le2 >> 1);
+		wr = cos(arg);
+		wi = sign * sin(arg);
+		for (j = 0; j < le2; j += 2) {
+			p1r = fftBuffer + j;
+			p1i = p1r + 1;
+			p2r = p1r + le2;
+			p2i = p2r + 1;
+			for (i = j; i < 2 * fftFrameSize; i += le) {
+				tr = *p2r * ur - *p2i * ui;
+				ti = *p2r * ui + *p2i * ur;
+				*p2r = *p1r - tr;
+				*p2i = *p1i - ti;
+				*p1r += tr;
+				*p1i += ti;
+				p1r += le;
+				p1i += le;
+				p2r += le;
+				p2i += le;
+			}
+			tr = ur * wr - ui * wi;
+			ui = ur * wi + ui * wr;
+			ur = tr;
+		}
+	}
+}
+void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
+
+	uint64_t time = OS::get_singleton()->get_ticks_usec();
+
+	//copy everything over first, since this only really does capture
+	for (int i = 0; i < p_frame_count; i++) {
+		p_dst_frames[i] = p_src_frames[i];
+	}
+
+	//capture spectrum
+	while (p_frame_count) {
+		int to_fill = fft_size * 2 - temporal_fft_pos;
+		to_fill = MIN(to_fill, p_frame_count);
+
+		float *fftw = temporal_fft.ptrw();
+		for (int i = 0; i < to_fill; i++) { //left and right buffers
+			fftw[(i + temporal_fft_pos) * 2] = p_src_frames[i].l;
+			fftw[(i + temporal_fft_pos) * 2 + 1] = 0;
+			fftw[(i + temporal_fft_pos + fft_size * 2) * 2] = p_src_frames[i].r;
+			fftw[(i + temporal_fft_pos + fft_size * 2) * 2 + 1] = 0;
+		}
+
+		p_src_frames += to_fill;
+		temporal_fft_pos += to_fill;
+		p_frame_count -= to_fill;
+
+		if (temporal_fft_pos == fft_size * 2) {
+			//time to do a FFT
+			smbFft(fftw, fft_size * 2, -1);
+			smbFft(fftw + fft_size * 4, fft_size * 2, -1);
+			int next = (fft_pos + 1) % fft_count;
+
+			AudioFrame *hw = (AudioFrame *)fft_history[next].ptr(); //do not use write, avoid cow
+
+			for (int i = 0; i < fft_size; i++) {
+				//abs(vec)/fft_size normalizes each frequency
+				float window = 1.0; //-.5 * Math::cos(2. * Math_PI * (double)i / (double)fft_size) + .5;
+				hw[i].l = window * Vector2(fftw[i * 2], fftw[i * 2 + 1]).length() / float(fft_size);
+				hw[i].r = window * Vector2(fftw[fft_size * 4 + i * 2], fftw[fft_size * 4 + i * 2 + 1]).length() / float(fft_size);
+			}
+
+			fft_pos = next; //swap
+			temporal_fft_pos = 0;
+		}
+	}
+
+	//determine time of capture
+	double remainer_sec = (temporal_fft_pos / mix_rate); //substract remainder from mix time
+	last_fft_time = time - uint64_t(remainer_sec * 1000000.0);
+}
+
+void AudioEffectSpectrumAnalyzerInstance::_bind_methods() {
+
+	ClassDB::bind_method(D_METHOD("get_magnitude_for_frequency_range", "from_hz", "to_hz", "mode"), &AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range, DEFVAL(MAGNITUDE_MAX));
+	BIND_ENUM_CONSTANT(MAGNITUDE_AVERAGE);
+	BIND_ENUM_CONSTANT(MAGNITUDE_MAX);
+}
+
+Vector2 AudioEffectSpectrumAnalyzerInstance::get_magnitude_for_frequency_range(float p_begin, float p_end, MagnitudeMode p_mode) const {
+
+	if (last_fft_time == 0) {
+		return Vector2();
+	}
+	uint64_t time = OS::get_singleton()->get_ticks_usec();
+	float diff = double(time - last_fft_time) / 1000000.0 + base->get_tap_back_pos();
+	diff -= AudioServer::get_singleton()->get_output_delay();
+	float fft_time_size = float(fft_size) / mix_rate;
+
+	int fft_index = fft_pos;
+
+	while (diff > fft_time_size) {
+		diff -= fft_time_size;
+		fft_index -= 1;
+		if (fft_index < 0) {
+			fft_index = fft_count - 1;
+		}
+	}
+
+	int begin_pos = p_begin * fft_size / (mix_rate * 0.5);
+	int end_pos = p_end * fft_size / (mix_rate * 0.5);
+
+	begin_pos = CLAMP(begin_pos, 0, fft_size - 1);
+	end_pos = CLAMP(end_pos, 0, fft_size - 1);
+
+	if (begin_pos > end_pos) {
+		SWAP(begin_pos, end_pos);
+	}
+	const AudioFrame *r = fft_history[fft_index].ptr();
+
+	if (p_mode == MAGNITUDE_AVERAGE) {
+		Vector2 avg;
+
+		for (int i = begin_pos; i <= end_pos; i++) {
+			avg += Vector2(r[i]);
+		}
+
+		avg /= float(end_pos - begin_pos + 1);
+
+		return avg;
+	} else {
+
+		Vector2 max;
+
+		for (int i = begin_pos; i <= end_pos; i++) {
+			max.x = MAX(max.x, r[i].l);
+			max.y = MAX(max.x, r[i].r);
+		}
+
+		return max;
+	}
+}
+
+Ref<AudioEffectInstance> AudioEffectSpectrumAnalyzer::instance() {
+
+	Ref<AudioEffectSpectrumAnalyzerInstance> ins;
+	ins.instance();
+	ins->base = Ref<AudioEffectSpectrumAnalyzer>(this);
+	static const int fft_sizes[FFT_SIZE_MAX] = { 256, 512, 1024, 2048, 4096 };
+	ins->fft_size = fft_sizes[fft_size];
+	ins->mix_rate = AudioServer::get_singleton()->get_mix_rate();
+	ins->fft_count = (buffer_length / (float(ins->fft_size) / ins->mix_rate)) + 1;
+	ins->fft_pos = 0;
+	ins->last_fft_time = 0;
+	ins->fft_history.resize(ins->fft_count);
+	ins->temporal_fft.resize(ins->fft_size * 8); //x2 stereo, x2 amount of samples for freqs, x2 for input
+	ins->temporal_fft_pos = 0;
+	for (int i = 0; i < ins->fft_count; i++) {
+		ins->fft_history.write[i].resize(ins->fft_size); //only magnitude matters
+		for (int j = 0; j < ins->fft_size; j++) {
+			ins->fft_history.write[i].write[j] = AudioFrame(0, 0);
+		}
+	}
+	return ins;
+}
+
+void AudioEffectSpectrumAnalyzer::set_buffer_length(float p_volume) {
+	buffer_length = p_volume;
+}
+
+float AudioEffectSpectrumAnalyzer::get_buffer_length() const {
+
+	return buffer_length;
+}
+
+void AudioEffectSpectrumAnalyzer::set_tap_back_pos(float p_seconds) {
+	tapback_pos = p_seconds;
+}
+
+float AudioEffectSpectrumAnalyzer::get_tap_back_pos() const {
+	return tapback_pos;
+}
+
+void AudioEffectSpectrumAnalyzer::set_fft_size(FFT_Size p_fft_size) {
+	ERR_FAIL_INDEX(p_fft_size, FFT_SIZE_MAX);
+	fft_size = p_fft_size;
+}
+
+AudioEffectSpectrumAnalyzer::FFT_Size AudioEffectSpectrumAnalyzer::get_fft_size() const {
+	return fft_size;
+}
+
+void AudioEffectSpectrumAnalyzer::_bind_methods() {
+
+	ClassDB::bind_method(D_METHOD("set_buffer_length", "seconds"), &AudioEffectSpectrumAnalyzer::set_buffer_length);
+	ClassDB::bind_method(D_METHOD("get_buffer_length"), &AudioEffectSpectrumAnalyzer::get_buffer_length);
+
+	ClassDB::bind_method(D_METHOD("set_tap_back_pos", "seconds"), &AudioEffectSpectrumAnalyzer::set_tap_back_pos);
+	ClassDB::bind_method(D_METHOD("get_tap_back_pos"), &AudioEffectSpectrumAnalyzer::get_tap_back_pos);
+
+	ClassDB::bind_method(D_METHOD("set_fft_size", "size"), &AudioEffectSpectrumAnalyzer::set_fft_size);
+	ClassDB::bind_method(D_METHOD("get_fft_size"), &AudioEffectSpectrumAnalyzer::get_fft_size);
+
+	ADD_PROPERTY(PropertyInfo(Variant::REAL, "buffer_length", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_buffer_length", "get_buffer_length");
+	ADD_PROPERTY(PropertyInfo(Variant::REAL, "tap_back_pos", PROPERTY_HINT_RANGE, "0.1,4,0.1"), "set_tap_back_pos", "get_tap_back_pos");
+	ADD_PROPERTY(PropertyInfo(Variant::INT, "fft_size", PROPERTY_HINT_ENUM, "256,512,1024,2048,4096"), "set_fft_size", "get_fft_size");
+}
+
+AudioEffectSpectrumAnalyzer::AudioEffectSpectrumAnalyzer() {
+	buffer_length = 2;
+	tapback_pos = 0.01;
+	fft_size = FFT_SIZE_1024;
+}