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eb0272e210
While the suspend modes NONE and PAUSED provided a low startup latency for connecting clients they did not ensure that streams started on fresh data. With this property we can maintain the low startup latency of those suspend modes while also ensuring that a stream starts on a key unit. Furthermore, by modifying the value of a new property, ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of a certain age but discard it if too much time has passed and instead force a new keyunit. Fixes #2443 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
72 lines
3.4 KiB
C
72 lines
3.4 KiB
C
/* GStreamer
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* Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_RTSP_SERVER_INTERNAL_H__
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#define __GST_RTSP_SERVER_INTERNAL_H__
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#include <glib.h>
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G_BEGIN_DECLS
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#include "rtsp-stream-transport.h"
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/* Internal GstRTSPStreamTransport interface */
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typedef gboolean (*GstRTSPBackPressureFunc) (guint8 channel, gpointer user_data);
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gboolean gst_rtsp_stream_transport_backlog_push (GstRTSPStreamTransport *trans,
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GstBuffer *buffer,
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GstBufferList *buffer_list,
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gboolean is_rtp);
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gboolean gst_rtsp_stream_transport_backlog_pop (GstRTSPStreamTransport *trans,
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GstBuffer **buffer,
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GstBufferList **buffer_list,
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gboolean *is_rtp);
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gboolean gst_rtsp_stream_transport_backlog_peek_is_rtp (GstRTSPStreamTransport * trans);
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gboolean gst_rtsp_stream_transport_backlog_is_empty (GstRTSPStreamTransport *trans);
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void gst_rtsp_stream_transport_clear_backlog (GstRTSPStreamTransport * trans);
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void gst_rtsp_stream_transport_lock_backlog (GstRTSPStreamTransport * trans);
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void gst_rtsp_stream_transport_unlock_backlog (GstRTSPStreamTransport * trans);
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void gst_rtsp_stream_transport_set_back_pressure_callback (GstRTSPStreamTransport *trans,
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GstRTSPBackPressureFunc back_pressure_func,
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gpointer user_data,
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GDestroyNotify notify);
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gboolean gst_rtsp_stream_transport_check_back_pressure (GstRTSPStreamTransport *trans,
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gboolean is_rtp);
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gboolean gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream);
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void gst_rtsp_media_set_enable_rtcp (GstRTSPMedia *media, gboolean enable);
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void gst_rtsp_stream_set_enable_rtcp (GstRTSPStream *stream, gboolean enable);
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void gst_rtsp_stream_set_drop_delta_units (GstRTSPStream * stream, gboolean drop);
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gboolean gst_rtsp_stream_install_drop_probe (GstRTSPStream * stream);
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G_END_DECLS
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#endif /* __GST_RTSP_SERVER_INTERNAL_H__ */
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