gstreamer/gst-libs/gst/rtp/gstbasertpdepayload.c
Wim Taymans 56e39e7c1c gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-04 16:18:48 +00:00

545 lines
15 KiB
C

/* 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
{
guint64 clock_base;
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
GstClockTime exttimestamp;
gboolean discont;
};
/* 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 void gst_base_rtp_depayload_set_gst_timestamp
(GstBaseRTPDepayload * filter, guint32 timestamp, GstBuffer * buf);
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.
*/
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,
DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE));
gstelement_class->change_state = gst_base_rtp_depayload_change_state;
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)
{
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;
guint val;
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 */
if (gst_structure_get_uint (caps_struct, "clock-base", &val))
priv->clock_base = val;
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;
priv->exttimestamp = -1;
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;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
if (filter->clock_rate == 0)
goto not_configured;
priv = filter->priv;
priv->discont = GST_BUFFER_IS_DISCONT (in);
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
/* let's send it out to processing */
out_buf = bclass->process (filter, in);
if (out_buf) {
guint32 timestamp;
timestamp = gst_rtp_buffer_get_timestamp (in);
/* 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 (in);
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;
}
}
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:
{
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.");
/* fallthrough */
}
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_push_full (GstBaseRTPDepayload * filter,
gboolean do_ts, guint32 timestamp, GstBuffer * out_buf)
{
GstFlowReturn ret;
GstCaps *srccaps;
GstBaseRTPDepayloadClass *bclass;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
/* 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);
if (priv->discont) {
GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* push it */
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 void
gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
guint32 timestamp, GstBuffer * buf)
{
GstClockTime ts, exttimestamp;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
/* no clock-base set, take first timestamp as base */
if (priv->clock_base == -1)
priv->clock_base = timestamp;
/* get extended timestamp */
exttimestamp = gst_rtp_buffer_ext_timestamp (&priv->exttimestamp, timestamp);
/* subtract clock-base to get a 0 based timestamp. Make sure we don't go
* negative. */
if (exttimestamp > priv->clock_base)
exttimestamp -= priv->clock_base;
else
exttimestamp = 0;
/* rtp timestamps are based on the clock_rate
* gst timesamps are in nanoseconds */
ts = gst_util_uint64_scale_int (exttimestamp, GST_SECOND, filter->clock_rate);
GST_DEBUG_OBJECT (filter,
"timestamp: %u, exttimestamp %" G_GUINT64_FORMAT ", clockrate : %u",
timestamp, exttimestamp, filter->clock_rate);
GST_DEBUG_OBJECT (filter, "RTP: %u, GST: %" GST_TIME_FORMAT ", ts %"
GST_TIME_FORMAT, timestamp, GST_TIME_ARGS (ts), GST_TIME_ARGS (ts));
GST_BUFFER_TIMESTAMP (buf) = ts;
/* if this is the first buffer send a NEWSEGMENT */
if (filter->need_newsegment) {
GstEvent *event;
GstClockTime stop, position;
if (priv->npt_stop != -1)
stop = priv->npt_stop - priv->npt_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, 0, 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 GstStateChangeReturn
gst_base_rtp_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPDepayload *filter;
GstStateChangeReturn ret;
filter = GST_BASE_RTP_DEPAYLOAD (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
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->priv->exttimestamp = -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:
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;
}
}