X-Git-Url: https://git.carlh.net/gitweb/?p=dcpomatic.git;a=blobdiff_plain;f=src%2Flib%2Fresampler.cc;h=2c6594a6df7cca9e519c2564040e8a94dd3a52ee;hp=c18bad3ac9eccfcb7b1b9c5ea6fc286207b94d89;hb=48eb118a26bbd98a7ac2d555c4add923b0f2311d;hpb=1858190cff2f960f3d1f0a5cc02c69da86088f5b diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc index c18bad3ac..2c6594a6d 100644 --- a/src/lib/resampler.cc +++ b/src/lib/resampler.cc @@ -1,22 +1,24 @@ /* - Copyright (C) 2013-2015 Carl Hetherington + Copyright (C) 2013-2021 Carl Hetherington - This program is free software; you can redistribute it and/or modify + This file is part of DCP-o-matic. + + DCP-o-matic is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. - This program is distributed in the hope that it will be useful, + DCP-o-matic is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + along with DCP-o-matic. If not, see . */ + #include "resampler.h" #include "audio_buffers.h" #include "exceptions.h" @@ -24,55 +26,80 @@ #include "dcpomatic_assert.h" #include #include +#include #include "i18n.h" + using std::cout; -using std::pair; using std::make_pair; -using boost::shared_ptr; +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. - * @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 StringError (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) { + 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)); + } } + shared_ptr Resampler::run (shared_ptr in) { + DCPOMATIC_ASSERT(in->channels() == _channels); + 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; - data.data_in = 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 = 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]; @@ -80,9 +107,10 @@ Resampler::run (shared_ptr 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; @@ -90,21 +118,26 @@ 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)"), src_strerror (r))); + throw EncodeError ( + String::compose ( + N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"), + src_strerror (r), + in_frames, + max_resampled_frames, + _channels + ) + ); } if (data.output_frames_gen == 0) { 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++; @@ -115,41 +148,39 @@ 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[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) { - 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++; @@ -157,7 +188,14 @@ Resampler::flush () } out_offset += data.output_frames_gen; - out->set_frames (out_offset); return out; } + + +void +Resampler::reset () +{ + src_reset (_src); +} +