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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 08:46:40 +00:00
stream-transport: add method to handle RTP/RTCP
Call new methods instead of poking into the structures directly.
This commit is contained in:
parent
883cf794e4
commit
75473fc88d
3 changed files with 53 additions and 4 deletions
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@ -170,3 +170,49 @@ gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
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gst_rtsp_transport_free (trans->transport);
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trans->transport = tr;
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}
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/**
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* gst_rtsp_stream_transport_send_rtp:
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* @trans: a #GstRTSPStreamTransport
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* @buffer: a #GstBuffer
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*
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* Send @buffer to the installed RTP callback for @trans.
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*
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* Returns: %TRUE on success
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*/
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gboolean
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gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
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GstBuffer * buffer)
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{
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gboolean res = FALSE;
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if (trans->send_rtp)
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res =
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trans->send_rtp (buffer, trans->transport->interleaved.min,
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trans->user_data);
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return res;
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}
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/**
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* gst_rtsp_stream_transport_send_rtcp:
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* @trans: a #GstRTSPStreamTransport
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* @buffer: a #GstBuffer
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*
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* Send @buffer to the installed RTCP callback for @trans.
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*
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* Returns: %TRUE on success
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*/
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gboolean
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gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
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GstBuffer * buffer)
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{
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gboolean res = FALSE;
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if (trans->send_rtcp)
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res =
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trans->send_rtcp (buffer, trans->transport->interleaved.max,
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trans->user_data);
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return res;
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}
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@ -105,6 +105,11 @@ void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamT
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gpointer user_data,
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GDestroyNotify notify);
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gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
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GstBuffer *buffer);
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gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
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GstBuffer *buffer);
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G_END_DECLS
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#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
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@ -524,11 +524,9 @@ handle_new_sample (GstAppSink * sink, gpointer user_data)
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GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
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if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
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if (tr->send_rtp)
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tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
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gst_rtsp_stream_transport_send_rtp (tr, buffer);
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} else {
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if (tr->send_rtcp)
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tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
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gst_rtsp_stream_transport_send_rtcp (tr, buffer);
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}
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}
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gst_sample_unref (sample);
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