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/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* Copyright (C) 2008 Max Kellermann <max@duempel.org>
* This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "pcm_resample.h"
#include "conf.h"
#include "log.h"
#include "utils.h"
#include <stdlib.h>
#include <string.h>
static int pcm_resample_get_converter(void)
{
const char *conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
long convalgo;
char *test;
const char *test2;
size_t len;
if (!conf) {
convalgo = SRC_SINC_FASTEST;
goto out;
}
convalgo = strtol(conf, &test, 10);
if (*test == '\0' && src_get_name(convalgo))
goto out;
len = strlen(conf);
for (convalgo = 0 ; ; convalgo++) {
test2 = src_get_name(convalgo);
if (!test2) {
convalgo = SRC_SINC_FASTEST;
break;
}
if (strncasecmp(test2, conf, len) == 0)
goto out;
}
ERROR("unknown samplerate converter \"%s\"\n", conf);
out:
DEBUG("selecting samplerate converter \"%s\"\n",
src_get_name(convalgo));
return convalgo;
}
size_t
pcm_resample_16(uint8_t channels,
unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate,
int16_t *dest_buffer, size_t dest_size,
struct pcm_resample_state *state)
{
static int convalgo = -1;
SRC_DATA *data = &state->data;
size_t data_in_size;
size_t data_out_size;
int error;
if (convalgo < 0)
convalgo = pcm_resample_get_converter();
/* (re)set the state/ratio if the in or out format changed */
if (channels != state->prev.channels ||
src_rate != state->prev.src_rate ||
dest_rate != state->prev.dest_rate) {
state->error = false;
state->prev.channels = channels;
state->prev.src_rate = src_rate;
state->prev.dest_rate = dest_rate;
if (state->state)
state->state = src_delete(state->state);
state->state = src_new(convalgo, channels, &error);
if (!state->state) {
ERROR("cannot create new libsamplerate state: %s\n",
src_strerror(error));
state->error = true;
return 0;
}
data->src_ratio = (double)dest_rate / (double)src_rate;
DEBUG("setting samplerate conversion ratio to %.2lf\n",
data->src_ratio);
src_set_ratio(state->state, data->src_ratio);
}
/* there was an error previously, and nothing has changed */
if (state->error)
return 0;
data->input_frames = src_size / 2 / channels;
data_in_size = data->input_frames * sizeof(float) * channels;
if (data_in_size > state->data_in_size) {
state->data_in_size = data_in_size;
data->data_in = xrealloc(data->data_in, data_in_size);
}
data->output_frames = dest_size / 2 / channels;
data_out_size = data->output_frames * sizeof(float) * channels;
if (data_out_size > state->data_out_size) {
state->data_out_size = data_out_size;
data->data_out = xrealloc(data->data_out, data_out_size);
}
src_short_to_float_array(src_buffer, data->data_in,
data->input_frames * channels);
error = src_process(state->state, data);
if (error) {
ERROR("error processing samples with libsamplerate: %s\n",
src_strerror(error));
state->error = true;
return 0;
}
src_float_to_short_array(data->data_out, dest_buffer,
data->output_frames_gen * channels);
return data->output_frames_gen * 2 * channels;
}
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