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"
#ifdef GST_DISABLE_DEPRECATED
#define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock))
#define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock))
#define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock))
#define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock))
#else
/* otherwise it's already been defined in the header (FIXME 0.11)*/
#endif
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
{
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
gboolean discont;
GstClockTime timestamp;
GstClockTime duration;
guint32 next_seqnum;
gboolean negotiated;
};
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_QUEUE_DELAY 0
enum
{
PROP_0,
PROP_QUEUE_DELAY,
PROP_LAST
};
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 void gst_base_rtp_depayload_set_gst_timestamp
(GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload *
filter, GstEvent * event);
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;
/**
* GstBaseRTPDepayload::queue-delay
*
* Control the amount of packets to buffer.
*
* Deprecated: Use a jitterbuffer or RTP session manager to delay packet
* playback. This property has no effect anymore since 0.10.15.
*/
#ifndef GST_REMOVE_DEPRECATED
g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
g_param_spec_uint ("queue-delay", "Queue Delay",
"Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
gstelement_class->change_state = gst_base_rtp_depayload_change_state;
klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
klass->packet_lost = gst_base_rtp_depayload_packet_lost;
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;
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
}
static void
gst_base_rtp_depayload_finalize (GObject * object)
{
GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
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 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;
GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
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;
GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
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);
if (!res) {
GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
caps);
}
} else {
res = TRUE;
}
priv->negotiated = res;
gst_object_unref (filter);
return res;
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime timestamp;
guint16 seqnum;
guint32 rtptime;
gboolean reset_seq, discont;
gint gap;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
priv = filter->priv;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
/* we must validate, it's possible that this element is plugged right after a
* network receiver and we don't want to operate on invalid data */
if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
goto invalid_buffer;
if (!priv->discont)
priv->discont = GST_BUFFER_IS_DISCONT (in);
timestamp = GST_BUFFER_TIMESTAMP (in);
/* convert to running_time and save the timestamp, this is the timestamp
* we put on outgoing buffers. */
timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
timestamp);
priv->timestamp = timestamp;
priv->duration = GST_BUFFER_DURATION (in);
seqnum = gst_rtp_buffer_get_seq (in);
rtptime = gst_rtp_buffer_get_timestamp (in);
reset_seq = TRUE;
discont = FALSE;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
GST_TIME_ARGS (timestamp));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are striclty increasing, dropping anything that is out of the ordinary. We
* can only do this when the next_seqnum is known. */
if (G_LIKELY (priv->next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
/* if we have no gap, all is fine */
if (G_UNLIKELY (gap != 0)) {
GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
priv->next_seqnum, gap);
if (gap < 0) {
/* seqnum > next_seqnum, we are missing some packets, this is always a
* DISCONT. */
GST_LOG_OBJECT (filter, "%d missing packets", gap);
discont = TRUE;
} else {
/* seqnum < next_seqnum, we have seen this packet before or the sender
* could be restarted. If the packet is not too old, we throw it away as
* a duplicate, otherwise we mark discont and continue. 100 misordered
* packets is a good threshold. See also RFC 4737. */
if (gap < 100)
goto dropping;
GST_LOG_OBJECT (filter,
"%d > 100, packet too old, sender likely restarted", gap);
discont = TRUE;
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
if (G_UNLIKELY (discont && !priv->discont)) {
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
/* we detected a seqnum discont but the buffer was not flagged with a discont,
* set the discont flag so that the subclass can throw away old data. */
priv->discont = TRUE;
in = gst_buffer_make_metadata_writable (in);
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (G_UNLIKELY (bclass->process == NULL))
goto no_process;
/* let's send it out to processing */
out_buf = bclass->process (filter, in);
if (out_buf) {
/* we pass rtptime as backward compatibility, in reality, the incomming
* buffer timestamp is always applied to the outgoing packet. */
ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
}
gst_buffer_unref (in);
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
if (GST_BUFFER_CAPS (in) == NULL) {
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
("No RTP format was negotiated."),
("Input buffers need to have RTP caps set on them. This is usually "
"achieved by setting the 'caps' property of the upstream source "
"element (often udpsrc or appsrc), or by putting a capsfilter "
"element before the depayloader and setting the 'caps' property "
"on that. Also see http://cgit.freedesktop.org/gstreamer/"
"gst-plugins-good/tree/gst/rtp/README"));
} else {
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
("No RTP format was negotiated."),
("RTP caps on input buffer were rejected, most likely because they "
"were incomplete or contained wrong values. Check the debug log "
"for more information."));
}
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (in);
return GST_FLOW_OK;
}
dropping:
{
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPDepayload *filter;
gboolean res = TRUE;
2009-08-31 18:31:56 +00:00
gboolean forward = TRUE;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
filter->need_newsegment = TRUE;
filter->priv->next_seqnum = -1;
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate;
GstFormat fmt;
gint64 start, stop, position;
gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
&position);
gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
start, stop, position);
/* don't pass the event downstream, we generate our own segment including
* the NTP time and other things we receive in caps */
2009-08-31 18:31:56 +00:00
forward = FALSE;
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
GstBaseRTPDepayloadClass *bclass;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (gst_event_has_name (event, "GstRTPPacketLost")) {
/* we get this event from the jitterbuffer when it considers a packet as
* being lost. We send it to our packet_lost vmethod. The default
* implementation will make time progress by pushing out a NEWSEGMENT
* update event. Subclasses can override and to one of the following:
* - Adjust timestamp/duration to something more accurate before
* calling the parent (default) packet_lost method.
* - do some more advanced error concealing on the already received
* (fragmented) packets.
* - ignore the packet lost.
*/
if (bclass->packet_lost)
res = bclass->packet_lost (filter, event);
2009-08-31 18:31:56 +00:00
forward = FALSE;
}
break;
}
default:
break;
}
2009-08-31 18:31:56 +00:00
if (forward)
res = gst_pad_push_event (filter->srcpad, event);
else
gst_event_unref (event);
return res;
}
static GstEvent *
create_segment_event (GstBaseRTPDepayload * filter, gboolean update,
GstClockTime position)
{
GstEvent *event;
GstClockTime stop;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
if (priv->npt_stop != -1)
stop = priv->npt_stop - priv->npt_start;
else
stop = -1;
event = gst_event_new_new_segment_full (update, priv->play_speed,
priv->play_scale, GST_FORMAT_TIME, position, stop,
position + priv->npt_start);
return event;
}
static GstFlowReturn
gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
gboolean do_ts, guint32 rtptime, GstBuffer * out_buf)
{
GstFlowReturn ret;
GstCaps *srccaps;
GstBaseRTPDepayloadClass *bclass;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
/* almost certainly required */
out_buf = gst_buffer_make_metadata_writable (out_buf);
/* set the caps if any */
srccaps = GST_PAD_CAPS (filter->srcpad);
if (G_LIKELY (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, rtptime, out_buf);
/* if this is the first buffer send a NEWSEGMENT */
if (G_UNLIKELY (filter->need_newsegment)) {
GstEvent *event;
event = create_segment_event (filter, FALSE, 0);
gst_pad_push_event (filter->srcpad, event);
filter->need_newsegment = FALSE;
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
}
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (filter, "Marking DISCONT on output buffer");
GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* 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);
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.
*
2010-03-08 11:11:01 +00:00
* Unlike gst_base_rtp_depayload_push(), this function will by default apply
* the last incomming timestamp on the outgoing buffer when it didn't have a
* timestamp already. The set_get_timestamp vmethod can be overwritten to change
* this behaviour (and take, for example, @timestamp into account).
*
* 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
2010-03-08 11:11:01 +00:00
* any timestamp on the outgoing buffer. Subclasses should therefore timestamp
* outgoing buffers themselves.
*
* 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);
}
/* convert the PacketLost event form a jitterbuffer to a segment update.
* subclasses can override this. */
static gboolean
gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter,
GstEvent * event)
{
GstClockTime timestamp, duration, position;
GstEvent *sevent;
const GstStructure *s;
s = gst_event_get_structure (event);
/* first start by parsing the timestamp and duration */
timestamp = -1;
duration = -1;
gst_structure_get_clock_time (s, "timestamp", &timestamp);
gst_structure_get_clock_time (s, "duration", &duration);
position = timestamp;
if (duration != -1)
position += duration;
/* update the current segment with the elapsed time */
sevent = create_segment_event (filter, TRUE, position);
return gst_pad_push_event (filter->srcpad, sevent);
}
static void
gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
guint32 rtptime, GstBuffer * buf)
{
GstBaseRTPDepayloadPrivate *priv;
GstClockTime timestamp, duration;
priv = filter->priv;
timestamp = GST_BUFFER_TIMESTAMP (buf);
duration = GST_BUFFER_DURATION (buf);
/* apply last incomming timestamp and duration to outgoing buffer if
* not otherwise set. */
if (!GST_CLOCK_TIME_IS_VALID (timestamp))
GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
if (!GST_CLOCK_TIME_IS_VALID (duration))
GST_BUFFER_DURATION (buf) = priv->duration;
}
static GstStateChangeReturn
gst_base_rtp_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstStateChangeReturn ret;
filter = GST_BASE_RTP_DEPAYLOAD (element);
priv = filter->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->need_newsegment = TRUE;
priv->npt_start = 0;
priv->npt_stop = -1;
priv->play_speed = 1.0;
priv->play_scale = 1.0;
priv->next_seqnum = -1;
priv->negotiated = FALSE;
priv->discont = FALSE;
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;
}
}