*
* 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp2tpay.h"
+#include "gstrtputils.h"
static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
);
static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
+ "payload = (int) " GST_RTP_PAYLOAD_MP2T_STRING ", "
+ "clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\" ; "
+ "application/x-rtp, "
+ "media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\"")
);
-static gboolean gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload *
+static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay);
static void gst_rtp_mp2t_pay_finalize (GObject * object);
#define gst_rtp_mp2t_pay_parent_class parent_class
-G_DEFINE_TYPE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GST_TYPE_BASE_RTP_PAYLOAD);
+G_DEFINE_TYPE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
- gstbasertppayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
+ gstrtpbasepayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_mp2t_pay_sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template));
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp2t_pay_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp2t_pay_src_template);
+ gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP",
"Payload-encodes MPEG2 TS into RTP packets (RFC 2250)",
"Wim Taymans <wim.taymans@gmail.com>");
static void
gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay)
{
- GST_BASE_RTP_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
- GST_BASE_RTP_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
+ GST_RTP_BASE_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
+ GST_RTP_BASE_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
rtpmp2tpay->adapter = gst_adapter_new ();
}
}
static gboolean
-gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
- gst_basertppayload_set_options (payload, "video", TRUE, "MP2T", 90000);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_rtp_base_payload_set_options (payload, "video",
+ payload->pt != GST_RTP_PAYLOAD_MP2T, "MP2T", 90000);
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
- guint avail;
- guint8 *payload;
- GstFlowReturn ret;
+ guint avail, mtu;
+ GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
- GstRTPBuffer rtp;
avail = gst_adapter_available (rtpmp2tpay->adapter);
- outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
- /* get payload */
- gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
- payload = gst_rtp_buffer_get_payload (&rtp);
+ mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
+
+ while (avail > 0 && (ret == GST_FLOW_OK)) {
+ guint towrite;
+ guint payload_len;
+ guint packet_len;
+ GstBuffer *paybuf;
+
+ /* this will be the total length of the packet */
+ packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
+
+ /* fill one MTU or all available bytes */
+ towrite = MIN (packet_len, mtu);
+
+ /* this is the payload length */
+ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
+ payload_len -= payload_len % 188;
- /* copy stuff from adapter to payload */
- gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
- gst_rtp_buffer_unmap (&rtp);
+ /* need whole packets */
+ if (!payload_len)
+ break;
- GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
- GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
+ /* create buffer to hold the payload */
+ outbuf =
+ gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+ (rtpmp2tpay), 0, 0, 0);
- GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
- gst_buffer_get_size (outbuf));
+ /* get payload */
+ paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
+ gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp2tpay), outbuf, paybuf, 0);
+ outbuf = gst_buffer_append (outbuf, paybuf);
+ avail -= payload_len;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
+ GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
+ GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
- /* flush the adapter content */
- gst_adapter_flush (rtpmp2tpay->adapter, avail);
+ GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
+ (guint) gst_buffer_get_size (outbuf));
+
+ ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
+ }
return ret;
}
static GstFlowReturn
-gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPMP2TPay *rtpmp2tpay;
rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
size = gst_buffer_get_size (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
+again:
ret = GST_FLOW_OK;
avail = gst_adapter_available (rtpmp2tpay->adapter);
rtpmp2tpay->duration = duration;
}
- /* get packet length of previous data and this new data,
- * payload length includes a 4 byte header */
- packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
+ /* get packet length of previous data and this new data */
+ packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
- /* if this buffer is going to overflow the packet, flush what we
- * have. */
- if (gst_basertppayload_is_filled (basepayload,
+ /* if this buffer is going to overflow the packet, flush what we have,
+ * or if upstream is handing us several packets, to keep latency low */
+ if (!size || gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmp2tpay->duration + duration)) {
ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
rtpmp2tpay->first_ts = timestamp;
}
/* copy buffer to adapter */
- gst_adapter_push (rtpmp2tpay->adapter, buffer);
+ if (buffer) {
+ gst_adapter_push (rtpmp2tpay->adapter, buffer);
+ buffer = NULL;
+ }
+
+ if (size >= (188 * 2)) {
+ size = 0;
+ goto again;
+ }
return ret;
projects / platform / upstream / gst-plugins-good.git / blobdiff