gstreamer/gst-libs/gst/rtp/gstbasertpdepayload.c

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/* GStreamer
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* 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.
*/
/**
* SECTION:gstbasertpdepayload
* @short_description: Base class for RTP depayloader
*
* <refsect2>
* <para>
* Provides a base class for RTP depayloaders
* </para>
* </refsect2>
*/
#include "gstbasertpdepayload.h"
GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
#define GST_CAT_DEFAULT (basertpdepayload_debug)
#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
struct _GstBaseRTPDepayloadPrivate
{
guint64 clock_base;
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
};
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_QUEUE_DELAY 0
enum
{
PROP_0,
PROP_QUEUE_DELAY,
};
static void gst_base_rtp_depayload_finalize (GObject * object);
static void gst_base_rtp_depayload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_base_rtp_depayload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
GstBuffer * in);
static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
GstEvent * event);
static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
element, GstStateChange transition);
static GstFlowReturn gst_base_rtp_depayload_add_to_queue (GstBaseRTPDepayload *
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
filter, GstBuffer * in);
static GstFlowReturn gst_base_rtp_depayload_process (GstBaseRTPDepayload *
filter, GstBuffer * rtp_buf);
static void gst_base_rtp_depayload_set_gst_timestamp
(GstBaseRTPDepayload * filter, guint32 timestamp, GstBuffer * buf);
static void gst_base_rtp_depayload_wait (GstBaseRTPDepayload * filter,
GstClockTime time);
GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
GST_TYPE_ELEMENT);
static void
gst_base_rtp_depayload_base_init (gpointer klass)
{
/*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
}
static void
gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
gobject_class->finalize = gst_base_rtp_depayload_finalize;
gobject_class->set_property = gst_base_rtp_depayload_set_property;
gobject_class->get_property = gst_base_rtp_depayload_get_property;
g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
g_param_spec_uint ("queue_delay", "Queue Delay",
"Amount of ms to queue/buffer", 0, G_MAXUINT, DEFAULT_QUEUE_DELAY,
G_PARAM_READWRITE));
gstelement_class->change_state = gst_base_rtp_depayload_change_state;
klass->add_to_queue = gst_base_rtp_depayload_add_to_queue;
klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
"Base class for RTP Depayloaders");
}
static void
gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
GstBaseRTPDepayloadClass * klass)
{
GstPadTemplate *pad_template;
GstBaseRTPDepayloadPrivate *priv;
priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
filter->priv = priv;
GST_DEBUG_OBJECT (filter, "init");
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
g_return_if_fail (pad_template != NULL);
filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_setcaps_function (filter->sinkpad,
gst_base_rtp_depayload_setcaps);
gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
gst_pad_set_event_function (filter->sinkpad,
gst_base_rtp_depayload_handle_sink_event);
gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
g_return_if_fail (pad_template != NULL);
filter->srcpad = gst_pad_new_from_template (pad_template, "src");
gst_pad_use_fixed_caps (filter->srcpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
filter->queue = g_queue_new ();
filter->queue_delay = DEFAULT_QUEUE_DELAY;
}
static void
gst_base_rtp_depayload_finalize (GObject * object)
{
GstBuffer *buf;
GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
while ((buf = g_queue_pop_head (filter->queue)))
gst_buffer_unref (buf);
g_queue_free (filter->queue);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadClass *bclass;
GstBaseRTPDepayloadPrivate *priv;
gboolean res;
GstStructure *caps_struct;
const GValue *value;
filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
priv = filter->priv;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
GST_DEBUG_OBJECT (filter, "Set caps");
caps_struct = gst_caps_get_structure (caps, 0);
/* get clock base if any, we need this for the newsegment */
value = gst_structure_get_value (caps_struct, "clock-base");
if (value && G_VALUE_HOLDS_UINT (value))
priv->clock_base = g_value_get_uint (value);
else
priv->clock_base = -1;
/* get other values for newsegment */
value = gst_structure_get_value (caps_struct, "npt-start");
if (value && G_VALUE_HOLDS_UINT64 (value))
priv->npt_start = g_value_get_uint64 (value);
else
priv->npt_start = 0;
value = gst_structure_get_value (caps_struct, "npt-stop");
if (value && G_VALUE_HOLDS_UINT64 (value))
priv->npt_stop = g_value_get_uint64 (value);
else
priv->npt_stop = -1;
value = gst_structure_get_value (caps_struct, "play-speed");
if (value && G_VALUE_HOLDS_DOUBLE (value))
priv->play_speed = g_value_get_double (value);
else
priv->play_speed = 1.0;
value = gst_structure_get_value (caps_struct, "play-scale");
if (value && G_VALUE_HOLDS_DOUBLE (value))
priv->play_scale = g_value_get_double (value);
else
priv->play_scale = 1.0;
if (bclass->set_caps)
res = bclass->set_caps (filter, caps);
else
res = TRUE;
gst_object_unref (filter);
return res;
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
if (filter->clock_rate == 0)
goto not_configured;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (filter->queue_delay == 0) {
GST_DEBUG_OBJECT (filter, "Pushing directly!");
ret = gst_base_rtp_depayload_process (filter, in);
} else {
if (bclass->add_to_queue)
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
ret = bclass->add_to_queue (filter, in);
else
goto no_delay;
}
return ret;
/* ERRORS */
not_configured:
{
GST_ELEMENT_ERROR (filter, STREAM, FORMAT,
(NULL), ("no clock rate was specified, likely incomplete input caps"));
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
no_delay:
{
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED,
(NULL), ("This element cannot operate with delay"));
gst_buffer_unref (in);
return GST_FLOW_NOT_SUPPORTED;
}
}
static gboolean
gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPDepayload *filter =
GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
gboolean res = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
/* intercept NEWSEGMENT events only if the packet scheduler thread
is active */
if (filter->thread) {
GST_DEBUG_OBJECT (filter,
"Upstream sent a NEWSEGMENT, handle in worker thread.");
/* the worker thread will assign a new RTP-TS<->GST-TS mapping
* based on the next processed RTP packet */
filter->need_newsegment = TRUE;
gst_event_unref (event);
break;
} else {
GstFormat format;
gst_event_parse_new_segment (event, NULL, NULL, &format, NULL, NULL,
NULL);
if (format != GST_FORMAT_TIME)
goto wrong_format;
GST_DEBUG_OBJECT (filter,
"Upstream sent a NEWSEGMENT, passing through.");
}
/* note: pass through to default if no thread running */
}
default:
/* pass other events forward */
res = gst_pad_push_event (filter->srcpad, event);
break;
}
return res;
/* ERRORS */
wrong_format:
{
GST_DEBUG_OBJECT (filter,
"Upstream sent a NEWSEGMENT in wrong format, dropping.");
gst_event_unref (event);
return TRUE;
}
}
static GstFlowReturn
gst_base_rtp_depayload_add_to_queue (GstBaseRTPDepayload * filter,
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
GstBuffer * in)
{
GQueue *queue = filter->queue;
int i;
/* our first packet, just push it */
QUEUE_LOCK (filter);
if (g_queue_is_empty (queue)) {
g_queue_push_tail (queue, in);
QUEUE_UNLOCK (filter);
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
} else {
guint16 seqnum, queueseq;
guint32 timestamp;
seqnum = gst_rtp_buffer_get_seq (in);
queueseq = gst_rtp_buffer_get_seq (GST_BUFFER (g_queue_peek_head (queue)));
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
/* look for right place to insert it */
i = 0;
/* Check for seqnum wraparound.
* Seqnums in the lowest quadrant of the 0-65535 space are considered to
* be greater than seqnums in the highest quadrant of this space. */
while (seqnum > queueseq || (seqnum < 16384 && queueseq > 49150)) {
gpointer data;
i++;
data = g_queue_peek_nth (queue, i);
if (!data)
break;
queueseq = gst_rtp_buffer_get_seq (GST_BUFFER (data));
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
}
/* now insert it at that place */
g_queue_push_nth (queue, in, i);
QUEUE_UNLOCK (filter);
timestamp = gst_rtp_buffer_get_timestamp (in);
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
GST_DEBUG_OBJECT (filter,
"Packet added to queue %d at pos %d timestamp %u sn %d",
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
g_queue_get_length (queue), i, timestamp, seqnum);
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
gboolean do_ts, guint32 timestamp, GstBuffer * out_buf)
{
GstFlowReturn ret;
GstCaps *srccaps;
GstBaseRTPDepayloadClass *bclass;
/* set the caps if any */
srccaps = GST_PAD_CAPS (filter->srcpad);
if (srccaps)
gst_buffer_set_caps (out_buf, srccaps);
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
/* set the timestamp if we must and can */
if (bclass->set_gst_timestamp && do_ts)
bclass->set_gst_timestamp (filter, timestamp, out_buf);
/* push it */
Printf format fixes. Original commit message from CVS: * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_device_property_probe_get_values): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad), (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain): * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad): * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream), (gst_ogg_parse_chain): * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header): * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup), (gst_vorbis_enc_buffer_check_discontinuous): * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_handle_track_seek): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_push_full): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push): * gst/audioresample/resample.c: (resample_input_pushthrough): * gst/playback/gstplaybasebin.c: (queue_out_of_data): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients): * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset), (wavpack_type_find): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create): * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy), (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/volume.c: (GST_START_TEST): Printf format fixes.
2006-10-05 15:55:21 +00:00
GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (out_buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)));
ret = gst_pad_push (filter->srcpad, out_buf);
GST_LOG_OBJECT (filter, "Pushed buffer: %s", gst_flow_get_name (ret));
return ret;
}
/**
* gst_base_rtp_depayload_push_ts:
* @filter: a #GstBaseRTPDepayload
* @timestamp: an RTP timestamp to apply
* @out_buf: a #GstBuffer
*
* Push @out_buf to the peer of @filter. This function takes ownership of
* @out_buf.
*
* Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp
* on the outgoing buffer, using the configured clock_rate to convert the
* timestamp to a valid GStreamer clock time.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
GstBuffer * out_buf)
{
return gst_base_rtp_depayload_push_full (filter, TRUE, timestamp, out_buf);
}
/**
* gst_base_rtp_depayload_push:
* @filter: a #GstBaseRTPDepayload
* @out_buf: a #GstBuffer
*
* Push @out_buf to the peer of @filter. This function takes ownership of
* @out_buf.
*
* Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
* any timestamp on the outgoing buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
{
return gst_base_rtp_depayload_push_full (filter, FALSE, 0, out_buf);
}
static GstFlowReturn
gst_base_rtp_depayload_process (GstBaseRTPDepayload * filter,
GstBuffer * rtp_buf)
{
GstBaseRTPDepayloadClass *bclass;
GstBuffer *out_buf;
GstFlowReturn ret = GST_FLOW_OK;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
/* let's send it out to processing */
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
out_buf = bclass->process (filter, rtp_buf);
if (out_buf) {
guint32 timestamp = gst_rtp_buffer_get_timestamp (rtp_buf);
/* push buffer with timestamp
* We are assuming here that the timestamp of the last RTP buffer
* is the same as the timestamp wanted on the collector. If this is not a
* desired result, the process function should push itself with another
* timestamp and return NULL.
*/
ret = gst_base_rtp_depayload_push_ts (filter, timestamp, out_buf);
}
gst_buffer_unref (rtp_buf);
return ret;
}
static void
gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
guint32 timestamp, GstBuffer * buf)
{
GstClockTime ts, adjusted;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
/* no clock-base set, take first timestamp as base */
if (priv->clock_base == -1)
priv->clock_base = timestamp;
/* rtp timestamps are based on the clock_rate
* gst timesamps are in nanoseconds */
ts = gst_util_uint64_scale_int (timestamp, GST_SECOND, filter->clock_rate);
GST_DEBUG_OBJECT (filter, "ts : timestamp : %u, clockrate : %u",
timestamp, filter->clock_rate);
/* add delay to timestamp */
adjusted = ts + (filter->queue_delay * GST_MSECOND);
GST_DEBUG_OBJECT (filter, "RTP: %u, GST: %" GST_TIME_FORMAT ", adjusted %"
GST_TIME_FORMAT, timestamp, GST_TIME_ARGS (ts), GST_TIME_ARGS (adjusted));
GST_BUFFER_TIMESTAMP (buf) = adjusted;
/* if this is the first buf send a NEWSEGMENT */
if (filter->need_newsegment) {
GstEvent *event;
GstClockTime start, stop, position;
start = gst_util_uint64_scale_int (priv->clock_base, GST_SECOND,
filter->clock_rate);
if (priv->npt_stop != -1)
stop = priv->npt_stop - priv->npt_start + start;
else
stop = -1;
position = priv->npt_start;
event =
gst_event_new_new_segment_full (FALSE, priv->play_speed,
priv->play_scale, GST_FORMAT_TIME, start, stop, position);
gst_pad_push_event (filter->srcpad, event);
filter->need_newsegment = FALSE;
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
}
}
static void
gst_base_rtp_depayload_queue_release (GstBaseRTPDepayload * filter)
{
GQueue *queue = filter->queue;
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
guint32 headts, tailts;
GstBaseRTPDepayloadClass *bclass;
gfloat q_size_secs;
guint maxtsunits;
if (g_queue_is_empty (queue))
return;
/* if our queue is getting to big (more than RTP_QUEUEDELAY ms of data)
* release heading buffers
*/
/*GST_DEBUG_OBJECT (filter, "clockrate %d, queue_delay %d", filter->clock_rate,
filter->queue_delay); */
q_size_secs = (gfloat) filter->queue_delay / 1000;
maxtsunits = (gfloat) filter->clock_rate * q_size_secs;
QUEUE_LOCK (filter);
headts =
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_head (queue)));
tailts =
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_tail (queue)));
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
/*GST_DEBUG("maxtsunit is %u %u %u %u", maxtsunits, headts, tailts, headts - tailts); */
gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes. Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_queue_release): * gst-libs/gst/rtp/gstbasertpdepayload.h: Fix for RTPBuffer changes. * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data), (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data), (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len), (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len), (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data), (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len), (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version), (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding), (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to), (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension), (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc), (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc), (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker), (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type), (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq), (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp), (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len), (gst_rtpbuffer_get_payload): * gst-libs/gst/rtp/gstrtpbuffer.h: Don't subclass GstBuffer but add methods and helper functions to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
while (headts - tailts > maxtsunits) {
GST_DEBUG_OBJECT (filter, "Poping packet from queue");
if (bclass->process) {
GstBuffer *in = g_queue_pop_head (queue);
gst_base_rtp_depayload_process (filter, in);
}
headts =
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_head (queue)));
}
QUEUE_UNLOCK (filter);
}
static gpointer
gst_base_rtp_depayload_thread (GstBaseRTPDepayload * filter)
{
while (filter->thread_running) {
gst_base_rtp_depayload_queue_release (filter);
/* sleep for 5msec (XXX: 5msec is a value that works for audio and video,
* should be adjusted based on frequency of incoming packet,
* or by data comsumption rate of the sink (depends on how
* clock-drift compensation is implemented) */
gst_base_rtp_depayload_wait (filter, GST_MSECOND * 5);
}
return NULL;
}
static gboolean
gst_base_rtp_depayload_start_thread (GstBaseRTPDepayload * filter)
{
/* only launch the thread if processing is needed */
if (filter->queue_delay) {
GST_DEBUG_OBJECT (filter, "Starting queue release thread");
filter->thread_running = TRUE;
filter->thread =
g_thread_create ((GThreadFunc) gst_base_rtp_depayload_thread, filter,
TRUE, NULL);
GST_DEBUG_OBJECT (filter, "Started queue release thread");
}
return TRUE;
}
static gboolean
gst_base_rtp_depayload_stop_thread (GstBaseRTPDepayload * filter)
{
filter->thread_running = FALSE;
if (filter->thread) {
g_thread_join (filter->thread);
filter->thread = NULL;
}
QUEUE_LOCK_FREE (filter);
return TRUE;
}
static void
gst_base_rtp_depayload_wait (GstBaseRTPDepayload * filter, GstClockTime time)
{
GstClockID id;
GstClock *clock;
GstClockTime base;
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (time));
GST_OBJECT_LOCK (filter);
if ((clock = GST_ELEMENT_CLOCK (filter)) == NULL)
goto no_clock;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (filter);
base = gst_clock_get_time (clock);
id = gst_clock_new_single_shot_id (clock, base + time);
gst_object_unref (clock);
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
return;
no_clock:
{
GST_DEBUG_OBJECT (filter, "No clock given yet");
GST_OBJECT_UNLOCK (filter);
return;
}
}
static GstStateChangeReturn
gst_base_rtp_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPDepayload *filter;
GstStateChangeReturn ret;
filter = GST_BASE_RTP_DEPAYLOAD (element);
/* we disallow changing the state from the thread */
if (g_thread_self () == filter->thread)
goto wrong_thread;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_base_rtp_depayload_start_thread (filter))
goto start_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* clock_rate needs to be overwritten by child */
filter->clock_rate = 0;
filter->priv->clock_base = -1;
filter->need_newsegment = TRUE;
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:
gst_base_rtp_depayload_stop_thread (filter);
break;
default:
break;
}
return ret;
/* ERRORS */
wrong_thread:
{
GST_ELEMENT_ERROR (filter, CORE, STATE_CHANGE,
(NULL), ("cannot perform a state change from this thread"));
return GST_STATE_CHANGE_FAILURE;
}
start_failed:
{
/* start method should have posted an error message */
return GST_STATE_CHANGE_FAILURE;
}
}
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;
}
}