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/*
* Copyright (C) 2003-2010 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 "pcm_mix.h"
#include "pcm_volume.h"
#include "pcm_utils.h"
#include "audio_format.h"
#include <glib.h>
#include <math.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "pcm"
static void
pcm_add_vol_8(int8_t *buffer1, const int8_t *buffer2,
unsigned num_samples, int volume1, int volume2)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 8);
--num_samples;
}
}
static void
pcm_add_vol_16(int16_t *buffer1, const int16_t *buffer2,
unsigned num_samples, int volume1, int volume2)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 16);
--num_samples;
}
}
static void
pcm_add_vol_24(int32_t *buffer1, const int32_t *buffer2,
unsigned num_samples, unsigned volume1, unsigned volume2)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 24);
--num_samples;
}
}
static void
pcm_add_vol_32(int32_t *buffer1, const int32_t *buffer2,
unsigned num_samples, unsigned volume1, unsigned volume2)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range_64(sample1, 32);
--num_samples;
}
}
static void
pcm_add_vol(void *buffer1, const void *buffer2, size_t size,
int vol1, int vol2,
const struct audio_format *format)
{
switch (format->format) {
case SAMPLE_FORMAT_S8:
pcm_add_vol_8((int8_t *)buffer1, (const int8_t *)buffer2,
size, vol1, vol2);
break;
case SAMPLE_FORMAT_S16:
pcm_add_vol_16((int16_t *)buffer1, (const int16_t *)buffer2,
size / 2, vol1, vol2);
break;
case SAMPLE_FORMAT_S24_P32:
pcm_add_vol_24((int32_t *)buffer1, (const int32_t *)buffer2,
size / 4, vol1, vol2);
break;
case SAMPLE_FORMAT_S32:
pcm_add_vol_32((int32_t *)buffer1, (const int32_t *)buffer2,
size / 4, vol1, vol2);
break;
default:
g_error("format %s not supported by pcm_add_vol",
sample_format_to_string(format->format));
}
}
static void
pcm_add_8(int8_t *buffer1, const int8_t *buffer2, unsigned num_samples)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 += sample2;
*buffer1++ = pcm_range(sample1, 8);
--num_samples;
}
}
static void
pcm_add_16(int16_t *buffer1, const int16_t *buffer2, unsigned num_samples)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 += sample2;
*buffer1++ = pcm_range(sample1, 16);
--num_samples;
}
}
static void
pcm_add_24(int32_t *buffer1, const int32_t *buffer2, unsigned num_samples)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 += sample2;
*buffer1++ = pcm_range(sample1, 24);
--num_samples;
}
}
static void
pcm_add_32(int32_t *buffer1, const int32_t *buffer2, unsigned num_samples)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 += sample2;
*buffer1++ = pcm_range_64(sample1, 32);
--num_samples;
}
}
static void
pcm_add(void *buffer1, const void *buffer2, size_t size,
const struct audio_format *format)
{
switch (format->format) {
case SAMPLE_FORMAT_S8:
pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2, size);
break;
case SAMPLE_FORMAT_S16:
pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2, size / 2);
break;
case SAMPLE_FORMAT_S24_P32:
pcm_add_24((int32_t *)buffer1, (const int32_t *)buffer2, size / 4);
break;
case SAMPLE_FORMAT_S32:
pcm_add_32((int32_t *)buffer1, (const int32_t *)buffer2, size / 4);
break;
default:
g_error("format %s not supported by pcm_add",
sample_format_to_string(format->format));
}
}
void
pcm_mix(void *buffer1, const void *buffer2, size_t size,
const struct audio_format *format, float portion1)
{
int vol1;
float s;
/* portion1 is between 0.0 and 1.0 for crossfading, MixRamp uses NaN
* to signal mixing rather than fading */
if (isnan(portion1)) {
pcm_add(buffer1, buffer2, size, format);
return;
}
s = sin(M_PI_2 * portion1);
s *= s;
vol1 = s * PCM_VOLUME_1 + 0.5;
vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1);
pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
}
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