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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/rtp/gstrtpbasedepayload.c
Sebastian Dröge 5ec73afe75 rtpbasedepayload: Add "extensions" property for the currently enabled extensions 
This works the same way as the one from `rtpbasepayload`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6163>
2024-02-21 12:17:24 +00:00

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstrtpbasedepayload
* @title: GstRTPBaseDepayload
* @short_description: Base class for RTP depayloader
*
* Provides a base class for RTP depayloaders
*
* In order to handle RTP header extensions correctly if the
* depayloader aggregates multiple RTP packet payloads into one output
* buffer this class provides the function
* gst_rtp_base_depayload_set_aggregate_hdrext_enabled(). If the
* aggregation is enabled the virtual functions
* @GstRTPBaseDepayload.process or
* @GstRTPBaseDepayload.process_rtp_packet must tell the base class
* what happens to the current RTP packet. By default the base class
* assumes that the packet payload is used with the next output
* buffer.
*
* If the RTP packet will not be used with an output buffer
* gst_rtp_base_depayload_dropped() must be called. A typical
* situation would be if we are waiting for a keyframe.
*
* If the RTP packet will be used but not with the current output
* buffer but with the next one gst_rtp_base_depayload_delayed() must
* be called. This may happen if the current RTP packet signals the
* start of a new output buffer and the currently processed output
* buffer will be pushed first. The undelay happens implicitly once
* the current buffer has been pushed or
* gst_rtp_base_depayload_flush() has been called.
*
* If gst_rtp_base_depayload_flush() is called all RTP packets that
* have not been dropped since the last output buffer are dropped,
* e.g. if an output buffer is discarded due to malformed data. This
* may or may not include the current RTP packet depending on the 2nd
* parameter @keep_current.
*
* Be aware that in case gst_rtp_base_depayload_push_list() is used
* each buffer will see the same list of RTP header extensions.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpbasedepayload.h"
#include "gstrtpmeta.h"
#include "gstrtphdrext.h"
GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
static GstStaticCaps ntp_reference_timestamp_caps =
GST_STATIC_CAPS ("timestamp/x-ntp");
struct _GstRTPBaseDepayloadPrivate
{
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
guint clock_base;
gboolean onvif_mode;
gboolean discont;
GstClockTime pts;
GstClockTime dts;
GstClockTime duration;
GstClockTime ref_ts;
guint32 last_ssrc;
guint32 last_seqnum;
guint32 last_rtptime;
guint32 next_seqnum;
gint max_reorder;
gboolean auto_hdr_ext;
gboolean negotiated;
GstCaps *last_caps;
GstEvent *segment_event;
guint32 segment_seqnum; /* Note: this is a GstEvent seqnum */
gboolean source_info;
GstBuffer *input_buffer;
GstFlowReturn process_flow_ret;
/* array of GstRTPHeaderExtension's * */
GPtrArray *header_exts;
/* maintain buffer list for header extensions read() */
gboolean hdrext_aggregate;
gboolean hdrext_seen;
GstBufferList *hdrext_buffers;
GstBuffer *hdrext_delayed;
GstBuffer *hdrext_outbuf;
gboolean hdrext_read_result;
};
/* Filter signals and args */
enum
{
SIGNAL_0,
SIGNAL_REQUEST_EXTENSION,
SIGNAL_ADD_EXTENSION,
SIGNAL_CLEAR_EXTENSIONS,
LAST_SIGNAL
};
static guint gst_rtp_base_depayload_signals[LAST_SIGNAL] = { 0 };
#define DEFAULT_SOURCE_INFO FALSE
#define DEFAULT_MAX_REORDER 100
#define DEFAULT_AUTO_HEADER_EXTENSION TRUE
enum
{
PROP_0,
PROP_STATS,
PROP_SOURCE_INFO,
PROP_MAX_REORDER,
PROP_AUTO_HEADER_EXTENSION,
PROP_EXTENSIONS,
PROP_LAST
};
static GParamSpec *gst_rtp_base_depayload_extensions_pspec;
static void gst_rtp_base_depayload_finalize (GObject * object);
static void gst_rtp_base_depayload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_rtp_base_depayload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
GstObject * parent, GstBuffer * in);
static GstFlowReturn gst_rtp_base_depayload_chain_list (GstPad * pad,
GstObject * parent, GstBufferList * list);
static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
filter, GstEvent * event);
static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
filter, GstEvent * event);
static GstElementClass *parent_class = NULL;
static gint private_offset = 0;
static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
klass);
static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
GstRTPBaseDepayloadClass * klass);
static GstEvent *create_segment_event (GstRTPBaseDepayload * filter,
guint rtptime, GstClockTime position);
static void gst_rtp_base_depayload_add_extension (GstRTPBaseDepayload *
rtpbasepayload, GstRTPHeaderExtension * ext);
static void gst_rtp_base_depayload_clear_extensions (GstRTPBaseDepayload *
rtpbasepayload);
static gboolean gst_rtp_base_depayload_operate_hdrext_buffer (GstBuffer **
buffer, guint idx, gpointer depayloader);
static void gst_rtp_base_depayload_reset_hdrext_buffers (GstRTPBaseDepayload *
rtpbasepayload);
GType
gst_rtp_base_depayload_get_type (void)
{
static GType rtp_base_depayload_type = 0;
if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
static const GTypeInfo rtp_base_depayload_info = {
sizeof (GstRTPBaseDepayloadClass),
NULL,
NULL,
(GClassInitFunc) gst_rtp_base_depayload_class_init,
NULL,
NULL,
sizeof (GstRTPBaseDepayload),
0,
(GInstanceInitFunc) gst_rtp_base_depayload_init,
};
GType _type;
_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload",
&rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT);
private_offset =
g_type_add_instance_private (_type,
sizeof (GstRTPBaseDepayloadPrivate));
g_once_init_leave ((gsize *) & rtp_base_depayload_type, _type);
}
return rtp_base_depayload_type;
}
static inline GstRTPBaseDepayloadPrivate *
gst_rtp_base_depayload_get_instance_private (GstRTPBaseDepayload * self)
{
return (G_STRUCT_MEMBER_P (self, private_offset));
}
static GstRTPHeaderExtension *
gst_rtp_base_depayload_request_extension_default (GstRTPBaseDepayload *
depayload, guint ext_id, const gchar * uri)
{
GstRTPHeaderExtension *ext = NULL;
if (!depayload->priv->auto_hdr_ext)
return NULL;
ext = gst_rtp_header_extension_create_from_uri (uri);
if (ext) {
GST_DEBUG_OBJECT (depayload,
"Automatically enabled extension %s for uri \'%s\'",
GST_ELEMENT_NAME (ext), uri);
gst_rtp_header_extension_set_id (ext, ext_id);
} else {
GST_DEBUG_OBJECT (depayload,
"Didn't find any extension implementing uri \'%s\'", uri);
}
return ext;
}
static gboolean
extension_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data)
{
gpointer ext;
/* Call default handler if user callback didn't create the extension */
ext = g_value_get_object (handler_return);
if (!ext)
return TRUE;
g_value_set_object (return_accu, ext);
return FALSE;
}
static void
gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * 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);
if (private_offset != 0)
g_type_class_adjust_private_offset (klass, &private_offset);
gobject_class->finalize = gst_rtp_base_depayload_finalize;
gobject_class->set_property = gst_rtp_base_depayload_set_property;
gobject_class->get_property = gst_rtp_base_depayload_get_property;
/**
* GstRTPBaseDepayload:stats:
*
* Various depayloader statistics retrieved atomically (and are therefore
* synchroized with each other). This property return a GstStructure named
* application/x-rtp-depayload-stats containing the following fields relating to
* the last processed buffer and current state of the stream being depayloaded:
*
* * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream
* * `npt-start`: #G_TYPE_UINT64, time of playback start
* * `npt-stop`: #G_TYPE_UINT64, time of playback stop
* * `play-speed`: #G_TYPE_DOUBLE, the playback speed
* * `play-scale`: #G_TYPE_DOUBLE, the playback scale
* * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
* last DTS
* * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
* last PTS
* * `seqnum`: #G_TYPE_UINT, the last seen seqnum
* * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBaseDepayload:source-info:
*
* Add RTP source information found in RTP header as meta to output buffer.
*
* Since: 1.16
**/
g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
g_param_spec_boolean ("source-info", "RTP source information",
"Add RTP source information as buffer meta",
DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
/**
* GstRTPBaseDepayload:max-reorder:
*
* Max seqnum reorder before the sender is assumed to have restarted.
*
* When max-reorder is set to 0 all reordered/duplicate packets are
* considered coming from a restarted sender.
*
* Since: 1.18
**/
g_object_class_install_property (gobject_class, PROP_MAX_REORDER,
g_param_spec_int ("max-reorder", "Max Reorder",
"Max seqnum reorder before assuming sender has restarted",
0, G_MAXINT, DEFAULT_MAX_REORDER, G_PARAM_READWRITE));
/**
* GstRTPBaseDepayload:auto-header-extension:
*
* If enabled, the depayloader will automatically try to enable all the
* RTP header extensions provided in the sink caps, saving the application
* the need to handle these extensions manually using the
* GstRTPBaseDepayload::request-extension: signal.
*
* Since: 1.20
*/
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_AUTO_HEADER_EXTENSION, g_param_spec_boolean ("auto-header-extension",
"Automatic RTP header extension",
"Whether RTP header extensions should be automatically enabled, if an implementation is available",
DEFAULT_AUTO_HEADER_EXTENSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBaseDepayload::request-extension:
* @object: the #GstRTPBaseDepayload
* @ext_id: the extension id being requested
* @ext_uri: (nullable): the extension URI being requested
*
* The returned @ext must be configured with the correct @ext_id and with the
* necessary attributes as required by the extension implementation.
*
* Returns: (transfer full) (nullable): the #GstRTPHeaderExtension for @ext_id, or %NULL
*
* Since: 1.20
*/
gst_rtp_base_depayload_signals[SIGNAL_REQUEST_EXTENSION] =
g_signal_new_class_handler ("request-extension",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_CALLBACK (gst_rtp_base_depayload_request_extension_default),
extension_accumulator, NULL, NULL,
GST_TYPE_RTP_HEADER_EXTENSION, 2, G_TYPE_UINT, G_TYPE_STRING);
/**
* GstRTPBaseDepayload::add-extension:
* @object: the #GstRTPBaseDepayload
* @ext: (transfer full): the #GstRTPHeaderExtension
*
* Add @ext as an extension for reading part of an RTP header extension from
* incoming RTP packets.
*
* Since: 1.20
*/
gst_rtp_base_depayload_signals[SIGNAL_ADD_EXTENSION] =
g_signal_new_class_handler ("add-extension", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_rtp_base_depayload_add_extension), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTP_HEADER_EXTENSION);
/**
* GstRTPBaseDepayload::clear-extensions:
* @object: the #GstRTPBaseDepayload
*
* Clear all RTP header extensions used by this depayloader.
*
* Since: 1.20
*/
gst_rtp_base_depayload_signals[SIGNAL_CLEAR_EXTENSIONS] =
g_signal_new_class_handler ("clear-extensions", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_rtp_base_depayload_clear_extensions), NULL, NULL, NULL,
G_TYPE_NONE, 0);
gst_rtp_base_depayload_extensions_pspec = gst_param_spec_array ("extensions",
"RTP header extensions",
"A list of already enabled RTP header extensions",
g_param_spec_object ("extension", "RTP header extension",
"An already enabled RTP extension", GST_TYPE_RTP_HEADER_EXTENSION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS),
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
/**
* GstRTPBaseDepayload:extensions:
*
* A list of already enabled RTP header extensions. This may be useful for finding
* out which extensions are already enabled (with add-extension signal) and picking a non-conflicting
* ID for a new extension that needs to be added on top of the existing ones.
*
* Note that the value returned by reading this property is not dynamically updated when the set of
* enabled extensions changes by any of existing action signals. Rather, it represents the current state
* at the time the property is read.
*
* Dynamic updates of this property can be received by subscribing to its corresponding "notify" signal, i.e.
* "notify::extensions".
*
* Since: 1.24
*/
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_EXTENSIONS, gst_rtp_base_depayload_extensions_pspec);
gstelement_class->change_state = gst_rtp_base_depayload_change_state;
klass->packet_lost = gst_rtp_base_depayload_packet_lost;
klass->handle_event = gst_rtp_base_depayload_handle_event;
GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0,
"Base class for RTP Depayloaders");
}
static void
gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * klass)
{
GstPadTemplate *pad_template;
GstRTPBaseDepayloadPrivate *priv;
priv = gst_rtp_base_depayload_get_instance_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_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
gst_pad_set_chain_list_function (filter->sinkpad,
gst_rtp_base_depayload_chain_list);
gst_pad_set_event_function (filter->sinkpad,
gst_rtp_base_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);
priv->npt_start = 0;
priv->npt_stop = -1;
priv->play_speed = 1.0;
priv->play_scale = 1.0;
priv->clock_base = -1;
priv->onvif_mode = FALSE;
priv->dts = -1;
priv->pts = -1;
priv->duration = -1;
priv->ref_ts = -1;
priv->source_info = DEFAULT_SOURCE_INFO;
priv->max_reorder = DEFAULT_MAX_REORDER;
priv->auto_hdr_ext = DEFAULT_AUTO_HEADER_EXTENSION;
priv->hdrext_aggregate = FALSE;
priv->hdrext_seen = FALSE;
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
priv->header_exts =
g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref);
priv->hdrext_buffers = gst_buffer_list_new ();
}
static void
gst_rtp_base_depayload_finalize (GObject * object)
{
GstRTPBaseDepayload *rtpbasedepayload = GST_RTP_BASE_DEPAYLOAD (object);
GstRTPBaseDepayloadPrivate *priv = rtpbasedepayload->priv;
g_ptr_array_unref (rtpbasedepayload->priv->header_exts);
gst_clear_buffer_list (&rtpbasedepayload->priv->hdrext_buffers);
if (priv->hdrext_delayed)
gst_buffer_unref (priv->hdrext_delayed);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
add_and_ref_item (GstRTPHeaderExtension * ext, GPtrArray * ret)
{
g_ptr_array_add (ret, gst_object_ref (ext));
}
static void
remove_item_from (GstRTPHeaderExtension * ext, GPtrArray * ret)
{
g_ptr_array_remove_fast (ret, ext);
}
static void
add_item_to (GstRTPHeaderExtension * ext, GPtrArray * ret)
{
g_ptr_array_add (ret, ext);
}
static gboolean
gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayloadPrivate *priv;
gboolean res = TRUE;
GstStructure *caps_struct;
const GValue *value;
priv = filter->priv;
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
GST_DEBUG_OBJECT (filter, "Set caps %" GST_PTR_FORMAT, caps);
if (priv->last_caps) {
if (gst_caps_is_equal (priv->last_caps, caps)) {
res = TRUE;
goto caps_not_changed;
} else {
gst_caps_unref (priv->last_caps);
priv->last_caps = NULL;
}
}
caps_struct = gst_caps_get_structure (caps, 0);
value = gst_structure_get_value (caps_struct, "onvif-mode");
if (value && G_VALUE_HOLDS_BOOLEAN (value))
priv->onvif_mode = g_value_get_boolean (value);
else
priv->onvif_mode = FALSE;
GST_DEBUG_OBJECT (filter, "Onvif mode: %d", priv->onvif_mode);
if (priv->onvif_mode)
filter->need_newsegment = FALSE;
/* 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;
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;
{
/* ensure we have header extension implementations for the list in the
* caps */
guint i, j, n_fields = gst_structure_n_fields (caps_struct);
GPtrArray *header_exts = g_ptr_array_new_with_free_func (gst_object_unref);
GPtrArray *to_add = g_ptr_array_new ();
GPtrArray *to_remove = g_ptr_array_new ();
GST_OBJECT_LOCK (filter);
g_ptr_array_foreach (filter->priv->header_exts,
(GFunc) add_and_ref_item, header_exts);
GST_OBJECT_UNLOCK (filter);
for (i = 0; i < n_fields; i++) {
const gchar *field_name = gst_structure_nth_field_name (caps_struct, i);
if (g_str_has_prefix (field_name, "extmap-")) {
const GValue *val;
const gchar *uri = NULL;
gchar *nptr;
guint ext_id;
GstRTPHeaderExtension *ext = NULL;
errno = 0;
ext_id = g_ascii_strtoull (&field_name[strlen ("extmap-")], &nptr, 10);
if (errno != 0 || (ext_id == 0 && field_name == nptr)) {
GST_WARNING_OBJECT (filter, "could not parse id from %s", field_name);
res = FALSE;
goto ext_out;
}
val = gst_structure_get_value (caps_struct, field_name);
if (G_VALUE_HOLDS_STRING (val)) {
uri = g_value_get_string (val);
} else if (GST_VALUE_HOLDS_ARRAY (val)) {
/* the uri is the second value in the array */
const GValue *str = gst_value_array_get_value (val, 1);
if (G_VALUE_HOLDS_STRING (str)) {
uri = g_value_get_string (str);
}
}
if (!uri) {
GST_WARNING_OBJECT (filter, "could not get extmap uri for "
"field %s", field_name);
res = FALSE;
goto ext_out;
}
/* try to find if this extension mapping already exists */
for (j = 0; j < header_exts->len; j++) {
ext = g_ptr_array_index (header_exts, j);
if (gst_rtp_header_extension_get_id (ext) == ext_id) {
if (g_strcmp0 (uri, gst_rtp_header_extension_get_uri (ext)) == 0) {
/* still matching, we're good, set attributes from caps in case
* the caps have changed */
if (!gst_rtp_header_extension_set_attributes_from_caps (ext,
caps)) {
GST_WARNING_OBJECT (filter,
"Failed to configure rtp header " "extension %"
GST_PTR_FORMAT " attributes from caps %" GST_PTR_FORMAT,
ext, caps);
res = FALSE;
goto ext_out;
}
break;
} else {
GST_DEBUG_OBJECT (filter, "extension id %u"
"was replaced with a different extension uri "
"original:\'%s' vs \'%s\'", ext_id,
gst_rtp_header_extension_get_uri (ext), uri);
g_ptr_array_add (to_remove, ext);
ext = NULL;
break;
}
} else {
ext = NULL;
}
}
/* if no extension, attempt to request one */
if (!ext) {
GST_DEBUG_OBJECT (filter, "requesting extension for id %u"
" and uri %s", ext_id, uri);
g_signal_emit (filter,
gst_rtp_base_depayload_signals[SIGNAL_REQUEST_EXTENSION], 0,
ext_id, uri, &ext);
GST_DEBUG_OBJECT (filter, "request returned extension %p \'%s\' "
"for id %u and uri %s", ext,
ext ? GST_OBJECT_NAME (ext) : "", ext_id, uri);
/* We require the caller to set the appropriate extension if it's required */
if (ext && gst_rtp_header_extension_get_id (ext) != ext_id) {
g_warning ("\'request-extension\' signal provided an rtp header "
"extension for uri \'%s\' that does not match the requested "
"extension id %u", uri, ext_id);
gst_clear_object (&ext);
}
if (ext && !gst_rtp_header_extension_set_attributes_from_caps (ext,
caps)) {
GST_WARNING_OBJECT (filter,
"Failed to configure rtp header " "extension %"
GST_PTR_FORMAT " attributes from caps %" GST_PTR_FORMAT,
ext, caps);
res = FALSE;
g_clear_object (&ext);
goto ext_out;
}
if (ext)
g_ptr_array_add (to_add, ext);
}
}
}
/* Note: we intentionally don't remove extensions that are not listed
* in caps */
GST_OBJECT_LOCK (filter);
g_ptr_array_foreach (to_remove, (GFunc) remove_item_from,
filter->priv->header_exts);
g_ptr_array_foreach (to_add, (GFunc) add_item_to,
filter->priv->header_exts);
GST_OBJECT_UNLOCK (filter);
g_object_notify_by_pspec (G_OBJECT (filter),
gst_rtp_base_depayload_extensions_pspec);
ext_out:
g_ptr_array_unref (to_add);
g_ptr_array_unref (to_remove);
g_ptr_array_unref (header_exts);
if (!res)
return res;
}
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;
if (priv->negotiated)
priv->last_caps = gst_caps_ref (caps);
return res;
caps_not_changed:
{
GST_DEBUG_OBJECT (filter, "Caps did not change");
return res;
}
}
/* takes ownership of the input buffer */
static GstFlowReturn
gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * bclass, GstBuffer * in)
{
GstBuffer *(*process_rtp_packet_func) (GstRTPBaseDepayload * base,
GstRTPBuffer * rtp_buffer);
GstBuffer *(*process_func) (GstRTPBaseDepayload * base, GstBuffer * in);
GstRTPBaseDepayloadPrivate *priv;
GstBuffer *out_buf;
guint32 ssrc;
guint16 seqnum;
guint32 rtptime;
gboolean discont, buf_discont;
gint gap;
GstRTPBuffer rtp = { NULL };
GstReferenceTimestampMeta *meta;
GstCaps *ref_caps;
priv = filter->priv;
priv->process_flow_ret = GST_FLOW_OK;
process_func = bclass->process;
process_rtp_packet_func = bclass->process_rtp_packet;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
/* Check for duplicate reference timestamp metadata */
ref_caps = gst_static_caps_get (&ntp_reference_timestamp_caps);
meta = gst_buffer_get_reference_timestamp_meta (in, ref_caps);
gst_caps_unref (ref_caps);
if (meta) {
guint64 ref_ts = meta->timestamp;
if (ref_ts == priv->ref_ts) {
/* Drop the redundant/duplicate reference timstamp metadata */
in = gst_buffer_make_writable (in);
gst_buffer_remove_meta (in, GST_META_CAST (meta));
} else {
priv->ref_ts = ref_ts;
}
}
if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
goto invalid_buffer;
buf_discont = GST_BUFFER_IS_DISCONT (in);
priv->pts = GST_BUFFER_PTS (in);
priv->dts = GST_BUFFER_DTS (in);
priv->duration = GST_BUFFER_DURATION (in);
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
seqnum = gst_rtp_buffer_get_seq (&rtp);
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
priv->last_seqnum = seqnum;
priv->last_rtptime = rtptime;
discont = buf_discont;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %"
GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime,
GST_TIME_ARGS (priv->pts), GST_TIME_ARGS (priv->dts));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are strictly 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)) {
if (ssrc != priv->last_ssrc) {
GST_LOG_OBJECT (filter,
"New ssrc %u (current ssrc %u), sender restarted",
ssrc, priv->last_ssrc);
discont = TRUE;
} else {
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, have a
* reordered packet 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 assuming the sender has restarted. See
* also RFC 4737. */
if (gap <= priv->max_reorder) {
GST_WARNING_OBJECT (filter, "got old packet %u, expected %u, "
"gap %d <= max_reorder (%d), dropping!",
seqnum, priv->next_seqnum, gap, priv->max_reorder);
goto dropping;
}
GST_WARNING_OBJECT (filter, "got old packet %u, expected %u, "
"marking discont", seqnum, priv->next_seqnum);
discont = TRUE;
}
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
priv->last_ssrc = ssrc;
if (G_UNLIKELY (discont)) {
priv->discont = TRUE;
if (!buf_discont) {
gpointer old_inbuf = in;
/* 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. */
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
in = gst_buffer_make_writable (in);
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
/* depayloaders will check flag on rtpbuffer->buffer, so if the input
* buffer was not writable already we need to remap to make our
* newly-flagged buffer current on the rtpbuffer */
if (in != old_inbuf) {
gst_rtp_buffer_unmap (&rtp);
if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
goto invalid_buffer;
}
}
}
/* prepare segment event if needed */
if (filter->need_newsegment) {
priv->segment_event = create_segment_event (filter, rtptime,
GST_BUFFER_PTS (in));
filter->need_newsegment = FALSE;
}
priv->input_buffer = in;
if (discont) {
gst_rtp_base_depayload_reset_hdrext_buffers (filter);
g_assert_null (priv->hdrext_delayed);
}
/* update RTP buffer cache for header extensions if any */
if (priv->hdrext_aggregate &&
!priv->hdrext_seen && gst_rtp_buffer_get_extension (&rtp)) {
GST_INFO_OBJECT (filter, "Activate RTP header ext aggregation");
priv->hdrext_seen = priv->hdrext_aggregate;
}
if (priv->hdrext_seen) {
GstBuffer *b = gst_buffer_new ();
/* make a copy of the buffer that only contains the RTP header
with the extensions to not waste too much memory */
guint s = gst_rtp_buffer_get_header_len (&rtp);
gst_buffer_copy_into (b, in,
GST_BUFFER_COPY_MEMORY | GST_BUFFER_COPY_DEEP, 0, s);
gst_buffer_list_add (priv->hdrext_buffers, b);
}
if (process_rtp_packet_func != NULL) {
out_buf = process_rtp_packet_func (filter, &rtp);
gst_rtp_buffer_unmap (&rtp);
} else if (process_func != NULL) {
gst_rtp_buffer_unmap (&rtp);
out_buf = process_func (filter, in);
} else {
goto no_process;
}
/* let's send it out to processing */
if (out_buf) {
if (priv->process_flow_ret == GST_FLOW_OK) {
priv->process_flow_ret = gst_rtp_base_depayload_push (filter, out_buf);
} else {
gst_buffer_unref (out_buf);
gst_rtp_base_depayload_reset_hdrext_buffers (filter);
}
}
/* if the current buffer is delayed the depayloader should either
have called gst_rtp_base_depayload_push() internally or returned
a buffer that's pushed, either way the buffer cache should be
empty here and we append the delayed buffer */
if (priv->hdrext_delayed) {
g_assert_true (gst_buffer_list_length (priv->hdrext_buffers) == 0);
gst_buffer_list_add (priv->hdrext_buffers, priv->hdrext_delayed);
priv->hdrext_delayed = NULL;
}
gst_buffer_unref (in);
priv->input_buffer = NULL;
return priv->process_flow_ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
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"));
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_rtp_buffer_unmap (&rtp);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
gst_rtp_buffer_unmap (&rtp);
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process or process_rtp_packet method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayload *basedepay;
GstFlowReturn flow_ret;
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, in);
return flow_ret;
}
static GstFlowReturn
gst_rtp_base_depayload_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * list)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayload *basedepay;
GstFlowReturn flow_ret;
GstBuffer *buffer;
guint i, len;
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
flow_ret = GST_FLOW_OK;
/* chain each buffer in list individually */
len = gst_buffer_list_length (list);
if (len == 0)
goto done;
for (i = 0; i < len; i++) {
buffer = gst_buffer_list_get (list, i);
/* handle_buffer takes ownership of input buffer */
/* FIXME: add a way to steal buffers from list as we will unref it anyway */
gst_buffer_ref (buffer);
/* Should we fix up any missing timestamps for list buffers here
* (e.g. set to first or previous timestamp in list) or just assume
* the's a jitterbuffer that will have done that for us? */
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, buffer);
if (flow_ret != GST_FLOW_OK)
break;
}
done:
gst_buffer_list_unref (list);
return flow_ret;
}
static gboolean
gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter,
GstEvent * event)
{
gboolean res = TRUE;
gboolean forward = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_OBJECT_LOCK (filter);
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
GST_OBJECT_UNLOCK (filter);
filter->need_newsegment = !filter->priv->onvif_mode;
filter->priv->next_seqnum = -1;
filter->priv->ref_ts = -1;
gst_event_replace (&filter->priv->segment_event, NULL);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_rtp_base_depayload_setcaps (filter, caps);
forward = FALSE;
break;
}
case GST_EVENT_SEGMENT:
{
GstSegment segment;
GST_OBJECT_LOCK (filter);
gst_event_copy_segment (event, &segment);
if (segment.format != GST_FORMAT_TIME) {
GST_ERROR_OBJECT (filter, "Segment with non-TIME format not supported");
res = FALSE;
}
filter->priv->segment_seqnum = gst_event_get_seqnum (event);
filter->segment = segment;
GST_OBJECT_UNLOCK (filter);
/* In ONVIF mode, upstream is expected to send us the correct segment */
if (!filter->priv->onvif_mode) {
/* don't pass the event downstream, we generate our own segment including
* the NTP time and other things we receive in caps */
forward = FALSE;
}
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
GstRTPBaseDepayloadClass *bclass;
bclass = GST_RTP_BASE_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 GAP event.
* Subclasses can override and do 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);
forward = FALSE;
}
break;
}
default:
break;
}
if (forward)
res = gst_pad_push_event (filter->srcpad, event);
else
gst_event_unref (event);
return res;
}
static gboolean
gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res = FALSE;
GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadClass *bclass;
filter = GST_RTP_BASE_DEPAYLOAD (parent);
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
if (bclass->handle_event)
res = bclass->handle_event (filter, event);
else
gst_event_unref (event);
return res;
}
static GstEvent *
create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
GstClockTime position)
{
GstEvent *event;
GstClockTime start, stop, running_time;
GstRTPBaseDepayloadPrivate *priv;
GstSegment segment;
priv = filter->priv;
/* We don't need the object lock around - the segment
* can't change here while we're holding the STREAM_LOCK
*/
/* determining the start of the segment */
start = filter->segment.start;
if (priv->clock_base != -1 && position != -1) {
GstClockTime exttime, gap;
exttime = priv->clock_base;
gst_rtp_buffer_ext_timestamp (&exttime, rtptime);
gap = gst_util_uint64_scale_int (exttime - priv->clock_base,
filter->clock_rate, GST_SECOND);
/* account for lost packets */
if (position > gap) {
GST_DEBUG_OBJECT (filter,
"Found gap of %" GST_TIME_FORMAT ", adjusting start: %"
GST_TIME_FORMAT " = %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (gap), GST_TIME_ARGS (position - gap),
GST_TIME_ARGS (position), GST_TIME_ARGS (gap));
start = position - gap;
}
}
/* determining the stop of the segment */
stop = filter->segment.stop;
if (priv->npt_stop != -1)
stop = start + (priv->npt_stop - priv->npt_start);
if (position == -1)
position = start;
running_time = gst_segment_to_running_time (&filter->segment,
GST_FORMAT_TIME, start);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.rate = priv->play_speed;
segment.applied_rate = priv->play_scale;
segment.start = start;
segment.stop = stop;
segment.time = priv->npt_start;
segment.position = position;
segment.base = running_time;
GST_DEBUG_OBJECT (filter, "Creating segment event %" GST_SEGMENT_FORMAT,
&segment);
event = gst_event_new_segment (&segment);
if (filter->priv->segment_seqnum != GST_SEQNUM_INVALID)
gst_event_set_seqnum (event, filter->priv->segment_seqnum);
return event;
}
static gboolean
foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
{
GType drop_api_type = (GType) user_data;
const GstMetaInfo *info = (*meta)->info;
if (info->api == drop_api_type)
*meta = NULL;
return TRUE;
}
static void
add_rtp_source_meta (GstBuffer * outbuf, GstBuffer * rtpbuf)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
GstRTPSourceMeta *meta;
guint32 ssrc;
GType source_meta_api = gst_rtp_source_meta_api_get_type ();
if (!gst_rtp_buffer_map (rtpbuf, GST_MAP_READ, &rtp))
return;
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
/* remove any pre-existing source-meta */
gst_buffer_foreach_meta (outbuf, foreach_metadata_drop,
(gpointer) source_meta_api);
meta = gst_buffer_add_rtp_source_meta (outbuf, &ssrc, NULL, 0);
if (meta != NULL) {
gint i;
gint csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
for (i = 0; i < csrc_count; i++) {
guint32 csrc = gst_rtp_buffer_get_csrc (&rtp, i);
gst_rtp_source_meta_append_csrc (meta, &csrc, 1);
}
}
gst_rtp_buffer_unmap (&rtp);
}
static void
gst_rtp_base_depayload_add_extension (GstRTPBaseDepayload * rtpbasepayload,
GstRTPHeaderExtension * ext)
{
g_return_if_fail (GST_IS_RTP_HEADER_EXTENSION (ext));
g_return_if_fail (gst_rtp_header_extension_get_id (ext) > 0);
/* XXX: check for duplicate ids? */
GST_OBJECT_LOCK (rtpbasepayload);
g_ptr_array_add (rtpbasepayload->priv->header_exts, gst_object_ref (ext));
GST_OBJECT_UNLOCK (rtpbasepayload);
g_object_notify_by_pspec (G_OBJECT (rtpbasepayload),
gst_rtp_base_depayload_extensions_pspec);
}
static void
gst_rtp_base_depayload_clear_extensions (GstRTPBaseDepayload * rtpbasepayload)
{
GST_OBJECT_LOCK (rtpbasepayload);
g_ptr_array_set_size (rtpbasepayload->priv->header_exts, 0);
GST_OBJECT_UNLOCK (rtpbasepayload);
g_object_notify_by_pspec (G_OBJECT (rtpbasepayload),
gst_rtp_base_depayload_extensions_pspec);
}
static void
gst_rtp_base_depayload_get_extensions (GstRTPBaseDepayload * depayload,
GValue * out_value)
{
GPtrArray *extensions;
guint i;
GST_OBJECT_LOCK (depayload);
extensions = depayload->priv->header_exts;
for (i = 0; i < extensions->len; ++i) {
GValue value = G_VALUE_INIT;
g_value_init (&value, GST_TYPE_RTP_HEADER_EXTENSION);
g_value_set_object (&value, g_ptr_array_index (extensions, i));
gst_value_array_append_and_take_value (out_value, &value);
}
GST_OBJECT_UNLOCK (depayload);
}
static gboolean
read_rtp_header_extensions (GstRTPBaseDepayload * depayload,
GstBuffer * input, GstBuffer * output)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint16 bit_pattern;
guint8 *pdata;
guint wordlen;
gboolean needs_src_caps_update = FALSE;
if (!input) {
GST_DEBUG_OBJECT (depayload, "no input buffer");
return needs_src_caps_update;
}
if (!gst_rtp_buffer_map (input, GST_MAP_READ, &rtp)) {
GST_WARNING_OBJECT (depayload, "Failed to map buffer");
return needs_src_caps_update;
}
if (gst_rtp_buffer_get_extension_data (&rtp, &bit_pattern, (gpointer) & pdata,
&wordlen)) {
GstRTPHeaderExtensionFlags ext_flags = 0;
gsize bytelen = wordlen * 4;
guint hdr_unit_bytes;
gsize offset = 0;
if (bit_pattern == 0xBEDE) {
/* one byte extensions */
hdr_unit_bytes = 1;
ext_flags |= GST_RTP_HEADER_EXTENSION_ONE_BYTE;
} else if (bit_pattern >> 4 == 0x100) {
/* two byte extensions */
hdr_unit_bytes = 2;
ext_flags |= GST_RTP_HEADER_EXTENSION_TWO_BYTE;
} else {
GST_DEBUG_OBJECT (depayload, "unknown extension bit pattern 0x%02x%02x",
bit_pattern >> 8, bit_pattern & 0xff);
goto out;
}
while (TRUE) {
guint8 read_id, read_len;
GstRTPHeaderExtension *ext = NULL;
guint i;
if (offset + hdr_unit_bytes >= bytelen)
/* not enough remaning data */
break;
if (ext_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
read_id = GST_READ_UINT8 (pdata + offset) >> 4;
read_len = (GST_READ_UINT8 (pdata + offset) & 0x0F) + 1;
offset += 1;
if (read_id == 0)
/* padding */
continue;
if (read_id == 15)
/* special id for possible future expansion */
break;
} else {
read_id = GST_READ_UINT8 (pdata + offset);
offset += 1;
if (read_id == 0)
/* padding */
continue;
read_len = GST_READ_UINT8 (pdata + offset);
offset += 1;
}
GST_TRACE_OBJECT (depayload, "found rtp header extension with id %u and "
"length %u", read_id, read_len);
/* Ignore extension headers where the size does not fit */
if (offset + read_len > bytelen) {
GST_WARNING_OBJECT (depayload, "Extension length extends past the "
"size of the extension data");
break;
}
GST_OBJECT_LOCK (depayload);
for (i = 0; i < depayload->priv->header_exts->len; i++) {
ext = g_ptr_array_index (depayload->priv->header_exts, i);
if (read_id == gst_rtp_header_extension_get_id (ext)) {
gst_object_ref (ext);
break;
}
ext = NULL;
}
if (ext) {
if (!gst_rtp_header_extension_read (ext, ext_flags, &pdata[offset],
read_len, output)) {
GST_WARNING_OBJECT (depayload, "RTP header extension (%s) could "
"not read payloaded data", GST_OBJECT_NAME (ext));
gst_object_unref (ext);
goto out;
}
if (gst_rtp_header_extension_wants_update_non_rtp_src_caps (ext)) {
needs_src_caps_update = TRUE;
}
gst_object_unref (ext);
}
GST_OBJECT_UNLOCK (depayload);
offset += read_len;
}
}
out:
gst_rtp_buffer_unmap (&rtp);
return needs_src_caps_update;
}
static gboolean
gst_rtp_base_depayload_operate_hdrext_buffer (GstBuffer ** buffer,
guint idx, gpointer depayloader)
{
GstRTPBaseDepayload *depayload = depayloader;
depayload->priv->hdrext_read_result |=
read_rtp_header_extensions (depayload, *buffer,
depayload->priv->hdrext_outbuf);
return TRUE;
}
static void
gst_rtp_base_depayload_reset_hdrext_buffers (GstRTPBaseDepayload * depayload)
{
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
gst_buffer_list_unref (priv->hdrext_buffers);
priv->hdrext_buffers = gst_buffer_list_new ();
}
static gboolean
gst_rtp_base_depayload_set_headers (GstRTPBaseDepayload * depayload,
GstBuffer * buffer)
{
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
GstClockTime pts, dts, duration;
gboolean ret = FALSE;
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* apply last incoming timestamp and duration to outgoing buffer if
* not otherwise set. */
if (!GST_CLOCK_TIME_IS_VALID (pts))
GST_BUFFER_PTS (buffer) = priv->pts;
if (!GST_CLOCK_TIME_IS_VALID (dts))
GST_BUFFER_DTS (buffer) = priv->dts;
if (!GST_CLOCK_TIME_IS_VALID (duration))
GST_BUFFER_DURATION (buffer) = priv->duration;
if (G_UNLIKELY (depayload->priv->discont)) {
GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
depayload->priv->discont = FALSE;
}
/* make sure we only set the timestamp on the first packet */
priv->pts = GST_CLOCK_TIME_NONE;
priv->dts = GST_CLOCK_TIME_NONE;
priv->duration = GST_CLOCK_TIME_NONE;
if (priv->input_buffer) {
if (priv->source_info)
add_rtp_source_meta (buffer, priv->input_buffer);
if (priv->hdrext_aggregate) {
priv->hdrext_read_result = FALSE;
priv->hdrext_outbuf = buffer;
/* if we have an empty list but a delayed RTP buffer let's use it */
if (!gst_buffer_list_length (priv->hdrext_buffers) &&
priv->hdrext_delayed) {
gst_buffer_list_add (priv->hdrext_buffers, priv->hdrext_delayed);
priv->hdrext_delayed = NULL;
}
gst_buffer_list_foreach (priv->hdrext_buffers,
gst_rtp_base_depayload_operate_hdrext_buffer, depayload);
ret = priv->hdrext_read_result;
priv->hdrext_outbuf = NULL;
} else {
ret = read_rtp_header_extensions (depayload, priv->input_buffer, buffer);
}
}
return ret;
}
static GstFlowReturn
gst_rtp_base_depayload_finish_push (GstRTPBaseDepayload * filter,
gboolean is_list, gpointer obj)
{
/* if this is the first buffer send a NEWSEGMENT */
if (G_UNLIKELY (filter->priv->segment_event)) {
gst_pad_push_event (filter->srcpad, filter->priv->segment_event);
filter->priv->segment_event = NULL;
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
}
if (is_list) {
GstBufferList *blist = obj;
return gst_pad_push_list (filter->srcpad, blist);
} else {
GstBuffer *buf = obj;
return gst_pad_push (filter->srcpad, buf);
}
}
static gboolean
gst_rtp_base_depayload_set_src_caps_from_hdrext (GstRTPBaseDepayload * filter)
{
gboolean update_ok = TRUE;
GstCaps *src_caps = gst_pad_get_current_caps (filter->srcpad);
if (src_caps) {
GstCaps *new_caps;
gint i;
new_caps = gst_caps_copy (src_caps);
for (i = 0; i < filter->priv->header_exts->len; i++) {
GstRTPHeaderExtension *ext;
ext = g_ptr_array_index (filter->priv->header_exts, i);
update_ok =
gst_rtp_header_extension_update_non_rtp_src_caps (ext, new_caps);
if (!update_ok) {
GST_ELEMENT_ERROR (filter, STREAM, DECODE,
("RTP header extension (%s) could not update src caps",
GST_OBJECT_NAME (ext)), (NULL));
break;
}
}
if (G_UNLIKELY (update_ok && !gst_caps_is_equal (src_caps, new_caps))) {
gst_pad_set_caps (filter->srcpad, new_caps);
}
gst_caps_unref (src_caps);
gst_caps_unref (new_caps);
}
return update_ok;
}
static GstFlowReturn
gst_rtp_base_depayload_do_push (GstRTPBaseDepayload * filter, gboolean is_list,
gpointer obj)
{
GstFlowReturn res;
if (is_list) {
GstBufferList *blist = obj;
guint i;
guint first_not_pushed_idx = 0;
for (i = 0; i < gst_buffer_list_length (blist); ++i) {
GstBuffer *buf = gst_buffer_list_get_writable (blist, i);
if (G_UNLIKELY (gst_rtp_base_depayload_set_headers (filter, buf))) {
/* src caps have changed; push the buffers preceding the current one,
* then apply the new caps on the src pad */
guint j;
for (j = first_not_pushed_idx; j < i; ++j) {
res = gst_rtp_base_depayload_finish_push (filter, FALSE,
gst_buffer_ref (gst_buffer_list_get (blist, j)));
if (G_UNLIKELY (res != GST_FLOW_OK)) {
goto error_list;
}
}
first_not_pushed_idx = i;
if (!gst_rtp_base_depayload_set_src_caps_from_hdrext (filter)) {
res = GST_FLOW_ERROR;
goto error_list;
}
}
}
if (G_LIKELY (first_not_pushed_idx == 0)) {
res = gst_rtp_base_depayload_finish_push (filter, TRUE, blist);
blist = NULL;
} else {
for (i = first_not_pushed_idx; i < gst_buffer_list_length (blist); ++i) {
res = gst_rtp_base_depayload_finish_push (filter, FALSE,
gst_buffer_ref (gst_buffer_list_get (blist, i)));
if (G_UNLIKELY (res != GST_FLOW_OK)) {
break;
}
}
}
error_list:
gst_clear_buffer_list (&blist);
} else {
GstBuffer *buf = obj;
if (G_UNLIKELY (gst_rtp_base_depayload_set_headers (filter, buf))) {
if (!gst_rtp_base_depayload_set_src_caps_from_hdrext (filter)) {
res = GST_FLOW_ERROR;
goto error_buffer;
}
}
res = gst_rtp_base_depayload_finish_push (filter, FALSE, buf);
buf = NULL;
error_buffer:
gst_clear_buffer (&buf);
}
gst_rtp_base_depayload_reset_hdrext_buffers (filter);
return res;
}
/**
* gst_rtp_base_depayload_push:
* @filter: a #GstRTPBaseDepayload
* @out_buf: (transfer full): a #GstBuffer
*
* Push @out_buf to the peer of @filter. This function takes ownership of
* @out_buf.
*
* This function will by default apply the last incoming timestamp on
* the outgoing buffer when it didn't have a timestamp already.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
{
GstFlowReturn res;
res = gst_rtp_base_depayload_do_push (filter, FALSE, out_buf);
if (res != GST_FLOW_OK)
filter->priv->process_flow_ret = res;
return res;
}
/**
* gst_rtp_base_depayload_push_list:
* @filter: a #GstRTPBaseDepayload
* @out_list: (transfer full): a #GstBufferList
*
* Push @out_list to the peer of @filter. This function takes ownership of
* @out_list.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
GstBufferList * out_list)
{
GstFlowReturn res;
res = gst_rtp_base_depayload_do_push (filter, TRUE, out_list);
if (res != GST_FLOW_OK)
filter->priv->process_flow_ret = res;
return res;
}
/* convert the PacketLost event from a jitterbuffer to a GAP event.
* subclasses can override this. */
static gboolean
gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter,
GstEvent * event)
{
GstClockTime timestamp, duration;
GstEvent *sevent;
const GstStructure *s;
gboolean might_have_been_fec;
gboolean res = TRUE;
s = gst_event_get_structure (event);
/* first start by parsing the timestamp and duration */
timestamp = -1;
duration = -1;
if (!gst_structure_get_clock_time (s, "timestamp", &timestamp) ||
!gst_structure_get_clock_time (s, "duration", &duration)) {
GST_ERROR_OBJECT (filter,
"Packet loss event without timestamp or duration");
return FALSE;
}
sevent = gst_pad_get_sticky_event (filter->srcpad, GST_EVENT_SEGMENT, 0);
if (G_UNLIKELY (!sevent)) {
/* Typically happens if lost event arrives before first buffer */
GST_DEBUG_OBJECT (filter,
"Ignore packet loss because segment event missing");
return FALSE;
}
gst_event_unref (sevent);
if (!gst_structure_get_boolean (s, "might-have-been-fec",
&might_have_been_fec) || !might_have_been_fec) {
/* send GAP event */
sevent = gst_event_new_gap (timestamp, duration);
gst_event_set_gap_flags (sevent, GST_GAP_FLAG_MISSING_DATA);
res = gst_pad_push_event (filter->srcpad, sevent);
}
return res;
}
static GstStateChangeReturn
gst_rtp_base_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadPrivate *priv;
GstStateChangeReturn ret;
filter = GST_RTP_BASE_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->clock_base = -1;
priv->ref_ts = -1;
priv->onvif_mode = FALSE;
priv->next_seqnum = -1;
priv->negotiated = FALSE;
priv->discont = FALSE;
priv->segment_seqnum = GST_SEQNUM_INVALID;
priv->hdrext_seen = FALSE;
if (priv->hdrext_delayed)
gst_buffer_unref (priv->hdrext_delayed);
gst_rtp_base_depayload_reset_hdrext_buffers (filter);
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:
gst_caps_replace (&priv->last_caps, NULL);
gst_event_replace (&priv->segment_event, NULL);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static GstStructure *
gst_rtp_base_depayload_create_stats (GstRTPBaseDepayload * depayload)
{
GstRTPBaseDepayloadPrivate *priv;
GstStructure *s;
GstClockTime pts = GST_CLOCK_TIME_NONE, dts = GST_CLOCK_TIME_NONE;
priv = depayload->priv;
GST_OBJECT_LOCK (depayload);
if (depayload->segment.format != GST_FORMAT_UNDEFINED) {
pts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
priv->pts);
dts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
priv->dts);
}
GST_OBJECT_UNLOCK (depayload);
s = gst_structure_new ("application/x-rtp-depayload-stats",
"clock_rate", G_TYPE_UINT, depayload->clock_rate,
"npt-start", G_TYPE_UINT64, priv->npt_start,
"npt-stop", G_TYPE_UINT64, priv->npt_stop,
"play-speed", G_TYPE_DOUBLE, priv->play_speed,
"play-scale", G_TYPE_DOUBLE, priv->play_scale,
"running-time-dts", G_TYPE_UINT64, dts,
"running-time-pts", G_TYPE_UINT64, pts,
"seqnum", G_TYPE_UINT, (guint) priv->last_seqnum,
"timestamp", G_TYPE_UINT, (guint) priv->last_rtptime, NULL);
return s;
}
static void
gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPBaseDepayload *depayload;
GstRTPBaseDepayloadPrivate *priv;
depayload = GST_RTP_BASE_DEPAYLOAD (object);
priv = depayload->priv;
switch (prop_id) {
case PROP_SOURCE_INFO:
gst_rtp_base_depayload_set_source_info_enabled (depayload,
g_value_get_boolean (value));
break;
case PROP_MAX_REORDER:
priv->max_reorder = g_value_get_int (value);
break;
case PROP_AUTO_HEADER_EXTENSION:
priv->auto_hdr_ext = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPBaseDepayload *depayload;
GstRTPBaseDepayloadPrivate *priv;
depayload = GST_RTP_BASE_DEPAYLOAD (object);
priv = depayload->priv;
switch (prop_id) {
case PROP_STATS:
g_value_take_boxed (value,
gst_rtp_base_depayload_create_stats (depayload));
break;
case PROP_SOURCE_INFO:
g_value_set_boolean (value,
gst_rtp_base_depayload_is_source_info_enabled (depayload));
break;
case PROP_MAX_REORDER:
g_value_set_int (value, priv->max_reorder);
break;
case PROP_AUTO_HEADER_EXTENSION:
g_value_set_boolean (value, priv->auto_hdr_ext);
break;
case PROP_EXTENSIONS:
gst_rtp_base_depayload_get_extensions (depayload, value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/**
* gst_rtp_base_depayload_set_source_info_enabled:
* @depayload: a #GstRTPBaseDepayload
* @enable: whether to add meta about RTP sources to buffer
*
* Enable or disable adding #GstRTPSourceMeta to depayloaded buffers.
*
* Since: 1.16
**/
void
gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
gboolean enable)
{
depayload->priv->source_info = enable;
}
/**
* gst_rtp_base_depayload_is_source_info_enabled:
* @depayload: a #GstRTPBaseDepayload
*
* Queries whether #GstRTPSourceMeta will be added to depayloaded buffers.
*
* Returns: %TRUE if source-info is enabled.
*
* Since: 1.16
**/
gboolean
gst_rtp_base_depayload_is_source_info_enabled (GstRTPBaseDepayload * depayload)
{
return depayload->priv->source_info;
}
/**
* gst_rtp_base_depayload_set_aggregate_hdrext_enabled:
* @depayload: a #GstRTPBaseDepayload
* @enable: whether to aggregate header extensions per output buffer
*
* Enable or disable aggregating header extensions.
*
* Since: 1.24
**/
void
gst_rtp_base_depayload_set_aggregate_hdrext_enabled (GstRTPBaseDepayload *
depayload, gboolean enable)
{
depayload->priv->hdrext_aggregate = enable;
if (!enable)
gst_rtp_base_depayload_reset_hdrext_buffers (depayload);
}
/**
* gst_rtp_base_depayload_is_aggregate_hdrext_enabled:
* @depayload: a #GstRTPBaseDepayload
*
* Queries whether header extensions will be aggregated per depayloaded buffers.
*
* Returns: %TRUE if aggregate-header-extension is enabled.
*
* Since: 1.24
**/
gboolean
gst_rtp_base_depayload_is_aggregate_hdrext_enabled (GstRTPBaseDepayload *
depayload)
{
return depayload->priv->hdrext_aggregate;
}
/**
* gst_rtp_base_depayload_dropped:
* @depayload: a #GstRTPBaseDepayload
*
* Called from @GstRTPBaseDepayload.process or
* @GstRTPBaseDepayload.process_rtp_packet if the depayloader does not
* use the current buffer for the output buffer. This will either drop
* the delayed buffer or the last buffer from the header extension
* cache.
*
* A typical use-case is when the depayloader implementation is
* dropping an input RTP buffer while waiting for the first keyframe.
*
* Must be called with the stream lock held.
*
* Since: 1.24
**/
void
gst_rtp_base_depayload_dropped (GstRTPBaseDepayload * depayload)
{
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
guint l = gst_buffer_list_length (priv->hdrext_buffers);
if (priv->hdrext_delayed) {
gst_clear_buffer (&priv->hdrext_delayed);
} else if (l) {
gst_buffer_list_remove (priv->hdrext_buffers, l - 1, 1);
}
}
/**
* gst_rtp_base_depayload_delayed:
* @depayload: a #GstRTPBaseDepayload
*
* Called from @GstRTPBaseDepayload.process or
* @GstRTPBaseDepayload.process_rtp_packet when the depayloader needs
* to keep the current input RTP header for use with the next output
* buffer.
*
* The delayed buffer will remain until the end of processing the
* current output buffer and then enqueued for processing with the
* next output buffer.
*
* A typical use-case is when the depayloader implementation will
* start a new output buffer for the current input RTP buffer but push
* the current output buffer first.
*
* Must be called with the stream lock held.
*
* Since: 1.24
**/
void
gst_rtp_base_depayload_delayed (GstRTPBaseDepayload * depayload)
{
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
guint l = gst_buffer_list_length (priv->hdrext_buffers);
if (l) {
priv->hdrext_delayed = gst_buffer_list_get (priv->hdrext_buffers, l - 1);
gst_buffer_ref (priv->hdrext_delayed);
gst_buffer_list_remove (priv->hdrext_buffers, l - 1, 1);
}
}
/**
* gst_rtp_base_depayload_flush:
* @depayload: a #GstRTPBaseDepayload
* @keep_current: if the current RTP buffer shall be kept
*
* If @GstRTPBaseDepayload.process or
* @GstRTPBaseDepayload.process_rtp_packet drop an output buffer this
* function tells the base class to flush header extension cache as
* well.
*
* This will not drop an input RTP header marked as delayed from
* gst_rtp_base_depayload_delayed().
*
* If @keep_current is %TRUE the current input RTP header will be kept
* and enqueued after flushing the previous input RTP headers.
*
* A typical use-case for @keep_current is when the depayloader
* implementation invalidates the current output buffer and starts a
* new one with the current RTP input buffer.
*
* Must be called with the stream lock held.
*
* Since: 1.24
**/
void
gst_rtp_base_depayload_flush (GstRTPBaseDepayload * depayload,
gboolean keep_current)
{
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
guint l = gst_buffer_list_length (priv->hdrext_buffers);
/* if the current buffer shall not be kept or has already been
removed from the cache clear the cache */
if (!keep_current || priv->hdrext_delayed) {
gst_rtp_base_depayload_reset_hdrext_buffers (depayload);
} else if (l) {
/* clear all cached buffers (if any) except the delayed */
GstBuffer *b = gst_buffer_list_get (priv->hdrext_buffers, l - 1);
gst_buffer_ref (b);
gst_rtp_base_depayload_reset_hdrext_buffers (depayload);
gst_buffer_list_add (priv->hdrext_buffers, b);
}
}
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