X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fresampler.cc;h=322c00c136779f009ad1163119bce5496a191dad;hb=e9cca90a162e3e4b1db4a5e5188831beaa69d44e;hp=db5552d15ab3159b17510ca1295704d8806507a0;hpb=a8a0dfd1b21de6c0facf965ab119833ff6f790bf;p=dcpomatic.git

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index db5552d15..322c00c13 100644
--- a/src/lib/resampler.cc
+++ b/src/lib/resampler.cc
@@ -25,6 +25,7 @@
 #include "dcpomatic_assert.h"
 #include <samplerate.h>
 #include <iostream>
+#include <cmath>
 
 #include "i18n.h"
 
@@ -37,23 +38,37 @@ using boost::shared_ptr;
 /** @param in Input sampling rate (Hz)
  *  @param out Output sampling rate (Hz)
  *  @param channels Number of channels.
- *  @param fast true to be fast rather than good.
  */
-Resampler::Resampler (int in, int out, int channels, bool fast)
+Resampler::Resampler (int in, int out, int channels)
 	: _in_rate (in)
 	, _out_rate (out)
 	, _channels (channels)
 {
 	int error;
-	_src = src_new (fast ? SRC_LINEAR : SRC_SINC_BEST_QUALITY, _channels, &error);
+	_src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
 	if (!_src) {
 		throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
 	}
 }
 
 Resampler::~Resampler ()
+{
+	if (_src) {
+		src_delete (_src);
+	}
+}
+
+void
+Resampler::set_fast ()
 {
 	src_delete (_src);
+	_src = 0;
+
+	int error;
+	_src = src_new (SRC_LINEAR, _channels, &error);
+	if (!_src) {
+		throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
+	}
 }
 
 shared_ptr<const AudioBuffers>
@@ -70,11 +85,11 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 		int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
 
 		SRC_DATA data;
-		data.data_in = new float[in_frames * _channels];
+		float* in_buffer = new float[in_frames * _channels];
 
 		{
 			float** p = in->data ();
-			float* q = data.data_in;
+			float* q = in_buffer;
 			for (int i = 0; i < in_frames; ++i) {
 				for (int j = 0; j < _channels; ++j) {
 					*q++ = p[j][in_offset + i];
@@ -82,6 +97,7 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 			}
 		}
 
+		data.data_in = in_buffer;
 		data.input_frames = in_frames;
 
 		data.data_out = new float[max_resampled_frames * _channels];
@@ -106,6 +122,8 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 		}
 
 		if (data.output_frames_gen == 0) {
+			delete[] data.data_in;
+			delete[] data.data_out;
 			break;
 		}
 
@@ -173,3 +191,9 @@ Resampler::flush ()
 	delete[] buffer;
 	return out;
 }
+
+void
+Resampler::reset ()
+{
+	src_reset (_src);
+}