238 lines
5.6 KiB
C
238 lines
5.6 KiB
C
/*
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* Copyright (C) 2003-2010 The Music Player Daemon Project
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* 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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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "pcm_mix.h"
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#include "pcm_volume.h"
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#include "pcm_utils.h"
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#include "audio_format.h"
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#include <glib.h>
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#include <math.h>
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#undef G_LOG_DOMAIN
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#define G_LOG_DOMAIN "pcm"
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static void
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pcm_add_vol_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_volume_dither() + PCM_VOLUME_1 / 2)
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/ PCM_VOLUME_1;
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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_vol_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_volume_dither() + PCM_VOLUME_1 / 2)
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/ PCM_VOLUME_1;
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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_vol_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_volume_dither() + PCM_VOLUME_1 / 2)
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/ PCM_VOLUME_1;
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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
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pcm_add_vol_32(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_volume_dither() + PCM_VOLUME_1 / 2)
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/ PCM_VOLUME_1;
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*buffer1++ = pcm_range_64(sample1, 32);
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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_vol(void *buffer1, const void *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->format) {
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case SAMPLE_FORMAT_S8:
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pcm_add_vol_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 SAMPLE_FORMAT_S16:
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pcm_add_vol_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 SAMPLE_FORMAT_S24_P32:
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pcm_add_vol_24((int32_t *)buffer1, (const int32_t *)buffer2,
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size / 4, vol1, vol2);
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break;
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case SAMPLE_FORMAT_S32:
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pcm_add_vol_32((int32_t *)buffer1, (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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g_error("format %s not supported by pcm_add_vol",
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sample_format_to_string(format->format));
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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, unsigned num_samples)
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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 += sample2;
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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, unsigned num_samples)
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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 += sample2;
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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, unsigned num_samples)
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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 += sample2;
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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
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pcm_add_32(int32_t *buffer1, const int32_t *buffer2, unsigned num_samples)
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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 += sample2;
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*buffer1++ = pcm_range_64(sample1, 32);
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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(void *buffer1, const void *buffer2, size_t size,
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const struct audio_format *format)
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{
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switch (format->format) {
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case SAMPLE_FORMAT_S8:
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pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2, size);
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break;
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case SAMPLE_FORMAT_S16:
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pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2, size / 2);
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break;
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case SAMPLE_FORMAT_S24_P32:
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pcm_add_24((int32_t *)buffer1, (const int32_t *)buffer2, size / 4);
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break;
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case SAMPLE_FORMAT_S32:
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pcm_add_32((int32_t *)buffer1, (const int32_t *)buffer2, size / 4);
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break;
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default:
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g_error("format %s not supported by pcm_add",
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sample_format_to_string(format->format));
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}
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}
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void
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pcm_mix(void *buffer1, const void *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;
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/* portion1 is between 0.0 and 1.0 for crossfading, MixRamp uses NaN
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* to signal mixing rather than fading */
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if (isnan(portion1)) {
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pcm_add(buffer1, buffer2, size, format);
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return;
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}
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s = sin(M_PI_2 * portion1);
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s *= s;
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vol1 = s * PCM_VOLUME_1 + 0.5;
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vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1);
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pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
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}
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