552 lines
17 KiB
C++
552 lines
17 KiB
C++
// Copyright (c) 2005, 2006, Rodrigo Braz Monteiro
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// Copyright (c) 2006, 2007, Niels Martin Hansen
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// All rights reserved.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// -----------------------------------------------------------------------------
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//
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// AEGISUB
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//
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// Website: http://aegisub.cellosoft.com
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// Contact: mailto:zeratul@cellosoft.com
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//
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#include <assert.h>
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#include <vector>
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#include <list>
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#include <utility>
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#include <algorithm>
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#include "audio_spectrum.h"
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#include "fft.h"
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#include "colorspace.h"
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#include "options.h"
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#include "utils.h"
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// Audio spectrum FFT data cache
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// Spectrum cache basically caches the raw result of FFT
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class AudioSpectrumCache {
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public:
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// Type of a single FFT result line
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typedef std::vector<float> CacheLine;
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// Types for cache aging
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typedef unsigned int CacheAccessTime;
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struct CacheAgeData {
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CacheAccessTime access_time;
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unsigned long first_line;
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unsigned long num_lines; // includes overlap-lines
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bool operator< (const CacheAgeData& second) { return access_time < second.access_time; }
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CacheAgeData(CacheAccessTime t, unsigned long first, unsigned long num) : access_time(t), first_line(first), num_lines(num) { }
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};
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typedef std::vector<CacheAgeData> CacheAgeList;
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// Get the overlap'th overlapping FFT in FFT group i, generating it if needed
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virtual CacheLine& GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time) = 0;
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// Get the total number of cache lines currently stored in this cache node's sub tree
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virtual size_t GetManagedLineCount() = 0;
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// Append to a list of last access times to the cache
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virtual void GetLineAccessTimes(CacheAgeList &ages) = 0;
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// Delete the cache storage starting with the given line id
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// Return true if the object called on is empty and can safely be deleted too
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virtual bool KillLine(unsigned long line_id) = 0;
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// Set the FFT size used
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static void SetLineLength(unsigned long new_length)
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{
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line_length = new_length;
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null_line.resize(new_length, 0);
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}
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virtual ~AudioSpectrumCache() {};
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protected:
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// A cache line containing only zero-values
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static CacheLine null_line;
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// The FFT size
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static unsigned long line_length;
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};
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AudioSpectrumCache::CacheLine AudioSpectrumCache::null_line;
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unsigned long AudioSpectrumCache::line_length;
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// Bottom level FFT cache, holds actual power data itself
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class FinalSpectrumCache : public AudioSpectrumCache {
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private:
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std::vector<CacheLine> data;
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unsigned long start, length; // start and end of range
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unsigned int overlaps;
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CacheAccessTime last_access;
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public:
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CacheLine& GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time)
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{
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last_access = access_time;
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// This check ought to be redundant
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if (i >= start && i-start < length)
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return data[i - start + overlap*length];
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else
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return null_line;
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}
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size_t GetManagedLineCount()
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{
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return data.size();
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}
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void GetLineAccessTimes(CacheAgeList &ages)
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{
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ages.push_back(CacheAgeData(last_access, start, data.size()));
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}
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bool KillLine(unsigned long line_id)
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{
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return start == line_id;
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}
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FinalSpectrumCache(AudioProvider *provider, unsigned long _start, unsigned long _length, unsigned int _overlaps)
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{
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start = _start;
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length = _length;
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overlaps = _overlaps;
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if (overlaps < 1) overlaps = 1;
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// Add an upper limit to number of overlaps or trust user to do sane things?
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// Any limit should probably be a function of length
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assert(length > 2);
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// First fill the data vector with blanks
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// Both start and end are included in the range stored, so we have end-start+1 elements
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data.resize(length*overlaps, null_line);
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unsigned int overlap_offset = line_length / overlaps * 2; // FIXME: the result seems weird/wrong without this factor 2, but why?
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// Raw sample data
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short *raw_sample_data = new short[line_length*2];
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float *sample_data = new float[line_length*2];
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// Real and imaginary components of the output
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float *out_r = new float[line_length*2];
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float *out_i = new float[line_length*2];
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FFT fft; // Use FFTW instead? A wavelet?
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for (unsigned int overlap = 0; overlap < overlaps; ++overlap) {
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// Start sample number of the next line calculated
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// line_length is half of the number of samples used to calculate a line, since half of the output from
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// a Fourier transform of real data is redundant, and not interesting for the purpose of creating
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// a frequenmcy/power spectrum.
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long long sample = start * line_length*2 + overlap*overlap_offset;
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for (unsigned long i = 0; i < length; ++i) {
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provider->GetAudio(raw_sample_data, sample, line_length*2);
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for (size_t j = 0; j < line_length; ++j) {
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sample_data[j*2] = (float)raw_sample_data[j*2];
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sample_data[j*2+1] = (float)raw_sample_data[j*2+1];
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}
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fft.Transform(line_length*2, sample_data, out_r, out_i);
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CacheLine &line = data[i + length*overlap];
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for (size_t j = 0; j < line_length; ++j) {
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line[j] = sqrt(out_r[j]*out_r[j] + out_i[j]*out_i[j]);
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}
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sample += line_length*2;
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}
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}
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delete[] raw_sample_data;
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delete[] sample_data;
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delete[] out_r;
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delete[] out_i;
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}
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virtual ~FinalSpectrumCache()
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{
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}
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};
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// Non-bottom-level cache, refers to other caches to do the work
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class IntermediateSpectrumCache : public AudioSpectrumCache {
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private:
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std::vector<AudioSpectrumCache*> sub_caches;
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unsigned long start, length, subcache_length;
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unsigned int overlaps;
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bool subcaches_are_final;
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int depth;
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AudioProvider *provider;
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public:
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CacheLine &GetLine(unsigned long i, unsigned int overlap, bool &created, CacheAccessTime access_time)
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{
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if (i >= start && i-start <= length) {
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// Determine which sub-cache this line resides in
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size_t subcache = (i-start) / subcache_length;
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assert(subcache < sub_caches.size());
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if (!sub_caches[subcache]) {
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created = true;
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if (subcaches_are_final) {
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sub_caches[subcache] = new FinalSpectrumCache(provider, start+subcache*subcache_length, subcache_length, overlaps);
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} else {
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sub_caches[subcache] = new IntermediateSpectrumCache(provider, start+subcache*subcache_length, subcache_length, overlaps, depth+1);
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}
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}
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return sub_caches[subcache]->GetLine(i, overlap, created, access_time);
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} else {
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return null_line;
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}
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}
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size_t GetManagedLineCount()
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{
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size_t res = 0;
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for (size_t i = 0; i < sub_caches.size(); ++i) {
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if (sub_caches[i])
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res += sub_caches[i]->GetManagedLineCount();
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}
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return res;
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}
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void GetLineAccessTimes(CacheAgeList &ages)
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{
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for (size_t i = 0; i < sub_caches.size(); ++i) {
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if (sub_caches[i])
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sub_caches[i]->GetLineAccessTimes(ages);
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}
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}
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bool KillLine(unsigned long line_id)
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{
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int sub_caches_left = 0;
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for (size_t i = 0; i < sub_caches.size(); ++i) {
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if (sub_caches[i]) {
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if (sub_caches[i]->KillLine(line_id)) {
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delete sub_caches[i];
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sub_caches[i] = 0;
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} else {
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sub_caches_left++;
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}
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}
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}
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return sub_caches_left == 0;
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}
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IntermediateSpectrumCache(AudioProvider *_provider, unsigned long _start, unsigned long _length, unsigned int _overlaps, int _depth)
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{
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provider = _provider;
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start = _start;
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length = _length;
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overlaps = _overlaps;
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depth = _depth;
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// FIXME: this calculation probably needs tweaking
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int num_subcaches = 1;
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unsigned long tmp = length;
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while (tmp > 0) {
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tmp /= 16;
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num_subcaches *= 2;
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}
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subcache_length = length / (num_subcaches-1);
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subcaches_are_final = num_subcaches <= 4;
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sub_caches.resize(num_subcaches, 0);
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}
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virtual ~IntermediateSpectrumCache()
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{
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for (size_t i = 0; i < sub_caches.size(); ++i)
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if (sub_caches[i])
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delete sub_caches[i];
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}
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};
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class AudioSpectrumCacheManager {
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private:
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IntermediateSpectrumCache *cache_root;
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unsigned long cache_hits, cache_misses;
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AudioSpectrumCache::CacheAccessTime cur_time;
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unsigned long max_lines_cached;
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public:
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AudioSpectrumCache::CacheLine &GetLine(unsigned long i, unsigned int overlap)
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{
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bool created = false;
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AudioSpectrumCache::CacheLine &res = cache_root->GetLine(i, overlap, created, cur_time++);
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if (created)
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cache_misses++;
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else
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cache_hits++;
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return res;
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}
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void Age()
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{
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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);
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// 0 means no limit
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if (max_lines_cached == 0)
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return;
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// No reason to proceed with complicated stuff if the count is too small
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// (FIXME: does this really pay off?)
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if (cache_root->GetManagedLineCount() < max_lines_cached)
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return;
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// Get and sort ages
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AudioSpectrumCache::CacheAgeList ages;
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cache_root->GetLineAccessTimes(ages);
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std::sort(ages.begin(), ages.end());
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// Number of lines we have found used so far
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// When this exceeds max_lines_caches go into kill-mode
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unsigned long cumulative_lines = 0;
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// Run backwards through the line age list (the most recently accessed items are at end)
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AudioSpectrumCache::CacheAgeList::reverse_iterator it = ages.rbegin();
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// Find the point where we have too many lines cached
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while (cumulative_lines < max_lines_cached) {
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if (it == ages.rend()) {
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wxLogDebug(_T("AudioSpectrumCacheManager done aging did not exceed max_lines_cached"));
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return;
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}
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cumulative_lines += it->num_lines;
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++it;
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}
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// By here, we have exceeded max_lines_cached so backtrack one
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--it;
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// And now start cleaning up
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for (; it != ages.rend(); ++it) {
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cache_root->KillLine(it->first_line);
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}
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wxLogDebug(_T("AudioSpectrumCacheManager done aging, managed lines now=%u (max=%u)"), cache_root->GetManagedLineCount(), max_lines_cached);
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assert(cache_root->GetManagedLineCount() < max_lines_cached);
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}
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AudioSpectrumCacheManager(AudioProvider *provider, unsigned long line_length, unsigned long num_lines, unsigned int num_overlaps)
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{
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cache_hits = cache_misses = 0;
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cur_time = 0;
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cache_root = new IntermediateSpectrumCache(provider, 0, num_lines, num_overlaps, 0);
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// option is stored in megabytes, but we want number of bytes
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unsigned long max_cache_size = Options.AsInt(_T("Audio Spectrum Memory Max")) * 1024 * 1024;
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unsigned long line_size = sizeof(AudioSpectrumCache::CacheLine::value_type) * line_length;
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max_lines_cached = max_cache_size / line_size;
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}
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~AudioSpectrumCacheManager()
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{
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delete cache_root;
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}
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};
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// AudioSpectrum
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AudioSpectrum::AudioSpectrum(AudioProvider *_provider)
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{
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provider = _provider;
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int quality_index = Options.AsInt(_T("Audio Spectrum Quality"));
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if (quality_index < 0) quality_index = 0;
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if (quality_index > 5) quality_index = 5; // no need to go freaking insane
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if (quality_index > 1)
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line_length = 1 << (8 + quality_index - 1);
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else
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line_length = 1 << 8;
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if (quality_index == 0)
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fft_overlaps = 1;
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else if (quality_index == 1)
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fft_overlaps = 4;
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else
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fft_overlaps = 1 << quality_index;
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long long _num_lines = provider->GetNumSamples() / line_length / 2;
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//assert (_num_lines < (1<<31)); // hope it fits into 32 bits...
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num_lines = (unsigned long)_num_lines;
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AudioSpectrumCache::SetLineLength(line_length);
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cache = new AudioSpectrumCacheManager(provider, line_length, num_lines, fft_overlaps);
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power_scale = 1;
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minband = Options.AsInt(_T("Audio Spectrum Cutoff"));
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maxband = line_length - minband * 2/3; // TODO: make this customisable?
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// Generate colour maps
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unsigned char *palptr = colours_normal;
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for (int i = 0; i < 256; i++) {
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//hsl_to_rgb(170 + i * 2/3, 128 + i/2, i, palptr+0, palptr+1, palptr+2); // Previous
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hsl_to_rgb((255+128-i)/2, 128 + i/2, MIN(255,2*i), palptr+0, palptr+1, palptr+2); // Icy blue
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palptr += 3;
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}
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palptr = colours_selected;
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for (int i = 0; i < 256; i++) {
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//hsl_to_rgb(170 + i * 2/3, 128 + i/2, i*3/4+64, palptr+0, palptr+1, palptr+2);
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hsl_to_rgb((255+128-i)/2, 128 + i/2, MIN(255,3*i/2+64), palptr+0, palptr+1, palptr+2); // Icy blue
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palptr += 3;
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}
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}
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AudioSpectrum::~AudioSpectrum()
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{
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delete cache;
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}
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void AudioSpectrum::RenderRange(long long range_start, long long range_end, bool selected, unsigned char *img, int imgleft, int imgwidth, int imgpitch, int imgheight)
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{
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unsigned long first_line = (unsigned long)(fft_overlaps * range_start / line_length / 2);
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unsigned long last_line = (unsigned long)(fft_overlaps * range_end / line_length / 2);
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float *power = new float[line_length];
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int last_imgcol_rendered = -1;
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unsigned char *palette;
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if (selected)
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palette = colours_selected;
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else
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palette = colours_normal;
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// Some scaling constants
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const int maxpower = (1 << (16 - 1))*256;
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const double upscale = power_scale * 16384 / line_length;
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const double onethirdmaxpower = maxpower / 3, twothirdmaxpower = maxpower * 2/3;
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const double logoverscale = log(maxpower*upscale - twothirdmaxpower);
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// Note that here "lines" are actually bands of power data
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unsigned long baseline = first_line / fft_overlaps;
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unsigned int overlap = first_line % fft_overlaps;
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for (unsigned long i = first_line; i <= last_line; ++i) {
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// Handle horizontal compression and don't unneededly re-render columns
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int imgcol = imgleft + imgwidth * (i - first_line) / (last_line - first_line + 1);
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if (imgcol <= last_imgcol_rendered)
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continue;
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AudioSpectrumCache::CacheLine &line = cache->GetLine(baseline, overlap);
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++overlap;
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if (overlap >= fft_overlaps) {
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overlap = 0;
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++baseline;
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}
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// Apply a "compressed" scaling to the signal power
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for (unsigned int j = 0; j < line_length; j++) {
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// First do a simple linear scale power calculation -- 8 gives a reasonable default scaling
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power[j] = line[j] * upscale;
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if (power[j] > maxpower * 2/3) {
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double p = power[j] - twothirdmaxpower;
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p = log(p) * onethirdmaxpower / logoverscale;
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power[j] = p + twothirdmaxpower;
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}
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}
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#define WRITE_PIXEL \
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if (intensity < 0) intensity = 0; \
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if (intensity > 255) intensity = 255; \
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img[((imgheight-y-1)*imgpitch+x)*3 + 0] = palette[intensity*3+0]; \
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img[((imgheight-y-1)*imgpitch+x)*3 + 1] = palette[intensity*3+1]; \
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img[((imgheight-y-1)*imgpitch+x)*3 + 2] = palette[intensity*3+2];
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// Handle horizontal expansion
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int next_line_imgcol = imgleft + imgwidth * (i - first_line + 1) / (last_line - first_line + 1);
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if (next_line_imgcol >= imgpitch)
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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;
|
|
}
|
|
|