Aegisub/aegisub/audio_spectrum.cpp

// Copyright (c) 2005, 2006, Rodrigo Braz Monteiro
// Copyright (c) 2006, 2007, Niels Martin Hansen
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
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//     this list of conditions and the following disclaimer.
//   * Redistributions in binary form must reproduce the above copyright notice,
//     this list of conditions and the following disclaimer in the documentation
//     and/or other materials provided with the distribution.
//   * Neither the name of the Aegisub Group nor the names of its contributors
//     may be used to endorse or promote products derived from this software
//     without specific prior written permission.
//
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// -----------------------------------------------------------------------------
//
// AEGISUB
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:zeratul@cellosoft.com
//

#include <assert.h>
#include "audio_spectrum.h"
#include "fft.h"
#include "colorspace.h"
#include "options.h"


// Audio spectrum FFT data cache

AudioSpectrumCache::CacheLine AudioSpectrumCache::null_line;
unsigned long AudioSpectrumCache::line_length;

void AudioSpectrumCache::SetLineLength(unsigned long new_length)
{
	line_length = new_length;
	null_line.resize(new_length, 0);
}


// Bottom level FFT cache, holds actual power data itself

class FinalSpectrumCache : public AudioSpectrumCache {
private:
	std::vector<CacheLine> data;
	unsigned long start, length; // start and end of range

public:
	CacheLine& GetLine(unsigned long i)
	{
		// This check ought to be redundant
		if (i >= start && i-start < length)
			return data[i - start];
		else
			return null_line;
	}

	FinalSpectrumCache(AudioProvider *provider, unsigned long _start, unsigned long _length)
	{
		start = _start;
		length = _length;

		assert(length > 2);

		// First fill the data vector with blanks
		// Both start and end are included in the range stored, so we have end-start+1 elements
		data.resize(length, null_line);

		// Start sample number of the next line calculated
		// line_length is half of the number of samples used to calculate a line, since half of the output from
		// a Fourier transform of real data is redundant, and not interesting for the purpose of creating
		// a frequenmcy/power spectrum.
		__int64 sample = start * line_length*2;

		// Raw sample data
		short *raw_sample_data = new short[line_length*2];
		float *sample_data = new float[line_length*2];
		// Real and imaginary components of the output
		float *out_r = new float[line_length*2];
		float *out_i = new float[line_length*2];

		FFT fft; // TODO: use FFTW instead? A wavelet?

		for (unsigned long i = 0; i < length; ++i) {
			provider->GetAudio(raw_sample_data, sample, line_length*2);
			for (size_t j = 0; j < line_length; ++j) {
				sample_data[j*2] = (float)raw_sample_data[j*2];
				sample_data[j*2+1] = (float)raw_sample_data[j*2+1];
			}

			fft.Transform(line_length*2, sample_data, out_r, out_i);

			CacheLine &line = data[i];
			for (size_t j = 0; j < line_length; ++j) {
				line[j] = sqrt(out_r[j]*out_r[j] + out_i[j]*out_i[j]);
			}

			sample += line_length*2;
		}

		delete[] raw_sample_data;
		delete[] sample_data;
		delete[] out_r;
		delete[] out_i;
	}

	virtual ~FinalSpectrumCache()
	{
	}

};


// Non-bottom-level cache, refers to other caches to do the work

class IntermediateSpectrumCache : public AudioSpectrumCache {
private:
	std::vector<AudioSpectrumCache*> sub_caches;
	unsigned long start, length, subcache_length;
	bool subcaches_are_final;
	int depth;
	AudioProvider *provider;

public:
	CacheLine &GetLine(unsigned long i)
	{
		if (i >= start && i-start <= length) {
			// Determine which sub-cache this line resides in
			size_t subcache = (i-start) / subcache_length;
			assert(subcache >= 0 && subcache < sub_caches.size());

			if (!sub_caches[subcache]) {
				if (subcaches_are_final) {
					sub_caches[subcache] = new FinalSpectrumCache(provider, start+subcache*subcache_length, subcache_length);
				} else {
					sub_caches[subcache] = new IntermediateSpectrumCache(provider, start+subcache*subcache_length, subcache_length, depth+1);
				}
			}

			return sub_caches[subcache]->GetLine(i);
		} else {
			return null_line;
		}
	}

	IntermediateSpectrumCache(AudioProvider *_provider, unsigned long _start, unsigned long _length, int _depth)
	{
		provider = _provider;
		start = _start;
		length = _length;
		depth = _depth;

		// FIXME: this calculation probably needs tweaking
		int num_subcaches = 1;
		unsigned long tmp = length;
		while (tmp > 0) {
			tmp /= 16;
			num_subcaches *= 2;
		}
		subcache_length = length / (num_subcaches-1);

		subcaches_are_final = num_subcaches <= 4;

		sub_caches.resize(num_subcaches, 0);
	}

	virtual ~IntermediateSpectrumCache()
	{
		for (size_t i = 0; i < sub_caches.size(); ++i)
			if (sub_caches[i])
				delete sub_caches[i];
	}

};


// AudioSpectrum

AudioSpectrum::AudioSpectrum(AudioProvider *_provider, unsigned long _line_length)
{
	provider = _provider;
	line_length = _line_length;

	__int64 _num_lines = provider->GetNumSamples() / line_length / 2;
	//assert (_num_lines < (1<<31)); // hope it fits into 32 bits...
	num_lines = (unsigned long)_num_lines;

	AudioSpectrumCache::SetLineLength(line_length);
	cache = new IntermediateSpectrumCache(provider, 0, num_lines, 0);

	power_scale = 1;
	minband = Options.AsInt(_T("Audio Spectrum Cutoff"));
	maxband = line_length - minband * 2/3; // TODO: make this customisable?

	// Generate colour maps
	unsigned char *palptr = colours_normal;
	for (int i = 0; i < 256; i++) {
		hsl_to_rgb(170 + i * 2/3, 128 + i/2, i, palptr+0, palptr+1, palptr+2);
		palptr += 3;
	}
	palptr = colours_selected;
	for (int i = 0; i < 256; i++) {
		hsl_to_rgb(170 + i * 2/3, 128 + i/2, i*3/4+64, palptr+0, palptr+1, palptr+2);
		palptr += 3;
	}
}


AudioSpectrum::~AudioSpectrum()
{
	delete cache;
}


void AudioSpectrum::RenderRange(__int64 range_start, __int64 range_end, bool selected, unsigned char *img, int imgleft, int imgwidth, int imgpitch, int imgheight)
{
	unsigned long first_line = (unsigned long)(range_start / line_length / 2);
	unsigned long last_line = (unsigned long)(range_end / line_length / 2);
	unsigned long lines_to_render = last_line - first_line + 1;

	float *power = new float[line_length];

	int last_imgcol_rendered = -1;

	unsigned char *palette;
	if (selected)
		palette = colours_selected;
	else
		palette = colours_normal;

	// Some scaling constants
	const int maxpower = (1 << (16 - 1))*256;

	const double upscale = power_scale * 16384 / line_length;
	const double onethirdmaxpower = maxpower / 3, twothirdmaxpower = maxpower * 2/3;
	const double logoverscale = log(maxpower*upscale - twothirdmaxpower);

	for (unsigned long i = first_line; i <= last_line; ++i) {
		// Handle horizontal compression and don't unneededly re-render columns
		int imgcol = imgleft + imgwidth * (i - first_line) / (last_line - first_line + 1);
		if (imgcol <= last_imgcol_rendered)
			continue;

		AudioSpectrumCache::CacheLine &line = cache->GetLine(i);

		// Calculate the signal power over frequency
		// "Compressed" scale
		for (unsigned int j = 0; j < line_length; j++) {
			// First do a simple linear scale power calculation -- 8 gives a reasonable default scaling
			power[j] = line[j] * upscale;
			if (power[j] > maxpower * 2/3) {
				double p = power[j] - twothirdmaxpower;
				p = log(p) * onethirdmaxpower / logoverscale;
				power[j] = p + twothirdmaxpower;
			}
		}

#define WRITE_PIXEL \
	if (intensity < 0) intensity = 0; \
	if (intensity > 255) intensity = 255; \
	img[((imgheight-y-1)*imgpitch+x)*3 + 0] = palette[intensity*3+0]; \
	img[((imgheight-y-1)*imgpitch+x)*3 + 1] = palette[intensity*3+1]; \
	img[((imgheight-y-1)*imgpitch+x)*3 + 2] = palette[intensity*3+2];

		// Handle horizontal expansion
		int next_line_imgcol = imgleft + imgwidth * (i - first_line + 1) / (last_line - first_line + 1);
		if (next_line_imgcol >= imgpitch)
			next_line_imgcol = imgpitch-1;

		for (int x = imgcol; x <= next_line_imgcol; ++x) {

			// Decide which rendering algo to use
			if (maxband - minband > imgheight) {
				// more than one frequency sample per pixel (vertically compress data)
				// pick the largest value per pixel for display

				// Iterate over pixels, picking a range of samples for each
				for (int y = 0; y < imgheight; ++y) {
					int sample1 = maxband * y/imgheight + minband;
					int sample2 = maxband * (y+1)/imgheight + minband;
					float maxval = 0;
					for (int samp = sample1; samp <= sample2; samp++) {
						if (power[samp] > maxval) maxval = power[samp];
					}
					int intensity = int(256 * maxval / maxpower);
					WRITE_PIXEL
				}
			}
			else {
				// less than one frequency sample per pixel (vertically expand data)
				// interpolate between pixels
				// can also happen with exactly one sample per pixel, but how often is that?

				// Iterate over pixels, picking the nearest power values
				for (int y = 0; y < imgheight; ++y) {
					float ideal = (float)(y+1.)/imgheight * maxband;
					float sample1 = power[(int)floor(ideal)+minband];
					float sample2 = power[(int)ceil(ideal)+minband];
					float frac = ideal - floor(ideal);
					int intensity = int(((1-frac)*sample1 + frac*sample2) / maxpower * 256);
					WRITE_PIXEL
				}
			}
		}

#undef WRITE_PIXEL

	}

	delete[] power;
}


void AudioSpectrum::SetScaling(float _power_scale)
{
	power_scale = _power_scale;
}