/* * 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); }