/* the Music Player Daemon (MPD) * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) * Copyright (C) 2008 Max Kellermann * 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 #include 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; }