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-rw-r--r--src/pcm_mix.c160
1 files changed, 138 insertions, 22 deletions
diff --git a/src/pcm_mix.c b/src/pcm_mix.c
index d1e716731..3145c07be 100644
--- a/src/pcm_mix.c
+++ b/src/pcm_mix.c
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * 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
@@ -17,10 +17,12 @@
* 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 "mpd_error.h"
#include <glib.h>
@@ -30,8 +32,8 @@
#define G_LOG_DOMAIN "pcm"
static void
-pcm_add_8(int8_t *buffer1, const int8_t *buffer2,
- unsigned num_samples, int volume1, int volume2)
+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;
@@ -47,8 +49,8 @@ pcm_add_8(int8_t *buffer1, const int8_t *buffer2,
}
static void
-pcm_add_16(int16_t *buffer1, const int16_t *buffer2,
- unsigned num_samples, int volume1, int volume2)
+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;
@@ -64,8 +66,8 @@ pcm_add_16(int16_t *buffer1, const int16_t *buffer2,
}
static void
-pcm_add_24(int32_t *buffer1, const int32_t *buffer2,
- unsigned num_samples, unsigned volume1, unsigned volume2)
+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;
@@ -81,29 +83,134 @@ pcm_add_24(int32_t *buffer1, const int32_t *buffer2,
}
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:
+ MPD_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,
- int vol1, int vol2,
const struct audio_format *format)
{
- switch (format->bits) {
- case 8:
- pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2,
- size, vol1, vol2);
+ switch (format->format) {
+ case SAMPLE_FORMAT_S8:
+ pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2, size);
break;
- case 16:
- pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2,
- size / 2, vol1, vol2);
+ case SAMPLE_FORMAT_S16:
+ pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2, size / 2);
break;
- case 24:
- pcm_add_24((int32_t*)buffer1,
- (const int32_t*)buffer2,
- size / 4, vol1, vol2);
+ 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("%u bits not supported by pcm_add!\n", format->bits);
+ MPD_ERROR("format %s not supported by pcm_add",
+ sample_format_to_string(format->format));
}
}
@@ -112,11 +219,20 @@ pcm_mix(void *buffer1, const void *buffer2, size_t size,
const struct audio_format *format, float portion1)
{
int vol1;
- float s = sin(M_PI_2 * portion1);
+ 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(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
+ pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
}