output/{alsa,oss}: move endian code to new library pcm_export

Makefile.am

@@ -399,6 +399,7 @@ libutil_a_SOURCES = \
 
 libpcm_a_SOURCES = \
 	src/pcm_buffer.c src/pcm_buffer.h \
+	src/pcm_export.c src/pcm_export.h \
 	src/pcm_convert.c src/pcm_convert.h \
 	src/dsd2pcm/dsd2pcm.c src/dsd2pcm/dsd2pcm.h \
 	src/pcm_dsd.c src/pcm_dsd.h \

src/output/alsa_output_plugin.c

@@ -21,8 +21,7 @@
 #include "alsa_output_plugin.h"
 #include "output_api.h"
 #include "mixer_list.h"
-#include "pcm_buffer.h"
-#include "pcm_byteswap.h"
+#include "pcm_export.h"
 
 #include <glib.h>
 #include <alsa/asoundlib.h>
@@ -47,12 +46,7 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
 struct alsa_data {
 	struct audio_output base;
 
-	/**
-	 * The buffer used to reverse the byte order.
-	 *
-	 * @see #reverse_endian
-	 */
-	struct pcm_buffer reverse_buffer;
+	struct pcm_export_state export;
 
 	/** the configured name of the ALSA device; NULL for the
 	    default device */
@@ -61,21 +55,6 @@ struct alsa_data {
 	/** use memory mapped I/O? */
 	bool use_mmap;
 
-	/**
-	 * Does ALSA expect samples in reverse byte order? (i.e. not
-	 * host byte order)
-	 *
-	 * This attribute is only valid while the device is open.
-	 */
-	bool reverse_endian;
-
-	/**
-	 * Which sample format is being sent to the play() method?
-	 *
-	 * This attribute is only valid while the device is open.
-	 */
-	enum sample_format sample_format;
-
 	/** libasound's buffer_time setting (in microseconds) */
 	unsigned int buffer_time;
 
@@ -196,7 +175,7 @@ alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
 {
 	struct alsa_data *ad = (struct alsa_data *)ao;
 
-	pcm_buffer_init(&ad->reverse_buffer);
+	pcm_export_init(&ad->export);
 	return true;
 }
 
@@ -205,7 +184,7 @@ alsa_output_disable(struct audio_output *ao)
 {
 	struct alsa_data *ad = (struct alsa_data *)ao;
 
-	pcm_buffer_deinit(&ad->reverse_buffer);
+	pcm_export_deinit(&ad->export);
 }
 
 static bool
@@ -434,8 +413,9 @@ configure_hw:
 		ad->writei = snd_pcm_writei;
 	}
 
+	bool reverse_endian;
 	err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
-				       &ad->reverse_endian);
+				       &reverse_endian);
 	if (err < 0) {
 		g_set_error(error, alsa_output_quark(), err,
 			    "ALSA device \"%s\" does not support format %s: %s",
@@ -445,8 +425,6 @@ configure_hw:
 		return false;
 	}
 
-	ad->sample_format = audio_format->format;
-
 	err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
 						  &channels);
 	if (err < 0) {
@@ -579,6 +557,9 @@ configure_hw:
 	ad->period_frames = alsa_period_size;
 	ad->period_position = 0;
 
+	pcm_export_open(&ad->export, audio_format->format,
+			reverse_endian);
+
 	return true;
 
 error:
@@ -710,9 +691,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 {
 	struct alsa_data *ad = (struct alsa_data *)ao;
 
-	if (ad->reverse_endian)
-		chunk = pcm_byteswap(&ad->reverse_buffer, ad->sample_format,
-				     chunk, size);
+	chunk = pcm_export(&ad->export, chunk, size, &size);
 
 	size /= ad->frame_size;
 

src/output/oss_output_plugin.c

@@ -52,20 +52,14 @@
 #endif
 
 #ifdef AFMT_S24_PACKED
-#include "pcm_buffer.h"
-#include "pcm_byteswap.h"
+#include "pcm_export.h"
 #endif
 
 struct oss_data {
 	struct audio_output base;
 
 #ifdef AFMT_S24_PACKED
-	/**
-	 * The buffer used to reverse the byte order.
-	 *
-	 * @see #reverse_endian
-	 */
-	struct pcm_buffer reverse_buffer;
+	struct pcm_export_state export;
 #endif
 
 	int fd;
@@ -76,16 +70,6 @@ struct oss_data {
 	 * the device after cancel().
 	 */
 	struct audio_format audio_format;
-
-#ifdef AFMT_S24_PACKED
-	/**
-	 * Does OSS expect samples in reverse byte order? (i.e. not
-	 * host byte order)
-	 *
-	 * This attribute is only valid while the device is open.
-	 */
-	bool reverse_endian;
-#endif
 };
 
 /**
@@ -252,7 +236,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
 {
 	struct oss_data *od = (struct oss_data *)ao;
 
-	pcm_buffer_init(&od->reverse_buffer);
+	pcm_export_init(&od->export);
 	return true;
 }
 
@@ -261,7 +245,7 @@ oss_output_disable(struct audio_output *ao)
 {
 	struct oss_data *od = (struct oss_data *)ao;
 
-	pcm_buffer_deinit(&od->reverse_buffer);
+	pcm_export_deinit(&od->export);
 }
 
 #endif
@@ -517,7 +501,7 @@ sample_format_from_oss(int format)
 static bool
 oss_setup_sample_format(int fd, struct audio_format *audio_format,
 #ifdef AFMT_S24_PACKED
-			bool *reverse_endian_r,
+			struct pcm_export_state *export,
 #endif
 			GError **error_r)
 {
@@ -537,8 +521,9 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 		audio_format->format = mpd_format;
 
 #ifdef AFMT_S24_PACKED
-		*reverse_endian_r = oss_format == AFMT_S24_PACKED &&
-			G_BYTE_ORDER != G_LITTLE_ENDIAN;
+		pcm_export_open(export, mpd_format,
+				oss_format == AFMT_S24_PACKED &&
+				G_BYTE_ORDER != G_LITTLE_ENDIAN);
 #endif
 		return true;
 
@@ -583,8 +568,9 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 			audio_format->format = mpd_format;
 
 #ifdef AFMT_S24_PACKED
-			*reverse_endian_r = oss_format == AFMT_S24_PACKED &&
-				G_BYTE_ORDER != G_LITTLE_ENDIAN;
+			pcm_export_open(export, mpd_format,
+					oss_format == AFMT_S24_PACKED &&
+					G_BYTE_ORDER != G_LITTLE_ENDIAN);
 #endif
 			return true;
 
@@ -611,7 +597,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
 		oss_setup_sample_rate(od->fd, audio_format, error_r) &&
 		oss_setup_sample_format(od->fd, audio_format,
 #ifdef AFMT_S24_PACKED
-					&od->reverse_endian,
+					&od->export,
 #endif
 					error_r);
 }
@@ -726,10 +712,7 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		return 0;
 
 #ifdef AFMT_S24_PACKED
-	if (od->reverse_endian)
-		chunk = pcm_byteswap(&od->reverse_buffer,
-				     od->audio_format.format,
-				     chunk, size);
+	chunk = pcm_export(&od->export, chunk, size, &size);
 #endif
 
 	while (true) {

src/pcm_export.c

+/*
+ * Copyright (C) 2003-2012 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_export.h"
+#include "pcm_byteswap.h"
+
+void
+pcm_export_init(struct pcm_export_state *state)
+{
+	pcm_buffer_init(&state->reverse_buffer);
+}
+
+void pcm_export_deinit(struct pcm_export_state *state)
+{
+	pcm_buffer_deinit(&state->reverse_buffer);
+}
+
+void
+pcm_export_open(struct pcm_export_state *state,
+		enum sample_format sample_format,
+		bool reverse_endian)
+{
+	state->sample_format = sample_format;
+	state->reverse_endian = reverse_endian;
+}
+
+const void *
+pcm_export(struct pcm_export_state *state, const void *data, size_t size,
+	   size_t *dest_size_r)
+{
+	if (state->reverse_endian)
+		data = pcm_byteswap(&state->reverse_buffer,
+				    state->sample_format, data, size);
+
+	*dest_size_r = size;
+	return data;
+}

src/pcm_export.h

+/*
+ * Copyright (C) 2003-2012 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 PCM_EXPORT_H
+#define PCM_EXPORT_H
+
+#include "check.h"
+#include "pcm_buffer.h"
+#include "audio_format.h"
+
+#include <stdbool.h>
+
+struct audio_format;
+
+/**
+ * An object that handles export of PCM samples to some instance
+ * outside of MPD.  It has a few more options to tweak the binary
+ * representation which are not supported by the pcm_convert library.
+ */
+struct pcm_export_state {
+	/**
+	 * The buffer used to reverse the byte order.
+	 *
+	 * @see #reverse_endian
+	 */
+	struct pcm_buffer reverse_buffer;
+
+	enum sample_format sample_format;
+
+	/**
+	 * Export the samples in reverse byte order?
+	 */
+	bool reverse_endian;
+};
+
+/**
+ * Initialize a #pcm_export_state object.
+ */
+void
+pcm_export_init(struct pcm_export_state *state);
+
+/**
+ * Deinitialize a #pcm_export_state object and free allocated memory.
+ */
+void
+pcm_export_deinit(struct pcm_export_state *state);
+
+/**
+ * Open the #pcm_export_state object.
+ *
+ * There is no "close" method.  This function may be called multiple
+ * times to reuse the object, until pcm_export_deinit() is called.
+ *
+ * This function cannot fail.
+ */
+void
+pcm_export_open(struct pcm_export_state *state,
+		enum sample_format sample_format,
+		bool reverse_endian);
+
+/**
+ * Export a PCM buffer.
+ *
+ * @param state an initialized and open pcm_export_state object
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer (may be a pointer to the source buffer)
+ */
+const void *
+pcm_export(struct pcm_export_state *state, const void *src, size_t src_size,
+	   size_t *dest_size_r);
+
+#endif