Add reverse_endian field to struct audio_format and handle conversion

Makefile.am

@@ -114,6 +114,7 @@ mpd_headers = \
 	src/pcm_convert.h \
 	src/pcm_volume.h \
 	src/pcm_mix.h \
+	src/pcm_byteswap.h \
 	src/pcm_channels.h \
 	src/pcm_format.h \
 	src/pcm_resample.h \
@@ -218,6 +219,7 @@ src_mpd_SOURCES = \
 	src/pcm_convert.c \
 	src/pcm_volume.c \
 	src/pcm_mix.c \
+	src/pcm_byteswap.c \
 	src/pcm_channels.c \
 	src/pcm_format.c \
 	src/pcm_resample.c \
@@ -708,7 +710,7 @@ test_run_filter_SOURCES = test/run_filter.c \
 	src/filter_plugin.c \
 	src/filter_registry.c \
 	src/conf.c src/buffer2array.c src/utils.c \
-	src/pcm_volume.c src/pcm_convert.c \
+	src/pcm_volume.c src/pcm_convert.c src/pcm_byteswap.c \
 	src/pcm_format.c src/pcm_channels.c src/pcm_dither.c \
 	src/pcm_resample.c src/pcm_resample_fallback.c \
 	src/audio_parser.c \

src/audio_format.h

@@ -27,6 +27,7 @@ struct audio_format {
 	uint32_t sample_rate;
 	uint8_t bits;
 	uint8_t channels;
+	uint8_t reverse_endian;
 };
 
 static inline void audio_format_clear(struct audio_format *af)
@@ -34,6 +35,7 @@ static inline void audio_format_clear(struct audio_format *af)
 	af->sample_rate = 0;
 	af->bits = 0;
 	af->channels = 0;
+	af->reverse_endian = 0;
 }
 
 static inline void audio_format_init(struct audio_format *af,
@@ -97,7 +99,8 @@ static inline bool audio_format_equals(const struct audio_format *a,
 {
 	return a->sample_rate == b->sample_rate &&
 		a->bits == b->bits &&
-		a->channels == b->channels;
+		a->channels == b->channels &&
+		a->reverse_endian == b->reverse_endian;
 }
 
 /**

src/filter/convert_filter_plugin.c

@@ -149,6 +149,7 @@ convert_filter_set(struct filter *_filter,
 	assert(audio_format_valid(&filter->out_audio_format));
 	assert(out_audio_format != NULL);
 	assert(audio_format_valid(out_audio_format));
+	assert(filter->in_audio_format.reverse_endian == 0);
 
 	filter->out_audio_format = *out_audio_format;
 }

src/output_thread.c

@@ -93,18 +93,20 @@ ao_open(struct audio_output *ao)
 	g_mutex_unlock(ao->mutex);
 
 	g_debug("opened plugin=%s name=\"%s\" "
-		"audio_format=%u:%u:%u",
+		"audio_format=%u:%u:%u:%u",
 		ao->plugin->name, ao->name,
 		ao->out_audio_format.sample_rate,
 		ao->out_audio_format.bits,
-		ao->out_audio_format.channels);
+		ao->out_audio_format.channels,
+		ao->out_audio_format.reverse_endian);
 
 	if (!audio_format_equals(&ao->in_audio_format,
 				 &ao->out_audio_format))
-		g_debug("converting from %u:%u:%u",
+		g_debug("converting from %u:%u:%u:%u",
 			ao->in_audio_format.sample_rate,
 			ao->in_audio_format.bits,
-			ao->in_audio_format.channels);
+			ao->in_audio_format.channels,
+			ao->in_audio_format.reverse_endian);
 }
 
 static void

src/pcm_byteswap.c

+/*
+ * Copyright (C) 2003-2009 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 "pcm_byteswap.h"
+#include "pcm_buffer.h"
+
+#include <glib.h>
+
+#include <assert.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "pcm"
+
+static inline uint16_t swab16(uint16_t x)
+{
+	return (x << 8) | (x >> 8);
+}
+
+const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
+			       const int16_t *src, size_t len)
+{
+	unsigned i;
+	int16_t *buf = pcm_buffer_get(buffer, len);
+
+	if (!buf)
+		return NULL;
+
+	for (i = 0; i < len / 2; i++)
+		buf[i] = swab16(src[i]);
+
+	return buf;
+}
+
+static inline uint32_t swab32(uint32_t x)
+{
+	return (x << 24) | 
+		((x & 0xff00) << 8) |
+		((x & 0xff0000) >> 8) |
+		(x >> 24);
+}
+
+const int32_t *pcm_byteswap_32(struct pcm_buffer *buffer,
+			       const int32_t *src, size_t len)
+{
+	unsigned i;
+	int32_t *buf = pcm_buffer_get(buffer, len);
+
+	if (!buf)
+		return NULL;
+
+	for (i = 0; i < len / 4; i++)
+		buf[i] = swab32(src[i]);
+
+	return buf;
+}

src/pcm_byteswap.h

+/*
+ * Copyright (C) 2003-2009 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.
+ */
+
+#ifndef MPD_PCM_BYTESWAP_H
+#define MPD_PCM_BYTESWAP_H
+
+#include <stdint.h>
+#include <stddef.h>
+
+struct pcm_buffer;
+
+/**
+ * Changes the endianness of 16 bit PCM data.
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @return the destination buffer
+ */
+const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
+			       const int16_t *src, size_t len);
+
+/**
+ * Changes the endianness of 32-bit (or 24-bit) PCM data.
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @return the destination buffer
+ */
+const int32_t *pcm_byteswap_32(struct pcm_buffer *buffer,
+			       const int32_t *src, size_t len);
+
+#endif

src/pcm_convert.c

@@ -20,6 +20,7 @@
 #include "pcm_convert.h"
 #include "pcm_channels.h"
 #include "pcm_format.h"
+#include "pcm_byteswap.h"
 #include "audio_format.h"
 
 #include <assert.h>
@@ -83,6 +84,12 @@ pcm_convert_16(struct pcm_convert_state *state,
 				      dest_format->sample_rate,
 				      &len);
 
+	if (dest_format->reverse_endian) {
+		buf = pcm_byteswap_16(&state->format_buffer, buf, len);
+		if (!buf)
+			g_error("pcm_byteswap_16() failed");
+	}
+
 	*dest_size_r = len;
 	return buf;
 }
@@ -120,6 +127,12 @@ pcm_convert_24(struct pcm_convert_state *state,
 				      dest_format->sample_rate,
 				      &len);
 
+	if (dest_format->reverse_endian) {
+		buf = pcm_byteswap_32(&state->format_buffer, buf, len);
+		if (!buf)
+			g_error("pcm_byteswap_32() failed");
+	}
+
 	*dest_size_r = len;
 	return buf;
 }
@@ -157,6 +170,12 @@ pcm_convert_32(struct pcm_convert_state *state,
 				      dest_format->sample_rate,
 				      &len);
 
+	if (dest_format->reverse_endian) {
+		buf = pcm_byteswap_32(&state->format_buffer, buf, len);
+		if (!buf)
+			g_error("pcm_byteswap_32() failed");
+	}
+
 	*dest_size_r = len;
 	return buf;
 }