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authorBent Bisballe Nyeng <deva@aasimon.org>2013-09-20 22:23:23 +0200
committerBent Bisballe Nyeng <deva@aasimon.org>2013-09-20 22:23:23 +0200
commit73f1a400608d93742e57c7ea3c4433eb1a2e1852 (patch)
treeeb6b500127e095e5bad124927054dc73f44814dc /src/audiofile.cc
parent3d0ed93889afb5ae739d2998e33178689c2de965 (diff)
New resample feature (greatly inspired by Cédric 'SxDx') based on libsamplerate.
Diffstat (limited to 'src/audiofile.cc')
-rw-r--r--src/audiofile.cc42
1 files changed, 38 insertions, 4 deletions
diff --git a/src/audiofile.cc b/src/audiofile.cc
index 37fb5cc..4ac1f94 100644
--- a/src/audiofile.cc
+++ b/src/audiofile.cc
@@ -31,9 +31,12 @@
#include <unistd.h>
#include <sndfile.h>
+#include <samplerate.h>
#include <hugin.hpp>
+#include "configuration.h"
+
#define LAZYLOAD
AudioFile::AudioFile(std::string filename)
@@ -63,13 +66,13 @@ void AudioFile::unload()
{
MutexAutolock l(mutex);
if(data == preloaded_data) {
- delete data;
+ delete[] data;
data = NULL;
size = 0;
}
else {
size = 0;
- delete data;
+ delete[] data;
data = NULL;
delete preloaded_data;
preloaded_data = NULL;
@@ -98,7 +101,7 @@ void AudioFile::init()
sample_t* data = new sample_t[size];
size = sf_read_float(fh, data, size);
-
+
//DEBUG(audiofile,"Lazy loaded %d samples\n", size);
sf_close(fh);
@@ -191,7 +194,13 @@ void AudioFile::load(int num_samples)
size = sf_info.frames;
- if(num_samples != ALL_SAMPLES && (int)size > num_samples) size = num_samples;
+ double ratio = (double)Conf::samplerate / (double)sf_info.samplerate;
+
+ if(num_samples != ALL_SAMPLES) {
+ // Make sure we read enough samples, even after conversion.
+ num_samples /= ratio;
+ if((int)size > num_samples) size = num_samples;
+ }
sample_t* data = new sample_t[size];
size = sf_read_float(fh, data, size);
@@ -200,6 +209,31 @@ void AudioFile::load(int num_samples)
sf_close(fh);
+ if(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", size, this);
+ }
+
+
mutex.lock();
this->data = data;
is_loaded = true;