move export-analysis implementation into cc-file.

lib/libfftw3f.a(apiplan.o):apiplan.c:(.text+0x430): multiple definition of `fftwf_destroy_plan'

This is because static symbols in a .dll have no fixed address and
are mapped when loading the dll. Static functions in .exe do have a fixed
address.
With a header-only implementation the functions are provided
libardour.dll and ardour.exe
This commit is contained in:
Robin Gareus 2016-02-10 15:10:40 +01:00
parent 7c3d3031dc
commit 6c8a062be9
3 changed files with 193 additions and 151 deletions

View File

@ -20,15 +20,14 @@
#define AUDIOGRAPHER_ANALYSER_H
#include <fftw3.h>
#include <vamp-hostsdk/PluginLoader.h>
#include <vamp-sdk/Plugin.h>
#include "audiographer/visibility.h"
#include "audiographer/sink.h"
#include "audiographer/routines.h"
#include "audiographer/utils/listed_source.h"
#include "pbd/fastlog.h"
#include "ardour/export_analysis.h"
namespace AudioGrapher
@ -37,152 +36,16 @@ namespace AudioGrapher
class /*LIBAUDIOGRAPHER_API*/ Analyser : public ListedSource<float>, public Sink<float>
{
public:
Analyser (float sample_rate, unsigned int channels, framecnt_t bufsize, framecnt_t n_samples)
: _ebur128_plugin (0)
, _sample_rate (sample_rate)
, _channels (channels)
, _bufsize (bufsize / channels)
, _n_samples (n_samples)
, _pos (0)
{
assert (bufsize % channels == 0);
//printf("NEW ANALYSER %p r:%.1f c:%d f:%ld l%ld\n", this, sample_rate, channels, bufsize, n_samples);
if (channels > 0 && channels <= 2) {
using namespace Vamp::HostExt;
PluginLoader* loader (PluginLoader::getInstance());
_ebur128_plugin = loader->loadPlugin ("libardourvampplugins:ebur128", sample_rate, PluginLoader::ADAPT_ALL_SAFE);
assert (_ebur128_plugin);
_ebur128_plugin->reset ();
_ebur128_plugin->initialise (channels, _bufsize, _bufsize);
}
_bufs[0] = (float*) malloc (sizeof(float) * _bufsize);
_bufs[1] = (float*) malloc (sizeof(float) * _bufsize);
const size_t peaks = sizeof(_result._peaks) / sizeof (ARDOUR::PeakData::PeakDatum) / 2;
_spp = ceil ((_n_samples + 1.f) / (float) peaks);
_fft_data_size = _bufsize / 2;
_fft_freq_per_bin = sample_rate / _fft_data_size / 2.f;
_fft_data_in = (float *) fftwf_malloc (sizeof(float) * _bufsize);
_fft_data_out = (float *) fftwf_malloc (sizeof(float) * _bufsize);
_fft_power = (float *) malloc (sizeof(float) * _fft_data_size);
for (uint32_t i = 0; i < _fft_data_size; ++i) {
_fft_power[i] = 0;
}
for (uint32_t i = 0; i < _bufsize; ++i) {
_fft_data_out[i] = 0;
}
_fft_plan = fftwf_plan_r2r_1d (_bufsize, _fft_data_in, _fft_data_out, FFTW_R2HC, FFTW_MEASURE);
_hann_window = (float *) malloc(sizeof(float) * _bufsize);
double sum = 0.0;
for (uint32_t i = 0; i < _bufsize; ++i) {
_hann_window[i] = 0.5f - (0.5f * (float) cos (2.0f * M_PI * (float)i / (float)(_bufsize)));
sum += _hann_window[i];
}
const double isum = 2.0 / sum;
for (uint32_t i = 0; i < _bufsize; ++i) {
_hann_window[i] *= isum;
}
}
~Analyser ()
{
delete _ebur128_plugin;
free (_bufs[0]);
free (_bufs[1]);
fftwf_destroy_plan (_fft_plan);
fftwf_free (_fft_data_in);
fftwf_free (_fft_data_out);
free (_fft_power);
free (_hann_window);
}
void process (ProcessContext<float> const & c)
{
framecnt_t n_samples = c.frames() / c.channels();
assert (c.frames() % c.channels() == 0);
assert (n_samples <= _bufsize);
//printf("PROC %p @%ld F: %ld, S: %ld C:%d\n", this, _pos, c.frames(), n_samples, c.channels());
float const * d = c.data ();
framecnt_t s;
for (s = 0; s < n_samples; ++s) {
_fft_data_in[s] = 0;
const framecnt_t pk = (_pos + s) / _spp;
for (unsigned int c = 0; c < _channels; ++c) {
const float v = *d;
_bufs[c][s] = v;
if (_result._peaks[pk].min > v) { _result._peaks[pk].min = *d; }
if (_result._peaks[pk].max < v) { _result._peaks[pk].max = *d; }
_fft_data_in[s] += v * _hann_window[s] / (float) _channels;
++d;
}
}
for (; s < _bufsize; ++s) {
for (unsigned int c = 0; c < _channels; ++c) {
_bufs[c][s] = 0.f;
_fft_data_in[s] = 0;
}
}
if (_ebur128_plugin) {
_ebur128_plugin->process (_bufs, Vamp::RealTime::fromSeconds ((double) _pos / _sample_rate));
}
fftwf_execute (_fft_plan);
_fft_power[0] = _fft_data_out[0] * _fft_data_out[0];
#define FRe (_fft_data_out[i])
#define FIm (_fft_data_out[_bufsize - i])
for (uint32_t i = 1; i < _fft_data_size - 1; ++i) {
_fft_power[i] = (FRe * FRe) + (FIm * FIm);
}
#undef FRe
#undef FIm
// TODO handle case where _pos / _spp != (_pos + _bufsize) / _spp
// TODO: get geometry from ExportAnalysis
const framecnt_t x = _pos / _spp;
const float range = 80; // dB
const double ypb = 256.0 / _fft_data_size;
for (uint32_t i = 1; i < _fft_data_size - 1; ++i) {
const float level = fft_power_at_bin (i, i);
if (level < -range) continue;
const float pk = level > 0.0 ? 1.0 : (range + level) / range;
const uint32_t y = 256 - ceil (i * ypb); // log-y?
assert (x >= 0 && x < 800);
assert (y < 256);
if (_result._spectrum[x][y] < pk) { _result._spectrum[x][y] = pk; }
}
_pos += n_samples;
/* pass audio audio through */
ListedSource<float>::output(c);
}
ARDOUR::ExportAnalysisPtr result () {
//printf("PROCESSED %ld / %ld samples\n", _pos, _n_samples);
if (_pos == 0) {
return ARDOUR::ExportAnalysisPtr ();
}
if (_ebur128_plugin) {
Vamp::Plugin::FeatureSet features = _ebur128_plugin->getRemainingFeatures ();
if (!features.empty() && features.size() == 2) {
_result.loudness = features[0][0].values[0];
_result.loudness_range = features[1][0].values[0];
_result.have_loudness = true;
}
}
return ARDOUR::ExportAnalysisPtr (new ARDOUR::ExportAnalysis (_result));
}
Analyser (float sample_rate, unsigned int channels, framecnt_t bufsize, framecnt_t n_samples);
~Analyser ();
void process (ProcessContext<float> const & c);
ARDOUR::ExportAnalysisPtr result ();
using Sink<float>::process;
private:
private:
float fft_power_at_bin (const uint32_t b, const float norm) const;
ARDOUR::ExportAnalysis _result;
Vamp::Plugin* _ebur128_plugin;
@ -203,11 +66,6 @@ class /*LIBAUDIOGRAPHER_API*/ Analyser : public ListedSource<float>, public Sink
float* _fft_power;
fftwf_plan _fft_plan;
inline float fft_power_at_bin (const uint32_t b, const float norm) const {
const float a = _fft_power[b] * norm;
return a > 1e-12 ? 10.0 * fast_log10(a) : -INFINITY;
}
};
} // namespace

View File

@ -0,0 +1,183 @@
/*
* Copyright (C) 2016 Robin Gareus <robin@gareus.org>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include "audiographer/general/analyser.h"
#include "pbd/fastlog.h"
using namespace AudioGrapher;
Analyser::Analyser (float sample_rate, unsigned int channels, framecnt_t bufsize, framecnt_t n_samples)
: _ebur128_plugin (0)
, _sample_rate (sample_rate)
, _channels (channels)
, _bufsize (bufsize / channels)
, _n_samples (n_samples)
, _pos (0)
{
assert (bufsize % channels == 0);
//printf("NEW ANALYSER %p r:%.1f c:%d f:%ld l%ld\n", this, sample_rate, channels, bufsize, n_samples);
if (channels > 0 && channels <= 2) {
using namespace Vamp::HostExt;
PluginLoader* loader (PluginLoader::getInstance());
_ebur128_plugin = loader->loadPlugin ("libardourvampplugins:ebur128", sample_rate, PluginLoader::ADAPT_ALL_SAFE);
assert (_ebur128_plugin);
_ebur128_plugin->reset ();
_ebur128_plugin->initialise (channels, _bufsize, _bufsize);
}
_bufs[0] = (float*) malloc (sizeof(float) * _bufsize);
_bufs[1] = (float*) malloc (sizeof(float) * _bufsize);
const size_t peaks = sizeof(_result.peaks) / sizeof (ARDOUR::PeakData::PeakDatum) / 2;
_spp = ceil ((_n_samples + 1.f) / (float) peaks);
_fft_data_size = _bufsize / 2;
_fft_freq_per_bin = sample_rate / _fft_data_size / 2.f;
_fft_data_in = (float *) fftwf_malloc (sizeof(float) * _bufsize);
_fft_data_out = (float *) fftwf_malloc (sizeof(float) * _bufsize);
_fft_power = (float *) malloc (sizeof(float) * _fft_data_size);
for (uint32_t i = 0; i < _fft_data_size; ++i) {
_fft_power[i] = 0;
}
for (uint32_t i = 0; i < _bufsize; ++i) {
_fft_data_out[i] = 0;
}
_fft_plan = fftwf_plan_r2r_1d (_bufsize, _fft_data_in, _fft_data_out, FFTW_R2HC, FFTW_MEASURE);
_hann_window = (float *) malloc(sizeof(float) * _bufsize);
double sum = 0.0;
for (uint32_t i = 0; i < _bufsize; ++i) {
_hann_window[i] = 0.5f - (0.5f * (float) cos (2.0f * M_PI * (float)i / (float)(_bufsize)));
sum += _hann_window[i];
}
const double isum = 2.0 / sum;
for (uint32_t i = 0; i < _bufsize; ++i) {
_hann_window[i] *= isum;
}
}
Analyser::~Analyser ()
{
delete _ebur128_plugin;
free (_bufs[0]);
free (_bufs[1]);
fftwf_destroy_plan (_fft_plan);
fftwf_free (_fft_data_in);
fftwf_free (_fft_data_out);
free (_fft_power);
free (_hann_window);
}
void
Analyser::process (ProcessContext<float> const & c)
{
framecnt_t n_samples = c.frames() / c.channels();
assert (c.frames() % c.channels() == 0);
assert (n_samples <= _bufsize);
//printf("PROC %p @%ld F: %ld, S: %ld C:%d\n", this, _pos, c.frames(), n_samples, c.channels());
float const * d = c.data ();
framecnt_t s;
for (s = 0; s < n_samples; ++s) {
_fft_data_in[s] = 0;
const framecnt_t pk = (_pos + s) / _spp;
for (unsigned int c = 0; c < _channels; ++c) {
const float v = *d;
_bufs[c][s] = v;
if (_result.peaks[pk].min > v) { _result.peaks[pk].min = *d; }
if (_result.peaks[pk].max < v) { _result.peaks[pk].max = *d; }
_fft_data_in[s] += v * _hann_window[s] / (float) _channels;
++d;
}
}
for (; s < _bufsize; ++s) {
for (unsigned int c = 0; c < _channels; ++c) {
_bufs[c][s] = 0.f;
_fft_data_in[s] = 0;
}
}
if (_ebur128_plugin) {
_ebur128_plugin->process (_bufs, Vamp::RealTime::fromSeconds ((double) _pos / _sample_rate));
}
fftwf_execute (_fft_plan);
_fft_power[0] = _fft_data_out[0] * _fft_data_out[0];
#define FRe (_fft_data_out[i])
#define FIm (_fft_data_out[_bufsize - i])
for (uint32_t i = 1; i < _fft_data_size - 1; ++i) {
_fft_power[i] = (FRe * FRe) + (FIm * FIm);
}
#undef FRe
#undef FIm
// TODO: get geometry from ExportAnalysis
const framecnt_t x0 = _pos / _spp;
const framecnt_t x1 = (_pos + n_samples) / _spp;
const float range = 80; // dB
const double ypb = 200.0 / _fft_data_size;
for (uint32_t i = 1; i < _fft_data_size - 1; ++i) {
const float level = fft_power_at_bin (i, i);
if (level < -range) continue;
const float pk = level > 0.0 ? 1.0 : (range + level) / range;
const uint32_t y = 200 - ceil (i * ypb); // log-y?
assert (y < 200);
for (int x = x0; x < x1; ++x) {
assert (x >= 0 && x < 800);
if (_result.spectrum[x][y] < pk) { _result.spectrum[x][y] = pk; }
}
}
_pos += n_samples;
/* pass audio audio through */
ListedSource<float>::output(c);
}
ARDOUR::ExportAnalysisPtr
Analyser::result ()
{
//printf("PROCESSED %ld / %ld samples\n", _pos, _n_samples);
if (_pos == 0) {
return ARDOUR::ExportAnalysisPtr ();
}
if (_ebur128_plugin) {
Vamp::Plugin::FeatureSet features = _ebur128_plugin->getRemainingFeatures ();
if (!features.empty() && features.size() == 3) {
_result.loudness = features[0][0].values[0];
_result.loudness_range = features[1][0].values[0];
assert (features[2][0].values.size() == 540);
for (int i = 0; i < 540; ++i) {
_result.loudness_hist[i] = features[2][0].values[i];
if (_result.loudness_hist[i] > _result.loudness_hist_max) {
_result.loudness_hist_max = _result.loudness_hist[i]; }
}
_result.have_loudness = true;
}
}
return ARDOUR::ExportAnalysisPtr (new ARDOUR::ExportAnalysis (_result));
}
float
Analyser::fft_power_at_bin (const uint32_t b, const float norm) const
{
const float a = _fft_power[b] * norm;
return a > 1e-12 ? 10.0 * fast_log10(a) : -INFINITY;
}

View File

@ -64,6 +64,7 @@ def build(bld):
'src/general/sample_format_converter.cc',
'src/routines.cc',
'src/debug_utils.cc',
'src/general/analyser.cc',
'src/general/broadcast_info.cc',
'src/general/normalizer.cc'
]
@ -83,7 +84,7 @@ def build(bld):
audiographer.name = 'libaudiographer'
audiographer.target = 'audiographer'
audiographer.export_includes = ['.', './src']
audiographer.includes = ['.', './src']
audiographer.includes = ['.', './src','../ardour','../timecode','../evoral']
audiographer.uselib = 'GLIB GLIBMM GTHREAD SAMPLERATE SNDFILE FFTW3F'
audiographer.use = 'libpbd'
audiographer.vnum = AUDIOGRAPHER_LIB_VERSION