/* * Copyright (C) 2003-2015 The Music Player Daemon Project * http://www.musicpd.org * * This program 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, * 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., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include "config.h" #include "LibsamplerateResampler.hxx" #include "util/ASCII.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" #include "Log.hxx" #include #include #include static constexpr Domain libsamplerate_domain("libsamplerate"); static int lsr_converter = SRC_SINC_FASTEST; static bool lsr_parse_converter(const char *s) { assert(s != nullptr); if (*s == 0) return true; char *endptr; long l = strtol(s, &endptr, 10); if (*endptr == 0 && src_get_name(l) != nullptr) { lsr_converter = l; return true; } size_t length = strlen(s); for (int i = 0;; ++i) { const char *name = src_get_name(i); if (name == nullptr) break; if (StringEqualsCaseASCII(s, name, length)) { lsr_converter = i; return true; } } return false; } bool pcm_resample_lsr_global_init(const char *converter, Error &error) { if (!lsr_parse_converter(converter)) { error.Format(libsamplerate_domain, "unknown samplerate converter '%s'", converter); return false; } FormatDebug(libsamplerate_domain, "libsamplerate converter '%s'", src_get_name(lsr_converter)); return true; } AudioFormat LibsampleratePcmResampler::Open(AudioFormat &af, unsigned new_sample_rate, Error &error) { assert(af.IsValid()); assert(audio_valid_sample_rate(new_sample_rate)); src_rate = af.sample_rate; dest_rate = new_sample_rate; channels = af.channels; /* libsamplerate works with floating point samples */ af.format = SampleFormat::FLOAT; int src_error; state = src_new(lsr_converter, channels, &src_error); if (!state) { error.Format(libsamplerate_domain, src_error, "libsamplerate initialization has failed: %s", src_strerror(src_error)); return AudioFormat::Undefined(); } memset(&data, 0, sizeof(data)); data.src_ratio = double(new_sample_rate) / double(af.sample_rate); FormatDebug(libsamplerate_domain, "setting samplerate conversion ratio to %.2lf", data.src_ratio); src_set_ratio(state, data.src_ratio); AudioFormat result = af; result.sample_rate = new_sample_rate; return result; } void LibsampleratePcmResampler::Close() { state = src_delete(state); } static bool src_process(SRC_STATE *state, SRC_DATA *data, Error &error) { int result = src_process(state, data); if (result != 0) { error.Format(libsamplerate_domain, result, "libsamplerate has failed: %s", src_strerror(result)); return false; } return true; } inline ConstBuffer LibsampleratePcmResampler::Resample2(ConstBuffer src, Error &error) { assert(src.size % channels == 0); const unsigned src_frames = src.size / channels; const unsigned dest_frames = (src_frames * dest_rate + src_rate - 1) / src_rate; size_t data_out_size = dest_frames * sizeof(float) * channels; data.data_in = const_cast(src.data); data.data_out = (float *)buffer.Get(data_out_size); data.input_frames = src_frames; data.output_frames = dest_frames; if (!src_process(state, &data, error)) return nullptr; return ConstBuffer(data.data_out, data.output_frames_gen * channels); } ConstBuffer LibsampleratePcmResampler::Resample(ConstBuffer src, Error &error) { return Resample2(ConstBuffer::FromVoid(src), error).ToVoid(); }