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-rw-r--r--Makefile.am4
-rw-r--r--NEWS1
-rw-r--r--configure.ac21
-rw-r--r--doc/user.xml32
-rw-r--r--src/output/solaris_output_plugin.c175
-rw-r--r--src/output_list.c4
6 files changed, 237 insertions, 0 deletions
diff --git a/Makefile.am b/Makefile.am
index 007c3f353..28d128249 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -519,6 +519,10 @@ OUTPUT_SRC += \
src/output/httpd_output_plugin.c
endif
+if ENABLE_SOLARIS_OUTPUT
+OUTPUT_SRC += src/output/solaris_output_plugin.c
+endif
+
#
# Sparse code analysis
diff --git a/NEWS b/NEWS
index 4da735343..a5b8ccc52 100644
--- a/NEWS
+++ b/NEWS
@@ -30,6 +30,7 @@ ver 0.15 - (200?/??/??)
- mvp: fixed default device detection
- pipe: new audio output plugin which runs a command
- alsa: better period_time default value for high sample rates
+ - solaris: new audio output plugin for Solaris /dev/audio
* commands:
- "playlistinfo" supports a range now
- added "sticker database", command "sticker", which allows clients
diff --git a/configure.ac b/configure.ac
index ed9703fed..cd14c42f6 100644
--- a/configure.ac
+++ b/configure.ac
@@ -594,6 +594,20 @@ if test x$enable_shout = xyes; then
enable_shout=no)
fi
+case "$host_os" in
+ solaris*)
+ AC_DEFINE(ENABLE_SOLARIS_OUTPUT, 1, [Define to enable Solaris /dev/audio support])
+ enable_solaris_output=yes
+ ;;
+
+ *)
+ enable_solaris_output=no
+ ;;
+esac
+echo "HOST_OS=$host_os"
+
+AM_CONDITIONAL(ENABLE_SOLARIS_OUTPUT, test x$enable_solaris_output = xyes)
+
if test x$enable_oggvorbis_encoder = xyes; then
if test x$enable_oggvorbis = xno; then
AC_MSG_WARN([disabling ogg shout streaming support because vorbis is not enabled])
@@ -1144,6 +1158,12 @@ else
echo " SHOUTcast support .............disabled"
fi
+if test x$enable_solaris_output = xyes; then
+ echo " Solaris /dev/audio support ....enabled"
+else
+ echo " Solaris /dev/audio support ....disabled"
+fi
+
echo ""
if
@@ -1151,6 +1171,7 @@ if
test x$enable_oss = xno &&
test x$enable_shout = xno &&
test x$enable_httpd_output = xno &&
+ test x$enable_solaris_output = xno &&
test x$enable_alsa = xno &&
test x$enable_osx = xno &&
test x$enable_pulse = xno &&
diff --git a/doc/user.xml b/doc/user.xml
index 6837181a9..7276429cd 100644
--- a/doc/user.xml
+++ b/doc/user.xml
@@ -424,6 +424,38 @@ cd mpd-0.14.2</programlisting>
or IceCast server. It forwards tags to this server.
</para>
</section>
+
+ <section>
+ <title><varname>solaris</varname></title>
+
+ <para>
+ The "Solaris" plugin runs only on SUN Solaris, and plays via
+ <filename>/dev/audio</filename>.
+ </para>
+
+ <informaltable>
+ <tgroup cols="2">
+ <thead>
+ <row>
+ <entry>Setting</entry>
+ <entry>Description</entry>
+ </row>
+ </thead>
+ <tbody>
+ <row>
+ <entry>
+ <varname>device</varname>
+ <parameter>PATH</parameter>
+ </entry>
+ <entry>
+ Sets the path of the audio device, defaults to
+ <filename>/dev/audio</filename>.
+ </entry>
+ </row>
+ </tbody>
+ </tgroup>
+ </informaltable>
+ </section>
</section>
</chapter>
</book>
diff --git a/src/output/solaris_output_plugin.c b/src/output/solaris_output_plugin.c
new file mode 100644
index 000000000..5febf0afc
--- /dev/null
+++ b/src/output/solaris_output_plugin.c
@@ -0,0 +1,175 @@
+/*
+ * 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 "output_api.h"
+
+#include <glib.h>
+
+#include <sys/audio.h>
+#include <sys/stropts.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "solaris_output"
+
+struct solaris_output {
+ /* configuration */
+ const char *device;
+
+ int fd;
+};
+
+/**
+ * The quark used for GError.domain.
+ */
+static inline GQuark
+solaris_output_quark(void)
+{
+ return g_quark_from_static_string("solaris_output");
+}
+
+static bool
+solaris_output_test_default_device(void)
+{
+ struct stat st;
+
+ return stat("/dev/audio", &st) == 0 && S_ISCHR(st.st_mode) &&
+ access("/dev/audio", W_OK) == 0;
+}
+
+static void *
+solaris_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+ const struct config_param *param,
+ G_GNUC_UNUSED GError **error)
+{
+ struct solaris_output *so = g_new(struct solaris_output, 1);
+
+ so->device = config_get_block_string(param, "device", "/dev/audio");
+
+ return so;
+}
+
+static void
+solaris_output_finish(void *data)
+{
+ struct solaris_output *so = data;
+
+ g_free(so);
+}
+
+static bool
+solaris_output_open(void *data, struct audio_format *audio_format,
+ GError **error)
+{
+ struct solaris_output *so = data;
+ struct audio_info info;
+ int ret, flags;
+
+ /* support only 16 bit mono/stereo for now; nothing else has
+ been tested */
+ audio_format->bits = 16;
+
+ /* open the device in non-blocking mode */
+
+ so->fd = open(so->device, O_WRONLY|O_NONBLOCK);
+ if (so->fd < 0) {
+ g_set_error(error, solaris_output_quark(), errno,
+ "Failed to open %s: %s",
+ so->device, g_strerror(errno));
+ return false;
+ }
+
+ /* restore blocking mode */
+
+ flags = fcntl(so->fd, F_GETFL);
+ if (flags > 0 && (flags & O_NONBLOCK) != 0)
+ fcntl(so->fd, F_SETFL, flags & ~O_NONBLOCK);
+
+ /* configure the audio device */
+
+ ret = ioctl(so->fd, AUDIO_GETINFO, &info);
+ if (ret < 0) {
+ g_set_error(error, solaris_output_quark(), errno,
+ "AUDIO_GETINFO failed: %s", g_strerror(errno));
+ close(so->fd);
+ return false;
+ }
+
+ info.play.sample_rate = audio_format->sample_rate;
+ info.play.channels = audio_format->channels;
+ info.play.precision = audio_format->bits;
+ info.play.encoding = AUDIO_ENCODING_LINEAR;
+
+ ret = ioctl(so->fd, AUDIO_SETINFO, &info);
+ if (ret < 0) {
+ g_set_error(error, solaris_output_quark(), errno,
+ "AUDIO_SETINFO failed: %s", g_strerror(errno));
+ close(so->fd);
+ return false;
+ }
+
+ return true;
+}
+
+static void
+solaris_output_close(void *data)
+{
+ struct solaris_output *so = data;
+
+ close(so->fd);
+}
+
+static size_t
+solaris_output_play(void *data, const void *chunk, size_t size, GError **error)
+{
+ struct solaris_output *so = data;
+ ssize_t nbytes;
+
+ nbytes = write(so->fd, chunk, size);
+ if (nbytes <= 0) {
+ g_set_error(error, solaris_output_quark(), errno,
+ "Write failed: %s", g_strerror(errno));
+ return 0;
+ }
+
+ return nbytes;
+}
+
+static void
+solaris_output_cancel(void *data)
+{
+ struct solaris_output *so = data;
+
+ ioctl(so->fd, I_FLUSH);
+}
+
+const struct audio_output_plugin solaris_output_plugin = {
+ .name = "solaris",
+ .test_default_device = solaris_output_test_default_device,
+ .init = solaris_output_init,
+ .finish = solaris_output_finish,
+ .open = solaris_output_open,
+ .close = solaris_output_close,
+ .play = solaris_output_play,
+ .cancel = solaris_output_cancel,
+};
diff --git a/src/output_list.c b/src/output_list.c
index f2e107584..f1569abf9 100644
--- a/src/output_list.c
+++ b/src/output_list.c
@@ -29,6 +29,7 @@ extern const struct audio_output_plugin alsaPlugin;
extern const struct audio_output_plugin ao_output_plugin;
extern const struct audio_output_plugin oss_output_plugin;
extern const struct audio_output_plugin osxPlugin;
+extern const struct audio_output_plugin solaris_output_plugin;
extern const struct audio_output_plugin pulse_plugin;
extern const struct audio_output_plugin mvp_output_plugin;
extern const struct audio_output_plugin jackPlugin;
@@ -57,6 +58,9 @@ const struct audio_output_plugin *audio_output_plugins[] = {
#ifdef HAVE_OSX
&osxPlugin,
#endif
+#ifdef ENABLE_SOLARIS_OUTPUT
+ &solaris_output_plugin,
+#endif
#ifdef HAVE_PULSE
&pulse_plugin,
#endif