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authorMax Kellermann <max@duempel.org>2012-03-21 19:54:35 +0100
committerMax Kellermann <max@duempel.org>2012-03-21 21:16:50 +0100
commit170635e3a6fbbdb2274f6ecd164c2c363d4e80b4 (patch)
tree1065b6a3651ff07416835f1bdc42a32f9befe87a
parenta5d1444ef480952c73b429d43aadb943925e5aa1 (diff)
downloadmpd-170635e3a6fbbdb2274f6ecd164c2c363d4e80b4.tar.gz
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output/{alsa,oss}: move endian code to new library pcm_export
Diffstat (limited to '')
-rw-r--r--Makefile.am1
-rw-r--r--src/output/alsa_output_plugin.c41
-rw-r--r--src/output/oss_output_plugin.c43
-rw-r--r--src/pcm_export.c54
-rw-r--r--src/pcm_export.h90
5 files changed, 168 insertions, 61 deletions
diff --git a/Makefile.am b/Makefile.am
index 7a59cb6a9..8a014f13e 100644
--- a/Makefile.am
+++ b/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 \
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index 00312c435..7714e8a66 100644
--- a/src/output/alsa_output_plugin.c
+++ b/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;
diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c
index ea15022d9..1b09c4b53 100644
--- a/src/output/oss_output_plugin.c
+++ b/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) {
diff --git a/src/pcm_export.c b/src/pcm_export.c
new file mode 100644
index 000000000..4c08394ea
--- /dev/null
+++ b/src/pcm_export.c
@@ -0,0 +1,54 @@
+/*
+ * 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;
+}
diff --git a/src/pcm_export.h b/src/pcm_export.h
new file mode 100644
index 000000000..198b1d6a8
--- /dev/null
+++ b/src/pcm_export.h
@@ -0,0 +1,90 @@
+/*
+ * 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