Aegisub/aegisub/audio_spectrum.cpp

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// 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 <vector>
#include <list>
#include <utility>
#include <algorithm>
#ifdef _OPENMP
#include <omp.h>
#endif
#include "audio_spectrum.h"
#include "fft.h"
#include "colorspace.h"
#include "options.h"
#include "utils.h"
#include <wx/log.h>
// Audio spectrum FFT data cache
// Spectrum cache basically caches the raw result of FFT
class AudioSpectrumCache {
public:
// Type of a single FFT result line
typedef std::vector<float> CacheLine;
// Types for cache aging
typedef unsigned int CacheAccessTime;
struct CacheAgeData {
CacheAccessTime access_time;
unsigned long first_line;
unsigned long num_lines; // includes overlap-lines
bool operator< (const CacheAgeData& second) const { return access_time < second.access_time; }
CacheAgeData(CacheAccessTime t, unsigned long first, unsigned long num) : access_time(t), first_line(first), num_lines(num) { }
};
typedef std::vector<CacheAgeData> CacheAgeList;
// Get the overlap'th overlapping FFT in FFT group i, generating it if needed
virtual CacheLine& GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time) = 0;
// Get the total number of cache lines currently stored in this cache node's sub tree
virtual size_t GetManagedLineCount() = 0;
// Append to a list of last access times to the cache
virtual void GetLineAccessTimes(CacheAgeList &ages) = 0;
// Delete the cache storage starting with the given line id
// Return true if the object called on is empty and can safely be deleted too
virtual bool KillLine(unsigned long line_id) = 0;
// Set the FFT size used
static void SetLineLength(unsigned long new_length)
{
line_length = new_length;
null_line.resize(new_length, 0);
}
virtual ~AudioSpectrumCache() {};
protected:
// A cache line containing only zero-values
static CacheLine null_line;
// The FFT size
static unsigned long line_length;
};
AudioSpectrumCache::CacheLine AudioSpectrumCache::null_line;
unsigned long AudioSpectrumCache::line_length;
// 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
unsigned int overlaps;
CacheAccessTime last_access;
public:
CacheLine& GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time)
{
last_access = access_time;
// This check ought to be redundant
if (i >= start && i-start < length)
return data[i - start + overlap*length];
else
return null_line;
}
size_t GetManagedLineCount()
{
return data.size();
}
void GetLineAccessTimes(CacheAgeList &ages)
{
ages.push_back(CacheAgeData(last_access, start, data.size()));
}
bool KillLine(unsigned long line_id)
{
return start == line_id;
}
FinalSpectrumCache(AudioProvider *provider, unsigned long _start, unsigned long _length, unsigned int _overlaps)
{
start = _start;
length = _length;
overlaps = _overlaps;
if (overlaps < 1) overlaps = 1;
// Add an upper limit to number of overlaps or trust user to do sane things?
// Any limit should probably be a function of 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*overlaps, null_line);
unsigned int overlap_offset = line_length / overlaps * 2; // FIXME: the result seems weird/wrong without this factor 2, but why?
// Raw sample data
short *raw_sample_data = NULL;
float *sample_data = NULL;
// Real and imaginary components of the output
float *out_r = NULL;
float *out_i = NULL;
FFT fft; // Use FFTW instead? A wavelet?
for (unsigned int overlap = 0; overlap < overlaps; ++overlap) {
// 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_t sample = start * line_length*2 + overlap*overlap_offset;
long len = length;
#ifdef _OPENMP
#pragma omp parallel shared(overlap,len,line_length)
#endif
{
short *raw_sample_data = new short[line_length*2];
float *sample_data = new float[line_length*2];
float *out_r = new float[line_length*2];
float *out_i = new float[line_length*2];
#ifdef _OPENMP
#pragma omp for
#endif
for (long i = 0; i < len; ++i) {
// Initialize
sample = start * line_length*2 + overlap*overlap_offset + i*line_length*2;
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 + length*overlap];
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;
unsigned int overlaps;
bool subcaches_are_final;
int depth;
AudioProvider *provider;
public:
CacheLine &GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time)
{
if (i >= start && i-start <= length) {
// Determine which sub-cache this line resides in
size_t subcache = (i-start) / subcache_length;
assert(subcache < sub_caches.size());
if (!sub_caches[subcache]) {
created = true;
if (subcaches_are_final) {
sub_caches[subcache] = new FinalSpectrumCache(provider, start+subcache*subcache_length, subcache_length, overlaps);
} else {
sub_caches[subcache] = new IntermediateSpectrumCache(provider, start+subcache*subcache_length, subcache_length, overlaps, depth+1);
}
}
return sub_caches[subcache]->GetLine(i, overlap, created, access_time);
} else {
return null_line;
}
}
size_t GetManagedLineCount()
{
size_t res = 0;
for (size_t i = 0; i < sub_caches.size(); ++i) {
if (sub_caches[i])
res += sub_caches[i]->GetManagedLineCount();
}
return res;
}
void GetLineAccessTimes(CacheAgeList &ages)
{
for (size_t i = 0; i < sub_caches.size(); ++i) {
if (sub_caches[i])
sub_caches[i]->GetLineAccessTimes(ages);
}
}
bool KillLine(unsigned long line_id)
{
int sub_caches_left = 0;
for (size_t i = 0; i < sub_caches.size(); ++i) {
if (sub_caches[i]) {
if (sub_caches[i]->KillLine(line_id)) {
delete sub_caches[i];
sub_caches[i] = 0;
} else {
sub_caches_left++;
}
}
}
return sub_caches_left == 0;
}
IntermediateSpectrumCache(AudioProvider *_provider, unsigned long _start, unsigned long _length, unsigned int _overlaps, int _depth)
{
provider = _provider;
start = _start;
length = _length;
overlaps = _overlaps;
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];
}
};
class AudioSpectrumCacheManager {
private:
IntermediateSpectrumCache *cache_root;
unsigned long cache_hits, cache_misses;
AudioSpectrumCache::CacheAccessTime cur_time;
unsigned long max_lines_cached;
public:
AudioSpectrumCache::CacheLine &GetLine(unsigned long i, unsigned int overlap)
{
bool created = false;
AudioSpectrumCache::CacheLine &res = cache_root->GetLine(i, overlap, created, cur_time++);
if (created)
cache_misses++;
else
cache_hits++;
return res;
}
void Age()
{
wxLogDebug(_T("AudioSpectrumCacheManager stats: hits=%u, misses=%u, misses%%=%f, managed lines=%u (max=%u)"), cache_hits, cache_misses, cache_misses/float(cache_hits+cache_misses)*100, cache_root->GetManagedLineCount(), max_lines_cached);
// 0 means no limit
if (max_lines_cached == 0)
return;
// No reason to proceed with complicated stuff if the count is too small
// (FIXME: does this really pay off?)
if (cache_root->GetManagedLineCount() < max_lines_cached)
return;
// Get and sort ages
AudioSpectrumCache::CacheAgeList ages;
cache_root->GetLineAccessTimes(ages);
std::sort(ages.begin(), ages.end());
// Number of lines we have found used so far
// When this exceeds max_lines_caches go into kill-mode
unsigned long cumulative_lines = 0;
// Run backwards through the line age list (the most recently accessed items are at end)
AudioSpectrumCache::CacheAgeList::reverse_iterator it = ages.rbegin();
// Find the point where we have too many lines cached
while (cumulative_lines < max_lines_cached) {
if (it == ages.rend()) {
wxLogDebug(_T("AudioSpectrumCacheManager done aging did not exceed max_lines_cached"));
return;
}
cumulative_lines += it->num_lines;
++it;
}
// By here, we have exceeded max_lines_cached so backtrack one
--it;
// And now start cleaning up
for (; it != ages.rend(); ++it) {
cache_root->KillLine(it->first_line);
}
wxLogDebug(_T("AudioSpectrumCacheManager done aging, managed lines now=%u (max=%u)"), cache_root->GetManagedLineCount(), max_lines_cached);
assert(cache_root->GetManagedLineCount() < max_lines_cached);
}
AudioSpectrumCacheManager(AudioProvider *provider, unsigned long line_length, unsigned long num_lines, unsigned int num_overlaps)
{
cache_hits = cache_misses = 0;
cur_time = 0;
cache_root = new IntermediateSpectrumCache(provider, 0, num_lines, num_overlaps, 0);
// option is stored in megabytes, but we want number of bytes
unsigned long max_cache_size = Options.AsInt(_T("Audio Spectrum Memory Max"));
// It can't go too low
if (max_cache_size < 5) max_cache_size = 128;
max_cache_size *= 1024 * 1024;
unsigned long line_size = sizeof(AudioSpectrumCache::CacheLine::value_type) * line_length;
max_lines_cached = max_cache_size / line_size;
}
~AudioSpectrumCacheManager()
{
delete cache_root;
}
};
// AudioSpectrum
AudioSpectrum::AudioSpectrum(AudioProvider *_provider)
{
provider = _provider;
// Determine the quality of the spectrum rendering based on an index
int quality_index = Options.AsInt(_T("Audio Spectrum Quality"));
if (quality_index < 0) quality_index = 0;
if (quality_index > 5) quality_index = 5; // no need to go freaking insane
// Line length determines the balance between resolution in the time and frequency domains.
// Larger line length gives better resolution in frequency domain,
// smaller gives better resolution in time domain.
// Any values uses the same amount of memory, but larger values takes (slightly) more CPU.
// Line lengths must be powers of 2 due to the FFT algorithm.
// 2^8 is a good compromise between time and frequency domain resolution, any smaller
// gives an unreasonably low resolution in the frequency domain.
// Increasing the number of overlaps gives better resolution in the time domain.
// Doubling the number of overlaps doubles memory and CPU use, and also
// doubles resolution in the time domain.
switch (quality_index) {
case 0:
// No overlaps, good comprimise between time/frequency resolution.
// 4 bytes used per sample.
line_length = 1<<8;
fft_overlaps = 1;
break;
case 1:
// Double frequency resolution, the resulting half time resolution
// is countered with an overlap.
// 8 bytes per sample.
line_length = 1<<9;
fft_overlaps = 2;
break;
case 2:
// Resulting double resolution in both domains.
// 16 bytes per sample.
line_length = 1<<9;
fft_overlaps = 4;
break;
case 3:
// Double frequency and quadrouble time resolution.
// 32 bytes per sample.
line_length = 1<<9;
fft_overlaps = 8;
break;
case 4:
// Quadrouble resolution in both domains.
// 64 bytes per sample.
line_length = 1<<10;
fft_overlaps = 16;
break;
case 5:
// Eight-double resolution in both domains.
// 256 bytes per sample.
line_length = 1<<11;
fft_overlaps = 64;
break;
default:
throw _T("Internal error in AudioSpectrum class - impossible quality index");
}
int64_t _num_lines = provider->GetNumSamples() / line_length / 2;
num_lines = (unsigned long)_num_lines;
AudioSpectrumCache::SetLineLength(line_length);
cache = new AudioSpectrumCacheManager(provider, line_length, num_lines, fft_overlaps);
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); // Previous
hsl_to_rgb((255+128-i)/2, 128 + i/2, MIN(255,2*i), palptr+0, palptr+1, palptr+2); // Icy blue
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);
hsl_to_rgb((255+128-i)/2, 128 + i/2, MIN(255,3*i/2+64), palptr+0, palptr+1, palptr+2); // Icy blue
palptr += 3;
}
}
AudioSpectrum::~AudioSpectrum()
{
delete cache;
}
void AudioSpectrum::RenderRange(int64_t range_start, int64_t range_end, bool selected, unsigned char *img, int imgleft, int imgwidth, int imgpitch, int imgheight)
{
unsigned long first_line = (unsigned long)(fft_overlaps * range_start / line_length / 2);
unsigned long last_line = (unsigned long)(fft_overlaps * range_end / line_length / 2);
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);
// Note that here "lines" are actually bands of power data
unsigned long baseline = first_line / fft_overlaps;
unsigned int overlap = first_line % fft_overlaps;
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(baseline, overlap);
++overlap;
if (overlap >= fft_overlaps) {
overlap = 0;
++baseline;
}
// Apply a "compressed" scaling to the signal power
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;
cache->Age();
}
void AudioSpectrum::SetScaling(float _power_scale)
{
power_scale = _power_scale;
}