From: Carl Hetherington Date: Mon, 5 Jul 2021 13:58:25 +0000 (+0200) Subject: Tidy a little and use some std::vector instead of raw arrays. X-Git-Tag: v2.15.156~10 X-Git-Url: https://main.carlh.net/gitweb/?p=dcpomatic.git;a=commitdiff_plain;h=db3866008bf2ab1b921c44c4e3c70a909304ac84 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 d5095c7e6..53d764a9b 100644 --- a/src/lib/audio_analyser.cc +++ b/src/lib/audio_analyser.cc @@ -60,8 +60,8 @@ AudioAnalyser::AudioAnalyser (shared_ptr film, shared_ptraudio_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 film, shared_ptrsetup (_filters); #endif - _current = new AudioPoint[_film->audio_channels()]; + _current = std::vector(_film->audio_channels()); if (!from_zero) { _start = _playlist->start().get_value_or(DCPTime()); @@ -127,12 +127,9 @@ AudioAnalyser::AudioAnalyser (shared_ptr film, shared_ptr (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; Frame _done = 0; - float* _sample_peak = nullptr; - Frame* _sample_peak_frame = nullptr; - AudioPoint* _current = nullptr; + std::vector _sample_peak; + std::vector _sample_peak_frame; + std::vector _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 + Copyright (C) 2013-2021 Carl Hetherington 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 Resampler::run (shared_ptr in) { int in_frames = in->frames (); int in_offset = 0; int out_offset = 0; - shared_ptr resampled (new AudioBuffers (_channels, 0)); + auto resampled = make_shared(_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(in_frames) * _out_rate / _in_rate) + 32; SRC_DATA data; - float* in_buffer = new float[in_frames * _channels]; + std::vector in_buffer(in_frames * _channels); + std::vector 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 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 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 in) } if (data.output_frames_gen == 0) { - delete[] data.data_in; - delete[] data.data_out; break; } @@ -131,8 +135,8 @@ Resampler::run (shared_ptr 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 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 Resampler::flush () { - shared_ptr out (new AudioBuffers (_channels, 0)); + auto out = make_shared(_channels, 0); int out_offset = 0; int64_t const output_size = 65536; float dummy[1]; - float* buffer = new float[output_size]; + std::vector 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 + Copyright (C) 2013-2021 Carl Hetherington 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;