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

diff --git a/src/lib/audio_analyser.cc b/src/lib/audio_analyser.cc
index d5095c7e67038fd1c95713709c1f41730cf83e9e..53d764a9b58dfbe55083719760e17f4a5db96334 100644 (file)
--- 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 e47ab94b424d64c4ffda4eb629f56ae4ac46b73a..14c7442856d9b3dd826988c4df37a49454c2ba6e 100644 (file)
--- 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 60eb7f5052a3e5ca4e47a14f850a83f762f75a05..056b2e1ee5c6416e779c7062f48346a858d77923 100644 (file)
--- 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 5a3a7fa40406c8dca974fa61da7cb4c18a71b8f6..0dbd0b491edaa3c3b64d0ad00291ef0dca95ce91 100644 (file)
--- 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;