Aegisub/aegisub/audio_spectrum.cpp

334 lines
10 KiB
C++

// 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:
//
// * Redistributions of source code must retain the above copyright notice,
// 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.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// -----------------------------------------------------------------------------
//
// 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;
}