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Re-work plugin-analysis (fix crashes)

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Plugin-analysis uses a fixed number of samples, which may be larger
than the session's block-size. This caused problems for some VST
plugins that use audioMasterCallback to query the session's block-size.

Plugin FFT analysis now processes in chunks of the session's block-size
to avoid this issue. This also allows to increase accuracy for
all plugin standards (FFT size was increased to 8192).
This commit is contained in:
Robin Gareus 2019-02-28 01:26:21 +01:00
parent 61623f730c
commit 06854e1315
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GPG key ID: A090BCE02CF57F04
2 changed files with 94 additions and 94 deletions
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184
gtk2_ardour/plugin_eq_gui.cc
View file

@ -1,22 +1,22 @@
/* /*
Copyright (C) 2008 Paul Davis * Copyright (C) 2018 Robin Gareus <robin@gareus.org>
Author: Sampo Savolainen * Copyright (C) 2008 Paul Davis
* Original Author: Sampo Savolainen
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 * This program is free software; you can redistribute it and/or
the Free Software Foundation; either version 2 of the License, or * modify it under the terms of the GNU General Public License
(at your option) any later version. * 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 * This program is distributed in the hope that it will be useful,
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * but WITHOUT ANY WARRANTY; without even the implied warranty of
GNU General Public License for more details. * 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 * You should have received a copy of the GNU General Public License
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/ */
#include <algorithm> #include <algorithm>
#include <math.h> #include <math.h>
@ -25,13 +25,13 @@
#include <sstream> #include <sstream>
#ifdef COMPILER_MSVC #ifdef COMPILER_MSVC
#include <float.h> # include <float.h>
/* isinf() & isnan() are C99 standards, which older MSVC doesn't provide */ /* isinf() & isnan() are C99 standards, which older MSVC doesn't provide */
#define ISINF(val) !((bool)_finite((double)val)) # define ISINF(val) !((bool)_finite((double)val))
#define ISNAN(val) (bool)_isnan((double)val) # define ISNAN(val) (bool)_isnan((double)val)
#else #else
#define ISINF(val) std::isinf((val)) # define ISINF(val) std::isinf((val))
#define ISNAN(val) std::isnan((val)) # define ISNAN(val) std::isnan((val))
#endif #endif
#include <gtkmm/box.h> #include <gtkmm/box.h>
@ -57,6 +57,7 @@ PluginEqGui::PluginEqGui(boost::shared_ptr<ARDOUR::PluginInsert> pluginInsert)
: _min_dB(-12.0) : _min_dB(-12.0)
, _max_dB(+12.0) , _max_dB(+12.0)
, _step_dB(3.0) , _step_dB(3.0)
, _block_size(0)
, _buffer_size(0) , _buffer_size(0)
, _signal_buffer_size(0) , _signal_buffer_size(0)
, _impulse_fft(0) , _impulse_fft(0)
@ -170,13 +171,14 @@ void
PluginEqGui::start_listening () PluginEqGui::start_listening ()
{ {
if (!_plugin) { if (!_plugin) {
_plugin = _plugin_insert->get_impulse_analysis_plugin(); _plugin = _plugin_insert->get_impulse_analysis_plugin ();
} }
_plugin->activate(); _plugin->activate ();
set_buffer_size(4096, 16384); set_buffer_size (8192, 16384);
_plugin->set_block_size (_buffer_size); _block_size = 0; // re-initialize the plugin next time.
// Connect the realtime signal collection callback
/* Connect the realtime signal collection callback */
_plugin_insert->AnalysisDataGathered.connect (analysis_connection, invalidator (*this), boost::bind (&PluginEqGui::signal_collect_callback, this, _1, _2), gui_context()); _plugin_insert->AnalysisDataGathered.connect (analysis_connection, invalidator (*this), boost::bind (&PluginEqGui::signal_collect_callback, this, _1, _2), gui_context());
} }
@ -290,7 +292,7 @@ PluginEqGui::set_buffer_size(uint32_t size, uint32_t signal_size)
_buffer_size = size; _buffer_size = size;
_signal_buffer_size = signal_size; _signal_buffer_size = signal_size;
// allocate separate in+out buffers, VST cannot process in-place /* allocate separate in+out buffers, VST cannot process in-place */
ARDOUR::ChanCount acount (_plugin->get_info()->n_inputs + _plugin->get_info()->n_outputs); ARDOUR::ChanCount acount (_plugin->get_info()->n_inputs + _plugin->get_info()->n_outputs);
ARDOUR::ChanCount ccount = ARDOUR::ChanCount::max (_plugin->get_info()->n_inputs, _plugin->get_info()->n_outputs); ARDOUR::ChanCount ccount = ARDOUR::ChanCount::max (_plugin->get_info()->n_inputs, _plugin->get_info()->n_outputs);
@ -363,91 +365,91 @@ PluginEqGui::run_impulse_analysis()
uint32_t inputs = _plugin->get_info()->n_inputs.n_audio(); uint32_t inputs = _plugin->get_info()->n_inputs.n_audio();
uint32_t outputs = _plugin->get_info()->n_outputs.n_audio(); uint32_t outputs = _plugin->get_info()->n_outputs.n_audio();
// Create the impulse, can't use silence() because consecutive calls won't work /* Create the impulse, can't use silence() because consecutive calls won't work */
for (uint32_t i = 0; i < inputs; ++i) { for (uint32_t i = 0; i < inputs; ++i) {
ARDOUR::AudioBuffer& buf = _bufferset.get_audio(i); ARDOUR::AudioBuffer& buf = _bufferset.get_audio(i);
ARDOUR::Sample* d = buf.data(); ARDOUR::Sample* d = buf.data();
memset(d, 0, sizeof(ARDOUR::Sample)*_buffer_size); memset (d, 0, sizeof(ARDOUR::Sample) * _buffer_size);
*d = 1.0; *d = 1.0;
} }
ARDOUR::ChanMapping in_map(_plugin->get_info()->n_inputs); /* Silence collect buffers to copy data to */
ARDOUR::ChanMapping out_map(_plugin->get_info()->n_outputs);
// map output buffers after input buffers (no inplace for VST)
out_map.offset_to (DataType::AUDIO, inputs);
_plugin->connect_and_run(_bufferset, 0, _buffer_size, 1.0, in_map, out_map, _buffer_size, 0);
samplecnt_t f = _plugin->signal_latency ();
// Adding user_latency() could be interesting
// Gather all output, taking latency into account.
_impulse_fft->reset();
// Silence collect buffers to copy data to, can't use silence() because consecutive calls won't work
for (uint32_t i = 0; i < outputs; ++i) { for (uint32_t i = 0; i < outputs; ++i) {
ARDOUR::AudioBuffer &buf = _collect_bufferset.get_audio(i); ARDOUR::AudioBuffer &buf = _collect_bufferset.get_audio(i);
ARDOUR::Sample *d = buf.data(); ARDOUR::Sample *d = buf.data();
memset(d, 0, sizeof(ARDOUR::Sample)*_buffer_size); memset(d, 0, sizeof(ARDOUR::Sample) * _buffer_size);
} }
if (f == 0) { /* create default linear I/O maps */
//std::cerr << "0: no latency, copying full buffer, trivial.." << std::endl; ARDOUR::ChanMapping in_map (_plugin->get_info()->n_inputs);
for (uint32_t i = 0; i < outputs; ++i) { ARDOUR::ChanMapping out_map (_plugin->get_info()->n_outputs);
memcpy(_collect_bufferset.get_audio(i).data(), /* map output buffers after input buffers (no inplace for VST) */
_bufferset.get_audio(inputs + i).data(), _buffer_size * sizeof(float)); out_map.offset_to (DataType::AUDIO, inputs);
/* run at most at session's block size chunks.
*
* This is important since VSTs may call audioMasterGetBlockSize
* or access various other /real/ session paramaters using the
* audioMasterCallback
*/
samplecnt_t block_size = ARDOUR_UI::instance()->the_session()->get_block_size();
if (_block_size != block_size) {
_block_size = block_size;
_plugin->set_block_size (block_size);
}
samplepos_t sample_pos = 0;
samplecnt_t latency = _plugin->signal_latency ();
samplecnt_t samples_remain = _buffer_size + latency;
_impulse_fft->reset();
while (samples_remain > 0) {
samplecnt_t n_samples = std::min (samples_remain, block_size);
_plugin->connect_and_run(_bufferset, sample_pos, sample_pos + n_samples, 1.0, in_map, out_map, n_samples, 0);
samples_remain -= n_samples;
/* zero input buffers */
if (sample_pos == 0 && samples_remain > 0) {
for (uint32_t i = 0; i < inputs; ++i) {
_bufferset.get_audio(i).data()[0] = 0.f;
}
} }
} else {
//int C = 0;
//std::cerr << (++C) << ": latency is " << f << " samples, doing split processing.." << std::endl;
samplecnt_t target_offset = 0;
samplecnt_t samples_left = _buffer_size; // refaktoroi
do {
if (f >= _buffer_size) {
//std::cerr << (++C) << ": f (=" << f << ") is larger than buffer_size, still trying to reach the actual output" << std::endl;
// there is no data in this buffer regarding to the input!
f -= _buffer_size;
} else {
// this buffer contains either the first, last or a whole bu the output of the impulse
// first part: offset is 0, so we copy to the start of _collect_bufferset
// we start at output offset "f"
// .. and copy "buffer size" - "f" - "offset" samples
samplecnt_t length = _buffer_size - f - target_offset; #ifndef NDEBUG
if (samples_remain > 0) {
//std::cerr << (++C) << ": copying " << length << " samples to _collect_bufferset.get_audio(i)+" << target_offset << " from bufferset at offset " << f << std::endl; for (uint32_t i = 0; i < inputs; ++i) {
for (uint32_t i = 0; i < outputs; ++i) { pframes_t unused;
memcpy(_collect_bufferset.get_audio(i).data(target_offset), assert (_bufferset.get_audio(i).check_silence (block_size, unused));
_bufferset.get_audio(inputs + i).data() + f,
length * sizeof(float));
}
target_offset += length;
samples_left -= length;
f = 0;
} }
if (samples_left > 0) { }
// Silence the buffers #endif
for (uint32_t i = 0; i < inputs; ++i) {
ARDOUR::AudioBuffer &buf = _bufferset.get_audio(i);
ARDOUR::Sample *d = buf.data();
memset(d, 0, sizeof(ARDOUR::Sample)*_buffer_size);
}
_plugin->connect_and_run (_bufferset, target_offset, target_offset + _buffer_size, 1.0, in_map, out_map, _buffer_size, 0); if (sample_pos + n_samples > latency) {
samplecnt_t dst_off = sample_pos >= latency ? sample_pos - latency : 0;
samplecnt_t src_off = sample_pos >= latency ? 0 : latency - sample_pos;
samplecnt_t n_copy = std::min (n_samples, sample_pos + n_samples - latency);
assert (dst_off + n_copy <= _buffer_size);
assert (src_off + n_copy <= _block_size);
for (uint32_t i = 0; i < outputs; ++i) {
memcpy(
&(_collect_bufferset.get_audio(i).data()[dst_off]),
&(_bufferset.get_audio(inputs + i).data()[src_off]),
n_copy * sizeof(float));
} }
} while ( samples_left > 0); }
sample_pos += n_samples;
} }
for (uint32_t i = 0; i < outputs; ++i) { for (uint32_t i = 0; i < outputs; ++i) {
_impulse_fft->analyze(_collect_bufferset.get_audio(i).data()); _impulse_fft->analyze (_collect_bufferset.get_audio(i).data());
} }
// normalize the output
_impulse_fft->calculate(); _impulse_fft->calculate();
// This signals calls expose_analysis_area()
_analysis_area->queue_draw(); _analysis_area->queue_draw();
ARDOUR_UI::instance()->drop_process_buffers (); ARDOUR_UI::instance()->drop_process_buffers ();

4
gtk2_ardour/plugin_eq_gui.h
View file

@ -15,7 +15,6 @@
You should have received a copy of the GNU General Public License You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/ */
#ifndef __ardour_plugin_eq_gui_h #ifndef __ardour_plugin_eq_gui_h
@ -99,6 +98,7 @@ private:
float _log_coeff; float _log_coeff;
float _log_max; float _log_max;
ARDOUR::samplecnt_t _block_size;
ARDOUR::samplecnt_t _buffer_size; ARDOUR::samplecnt_t _buffer_size;
ARDOUR::samplecnt_t _signal_buffer_size; ARDOUR::samplecnt_t _signal_buffer_size;
@ -106,7 +106,6 @@ private:
ARDOUR::BufferSet _bufferset; ARDOUR::BufferSet _bufferset;
ARDOUR::BufferSet _collect_bufferset; ARDOUR::BufferSet _collect_bufferset;
// dimensions // dimensions
float _analysis_width; float _analysis_width;
float _analysis_height; float _analysis_height;
@ -155,4 +154,3 @@ private:
}; };
#endif #endif