<URL: http://bugs.freeciv.org/Ticket/Display.html?id=40582 >
Hello all,
I tried to make a GStreamer audio plugin for Freeciv. It's a work in
progress but it already works well. It's my first contribution so,
please let me know if there are any rules I break.
I'm working on a Linux machine so I didn't test this on other
platforms but because GStreamer can run on all the other supported
platforms it should not be a problem I think.
Gr,
Thijs
---
client/Makefile.am | 20 ++++-
client/audio.c | 17 ++++-
client/audio_gst.c | 180 +++++++++++++++++++++++++++++++++++++++++++++++++++++
client/audio_gst.h | 18 +++++
configure.ac | 13 +++
5 files changed, 240 insertions(+), 8 deletions(-)
--- freeciv.orig/client/Makefile.am
+++ freeciv/client/Makefile.am
@@ -33,11 +33,17 @@
AUDIO_SDL_FILES=$(ALL_AUDIO_SDL_FILES)
endif
+ALL_AUDIO_GST_FILES=audio_gst.c audio_gst.h
+
+if AUDIO_GST
+AUDIO_GST_FILES=$(ALL_AUDIO_GST_FILES)
+endif
+
if MINGW32
CLIENTICON=../win32/clienticon.o
endif
-EXTRA_DIST= $(ALL_AUDIO_SDL_FILES)
+EXTRA_DIST= $(ALL_AUDIO_SDL_FILES) $(ALL_AUDIO_GST_FILES)
## This is usually false, so "include" is not recursed into
## by 'make', but it can be enabled in configure, and it is
@@ -53,9 +59,10 @@
bin_PROGRAMS = civclient
-AM_CPPFLAGS = -I$(top_srcdir)/utility -I$(srcdir)/include -I$(top_srcdir)/common -I$(top_srcdir)/common/aicore -I$(srcdir)/agents $(CLIENT_CFLAGS) $(SOUND_CFLAGS) $(LIBGGZ_INCLUDES) $(GGZMOD_INCLUDES) $(GGZ_GTK_INCLUDES)
+AM_CPPFLAGS = -I$(top_srcdir)/utility -I$(srcdir)/include -I$(top_srcdir)/common -I$(top_srcdir)/common/aicore -I$(srcdir)/agents $(CLIENT_CFLAGS) $(SOUND_CFLAGS) $(LIBGGZ_INCLUDES) $(GGZMOD_INCLUDES) $(GGZ_GTK_INCLUDES) $(GSTREAMER_CFLAGS)
civclient_SOURCES = $(AUDIO_SDL_FILES) \
+ $(AUDIO_GST_FILES) \
attribute.h \
attribute.c \
citydlg_common.c \
@@ -114,9 +121,9 @@
themes_common.h \
tilespec.c \
tilespec.h \
- audio.c \
- audio.h \
- audio_none.c \
+ audio.c \
+ audio.h \
+ audio_none.c \
audio_none.h
# packhand_gen.c & packhand_gen.h are generated files, but as they are
@@ -132,7 +139,8 @@
$(gui_sources)/libguiclient.a
civclient_DEPENDENCIES = $(fc_civclient_libs)
civclient_LDADD = $(fc_civclient_libs) $(fc_civclient_libs) \
- $(INTLLIBS) $(CLIENT_LIBS) $(SOUND_LIBS) $(LIB_GGZMOD) $(CLIENTICON)
+ $(INTLLIBS) $(CLIENT_LIBS) $(SOUND_LIBS) $(LIB_GGZMOD) $(CLIENTICON) \
+ $(GSTREAMER_LIBS)
desktopfiledir = $(prefix)/share/applications
desktopfile_DATA = \
freeciv.desktop
--- freeciv.orig/client/audio.c
+++ freeciv/client/audio.c
@@ -29,13 +29,18 @@
#include "support.h"
#include "audio_none.h"
+
+#ifdef AUDIO_GST
+#include "audio_gst.h"
+#endif
+
#ifdef AUDIO_SDL
#include "audio_sdl.h"
#endif
#include "audio.h"
-#define MAX_NUM_PLUGINS 2
+#define MAX_NUM_PLUGINS 3
#define SNDSPEC_SUFFIX ".soundspec"
/* keep it open throughout */
@@ -142,6 +147,10 @@
assert(num_plugins_used == 1);
selected_plugin = 0;
+#ifdef AUDIO_GST
+ audio_gst_init ();
+#endif
+
#ifdef AUDIO_SDL
audio_sdl_init();
#endif
@@ -248,8 +257,12 @@
return;
}
+#ifdef AUDIO_GST
+ if (audio_select_plugin ("gst")) return;
+#endif
+
#ifdef AUDIO_SDL
- if (audio_select_plugin("sdl")) return;
+ if (audio_select_plugin("sdl")) return;
#endif
freelog(LOG_NORMAL, _("No real audio subsystem managed to initialize!"));
freelog(LOG_NORMAL,
--- /dev/null
+++ freeciv/client/audio_gst.c
@@ -0,0 +1,180 @@
+/**********************************************************************
+ Freeciv - Copyright (C) 2008 Thijs Vermeir <[EMAIL PROTECTED]>
+ 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, 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.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <string.h>
+
+#include "support.h"
+
+#include "audio.h"
+#include "gui_main_g.h"
+
+#include "audio_gst.h"
+
+#include <gst/gst.h>
+
+static void my_stop(void);
+static void my_wait(void);
+
+GstElement *pipeline;
+GstElement *playbin;
+gchar *last_fullpath;
+GMutex *audio_lock;
+
+static void my_shutdown(void)
+{
+ GstState new_state;
+ GstStateChangeReturn ret;
+
+ gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
+ gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+
+ gst_object_unref (pipeline);
+ gst_object_unref (playbin);
+ if (last_fullpath)
+ g_free (last_fullpath);
+ if (audio_lock)
+ g_mutex_free (audio_lock);
+}
+
+static void my_stop(void)
+{
+ g_mutex_lock (audio_lock);
+ gst_element_set_state (pipeline, GST_STATE_READY);
+ g_mutex_unlock (audio_lock);
+}
+
+static void my_wait(void)
+{
+ GstState new_state;
+ GstStateChangeReturn ret;
+
+ g_mutex_lock (audio_lock);
+ ret = gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+ while (ret == GST_STATE_CHANGE_SUCCESS && new_state == GST_STATE_PLAYING) {
+ ret = gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+ }
+ g_mutex_unlock (audio_lock);
+}
+
+static bool my_play(const char *const tag, const char *const fullpath,
+ bool repeat)
+{
+ if (fullpath) {
+ gboolean last_fullpath_changed = FALSE;
+ GstState new_state;
+ GstStateChangeReturn ret;
+
+ g_mutex_lock (audio_lock);
+
+ if (!last_fullpath || g_strcmp0 (fullpath, last_fullpath) != 0) {
+ if (last_fullpath)
+ g_free (last_fullpath);
+ last_fullpath = g_strdup (fullpath);
+ gchar *full_uri = g_strdup_printf ("file://%s", fullpath);
+ g_object_set (G_OBJECT (playbin), "uri", full_uri, NULL);
+ g_free (full_uri);
+ last_fullpath_changed = TRUE;
+ }
+
+ ret = gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ goto no_change_succes;
+
+ if (new_state != GST_STATE_PAUSED || new_state != GST_STATE_PLAYING ||
+ last_fullpath_changed) {
+ gst_element_set_state (pipeline , GST_STATE_READY);
+ ret = gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ goto no_change_succes;
+
+ gst_element_set_state (pipeline , GST_STATE_PAUSED);
+ ret = gst_element_get_state (pipeline, &new_state, NULL, GST_CLOCK_TIME_NONE);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ goto no_change_succes;
+
+ }
+ gst_element_seek_simple (pipeline, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, 0 * GST_SECOND);
+ gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
+
+no_change_succes:
+ g_mutex_unlock (audio_lock);
+ return TRUE;
+ }
+
+ return FALSE;
+}
+
+static void
+message_cb (GstBus *bus, GstMessage *message, gpointer user_data) {
+
+ g_mutex_lock (audio_lock);
+ switch (GST_MESSAGE_TYPE (message)) {
+ default:
+ break;
+ }
+ g_mutex_unlock (audio_lock);
+}
+
+static bool my_init(void)
+{
+ GstBus *bus;
+
+ pipeline = gst_pipeline_new ("freeciv_pipeline");
+ playbin = gst_element_factory_make ("playbin", NULL);
+ gst_object_ref (playbin);
+ gst_bin_add (GST_BIN (pipeline), playbin);
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+
+ gst_bus_add_signal_watch (bus);
+ g_signal_connect (bus, "message", (GCallback) message_cb,
+ NULL);
+
+ last_fullpath = NULL;
+ audio_lock = g_mutex_new ();
+
+ gst_object_unref (bus);
+
+ return TRUE;
+}
+
+static void my_set_volume (double volume) {
+ g_object_set (G_OBJECT (playbin), "volume", volume, NULL);
+}
+
+static double my_get_volume (void) {
+ double temp;
+ g_object_get (G_OBJECT (playbin), "volume", &temp, NULL);
+ return temp;
+}
+
+void audio_gst_init(void)
+{
+ gst_init (NULL, NULL);
+
+ struct audio_plugin self;
+
+ sz_strlcpy(self.name, "gst");
+ sz_strlcpy(self.descr, "GStreamer audio plugin");
+ self.init = my_init;
+ self.shutdown = my_shutdown;
+ self.stop = my_stop;
+ self.wait = my_wait;
+ self.play = my_play;
+ self.get_volume = my_get_volume;
+ self.set_volume = my_set_volume;
+ audio_add_plugin(&self);
+}
--- /dev/null
+++ freeciv/client/audio_gst.h
@@ -0,0 +1,18 @@
+/**********************************************************************
+ Freeciv - Copyright (C) 2008 - Thijs Vermeir <[EMAIL PROTECTED]>
+ 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, 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.
+***********************************************************************/
+#ifndef FC__AUDIO_GST_H
+#define FC__AUDIO_GST_H
+
+void audio_gst_init(void);
+
+#endif /* FC__AUDIO_GST_H */
--- freeciv.orig/configure.ac
+++ freeciv/configure.ac
@@ -566,6 +566,16 @@
AC_CHECK_FUNCS([chdir mkdtemp])
fi
+dnl Check for GStreamer
+PKG_CHECK_MODULES(GSTREAMER,
+ gstreamer-0.10,
+ HAVE_GST="yes",
+ HAVE_GST="no")
+
+if test "x$HAVE_GST" = "xyes" ; then
+ AC_DEFINE(AUDIO_GST, 1, [Enable GStreamer audio])
+fi
+
AC_SUBST(gui_sources)
AC_SUBST(CLIENT_CFLAGS)
AC_SUBST(CLIENT_CXXFLAGS)
@@ -577,7 +587,10 @@
AC_SUBST([VERSION_WITHOUT_LABEL])
AC_SUBST([VERSION_LABEL])
AC_SUBST([HOST_PATH_SEPARATOR])
+AC_SUBST(GSTREAMER_CFLAGS)
+AC_SUBST(GSTREAMER_LIBS)
AM_CONDITIONAL(AUDIO_SDL, test "x$SDL_mixer" = "xyes")
+AM_CONDITIONAL(AUDIO_GST, test "x$HAVE_GST" = "xyes")
AM_CONDITIONAL(CLIENT_GUI_SDL, test "$gui_sources" = "gui-sdl")
AM_CONDITIONAL(CLIENT_GUI_GTK_2_0, test "$gui_sources" = "gui-gtk-2.0")
AM_CONDITIONAL(CLIENT_GUI_XAW, test "$gui_sources" = "gui-xaw")
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