X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fresampler.cc;h=4447ccf0dfc2728ad6cffe3b78fee435c2f3ec09;hb=f706bbb9afd10472e81a051cd5db601d6404377c;hp=e991591c24544b5bf77849eb7eacd5d564849ded;hpb=7a68de9aa2aba678f9ae9c6f9e11d9fc20c1c8e2;p=dcpomatic.git diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc index e991591c2..4447ccf0d 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" @@ -29,11 +30,14 @@ #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) @@ -47,55 +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)); } } -pair, Frame> -Resampler::run (shared_ptr in, Frame frame) -{ - if (!_next_in || !_next_out || _next_in.get() != frame) { - /* Either there was a discontinuity in the input or this is the first input; - reset _next_out. - */ - _next_out = lrintf (frame * _out_rate / _in_rate); - } - - /* Expected next input frame */ - _next_in = frame + in->frames (); +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]; @@ -103,10 +105,10 @@ Resampler::run (shared_ptr in, Frame frame) } } - 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; @@ -114,8 +116,6 @@ Resampler::run (shared_ptr in, Frame frame) 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]"), @@ -128,17 +128,14 @@ Resampler::run (shared_ptr in, Frame frame) } if (data.output_frames_gen == 0) { - delete[] data.data_in; - delete[] data.data_out; 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++; @@ -149,47 +146,39 @@ Resampler::run (shared_ptr in, Frame frame) 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; } - Frame out_frame = _next_out.get (); - - /* Expected next output frame */ - _next_out = _next_out.get() + resampled->frames(); - - return make_pair (resampled, out_frame); + return resampled; } -pair, Frame> + +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); + 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++; @@ -197,14 +186,14 @@ Resampler::flush () } out_offset += data.output_frames_gen; - out->set_frames (out_offset); - delete[] buffer; - return make_pair (out, _next_out.get ()); + return out; } + void Resampler::reset () { src_reset (_src); } +