152 lines
3.8 KiB
C++
152 lines
3.8 KiB
C++
/*
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* Copyright 2003-2017 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "LibsamplerateResampler.hxx"
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#include "config/Block.hxx"
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#include "util/ASCII.hxx"
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#include "util/RuntimeError.hxx"
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#include "util/Domain.hxx"
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#include "Log.hxx"
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#include <assert.h>
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#include <stdlib.h>
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#include <string.h>
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static constexpr Domain libsamplerate_domain("libsamplerate");
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static int lsr_converter = SRC_SINC_FASTEST;
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static bool
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lsr_parse_converter(const char *s)
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{
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assert(s != nullptr);
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if (*s == 0)
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return true;
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char *endptr;
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long l = strtol(s, &endptr, 10);
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if (*endptr == 0 && src_get_name(l) != nullptr) {
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lsr_converter = l;
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return true;
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}
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size_t length = strlen(s);
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for (int i = 0;; ++i) {
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const char *name = src_get_name(i);
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if (name == nullptr)
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break;
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if (StringEqualsCaseASCII(s, name, length)) {
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lsr_converter = i;
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return true;
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}
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}
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return false;
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}
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void
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pcm_resample_lsr_global_init(const ConfigBlock &block)
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{
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const char *converter = block.GetBlockValue("type", "2");
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if (!lsr_parse_converter(converter))
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throw FormatRuntimeError("unknown samplerate converter '%s'",
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converter);
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FormatDebug(libsamplerate_domain,
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"libsamplerate converter '%s'",
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src_get_name(lsr_converter));
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}
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AudioFormat
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LibsampleratePcmResampler::Open(AudioFormat &af, unsigned new_sample_rate)
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{
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assert(af.IsValid());
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assert(audio_valid_sample_rate(new_sample_rate));
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src_rate = af.sample_rate;
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dest_rate = new_sample_rate;
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channels = af.channels;
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/* libsamplerate works with floating point samples */
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af.format = SampleFormat::FLOAT;
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int src_error;
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state = src_new(lsr_converter, channels, &src_error);
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if (!state)
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throw FormatRuntimeError("libsamplerate initialization has failed: %s",
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src_strerror(src_error));
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memset(&data, 0, sizeof(data));
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data.src_ratio = double(new_sample_rate) / double(af.sample_rate);
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FormatDebug(libsamplerate_domain,
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"setting samplerate conversion ratio to %.2lf",
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data.src_ratio);
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src_set_ratio(state, data.src_ratio);
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AudioFormat result = af;
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result.sample_rate = new_sample_rate;
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return result;
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}
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void
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LibsampleratePcmResampler::Close() noexcept
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{
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state = src_delete(state);
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}
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void
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LibsampleratePcmResampler::Reset() noexcept
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{
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src_reset(state);
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}
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inline ConstBuffer<float>
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LibsampleratePcmResampler::Resample2(ConstBuffer<float> src)
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{
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assert(src.size % channels == 0);
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const unsigned src_frames = src.size / channels;
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const unsigned dest_frames =
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(src_frames * dest_rate + src_rate - 1) / src_rate;
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size_t data_out_size = dest_frames * sizeof(float) * channels;
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data.data_in = const_cast<float *>(src.data);
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data.data_out = (float *)buffer.Get(data_out_size);
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data.input_frames = src_frames;
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data.output_frames = dest_frames;
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int result = src_process(state, &data);
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if (result != 0)
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throw FormatRuntimeError("libsamplerate has failed: %s",
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src_strerror(result));
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return ConstBuffer<float>(data.data_out,
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data.output_frames_gen * channels);
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}
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ConstBuffer<void>
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LibsampleratePcmResampler::Resample(ConstBuffer<void> src)
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{
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return Resample2(ConstBuffer<float>::FromVoid(src)).ToVoid();
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}
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