X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fresampler.cc;h=4447ccf0dfc2728ad6cffe3b78fee435c2f3ec09;hb=ff639b3cf30afcc097bfd21d39c8d15f466cadd6;hp=ff93d1609ae5e68590c3dcb76f8e2a337be216bf;hpb=76f83b97c401c24b3c93baee0665e84be05f43ea;p=dcpomatic.git

diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index ff93d1609..4447ccf0d 100644
--- a/src/lib/resampler.cc
+++ b/src/lib/resampler.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2013-2015 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2013-2021 Carl Hetherington <cth@carlh.net>
 
     This file is part of DCP-o-matic.
 
@@ -18,6 +18,7 @@
 
 */
 
+
 #include "resampler.h"
 #include "audio_buffers.h"
 #include "exceptions.h"
@@ -25,14 +26,18 @@
 #include "dcpomatic_assert.h"
 #include <samplerate.h>
 #include <iostream>
+#include <cmath>
 
 #include "i18n.h"
 
+
 using std::cout;
-using std::pair;
 using std::make_pair;
+using std::make_shared;
+using std::pair;
 using std::runtime_error;
-using boost::shared_ptr;
+using std::shared_ptr;
+
 
 /** @param in Input sampling rate (Hz)
  *  @param out Output sampling rate (Hz)
@@ -46,45 +51,53 @@ Resampler::Resampler (int in, int out, int channels)
 	int 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));
+		throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
 	}
 }
 
+
 Resampler::~Resampler ()
 {
-	src_delete (_src);
+	if (_src) {
+		src_delete (_src);
+	}
 }
 
+
 void
 Resampler::set_fast ()
 {
 	src_delete (_src);
+	_src = nullptr;
+
 	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));
+		throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
 	}
 }
 
+
 shared_ptr<const AudioBuffers>
 Resampler::run (shared_ptr<const AudioBuffers> in)
 {
 	int in_frames = in->frames ();
 	int in_offset = 0;
 	int out_offset = 0;
-	shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
+	auto resampled = make_shared<AudioBuffers>(_channels, 0);
 
 	while (in_frames > 0) {
 
 		/* Compute the resampled frames count and add 32 for luck */
-		int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
+		int const max_resampled_frames = ceil (static_cast<double>(in_frames) * _out_rate / _in_rate) + 32;
 
 		SRC_DATA data;
-		data.data_in = new float[in_frames * _channels];
+		std::vector<float> in_buffer(in_frames * _channels);
+		std::vector<float> out_buffer(max_resampled_frames * _channels);
 
 		{
-			float** p = in->data ();
-			float* q = data.data_in;
+			auto p = in->data ();
+			auto q = in_buffer.data();
 			for (int i = 0; i < in_frames; ++i) {
 				for (int j = 0; j < _channels; ++j) {
 					*q++ = p[j][in_offset + i];
@@ -92,9 +105,10 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 			}
 		}
 
+		data.data_in = in_buffer.data();
 		data.input_frames = in_frames;
 
-		data.data_out = new float[max_resampled_frames * _channels];
+		data.data_out = out_buffer.data();
 		data.output_frames = max_resampled_frames;
 
 		data.end_of_input = 0;
@@ -102,8 +116,6 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 
 		int const r = src_process (_src, &data);
 		if (r) {
-			delete[] data.data_in;
-			delete[] data.data_out;
 			throw EncodeError (
 				String::compose (
 					N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
@@ -119,12 +131,11 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 			break;
 		}
 
-		resampled->ensure_size (out_offset + data.output_frames_gen);
 		resampled->set_frames (out_offset + data.output_frames_gen);
 
 		{
-			float* p = data.data_out;
-			float** q = resampled->data ();
+			auto p = data.data_out;
+			auto q = resampled->data ();
 			for (int i = 0; i < data.output_frames_gen; ++i) {
 				for (int j = 0; j < _channels; ++j) {
 					q[j][out_offset + i] = *p++;
@@ -135,42 +146,39 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 		in_frames -= data.input_frames_used;
 		in_offset += data.input_frames_used;
 		out_offset += data.output_frames_gen;
-
-		delete[] data.data_in;
-		delete[] data.data_out;
 	}
 
 	return resampled;
 }
 
+
 shared_ptr<const AudioBuffers>
 Resampler::flush ()
 {
-	shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
+	auto out = make_shared<AudioBuffers>(_channels, 0);
 	int out_offset = 0;
 	int64_t const output_size = 65536;
 
 	float dummy[1];
-	float* buffer = new float[output_size];
+	std::vector<float> buffer(output_size);
 
 	SRC_DATA data;
 	data.data_in = dummy;
 	data.input_frames = 0;
-	data.data_out = buffer;
+	data.data_out = buffer.data();
 	data.output_frames = output_size;
 	data.end_of_input = 1;
 	data.src_ratio = double (_out_rate) / _in_rate;
 
 	int const r = src_process (_src, &data);
 	if (r) {
-		delete[] buffer;
-		throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
+		throw EncodeError (String::compose(N_("could not run sample-rate converter (%1)"), src_strerror(r)));
 	}
 
-	out->ensure_size (out_offset + data.output_frames_gen);
+	out->set_frames (out_offset + data.output_frames_gen);
 
-	float* p = data.data_out;
-	float** q = out->data ();
+	auto p = data.data_out;
+	auto q = out->data ();
 	for (int i = 0; i < data.output_frames_gen; ++i) {
 		for (int j = 0; j < _channels; ++j) {
 			q[j][out_offset + i] = *p++;
@@ -178,8 +186,14 @@ Resampler::flush ()
 	}
 
 	out_offset += data.output_frames_gen;
-	out->set_frames (out_offset);
 
-	delete[] buffer;
 	return out;
 }
+
+
+void
+Resampler::reset ()
+{
+	src_reset (_src);
+}
+