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authorBent Bisballe Nyeng <deva@aasimon.org>2014-05-17 21:27:52 +0200
committerBent Bisballe Nyeng <deva@aasimon.org>2014-05-17 21:27:52 +0200
commit0e048dee411cc20beb2a8667e5bb339f9b7fb939 (patch)
treec9bacf6f2491f5ae6feb4f4786688ef5659f5c9b /src
parentb6405171c6a74a5227555a9b75c03f88a5f21ed2 (diff)
New resample feature on output channels.
Diffstat (limited to 'src')
-rw-r--r--src/Makefile.am.drumgizmo2
-rw-r--r--src/audiofile.cc36
-rw-r--r--src/audiooutputengine.h7
-rw-r--r--src/drumgizmo.cc82
-rw-r--r--src/drumkit.cc6
-rw-r--r--src/drumkit.h3
-rw-r--r--src/drumkitparser.cc8
7 files changed, 95 insertions, 49 deletions
diff --git a/src/Makefile.am.drumgizmo b/src/Makefile.am.drumgizmo
index c48c478..da5d839 100644
--- a/src/Makefile.am.drumgizmo
+++ b/src/Makefile.am.drumgizmo
@@ -26,4 +26,4 @@ DRUMGIZMO_SOURCES = \
$(top_srcdir)/src/velocity.cc \
$(top_srcdir)/src/versionstr.cc
-DRUMGIZMO_LIBS = $(SNDFILE_LIBS) $(EXPAT_LIBS) $(SAMPLERATE_LIBS) \ No newline at end of file
+DRUMGIZMO_LIBS = $(ZITA_LIBS) $(SNDFILE_LIBS) $(EXPAT_LIBS) $(SAMPLERATE_LIBS) \ No newline at end of file
diff --git a/src/audiofile.cc b/src/audiofile.cc
index 874b9c5..59e0c14 100644
--- a/src/audiofile.cc
+++ b/src/audiofile.cc
@@ -34,9 +34,6 @@
#include <stdlib.h>
#include <unistd.h>
#include <sndfile.h>
-#ifdef WITH_RESAMPLE
-#include <samplerate.h>
-#endif/*WITH_RESAMPLE*/
#include <hugin.hpp>
#include "configuration.h"
@@ -154,39 +151,6 @@ void AudioFile::load(int num_samples)
sf_close(fh);
-#ifdef WITH_RESAMPLE
-
- // Check environment to see if resample should be disabled.
- // Defaults to "1" which is 'enable'. All other values are 'disabled'.
- const char *env_res = getenv("DRUMGIZMO_RESAMPLE");
- if(env_res == NULL) env_res = "1";
-
- if( (strcmp(env_res, "1") == 0) &&
- Conf::samplerate != sf_info.samplerate) {
- // Resample data...
- size_t osize = size * ratio;
- sample_t *odata = new sample_t[osize];
-
- SRC_DATA src;
- src.data_in = data;
- src.input_frames = size;
-
- src.data_out = odata;
- src.output_frames = osize;
-
- src.src_ratio = ratio;
-
- // Do the conversion
- src_simple(&src, SRC_SINC_BEST_QUALITY, 1);
-
- delete[] data;
- data = odata;
- size = src.output_frames;
-
- DEBUG(audiofile,"Converted into %d samples %p\n", (int)size, this);
- }
-#endif/*WITH_RESAMPLE*/
-
this->data = data;
is_loaded = true;
diff --git a/src/audiooutputengine.h b/src/audiooutputengine.h
index 7f15e49..25b6ea0 100644
--- a/src/audiooutputengine.h
+++ b/src/audiooutputengine.h
@@ -33,6 +33,8 @@
#include "channel.h"
+#define UNKNOWN_SAMPLERATE 0
+
class AudioOutputEngine {
public:
virtual ~AudioOutputEngine() {}
@@ -55,6 +57,11 @@ public:
* Overload this method to force engine to use different buffer size.
*/
virtual size_t getBufferSize() { return 1024; }
+
+ /**
+ * Overload this method to report output engine samplerate.
+ */
+ virtual size_t samplerate() { return UNKNOWN_SAMPLERATE; }
};
#endif/*__DRUMGIZMO_AUDIOOUTPUTENGINE_H__*/
diff --git a/src/drumgizmo.cc b/src/drumgizmo.cc
index 5d67c86..3ffcba5 100644
--- a/src/drumgizmo.cc
+++ b/src/drumgizmo.cc
@@ -36,6 +36,19 @@
#include <hugin.hpp>
+#include <config.h>
+
+//
+// Warning: Zita currently not working...
+//
+#ifdef WITH_RESAMPLE
+#ifdef ZITA
+#include <zita-resampler/resampler.h>
+#else
+#include <samplerate.h>
+#endif/*ZITA*/
+#endif/*WITH_RESAMPLE*/
+
#include "drumkitparser.h"
#include "audioinputenginemidi.h"
#include "configuration.h"
@@ -158,6 +171,13 @@ void DrumGizmo::handleMessage(Message *msg)
bool DrumGizmo::run(size_t pos, sample_t *samples, size_t nsamples)
{
+ double samplerate_scale = 1.0;
+#ifndef WITH_RESAMPLE
+ if(oe->samplerate() != UNKNOWN_SAMPLERATE) {
+ samplerate_scale = (double)kit.samplerate() / (double)oe->samplerate();
+ }
+#endif/*WITH_RESAMPLE*/
+
// Handle engine messages, at most one in each iteration:
handleMessages(1);
@@ -240,7 +260,7 @@ bool DrumGizmo::run(size_t pos, sample_t *samples, size_t nsamples)
} else {
//DEBUG(drumgizmo, "Adding event %d.\n", evs[e].offset);
Event *evt = new EventSample(ch.num, 1.0, af, i->group(), i);
- evt->offset = evs[e].offset + pos;
+ evt->offset = (evs[e].offset + pos) * samplerate_scale;
activeevents[ch.num].push_back(evt);
}
j++;
@@ -255,22 +275,60 @@ bool DrumGizmo::run(size_t pos, sample_t *samples, size_t nsamples)
free(evs);
-
//
// Write audio
//
for(size_t c = 0; c < kit.channels.size(); c++) {
- sample_t *buf = samples;
- bool internal = false;
- if(oe->getBuffer(c)) {
- buf = oe->getBuffer(c);
- internal = true;
- }
- if(buf) {
- memset(buf, 0, nsamples * sizeof(sample_t));
+ if(samplerate_scale == 1.0) {
+ // No resampling needed
+ sample_t *buf = samples;
+ bool internal = false;
+ if(oe->getBuffer(c)) {
+ buf = oe->getBuffer(c);
+ internal = true;
+ }
+ if(buf) {
+ memset(buf, 0, nsamples * sizeof(sample_t));
- getSamples(c, pos, buf, nsamples);
- if(!internal) oe->run(c, samples, nsamples);
+ getSamples(c, pos, buf, nsamples);
+
+ if(!internal) oe->run(c, samples, nsamples);
+ }
+ } else {
+#ifdef WITH_RESAMPLE
+ // Resampling needed
+ size_t nkitsamples = nsamples * samplerate_scale;
+ sample_t kitsamples[nkitsamples];
+
+ memset(rs, 0, nkitsamples * sizeof(sample_t));
+ getSamples(c, pos * samplerate_scale, rs, nkitsamples);
+
+#ifdef ZITA
+ Resampler resampler;
+ resampler.setup(kit.samplerate(), oe->samplerate(), 1, 96);
+
+ resampler.inp_data = kitsamples;
+ resampler.inp_count = nkitsamples;
+
+ resampler.out_data = samples;
+ resampler.out_count = nsamples;
+
+ resampler.process();
+#else
+ SRC_DATA src;
+ src.data_in = kitsamples;
+ src.input_frames = nkitsamples;
+
+ src.data_out = samples;
+ src.output_frames = nsamples;
+
+ src.src_ratio = 1.0 / samplerate_scale;
+
+ src_simple(&src, SRC_SINC_BEST_QUALITY, 1);
+#endif/*ZITA*/
+
+ oe->run(c, samples, nsamples);
+#endif/*WITH_RESAMPLE*/
}
}
diff --git a/src/drumkit.cc b/src/drumkit.cc
index c2aa221..4d75f3b 100644
--- a/src/drumkit.cc
+++ b/src/drumkit.cc
@@ -50,6 +50,7 @@ void DrumKit::clear()
_name = "";
_description = "";
+ _samplerate = 44100;
}
bool DrumKit::isValid()
@@ -72,6 +73,11 @@ std::string DrumKit::description()
return _description;
}
+size_t DrumKit::samplerate()
+{
+ return _samplerate;
+}
+
#ifdef TEST_DRUMKIT
//Additional dependency files
//deps:
diff --git a/src/drumkit.h b/src/drumkit.h
index 82fe69b..04b2c56 100644
--- a/src/drumkit.h
+++ b/src/drumkit.h
@@ -53,6 +53,8 @@ public:
bool isValid();
+ size_t samplerate();
+
private:
void *magic;
@@ -60,6 +62,7 @@ private:
std::string _name;
std::string _description;
+ size_t _samplerate;
VersionStr _version;
};
diff --git a/src/drumkitparser.cc b/src/drumkitparser.cc
index 1b7ecaf..2c21c52 100644
--- a/src/drumkitparser.cc
+++ b/src/drumkitparser.cc
@@ -60,6 +60,14 @@ void DrumKitParser::startTag(std::string name,
if(attr.find("name") != attr.end())
kit._name = attr["name"];
+ if(attr.find("samplerate") != attr.end()) {
+ kit._samplerate = atoi(attr["samplerate"].c_str());
+ } else {
+ // If 'samplerate' attribute is missing, assume 44k1Hz
+ // TODO: Ask instrument what samplerate is in the audiofiles...
+ kit._samplerate = 44100;
+ }
+
if(attr.find("description") != attr.end())
kit._description = attr["description"];