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/*
* Copyright (C) 2003-2014 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 <assert.h>
#include <stdlib.h>
#include <string.h>
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<float>
LibsampleratePcmResampler::Resample2(ConstBuffer<float> 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<float *>(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<float>(data.data_out,
data.output_frames_gen * channels);
}
ConstBuffer<void>
LibsampleratePcmResampler::Resample(ConstBuffer<void> src, Error &error)
{
return Resample2(ConstBuffer<float>::FromVoid(src), error).ToVoid();
}
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