[Sugar-devel] [PATCH gst-plugins-espeak] Port to gstreamer 1.0
Daniel Narvaez
dwnarvaez at gmail.com
Sat Aug 11 03:58:21 EDT 2012
From: Daniel Narvaez <dwnarvaez at gmail.com>
---
configure.ac | 2 +-
src/espeak.c | 9 +++---
src/gstespeak.c | 84 ++++++++++++++++++++++++++-----------------------------
3 files changed, 45 insertions(+), 50 deletions(-)
diff --git a/configure.ac b/configure.ac
index 84c98c2..fc4b799 100644
--- a/configure.ac
+++ b/configure.ac
@@ -10,7 +10,7 @@ m4_ifdef([AM_SILENT_RULES], [AM_SILENT_RULES([yes])])
AC_PROG_CC
AC_PROG_LIBTOOL
-GST_MAJORMINOR=0.10
+GST_MAJORMINOR=1.0
PKG_CHECK_MODULES(GST, gstreamer-$GST_MAJORMINOR)
PKG_CHECK_MODULES(GST_AUDIO, gstreamer-audio-$GST_MAJORMINOR, have_audio=yes, have_audio=no)
diff --git a/src/espeak.c b/src/espeak.c
index 5cceffa..004fdf1 100644
--- a/src/espeak.c
+++ b/src/espeak.c
@@ -257,10 +257,9 @@ GstBuffer *play (Econtext * self, Espin * spin, gsize size_to_play) {
espeak_EVENT *event = &g_array_index (spin->events, espeak_EVENT,
spin->events_pos);
- GstBuffer *out = gst_buffer_new ();
+ GstBuffer *out = gst_buffer_new_and_alloc (size_to_play);
+ gst_buffer_fill(out, 0, spin->sound->data + spin->sound_offset, size_to_play);
GST_BUFFER_OFFSET (out) = spin->sound_offset;
- GST_BUFFER_DATA (out) = spin->sound->data + spin->sound_offset;
- GST_BUFFER_SIZE (out) = size_to_play;
GST_BUFFER_TIMESTAMP (out) = spin->audio_position;
spin->audio_position =
gst_util_uint64_scale_int (event->audio_position, GST_SECOND, 1000);
@@ -270,8 +269,8 @@ GstBuffer *play (Econtext * self, Espin * spin, gsize size_to_play) {
spin->sound_offset += size_to_play;
spin->events_pos += 1;
- GST_DEBUG ("out=%p size_to_play=%zd tell=%zd ts=%" G_GUINT64_FORMAT " dur=%"
- G_GUINT64_FORMAT, GST_BUFFER_DATA (out), size_to_play,
+ GST_DEBUG ("size_to_play=%zd tell=%zd ts=%" G_GUINT64_FORMAT " dur=%"
+ G_GUINT64_FORMAT, size_to_play,
spin->sound_offset, GST_BUFFER_TIMESTAMP (out),
GST_BUFFER_DURATION (out));
diff --git a/src/gstespeak.c b/src/gstespeak.c
index f7a05fa..d86e465 100644
--- a/src/gstespeak.c
+++ b/src/gstespeak.c
@@ -58,6 +58,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
+static void gst_espeak_uri_handler_init (gpointer g_iface, gpointer iface_data);
static GstFlowReturn gst_espeak_create (GstBaseSrc *,
guint64, guint, GstBuffer **);
static gboolean gst_espeak_start (GstBaseSrc *);
@@ -68,33 +69,19 @@ static void gst_espeak_finalize (GObject *);
static void gst_espeak_set_property (GObject *, guint, const GValue *,
GParamSpec *);
static void gst_espeak_get_property (GObject *, guint, GValue *, GParamSpec *);
-static GstCaps *gst_espeak_getcaps (GstBaseSrc *);
+static GstCaps *gst_espeak_getcaps (GstBaseSrc *self_, GstCaps *filter);
-GST_BOILERPLATE_FULL (GstEspeak, gst_espeak, GstBaseSrc, GST_TYPE_BASE_SRC,
- gst_espeak_init_uri);
+G_DEFINE_TYPE_EXTENDED (GstEspeak, gst_espeak, GST_TYPE_BASE_SRC, 0,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
+ gst_espeak_uri_handler_init));
/******************************************************************************/
-static void gst_espeak_base_init (gpointer gclass) {
- GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
-
- static GstElementDetails details = {
- "Espeak",
- "Source",
- "Uses eSpeak library as a sound source for GStreamer",
- "Aleksey S. Lim <alsroot at member.fsf.org>"
- };
-
- gst_element_class_set_details (element_class, &details);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
-}
-
/* initialize the espeak's class */
static void gst_espeak_class_init (GstEspeakClass * klass) {
GObjectClass *gobject_class = (GObjectClass *) klass;
GstBaseSrcClass *basesrc_class = (GstBaseSrcClass *) klass;
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
basesrc_class->create = gst_espeak_create;
basesrc_class->start = gst_espeak_start;
@@ -138,6 +125,16 @@ static void gst_espeak_class_init (GstEspeakClass * klass) {
g_param_spec_boxed ("caps", "Caps",
"Caps describing the format of the data", GST_TYPE_CAPS,
G_PARAM_READABLE));
+
+ gst_element_class_set_static_metadata (
+ element_class,
+ "Espeak",
+ "Source",
+ "Uses eSpeak library as a sound source for GStreamer",
+ "Aleksey S. Lim <alsroot at member.fsf.org>");
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_factory));
}
/* initialize the new element
@@ -145,7 +142,7 @@ static void gst_espeak_class_init (GstEspeakClass * klass) {
* set pad calback functions
* initialize instance structure
*/
-static void gst_espeak_init (GstEspeak * self, GstEspeakClass * gclass) {
+static void gst_espeak_init (GstEspeak * self) {
self->text = NULL;
self->pitch = 0;
self->rate = 0;
@@ -153,14 +150,17 @@ static void gst_espeak_init (GstEspeak * self, GstEspeakClass * gclass) {
self->voices = espeak_get_voices ();
self->speak = espeak_new (GST_ELEMENT (self));
- self->caps = gst_caps_new_simple ("audio/x-raw-int",
+ self->caps = gst_caps_new_simple ("audio/x-raw",
"rate", G_TYPE_INT, espeak_get_sample_rate (),
"channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+ "format", G_TYPE_STRING, "S16LE",
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
- gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_DEFAULT);
+ GstPad *pad = gst_element_get_static_pad(GST_ELEMENT(self), "src");
+ gst_pad_set_caps(pad, self->caps);
+
+ gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
}
static void gst_espeak_finalize (GObject * self_) {
@@ -177,7 +177,7 @@ static void gst_espeak_finalize (GObject * self_) {
g_value_array_free (self->voices);
self->voices = NULL;
- G_OBJECT_CLASS (parent_class)->dispose (self_);
+ G_OBJECT_CLASS (gst_espeak_parent_class)->dispose (self_);
}
/******************************************************************************/
@@ -268,10 +268,9 @@ gst_espeak_create (GstBaseSrc * self_, guint64 offset, guint size,
*buf = espeak_out (self->speak, size);
if (*buf) {
- gst_buffer_set_caps (*buf, self->caps);
return GST_FLOW_OK;
} else
- return GST_FLOW_UNEXPECTED;
+ return GST_FLOW_EOS;
}
static gboolean gst_espeak_start (GstBaseSrc * self_) {
@@ -292,24 +291,24 @@ static gboolean gst_espeak_is_seekable (GstBaseSrc * src) {
return FALSE;
}
-static GstCaps *gst_espeak_getcaps (GstBaseSrc * self_) {
+static GstCaps *gst_espeak_getcaps (GstBaseSrc * self_, GstCaps *filter) {
GstEspeak *self = GST_ESPEAK (self_);
return gst_caps_ref (self->caps);
}
/******************************************************************************/
-static GstURIType gst_espeak_uri_get_type (void) {
+static GstURIType gst_espeak_uri_get_type (GType type) {
return GST_URI_SRC;
}
-static gchar **gst_espeak_uri_get_protocols (void) {
- static gchar *protocols[] = { "espeak", NULL };
+static const gchar * const * gst_espeak_uri_get_protocols (GType type) {
+ static const gchar *protocols[] = { "espeak", NULL };
return protocols;
}
static gboolean
-gst_espeak_uri_set_uri (GstURIHandler * handler, const gchar * uri) {
+gst_espeak_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error) {
gchar *protocol, *text;
gboolean ret;
@@ -330,22 +329,19 @@ gst_espeak_uri_set_uri (GstURIHandler * handler, const gchar * uri) {
return TRUE;
}
+static gchar *
+gst_espeak_uri_get_uri (GstURIHandler * handler)
+{
+ return g_strdup_printf("espeak://%s", GST_ESPEAK (handler)->text);
+}
+
static void gst_espeak_uri_handler_init (gpointer g_iface, gpointer iface_data) {
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_espeak_uri_get_type;
iface->get_protocols = gst_espeak_uri_get_protocols;
iface->set_uri = gst_espeak_uri_set_uri;
-}
-
-static void gst_espeak_init_uri (GType filesrc_type) {
- static const GInterfaceInfo urihandler_info = {
- gst_espeak_uri_handler_init,
- NULL,
- NULL
- };
- g_type_add_interface_static (filesrc_type, GST_TYPE_URI_HANDLER,
- &urihandler_info);
+ iface->get_uri = gst_espeak_uri_get_uri;
}
/******************************************************************************/
@@ -367,7 +363,7 @@ static gboolean espeak_init (GstPlugin * espeak) {
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
- "espeak",
+ espeak,
"Uses eSpeak library as a sound source for GStreamer",
espeak_init,
PACKAGE_VERSION,
--
1.7.10.4
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