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Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11

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This commit is contained in:
David Schleef 2012-03-06 21:36:02 -08:00
commit 7831feced5
81 changed files with 2085 additions and 2607 deletions
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3
autogen.sh
View file

@ -29,6 +29,9 @@ then
ln -s ../../common/hooks/pre-commit.hook .git/hooks/pre-commit
fi
# GNU gettext automake support doesn't get along with git.
# https://bugzilla.gnome.org/show_bug.cgi?id=661128
touch -t 200001010000 po/gst-plugins-good-0.10.pot
CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc'

3
configure.ac
View file

@ -314,9 +314,8 @@ dnl Make sure you have a space before and after all plugins
GST_PLUGINS_NONPORTED="deinterlace interleave flx \
imagefreeze smpte \
videobox \
cairo cairo_gobject dv1394 gdk_pixbuf \
cairo cairo_gobject dv1394 \
oss oss4 \
wavpack \
osx_video osx_audio "
AC_SUBST(GST_PLUGINS_NONPORTED)

1
docs/plugins/Makefile.am
View file

@ -84,7 +84,6 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/speex/gstspeexdec.h \
$(top_srcdir)/ext/wavpack/gstwavpackdec.h \
$(top_srcdir)/ext/wavpack/gstwavpackenc.h \
$(top_srcdir)/ext/wavpack/gstwavpackparse.h \
$(top_srcdir)/gst/alpha/gstalpha.h \
$(top_srcdir)/gst/alpha/gstalphacolor.h \
$(top_srcdir)/gst/apetag/gstapedemux.h \

5
ext/aalib/gstaasink.c
View file

@ -273,7 +273,7 @@ gst_aasink_init (GstAASink * aasink)
static void
gst_aasink_scale (GstAASink * aasink, guchar * src, guchar * dest,
gint sw, gint sh, gint dw, gint dh)
gint sw, gint sh, gint ss, gint dw, gint dh)
{
gint ypos, yinc, y;
gint xpos, xinc, x;
@ -287,7 +287,7 @@ gst_aasink_scale (GstAASink * aasink, guchar * src, guchar * dest,
for (y = dh; y; y--) {
while (ypos > 0x10000) {
ypos -= 0x10000;
src += sw;
src += ss;
}
xpos = 0x10000;
{
@ -334,6 +334,7 @@ gst_aasink_render (GstBaseSink * basesink, GstBuffer * buffer)
aa_image (aasink->context), /* dest */
GST_VIDEO_INFO_WIDTH (&aasink->info), /* sw */
GST_VIDEO_INFO_HEIGHT (&aasink->info), /* sh */
GST_VIDEO_FRAME_PLANE_STRIDE (&frame, 0), /* ss */
aa_imgwidth (aasink->context), /* dw */
aa_imgheight (aasink->context)); /* dh */

4
ext/annodex/gstannodex.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
ext/annodex/gstcmmlparser.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <stdarg.h>
#include <gst/gst.h>

4
ext/annodex/gstcmmltag.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

41
ext/flac/gstflacenc.c
View file

@ -761,7 +761,6 @@ gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
encoder_already_initialized:
{
g_warning ("flac already initialized -- fixme allow this");
gst_object_unref (flacenc);
return FALSE;
}
setting_src_caps_failed:
@ -769,14 +768,12 @@ setting_src_caps_failed:
GST_DEBUG_OBJECT (flacenc,
"Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
gst_object_unref (flacenc);
return FALSE;
}
failed_to_initialize:
{
GST_ELEMENT_ERROR (flacenc, LIBRARY, INIT, (NULL),
("could not initialize encoder (wrong parameters?)"));
gst_object_unref (flacenc);
return FALSE;
}
}
@ -1104,43 +1101,9 @@ gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
gint64 start, stream_time;
if (flacenc->offset == 0) {
gst_event_copy_segment (event, &seg);
start = seg.start;
stream_time = seg.time;
} else {
start = -1;
stream_time = -1;
}
if (start > 0) {
if (flacenc->offset > 0)
GST_DEBUG ("Not handling mid-stream newsegment event");
else
GST_DEBUG ("Not handling newsegment event with non-zero start");
} else {
GstEvent *e;
gst_segment_init (&seg, GST_FORMAT_BYTES);
e = gst_event_new_segment (&seg);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), e);
}
if (stream_time > 0) {
GST_DEBUG ("Not handling non-zero stream time");
}
/* don't push it downstream, we'll generate our own via seek to 0 */
gst_event_unref (event);
ret = TRUE;
break;
}
case GST_EVENT_EOS:
flacenc->eos = TRUE;
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event);
break;
case GST_EVENT_TAG:
if (flacenc->tags) {
@ -1150,8 +1113,10 @@ gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
} else {
g_assert_not_reached ();
}
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event);
break;
default:
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event);
break;
}

310
ext/gdk_pixbuf/gstgdkpixbuf.c
View file

@ -36,7 +36,6 @@ GST_DEBUG_CATEGORY_STATIC (gst_gdk_pixbuf_debug);
enum
{
ARG_0,
ARG_SILENT /* FIXME 0.11: remove */
};
static GstStaticPadTemplate gst_gdk_pixbuf_sink_template =
@ -67,44 +66,40 @@ static GstStaticPadTemplate gst_gdk_pixbuf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB "; " GST_VIDEO_CAPS_RGBA)
GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB") "; "
GST_VIDEO_CAPS_MAKE ("RGBA"))
);
static void gst_gdk_pixbuf_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_gdk_pixbuf_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn
gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition);
static GstFlowReturn gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static gboolean gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
#ifdef enable_typefind
static void gst_gdk_pixbuf_type_find (GstTypeFind * tf, gpointer ignore);
#endif
GST_BOILERPLATE (GstGdkPixbuf, gst_gdk_pixbuf, GstElement, GST_TYPE_ELEMENT);
#define gst_gdk_pixbuf_parent_class parent_class
G_DEFINE_TYPE (GstGdkPixbuf, gst_gdk_pixbuf, GST_TYPE_ELEMENT);
static gboolean
gst_gdk_pixbuf_sink_setcaps (GstPad * pad, GstCaps * caps)
gst_gdk_pixbuf_sink_setcaps (GstGdkPixbuf * filter, GstCaps * caps)
{
GstGdkPixbuf *filter;
const GValue *framerate;
GstStructure *s;
filter = GST_GDK_PIXBUF (GST_PAD_PARENT (pad));
s = gst_caps_get_structure (caps, 0);
if ((framerate = gst_structure_get_value (s, "framerate")) != NULL) {
filter->framerate_numerator = gst_value_get_fraction_numerator (framerate);
filter->framerate_denominator =
gst_value_get_fraction_denominator (framerate);
filter->in_fps_n = gst_value_get_fraction_numerator (framerate);
filter->in_fps_d = gst_value_get_fraction_denominator (framerate);
GST_DEBUG_OBJECT (filter, "got framerate of %d/%d fps => packetized mode",
filter->framerate_numerator, filter->framerate_denominator);
filter->in_fps_n, filter->in_fps_d);
} else {
filter->framerate_numerator = 0;
filter->framerate_denominator = 1;
filter->in_fps_n = 0;
filter->in_fps_d = 1;
GST_DEBUG_OBJECT (filter, "no framerate, assuming single image");
}
@ -112,7 +107,7 @@ gst_gdk_pixbuf_sink_setcaps (GstPad * pad, GstCaps * caps)
}
static GstCaps *
gst_gdk_pixbuf_get_capslist (void)
gst_gdk_pixbuf_get_capslist (GstCaps * filter)
{
GSList *slist;
GSList *slist0;
@ -132,7 +127,7 @@ gst_gdk_pixbuf_get_capslist (void)
mimetypes = gdk_pixbuf_format_get_mime_types (pixbuf_format);
for (mimetype = mimetypes; *mimetype; mimetype++) {
gst_caps_append_structure (capslist, gst_structure_new (*mimetype, NULL));
gst_caps_append_structure (capslist, gst_structure_new_empty (*mimetype));
}
g_strfreev (mimetypes);
}
@ -143,61 +138,72 @@ gst_gdk_pixbuf_get_capslist (void)
gst_caps_unref (tmpl_caps);
gst_caps_unref (capslist);
if (filter && return_caps) {
GstCaps *temp;
temp = gst_caps_intersect (return_caps, filter);
gst_caps_unref (return_caps);
return_caps = temp;
}
return return_caps;
}
static GstCaps *
gst_gdk_pixbuf_sink_getcaps (GstPad * pad)
static gboolean
gst_gdk_pixbuf_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
return gst_gdk_pixbuf_get_capslist ();
gboolean res;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_gdk_pixbuf_get_capslist (filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static void
gst_gdk_pixbuf_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_gdk_pixbuf_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_gdk_pixbuf_sink_template));
gst_element_class_set_details_simple (element_class,
"GdkPixbuf image decoder", "Codec/Decoder/Image",
"Decodes images in a video stream using GdkPixbuf",
"David A. Schleef <ds@schleef.org>, Renato Filho <renato.filho@indt.org.br>");
}
/* initialize the plugin's class */
static void
gst_gdk_pixbuf_class_init (GstGdkPixbufClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_gdk_pixbuf_set_property;
gobject_class->get_property = gst_gdk_pixbuf_get_property;
g_object_class_install_property (gobject_class, ARG_SILENT,
g_param_spec_boolean ("silent", "Silent",
"Produce verbose output ? (deprecated)", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_change_state);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_gdk_pixbuf_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_gdk_pixbuf_sink_template));
gst_element_class_set_details_simple (gstelement_class,
"GdkPixbuf image decoder", "Codec/Decoder/Image",
"Decodes images in a video stream using GdkPixbuf",
"David A. Schleef <ds@schleef.org>, Renato Filho <renato.filho@indt.org.br>");
}
static void
gst_gdk_pixbuf_init (GstGdkPixbuf * filter, GstGdkPixbufClass * klass)
gst_gdk_pixbuf_init (GstGdkPixbuf * filter)
{
filter->sinkpad =
gst_pad_new_from_static_template (&gst_gdk_pixbuf_sink_template, "sink");
gst_pad_set_setcaps_function (filter->sinkpad,
GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_setcaps));
gst_pad_set_getcaps_function (filter->sinkpad,
GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_getcaps));
gst_pad_set_query_function (filter->sinkpad,
GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_query));
gst_pad_set_chain_function (filter->sinkpad,
GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_chain));
gst_pad_set_event_function (filter->sinkpad,
@ -213,6 +219,55 @@ gst_gdk_pixbuf_init (GstGdkPixbuf * filter, GstGdkPixbufClass * klass)
filter->pixbuf_loader = NULL;
}
static gboolean
gst_gdk_pixbuf_setup_pool (GstGdkPixbuf * filter, GstVideoInfo * info)
{
GstCaps *target;
GstQuery *query;
GstBufferPool *pool = NULL;
guint size, min, max, prefix, alignment;
target = gst_pad_get_current_caps (filter->srcpad);
/* try to get a bufferpool now */
/* find a pool for the negotiated caps now */
query = gst_query_new_allocation (target, TRUE);
if (gst_pad_peer_query (filter->srcpad, query)) {
/* we got configuration from our peer, parse them */
gst_query_parse_allocation_params (query, &size, &min, &max, &prefix,
&alignment, &pool);
} else {
size = info->size;
min = max = 0;
prefix = 0;
alignment = 0;
}
if (pool == NULL) {
GstStructure *config;
/* we did not get a pool, make one ourselves then */
pool = gst_buffer_pool_new ();
config = gst_buffer_pool_get_config (pool);
gst_buffer_pool_config_set (config, target, size, min, max, prefix,
alignment);
gst_buffer_pool_set_config (pool, config);
}
if (filter->pool)
gst_object_unref (filter->pool);
filter->pool = pool;
/* and activate */
gst_buffer_pool_set_active (filter->pool, TRUE);
gst_caps_unref (target);
return TRUE;
}
static GstFlowReturn
gst_gdk_pixbuf_flush (GstGdkPixbuf * filter)
{
@ -221,48 +276,54 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter)
int y;
guint8 *out_pix;
guint8 *in_pix;
int in_rowstride;
int in_rowstride, out_rowstride;
GstFlowReturn ret;
GstCaps *caps = NULL;
gint width, height;
gint n_channels;
GstVideoFrame frame;
pixbuf = gdk_pixbuf_loader_get_pixbuf (filter->pixbuf_loader);
if (pixbuf == NULL)
goto no_pixbuf;
if (filter->image_size == 0) {
filter->width = gdk_pixbuf_get_width (pixbuf);
filter->height = gdk_pixbuf_get_height (pixbuf);
filter->rowstride = gdk_pixbuf_get_rowstride (pixbuf);
filter->image_size = filter->rowstride * filter->height;
width = gdk_pixbuf_get_width (pixbuf);
height = gdk_pixbuf_get_height (pixbuf);
if (GST_VIDEO_INFO_FORMAT (&filter->info) == GST_VIDEO_FORMAT_UNKNOWN) {
GstVideoInfo info;
GstVideoFormat fmt;
GST_DEBUG ("Set size to %dx%d", width, height);
n_channels = gdk_pixbuf_get_n_channels (pixbuf);
switch (n_channels) {
case 3:
caps = gst_caps_from_string (GST_VIDEO_CAPS_RGB);
fmt = GST_VIDEO_FORMAT_RGB;
break;
case 4:
caps = gst_caps_from_string (GST_VIDEO_CAPS_RGBA);
fmt = GST_VIDEO_FORMAT_RGBA;
break;
default:
goto channels_not_supported;
}
gst_caps_set_simple (caps,
"width", G_TYPE_INT, filter->width,
"height", G_TYPE_INT, filter->height,
"framerate", GST_TYPE_FRACTION, filter->framerate_numerator,
filter->framerate_denominator, NULL);
GST_DEBUG ("Set size to %dx%d", filter->width, filter->height);
gst_video_info_init (&info);
gst_video_info_set_format (&info, fmt, width, height);
info.fps_n = filter->in_fps_n;
info.fps_d = filter->in_fps_d;
caps = gst_video_info_to_caps (&info);
filter->info = info;
gst_pad_set_caps (filter->srcpad, caps);
gst_caps_unref (caps);
gst_gdk_pixbuf_setup_pool (filter, &info);
}
ret = gst_pad_alloc_buffer_and_set_caps (filter->srcpad,
GST_BUFFER_OFFSET_NONE,
filter->image_size, GST_PAD_CAPS (filter->srcpad), &outbuf);
ret = gst_buffer_pool_acquire_buffer (filter->pool, &outbuf, NULL);
if (ret != GST_FLOW_OK)
goto no_buffer;
@ -271,16 +332,20 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter)
in_pix = gdk_pixbuf_get_pixels (pixbuf);
in_rowstride = gdk_pixbuf_get_rowstride (pixbuf);
out_pix = GST_BUFFER_DATA (outbuf);
/* FIXME, last line might not have rowstride pixels */
for (y = 0; y < filter->height; y++) {
memcpy (out_pix, in_pix, filter->rowstride);
gst_video_frame_map (&frame, &filter->info, outbuf, GST_MAP_WRITE);
out_pix = GST_VIDEO_FRAME_PLANE_DATA (&frame, 0);
out_rowstride = GST_VIDEO_FRAME_PLANE_STRIDE (&frame, 0);
for (y = 0; y < height; y++) {
memcpy (out_pix, in_pix, width * GST_VIDEO_FRAME_COMP_PSTRIDE (&frame, 0));
in_pix += in_rowstride;
out_pix += filter->rowstride;
out_pix += out_rowstride;
}
GST_DEBUG ("pushing... %d bytes", GST_BUFFER_SIZE (outbuf));
gst_video_frame_unmap (&frame);
GST_DEBUG ("pushing... %d bytes", gst_buffer_get_size (outbuf));
ret = gst_pad_push (filter->srcpad, outbuf);
if (ret != GST_FLOW_OK)
@ -308,15 +373,24 @@ no_buffer:
}
static gboolean
gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event)
gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstFlowReturn res = GST_FLOW_OK;
gboolean ret = TRUE;
gboolean ret = TRUE, forward = TRUE;
GstGdkPixbuf *pixbuf;
pixbuf = GST_GDK_PIXBUF (gst_pad_get_parent (pad));
pixbuf = GST_GDK_PIXBUF (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_gdk_pixbuf_sink_setcaps (pixbuf, caps);
forward = FALSE;
break;
}
case GST_EVENT_EOS:
if (pixbuf->pixbuf_loader != NULL) {
gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL);
@ -329,10 +403,12 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event)
if (res != GST_FLOW_OK && res != GST_FLOW_FLUSHING) {
GST_ELEMENT_ERROR (pixbuf, STREAM, FAILED, (NULL),
("Flow: %s", gst_flow_get_name (res)));
forward = FALSE;
ret = FALSE;
}
}
break;
case GST_EVENT_NEWSEGMENT:
case GST_EVENT_SEGMENT:
case GST_EVENT_FLUSH_STOP:
if (pixbuf->pixbuf_loader != NULL) {
gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL);
@ -343,29 +419,24 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event)
default:
break;
}
if (res == GST_FLOW_OK) {
ret = gst_pad_event_default (pad, event);
if (forward) {
ret = gst_pad_event_default (pad, parent, event);
} else {
ret = FALSE;
gst_event_unref (event);
}
gst_object_unref (pixbuf);
return ret;
}
static GstFlowReturn
gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buf)
gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstGdkPixbuf *filter;
GstFlowReturn ret = GST_FLOW_OK;
GError *error = NULL;
GstClockTime timestamp;
guint8 *data;
guint size;
GstMapInfo map;
filter = GST_GDK_PIXBUF (gst_pad_get_parent (pad));
filter = GST_GDK_PIXBUF (parent);
timestamp = GST_BUFFER_TIMESTAMP (buf);
@ -378,23 +449,23 @@ gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buf)
if (filter->pixbuf_loader == NULL)
filter->pixbuf_loader = gdk_pixbuf_loader_new ();
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
gst_buffer_map (buf, &map, GST_MAP_READ);
GST_LOG_OBJECT (filter, "Writing buffer size %d", size);
if (!gdk_pixbuf_loader_write (filter->pixbuf_loader, data, size, &error))
GST_LOG_OBJECT (filter, "Writing buffer size %d", (gint) map.size);
if (!gdk_pixbuf_loader_write (filter->pixbuf_loader, map.data, map.size,
&error))
goto error;
/* packetised mode? */
if (filter->framerate_numerator != 0) {
if (filter->in_fps_n != 0) {
gdk_pixbuf_loader_close (filter->pixbuf_loader, NULL);
ret = gst_gdk_pixbuf_flush (filter);
g_object_unref (filter->pixbuf_loader);
filter->pixbuf_loader = NULL;
}
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
gst_object_unref (filter);
return ret;
@ -404,40 +475,12 @@ error:
GST_ELEMENT_ERROR (filter, STREAM, DECODE, (NULL),
("gdk_pixbuf_loader_write error: %s", error->message));
g_error_free (error);
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
gst_object_unref (filter);
return GST_FLOW_ERROR;
}
}
static void
gst_gdk_pixbuf_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
case ARG_SILENT:
/* filter->silent = g_value_get_boolean (value); */
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_gdk_pixbuf_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
case ARG_SILENT:
/* g_value_set_boolean (value, filter->silent); */
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition)
{
@ -447,8 +490,9 @@ gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition)
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* default to single image mode, setcaps function might not be called */
dec->framerate_numerator = 0;
dec->framerate_denominator = 1;
dec->in_fps_n = 0;
dec->in_fps_d = 1;
gst_video_info_init (&dec->info);
break;
default:
break;
@ -460,8 +504,12 @@ gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition)
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
dec->framerate_numerator = 0;
dec->framerate_denominator = 0;
dec->in_fps_n = 0;
dec->in_fps_d = 0;
if (dec->pool) {
gst_buffer_pool_set_active (dec->pool, FALSE);
gst_object_replace ((GstObject **) & dec->pool, NULL);
}
break;
default:
break;

9
ext/gdk_pixbuf/gstgdkpixbuf.h
View file

@ -49,13 +49,10 @@ struct _GstGdkPixbuf
GstClockTime last_timestamp;
GdkPixbufLoader *pixbuf_loader;
int width;
int height;
int rowstride;
unsigned int image_size;
gint in_fps_n, in_fps_d;
gint framerate_numerator;
gint framerate_denominator;
GstVideoInfo info;
GstBufferPool *pool;
};
struct _GstGdkPixbufClass

85
ext/gdk_pixbuf/gstgdkpixbufsink.c
View file

@ -103,15 +103,13 @@
enum
{
PROP_0,
PROP_SEND_MESSAGES,
PROP_POST_MESSAGES,
PROP_LAST_PIXBUF,
PROP_LAST
};
GST_BOILERPLATE (GstGdkPixbufSink, gst_gdk_pixbuf_sink, GstVideoSink,
GST_TYPE_VIDEO_SINK);
G_DEFINE_TYPE (GstGdkPixbufSink, gst_gdk_pixbuf_sink, GST_TYPE_VIDEO_SINK);
static void gst_gdk_pixbuf_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
@ -129,18 +127,20 @@ static GstFlowReturn gst_gdk_pixbuf_sink_preroll (GstBaseSink * bsink,
static GdkPixbuf *gst_gdk_pixbuf_sink_get_pixbuf_from_buffer (GstGdkPixbufSink *
sink, GstBuffer * buf);
#define WxH ", width = (int) [ 16, 4096 ], height = (int) [ 16, 4096 ]"
static GstStaticPadTemplate pixbufsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB WxH ";" GST_VIDEO_CAPS_RGBA WxH));
GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB") ";"
GST_VIDEO_CAPS_MAKE ("RGBA"))
);
static void
gst_gdk_pixbuf_sink_base_init (gpointer g_class)
gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gst_element_class_set_details_simple (element_class, "GdkPixbuf sink",
"Sink/Video", "Output images as GdkPixbuf objects in bus messages",
@ -148,23 +148,10 @@ gst_gdk_pixbuf_sink_base_init (gpointer g_class)
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&pixbufsink_sink_factory));
}
static void
gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass)
{
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gobject_class->set_property = gst_gdk_pixbuf_sink_set_property;
gobject_class->get_property = gst_gdk_pixbuf_sink_get_property;
/* FIXME 0.11, remove in favour of post-messages */
g_object_class_install_property (gobject_class, PROP_SEND_MESSAGES,
g_param_spec_boolean ("send-messages", "Send Messages",
"Whether to post messages containing pixbufs on the bus "
" (deprecated, use post-messages)",
DEFAULT_SEND_MESSAGES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstGdkPixbuf:post-messages:
*
@ -190,8 +177,7 @@ gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass)
}
static void
gst_gdk_pixbuf_sink_init (GstGdkPixbufSink * sink,
GstGdkPixbufSinkClass * klass)
gst_gdk_pixbuf_sink_init (GstGdkPixbufSink * sink)
{
sink->par_n = 0;
sink->par_d = 0;
@ -240,46 +226,51 @@ static gboolean
gst_gdk_pixbuf_sink_set_caps (GstBaseSink * basesink, GstCaps * caps)
{
GstGdkPixbufSink *sink = GST_GDK_PIXBUF_SINK (basesink);
GstVideoInfo info;
GstVideoFormat fmt;
gint w, h, par_n, par_d;
gint w, h, s, par_n, par_d;
GST_LOG_OBJECT (sink, "caps: %" GST_PTR_FORMAT, caps);
if (!gst_video_format_parse_caps (caps, &fmt, &w, &h)) {
if (!gst_video_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (sink, "parse_caps failed");
return FALSE;
}
if (!gst_video_parse_caps_pixel_aspect_ratio (caps, &par_n, &par_d)) {
GST_LOG_OBJECT (sink, "no pixel aspect ratio");
return FALSE;
}
fmt = GST_VIDEO_INFO_FORMAT (&info);
w = GST_VIDEO_INFO_WIDTH (&info);
h = GST_VIDEO_INFO_HEIGHT (&info);
s = GST_VIDEO_INFO_COMP_PSTRIDE (&info, 0);
par_n = GST_VIDEO_INFO_PAR_N (&info);
par_d = GST_VIDEO_INFO_PAR_N (&info);
g_assert ((fmt == GST_VIDEO_FORMAT_RGB &&
gst_video_format_get_pixel_stride (fmt, 0) == 3) ||
(fmt == GST_VIDEO_FORMAT_RGBA &&
gst_video_format_get_pixel_stride (fmt, 0) == 4));
g_assert ((fmt == GST_VIDEO_FORMAT_RGB && s == 3) ||
(fmt == GST_VIDEO_FORMAT_RGBA && s == 4));
GST_VIDEO_SINK_WIDTH (sink) = w;
GST_VIDEO_SINK_HEIGHT (sink) = h;
sink->rowstride = gst_video_format_get_row_stride (fmt, 0, w);
sink->has_alpha = (fmt == GST_VIDEO_FORMAT_RGBA);
sink->par_n = par_n;
sink->par_d = par_d;
sink->has_alpha = GST_VIDEO_INFO_HAS_ALPHA (&info);
GST_INFO_OBJECT (sink, "format : %d", fmt);
GST_INFO_OBJECT (sink, "width x height : %d x %d", w, h);
GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_n, par_d);
sink->info = info;
return TRUE;
}
static void
gst_gdk_pixbuf_sink_pixbuf_destroy_notify (guchar * pixels, GstBuffer * buf)
gst_gdk_pixbuf_sink_pixbuf_destroy_notify (guchar * pixels,
GstVideoFrame * frame)
{
gst_buffer_unref (buf);
gst_video_frame_unmap (frame);
gst_buffer_unref (frame->buffer);
g_slice_free (GstVideoFrame, frame);
}
static GdkPixbuf *
@ -287,24 +278,30 @@ gst_gdk_pixbuf_sink_get_pixbuf_from_buffer (GstGdkPixbufSink * sink,
GstBuffer * buf)
{
GdkPixbuf *pix = NULL;
GstVideoFrame *frame;
gint minsize, bytes_per_pixel;
g_return_val_if_fail (GST_VIDEO_SINK_WIDTH (sink) > 0, NULL);
g_return_val_if_fail (GST_VIDEO_SINK_HEIGHT (sink) > 0, NULL);
frame = g_slice_new0 (GstVideoFrame);
gst_video_frame_map (frame, &sink->info, buf, GST_MAP_READ);
bytes_per_pixel = (sink->has_alpha) ? 4 : 3;
/* last row needn't have row padding */
minsize = (sink->rowstride * (GST_VIDEO_SINK_HEIGHT (sink) - 1)) +
minsize = (GST_VIDEO_FRAME_COMP_STRIDE (frame, 0) *
(GST_VIDEO_SINK_HEIGHT (sink) - 1)) +
(bytes_per_pixel * GST_VIDEO_SINK_WIDTH (sink));
g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= minsize, NULL);
g_return_val_if_fail (gst_buffer_get_size (buf) >= minsize, NULL);
pix = gdk_pixbuf_new_from_data (GST_BUFFER_DATA (buf),
gst_buffer_ref (buf);
pix = gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (frame, 0),
GDK_COLORSPACE_RGB, sink->has_alpha, 8, GST_VIDEO_SINK_WIDTH (sink),
GST_VIDEO_SINK_HEIGHT (sink), sink->rowstride,
GST_VIDEO_SINK_HEIGHT (sink), GST_VIDEO_FRAME_COMP_STRIDE (frame, 0),
(GdkPixbufDestroyNotify) gst_gdk_pixbuf_sink_pixbuf_destroy_notify,
gst_buffer_ref (buf));
frame);
return pix;
}
@ -387,7 +384,6 @@ gst_gdk_pixbuf_sink_set_property (GObject * object, guint prop_id,
sink = GST_GDK_PIXBUF_SINK (object);
switch (prop_id) {
case PROP_SEND_MESSAGES:
case PROP_POST_MESSAGES:
GST_OBJECT_LOCK (sink);
sink->post_messages = g_value_get_boolean (value);
@ -408,7 +404,6 @@ gst_gdk_pixbuf_sink_get_property (GObject * object, guint prop_id,
sink = GST_GDK_PIXBUF_SINK (object);
switch (prop_id) {
case PROP_SEND_MESSAGES:
case PROP_POST_MESSAGES:
GST_OBJECT_LOCK (sink);
g_value_set_boolean (value, sink->post_messages);

4
ext/gdk_pixbuf/gstgdkpixbufsink.h
View file

@ -21,7 +21,7 @@
#define GST_GDK_PIXBUF_SINK_H
#include <gst/gst.h>
#include <gst/video/video.h>
#include <gst/video/gstvideosink.h>
#include <gdk-pixbuf/gdk-pixbuf.h>
@ -47,9 +47,9 @@ struct _GstGdkPixbufSink
/*< private >*/
/* current caps */
GstVideoInfo info;
gint width;
gint height;
gint rowstride;
gint par_n;
gint par_d;
gboolean has_alpha;

187
ext/gdk_pixbuf/pixbufscale.c
View file

@ -57,14 +57,14 @@ static GstStaticPadTemplate gst_pixbufscale_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB)
GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB"))
);
static GstStaticPadTemplate gst_pixbufscale_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB)
GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB"))
);
#define GST_TYPE_PIXBUFSCALE_METHOD (gst_pixbufscale_method_get_type())
@ -93,28 +93,32 @@ static void gst_pixbufscale_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_pixbufscale_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps);
static gboolean gst_pixbufscale_set_caps (GstBaseTransform * trans,
GstCaps * in, GstCaps * out);
static gboolean gst_pixbufscale_get_unit_size (GstBaseTransform * trans,
GstCaps * caps, guint * size);
static void gst_pixbufscale_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static gboolean gst_pixbufscale_set_info (GstVideoFilter * filter,
GstCaps * in, GstVideoInfo * in_info, GstCaps * out,
GstVideoInfo * out_info);
static GstCaps *gst_pixbufscale_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
static GstFlowReturn gst_pixbufscale_transform (GstBaseTransform * trans,
GstBuffer * in, GstBuffer * out);
static gboolean gst_pixbufscale_handle_src_event (GstPad * pad,
static GstFlowReturn gst_pixbufscale_transform_frame (GstVideoFilter * filter,
GstVideoFrame * in, GstVideoFrame * out);
static gboolean gst_pixbufscale_handle_src_event (GstBaseTransform * trans,
GstEvent * event);
static gboolean parse_caps (GstCaps * caps, gint * width, gint * height);
GST_BOILERPLATE (GstPixbufScale, gst_pixbufscale, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM);
#define gst_pixbufscale_parent_class parent_class
G_DEFINE_TYPE (GstPixbufScale, gst_pixbufscale, GST_TYPE_VIDEO_FILTER);
static void
gst_pixbufscale_base_init (gpointer g_class)
gst_pixbufscale_class_init (GstPixbufScaleClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GObjectClass *gobject_class;
GstElementClass *element_class;
GstBaseTransformClass *trans_class;
GstVideoFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
element_class = (GstElementClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstVideoFilterClass *) klass;
gst_element_class_set_details_simple (element_class, "GdkPixbuf image scaler",
"Filter/Effect/Video", "Resizes video",
@ -126,16 +130,6 @@ gst_pixbufscale_base_init (gpointer g_class)
gst_static_pad_template_get (&gst_pixbufscale_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_pixbufscale_sink_template));
}
static void
gst_pixbufscale_class_init (GstPixbufScaleClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
gobject_class->set_property = gst_pixbufscale_set_property;
gobject_class->get_property = gst_pixbufscale_get_property;
@ -148,26 +142,21 @@ gst_pixbufscale_class_init (GstPixbufScaleClass * klass)
trans_class->transform_caps =
GST_DEBUG_FUNCPTR (gst_pixbufscale_transform_caps);
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_pixbufscale_set_caps);
trans_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_pixbufscale_get_unit_size);
trans_class->transform = GST_DEBUG_FUNCPTR (gst_pixbufscale_transform);
trans_class->fixate_caps = GST_DEBUG_FUNCPTR (gst_pixbufscale_fixate_caps);
trans_class->src_event = GST_DEBUG_FUNCPTR (gst_pixbufscale_handle_src_event);
trans_class->passthrough_on_same_caps = TRUE;
filter_class->set_info = GST_DEBUG_FUNCPTR (gst_pixbufscale_set_info);
filter_class->transform_frame =
GST_DEBUG_FUNCPTR (gst_pixbufscale_transform_frame);
parent_class = g_type_class_peek_parent (klass);
}
static void
gst_pixbufscale_init (GstPixbufScale * pixbufscale,
GstPixbufScaleClass * kclass)
gst_pixbufscale_init (GstPixbufScale * pixbufscale)
{
GstBaseTransform *trans;
GST_DEBUG_OBJECT (pixbufscale, "_init");
trans = GST_BASE_TRANSFORM (pixbufscale);
gst_pad_set_event_function (trans->srcpad, gst_pixbufscale_handle_src_event);
pixbufscale->method = GST_PIXBUFSCALE_TILES;
pixbufscale->gdk_method = GDK_INTERP_TILES;
@ -227,7 +216,7 @@ gst_pixbufscale_get_property (GObject * object, guint prop_id, GValue * value,
static GstCaps *
gst_pixbufscale_transform_caps (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * caps)
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstCaps *ret;
int i;
@ -243,76 +232,46 @@ gst_pixbufscale_transform_caps (GstBaseTransform * trans,
gst_structure_remove_field (structure, "pixel-aspect-ratio");
}
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (ret);
ret = intersection;
}
GST_DEBUG_OBJECT (trans, "returning caps: %" GST_PTR_FORMAT, ret);
return ret;
}
static gboolean
parse_caps (GstCaps * caps, gint * width, gint * height)
gst_pixbufscale_set_info (GstVideoFilter * filter, GstCaps * in,
GstVideoInfo * in_info, GstCaps * out, GstVideoInfo * out_info)
{
gboolean ret;
GstStructure *structure;
if (in_info->width == out_info->width && in_info->height == out_info->height) {
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), TRUE);
} else {
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), FALSE);
}
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", width);
ret &= gst_structure_get_int (structure, "height", height);
return ret;
}
static gboolean
gst_pixbufscale_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
GstPixbufScale *pixbufscale;
gboolean ret;
pixbufscale = GST_PIXBUFSCALE (trans);
ret = parse_caps (in, &pixbufscale->from_width, &pixbufscale->from_height);
ret &= parse_caps (out, &pixbufscale->to_width, &pixbufscale->to_height);
if (!ret)
goto done;
pixbufscale->from_stride = GST_ROUND_UP_4 (pixbufscale->from_width * 3);
pixbufscale->from_buf_size =
pixbufscale->from_stride * pixbufscale->from_height;
pixbufscale->to_stride = GST_ROUND_UP_4 (pixbufscale->to_width * 3);
pixbufscale->to_buf_size = pixbufscale->to_stride * pixbufscale->to_height;
GST_DEBUG_OBJECT (pixbufscale, "from=%dx%d, size %d -> to=%dx%d, size %d",
pixbufscale->from_width, pixbufscale->from_height,
pixbufscale->from_buf_size, pixbufscale->to_width, pixbufscale->to_height,
pixbufscale->to_buf_size);
done:
return ret;
}
static gboolean
gst_pixbufscale_get_unit_size (GstBaseTransform * trans,
GstCaps * caps, guint * size)
{
gint width, height;
g_assert (size);
if (!parse_caps (caps, &width, &height))
return FALSE;
*size = GST_ROUND_UP_4 (width * 3) * height;
GST_DEBUG_OBJECT (filter, "from=%dx%d, size %" G_GSIZE_FORMAT
" -> to=%dx%d, size %" G_GSIZE_FORMAT,
in_info->width, in_info->height, in_info->size,
out_info->width, out_info->height, out_info->size);
return TRUE;
}
static void
static GstCaps *
gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction,
GstCaps * caps, GstCaps * othercaps)
{
GstStructure *ins, *outs;
const GValue *from_par, *to_par;
g_return_if_fail (gst_caps_is_fixed (caps));
othercaps = gst_caps_make_writable (othercaps);
gst_caps_truncate (othercaps);
GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
" based on caps %" GST_PTR_FORMAT, othercaps, caps);
@ -337,7 +296,7 @@ gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction,
if (count == 2) {
GST_DEBUG_OBJECT (base, "dimensions already set to %dx%d, not fixating",
w, h);
return;
goto done;
}
gst_structure_get_int (ins, "width", &from_w);
@ -396,33 +355,39 @@ gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction,
}
}
done:
GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, othercaps);
return othercaps;
}
static GstFlowReturn
gst_pixbufscale_transform (GstBaseTransform * trans,
GstBuffer * in, GstBuffer * out)
gst_pixbufscale_transform_frame (GstVideoFilter * filter,
GstVideoFrame * in, GstVideoFrame * out)
{
GstPixbufScale *scale;
GdkPixbuf *src_pixbuf, *dest_pixbuf;
scale = GST_PIXBUFSCALE (trans);
scale = GST_PIXBUFSCALE (filter);
src_pixbuf =
gdk_pixbuf_new_from_data (GST_BUFFER_DATA (in), GDK_COLORSPACE_RGB, FALSE,
8, scale->from_width, scale->from_height,
GST_RGB24_ROWSTRIDE (scale->from_width), NULL, NULL);
gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (in, 0),
GDK_COLORSPACE_RGB, FALSE, 8, GST_VIDEO_FRAME_WIDTH (in),
GST_VIDEO_FRAME_HEIGHT (in),
GST_VIDEO_FRAME_COMP_STRIDE (in, 0), NULL, NULL);
dest_pixbuf =
gdk_pixbuf_new_from_data (GST_BUFFER_DATA (out), GDK_COLORSPACE_RGB,
FALSE, 8, scale->to_width, scale->to_height,
GST_RGB24_ROWSTRIDE (scale->to_width), NULL, NULL);
gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (out, 0),
GDK_COLORSPACE_RGB, FALSE, 8, GST_VIDEO_FRAME_WIDTH (out),
GST_VIDEO_FRAME_HEIGHT (out),
GST_VIDEO_FRAME_COMP_STRIDE (out, 0), NULL, NULL);
gdk_pixbuf_scale (src_pixbuf, dest_pixbuf, 0, 0,
scale->to_width,
scale->to_height, 0, 0,
(double) scale->to_width / scale->from_width,
(double) scale->to_height / scale->from_height, scale->gdk_method);
GST_VIDEO_FRAME_WIDTH (out),
GST_VIDEO_FRAME_HEIGHT (out), 0, 0,
(double) GST_VIDEO_FRAME_WIDTH (out) / GST_VIDEO_FRAME_WIDTH (in),
(double) GST_VIDEO_FRAME_HEIGHT (out) / GST_VIDEO_FRAME_HEIGHT (in),
scale->gdk_method);
g_object_unref (src_pixbuf);
g_object_unref (dest_pixbuf);
@ -431,14 +396,14 @@ gst_pixbufscale_transform (GstBaseTransform * trans,
}
static gboolean
gst_pixbufscale_handle_src_event (GstPad * pad, GstEvent * event)
gst_pixbufscale_handle_src_event (GstBaseTransform * trans, GstEvent * event)
{
GstPixbufScale *pixbufscale;
gboolean ret;
double a;
GstStructure *structure;
pixbufscale = GST_PIXBUFSCALE (gst_pad_get_parent (pad));
pixbufscale = GST_PIXBUFSCALE (trans);
GST_DEBUG_OBJECT (pixbufscale, "handling %s event",
GST_EVENT_TYPE_NAME (event));
@ -462,9 +427,7 @@ gst_pixbufscale_handle_src_event (GstPad * pad, GstEvent * event)
break;
}
ret = gst_pad_event_default (pad, event);
gst_object_unref (pixbufscale);
ret = GST_BASE_TRANSFORM_CLASS (parent_class)->src_event (trans, event);
return ret;
}

6
ext/gdk_pixbuf/pixbufscale.h
View file

@ -25,6 +25,8 @@
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/video/gstvideofilter.h>
#include <gdk-pixbuf/gdk-pixbuf.h>
G_BEGIN_DECLS
@ -52,7 +54,7 @@ typedef struct _GstPixbufScale GstPixbufScale;
typedef struct _GstPixbufScaleClass GstPixbufScaleClass;
struct _GstPixbufScale {
GstBaseTransform element;
GstVideoFilter element;
/* video state */
gint to_width;
@ -72,7 +74,7 @@ struct _GstPixbufScale {
};
struct _GstPixbufScaleClass {
GstBaseTransformClass parent_class;
GstVideoFilterClass parent_class;
};
GType gst_pixbufscale_get_type(void);

6
ext/jack/gstjack.c
View file

@ -52,16 +52,14 @@ gst_jack_transport_get_type (void)
static volatile gsize type = 0;
if (g_once_init_enter (&type)) {
static const GEnumValue enum_values[] = {
{GST_JACK_TRANSPORT_AUTONOMOUS,
"No transport support", "autonomous"},
static const GFlagsValue flag_values[] = {
{GST_JACK_TRANSPORT_MASTER,
"Start and stop transport with state changes", "master"},
{GST_JACK_TRANSPORT_SLAVE,
"Follow transport state changes", "slave"},
{0, NULL, NULL},
};
GType tmp = g_enum_register_static ("GstJackTransport", enum_values);
GType tmp = g_flags_register_static ("GstJackTransport", flag_values);
g_once_init_leave (&type, tmp);
}
return (GType) type;

6
ext/jack/gstjack.h
View file

@ -53,9 +53,9 @@ typedef enum {
* client behaviour regarding to the transport mechanism.
*/
typedef enum {
GST_JACK_TRANSPORT_AUTONOMOUS,
GST_JACK_TRANSPORT_MASTER,
GST_JACK_TRANSPORT_SLAVE
GST_JACK_TRANSPORT_AUTONOMOUS = 0,
GST_JACK_TRANSPORT_MASTER = (1 << 0),
GST_JACK_TRANSPORT_SLAVE = (1 << 1),
} GstJackTransport;
typedef jack_default_audio_sample_t sample_t;

4
ext/jack/gstjackaudioclient.c
View file

@ -88,10 +88,10 @@ static gboolean
jack_handle_transport_change (GstJackAudioClient * client, GstState state)
{
GstObject *obj = GST_OBJECT_PARENT (client->user_data);
GstJackTransport mode;
guint mode;
g_object_get (obj, "transport", &mode, NULL);
if ((mode == GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) {
if ((mode & GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) {
GST_INFO_OBJECT (obj, "requesting state change: %s",
gst_element_state_get_name (state));
gst_element_post_message (GST_ELEMENT (obj),

12
ext/jack/gstjackaudiosink.c
View file

@ -564,7 +564,7 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "start");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
@ -583,7 +583,7 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "pause");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
@ -602,7 +602,7 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (sink, "stop");
if (sink->transport == GST_JACK_TRANSPORT_MASTER) {
if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
@ -761,7 +761,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
* Since: 0.10.31
*/
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_enum ("transport", "Transport mode",
g_param_spec_flags ("transport", "Transport mode",
"Jack transport behaviour of the client",
GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -840,7 +840,7 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
}
break;
case PROP_TRANSPORT:
sink->transport = g_value_get_enum (value);
sink->transport = g_value_get_flags (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -870,7 +870,7 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
g_value_set_boxed (value, sink->jclient);
break;
case PROP_TRANSPORT:
g_value_set_enum (value, sink->transport);
g_value_set_flags (value, sink->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

2
ext/jack/gstjackaudiosink.h
View file

@ -59,7 +59,7 @@ struct _GstJackAudioSink {
gchar *server;
jack_client_t *jclient;
gchar *client_name;
GstJackTransport transport;
guint transport;
/* our client */
GstJackAudioClient *client;

12
ext/jack/gstjackaudiosrc.c
View file

@ -576,7 +576,7 @@ gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "start");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
@ -595,7 +595,7 @@ gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "pause");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
@ -614,7 +614,7 @@ gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
GST_DEBUG_OBJECT (src, "stop");
if (src->transport == GST_JACK_TRANSPORT_MASTER) {
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
@ -782,7 +782,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
* Since: 0.10.31
*/
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_enum ("transport", "Transport mode",
g_param_spec_flags ("transport", "Transport mode",
"Jack transport behaviour of the client",
GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -859,7 +859,7 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id,
}
break;
case PROP_TRANSPORT:
src->transport = g_value_get_enum (value);
src->transport = g_value_get_flags (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -887,7 +887,7 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id,
g_value_set_boxed (value, src->jclient);
break;
case PROP_TRANSPORT:
g_value_set_enum (value, src->transport);
g_value_set_flags (value, src->transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

4
ext/jack/gstjackaudiosrc.h
View file

@ -72,11 +72,11 @@ struct _GstJackAudioSrc
GstCaps *caps;
/* properties */
GstJackConnect connect;
GstJackConnect connect;
gchar *server;
jack_client_t *jclient;
gchar *client_name;
GstJackTransport transport;
guint transport;
/* our client */
GstJackAudioClient *client;

18
ext/pulse/pulsemixerctrl.h
View file

@ -92,7 +92,7 @@ void gst_pulsemixer_ctrl_set_record (GstPulseMixerCtrl * mixer,
GstMixerTrack * track, gboolean record);
GstMixerFlags gst_pulsemixer_ctrl_get_mixer_flags (GstPulseMixerCtrl * mixer);
#define GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(Type, interface_as_function) \
#define GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(Type, interface_as_function) \
static const GList* \
interface_as_function ## _list_tracks (GstMixer * mixer) \
{ \
@ -155,19 +155,23 @@ interface_as_function ## _get_mixer_flags (GstMixer * mixer)
g_return_val_if_fail (this != NULL, GST_MIXER_FLAG_NONE); \
g_return_val_if_fail (this->mixer != NULL, GST_MIXER_FLAG_NONE); \
\
return gst_pulsemixer_ctrl_get_mixer_flags (this->mixer); \
return gst_pulsemixer_ctrl_get_mixer_flags (this->mixer); \
} \
static void \
interface_as_function ## _mixer_interface_init (GstMixerInterface * iface) \
static GstMixerType \
interface_as_function ## _get_mixer_type (GstMixer * mixer) \
{ \
return GST_MIXER_HARDWARE; \
} \
static void \
interface_as_function ## _mixer_interface_init (GstMixerInterface * iface) \
{ \
GST_MIXER_TYPE (iface) = GST_MIXER_HARDWARE; \
\
iface->list_tracks = interface_as_function ## _list_tracks; \
iface->set_volume = interface_as_function ## _set_volume; \
iface->get_volume = interface_as_function ## _get_volume; \
iface->set_mute = interface_as_function ## _set_mute; \
iface->set_record = interface_as_function ## _set_record; \
iface->get_mixer_flags = interface_as_function ## _get_mixer_flags; \
iface->get_mixer_flags = interface_as_function ## _get_mixer_flags; \
iface->get_mixer_type = interface_as_function ## _get_mixer_type; \
}
G_END_DECLS

4
ext/pulse/pulseprobe.c
View file

@ -21,6 +21,10 @@
* USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

2
ext/speex/gstspeexenc.c
View file

@ -530,7 +530,7 @@ gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
}
/* we only peeked, let base class handle it */
return FALSE;
return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event);
}
static GstFlowReturn

2
ext/wavpack/Makefile.am
View file

@ -3,7 +3,6 @@ plugin_LTLIBRARIES = libgstwavpack.la
libgstwavpack_la_SOURCES = \
gstwavpack.c \
gstwavpackcommon.c \
gstwavpackparse.c \
gstwavpackdec.c \
gstwavpackenc.c \
gstwavpackstreamreader.c
@ -16,7 +15,6 @@ libgstwavpack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstwavpack_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = \
gstwavpackparse.h \
gstwavpackdec.h \
gstwavpackenc.h \
gstwavpackcommon.h \

4
ext/wavpack/gstwavpack.c
View file

@ -25,7 +25,6 @@
#include <gst/gst-i18n-plugin.h>
#include "gstwavpackparse.h"
#include "gstwavpackdec.h"
#include "gstwavpackenc.h"
@ -44,8 +43,7 @@ plugin_init (GstPlugin * plugin)
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif
return (gst_wavpack_parse_plugin_init (plugin)
&& gst_wavpack_dec_plugin_init (plugin)
return (gst_wavpack_dec_plugin_init (plugin)
&& gst_wavpack_enc_plugin_init (plugin));
}

40
ext/wavpack/gstwavpackcommon.c
View file

@ -29,7 +29,6 @@
#include <string.h>
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
#define GST_CAT_DEFAULT wavpack_debug
@ -147,7 +146,7 @@ static const struct
0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
@ -155,37 +154,25 @@ static const struct
0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */
{
0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */
{
0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */
{
0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */
{
0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */
{
0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */
{
0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */
0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
};
#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
gboolean
gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
gst_wavpack_get_channel_positions (gint num_channels, gint layout,
GstAudioChannelPosition * pos)
{
GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS];
GstStructure *s;
gint num_channels, i, p;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &num_channels))
g_return_val_if_reached (FALSE);
gint i, p;
if (num_channels == 1 && layout == 0x00004) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
gst_audio_set_channel_positions (s, pos);
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
return TRUE;
}
@ -214,7 +201,6 @@ gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
return FALSE;
}
gst_audio_set_channel_positions (s, pos);
return TRUE;
}
@ -242,7 +228,7 @@ gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos,
gint channel_mask = 0;
gint i, j;
if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) {
if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
channel_mask = 0x00000004;
return channel_mask;
}

4
ext/wavpack/gstwavpackcommon.h
View file

@ -24,7 +24,7 @@
#define __GST_WAVPACK_COMMON_H__
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <gst/audio/audio.h>
#include <wavpack/wavpack.h>
typedef struct
@ -67,7 +67,7 @@ gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf);
gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta,
guint8 * header_data, guint8 ** p_data);
gint gst_wavpack_get_default_channel_mask (gint nchannels);
gboolean gst_wavpack_set_channel_layout (GstCaps * caps, gint layout);
gboolean gst_wavpack_get_channel_positions (gint nchannels, gint layout, GstAudioChannelPosition *pos);
GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels);
gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels);
gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping);

455
ext/wavpack/gstwavpackdec.c
View file

@ -44,7 +44,6 @@
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <math.h>
#include <string.h>
@ -55,8 +54,6 @@
#include "gstwavpackstreamreader.h"
#define WAVPACK_DEC_MAX_ERRORS 16
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
@ -64,7 +61,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
@ -72,28 +69,41 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32 ], "
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S8, "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
"rate = (int) [ 6000, 192000 ]; "
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]; "
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S32) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec);
static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
GstCaps * caps);
static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void gst_wavpack_dec_finalize (GObject * object);
static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
GstStateChange transition);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
#define gst_wavpack_dec_parent_class parent_class
G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER);
static void
gst_wavpack_dec_base_init (gpointer klass)
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) (klass);
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
@ -104,17 +114,13 @@ gst_wavpack_dec_base_init (gpointer klass)
"Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
gobject_class->finalize = gst_wavpack_dec_finalize;
base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
}
static void
@ -123,33 +129,15 @@ gst_wavpack_dec_reset (GstWavpackDec * dec)
dec->wv_id.buffer = NULL;
dec->wv_id.position = dec->wv_id.length = 0;
dec->error_count = 0;
dec->channels = 0;
dec->channel_mask = 0;
dec->sample_rate = 0;
dec->depth = 0;
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
dec->next_block_index = 0;
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
gst_pad_set_setcaps_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (dec->srcpad);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
@ -168,54 +156,85 @@ gst_wavpack_dec_finalize (GObject * object)
}
static gboolean
gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
gst_wavpack_dec_start (GstAudioDecoder * dec)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (dec, "start");
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
GstCaps *caps;
GstAudioChannelPosition *pos;
/* never mind a few errors */
gst_audio_decoder_set_max_errors (dec, 16);
/* don't bother us with flushing */
gst_audio_decoder_set_drainable (dec, FALSE);
/* aim for some perfect timestamping */
gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
return TRUE;
}
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
pos = gst_audio_get_channel_positions (structure);
if (pos != NULL && dec->channels > 2) {
GstStructure *new_str = gst_caps_get_structure (caps, 0);
static gboolean
gst_wavpack_dec_stop (GstAudioDecoder * dec)
{
GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
gst_audio_set_channel_positions (new_str, pos);
dec->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
}
GST_DEBUG_OBJECT (dec, "stop");
if (pos != NULL)
g_free (pos);
}
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
if (wpdec->context) {
WavpackCloseFile (wpdec->context);
wpdec->context = NULL;
}
gst_object_unref (dec);
gst_wavpack_dec_reset (wpdec);
return TRUE;
}
static void
gst_wavpack_dec_negotiate (GstWavpackDec * dec)
{
GstAudioInfo info;
GstAudioFormat fmt;
GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
/* arrange for 1, 2 or 4-byte width == depth output */
dec->width = dec->depth;
switch (dec->depth) {
case 8:
fmt = GST_AUDIO_FORMAT_S8;
break;
case 16:
fmt = _GST_AUDIO_FORMAT_NE (S16);
break;
case 24:
case 32:
fmt = _GST_AUDIO_FORMAT_NE (S32);
dec->width = 32;
break;
default:
g_assert_not_reached ();
break;
}
g_assert (dec->channel_mask != 0);
if (!gst_wavpack_get_channel_positions (dec->channels,
dec->channel_mask, pos))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
gst_audio_channel_positions_to_valid_order (info.position, info.channels);
gst_audio_get_channel_reorder_map (info.channels,
info.position, pos, dec->channel_reorder_map);
/* should always succeed */
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
}
static gboolean
gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
/* pretty much nothing to do here,
* we'll parse it all from the stream and setup then */
return TRUE;
}
@ -227,28 +246,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
list = gst_tag_list_new ();
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
/* try to estimate the average bitrate */
if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
size > 0 && duration > 0) {
if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_bytes, &size) &&
gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate;
list = gst_tag_list_new_empty ();
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
}
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_tag (GST_OBJECT (dec), list));
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_tag (GST_OBJECT (dec), list));
}
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf = NULL;
@ -256,24 +273,32 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
gint width, depth, i, j, max;
gint32 *dec_data = NULL;
guint8 *out_data;
GstMapInfo map, omap;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
dec = GST_WAVPACK_DEC (bdec);
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
gst_buffer_map (buf, &map, GST_MAP_READ);
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
if (map.size < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
if (!gst_wavpack_read_header (&wph, map.data))
goto invalid_header;
if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
if (map.size < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf);
dec->wv_id.length = GST_BUFFER_SIZE (buf);
dec->wv_id.buffer = map.data;
dec->wv_id.length = map.size;
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
@ -285,46 +310,32 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
/* expect this to work */
if (!dec->context) {
GST_WARNING ("Couldn't decode buffer: %s", error_msg);
dec->error_count++;
if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
goto out; /* just return OK for now */
} else {
goto decode_error;
}
GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
goto context_failed;
}
}
g_assert (dec->context != NULL);
dec->error_count = 0;
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBitsPerSample (dec->context)) ||
(dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
#ifdef WAVPACK_OLD_API
(dec->channel_mask != dec->context->config.channel_mask);
#else
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) ||
format_changed) {
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
#ifdef WAVPACK_OLD_API
channel_mask = dec->context->config.channel_mask;
@ -336,17 +347,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
dec->channel_mask = channel_mask;
/* Only set the channel layout for more than two channels
* otherwise things break unfortunately */
if (channel_mask != 0 && dec->channels > 2)
if (!gst_wavpack_set_channel_layout (caps, channel_mask))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
gst_wavpack_dec_negotiate (dec);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
@ -354,58 +355,97 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
}
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
/* If we got a DISCONT buffer forward the flag. Nothing else
* has to be done as libwavpack doesn't store state between
* Wavpack blocks */
if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->next_block_index = wph.block_index + wph.block_samples;
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
dec->sample_rate, 4 * dec->channels))) {
GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (dec->srcpad, outbuf);
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
}
} else {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]);
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
buf = NULL;
g_free (dec_data);
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
out:
if (buf)
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
gst_buffer_unref (buf);
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto out;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto out;
}
context_failed:
{
GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
("error creating Wavpack context"), ret);
goto out;
}
decode_error:
{
@ -420,95 +460,22 @@ decode_error:
} else {
reason = "couldn't create decoder context";
}
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream: %s", reason));
if (outbuf)
gst_buffer_unref (outbuf);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("decoding error: %s", reason), ret);
g_free (dec_data);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
goto out;
}
}
static gboolean
gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat fmt;
gboolean is_update;
gint64 start, end, base;
gdouble rate;
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
&end, &base);
if (fmt == GST_FORMAT_TIME) {
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
GST_TIME_ARGS (end));
gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt,
start, end, base);
} else {
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
}
break;
}
default:
break;
}
gst_object_unref (dec);
return gst_pad_event_default (pad, event);
}
static GstStateChangeReturn
gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackDec *dec = GST_WAVPACK_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (dec->context) {
WavpackCloseFile (dec->context);
dec->context = NULL;
}
gst_wavpack_dec_reset (dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
"Wavpack decoder");
return TRUE;
}

14
ext/wavpack/gstwavpackdec.h
View file

@ -24,6 +24,7 @@
#define __GST_WAVPACK_DEC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
#include <wavpack/wavpack.h>
@ -45,31 +46,28 @@ typedef struct _GstWavpackDecClass GstWavpackDecClass;
struct _GstWavpackDec
{
GstElement element;
GstAudioDecoder element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
WavpackContext *context;
WavpackStreamReader *stream_reader;
read_id wv_id;
GstSegment segment; /* used for clipping, TIME format */
guint32 next_block_index;
gint sample_rate;
gint depth;
gint width;
gint channels;
gint channel_mask;
gint error_count;
gint channel_reorder_map[64];
};
struct _GstWavpackDecClass
{
GstElementClass parent;
GstAudioDecoderClass parent;
};
GType gst_wavpack_dec_get_type (void);

560
ext/wavpack/gstwavpackenc.c
View file

@ -31,7 +31,7 @@
* |[
* gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
* as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
* as the Wavpack encoder only accepts input with 32 bit width.
* |[
* gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
@ -55,12 +55,18 @@
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
@ -84,20 +90,18 @@ GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32], "
"endianness = (int) BYTE_ORDER, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S32) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
@ -196,26 +200,15 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
return qtype;
}
static void
_do_init (GType object_type)
{
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
g_type_add_interface_static (object_type, GST_TYPE_PRESET,
&preset_interface_info);
}
GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement,
GST_TYPE_ELEMENT, _do_init);
#define gst_wavpack_enc_parent_class parent_class
G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
static void
gst_wavpack_enc_base_init (gpointer klass)
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) (klass);
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
/* add pad templates */
gst_element_class_add_pad_template (element_class,
@ -230,25 +223,17 @@ gst_wavpack_enc_base_init (gpointer klass)
"Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
/* set state change handler */
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
/* set property handlers */
gobject_class->set_property = gst_wavpack_enc_set_property;
gobject_class->get_property = gst_wavpack_enc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
base_class->event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
@ -306,6 +291,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
g_checksum_free (enc->md5_context);
enc->md5_context = NULL;
}
if (enc->pending_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = NULL;
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
@ -330,20 +318,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
}
static void
gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
gst_wavpack_enc_init (GstWavpackEnc * enc)
{
enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
gst_pad_set_event_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
/* setup src pad */
enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
/* initialize object attributes */
enc->wp_config = NULL;
@ -367,37 +344,53 @@ gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
enc->md5 = FALSE;
enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
/* require perfect ts */
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
}
static gboolean
gst_wavpack_enc_start (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (enc, "start");
return TRUE;
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
gst_wavpack_enc_stop (GstAudioEncoder * enc)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gst_wavpack_enc_reset (wpenc);
return TRUE;
}
static gboolean
gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos;
GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
GstCaps *caps;
guint64 mask = 0;
if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
!gst_structure_get_int (structure, "rate", &enc->samplerate) ||
!gst_structure_get_int (structure, "depth", &enc->depth)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("got invalid caps: %" GST_PTR_FORMAT, caps));
gst_object_unref (enc);
return FALSE;
}
/* we may be configured again, but that change should have cleanup context */
g_assert (enc->wp_context == NULL);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->depth = GST_AUDIO_INFO_DEPTH (info);
enc->samplerate = GST_AUDIO_INFO_RATE (info);
pos = info->position;
g_assert (pos);
pos = gst_audio_get_channel_positions (structure);
/* If one channel is NONE they'll be all undefined */
if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
g_free (pos);
pos = NULL;
}
if (pos == NULL) {
GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
("input has no valid channel layout"));
gst_object_unref (enc);
return FALSE;
goto invalid_channels;
}
enc->channel_mask =
@ -405,29 +398,41 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
enc->need_channel_remap =
gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping);
g_free (pos);
/* wavpack caps hold gst mask, not wavpack mask */
gst_audio_channel_positions_to_mask (opos, enc->channels, &mask);
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
"width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
"depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
GST_WARNING_OBJECT (enc, "setting channel layout failed");
if (mask)
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
if (!gst_pad_set_caps (enc->srcpad, caps)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("setting caps failed: %" GST_PTR_FORMAT, caps));
gst_caps_unref (caps);
gst_object_unref (enc);
return FALSE;
}
gst_pad_use_fixed_caps (enc->srcpad);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
goto setting_src_caps_failed;
gst_caps_unref (caps);
gst_object_unref (enc);
/* no special feedback to base class; should provide all available samples */
return TRUE;
/* ERRORS */
setting_src_caps_failed:
{
GST_DEBUG_OBJECT (enc,
"Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
return FALSE;
}
invalid_channels:
{
GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
return FALSE;
}
}
static void
@ -549,22 +554,15 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
gint samples = 0;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
srcpad_last_return;
(wid->correction) ? &enc->
wvcsrcpad_last_return : &enc->srcpad_last_return;
*flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
count, GST_PAD_CAPS (pad), &buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
buffer = gst_buffer_new_and_alloc (count);
gst_buffer_fill (buffer, 0, data, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
@ -599,44 +597,50 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
enc->pending_buffer = NULL;
enc->pending_offset = 0;
/* if it's the first wavpack block, send a NEW_SEGMENT event */
if (wph.block_index == 0) {
gst_pad_push_event (pad,
gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
/* only send segment on correction pad,
* regular pad is handled normally by baseclass */
if (wid->correction && enc->pending_segment) {
gst_pad_push_event (pad, enc->pending_segment);
enc->pending_segment = NULL;
}
if (wph.block_index == 0) {
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
enc->first_block =
g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
enc->first_block_size = GST_BUFFER_SIZE (buffer);
GstMapInfo map;
gst_buffer_map (buffer, &map, GST_MAP_READ);
enc->first_block = g_memdup (map.data, map.size);
enc->first_block_size = map.size;
gst_buffer_unmap (buffer, &map);
}
}
}
samples = wph.block_samples;
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
enc->samplerate);
/* decorate buffer */
/* NOTE: this will get overwritten by baseclass, but stay for those
* that are pushed directly
* FIXME: add setting to baseclass to avoid overwriting it ?? */
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_pad_push (pad, buffer);
if (wid->correction || wid->passthrough) {
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
gst_buffer_get_size (buffer));
*flow = gst_pad_push (pad, buffer);
} else {
GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
gst_buffer_get_size (buffer));
*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
samples);
}
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
@ -666,18 +670,26 @@ gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
uint32_t sample_count;
GstFlowReturn ret;
GstMapInfo map;
/* base class ensures configuration */
g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
GST_DEBUG ("got %u raw samples", sample_count);
if (G_UNLIKELY (!buf))
return gst_wavpack_enc_drain (enc);
sample_count = gst_buffer_get_size (buf) / 4;
GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
@ -685,13 +697,8 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
if (!enc->wp_context) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
if (!enc->wp_context)
goto context_failed;
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
@ -701,99 +708,36 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
goto config_failed;
}
GST_DEBUG ("setup of encoding context successfull");
GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
}
/* Save the timestamp of the first buffer. This will be later
* used as offset for all following buffers */
if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
enc->next_ts = GST_BUFFER_TIMESTAMP (buf);
} else {
enc->timestamp_offset = 0;
enc->next_ts = 0;
}
}
/* Check if we have a continous stream, if not drop some samples or the buffer or
* insert some silence samples */
if (enc->next_ts != GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
guint64 diff_bytes;
GST_WARNING_OBJECT (enc, "Buffer is older than previous "
"timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
"), cannot handle. Clipping buffer.",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (enc->next_ts));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2;
if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
gst_buffer_unref (buf);
return GST_FLOW_OK;
}
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_DATA (buf) += diff_bytes;
GST_BUFFER_SIZE (buf) -= diff_bytes;
GST_BUFFER_TIMESTAMP (buf) += diff;
if (GST_BUFFER_DURATION_IS_VALID (buf))
GST_BUFFER_DURATION (buf) -= diff;
}
/* Allow a diff of at most 5 ms */
if (enc->next_ts != GST_CLOCK_TIME_NONE
&& GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) {
GST_WARNING_OBJECT (enc,
"Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND);
WavpackFlushSamples (enc->wp_context);
enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts);
}
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
&& GST_BUFFER_DURATION_IS_VALID (buf))
enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
else
enc->next_ts = GST_CLOCK_TIME_NONE;
if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
sample_count);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
gst_buffer_unmap (buf, &map);
}
gst_buffer_map (buf, &map, GST_MAP_READ);
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
g_checksum_update (enc->md5_context, map.data, map.size);
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
GST_DEBUG_OBJECT (enc, "encoding samples successful");
gst_buffer_unmap (buf, &map);
ret = GST_FLOW_OK;
} else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
(enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
ret = GST_FLOW_RESEND;
} else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
gst_buffer_unmap (buf, &map);
if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
@ -803,22 +747,41 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
(enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
ret = GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
goto encoding_failed;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
exit:
return ret;
/* ERRORS */
encoding_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
goto exit;
}
config_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
ret = GST_FLOW_ERROR;
goto exit;
}
context_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
ret = GST_FLOW_ERROR;
goto exit;
}
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{
GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
0, GST_BUFFER_OFFSET_NONE, 0);
GstSegment segment;
gboolean ret;
g_return_if_fail (enc);
@ -828,7 +791,9 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
ret = gst_pad_push_event (enc->srcpad, event);
gst_segment_init (&segment, GST_FORMAT_BYTES);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (&segment));
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
@ -836,111 +801,84 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
enc->wv_id.passthrough = FALSE;
g_free (enc->first_block);
enc->first_block = NULL;
} else {
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
"Seeking to first block failed!");
}
}
static gboolean
gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
static GstFlowReturn
gst_wavpack_enc_drain (GstWavpackEnc * enc)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
gboolean ret = TRUE;
if (!enc->wp_context)
return GST_FLOW_OK;
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
GST_DEBUG_OBJECT (enc, "draining");
/* Encode all remaining samples and flush them to the src pads */
WavpackFlushSamples (enc->wp_context);
/* Drop all remaining data, this is no complete block otherwise
* it would've been pushed already */
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = NULL;
enc->pending_offset = 0;
}
/* write the MD5 sum if we have to write one */
if ((enc->md5) && (enc->md5_context)) {
guint8 md5_digest[16];
gsize digest_len = sizeof (md5_digest);
g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
if (digest_len == sizeof (md5_digest)) {
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
WavpackFlushSamples (enc->wp_context);
} else
GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
}
/* Try to rewrite the first frame with the correct sample number */
if (enc->first_block)
gst_wavpack_enc_rewrite_first_block (enc);
/* close the context if not already happened */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
return GST_FLOW_OK;
}
static gboolean
gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* Encode all remaining samples and flush them to the src pads */
WavpackFlushSamples (enc->wp_context);
/* Drop all remaining data, this is no complete block otherwise
* it would've been pushed already */
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = NULL;
enc->pending_offset = 0;
}
/* write the MD5 sum if we have to write one */
if ((enc->md5) && (enc->md5_context)) {
guint8 md5_digest[16];
gsize digest_len = sizeof (md5_digest);
g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
if (digest_len == sizeof (md5_digest))
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
else
GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
}
/* Try to rewrite the first frame with the correct sample number */
if (enc->first_block)
gst_wavpack_enc_rewrite_first_block (enc);
/* close the context if not already happened */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
case GST_EVENT_NEWSEGMENT:
case GST_EVENT_SEGMENT:
if (enc->wp_context) {
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
"already started");
}
/* drop NEWSEGMENT events, we create our own when pushing
* the first buffer to the pads */
gst_event_unref (event);
ret = TRUE;
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (enc);
return ret;
}
static GstStateChangeReturn
gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
* as they're only set to something else in WavpackPackSamples() or more
* specific gst_wavpack_enc_push_block() and nothing happened there yet */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavpack_enc_reset (enc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
/* peek and hold NEWSEGMENT events for sending on correction pad */
if (enc->pending_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = gst_event_ref (event);
break;
default:
break;
}
return ret;
/* baseclass handles rest */
return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event);
}
static void
@ -1044,7 +982,7 @@ gst_wavpack_enc_plugin_init (GstPlugin * plugin)
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
"Wavpack encoder");
return TRUE;

7
ext/wavpack/gstwavpackenc.h
View file

@ -23,6 +23,7 @@
#define __GST_WAVPACK_ENC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <wavpack/wavpack.h>
@ -50,10 +51,9 @@ typedef struct
struct _GstWavpackEnc
{
GstElement element;
GstAudioEncoder element;
/*< private > */
GstPad *sinkpad, *srcpad;
GstPad *wvcsrcpad;
GstFlowReturn srcpad_last_return;
@ -86,6 +86,7 @@ struct _GstWavpackEnc
GstBuffer *pending_buffer;
gint32 pending_offset;
GstEvent *pending_segment;
GstClockTime timestamp_offset;
GstClockTime next_ts;
@ -93,7 +94,7 @@ struct _GstWavpackEnc
struct _GstWavpackEncClass
{
GstElementClass parent;
GstAudioEncoderClass parent;
};
GType gst_wavpack_enc_get_type (void);

1341
ext/wavpack/gstwavpackparse.c
View file

File diff suppressed because it is too large Load diff

97
ext/wavpack/gstwavpackparse.h
View file

@ -1,97 +0,0 @@
/* GStreamer wavpack plugin
* (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
*
* gstwavpackparse.h: wavpack file parser
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_WAVPACK_PARSE_H__
#define __GST_WAVPACK_PARSE_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_PARSE \
(gst_wavpack_parse_get_type())
#define GST_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_PARSE,GstWavpackParse))
#define GST_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_PARSE,GstWavpackParseClass))
#define GST_IS_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_PARSE))
#define GST_IS_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_PARSE))
typedef struct _GstWavpackParse GstWavpackParse;
typedef struct _GstWavpackParseClass GstWavpackParseClass;
typedef struct _GstWavpackParseIndexEntry GstWavpackParseIndexEntry;
struct _GstWavpackParseIndexEntry
{
gint64 byte_offset; /* byte offset of this chunk */
gint64 sample_offset; /* first sample in this chunk */
gint64 sample_offset_end; /* first sample in next chunk */
};
struct _GstWavpackParse
{
GstElement element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
guint samplerate;
guint channels;
gint64 total_samples;
gboolean need_newsegment;
gboolean discont;
gint64 current_offset; /* byte offset on sink pad */
gint64 upstream_length; /* length of file in bytes */
GstSegment segment; /* the currently configured segment, in
* samples/audio frames (DEFAULT format) */
GstBuffer *pending_buffer;
gint32 pending_offset;
guint32 next_block_index;
GstAdapter *adapter; /* when operating chain-based, otherwise NULL */
/* List of GstWavpackParseIndexEntry structs, mapping known
* sample offsets to byte offsets. Is kept increasing without
* gaps (ie. append only and consecutive entries must always
* map to consecutive chunks in the file). */
GSList *entries;
/* Queued events (e.g. tag events we receive before we create the src pad) */
GList *queued_events; /* STREAM_LOCK */
};
struct _GstWavpackParseClass
{
GstElementClass parent;
};
GType gst_wavpack_parse_get_type (void);
gboolean gst_wavpack_parse_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_WAVPACK_PARSE_H__ */

2
gst/alpha/gstalpha.c
View file

@ -393,7 +393,7 @@ gst_alpha_set_property (GObject * object, guint prop_id,
}
if (reconfigure)
gst_base_transform_reconfigure (GST_BASE_TRANSFORM_CAST (alpha));
gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM_CAST (alpha));
GST_ALPHA_UNLOCK (alpha);
}

6
gst/audiofx/audiofirfilter.c
View file

@ -45,6 +45,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
@ -130,7 +134,7 @@ gst_audio_fir_filter_class_init (GstAudioFIRFilterClass * klass)
gst_audio_fir_filter_signals[SIGNAL_RATE_CHANGED] =
g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioFIRFilterClass, rate_changed),
NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_INT);
gst_element_class_set_details_simple (gstelement_class,
"Audio FIR filter", "Filter/Effect/Audio",

6
gst/audiofx/audioiirfilter.c
View file

@ -41,6 +41,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
@ -128,7 +132,7 @@ gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass)
gst_audio_iir_filter_signals[SIGNAL_RATE_CHANGED] =
g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioIIRFilterClass, rate_changed),
NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_INT);
gst_element_class_set_details_simple (gstelement_class,
"Audio IIR filter", "Filter/Effect/Audio",

4
gst/audioparsers/Makefile.am
View file

@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la
libgstaudioparsers_la_SOURCES = \
gstaacparse.c gstamrparse.c gstac3parse.c \
gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \
plugin.c
gstwavpackparse.c plugin.c
libgstaudioparsers_la_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
@ -15,4 +15,4 @@ libgstaudioparsers_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioparsers_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \
gstdcaparse.h gstflacparse.h gstmpegaudioparse.h
gstdcaparse.h gstflacparse.h gstmpegaudioparse.h gstwavpackparse.h

169
gst/audioparsers/gstmpegaudioparse.c
View file

@ -196,6 +196,7 @@ gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
mp3parse->sent_codec_tag = FALSE;
mp3parse->last_posted_crc = CRC_UNKNOWN;
mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
mp3parse->freerate = 0;
mp3parse->hdr_bitrate = 0;
@ -302,14 +303,16 @@ mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
/* The caller has ensured we have a valid header, so bitrate can't be
zero here. */
g_assert (bitrate != 0);
if (!bitrate) {
GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
bitrate = mp3parse->freerate;
}
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
/* force 0 length if 0 bitrate */
padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
@ -414,8 +417,7 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
(guint) next_header & HDRMASK, bpf);
*valid = FALSE;
goto cleanup;
} else if ((((next_header >> 12) & 0xf) == 0) ||
(((next_header >> 12) & 0xf) == 0xf)) {
} else if (((next_header >> 12) & 0xf) == 0xf) {
/* The essential parts were the same, but the bitrate held an
invalid value - also reject */
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
@ -426,6 +428,13 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
NULL, NULL, NULL, NULL, NULL, NULL, NULL);
/* if no bitrate, and no freeform rate known, then fail */
if (G_UNLIKELY (!bpf)) {
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
*valid = FALSE;
return TRUE;
}
offset += bpf;
frames_found++;
}
@ -459,11 +468,6 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
return FALSE;
}
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
"Free format files are not supported yet", (head >> 12) & 0xf);
return FALSE;
}
if (((head >> 12) & 0xf) == 0xf) {
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
return FALSE;
@ -484,6 +488,115 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
return TRUE;
}
/* Determines possible freeform frame rate/size by looking for next
* header with valid bitrate (0 or otherwise valid) (and sufficiently
* matching current header).
*
* Returns TRUE if we've found such one, and *rate then contains rate
* (or *rate contains 0 if decided no freeframe size could be determined).
* If not enough data, returns FALSE.
*/
static gboolean
gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map,
guint32 header, gboolean at_eos, gint * _rate)
{
guint32 next_header;
const guint8 *data;
guint available;
int offset = 4;
gulong samplerate, rate, layer, padding;
gboolean valid;
gint lsf, mpg25;
available = map->size;
data = map->data;
*_rate = 0;
/* pick apart header again partially */
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
layer = 4 - ((header >> 17) & 0x3);
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
for (; offset < available; ++offset) {
/* Check if we have enough data for all these frames, plus the next
frame header. */
if (available < offset + 4) {
if (at_eos) {
/* Running out of data; failed to determine size */
return TRUE;
} else {
return FALSE;
}
}
valid = FALSE;
next_header = GST_READ_UINT32_BE (data + offset);
if ((next_header & 0xFFE00000) != 0xFFE00000)
goto next;
GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
offset, (unsigned int) header, (unsigned int) next_header);
if ((next_header & HDRMASK) != (header & HDRMASK)) {
/* If any of the unmasked bits don't match, then it's not valid */
GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
"(header=%08X (%08X), header2=%08X (%08X))",
(guint) header, (guint) header & HDRMASK, (guint) next_header,
(guint) next_header & HDRMASK);
goto next;
} else if (((next_header >> 12) & 0xf) == 0xf) {
/* The essential parts were the same, but the bitrate held an
invalid value - also reject */
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
goto next;
}
valid = TRUE;
next:
/* almost accept as free frame */
if (layer == 1) {
rate = samplerate * (offset - 4 * padding + 4) / 48000;
} else {
rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
}
if (valid) {
GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000);
if (rate < 8 || (layer == 3 && rate > 640)) {
GST_DEBUG_OBJECT (mp3parse, "rate invalid");
if (rate < 8) {
/* maybe some hope */
continue;
} else {
GST_DEBUG_OBJECT (mp3parse, "aborting");
/* give up */
break;
}
}
*_rate = rate * 1000;
break;
} else {
/* avoid indefinite searching */
if (rate > 1000) {
GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
break;
}
}
}
return TRUE;
}
static GstFlowReturn
gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
@ -532,9 +645,14 @@ gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GST_LOG_OBJECT (parse, "got frame");
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
if (G_UNLIKELY (lost_sync))
mp3parse->freerate = 0;
bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
g_assert (bpf != 0);
if (channels != mp3parse->channels || rate != mp3parse->rate ||
layer != mp3parse->layer || version != mp3parse->version)
@ -542,8 +660,31 @@ gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
else
caps_change = FALSE;
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
/* maybe free format */
if (bpf == 0) {
GST_LOG_OBJECT (mp3parse, "possibly free format");
if (lost_sync || mp3parse->freerate == 0) {
GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining,
&valid)) {
/* not enough data */
gst_base_parse_set_min_frame_size (parse, valid);
*skipsize = 0;
return FALSE;
} else {
GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
mp3parse->freerate = valid;
}
}
/* try again */
bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
if (!bpf) {
/* did not come up with valid freeform length, reject after all */
*skipsize = 1;
return FALSE;
}
}
if (!draining && (lost_sync || caps_change)) {
if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,

2
gst/audioparsers/gstmpegaudioparse.h
View file

@ -60,6 +60,8 @@ struct _GstMpegAudioParse {
/* samples per frame */
gint spf;
gint freerate;
gboolean sent_codec_tag;
guint last_posted_bitrate;
gint last_posted_crc, last_crc;

640
gst/audioparsers/gstwavpackparse.c Normal file
View file

@ -0,0 +1,640 @@
/* GStreamer Wavpack parser
* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2012 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-wavpackparse
* @short_description: Wavpack parser
* @see_also: #GstAmrParse, #GstAACParse
*
* This is an Wavpack parser.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstwavpackparse.h"
#include <gst/base/gstbytereader.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug);
#define GST_CAT_DEFAULT wavpack_parse_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; "
"audio/x-wavpack-correction, " "framed = (boolean) TRUE")
);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack"));
static void gst_wavpack_parse_finalize (GObject * object);
static gboolean gst_wavpack_parse_start (GstBaseParse * parse);
static gboolean gst_wavpack_parse_stop (GstBaseParse * parse);
static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse,
GstCaps * filter);
#define gst_wavpack_parse_parent_class parent_class
G_DEFINE_TYPE (GstWavpackParse, gst_wavpack_parse, GST_TYPE_BASE_PARSE);
static void
gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
{
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (wavpack_parse_debug, "wavpackparse", 0,
"Wavpack audio stream parser");
object_class->finalize = gst_wavpack_parse_finalize;
parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop);
parse_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_handle_frame);
parse_class->get_sink_caps =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class,
"Wavpack audio stream parser", "Codec/Parser/Audio",
"Wavpack parser", "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
}
static void
gst_wavpack_parse_reset (GstWavpackParse * wvparse)
{
wvparse->channels = -1;
wvparse->channel_mask = 0;
wvparse->sample_rate = -1;
wvparse->width = -1;
wvparse->total_samples = 0;
}
static void
gst_wavpack_parse_init (GstWavpackParse * wvparse)
{
gst_wavpack_parse_reset (wvparse);
}
static void
gst_wavpack_parse_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_wavpack_parse_start (GstBaseParse * parse)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GST_DEBUG_OBJECT (parse, "starting");
gst_wavpack_parse_reset (wvparse);
/* need header at least */
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse),
sizeof (WavpackHeader));
/* inform baseclass we can come up with ts, based on counters in packets */
gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE);
gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE);
return TRUE;
}
static gboolean
gst_wavpack_parse_stop (GstBaseParse * parse)
{
GST_DEBUG_OBJECT (parse, "stopping");
return TRUE;
}
static gint
gst_wavpack_get_default_channel_mask (gint nchannels)
{
gint channel_mask = 0;
/* Set the default channel mask for the given number of channels.
* It's the same as for WAVE_FORMAT_EXTENDED:
* http://www.microsoft.com/whdc/device/audio/multichaud.mspx
*/
switch (nchannels) {
case 11:
channel_mask |= 0x00400;
channel_mask |= 0x00200;
case 9:
channel_mask |= 0x00100;
case 8:
channel_mask |= 0x00080;
channel_mask |= 0x00040;
case 6:
channel_mask |= 0x00020;
channel_mask |= 0x00010;
case 4:
channel_mask |= 0x00008;
case 3:
channel_mask |= 0x00004;
case 2:
channel_mask |= 0x00002;
channel_mask |= 0x00001;
break;
case 1:
/* For mono use front center */
channel_mask |= 0x00004;
break;
}
return channel_mask;
}
static const struct
{
const guint32 ms_mask;
const GstAudioChannelPosition gst_pos;
} layout_mapping[] = {
{
0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
};
#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
static gboolean
gst_wavpack_get_channel_positions (gint num_channels, gint layout,
GstAudioChannelPosition * pos)
{
gint i, p;
if (num_channels == 1 && layout == 0x00004) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
return TRUE;
}
p = 0;
for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) {
if ((layout & layout_mapping[i].ms_mask) != 0) {
if (p >= num_channels) {
GST_WARNING ("More bits set in the channel layout map than there "
"are channels! Broken file");
return FALSE;
}
if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel "
"layout map - ignoring those channels", layout_mapping[i].ms_mask);
/* what to do? just ignore it and let downstream deal with a channel
* layout that has INVALID positions in it for now ... */
}
pos[p] = layout_mapping[i].gst_pos;
++p;
}
}
if (p != num_channels) {
GST_WARNING ("Only %d bits set in the channel layout map, but there are "
"supposed to be %d channels! Broken file", p, num_channels);
return FALSE;
}
return TRUE;
}
static const guint32 sample_rates[] = {
6000, 8000, 9600, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000
};
#define CHECK(call) { \
if (!call) \
goto read_failed; \
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * wph, WavpackInfo * wpi)
{
GstByteReader br;
gint i;
GstMapInfo map;
g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE);
g_return_val_if_fail (gst_buffer_get_size (buf) >=
skip + sizeof (WavpackHeader), FALSE);
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&br, map.data + skip, wph->ckSize + 8);
/* skip past header */
gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader));
/* get some basics from header */
i = (wph->flags >> 23) & 0xF;
if (!wpi->rate)
wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100;
wpi->width = ((wph->flags & 0x3) + 1) * 8;
if (!wpi->channels)
wpi->channels = (wph->flags & 0x4) ? 1 : 2;
if (!wpi->channel_mask)
wpi->channel_mask = 5 - wpi->channels;
/* need to dig metadata blocks for some more */
while (gst_byte_reader_get_remaining (&br)) {
gint size = 0;
guint16 size2 = 0;
guint8 c, id;
const guint8 *data;
GstByteReader mbr;
CHECK (gst_byte_reader_get_uint8 (&br, &id));
CHECK (gst_byte_reader_get_uint8 (&br, &c));
if (id & ID_LARGE)
CHECK (gst_byte_reader_get_uint16_le (&br, &size2));
size = size2;
size <<= 8;
size += c;
size <<= 1;
if (id & ID_ODD_SIZE)
size--;
CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data));
gst_byte_reader_init (&mbr, data, size);
switch (id) {
case ID_WVC_BITSTREAM:
GST_LOG_OBJECT (parse, "correction bitstream");
wpi->correction = TRUE;
break;
case ID_WV_BITSTREAM:
case ID_WVX_BITSTREAM:
break;
case ID_SAMPLE_RATE:
if (size == 3) {
CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate));
GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata");
}
break;
case ID_CHANNEL_INFO:
{
guint16 channels;
guint32 mask = 0;
if (size == 6) {
CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels));
channels = channels & 0xFFF;
CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask));
} else if (size) {
CHECK (gst_byte_reader_get_uint8 (&mbr, &c));
channels = c;
while (gst_byte_reader_get_uint8 (&mbr, &c))
mask |= (((guint32) c) << 8);
} else {
GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata");
break;
}
wpi->channels = channels;
wpi->channel_mask = mask;
break;
}
default:
GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id);
break;
}
}
gst_buffer_unmap (buf, &map);
return TRUE;
/* ERRORS */
read_failed:
{
gst_buffer_unmap (buf, &map);
GST_DEBUG_OBJECT (parse, "short read while parsing metadata");
/* let's look the other way anyway */
return TRUE;
}
}
/* caller ensures properly sync'ed with enough data */
static gboolean
gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf,
gint skip, WavpackHeader * _wph)
{
GstByteReader br;
WavpackHeader wph;
GstMapInfo map;
g_return_val_if_fail (gst_buffer_get_size (buf) >=
skip + sizeof (WavpackHeader), FALSE);
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&br, map.data, map.size);
/* marker */
gst_byte_reader_skip_unchecked (&br, skip + 4);
/* read */
gst_byte_reader_get_uint32_le (&br, &wph.ckSize);
gst_byte_reader_get_uint16_le (&br, &wph.version);
gst_byte_reader_get_uint8 (&br, &wph.track_no);
gst_byte_reader_get_uint8 (&br, &wph.index_no);
gst_byte_reader_get_uint32_le (&br, &wph.total_samples);
gst_byte_reader_get_uint32_le (&br, &wph.block_index);
gst_byte_reader_get_uint32_le (&br, &wph.block_samples);
gst_byte_reader_get_uint32_le (&br, &wph.flags);
gst_byte_reader_get_uint32_le (&br, &wph.crc);
/* dump */
GST_LOG_OBJECT (parse, "size %d", wph.ckSize);
GST_LOG_OBJECT (parse, "version 0x%x", wph.version);
GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples);
GST_LOG_OBJECT (parse, "block index %d", wph.block_index);
GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples);
GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags);
GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags);
if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) {
GST_DEBUG_OBJECT (parse, "determined duration of %u samples",
wph.total_samples);
parse->total_samples = wph.total_samples;
}
if (_wph)
*_wph = wph;
gst_buffer_unmap (buf, &map);
return TRUE;
}
static GstFlowReturn
gst_wavpack_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
GstBuffer *buf = frame->buffer;
GstByteReader reader;
gint off;
guint rate, chans, width, mask;
gboolean lost_sync, draining, final;
guint frmsize = 0;
WavpackHeader wph;
WavpackInfo wpi = { 0, };
GstMapInfo map;
if (G_UNLIKELY (gst_buffer_get_size (buf) < sizeof (WavpackHeader)))
return FALSE;
gst_buffer_map (buf, &map, GST_MAP_READ);
gst_byte_reader_init (&reader, map.data, map.size);
/* scan for 'wvpk' marker */
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b,
0, map.size);
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
*skipsize = map.size - 3;
goto skip;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
goto skip;
}
/* make sure the values in the frame header look sane */
gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph);
frmsize = wph.ckSize + 8;
/* need the entire frame for parsing */
if (gst_byte_reader_get_remaining (&reader) < frmsize)
goto more;
/* got a frame, now we can dig for some more metadata */
GST_LOG_OBJECT (parse, "got frame");
gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi);
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
draining = GST_BASE_PARSE_DRAINING (parse);
while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) {
guint32 word = 0;
GST_LOG_OBJECT (wvparse, "checking next frame syncword; "
"lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final);
if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) ||
!gst_byte_reader_peek_uint32_be (&reader, &word)) {
GST_DEBUG_OBJECT (wvparse, "... but not sufficient data");
frmsize += 4;
goto more;
} else {
if (word != 0x7776706b) {
GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word);
*skipsize = off + 2;
goto skip;
}
/* need to parse each frame/block for metadata if several ones */
if (!final) {
gint av;
GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)",
frmsize, frmsize);
av = gst_byte_reader_get_remaining (&reader);
if (av < sizeof (WavpackHeader)) {
frmsize += sizeof (WavpackHeader);
goto more;
}
gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph);
off = frmsize;
frmsize += wph.ckSize + 8;
if (av < wph.ckSize + 8)
goto more;
gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi);
/* could also check for matching block_index and block_samples ?? */
}
}
/* resynced if we make it here */
lost_sync = FALSE;
}
rate = wpi.rate;
width = wpi.width;
chans = wpi.channels;
mask = wpi.channel_mask;
GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans);
GST_BUFFER_TIMESTAMP (buf) =
gst_util_uint64_scale_int (wph.block_index, GST_SECOND, rate);
GST_BUFFER_DURATION (buf) =
gst_util_uint64_scale_int (wph.block_index + wph.block_samples,
GST_SECOND, rate) - GST_BUFFER_TIMESTAMP (buf);
if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans
|| wvparse->width != width || wvparse->channel_mask != mask)) {
GstCaps *caps;
if (wpi.correction) {
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
} else {
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, chans,
"rate", G_TYPE_INT, rate,
"depth", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!mask)
mask = gst_wavpack_get_default_channel_mask (wvparse->channels);
if (mask != 0) {
GstAudioChannelPosition pos[64] =
{ GST_AUDIO_CHANNEL_POSITION_INVALID, };
guint64 gmask;
if (!gst_wavpack_get_channel_positions (chans, mask, pos)) {
GST_WARNING_OBJECT (wvparse, "Failed to determine channel layout");
} else {
gst_audio_channel_positions_to_mask (pos, chans, &gmask);
if (gmask)
gst_caps_set_simple (caps,
"channel-mask", GST_TYPE_BITMASK, gmask, NULL);
}
}
}
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
wvparse->sample_rate = rate;
wvparse->channels = chans;
wvparse->width = width;
wvparse->channel_mask = mask;
if (wvparse->total_samples) {
GST_DEBUG_OBJECT (wvparse, "setting duration");
gst_base_parse_set_duration (GST_BASE_PARSE (wvparse),
GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples,
GST_SECOND, wvparse->sample_rate), 0);
}
}
/* return to normal size */
gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader));
gst_buffer_unmap (buf, &map);
return gst_base_parse_finish_frame (parse, frame, frmsize);
skip:
gst_buffer_unmap (buf, &map);
GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize);
return GST_FLOW_OK;
more:
gst_buffer_unmap (buf, &map);
GST_LOG_OBJECT (wvparse, "need at least %u", frmsize);
gst_base_parse_set_min_frame_size (parse, frmsize);
*skipsize = 0;
return GST_FLOW_OK;
}
static GstCaps *
gst_wavpack_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
/* FIXME: handle filter caps */
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
/* Remove the framed field */
peercaps = gst_caps_make_writable (peercaps);
n = gst_caps_get_size (peercaps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (peercaps, i);
gst_structure_remove_field (s, "framed");
}
res =
gst_caps_intersect_full (peercaps,
gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
} else {
res =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
(parse)));
}
return res;
}

131
gst/audioparsers/gstwavpackparse.h Normal file
View file

@ -0,0 +1,131 @@
/* GStreamer Wavpack parser
* Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2012 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_WAVPACK_PARSE_H__
#define __GST_WAVPACK_PARSE_H__
#include <gst/gst.h>
#include <gst/base/gstbaseparse.h>
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_PARSE \
(gst_wavpack_parse_get_type())
#define GST_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_WAVPACK_PARSE, GstWavpackParse))
#define GST_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_WAVPACK_PARSE, GstWavpackParseClass))
#define GST_IS_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_WAVPACK_PARSE))
#define GST_IS_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_WAVPACK_PARSE))
#define ID_UNIQUE 0x3f
#define ID_OPTIONAL_DATA 0x20
#define ID_ODD_SIZE 0x40
#define ID_LARGE 0x80
#define ID_DUMMY 0x0
#define ID_ENCODER_INFO 0x1
#define ID_DECORR_TERMS 0x2
#define ID_DECORR_WEIGHTS 0x3
#define ID_DECORR_SAMPLES 0x4
#define ID_ENTROPY_VARS 0x5
#define ID_HYBRID_PROFILE 0x6
#define ID_SHAPING_WEIGHTS 0x7
#define ID_FLOAT_INFO 0x8
#define ID_INT32_INFO 0x9
#define ID_WV_BITSTREAM 0xa
#define ID_WVC_BITSTREAM 0xb
#define ID_WVX_BITSTREAM 0xc
#define ID_CHANNEL_INFO 0xd
#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3)
#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4)
#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7)
#define FLAG_FINAL_BLOCK (1 << 12)
typedef struct {
char ckID [4]; /* "wvpk" */
guint32 ckSize; /* size of entire block (minus 8, of course) */
guint16 version; /* 0x402 to 0x410 are currently valid for decode */
guchar track_no; /* track number (0 if not used, like now) */
guchar index_no; /* track sub-index (0 if not used, like now) */
guint32 total_samples; /* total samples for entire file, but this is
* only valid if block_index == 0 and a value of
* -1 indicates unknown length */
guint32 block_index; /* index of first sample in block relative to
* beginning of file (normally this would start
* at 0 for the first block) */
guint32 block_samples; /* number of samples in this block (0 = no audio) */
guint32 flags; /* various flags for id and decoding */
guint32 crc; /* crc for actual decoded data */
} WavpackHeader;
typedef struct {
gboolean correction;
guint rate;
guint width;
guint channels;
guint channel_mask;
} WavpackInfo;
typedef struct _GstWavpackParse GstWavpackParse;
typedef struct _GstWavpackParseClass GstWavpackParseClass;
/**
* GstWavpackParse:
*
* The opaque GstWavpackParse object
*/
struct _GstWavpackParse {
GstBaseParse baseparse;
/*< private >*/
gint sample_rate;
gint channels;
gint width;
gint channel_mask;
guint total_samples;
};
/**
* GstWavpackParseClass:
* @parent_class: Element parent class.
*
* The opaque GstWavpackParseClass data structure.
*/
struct _GstWavpackParseClass {
GstBaseParseClass baseparse_class;
};
GType gst_wavpack_parse_get_type (void);
G_END_DECLS
#endif /* __GST_WAVPACK_PARSE_H__ */

3
gst/audioparsers/plugin.c
View file

@ -27,6 +27,7 @@
#include "gstdcaparse.h"
#include "gstflacparse.h"
#include "gstmpegaudioparse.h"
#include "gstwavpackparse.h"
static gboolean
plugin_init (GstPlugin * plugin)
@ -45,6 +46,8 @@ plugin_init (GstPlugin * plugin)
GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE);
ret &= gst_element_register (plugin, "mpegaudioparse",
GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE);
ret &= gst_element_register (plugin, "wavpackparse",
GST_RANK_PRIMARY + 1, GST_TYPE_WAVPACK_PARSE);
return ret;
}

2
gst/avi/gstavimux.c
View file

@ -178,7 +178,7 @@ static GstStaticPadTemplate audio_sink_factory =
"rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]; "
#endif*/
"audio/x-ac3, "
"rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]; "
"rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 6 ]; "
"audio/x-alaw, "
"rate = (int) [ 1000, 48000 ], " "channels = (int) [ 1, 2 ]; "
"audio/x-mulaw, "

2
gst/debugutils/gstcapssetter.c
View file

@ -294,7 +294,7 @@ gst_caps_setter_set_property (GObject * object, guint prop_id,
}
/* try to activate these new caps next time around */
gst_base_transform_reconfigure (GST_BASE_TRANSFORM (filter));
gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM (filter));
break;
}
case PROP_JOIN:

23
gst/flv/gstflvmux.c
View file

@ -1144,6 +1144,29 @@ gst_flv_mux_write_header (GstFlvMux * mux)
GSList *l;
GstFlowReturn ret;
/* if not streaming, check if downstream is seekable */
if (!mux->streamable) {
gboolean seekable;
GstQuery *query;
query = gst_query_new_seeking (GST_FORMAT_BYTES);
if (gst_pad_peer_query (mux->srcpad, query)) {
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not ");
} else {
/* have to assume seeking is supported if query not handled downstream */
GST_WARNING_OBJECT (mux, "downstream did not handle seeking query");
seekable = FALSE;
}
if (!seekable) {
mux->streamable = TRUE;
g_object_notify (G_OBJECT (mux), "streamable");
GST_WARNING_OBJECT (mux, "downstream is not seekable, but "
"streamable=false. Will ignore that and create streamable output "
"instead");
}
}
header = gst_flv_mux_create_header (mux);
metadata = gst_flv_mux_create_metadata (mux, TRUE);
video_codec_data = NULL;

2
gst/flv/gstindex.c
View file

@ -172,7 +172,7 @@ gst_index_class_init (GstIndexClass * klass)
gst_index_signals[ENTRY_ADDED] =
g_signal_new ("entry-added", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstIndexClass, entry_added), NULL, NULL,
gst_marshal_VOID__BOXED, G_TYPE_NONE, 1, GST_TYPE_INDEX_ENTRY);
NULL, G_TYPE_NONE, 1, GST_TYPE_INDEX_ENTRY);
gobject_class->set_property = gst_index_set_property;
gobject_class->get_property = gst_index_get_property;

2
gst/goom2k1/gstgoom.c
View file

@ -508,7 +508,7 @@ gst_goom_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
}
GST_DEBUG_OBJECT (goom,
"Input buffer has %d samples, time=%" G_GUINT64_FORMAT,
"Input buffer has %" G_GSIZE_FORMAT " samples, time=%" G_GUINT64_FORMAT,
gst_buffer_get_size (buffer) / goom->bps, GST_BUFFER_TIMESTAMP (buffer));
/* Collect samples until we have enough for an output frame */

4
gst/interleave/interleave.c
View file

@ -57,6 +57,10 @@
* </refsect2>
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

24
gst/isomp4/gstqtmux.c
View file

@ -1629,6 +1629,30 @@ gst_qt_mux_start_file (GstQTMux * qtmux)
gst_pad_set_caps (qtmux->srcpad, caps);
gst_caps_unref (caps);
/* if not streaming, check if downstream is seekable */
if (!qtmux->streamable) {
gboolean seekable;
GstQuery *query;
query = gst_query_new_seeking (GST_FORMAT_BYTES);
if (gst_pad_peer_query (qtmux->srcpad, query)) {
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
GST_INFO_OBJECT (qtmux, "downstream is %sseekable",
seekable ? "" : "not ");
} else {
/* have to assume seeking is supported if query not handled downstream */
GST_WARNING_OBJECT (qtmux, "downstream did not handle seeking query");
seekable = FALSE;
}
if (!seekable) {
qtmux->streamable = TRUE;
g_object_notify (G_OBJECT (qtmux), "streamable");
GST_WARNING_OBJECT (qtmux, "downstream is not seekable, but "
"streamable=false. Will ignore that and create streamable output "
"instead");
}
}
/* let downstream know we think in BYTES and expect to do seeking later on */
gst_segment_init (&segment, GST_FORMAT_BYTES);
gst_pad_push_event (qtmux->srcpad, gst_event_new_segment (&segment));

23
gst/matroska/matroska-mux.c
View file

@ -2340,6 +2340,29 @@ gst_matroska_mux_start (GstMatroskaMux * mux)
guint32 segment_uid[4];
GTimeVal time = { 0, 0 };
/* if not streaming, check if downstream is seekable */
if (!mux->streamable) {
gboolean seekable;
GstQuery *query;
query = gst_query_new_seeking (GST_FORMAT_BYTES);
if (gst_pad_peer_query (mux->srcpad, query)) {
gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL);
GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not ");
} else {
/* have to assume seeking is supported if query not handled downstream */
GST_WARNING_OBJECT (mux, "downstream did not handle seeking query");
seekable = FALSE;
}
if (!seekable) {
mux->streamable = TRUE;
g_object_notify (G_OBJECT (mux), "streamable");
GST_WARNING_OBJECT (mux, "downstream is not seekable, but "
"streamable=false. Will ignore that and create streamable output "
"instead");
}
}
if (!strcmp (mux->doctype, GST_MATROSKA_DOCTYPE_WEBM)) {
ebml->caps = gst_caps_new_empty_simple ("video/webm");
} else {

4
gst/rtpmanager/rtpsession.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>

4
gst/udp/gstdynudpsink.c
View file

@ -21,6 +21,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
gst/udp/gstmultiudpsink.c
View file

@ -29,6 +29,10 @@
* It can be combined with rtp payload encoders to implement RTP streaming.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

2
gst/videocrop/gstvideocrop.c
View file

@ -711,7 +711,7 @@ gst_video_crop_set_property (GObject * object, guint prop_id,
video_crop->crop_top);
GST_OBJECT_UNLOCK (video_crop);
gst_base_transform_reconfigure (GST_BASE_TRANSFORM (video_crop));
gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM (video_crop));
g_mutex_unlock (&video_crop->lock);
}

8
gst/videofilter/gstvideobalance.c
View file

@ -637,13 +637,19 @@ gst_video_balance_colorbalance_get_value (GstColorBalance * balance,
return value;
}
static GstColorBalanceType
gst_video_balance_colorbalance_get_balance_type (GstColorBalance * balance)
{
return GST_COLOR_BALANCE_SOFTWARE;
}
static void
gst_video_balance_colorbalance_init (GstColorBalanceInterface * iface)
{
GST_COLOR_BALANCE_TYPE (iface) = GST_COLOR_BALANCE_SOFTWARE;
iface->list_channels = gst_video_balance_colorbalance_list_channels;
iface->set_value = gst_video_balance_colorbalance_set_value;
iface->get_value = gst_video_balance_colorbalance_get_value;
iface->get_balance_type = gst_video_balance_colorbalance_get_balance_type;
}
static GstColorBalanceChannel *

2
gst/videofilter/gstvideoflip.c
View file

@ -965,7 +965,7 @@ gst_video_flip_set_property (GObject * object, guint prop_id,
gst_base_transform_set_passthrough (btrans,
method == GST_VIDEO_FLIP_METHOD_IDENTITY);
gst_base_transform_reconfigure (btrans);
gst_base_transform_reconfigure_src (btrans);
} else {
GST_OBJECT_UNLOCK (videoflip);
}

4
sys/oss4/oss4-audio.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

4
sys/oss4/oss4-property-probe.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif

24
sys/v4l2/gstv4l2bufferpool.c
View file

@ -57,6 +57,19 @@ GST_DEBUG_CATEGORY_EXTERN (v4l2_debug);
/*
* GstV4l2Buffer:
*/
GType
gst_v4l2_meta_api_get_type (void)
{
static volatile GType type;
static const gchar *tags[] = { "memory", NULL };
if (g_once_init_enter (&type)) {
GType _type = gst_meta_api_type_register ("GstV4l2MetaAPI", tags);
g_once_init_leave (&type, _type);
}
return type;
}
const GstMetaInfo *
gst_v4l2_meta_get_info (void)
{
@ -64,10 +77,9 @@ gst_v4l2_meta_get_info (void)
if (meta_info == NULL) {
meta_info =
gst_meta_register ("GstV4l2Meta", "GstV4l2Meta",
gst_meta_register (gst_v4l2_meta_api_get_type (), "GstV4l2Meta",
sizeof (GstV4l2Meta), (GstMetaInitFunction) NULL,
(GstMetaFreeFunction) NULL, (GstMetaCopyFunction) NULL,
(GstMetaTransformFunction) NULL);
(GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL);
}
return meta_info;
}
@ -864,6 +876,7 @@ GstBufferPool *
gst_v4l2_buffer_pool_new (GstV4l2Object * obj, GstCaps * caps)
{
GstV4l2BufferPool *pool;
GstStructure *s;
gint fd;
fd = v4l2_dup (obj->video_fd);
@ -874,8 +887,9 @@ gst_v4l2_buffer_pool_new (GstV4l2Object * obj, GstCaps * caps)
pool->video_fd = fd;
pool->obj = obj;
gst_buffer_pool_config_set (GST_BUFFER_POOL_CAST (pool)->config, caps,
obj->sizeimage, 2, 0, 0, 0);
s = gst_buffer_pool_get_config (GST_BUFFER_POOL_CAST (pool));
gst_buffer_pool_config_set (s, caps, obj->sizeimage, 2, 0, 0, 0);
gst_buffer_pool_set_config (GST_BUFFER_POOL_CAST (pool), s);
return GST_BUFFER_POOL (pool);

3
sys/v4l2/gstv4l2bufferpool.h
View file

@ -79,8 +79,9 @@ struct _GstV4l2Meta {
struct v4l2_buffer vbuffer;
};
GType gst_v4l2_meta_api_get_type (void);
const GstMetaInfo * gst_v4l2_meta_get_info (void);
#define GST_V4L2_META_GET(buf) ((GstV4l2Meta *)gst_buffer_get_meta(buf,gst_v4l2_meta_get_info()))
#define GST_V4L2_META_GET(buf) ((GstV4l2Meta *)gst_buffer_get_meta(buf,gst_v4l2_meta_api_get_type()))
#define GST_V4L2_META_ADD(buf) ((GstV4l2Meta *)gst_buffer_add_meta(buf,gst_v4l2_meta_get_info(),NULL))
GType gst_v4l2_buffer_pool_get_type (void);

9
sys/v4l2/gstv4l2colorbalance.h
View file

@ -90,15 +90,20 @@ interface_as_function ## _color_balance_get_value (GstColorBalance * balance,
return gst_v4l2_color_balance_get_value(this->v4l2object, channel); \
} \
\
static GstColorBalanceType \
interface_as_function ## _color_balance_get_balance_type (GstColorBalance * balance) \
{ \
return GST_COLOR_BALANCE_HARDWARE; \
} \
\
static void \
interface_as_function ## _color_balance_interface_init (GstColorBalanceInterface * iface) \
{ \
GST_COLOR_BALANCE_TYPE (iface) = GST_COLOR_BALANCE_HARDWARE; \
\
/* default virtual functions */ \
iface->list_channels = interface_as_function ## _color_balance_list_channels; \
iface->set_value = interface_as_function ## _color_balance_set_value; \
iface->get_value = interface_as_function ## _color_balance_get_value; \
iface->get_balance_type = interface_as_function ## _color_balance_get_balance_type; \
} \
#endif /* __GST_V4L2_COLOR_BALANCE_H__ */

8
sys/v4l2/gstv4l2object.c
View file

@ -18,6 +18,10 @@
* USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
@ -1307,7 +1311,7 @@ gst_v4l2_object_v4l2fourcc_to_structure (guint32 fourcc)
break;
#ifdef V4L2_PIX_FMT_SBGGR8
case V4L2_PIX_FMT_SBGGR8:
structure = gst_structure_new_empty ("video/x-raw-bayer");
structure = gst_structure_new_empty ("video/x-bayer");
break;
#endif
#ifdef V4L2_PIX_FMT_SN9C10X
@ -1468,7 +1472,7 @@ gst_v4l2_object_get_caps_info (GstV4l2Object * v4l2object, GstCaps * caps,
} else if (g_str_equal (mimetype, "image/jpeg")) {
fourcc = V4L2_PIX_FMT_JPEG;
#ifdef V4L2_PIX_FMT_SBGGR8
} else if (g_str_equal (mimetype, "video/x-raw-bayer")) {
} else if (g_str_equal (mimetype, "video/x-bayer")) {
fourcc = V4L2_PIX_FMT_SBGGR8;
#endif
#ifdef V4L2_PIX_FMT_SN9C10X

4
sys/v4l2/gstv4l2sink.c
View file

@ -653,8 +653,8 @@ gst_v4l2sink_propose_allocation (GstBaseSink * bsink, GstQuery * query)
gst_query_set_allocation_params (query, size, 2, 0, 0, 0, pool);
/* we also support various metadata */
gst_query_add_allocation_meta (query, GST_VIDEO_META_API);
gst_query_add_allocation_meta (query, GST_VIDEO_CROP_META_API);
gst_query_add_allocation_meta (query, GST_VIDEO_META_API_TYPE);
gst_query_add_allocation_meta (query, GST_VIDEO_CROP_META_API_TYPE);
if (pool)
gst_object_unref (pool);

2
sys/v4l2/gstv4l2src.c
View file

@ -571,7 +571,7 @@ gst_v4l2src_decide_allocation (GstBaseSrc * bsrc, GstQuery * query)
alignment);
/* if downstream supports video metadata, add this to the pool config */
if (gst_query_has_allocation_meta (query, GST_VIDEO_META_API))
if (gst_query_has_allocation_meta (query, GST_VIDEO_META_API_TYPE))
gst_buffer_pool_config_add_option (config,
GST_BUFFER_POOL_OPTION_VIDEO_META);

4
sys/v4l2/gstv4l2videooverlay.c
View file

@ -31,6 +31,8 @@
#include <X11/extensions/Xv.h>
#include <X11/extensions/Xvlib.h>
#include <sys/stat.h>
/* for XkbKeycodeToKeysym */
#include <X11/XKBlib.h>
#include <gst/interfaces/navigation.h>
@ -323,7 +325,7 @@ event_refresh (gpointer data)
case KeyPress:
case KeyRelease:
g_mutex_lock (&v4l2xv->mutex);
keysym = XKeycodeToKeysym (v4l2xv->dpy, e.xkey.keycode, 0);
keysym = XkbKeycodeToKeysym (v4l2xv->dpy, e.xkey.keycode, 0, 0);
if (keysym != NoSymbol) {
key_str = XKeysymToString (keysym);
} else {

2
sys/ximage/Makefile.am
View file

@ -7,7 +7,7 @@ libgstximagesrc_la_CFLAGS = \
$(GST_CFLAGS) \
$(X_CFLAGS) $(XFIXES_CFLAGS) $(XDAMAGE_CFLAGS)
libgstximagesrc_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) \
$(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) \
$(GST_BASE_LIBS) \
$(X_LIBS) $(XSHM_LIBS) $(XFIXES_LIBS) $(XDAMAGE_LIBS)
libgstximagesrc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)

21
sys/ximage/gstximagesrc.c
View file

@ -30,7 +30,7 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch ximagesrc ! video/x-raw-rgb,framerate=5/1 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=desktop.ogg
* gst-launch ximagesrc ! video/x-raw,framerate=5/1 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=desktop.ogg
* ]| Encodes your X display to an Ogg theora video at 5 frames per second.
* </refsect2>
*/
@ -48,6 +48,7 @@
#include <gst/gst.h>
#include <gst/gst-i18n-plugin.h>
#include <gst/video/video.h>
#include "gst/glib-compat-private.h"
@ -56,7 +57,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_debug_ximage_src);
static GstStaticPadTemplate t =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-raw-rgb, "
GST_STATIC_CAPS ("video/x-raw, "
"framerate = (fraction) [ 0, MAX ], "
"width = (int) [ 1, MAX ], " "height = (int) [ 1, MAX ], "
"pixel-aspect-ratio = (fraction) [ 0, MAX ]"));
@ -1018,6 +1019,7 @@ gst_ximage_src_get_caps (GstBaseSrc * bs, GstCaps * filter)
GstXImageSrc *s = GST_XIMAGE_SRC (bs);
GstXContext *xcontext;
gint width, height;
GstVideoFormat format;
if ((!s->xcontext) && (!gst_ximage_src_open_display (s, s->display_name)))
return
@ -1072,13 +1074,14 @@ gst_ximage_src_get_caps (GstBaseSrc * bs, GstCaps * filter)
s->endy = height - 1;
}
GST_DEBUG ("width = %d, height=%d", width, height);
return gst_caps_new_simple ("video/x-raw-rgb",
"bpp", G_TYPE_INT, xcontext->bpp,
"depth", G_TYPE_INT, xcontext->depth,
"endianness", G_TYPE_INT, xcontext->endianness,
"red_mask", G_TYPE_INT, xcontext->r_mask_output,
"green_mask", G_TYPE_INT, xcontext->g_mask_output,
"blue_mask", G_TYPE_INT, xcontext->b_mask_output,
format =
gst_video_format_from_masks (xcontext->depth, xcontext->bpp,
xcontext->endianness, xcontext->r_mask_output, xcontext->g_mask_output,
xcontext->b_mask_output, 0);
return gst_caps_new_simple ("video/x-raw",
"format", G_TYPE_STRING, gst_video_format_to_string (format),
"width", G_TYPE_INT, width,
"height", G_TYPE_INT, height,
"framerate", GST_TYPE_FRACTION_RANGE, 1, G_MAXINT, G_MAXINT, 1,

18
sys/ximage/ximageutil.c
View file

@ -23,6 +23,19 @@
#include "ximageutil.h"
GType
gst_meta_ximage_api_get_type (void)
{
static volatile GType type;
static const gchar *tags[] = { "memory", NULL };
if (g_once_init_enter (&type)) {
GType _type = gst_meta_api_type_register ("GstMetaXImageSrcAPI", tags);
g_once_init_leave (&type, _type);
}
return type;
}
const GstMetaInfo *
gst_meta_ximage_get_info (void)
{
@ -30,10 +43,9 @@ gst_meta_ximage_get_info (void)
if (meta_ximage_info == NULL) {
meta_ximage_info =
gst_meta_register ("GstMetaXImageSrc", "GstMetaXImageSrc",
gst_meta_register (gst_meta_ximage_api_get_type (), "GstMetaXImageSrc",
sizeof (GstMetaXImage), (GstMetaInitFunction) NULL,
(GstMetaFreeFunction) NULL, (GstMetaCopyFunction) NULL,
(GstMetaTransformFunction) NULL);
(GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL);
}
return meta_ximage_info;
}

3
sys/ximage/ximageutil.h
View file

@ -160,8 +160,9 @@ struct _GstMetaXImage {
BufferReturnFunc return_func;
};
GType gst_meta_ximage_api_get_type (void);
const GstMetaInfo * gst_meta_ximage_get_info (void);
#define GST_META_XIMAGE_GET(buf) ((GstMetaXImage *)gst_buffer_get_meta(buf,gst_meta_ximage_get_info()))
#define GST_META_XIMAGE_GET(buf) ((GstMetaXImage *)gst_buffer_get_meta(buf,gst_meta_ximage_api_get_type()))
#define GST_META_XIMAGE_ADD(buf) ((GstMetaXImage *)gst_buffer_add_meta(buf,gst_meta_ximage_get_info(),NULL))
GstBuffer *gst_ximageutil_ximage_new (GstXContext *xcontext,

2
tests/check/Makefile.am
View file

@ -83,7 +83,6 @@ endif
if USE_WAVPACK
check_wavpack = \
elements/wavpackparse \
elements/wavpackdec \
elements/wavpackenc \
pipelines/wavpack
@ -150,6 +149,7 @@ check_PROGRAMS = \
elements/udpsrc \
elements/videocrop \
elements/videofilter \
elements/wavpackparse \
elements/y4menc \
pipelines/simple-launch-lines \
pipelines/effectv \

4
tests/check/elements/interleave.c
View file

@ -18,6 +18,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

54
tests/check/elements/wavpackdec.c
View file

@ -29,6 +29,12 @@
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
#if G_BYTE_ORDER == G_BIG_ENDIAN
#define AUDIO_FORMAT "S16BE"
#else
#define AUDIO_FORMAT "S16LE"
#endif
guint8 test_frame[] = {
0x77, 0x76, 0x70, 0x6B, /* "wvpk" */
0x2E, 0x00, 0x00, 0x00, /* ckSize */
@ -51,13 +57,12 @@ guint8 test_frame[] = {
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " AUDIO_FORMAT ", "
"layout = (string) interleaved, "
"channels = (int) 1, " "rate = (int) 44100")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
@ -73,8 +78,8 @@ setup_wavpackdec (void)
GST_DEBUG ("setup_wavpackdec");
wavpackdec = gst_check_setup_element ("wavpackdec");
mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate, NULL);
mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate);
mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
@ -100,6 +105,7 @@ GST_START_TEST (test_decode_frame)
GstBuffer *inbuffer, *outbuffer;
GstBus *bus;
int i;
GstMapInfo map;
wavpackdec = setup_wavpackdec ();
@ -109,27 +115,29 @@ GST_START_TEST (test_decode_frame)
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
/* uncompressed data should be 102400 bytes */
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 102400);
/* and all 102400 bytes must be 0, i.e. silence */
for (i = 0; i < 102400; i++)
fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0);
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
/* uncompressed data should be 102400 bytes */
fail_unless_equals_int (map.size, 51200);
/* and all bytes must be 0, i.e. silence */
for (i = 0; i < 51200; i++)
fail_unless_equals_int (map.data[i], 0);
gst_buffer_unmap (outbuffer, &map);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
gst_buffer_unref (outbuffer);
@ -161,19 +169,16 @@ GST_START_TEST (test_decode_frame_with_broken_header)
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame));
/* break header */
GST_BUFFER_DATA (inbuffer)[2] = 'e';
gst_buffer_memset (inbuffer, 2, 'e', 1);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
@ -201,17 +206,14 @@ GST_START_TEST (test_decode_frame_with_incomplete_frame)
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame) - 2);
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2);
gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame) - 2);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);

46
tests/check/elements/wavpackenc.c
View file

@ -30,16 +30,20 @@
static GstPad *mysrcpad, *mysinkpad;
static GstBus *bus;
#define RAW_CAPS_STRING "audio/x-raw-int, " \
"width = (int) 32, " \
"depth = (int) 16, " \
#if G_BYTE_ORDER == G_BIG_ENDIAN
#define AUDIO_FORMAT "S32BE"
#else
#define AUDIO_FORMAT "S32LE"
#endif
#define RAW_CAPS_STRING "audio/x-raw, " \
"format = (string) " AUDIO_FORMAT ", " \
"layout = (string) interleaved, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true"
"rate = (int) 44100"
#define WAVPACK_CAPS_STRING "audio/x-wavpack, " \
"width = (int) 16, " \
"width = (int) 32, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"framed = (boolean) true"
@ -48,19 +52,17 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"width = (int) 32, "
"channels = (int) 1, "
"rate = (int) 44100, " "framed = (boolean) true"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true"));
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " AUDIO_FORMAT ", "
"layout = (string) interleaved, "
"channels = (int) 1, " "rate = (int) 44100"));
static GstElement *
setup_wavpackenc (void)
@ -69,8 +71,8 @@ setup_wavpackenc (void)
GST_DEBUG ("setup_wavpackenc");
wavpackenc = gst_check_setup_element ("wavpackenc");
mysrcpad = gst_check_setup_src_pad (wavpackenc, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (wavpackenc, &sinktemplate, NULL);
mysrcpad = gst_check_setup_src_pad (wavpackenc, &srctemplate);
mysinkpad = gst_check_setup_sink_pad (wavpackenc, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
@ -111,20 +113,16 @@ GST_START_TEST (test_encode_silence)
wavpackenc = setup_wavpackenc ();
inbuffer = gst_buffer_new_and_alloc (1000);
for (i = 0; i < 1000; i++)
GST_BUFFER_DATA (inbuffer)[i] = 0;
gst_buffer_memset (inbuffer, 0, 0, 1000);
caps = gst_caps_from_string (RAW_CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
fail_unless (gst_pad_set_caps (mysrcpad, caps));
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackenc, bus);
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE);
@ -134,11 +132,9 @@ GST_START_TEST (test_encode_silence)
fail_if (outbuffer == NULL);
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0,
"Failed to encode to valid Wavpack frames");
/* free all buffers */

123
tests/check/elements/wavpackparse.c
View file

@ -28,8 +28,6 @@
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
static GstBus *bus;
static GstElement *wavpackparse;
/* Wavpack file with 2 frames of silence */
guint8 test_file[] = {
@ -69,99 +67,53 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack"));
static void
wavpackparse_found_pad (GstElement * src, GstPad * pad, gpointer data)
{
GstPad *srcpad;
mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
fail_if (mysinkpad == NULL, "Couldn't create sinkpad");
srcpad = gst_element_get_static_pad (wavpackparse, "src");
fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse");
gst_pad_set_chain_function (mysinkpad, gst_check_chain_func);
fail_unless (gst_pad_link (srcpad, mysinkpad) == GST_PAD_LINK_OK,
"Failed to link pads");
gst_pad_set_active (mysinkpad, TRUE);
gst_object_unref (srcpad);
}
static void
static GstElement *
setup_wavpackparse (void)
{
GstPad *sinkpad;
GstElement *wavpackparse;
GST_DEBUG ("setup_wavpackparse");
wavpackparse = gst_element_factory_make ("wavpackparse", "wavpackparse");
fail_if (wavpackparse == NULL, "Could not create wavpackparse");
wavpackparse = gst_check_setup_element ("wavpackparse");
mysrcpad = gst_check_setup_src_pad (wavpackparse, &srctemplate);
mysinkpad = gst_check_setup_sink_pad (wavpackparse, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
fail_if (mysrcpad == NULL, "Could not create srcpad");
sinkpad = gst_element_get_static_pad (wavpackparse, "sink");
fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse");
fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK,
"Failed to link pads");
gst_object_unref (sinkpad);
g_signal_connect (wavpackparse, "pad-added",
G_CALLBACK (wavpackparse_found_pad), NULL);
bus = gst_bus_new ();
gst_element_set_bus (wavpackparse, bus);
fail_unless (gst_element_set_state (wavpackparse,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
return wavpackparse;
}
static void
cleanup_wavpackparse (void)
cleanup_wavpackparse (GstElement * wavpackparse)
{
GstPad *sinkpad, *srcpad;
GST_DEBUG ("cleanup_wavpackparse");
gst_element_set_state (wavpackparse, GST_STATE_NULL);
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (wavpackparse, NULL);
gst_object_unref (GST_OBJECT (bus));
sinkpad = gst_element_get_static_pad (wavpackparse, "sink");
fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse");
fail_unless (gst_pad_unlink (mysrcpad, sinkpad), "Failed to unlink pads");
gst_pad_set_caps (mysrcpad, NULL);
gst_object_unref (sinkpad);
gst_object_unref (mysrcpad);
srcpad = gst_element_get_static_pad (wavpackparse, "src");
fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse");
fail_unless (gst_pad_unlink (srcpad, mysinkpad), "Failed to unlink pads");
gst_pad_set_caps (mysinkpad, NULL);
gst_object_unref (srcpad);
gst_object_unref (mysinkpad);
fail_unless (gst_element_set_state (wavpackparse, GST_STATE_NULL) ==
GST_STATE_CHANGE_SUCCESS, "could not set to null");
gst_object_unref (wavpackparse);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (wavpackparse);
gst_check_teardown_sink_pad (wavpackparse);
gst_check_teardown_element (wavpackparse);
}
GST_START_TEST (test_parsing_valid_frames)
{
GstElement *wavpackparse;
GstBuffer *inbuffer, *outbuffer;
int i, num_buffers;
GstFormat format = GST_FORMAT_DEFAULT;
GstFormat format = GST_FORMAT_TIME;
gint64 pos;
setup_wavpackparse ();
wavpackparse = setup_wavpackparse ();
fail_unless (gst_element_set_state (wavpackparse,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (sizeof (test_file));
memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file));
gst_buffer_ref (inbuffer);
gst_buffer_fill (inbuffer, 0, test_file, sizeof (test_file));
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
gst_buffer_unref (inbuffer);
num_buffers = g_list_length (buffers);
/* should get 2 buffers, each one complete wavpack frame */
@ -171,21 +123,17 @@ GST_START_TEST (test_parsing_valid_frames)
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0,
"Buffer contains no Wavpack frame");
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866);
switch (i) {
case 0:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 25600);
break;
}
case 1:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 51200);
break;
}
}
@ -196,35 +144,37 @@ GST_START_TEST (test_parsing_valid_frames)
outbuffer = NULL;
}
fail_unless (gst_element_query_position (wavpackparse, &format, &pos),
fail_unless (gst_element_query_position (wavpackparse, format, &pos),
"Position query failed");
fail_unless_equals_int (pos, 51200);
fail_unless (gst_element_query_duration (wavpackparse, &format, NULL),
fail_unless_equals_int64 (pos, 580498866 * 2);
fail_unless (gst_element_query_duration (wavpackparse, format, NULL),
"Duration query failed");
g_list_free (buffers);
buffers = NULL;
cleanup_wavpackparse ();
cleanup_wavpackparse (wavpackparse);
}
GST_END_TEST;
GST_START_TEST (test_parsing_invalid_first_header)
{
GstElement *wavpackparse;
GstBuffer *inbuffer, *outbuffer;
int i, num_buffers;
setup_wavpackparse ();
wavpackparse = setup_wavpackparse ();
fail_unless (gst_element_set_state (wavpackparse,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (sizeof (test_file));
memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file));
GST_BUFFER_DATA (inbuffer)[0] = 'k';
gst_buffer_ref (inbuffer);
gst_buffer_fill (inbuffer, 0, test_file, sizeof (test_file));
gst_buffer_memset (inbuffer, 0, 'k', 1);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
gst_buffer_unref (inbuffer);
num_buffers = g_list_length (buffers);
@ -235,14 +185,13 @@ GST_START_TEST (test_parsing_invalid_first_header)
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0,
"Buffer contains no Wavpack frame");
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866);
switch (i) {
case 0:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600);
break;
}
}
@ -256,7 +205,7 @@ GST_START_TEST (test_parsing_invalid_first_header)
g_list_free (buffers);
buffers = NULL;
cleanup_wavpackparse ();
cleanup_wavpackparse (wavpackparse);
}
GST_END_TEST;

8
tests/check/pipelines/wavpack.c
View file

@ -96,6 +96,7 @@ GST_START_TEST (test_encode_decode)
GMainLoop *loop;
GstBuffer *in, *out;
guint bus_watch = 0;
GstMapInfo map;
srcadapter = gst_adapter_new ();
fail_unless (srcadapter != NULL);
@ -170,9 +171,10 @@ GST_START_TEST (test_encode_decode)
gst_adapter_available (sinkadapter));
fail_unless (out != NULL);
fail_unless_equals_int (GST_BUFFER_SIZE (in), GST_BUFFER_SIZE (out));
fail_unless (memcmp (GST_BUFFER_DATA (in), GST_BUFFER_DATA (out),
GST_BUFFER_SIZE (in)) == 0);
fail_unless_equals_int (gst_buffer_get_size (in), gst_buffer_get_size (out));
gst_buffer_map (out, &map, GST_MAP_READ);
fail_unless (gst_buffer_memcmp (in, 0, map.data, map.size) == 0);
gst_buffer_unmap (out, &map);
gst_buffer_unref (in);
gst_buffer_unref (out);

4
tests/examples/audiofx/firfilter-example.c
View file

@ -21,6 +21,10 @@
* by transforming the frequency response to the filter kernel.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/examples/audiofx/iirfilter-example.c
View file

@ -23,6 +23,10 @@
* of the IIR filter that is used.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/examples/pulse/pulse.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <gst/gst.h>
#if 0

4
tests/examples/rtp/server-alsasrc-PCMA.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <math.h>

4
tests/icles/test-oss4.c
View file

@ -17,6 +17,10 @@
* Boston, MA 02111-1307, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif