a7924d141d
Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2() and pcm_convert_channels_2_to_1(). Improved the quality of pcm_convert_channels_2_to_1() by calculating the arithmetic mean value of both samples.
577 lines
14 KiB
C
577 lines
14 KiB
C
/* the Music Player Daemon (MPD)
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* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: 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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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "pcm_utils.h"
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#include "log.h"
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#include "utils.h"
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#include "conf.h"
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#include "audio_format.h"
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#include <assert.h>
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#include <string.h>
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#include <math.h>
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static inline int
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pcm_dither(void)
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{
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return (rand() & 511) - (rand() & 511);
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}
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/**
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* Check if the value is within the range of the provided bit size,
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* and caps it if necessary.
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*/
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static int32_t
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pcm_range(int32_t sample, unsigned bits)
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{
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if (mpd_unlikely(sample < (-1 << (bits - 1))))
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return -1 << (bits - 1);
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if (mpd_unlikely(sample >= (1 << (bits - 1))))
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return (1 << (bits - 1)) - 1;
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return sample;
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}
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static void
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pcm_volume_change_8(int8_t *buffer, unsigned num_samples, int volume)
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{
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while (num_samples > 0) {
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int32_t sample = *buffer;
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sample = (sample * volume + pcm_dither() + 500) / 1000;
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*buffer++ = pcm_range(sample, 8);
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--num_samples;
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}
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}
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static void
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pcm_volume_change_16(int16_t *buffer, unsigned num_samples, int volume)
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{
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while (num_samples > 0) {
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int32_t sample = *buffer;
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sample = (sample * volume + pcm_dither() + 500) / 1000;
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*buffer++ = pcm_range(sample, 16);
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--num_samples;
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}
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}
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static void
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pcm_volume_change_24(int32_t *buffer, unsigned num_samples, int volume)
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{
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while (num_samples > 0) {
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int64_t sample = *buffer;
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sample = (sample * volume + pcm_dither() + 500) / 1000;
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*buffer++ = pcm_range(sample, 24);
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--num_samples;
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}
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}
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void pcm_volumeChange(char *buffer, int bufferSize,
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const struct audio_format *format,
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int volume)
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{
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if (volume >= 1000)
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return;
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if (volume <= 0) {
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memset(buffer, 0, bufferSize);
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return;
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}
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switch (format->bits) {
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case 8:
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pcm_volume_change_8((int8_t *)buffer, bufferSize, volume);
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break;
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case 16:
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pcm_volume_change_16((int16_t *)buffer, bufferSize / 2,
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volume);
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break;
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case 24:
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pcm_volume_change_24((int32_t*)buffer, bufferSize / 4,
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volume);
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break;
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default:
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FATAL("%u bits not supported by pcm_volumeChange!\n",
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format->bits);
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}
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}
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static void
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pcm_add_8(int8_t *buffer1, const int8_t *buffer2,
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unsigned num_samples, int volume1, int volume2)
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{
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while (num_samples > 0) {
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int32_t sample1 = *buffer1;
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int32_t sample2 = *buffer2++;
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sample1 = ((sample1 * volume1 + sample2 * volume2) +
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pcm_dither() + 500) / 1000;
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*buffer1++ = pcm_range(sample1, 8);
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--num_samples;
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}
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}
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static void
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pcm_add_16(int16_t *buffer1, const int16_t *buffer2,
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unsigned num_samples, int volume1, int volume2)
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{
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while (num_samples > 0) {
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int32_t sample1 = *buffer1;
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int32_t sample2 = *buffer2++;
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sample1 = ((sample1 * volume1 + sample2 * volume2) +
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pcm_dither() + 500) / 1000;
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*buffer1++ = pcm_range(sample1, 16);
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--num_samples;
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}
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}
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static void
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pcm_add_24(int32_t *buffer1, const int32_t *buffer2,
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unsigned num_samples, unsigned volume1, unsigned volume2)
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{
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while (num_samples > 0) {
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int64_t sample1 = *buffer1;
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int64_t sample2 = *buffer2++;
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sample1 = ((sample1 * volume1 + sample2 * volume2) +
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pcm_dither() + 500) / 1000;
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*buffer1++ = pcm_range(sample1, 24);
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--num_samples;
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}
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}
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static void pcm_add(char *buffer1, const char *buffer2, size_t size,
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int vol1, int vol2,
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const struct audio_format *format)
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{
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switch (format->bits) {
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case 8:
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pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2,
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size, vol1, vol2);
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break;
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case 16:
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pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2,
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size / 2, vol1, vol2);
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break;
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case 24:
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pcm_add_24((int32_t*)buffer1,
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(const int32_t*)buffer2,
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size / 4, vol1, vol2);
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break;
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default:
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FATAL("%u bits not supported by pcm_add!\n", format->bits);
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}
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}
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void pcm_mix(char *buffer1, const char *buffer2, size_t size,
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const struct audio_format *format, float portion1)
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{
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int vol1;
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float s = sin(M_PI_2 * portion1);
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s *= s;
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vol1 = s * 1000 + 0.5;
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vol1 = vol1 > 1000 ? 1000 : (vol1 < 0 ? 0 : vol1);
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pcm_add(buffer1, buffer2, size, vol1, 1000 - vol1, format);
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}
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#ifdef HAVE_LIBSAMPLERATE
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static int pcm_getSampleRateConverter(void)
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{
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const char *conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
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long convalgo;
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char *test;
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const char *test2;
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size_t len;
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if (!conf) {
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convalgo = SRC_SINC_FASTEST;
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goto out;
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}
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convalgo = strtol(conf, &test, 10);
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if (*test == '\0' && src_get_name(convalgo))
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goto out;
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len = strlen(conf);
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for (convalgo = 0 ; ; convalgo++) {
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test2 = src_get_name(convalgo);
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if (!test2) {
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convalgo = SRC_SINC_FASTEST;
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break;
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}
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if (strncasecmp(test2, conf, len) == 0)
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goto out;
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}
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ERROR("unknown samplerate converter \"%s\"\n", conf);
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out:
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DEBUG("selecting samplerate converter \"%s\"\n",
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src_get_name(convalgo));
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return convalgo;
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}
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#endif
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#ifdef HAVE_LIBSAMPLERATE
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static size_t pcm_convertSampleRate(int8_t channels, uint32_t inSampleRate,
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const int16_t *inBuffer, size_t inSize,
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uint32_t outSampleRate, int16_t *outBuffer,
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size_t outSize, ConvState *convState)
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{
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static int convalgo = -1;
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SRC_DATA *data = &convState->data;
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size_t dataInSize;
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size_t dataOutSize;
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int error;
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if (convalgo < 0)
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convalgo = pcm_getSampleRateConverter();
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/* (re)set the state/ratio if the in or out format changed */
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if ((channels != convState->lastChannels) ||
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(inSampleRate != convState->lastInSampleRate) ||
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(outSampleRate != convState->lastOutSampleRate)) {
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convState->error = 0;
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convState->lastChannels = channels;
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convState->lastInSampleRate = inSampleRate;
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convState->lastOutSampleRate = outSampleRate;
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if (convState->state)
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convState->state = src_delete(convState->state);
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convState->state = src_new(convalgo, channels, &error);
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if (!convState->state) {
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ERROR("cannot create new libsamplerate state: %s\n",
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src_strerror(error));
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convState->error = 1;
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return 0;
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}
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data->src_ratio = (double)outSampleRate / (double)inSampleRate;
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DEBUG("setting samplerate conversion ratio to %.2lf\n",
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data->src_ratio);
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src_set_ratio(convState->state, data->src_ratio);
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}
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/* there was an error previously, and nothing has changed */
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if (convState->error)
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return 0;
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data->input_frames = inSize / 2 / channels;
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dataInSize = data->input_frames * sizeof(float) * channels;
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if (dataInSize > convState->dataInSize) {
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convState->dataInSize = dataInSize;
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data->data_in = xrealloc(data->data_in, dataInSize);
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}
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data->output_frames = outSize / 2 / channels;
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dataOutSize = data->output_frames * sizeof(float) * channels;
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if (dataOutSize > convState->dataOutSize) {
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convState->dataOutSize = dataOutSize;
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data->data_out = xrealloc(data->data_out, dataOutSize);
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}
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src_short_to_float_array((const short *)inBuffer, data->data_in,
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data->input_frames * channels);
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error = src_process(convState->state, data);
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if (error) {
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ERROR("error processing samples with libsamplerate: %s\n",
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src_strerror(error));
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convState->error = 1;
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return 0;
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}
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src_float_to_short_array(data->data_out, (short *)outBuffer,
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data->output_frames_gen * channels);
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return data->output_frames_gen * 2 * channels;
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}
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#else /* !HAVE_LIBSAMPLERATE */
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/* resampling code blatantly ripped from ESD */
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static size_t pcm_convertSampleRate(int8_t channels, uint32_t inSampleRate,
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const int16_t *inBuffer,
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mpd_unused size_t inSize,
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uint32_t outSampleRate, char *outBuffer,
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size_t outSize,
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mpd_unused ConvState *convState)
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{
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uint32_t rd_dat = 0;
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uint32_t wr_dat = 0;
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const int16_t *in = (const int16_t *)inBuffer;
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int16_t *out = (int16_t *)outBuffer;
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uint32_t nlen = outSize / 2;
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int16_t lsample, rsample;
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switch (channels) {
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case 1:
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while (wr_dat < nlen) {
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rd_dat = wr_dat * inSampleRate / outSampleRate;
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lsample = in[rd_dat++];
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out[wr_dat++] = lsample;
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}
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break;
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case 2:
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while (wr_dat < nlen) {
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rd_dat = wr_dat * inSampleRate / outSampleRate;
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rd_dat &= ~1;
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lsample = in[rd_dat++];
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rsample = in[rd_dat++];
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out[wr_dat++] = lsample;
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out[wr_dat++] = rsample;
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}
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break;
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}
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return outSize;
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}
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#endif /* !HAVE_LIBSAMPLERATE */
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static void
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pcm_convert_channels_1_to_2(int16_t *dest, const int16_t *src,
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unsigned num_frames)
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{
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while (num_frames-- > 0) {
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int16_t value = *src++;
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*dest++ = value;
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*dest++ = value;
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}
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}
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static void
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pcm_convert_channels_2_to_1(int16_t *dest, const int16_t *src,
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unsigned num_frames)
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{
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while (num_frames-- > 0) {
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int32_t a = *src++, b = *src++;
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*dest++ = (a + b) / 2;
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}
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}
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static const int16_t *
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pcm_convertChannels(int8_t channels, const int16_t *inBuffer,
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size_t inSize, size_t *outSize)
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{
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static int16_t *buf;
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static size_t len;
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int16_t *outBuffer = NULL;
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switch (channels) {
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/* convert from 1 -> 2 channels */
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case 1:
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*outSize = (inSize >> 1) << 2;
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if (*outSize > len) {
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len = *outSize;
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buf = xrealloc(buf, len);
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}
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outBuffer = buf;
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pcm_convert_channels_1_to_2((int16_t *)buf,
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(const int16_t *)inBuffer,
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inSize >> 1);
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break;
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/* convert from 2 -> 1 channels */
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case 2:
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*outSize = inSize >> 1;
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if (*outSize > len) {
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len = *outSize;
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buf = xrealloc(buf, len);
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}
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outBuffer = buf;
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pcm_convert_channels_2_to_1((int16_t *)buf,
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(const int16_t *)inBuffer,
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inSize >> 2);
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break;
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default:
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ERROR("only 1 or 2 channels are supported for conversion!\n");
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}
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return outBuffer;
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}
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static void
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pcm_convert_8_to_16(int16_t *out, const int8_t *in,
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unsigned num_samples)
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{
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while (num_samples > 0) {
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*out++ = *in++ << 8;
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--num_samples;
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}
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}
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static void
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pcm_convert_24_to_16(int16_t *out, const int32_t *in,
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unsigned num_samples)
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{
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while (num_samples > 0) {
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*out++ = *in++ >> 8;
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--num_samples;
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}
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}
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static const int16_t *
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pcm_convertTo16bit(uint8_t bits, const void *inBuffer,
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size_t inSize, size_t *outSize)
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{
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static int16_t *buf;
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static size_t len;
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unsigned num_samples;
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switch (bits) {
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case 8:
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num_samples = inSize;
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*outSize = inSize << 1;
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if (*outSize > len) {
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len = *outSize;
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buf = xrealloc(buf, len);
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}
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pcm_convert_8_to_16((int16_t *)buf,
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(const int8_t *)inBuffer,
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num_samples);
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return buf;
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case 16:
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*outSize = inSize;
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return inBuffer;
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case 24:
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num_samples = inSize / 4;
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*outSize = num_samples * 2;
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if (*outSize > len) {
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len = *outSize;
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buf = xrealloc(buf, len);
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}
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pcm_convert_24_to_16((int16_t *)buf,
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(const int32_t *)inBuffer,
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num_samples);
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return buf;
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}
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ERROR("only 8 or 16 bits are supported for conversion!\n");
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return NULL;
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}
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/* outFormat bits must be 16 and channels must be 1 or 2! */
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size_t pcm_convertAudioFormat(const struct audio_format *inFormat,
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const char *inBuffer, size_t inSize,
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const struct audio_format *outFormat,
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char *outBuffer, ConvState *convState)
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{
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const int16_t *buf;
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size_t len = 0;
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size_t outSize = pcm_sizeOfConvBuffer(inFormat, inSize, outFormat);
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assert(outFormat->bits == 16);
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assert(outFormat->channels == 2 || outFormat->channels == 1);
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/* everything else supports 16 bit only, so convert to that first */
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buf = pcm_convertTo16bit(inFormat->bits, inBuffer, inSize, &len);
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if (!buf)
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exit(EXIT_FAILURE);
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if (inFormat->channels != outFormat->channels) {
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buf = pcm_convertChannels(inFormat->channels, buf, len, &len);
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if (!buf)
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exit(EXIT_FAILURE);
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}
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if (inFormat->sample_rate == outFormat->sample_rate) {
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assert(outSize >= len);
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memcpy(outBuffer, buf, len);
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} else {
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len = pcm_convertSampleRate(outFormat->channels,
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inFormat->sample_rate, buf, len,
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outFormat->sample_rate,
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(int16_t*)outBuffer,
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outSize, convState);
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if (len == 0)
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exit(EXIT_FAILURE);
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}
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return len;
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}
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size_t pcm_sizeOfConvBuffer(const struct audio_format *inFormat, size_t inSize,
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const struct audio_format *outFormat)
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{
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const double ratio = (double)outFormat->sample_rate /
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(double)inFormat->sample_rate;
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const int shift = 2 * outFormat->channels;
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size_t outSize = inSize;
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|
|
switch (inFormat->bits) {
|
|
case 8:
|
|
outSize <<= 1;
|
|
break;
|
|
case 16:
|
|
break;
|
|
case 24:
|
|
outSize = (outSize / 4) * 2;
|
|
break;
|
|
default:
|
|
FATAL("only 8 or 16 bits are supported for conversion!\n");
|
|
}
|
|
|
|
if (inFormat->channels != outFormat->channels) {
|
|
switch (inFormat->channels) {
|
|
case 1:
|
|
outSize = (outSize >> 1) << 2;
|
|
break;
|
|
case 2:
|
|
outSize >>= 1;
|
|
break;
|
|
default:
|
|
FATAL("only 1 or 2 channels are supported "
|
|
"for conversion!\n");
|
|
}
|
|
}
|
|
|
|
outSize /= shift;
|
|
outSize = floor(0.5 + (double)outSize * ratio);
|
|
outSize *= shift;
|
|
|
|
return outSize;
|
|
}
|