mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ffa52e2eac
Original commit message from CVS: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/gstaudiofilter.h: * gst-libs/gst/interfaces/mixer.h: * gst-libs/gst/rtp/gstbasertpdepayload.c: * gst-libs/gst/rtp/gstbasertpdepayload.h: * gst-libs/gst/sdp/gstsdpmessage.c: Fix the docs according to what gtk-doc complained about.
523 lines
15 KiB
C
523 lines
15 KiB
C
/* GStreamer
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* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
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* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstbasertpdepayload
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* @short_description: Base class for RTP depayloader
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*
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* <refsect2>
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* <para>
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* Provides a base class for RTP depayloaders
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* </para>
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* </refsect2>
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*/
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#include "gstbasertpdepayload.h"
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#ifdef GST_DISABLE_DEPRECATED
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#define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock))
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#define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock))
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#define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock))
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#define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock))
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#else
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/* otherwise it's already been defined in the header (FIXME 0.11)*/
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#endif
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GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
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#define GST_CAT_DEFAULT (basertpdepayload_debug)
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#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
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struct _GstBaseRTPDepayloadPrivate
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{
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GstClockTime npt_start;
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GstClockTime npt_stop;
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gdouble play_speed;
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gdouble play_scale;
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gboolean discont;
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GstClockTime timestamp;
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GstClockTime duration;
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};
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_QUEUE_DELAY 0
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enum
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{
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PROP_0,
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PROP_QUEUE_DELAY
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};
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static void gst_base_rtp_depayload_finalize (GObject * object);
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static void gst_base_rtp_depayload_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_base_rtp_depayload_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
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GstBuffer * in);
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static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
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GstEvent * event);
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static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
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element, GstStateChange transition);
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static void gst_base_rtp_depayload_set_gst_timestamp
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(GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
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GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
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GST_TYPE_ELEMENT);
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static void
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gst_base_rtp_depayload_base_init (gpointer klass)
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{
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/*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
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}
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static void
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gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
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gobject_class->finalize = gst_base_rtp_depayload_finalize;
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gobject_class->set_property = gst_base_rtp_depayload_set_property;
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gobject_class->get_property = gst_base_rtp_depayload_get_property;
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/**
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* GstBaseRTPDepayload::queue-delay
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*
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* Control the amount of packets to buffer.
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*
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* Deprecated: Use a jitterbuffer or RTP session manager to delay packet
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* playback. This property has no effect anymore since 0.10.15.
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*/
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#ifndef GST_REMOVE_DEPRECATED
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g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
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g_param_spec_uint ("queue-delay", "Queue Delay",
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"Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
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DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE));
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#endif
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gstelement_class->change_state = gst_base_rtp_depayload_change_state;
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klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
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GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
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"Base class for RTP Depayloaders");
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}
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static void
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gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
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GstBaseRTPDepayloadClass * klass)
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{
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GstPadTemplate *pad_template;
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GstBaseRTPDepayloadPrivate *priv;
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priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
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filter->priv = priv;
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GST_DEBUG_OBJECT (filter, "init");
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
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g_return_if_fail (pad_template != NULL);
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filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
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gst_pad_set_setcaps_function (filter->sinkpad,
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gst_base_rtp_depayload_setcaps);
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gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
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gst_pad_set_event_function (filter->sinkpad,
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gst_base_rtp_depayload_handle_sink_event);
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gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
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g_return_if_fail (pad_template != NULL);
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filter->srcpad = gst_pad_new_from_template (pad_template, "src");
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gst_pad_use_fixed_caps (filter->srcpad);
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gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
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filter->queue = g_queue_new ();
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filter->queue_delay = DEFAULT_QUEUE_DELAY;
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gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
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}
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static void
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gst_base_rtp_depayload_finalize (GObject * object)
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{
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GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
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g_queue_free (filter->queue);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstBaseRTPDepayload *filter;
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GstBaseRTPDepayloadClass *bclass;
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GstBaseRTPDepayloadPrivate *priv;
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gboolean res;
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GstStructure *caps_struct;
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const GValue *value;
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filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
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priv = filter->priv;
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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GST_DEBUG_OBJECT (filter, "Set caps");
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caps_struct = gst_caps_get_structure (caps, 0);
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/* get other values for newsegment */
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value = gst_structure_get_value (caps_struct, "npt-start");
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if (value && G_VALUE_HOLDS_UINT64 (value))
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priv->npt_start = g_value_get_uint64 (value);
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else
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priv->npt_start = 0;
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GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
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value = gst_structure_get_value (caps_struct, "npt-stop");
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if (value && G_VALUE_HOLDS_UINT64 (value))
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priv->npt_stop = g_value_get_uint64 (value);
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else
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priv->npt_stop = -1;
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GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
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value = gst_structure_get_value (caps_struct, "play-speed");
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if (value && G_VALUE_HOLDS_DOUBLE (value))
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priv->play_speed = g_value_get_double (value);
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else
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priv->play_speed = 1.0;
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value = gst_structure_get_value (caps_struct, "play-scale");
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if (value && G_VALUE_HOLDS_DOUBLE (value))
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priv->play_scale = g_value_get_double (value);
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else
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priv->play_scale = 1.0;
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if (bclass->set_caps)
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res = bclass->set_caps (filter, caps);
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else
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res = TRUE;
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gst_object_unref (filter);
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return res;
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}
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static GstFlowReturn
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gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
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{
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GstBaseRTPDepayload *filter;
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GstBaseRTPDepayloadPrivate *priv;
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GstBaseRTPDepayloadClass *bclass;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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GstClockTime timestamp;
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filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
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priv = filter->priv;
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priv->discont = GST_BUFFER_IS_DISCONT (in);
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/* convert to running_time and save the timestamp, this is the timestamp
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* we put on outgoing buffers. */
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timestamp = GST_BUFFER_TIMESTAMP (in);
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timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
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timestamp);
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priv->timestamp = timestamp;
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priv->duration = GST_BUFFER_DURATION (in);
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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/* let's send it out to processing */
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out_buf = bclass->process (filter, in);
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if (out_buf) {
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guint32 rtptime;
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rtptime = gst_rtp_buffer_get_timestamp (in);
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/* we pass rtptime as backward compatibility, in reality, the incomming
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* buffer timestamp is always applied to the outgoing packet. */
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ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
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}
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gst_buffer_unref (in);
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return ret;
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}
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static gboolean
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gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
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{
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GstBaseRTPDepayload *filter;
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gboolean res = TRUE;
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filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_STOP:
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res = gst_pad_push_event (filter->srcpad, event);
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gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
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filter->need_newsegment = TRUE;
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break;
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case GST_EVENT_NEWSEGMENT:
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{
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gboolean update;
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gdouble rate;
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GstFormat fmt;
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gint64 start, stop, position;
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gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
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&position);
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gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
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start, stop, position);
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/* don't pass the event downstream, we generate our own segment including
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* the NTP time and other things we receive in caps */
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gst_event_unref (event);
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break;
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}
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default:
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/* pass other events forward */
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res = gst_pad_push_event (filter->srcpad, event);
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break;
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}
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return res;
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}
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static GstFlowReturn
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gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
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gboolean do_ts, guint32 rtptime, GstBuffer * out_buf)
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{
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GstFlowReturn ret;
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GstCaps *srccaps;
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GstBaseRTPDepayloadClass *bclass;
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GstBaseRTPDepayloadPrivate *priv;
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priv = filter->priv;
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/* set the caps if any */
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srccaps = GST_PAD_CAPS (filter->srcpad);
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if (srccaps)
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gst_buffer_set_caps (out_buf, srccaps);
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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/* set the timestamp if we must and can */
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if (bclass->set_gst_timestamp && do_ts)
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bclass->set_gst_timestamp (filter, rtptime, out_buf);
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if (priv->discont) {
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GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
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priv->discont = FALSE;
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}
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/* push it */
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GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
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GST_BUFFER_SIZE (out_buf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)));
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ret = gst_pad_push (filter->srcpad, out_buf);
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GST_LOG_OBJECT (filter, "Pushed buffer: %s", gst_flow_get_name (ret));
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return ret;
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}
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/**
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* gst_base_rtp_depayload_push_ts:
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* @filter: a #GstBaseRTPDepayload
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* @timestamp: an RTP timestamp to apply
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* @out_buf: a #GstBuffer
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*
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* Push @out_buf to the peer of @filter. This function takes ownership of
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* @out_buf.
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*
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* Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp
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* on the outgoing buffer, using the configured clock_rate to convert the
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* timestamp to a valid GStreamer clock time.
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*
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* Returns: a #GstFlowReturn.
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*/
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GstFlowReturn
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gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
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GstBuffer * out_buf)
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{
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return gst_base_rtp_depayload_push_full (filter, TRUE, timestamp, out_buf);
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}
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/**
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* gst_base_rtp_depayload_push:
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* @filter: a #GstBaseRTPDepayload
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* @out_buf: a #GstBuffer
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*
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* Push @out_buf to the peer of @filter. This function takes ownership of
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* @out_buf.
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*
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* Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
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* any timestamp on the outgoing buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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GstFlowReturn
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gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
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{
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return gst_base_rtp_depayload_push_full (filter, FALSE, 0, out_buf);
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}
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static void
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gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
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guint32 rtptime, GstBuffer * buf)
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{
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GstBaseRTPDepayloadPrivate *priv;
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GstClockTime timestamp, duration;
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priv = filter->priv;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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duration = GST_BUFFER_DURATION (buf);
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/* apply last incomming timestamp and duration to outgoing buffer if
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* not otherwise set. */
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if (!GST_CLOCK_TIME_IS_VALID (timestamp))
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GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
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if (!GST_CLOCK_TIME_IS_VALID (duration))
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GST_BUFFER_DURATION (buf) = priv->duration;
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/* if this is the first buffer send a NEWSEGMENT */
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if (filter->need_newsegment) {
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GstEvent *event;
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GstClockTime stop, position;
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if (priv->npt_stop != -1)
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stop = priv->npt_stop - priv->npt_start;
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else
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stop = -1;
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position = priv->npt_start;
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event =
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gst_event_new_new_segment_full (FALSE, priv->play_speed,
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priv->play_scale, GST_FORMAT_TIME, 0, stop, position);
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gst_pad_push_event (filter->srcpad, event);
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filter->need_newsegment = FALSE;
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GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
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}
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}
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static GstStateChangeReturn
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gst_base_rtp_depayload_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstBaseRTPDepayload *filter;
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GstBaseRTPDepayloadPrivate *priv;
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GstStateChangeReturn ret;
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filter = GST_BASE_RTP_DEPAYLOAD (element);
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priv = filter->priv;
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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filter->need_newsegment = TRUE;
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priv->npt_start = 0;
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priv->npt_stop = -1;
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priv->play_speed = 1.0;
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|
priv->play_scale = 1.0;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseRTPDepayload *filter;
|
|
|
|
filter = GST_BASE_RTP_DEPAYLOAD (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUEUE_DELAY:
|
|
filter->queue_delay = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseRTPDepayload *filter;
|
|
|
|
filter = GST_BASE_RTP_DEPAYLOAD (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUEUE_DELAY:
|
|
g_value_set_uint (value, filter->queue_delay);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|