static GstStateChangeReturn
gst_rtp_mpv_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRTPMPVPay *rtpmpvpay;
GstStateChangeReturn ret;
rtpmpvpay = GST_RTP_MPV_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rtp_mpv_pay_reset (rtpmpvpay);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_mpv_pay_reset (rtpmpvpay);
break;
default:
break;
}
return ret;
}
static void
gst_rtp_mpv_pay_finalize (GObject * object)
{
GstRTPMPVPay *rtpmpvpay;
rtpmpvpay = GST_RTP_MPV_PAY (object);
g_object_unref (rtpmpvpay->adapter);
rtpmpvpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstFlowReturn
gst_rtp_mpv_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPMPVPay *rtpmpvpay;
guint avail, packet_len;
GstClockTime timestamp, duration;
GstFlowReturn ret = GST_FLOW_OK;
rtpmpvpay = GST_RTP_MPV_PAY (basepayload);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (rtpmpvpay, "DISCONT");
gst_rtp_mpv_pay_reset (rtpmpvpay);
}
avail = gst_adapter_available (rtpmpvpay->adapter);
if (duration == -1)
duration = 0;
if (rtpmpvpay->first_ts == GST_CLOCK_TIME_NONE || avail == 0)
rtpmpvpay->first_ts = timestamp;
if (avail == 0) {
rtpmpvpay->duration = duration;
} else {
rtpmpvpay->duration += duration;
}
gst_adapter_push (rtpmpvpay->adapter, buffer);
avail = gst_adapter_available (rtpmpvpay->adapter);
/* get packet length of previous data and this new data,
* payload length includes a 4 byte MPEG video-specific header */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
GST_LOG_OBJECT (rtpmpvpay, "available %d, rtp packet length %d", avail,
packet_len);
if (gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmpvpay->duration)) {
ret = gst_rtp_mpv_pay_flush (rtpmpvpay);
} else {
rtpmpvpay->first_ts = timestamp;
}
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPMPVPay *rtpmpvpay;
guint size, avail, packet_len;
guint8 *data;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpmpvpay = GST_RTP_MPV_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
gst_adapter_push (rtpmpvpay->adapter, buffer);
avail = gst_adapter_available (rtpmpvpay->adapter);
/* Initialize new RTP payload */
if (avail == 0) {
rtpmpvpay->first_ts = timestamp;
rtpmpvpay->duration = duration;
}
/* get packet length of previous data and this new data,
* payload length includes a 4 byte MPEG video-specific header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmpvpay->duration + duration)) {
ret = gst_rtp_mpv_pay_flush (rtpmpvpay, timestamp, duration);
} else {
if (GST_CLOCK_TIME_IS_VALID (duration))
rtpmpvpay->duration += duration;
ret = GST_FLOW_OK;
}
return ret;
}
static gboolean
gst_rtp_mpv_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean ret;
GstRTPMPVPay *rtpmpvpay;
rtpmpvpay = GST_RTP_MPV_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* make sure we push the last packets in the adapter on EOS */
gst_rtp_mpv_pay_flush (rtpmpvpay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mpv_pay_reset (rtpmpvpay);
break;
default:
break;
}
ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return ret;
}
