Tidy a little and use some std::vector instead of raw arrays.

4 files changed, 39 insertions, 40 deletions

diff --git a/src/lib/audio_analyser.cc b/src/lib/audio_analyser.cc
index d5095c7e6..53d764a9b 100644
--- a/src/lib/audio_analyser.cc
+++ b/src/lib/audio_analyser.cc
@@ -60,8 +60,8 @@ AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Play
 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
 	, _ebur128 (new AudioFilterGraph(film->audio_frame_rate(), film->audio_channels()))
 #endif
-	, _sample_peak (new float[film->audio_channels()])
-	, _sample_peak_frame (new Frame[film->audio_channels()])
+	, _sample_peak (film->audio_channels())
+	, _sample_peak_frame (film->audio_channels())
 	, _analysis (film->audio_channels())
 {
 
@@ -70,7 +70,7 @@ AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Play
 	_ebur128->setup (_filters);
 #endif
 
-	_current = new AudioPoint[_film->audio_channels()];
+	_current = std::vector<AudioPoint>(_film->audio_channels());
 
 	if (!from_zero) {
 		_start = _playlist->start().get_value_or(DCPTime());
@@ -127,12 +127,9 @@ AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Play
 
 AudioAnalyser::~AudioAnalyser ()
 {
-	delete[] _current;
 	for (auto i: _filters) {
 		delete const_cast<Filter*> (i);
 	}
-	delete[] _sample_peak;
-	delete[] _sample_peak_frame;
 }
 
 
diff --git a/src/lib/audio_analyser.h b/src/lib/audio_analyser.h
index e47ab94b4..14c744285 100644
--- a/src/lib/audio_analyser.h
+++ b/src/lib/audio_analyser.h
@@ -72,9 +72,9 @@ private:
 
 	boost::scoped_ptr<leqm_nrt::Calculator> _leqm;
 	Frame _done = 0;
-	float* _sample_peak = nullptr;
-	Frame* _sample_peak_frame = nullptr;
-	AudioPoint* _current = nullptr;
+	std::vector<float> _sample_peak;
+	std::vector<Frame> _sample_peak_frame;
+	std::vector<AudioPoint> _current;
 
 	AudioAnalysis _analysis;
 };
diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
index 60eb7f505..056b2e1ee 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"
@@ -29,12 +30,15 @@
 
 #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 std::shared_ptr;
 
+
 /** @param in Input sampling rate (Hz)
  *  @param out Output sampling rate (Hz)
  *  @param channels Number of channels.
@@ -47,10 +51,11 @@ 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 ()
 {
 	if (_src) {
@@ -58,38 +63,41 @@ Resampler::~Resampler ()
 	}
 }
 
+
 void
 Resampler::set_fast ()
 {
 	src_delete (_src);
-	_src = 0;
+	_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;
-		float* in_buffer = 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 = in_buffer;
+			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];
@@ -97,10 +105,10 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 			}
 		}
 
-		data.data_in = in_buffer;
+		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;
@@ -108,8 +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]"),
@@ -122,8 +128,6 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 		}
 
 		if (data.output_frames_gen == 0) {
-			delete[] data.data_in;
-			delete[] data.data_out;
 			break;
 		}
 
@@ -131,8 +135,8 @@ Resampler::run (shared_ptr<const AudioBuffers> in)
 		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++;
@@ -143,42 +147,39 @@ 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);
 
-	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++;
@@ -188,12 +189,13 @@ Resampler::flush ()
 	out_offset += data.output_frames_gen;
 	out->set_frames (out_offset);
 
-	delete[] buffer;
 	return out;
 }
 
+
 void
 Resampler::reset ()
 {
 	src_reset (_src);
 }
+
diff --git a/src/lib/resampler.h b/src/lib/resampler.h
index 5a3a7fa40..0dbd0b491 100644
--- a/src/lib/resampler.h
+++ b/src/lib/resampler.h
@@ -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.
 
@@ -41,7 +41,7 @@ public:
 	void set_fast ();
 
 private:
-	SRC_STATE* _src;
+	SRC_STATE* _src = nullptr;
 	int _in_rate;
 	int _out_rate;
 	int _channels;