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[Gnash-commit] gnash gui/gtk.cpp libbase/gstgnashsrc.c libbase...
|
From: |
Bastiaan Jacques |
|
Subject: |
[Gnash-commit] gnash gui/gtk.cpp libbase/gstgnashsrc.c libbase... |
|
Date: |
Wed, 20 Feb 2008 21:39:45 +0000 |
CVSROOT: /sources/gnash
Module name: gnash
Changes by: Bastiaan Jacques <bjacques> 08/02/20 21:39:45
Modified files:
gui : gtk.cpp
Removed files:
libbase : gstgnashsrc.c gstgnashsrc.h
libmedia/gst : sound_handler_gst.cpp sound_handler_gst.h
Log message:
Remove unused files.
CVSWeb URLs:
http://cvs.savannah.gnu.org/viewcvs/gnash/gui/gtk.cpp?cvsroot=gnash&r1=1.148&r2=1.149
http://cvs.savannah.gnu.org/viewcvs/gnash/libbase/gstgnashsrc.c?cvsroot=gnash&r1=1.5&r2=0
http://cvs.savannah.gnu.org/viewcvs/gnash/libbase/gstgnashsrc.h?cvsroot=gnash&r1=1.10&r2=0
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/gst/sound_handler_gst.cpp?cvsroot=gnash&r1=1.11&r2=0
http://cvs.savannah.gnu.org/viewcvs/gnash/libmedia/gst/sound_handler_gst.h?cvsroot=gnash&r1=1.9&r2=0
Patches:
Index: gui/gtk.cpp
===================================================================
RCS file: /sources/gnash/gnash/gui/gtk.cpp,v
retrieving revision 1.148
retrieving revision 1.149
diff -u -b -r1.148 -r1.149
--- gui/gtk.cpp 20 Feb 2008 19:49:49 -0000 1.148
+++ gui/gtk.cpp 20 Feb 2008 21:39:44 -0000 1.149
@@ -17,7 +17,7 @@
// Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
//
-/* $Id: gtk.cpp,v 1.148 2008/02/20 19:49:49 bjacques Exp $ */
+/* $Id: gtk.cpp,v 1.149 2008/02/20 21:39:44 bjacques Exp $ */
#ifdef HAVE_CONFIG_H
#include "gnashconfig.h"
@@ -117,6 +117,7 @@
log_debug (_("Created top level window"));
}
+
// XXXbjacques: why do we need this?
//gtk_container_set_reallocate_redraws(GTK_CONTAINER (_window), TRUE);
@@ -161,6 +162,14 @@
setupEvents();
gtk_widget_realize(_window);
+ gdk_window_set_events(_window->window,
+ GdkEventMask(GDK_EXPOSURE_MASK | GDK_VISIBILITY_NOTIFY_MASK));
+
+ GdkWindow* parent = gdk_window_get_parent(_window->window);
+ gdk_window_set_events(parent,
+ GdkEventMask(GDK_EXPOSURE_MASK | GDK_VISIBILITY_NOTIFY_MASK));
+
+
gtk_widget_show(_drawingArea);
gtk_widget_show(_window);
@@ -392,7 +401,7 @@
G_CALLBACK (realize_event), NULL);
g_signal_connect(G_OBJECT (_drawingArea), "configure_event",
G_CALLBACK (configure_event), this);
- g_signal_connect(G_OBJECT (_drawingArea), "expose_event",
+ g_signal_connect(G_OBJECT (_window), "expose_event",
G_CALLBACK (expose_event), this);
return true;
@@ -1615,7 +1624,7 @@
GdkEventExpose *const event,
const gpointer data)
{
-// GNASH_REPORT_FUNCTION;
+ GNASH_REPORT_FUNCTION;
GtkGui* gui = static_cast<GtkGui*>(data);
Index: libbase/gstgnashsrc.c
===================================================================
RCS file: libbase/gstgnashsrc.c
diff -N libbase/gstgnashsrc.c
--- libbase/gstgnashsrc.c 1 Jul 2007 10:54:08 -0000 1.5
+++ /dev/null 1 Jan 1970 00:00:00 -0000
@@ -1,328 +0,0 @@
-//
-// Copyright (C) 2005, 2006 Free Software Foundation, Inc.
-//
-// 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 3 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 St, Fifth Floor, Boston, MA 02110-1301 USA
-//
-// Based on the filesrc and fdsrc element in Gstreamer-core.
-//
-
-/* $Id: gstgnashsrc.c,v 1.5 2007/07/01 10:54:08 bjacques Exp $ */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gstgnashsrc.h"
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS_ANY);
-
-
-GST_DEBUG_CATEGORY_STATIC (gst_gnash_src_debug);
-#define GST_CAT_DEFAULT gst_gnash_src_debug
-
-static const GstElementDetails gst_gnash_src_details =
-GST_ELEMENT_DETAILS ("Gnash source",
- "Gnash",
- "Use callback to read from Gnash",
- "Gnash team");
-
-/* GnashSrc signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
-
-#define DEFAULT_BLOCKSIZE 4*1024
-#define DEFAULT_DATA NULL
-#define DEFAULT_CALLBACKS NULL
-
-enum
-{
- ARG_0,
- ARG_DATA,
- ARG_CALLBACKS
-};
-
-static void gst_gnash_src_finalize (GObject * object);
-
-static void gst_gnash_src_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_gnash_src_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-
-static gboolean gst_gnash_src_start (GstBaseSrc * basesrc);
-static gboolean gst_gnash_src_stop (GstBaseSrc * basesrc);
-
-static gboolean gst_gnash_src_is_seekable (GstBaseSrc * src);
-static gboolean gst_gnash_src_get_size (GstBaseSrc * src, guint64 * size);
-static GstFlowReturn gst_gnash_src_create (GstPushSrc * src, GstBuffer **
buffer);
-static gboolean gst_gnash_src_do_seek (GstBaseSrc * src, GstSegment * s);
-
-static void
-_do_init (GType gnashsrc_type)
-{
- UNUSEDPAR(gnashsrc_type);
-/* static const GInterfaceInfo urihandler_info = {
- gst_gnash_src_uri_handler_init,
- NULL,
- NULL
- };*/
-
- GST_DEBUG_CATEGORY_INIT (gst_gnash_src_debug, "gnashsrc", 0, "gnashsrc
element");
-}
-
-GST_BOILERPLATE_FULL (GstGnashSrc, gst_gnash_src, GstElement,
GST_TYPE_PUSH_SRC, _do_init);
-
-static void
-gst_gnash_src_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&srctemplate));
-
- gst_element_class_set_details (gstelement_class, &gst_gnash_src_details);
-}
-
-static void
-gst_gnash_src_class_init (GstGnashSrcClass * klass)
-{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
- GstBaseSrcClass *gstbasesrc_class;
- GstPushSrcClass *gstpush_src_class;
-
- gobject_class = G_OBJECT_CLASS (klass);
- gstelement_class = GST_ELEMENT_CLASS (klass);
- gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
- gstpush_src_class = GST_PUSH_SRC_CLASS (klass);
-
- gobject_class->set_property = gst_gnash_src_set_property;
- gobject_class->get_property = gst_gnash_src_get_property;
-
- g_object_class_install_property (gobject_class, ARG_DATA,
- g_param_spec_pointer ("data", NULL, NULL,
(GParamFlags)G_PARAM_READWRITE));
-
- g_object_class_install_property (gobject_class, ARG_CALLBACKS,
- g_param_spec_pointer ("callbacks", NULL, NULL,
(GParamFlags)G_PARAM_READWRITE));
-
- gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_gnash_src_finalize);
-
- gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_gnash_src_start);
- gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_gnash_src_stop);
- gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR
(gst_gnash_src_is_seekable);
- gstbasesrc_class->get_size = GST_DEBUG_FUNCPTR (gst_gnash_src_get_size);
- gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_gnash_src_do_seek);
-
- gstpush_src_class->create = GST_DEBUG_FUNCPTR (gst_gnash_src_create);
-}
-
-static void
-gst_gnash_src_init (GstGnashSrc * src, GstGnashSrcClass * g_class)
-{
- UNUSEDPAR(g_class);
- src->data = NULL;
- src->callbacks = NULL;
- src->read_position = 0;
-}
-
-static void
-gst_gnash_src_finalize (GObject * object)
-{
- GstGnashSrc *src;
-
- src = GST_GNASH_SRC (object);
-
- free (src->callbacks);
- G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_gnash_src_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstGnashSrc *src;
-
- g_return_if_fail (GST_IS_GNASH_SRC (object));
-
- src = GST_GNASH_SRC (object);
-
- switch (prop_id) {
- case ARG_DATA:
- src->data = g_value_get_pointer (value);
- break;
- case ARG_CALLBACKS:
- src->callbacks = g_value_get_pointer (value);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_gnash_src_get_property (GObject * object, guint prop_id, GValue * value,
- GParamSpec * pspec)
-{
- GstGnashSrc *src;
- UNUSEDPAR(value);
- g_return_if_fail (GST_IS_GNASH_SRC (object));
-
- src = GST_GNASH_SRC (object);
-
- switch (prop_id) {
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-//
-// Below is the functions used get make this plugin work.
-//
-
-// Used for seeking
-static gboolean
-gst_gnash_src_do_seek (GstBaseSrc * basesrc, GstSegment * seg)
-{
-
- GstGnashSrc *src;
- off_t res;
- src = GST_GNASH_SRC (basesrc);
-
- if (seg->format != GST_FORMAT_BYTES) return FALSE;
-
- struct gnashsrc_callback *gc;
- gc = src->callbacks;
- res = gc->seek (src->data, seg->start, SEEK_CUR);
- return TRUE;
-
-}
-
-// Output the next buffer
-static GstFlowReturn
-gst_gnash_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
-{
- GstGnashSrc *src;
- src = GST_GNASH_SRC (psrc);
- int ret, blocksize;
- GstBuffer *buf;
-
- struct gnashsrc_callback *gc;
- gc = src->callbacks;
-
- blocksize = GST_BASE_SRC (src)->blocksize;
- buf = gst_buffer_new_and_alloc (blocksize);
-
- GST_LOG_OBJECT (src, "Reading %d bytes", blocksize);
-
- ret = gc->read(src->data, (void*)GST_BUFFER_DATA(buf), blocksize);
-
- if (G_UNLIKELY (ret < 0)) goto could_not_read;
-
- /* other files should eos if they read 0 and more was requested */
- if (G_UNLIKELY (ret == 0 && blocksize > 0))
- goto eos;
-
- blocksize = ret;
-
- GST_BUFFER_SIZE (buf) = blocksize;
- GST_BUFFER_OFFSET (buf) = src->read_position;
- GST_BUFFER_OFFSET_END (buf) = src->read_position + blocksize;
-
- *outbuf = buf;
-
- src->read_position += blocksize;
-
- return GST_FLOW_OK;
-
- /* ERROR */
-could_not_read:
- {
- GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
-eos:
- {
- GST_DEBUG ("non-regular file hits EOS");
- gst_buffer_unref (buf);
- return GST_FLOW_UNEXPECTED;
- }
-}
-
-static gboolean
-gst_gnash_src_is_seekable (GstBaseSrc * basesrc)
-{
- GstGnashSrc *src = GST_GNASH_SRC (basesrc);
-
- return src->seekable;
-}
-
-// Get size of the file. Not sure how we shall handles this...
-static gboolean
-gst_gnash_src_get_size (GstBaseSrc * basesrc, guint64 * size)
-{
- GstGnashSrc *src;
-
- src = GST_GNASH_SRC (basesrc);
-
- if (!src->seekable) {
- /* If it isn't seekable, we won't know the length (but fstat will still
- * succeed, and wrongly say our length is zero. */
- return FALSE;
- }
- return FALSE;
-
- // Since it's a streamed video file we probably don't know the length, so we
- // tell it's 50000. Maybe we should just return FALSE?
- *size = 500000;
- return TRUE;
-
-}
-
-/* open the file and mmap it, necessary to go to READY state */
-static gboolean
-gst_gnash_src_start (GstBaseSrc * basesrc)
-{
- GstGnashSrc *src = GST_GNASH_SRC (basesrc);
-
- if (src->data == NULL || src->callbacks == NULL) {
- GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,(("No data or callback struct
supplied.")), (NULL));
- return FALSE;
- }
-
- src->read_position = 0;
- GST_INFO_OBJECT (src, "Ready for reading using callbacks");
-
- // TODO: set seekable to false when real streaming
- src->seekable = TRUE;
-
- return TRUE;
-
-}
-
-/* stop and free */
-static gboolean
-gst_gnash_src_stop (GstBaseSrc * basesrc)
-{
- GstGnashSrc *src = GST_GNASH_SRC (basesrc);
- UNUSEDPAR(src);
- return TRUE;
-}
-
Index: libbase/gstgnashsrc.h
===================================================================
RCS file: libbase/gstgnashsrc.h
diff -N libbase/gstgnashsrc.h
--- libbase/gstgnashsrc.h 21 Jan 2008 20:55:44 -0000 1.10
+++ /dev/null 1 Jan 1970 00:00:00 -0000
@@ -1,81 +0,0 @@
-//
-// Copyright (C) 2005, 2006, 2007, 2008 Free Software Foundation, Inc.
-//
-// 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 3 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 St, Fifth Floor, Boston, MA 02110-1301 USA
-//
-// Based on the filesrc and fdsrc element in Gstreamer-core
-//
-
-/* $Id: gstgnashsrc.h,v 1.10 2008/01/21 20:55:44 rsavoye Exp $ */
-
-#ifndef __GST_GNASH_SRC_H__
-#define __GST_GNASH_SRC_H__
-
-#define UNUSEDPAR(x) { x = x; }
-
-
-#include <gst/gst.h>
-#include <gst/base/gstpushsrc.h>
-
-// Struct to contain the callback functions
-struct gnashsrc_callback {
- int (*read)(void* data, char* buf, int buf_size);
- int (*seek)(void* data, int offset, int whence);
-};
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_GNASH_SRC \
- (gst_gnash_src_get_type())
-#define GST_GNASH_SRC(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_GNASH_SRC,GstGnashSrc))
-#define GST_GNASH_SRC_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_GNASH_SRC,GstGnashSrcClass))
-#define GST_IS_GNASH_SRC(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_GNASH_SRC))
-#define GST_IS_GNASH_SRC_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_GNASH_SRC))
-
-typedef struct _GstGnashSrc GstGnashSrc;
-typedef struct _GstGnashSrcClass GstGnashSrcClass;
-
-/**
- * GstGnashSrc:
- *
- * Opaque #GstGnashSrc structure.
- */
-struct _GstGnashSrc {
- GstPushSrc element;
-
- /*< private >*/
-
- guint64 read_position; // position in the stream
-
- gpointer data; // data passes with the callbacks
- gpointer callbacks; // struct with the callbacks
-
- gboolean seekable; // seekable or not
-
-};
-
-struct _GstGnashSrcClass {
- GstPushSrcClass parent_class;
-};
-
-GType gst_gnash_src_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_GNASH_SRC_H__ */
Index: libmedia/gst/sound_handler_gst.cpp
===================================================================
RCS file: libmedia/gst/sound_handler_gst.cpp
diff -N libmedia/gst/sound_handler_gst.cpp
--- libmedia/gst/sound_handler_gst.cpp 21 Jan 2008 23:10:15 -0000 1.11
+++ /dev/null 1 Jan 1970 00:00:00 -0000
@@ -1,906 +0,0 @@
-// sound_handler_gst.cpp: Audio output via GStreamer, for Gnash.
-//
-// Copyright (C) 2005, 2006, 2007, 2008 Free Software Foundation, Inc.
-//
-// 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 3 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 St, Fifth Floor, Boston, MA 02110-1301 USA
-//
-
-// Based on sound_handler_sdl.cpp by Thatcher Ulrich http://tulrich.com 2003
-// which has been donated to the Public Domain.
-
-/* $Id: sound_handler_gst.cpp,v 1.11 2008/01/21 23:10:15 rsavoye Exp $ */
-
-#ifdef HAVE_CONFIG_H
-#include "gnashconfig.h"
-#endif
-
-#include "utility.h" // for convert_raw_data
-
-#include <utility> // for std::make_pair
-
-// Assume people running --enable-media=gst know what they are doing
-// (HAVE_GST_GST_H seems broken atm, specifically when an older glib
-// install is around)
-//
-
-#ifdef SOUND_GST
-
-#include "sound_handler_gst.h"
-#include "gnash.h"
-#include "log.h"
-#include "types.h" // for IF_VERBOSE_* macros
-#include <cmath>
-#include <vector>
-
-#include <gst/gst.h>
-
-#define BUFFER_SIZE 5000
-
-using namespace boost;
-
-namespace gnash {
-namespace media {
-
-GST_sound_handler::GST_sound_handler()
- : looping(false),
- muted(false)
-{
- // init gstreamer
- gst_init(NULL, NULL);
-}
-
-void
-GST_sound_handler::delete_all_sounds()
-{
- for (size_t i=0, e=m_sound_data.size(); i < e; ++i) {
- stop_sound(i);
- delete_sound(i);
- }
- m_sound_data.clear();
-}
-
-GST_sound_handler::~GST_sound_handler()
-{
- delete_all_sounds();
-}
-
-
-int GST_sound_handler::create_sound(
- void* data_,
- unsigned int data_bytes,
- std::auto_ptr<SoundInfo> sinfo)
-// Called to create a sample. We'll return a sample ID that
-// can be use for playing it.
-{
-
- try_mutex::scoped_lock lock(_mutex);
-
- unsigned char* data = static_cast<unsigned char*>(data_);
-
- assert(sinfo.get());
- sound_data *sounddata = new sound_data;
- if (!sounddata) {
- log_error(_("could not allocate memory for sound data"));
- return -1;
- }
-
- sounddata->volume = 100;
- sounddata->soundinfo = sinfo;
-
- switch (sounddata->soundinfo->getFormat())
- {
- case AUDIO_CODEC_MP3:
- case AUDIO_CODEC_RAW:
- case AUDIO_CODEC_ADPCM:
- case AUDIO_CODEC_UNCOMPRESSED:
- case AUDIO_CODEC_NELLYMOSER:
- case AUDIO_CODEC_NELLYMOSER_8HZ_MONO:
- if ( data ) sounddata->append(data, data_bytes);
- break;
-
-
- default:
- // Unhandled format.
- log_error(_("Unknown sound format %d requested; gnash does not
handle it"), (int)sounddata->soundinfo->getFormat());
- return -1; // Unhandled format, set to NULL.
- }
-
- m_sound_data.push_back(sounddata);
-
- return m_sound_data.size()-1;
-}
-
-
-// this gets called when a stream gets more data
-long GST_sound_handler::fill_stream_data(unsigned char* data, unsigned int
data_bytes, unsigned int /*sample_count*/, int handle_id)
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // @@ does a negative handle_id have any meaning ?
- // should we change it to unsigned instead ?
- if (handle_id >= 0 && (unsigned int) handle_id < m_sound_data.size())
- {
- sound_data* sounddata = m_sound_data[handle_id];
-
- long startSize = sounddata->dataSize();
-
- sounddata->append(data, data_bytes);
-
- // If playback has already started, we also update the active
sounds
- for (size_t i=0, e=sounddata->m_gst_elements.size(); i < e;
++i) {
- gst_elements* sound = sounddata->m_gst_elements[i];
- sound->data_size = sounddata->dataSize();
- sound->set_data(sounddata->data());
- }
-
- return startSize;
- }
- else
- {
- delete [] data;
- return 0;
- }
-}
-
-// This stops sounds when they are done playing
-static gboolean sound_killer (gpointer user_data)
-{
- gst_elements *gstelements = static_cast<gst_elements*>(user_data);
- gst_element_set_state (GST_ELEMENT (gstelements->pipeline),
GST_STATE_NULL);
- return false;
-}
-
-// The callback function which refills the buffer with data
-void GST_sound_handler::callback_handoff (GstElement * /*c*/, GstBuffer
*buffer, GstPad* /*pad*/, gpointer user_data)
-{
- gst_elements *gstelements = static_cast<gst_elements*>(user_data);
-
- try_mutex::scoped_try_lock lock(gstelements->handler->_mutex);
-
- // If we couldn't obtain a lock return to avoid a deadlock
- if (!lock.locked()) {
-
- // We return nothing in this case to avoid noise being decoded
and played
- if (GST_BUFFER_SIZE(buffer) != 0 && GST_BUFFER_DATA(buffer)) {
- GST_BUFFER_DATA(buffer) = 0;
- GST_BUFFER_SIZE(buffer) = 0;
- }
- return;
- }
-
- // First callback or after a couldn't-get-lock-return
- if (GST_BUFFER_SIZE(buffer) == 0) {
- if (gstelements->data_size > BUFFER_SIZE) {
- GST_BUFFER_SIZE(buffer) = BUFFER_SIZE;
- } else {
- GST_BUFFER_SIZE(buffer) = gstelements->data_size;
- }
-
- // Reallocate the required memory.
- guint8* tmp_buf = new guint8[GST_BUFFER_SIZE(buffer)];
- memcpy(tmp_buf, GST_BUFFER_DATA(buffer), sizeof(buffer));
-
- delete [] GST_BUFFER_DATA(buffer);
- GST_BUFFER_DATA(buffer) = tmp_buf;
- }
-
- // All the data has been given to the pipeline, so now we need to stop
- // the pipeline. g_idle_add() makes sure sound_killer is called soon.
- if (gstelements->position > gstelements->data_size) {
- g_idle_add(sound_killer, user_data);
- GST_BUFFER_SIZE(buffer) = 0;
- GST_BUFFER_DATA(buffer) = 0;
- return;
- }
-
- const guint8* data_pos =
gstelements->get_data_ptr(gstelements->position);
-
- // Last callback - the last re-fill
- if (gstelements->position+BUFFER_SIZE > gstelements->data_size) {
-
- unsigned int chunk_size =
gstelements->data_size-gstelements->position;
- // Check if we should loop. If loop_count is 0 we have we just
- // played the sound for the last (and perhaps first) time.
- // If loop_count is anything else we continue to loop.
- if (gstelements->loop_count == 0) {
- GST_BUFFER_SIZE(buffer) = chunk_size;
- memcpy(GST_BUFFER_DATA(buffer), data_pos, chunk_size);
- gstelements->position += BUFFER_SIZE;
-
- gst_element_set_state (GST_ELEMENT
(gstelements->input), GST_STATE_PAUSED);
-
- } else {
- // Copy what's left of the data, and then fill the rest
with "new" data.
- memcpy(GST_BUFFER_DATA(buffer), data_pos, chunk_size);
- memcpy(GST_BUFFER_DATA(buffer) + chunk_size,
gstelements->get_data_ptr(0), GST_BUFFER_SIZE(buffer)- chunk_size);
- gstelements->position = GST_BUFFER_SIZE(buffer) -
chunk_size;
- gstelements->loop_count--;
-
- }
-
- return;
-
- }
-
- // Standard re-fill
- memcpy(GST_BUFFER_DATA(buffer), data_pos, BUFFER_SIZE);
- gstelements->position += BUFFER_SIZE;
-
-}
-
-
-void GST_sound_handler::play_sound(int sound_handle, int loop_count, int
/*offset*/, long start_position, const std::vector<sound_envelope>*
/*envelopes*/)
-// Play the index'd sample.
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists, or if audio is muted
- if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size() || muted)
- {
- // Invalid handle, or audio is muted.
- return;
- }
-
- sound_data* sounddata = m_sound_data[sound_handle];
-
- // If this is called from a streamsoundblocktag, we only start if this
- // sound isn't already playing.
- if (start_position > 0 && sounddata->m_gst_elements.size() > 0) {
- return;
- }
- // Make sure sound actually got some data
- if (sounddata->dataSize() < 1) {
- IF_VERBOSE_MALFORMED_SWF(
- log_swferror(_("Trying to play sound with size 0"));
- );
- return;
- }
-
- // Make a "gst_elements" for this sound which is latter placed on the
vector of instances of this sound being played
- gst_elements* gst_element = new gst_elements;
- if (gst_element == NULL) {
- log_error (_("Could not allocate memory for gst_element"));
- return;
- }
-
- // Set the handler
- gst_element->handler = this;
-
- // Copy data-info to the "gst_elements"
- gst_element->data_size = sounddata->dataSize();
- gst_element->set_data(sounddata->data());
- gst_element->position = start_position;
-
- // Set number of loop we should do. -1 is infinte loop, 0 plays it
once, 1 twice etc.
- gst_element->loop_count = loop_count;
-
- // create main pipeline
- gst_element->pipeline = gst_pipeline_new (NULL);
-
- // create an audio sink - use oss, alsa or...? make a commandline
option?
- // we first try atudetect, then alsa, then oss, then esd, then...?
-#if !defined(__NetBSD__)
- gst_element->audiosink = gst_element_factory_make ("autoaudiosink",
NULL);
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("alsasink", NULL);
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("osssink", NULL);
-#endif
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("esdsink", NULL);
-
- // Check if the creation of the gstreamer pipeline, adder and audiosink
was a succes
- if (!gst_element->pipeline) {
- log_error(_("The gstreamer pipeline element could not be
created"));
- }
- if (!gst_element->audiosink) {
- log_error(_("The gstreamer audiosink element could not be
created"));
- }
-
- // link adder and output to bin
- gst_bin_add (GST_BIN (gst_element->pipeline), gst_element->audiosink);
-
- gst_element->bin = gst_bin_new(NULL);
- gst_element->input = gst_element_factory_make ("fakesrc", NULL);
- gst_element->capsfilter = gst_element_factory_make ("capsfilter", NULL);
- gst_element->audioconvert = gst_element_factory_make ("audioconvert",
NULL);
- gst_element->audioresample = gst_element_factory_make ("audioresample",
NULL);
- gst_element->volume = gst_element_factory_make ("volume", NULL);
-
- // Put the gstreamer elements in the pipeline
- gst_bin_add_many (GST_BIN (gst_element->bin), gst_element->input,
- gst_element->capsfilter,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
-
- // Test if the fakesrc, typefind and audio* elements was correctly
created
- if (!gst_element->input
- || !gst_element->capsfilter
- || !gst_element->audioconvert
- || !gst_element->audioresample) {
-
- log_error(_("Gstreamer element for audio handling could not be
created"));
- return;
- }
-
- // Create a gstreamer decoder for the chosen sound.
-
- // Temp variables to make the code simpler and easier to read
- audioCodecType soundFormat = sounddata->soundinfo->getFormat();
- bool soundStereo = sounddata->soundinfo->isStereo();
- boost::uint32_t soundSampleRate = sounddata->soundinfo->getSampleRate();
-
- if (soundFormat == AUDIO_CODEC_MP3) {
-
- gst_element->decoder = gst_element_factory_make ("mad", NULL);
- if (gst_element->decoder == NULL) {
- gst_element->decoder = gst_element_factory_make
("flump3dec", NULL);
- if (gst_element->decoder != NULL &&
!gst_default_registry_check_feature_version("flump3dec", 0, 10, 4))
- {
- static bool warned=false;
- if ( ! warned )
- {
- // I keep getting these messages even
if I hear sound... too much paranoia ?
- log_debug(_("This version of fluendos
mp3 plugin does not support flash streaming sounds, please upgrade to version
0.10.4 or higher"));
- warned=true;
- }
- }
- }
- // Check if the element was correctly created
- if (!gst_element->decoder) {
- log_error(_("A gstreamer mp3-decoder element could not
be created. You probably need to install a mp3-decoder plugin like
gstreamer0.10-mad or gstreamer0.10-fluendo-mp3."));
- return;
- }
- gst_bin_add (GST_BIN (gst_element->bin), gst_element->decoder);
-
- // Set the info about the stream so that gstreamer knows what
it is.
- GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
- "mpegversion", G_TYPE_INT, 1,
- "layer", G_TYPE_INT, 3,
- "rate", G_TYPE_INT, soundSampleRate,
- "channels", G_TYPE_INT, soundStereo ? 2 : 1, NULL);
- g_object_set (G_OBJECT (gst_element->capsfilter), "caps", caps,
NULL);
- gst_caps_unref (caps);
-
- // setup fake source
- g_object_set (G_OBJECT (gst_element->input),
- "sizetype", 2, "can-activate-pull",
FALSE, "signal-handoffs", TRUE,
- "sizemax", BUFFER_SIZE, NULL);
- // Setup the callback
- gst_element->handoff_signal_id = g_signal_connect
(gst_element->input, "handoff", G_CALLBACK (callback_handoff), gst_element);
-
- // link data, decoder, audio* and adder
- gst_element_link_many (gst_element->input,
- gst_element->capsfilter,
- gst_element->decoder,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
-
- } else if (soundFormat == AUDIO_CODEC_ADPCM) {
- gst_element->decoder = gst_element_factory_make
("ffdec_adpcm_swf", NULL);
-
- // Check if the element was correctly created
- if (!gst_element->decoder) {
- log_error(_("A gstreamer adpcm-decoder element could
not be created. You probably need to install gst-ffmpeg."));
- return;
- }
- gst_bin_add (GST_BIN (gst_element->bin), gst_element->decoder);
-
- // Set the info about the stream so that gstreamer knows what
it is.
- GstCaps *caps = gst_caps_new_simple ("audio/x-adpcm",
- "rate", G_TYPE_INT, soundSampleRate,
- "channels", G_TYPE_INT, soundStereo ? 2 : 1, NULL);
- g_object_set (G_OBJECT (gst_element->capsfilter), "caps", caps,
NULL);
- gst_caps_unref (caps);
-
- // setup fake source
- g_object_set (G_OBJECT (gst_element->input),
- "sizetype", 2, "can-activate-pull",
FALSE, "signal-handoffs", TRUE,
- "sizemax", BUFFER_SIZE, NULL);
- // Setup the callback
- gst_element->handoff_signal_id = g_signal_connect
(gst_element->input, "handoff", G_CALLBACK (callback_handoff), gst_element);
-
- // link data, decoder, audio* and adder
- gst_element_link_many (gst_element->input,
- gst_element->capsfilter,
- gst_element->decoder,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
-
- } else if (soundFormat == AUDIO_CODEC_NELLYMOSER_8HZ_MONO ||
soundFormat == AUDIO_CODEC_NELLYMOSER) {
- return;
- } else {
-
- // Set the info about the stream so that gstreamer knows what
it is.
- GstCaps *caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, soundSampleRate,
- "channels", G_TYPE_INT, soundStereo ? 2 : 1,
- "endianness", G_TYPE_INT, G_BIG_ENDIAN,
- "width", G_TYPE_INT, (sounddata->soundinfo->is16bit() ?
16 : 8),
- "depth", G_TYPE_INT, 16,
- //"signed", G_TYPE_INT, 1,
- NULL);
- g_object_set (G_OBJECT (gst_element->capsfilter), "caps", caps,
NULL);
- gst_caps_unref (caps);
-
- // setup fake source
- g_object_set (G_OBJECT (gst_element->input),
- "sizetype", 2, "can-activate-pull",
FALSE, "signal-handoffs", TRUE,
- "sizemax", BUFFER_SIZE, NULL);
- // Setup the callback
- gst_element->handoff_signal_id = g_signal_connect
(gst_element->input, "handoff", G_CALLBACK (callback_handoff), gst_element);
-
- // Raw native sound-data, output directly
- gst_element_link_many (gst_element->input,
- gst_element->capsfilter,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
- }
- // Add ghostpad
- GstPad *pad = gst_element_get_pad (gst_element->volume, "src");
- gst_element_add_pad (gst_element->bin, gst_ghost_pad_new ("src", pad));
- gst_object_unref (GST_OBJECT (pad));
-
- // Add the bin to the main pipeline
- gst_bin_add(GST_BIN (gst_element->pipeline), gst_element->bin);
- // Link to the adder sink pad
- GstPad *sinkpad = gst_element_get_pad (gst_element->audiosink, "sink");
- GstPad *srcpad = gst_element_get_pad (gst_element->bin, "src");
- gst_pad_link (srcpad, sinkpad);
- gst_object_unref (GST_OBJECT (srcpad));
- gst_object_unref (GST_OBJECT (sinkpad));
-
- // Set the volume
- g_object_set (G_OBJECT (gst_element->volume), "volume",
static_cast<double>(sounddata->volume) / 100.0, NULL);
-
- //gst_pad_add_event_probe(pad, G_CALLBACK(event_callback), sounddata);
-
- // Put the gst_element on the vector
- sounddata->m_gst_elements.push_back(gst_element);
-
- // If not already playing, start doing it
- gst_element_set_state (GST_ELEMENT (gst_element->pipeline),
GST_STATE_PLAYING);
-
- ++_soundsStarted;
-
-}
-
-
-void GST_sound_handler::stop_sound(int sound_handle)
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
- {
- // Invalid handle.
- return;
- }
-
- sound_data* sounddata = m_sound_data[sound_handle];
-
- // Stop all the instances of this sound.
- // TODO: fix the loop to use size_t instead of i
- for (int i = sounddata->m_gst_elements.size()-1; i >= 0 ; i--)
- {
- gst_elements* elements = sounddata->m_gst_elements[i];
-
- // Check if we can succesfully stop the elements
- // playback - if not we skip cleaning this for now
- // FIXME: what if it ain't possible to stop an element when
this is called from ~GST_sound_handler
-
- // Disconnect signals
- g_signal_handler_disconnect (elements->input,
elements->handoff_signal_id);
-
- gst_element_set_state (GST_ELEMENT (elements->pipeline),
GST_STATE_NULL);
- gst_object_unref (GST_OBJECT (elements->pipeline));
-
- // Delete the gst_element struct
- // @@ we're deleting the elements from the start, so half-way
of the loop we will be referring to undefined elements. Is this intended ?
--strk;
- delete elements;
-
sounddata->m_gst_elements.erase(sounddata->m_gst_elements.begin() + i);
- }
-
- ++_soundsStopped;
-}
-
-
-void GST_sound_handler::delete_sound(int sound_handle)
-// this gets called when it's done with a sample.
-{
- try_mutex::scoped_lock lock(_mutex);
-
- if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
- {
- delete m_sound_data[sound_handle];
- m_sound_data.erase (m_sound_data.begin() + sound_handle);
- }
-
-}
-
-// This will stop all sounds playing. Will cause problems if the soundhandler
is made static
-// and supplys sound_handling for many SWF's, since it will stop all sounds
with no regard
-// for what sounds is associated with what SWF.
-void GST_sound_handler::stop_all_sounds()
-{
- for (size_t i=0, e=m_sound_data.size(); i < e; ++i)
- stop_sound(i);
-}
-
-void
-GST_sound_handler::reset()
-{
- stop_all_sounds();
-}
-
-
-// returns the sound volume level as an integer from 0 to 100,
-// where 0 is off and 100 is full volume. The default setting is 100.
-int GST_sound_handler::get_volume(int sound_handle) {
-
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
- {
- return m_sound_data[sound_handle]->volume;
- } else {
- return 0; // Invalid handle
- }
-}
-
-
-// A number from 0 to 100 representing a volume level.
-// 100 is full volume and 0 is no volume. The default setting is 100.
-void GST_sound_handler::set_volume(int sound_handle, int volume) {
-
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
- {
- // Invalid handle.
- return;
- }
-
- sound_data* sd = m_sound_data[sound_handle];
-
- // Set volume for this sound. Should this only apply to the active
sounds?
-
- sd->volume = volume;
-
- for (size_t i=0, n=sd->m_gst_elements.size(); i<n; ++i)
- {
- g_object_set (
- G_OBJECT (sd->m_gst_elements[i]->volume),
- "volume",
- static_cast<double>(volume/100.0),
- NULL);
- }
-
-}
-
-SoundInfo* GST_sound_handler::get_sound_info(int sound_handle) {
-
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle >= 0 && static_cast<unsigned int>(sound_handle) <
m_sound_data.size())
- {
- return m_sound_data[sound_handle]->soundinfo.get();
- } else {
- return NULL;
- }
-
-}
-
-// gnash calls this to mute audio
-void GST_sound_handler::mute() {
- stop_all_sounds();
- muted = true;
-}
-
-// gnash calls this to unmute audio
-void GST_sound_handler::unmute() {
- muted = false;
-}
-
-bool GST_sound_handler::is_muted() {
- return muted;
-}
-
-
-
-// The callback function which refills the buffer with data
-void GST_sound_handler::callback_as_handoff (GstElement * /*c*/, GstBuffer
*buffer, GstPad* /*pad*/, gpointer user_data)
-{
- gst_elements *gstelements = static_cast<gst_elements*>(user_data);
-
- try_mutex::scoped_try_lock lock(gstelements->handler->_mutex);
-
- // If we couldn't obtain a lock return to avoid a deadlock
- if (!lock.locked()) {
-
- // We return nothing in this case to avoid noise being decoded
and played
- if (GST_BUFFER_SIZE(buffer) != 0 && GST_BUFFER_DATA(buffer)) {
- GST_BUFFER_DATA(buffer) = 0;
- GST_BUFFER_SIZE(buffer) = 0;
- }
- return;
- }
-
- // First callback or after a couldn't-get-lock-return
- if (GST_BUFFER_SIZE(buffer) == 0) {
- if (gstelements->data_size > BUFFER_SIZE) {
- GST_BUFFER_SIZE(buffer) = BUFFER_SIZE;
- } else {
- GST_BUFFER_SIZE(buffer) = gstelements->data_size;
- }
-
- // Reallocate the required memory.
- guint8* tmp_buf = new guint8[GST_BUFFER_SIZE(buffer)];
- memcpy(tmp_buf, GST_BUFFER_DATA(buffer), sizeof(buffer));
-
- delete [] GST_BUFFER_DATA(buffer);
- GST_BUFFER_DATA(buffer) = tmp_buf;
- }
-
- aux_streamer_ptr aux_streamer =
gstelements->handler->m_aux_streamer[gstelements->owner];
-
- // If false is returned the sound doesn't want to be attached anymore
- bool ret = (aux_streamer)(gstelements->owner, GST_BUFFER_DATA(buffer),
GST_BUFFER_SIZE(buffer));
-
-}
-
-void GST_sound_handler::attach_aux_streamer(aux_streamer_ptr ptr, void* owner)
-{
- try_mutex::scoped_lock lock(_mutex);
- assert(owner);
- assert(ptr);
-
- if ( ! m_aux_streamer.insert(std::make_pair(owner, ptr)).second )
- {
- // Already in the hash.
- return;
- }
-
- // Make a pipeline that can play the raw data
-
- // Make a "gst_elements" for this sound which is latter placed on the
vector of instances of this sound being played
- gst_elements* gst_element = new gst_elements;
- if (gst_element == NULL) {
- log_error (_("Could not allocate memory for gst_element"));
- return;
- }
-
- // Set the handler
- gst_element->handler = this;
-
- // create main pipeline
- gst_element->pipeline = gst_pipeline_new (NULL);
-
- // create an audio sink - use oss, alsa or...? make a commandline
option?
- // we first try atudetect, then alsa, then oss, then esd, then...?
-#if !defined(__NetBSD__)
- gst_element->audiosink = gst_element_factory_make ("autoaudiosink",
NULL);
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("alsasink", NULL);
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("osssink", NULL);
-#endif
- if (!gst_element->audiosink) gst_element->audiosink =
gst_element_factory_make ("esdsink", NULL);
-
- // Check if the creation of the gstreamer pipeline, adder and audiosink
was a succes
- if (!gst_element->pipeline) {
- log_error(_("The gstreamer pipeline element could not be
created"));
- }
- if (!gst_element->audiosink) {
- log_error(_("The gstreamer audiosink element could not be
created"));
- }
-
- // link adder and output to bin
- gst_bin_add (GST_BIN (gst_element->pipeline), gst_element->audiosink);
-
- gst_element->bin = gst_bin_new(NULL);
- gst_element->input = gst_element_factory_make ("fakesrc", NULL);
- gst_element->capsfilter = gst_element_factory_make ("capsfilter", NULL);
- gst_element->audioconvert = gst_element_factory_make ("audioconvert",
NULL);
- gst_element->audioresample = gst_element_factory_make ("audioresample",
NULL);
- gst_element->volume = gst_element_factory_make ("volume", NULL);
-
- // Put the gstreamer elements in the pipeline
- gst_bin_add_many (GST_BIN (gst_element->bin), gst_element->input,
- gst_element->capsfilter,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
-
- // Test if the fakesrc, typefind and audio* elements was correctly
created
- if (!gst_element->input
- || !gst_element->capsfilter
- || !gst_element->audioconvert
- || !gst_element->audioresample) {
-
- log_error(_("Gstreamer element for audio handling could not be
created"));
- return;
- }
-
- // Create a gstreamer decoder for the chosen sound.
-
- // Set the info about the stream so that gstreamer knows what it is.
- // We know what the pre-decoded data format is.
- GstCaps *caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, 44100,
- "channels", G_TYPE_INT, 2,
- "endianness", G_TYPE_INT, G_BIG_ENDIAN,
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16,
- //"signed", G_TYPE_INT, 1,
- NULL);
- g_object_set (G_OBJECT (gst_element->capsfilter), "caps", caps, NULL);
- gst_caps_unref (caps);
-
- // setup fake source
- g_object_set (G_OBJECT (gst_element->input),
- "sizetype", 2, "can-activate-pull", FALSE,
"signal-handoffs", TRUE,
- "sizemax", BUFFER_SIZE, NULL);
- // Setup the callback
- gst_element->handoff_signal_id = g_signal_connect (gst_element->input,
"handoff", G_CALLBACK (callback_as_handoff), gst_element);
-
- // we receive raw native sound-data, output directly
- gst_element_link_many (gst_element->input,
- gst_element->capsfilter,
- gst_element->audioconvert,
- gst_element->audioresample,
- gst_element->volume, NULL);
-
- // Add ghostpad
- GstPad *pad = gst_element_get_pad (gst_element->volume, "src");
- gst_element_add_pad (gst_element->bin, gst_ghost_pad_new ("src", pad));
- gst_object_unref (GST_OBJECT (pad));
-
- // Add the bin to the main pipeline
- gst_bin_add(GST_BIN (gst_element->pipeline), gst_element->bin);
- // Link to the adder sink pad
- GstPad *sinkpad = gst_element_get_pad (gst_element->audiosink, "sink");
- GstPad *srcpad = gst_element_get_pad (gst_element->bin, "src");
- gst_pad_link (srcpad, sinkpad);
- gst_object_unref (GST_OBJECT (srcpad));
- gst_object_unref (GST_OBJECT (sinkpad));
-
- gst_element->owner = owner;
-
- // Put the gst_element in the map
- m_aux_streamer_gstelements[owner] = gst_element;
-
- // If not already playing, start doing it
- gst_element_set_state (GST_ELEMENT (gst_element->pipeline),
GST_STATE_PLAYING);
-printf("pipeline stated playing\n");
-}
-
-void GST_sound_handler::detach_aux_streamer(void* owner)
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // TODO: stuff both gstelements and callbacks in the same container !
-
- GstElementsMap::iterator it=m_aux_streamer_gstelements.find(owner);
- if ( it != m_aux_streamer_gstelements.end() )
- {
- delete it->second;
- // WARNING: erasing would break any iteration in the map
- m_aux_streamer_gstelements.erase(it);
- }
-
- CallbacksMap::iterator it2=m_aux_streamer.find(owner);
- if ( it2 != m_aux_streamer.end() )
- {
- // WARNING: erasing would break any iteration in the map
- m_aux_streamer.erase(it2);
- }
-}
-
-unsigned int GST_sound_handler::get_duration(int sound_handle)
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
- {
- // Invalid handle.
- return 0;
- }
-
- sound_data* sounddata = m_sound_data[sound_handle];
-
- boost::uint32_t sampleCount = sounddata->soundinfo->getSampleCount();
- boost::uint32_t sampleRate = sounddata->soundinfo->getSampleRate();
-
- // Return the sound duration in milliseconds
- if (sampleCount > 0 && sampleRate > 0) {
- unsigned int ret = sampleCount / sampleRate * 1000;
- ret += ((sampleCount % sampleRate) * 1000) / sampleRate;
- if (sounddata->soundinfo->isStereo()) ret = ret / 2;
- return ret;
- } else {
- return 0;
- }
-}
-
-unsigned int GST_sound_handler::get_position(int sound_handle)
-{
- try_mutex::scoped_lock lock(_mutex);
-
- // Check if the sound exists.
- if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
- {
- // Invalid handle.
- return 0;
- }
-
- sound_data* sounddata = m_sound_data[sound_handle];
-
- // If there is no active sounds, return 0
- if (sounddata->m_gst_elements.size() == 0) {
- return 0;
- }
-
- // return the position of the last element added
- GstElement *pipeline,*audioconvert;
- GstStateChangeReturn ret;
- GstState current, pending;
- boost::int64_t pos;
- GstFormat fmt = GST_FORMAT_TIME;
-
- pipeline =
sounddata->m_gst_elements[sounddata->m_gst_elements.size()-1]->pipeline;
-
- ret = gst_element_get_state (GST_ELEMENT (pipeline), ¤t,
&pending, 0);
-
- if (current != GST_STATE_NULL) {
- audioconvert =
sounddata->m_gst_elements[sounddata->m_gst_elements.size()-1]->audioconvert;
- if (gst_element_query_position (audioconvert, &fmt, &pos)) {
- return static_cast<unsigned int>(pos / GST_MSECOND);
- } else {
- return 0;
- }
- }
- return 0;
-}
-
-// Pointer handling and checking functions
-const boost::uint8_t* gst_elements::get_data_ptr(unsigned long int pos)
-{
- assert(data_size > pos);
- return data + pos;
-}
-
-void gst_elements::set_data(const boost::uint8_t* idata) {
- data = idata;
-}
-
-sound_handler* create_sound_handler_gst()
-// Factory.
-{
- return new GST_sound_handler;
-}
-
-} // gnash.media namespace
-} // namespace gnash
-
-#endif // SOUND_GST
-
-// Local Variables:
-// mode: C++
-// End:
-
Index: libmedia/gst/sound_handler_gst.h
===================================================================
RCS file: libmedia/gst/sound_handler_gst.h
diff -N libmedia/gst/sound_handler_gst.h
--- libmedia/gst/sound_handler_gst.h 21 Jan 2008 23:10:15 -0000 1.9
+++ /dev/null 1 Jan 1970 00:00:00 -0000
@@ -1,229 +0,0 @@
-// Copyright (C) 2005, 2006, 2007, 2008 Free Software Foundation, Inc.
-//
-// 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 3 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 St, Fifth Floor, Boston, MA 02110-1301 USA
-
-// $Id: sound_handler_gst.h,v 1.9 2008/01/21 23:10:15 rsavoye Exp $
-
-#ifndef SOUND_HANDLER_GST_H
-#define SOUND_HANDLER_GST_H
-
-//#include "gnash.h"
-#include "sound_handler.h" // for inheritance
-
-#include <vector>
-#include <map>
-
-#include <gst/gst.h>
-#include <boost/thread/thread.hpp>
-#include <boost/bind.hpp>
-#include <boost/thread/mutex.hpp>
-
-#define BUFFER_SIZE 5000
-
-namespace gnash {
-namespace media {
-
-// forward declaration
-class GST_sound_handler;
-
-// Used to hold the gstreamer when doing on-demand-decoding
-class gst_elements
-{
-public:
- // gstreamer pipeline objects
-
- // the main bin containing the adder and output (sink)
- GstElement *pipeline;
- GstElement *audiosink;
-
- // gstreamer objects
- GstElement *input;
- GstElement *decoder;
- GstElement *capsfilter;
- GstElement *audioconvert;
- GstElement *audioresample;
- GstElement *volume;
- GstElement *bin;
- GstPad *addersinkpad;
-
- // position in the stream
- unsigned long position;
-
- // data size
- unsigned long data_size;
-
- long loop_count;
-
- // signal id
- guint handoff_signal_id;
-
- // The sound handler. Used to get access to the
GST_sound_handler->_mutex
- GST_sound_handler* handler;
-
- /// Returns the data pointer in the undecoded datastream
- /// for the given position. Boundaries are checked.
- const boost::uint8_t* get_data_ptr(unsigned long int pos);
-
- /// Set the undecoded data pointer
- void set_data(const boost::uint8_t*);
-
- /// The owner of the audio data this element will get data from.
- /// Only used when getting data from NetStream or Sound.
- void* owner;
-
-private:
- // The (un)compressed data
- const guint8* data;
-
-};
-
-
-// Used to hold the sounddata when doing on-demand-decoding
-class sound_data
-{
- // The (un)compressed data
- Buffer _data;
-
-public:
-
- sound_data()
- {}
-
- /// Append size bytes to this sound
- //
- /// @param data
- /// Data bytes, allocated with new[]. Ownership transferred.
- ///
- /// @param size
- /// Size of the 'data' buffer.
- ///
- void append(unsigned char* data, unsigned int size)
- {
- _data.append(data, size);
- }
-
- /// Return data size
- size_t dataSize() const { return _data.size(); }
-
- /// Return data buffer
- const boost::uint8_t* data() { return _data.data(); }
-
- // Object holding information about the sound
- std::auto_ptr<SoundInfo> soundinfo;
-
- // Volume, SWF range: 0-100, GST range 0-10 (we only use 0-1, the rest
is amplified)
- // It's the SWF range that is represented here
- int volume;
-
- // gstreamer objects
- std::vector<gst_elements*> m_gst_elements;
-
-};
-
-// Use gstreamer to handle sounds.
-class GST_sound_handler : public sound_handler
-{
-private:
- /// AS classes (NetStream, Sound) audio callbacks
- typedef std::map< void* /* owner */, aux_streamer_ptr /* callback */>
CallbacksMap;
- CallbacksMap m_aux_streamer;
-
- typedef std::map< void* /* owner */, gst_elements*> GstElementsMap;
- GstElementsMap m_aux_streamer_gstelements;
-
- /// Vector containing all the sounds
- std::vector<sound_data*> m_sound_data;
-
- /// Is the loop running?
- bool looping;
-
- /// Is the audio muted?
- bool muted;
-
- /// Mutex for making sure threads doesn't mess things up
- boost::try_mutex _mutex;
-
- /// stop and delete all sounds
- void delete_all_sounds();
-
-public:
-
- /// Gstreamer callback function
- static void callback_handoff (GstElement * /*c*/, GstBuffer *buffer,
GstPad* /*pad*/, gpointer user_data);
-
- /// Gstreamer callback function when using ActionScript audio source
(NetStream, Sound)
- static void callback_as_handoff (GstElement * /*c*/, GstBuffer *buffer,
GstPad* /*pad*/, gpointer user_data);
-
- GST_sound_handler();
- virtual ~GST_sound_handler();
-
- /// Called to create a sound.
- virtual int create_sound(void* data, unsigned int data_bytes,
- std::auto_ptr<SoundInfo> sinfo);
-
- /// this gets called when a stream gets more data
- virtual long fill_stream_data(unsigned char* data, unsigned int
data_bytes,
- unsigned int sample_count, int
handle_id);
-
- /// Play the index'd sample.
- virtual void play_sound(int sound_handle, int loop_count, int offset,
- long start_position, const
std::vector<sound_envelope>* envelopes);
-
- /// Stop the index'd sample.
- virtual void stop_sound(int sound_handle);
-
- /// This gets called when it's done with a sample.
- virtual void delete_sound(int sound_handle);
-
- /// This will stop all sounds playing.
- virtual void stop_all_sounds();
-
- // See dox in sound_handler.h
- virtual void reset();
-
- /// Returns the sound volume level as an integer from 0 to 100.
AS-script only.
- virtual int get_volume(int sound_handle);
-
- /// Sets the sound volume level as an integer from 0 to 100. AS-script
only.
- virtual void set_volume(int sound_handle, int volume);
-
- /// Gnash uses this to get info about a sound. Used when a stream needs
more data.
- virtual SoundInfo* get_sound_info(int sound_handle);
-
- /// Gnash calls this to mute audio.
- virtual void mute();
-
- /// Gnash calls this to unmute audio.
- virtual void unmute();
-
- /// Gnash calls this to get the mute state.
- virtual bool is_muted();
-
- /// Gets the duration in milliseconds of an event sound connected to an
AS Sound obejct.
- virtual unsigned int get_duration(int sound_handle);
-
- /// Gets the playhead position in milliseconds of an event sound
connected to an AS Soound obejct.
- virtual unsigned int get_position(int sound_handle);
-
- virtual void attach_aux_streamer(aux_streamer_ptr ptr, void* owner);
- virtual void detach_aux_streamer(void* owner);
-
-};
-
-} // gnash.media namespace
-} // namespace gnash
-
-#endif // SOUND_HANDLER_GST_H
-
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