*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
"bitrate = (string) \"16000\", " "dct-length = (int) 320")
);
-static gboolean gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_siren_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
G_DEFINE_TYPE (GstRTPSirenPay, gst_rtp_siren_pay,
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+ GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_siren_pay_class_init (GstRTPSirenPayClass * klass)
{
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
- gstbasertppayload_class->set_caps = gst_rtp_siren_pay_setcaps;
+ gstrtpbasepayload_class->set_caps = gst_rtp_siren_pay_setcaps;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_siren_pay_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_siren_pay_src_template));
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_set_static_metadata (gstelement_class,
"RTP Payloader for Siren Audio", "Codec/Payloader/Network/RTP",
"Packetize Siren audio streams into RTP packets",
"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
static void
gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstRTPBasePayload *rtpbasepayload;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
- basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
+ rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpsirenpay);
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpsirenpay);
/* we don't set the payload type, it should be set by the application using
* the pt property or the default 96 will be used */
- basertppayload->clock_rate = 16000;
+ rtpbasepayload->clock_rate = 16000;
- /* tell basertpaudiopayload that this is a frame based codec */
- gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
+ /* tell rtpbaseaudiopayload that this is a frame based codec */
+ gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
}
static gboolean
-gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+gst_rtp_siren_pay_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
{
GstRTPSirenPay *rtpsirenpay;
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gint dct_length;
GstStructure *structure;
const char *payload_name;
- rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+ rtpsirenpay = GST_RTP_SIREN_PAY (rtpbasepayload);
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
goto wrong_caps;
- gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
+ gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "SIREN",
16000);
/* set options for this frame based audio codec */
- gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
+ gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, 20, 40);
- return gst_basertppayload_set_outcaps (basertppayload, NULL);
+ return gst_rtp_base_payload_set_outcaps (rtpbasepayload, NULL);
/* ERRORS */
wrong_dct: