*
* 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
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
#include "gstrtpgsmpay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
#define GST_CAT_DEFAULT (rtpgsmpay_debug)
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
);
-static gboolean gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * payload,
+static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define gst_rtp_gsm_pay_parent_class parent_class
-G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_BASE_RTP_PAYLOAD);
+G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
{
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
"GSM Audio RTP Payloader");
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_gsm_pay_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_gsm_pay_src_template);
- gst_element_class_set_details_simple (gstelement_class, "RTP GSM payloader",
+ gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
"Codec/Payloader/Network/RTP",
"Payload-encodes GSM audio into a RTP packet",
"Zeeshan Ali <zeenix@gmail.com>");
- gstbasertppayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
+ gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
}
static void
gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
{
- GST_BASE_RTP_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
- GST_BASE_RTP_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
+ GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
+ GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
}
static gboolean
-gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
const char *stname;
GstStructure *structure;
if (strcmp ("audio/x-gsm", stname))
goto invalid_type;
- gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_rtp_base_payload_set_options (payload, "audio",
+ payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
-gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPGSMPay *rtpgsmpay;
guint payload_len;
GstBuffer *outbuf;
- guint8 *payload, *data;
GstClockTime timestamp, duration;
GstFlowReturn ret;
- gsize size;
- GstRTPBuffer rtp = { NULL };
rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
- data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
-
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one GSM frame per RTP packet for now */
- payload_len = size;
+ payload_len = gst_buffer_get_size (buffer);
/* FIXME, just error out for now */
- if (payload_len > GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay))
+ if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
goto too_big;
- outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+ outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* copy timestamp and duration */
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
- /* get payload */
- gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
-
- /* copy data in payload */
- payload = gst_rtp_buffer_get_payload (&rtp);
- memcpy (payload, data, size);
-
- gst_rtp_buffer_unmap (&rtp);
+ gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
- gst_buffer_unmap (buffer, data, size);
- gst_buffer_unref (buffer);
+ /* append payload */
+ outbuf = gst_buffer_append (outbuf, buffer);
- GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
+ GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
- ret = gst_basertppayload_push (basepayload, outbuf);
+ ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
{
GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
("payload_len %u > mtu %u", payload_len,
- GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)));
- gst_buffer_unmap (buffer, data, size);
+ GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
return GST_FLOW_ERROR;
}
}
projects / platform / upstream / gst-plugins-good.git / blobdiff