<SECTION>
<FILE>gstbasertpaudiopayload</FILE>
<INCLUDE>gst/rtp/gstbasertpaudiopayload.h</INCLUDE>
-GstBaseRTPAudioPayload
-GstBaseRTPAudioPayloadClass
-
-gst_base_rtp_audio_payload_set_frame_based
-gst_base_rtp_audio_payload_set_frame_options
-gst_base_rtp_audio_payload_set_sample_based
-gst_base_rtp_audio_payload_set_sample_options
-gst_base_rtp_audio_payload_get_adapter
-gst_base_rtp_audio_payload_push
-gst_base_rtp_audio_payload_flush
-gst_base_rtp_audio_payload_set_samplebits_options
+GstRTPBaseAudioPayload
+GstRTPBaseAudioPayloadClass
+
+gst_rtp_base_audio_payload_set_frame_based
+gst_rtp_base_audio_payload_set_frame_options
+gst_rtp_base_audio_payload_set_sample_based
+gst_rtp_base_audio_payload_set_sample_options
+gst_rtp_base_audio_payload_get_adapter
+gst_rtp_base_audio_payload_push
+gst_rtp_base_audio_payload_flush
+gst_rtp_base_audio_payload_set_samplebits_options
<SUBSECTION Standard>
-GST_TYPE_BASE_RTP_AUDIO_PAYLOAD
-GST_BASE_RTP_AUDIO_PAYLOAD
-GST_BASE_RTP_AUDIO_PAYLOAD_CLASS
-GST_IS_BASE_RTP_AUDIO_PAYLOAD
-GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS
-GST_BASE_RTP_AUDIO_PAYLOAD_CAST
-gst_base_rtp_audio_payload_get_type
-GstBaseRTPAudioPayloadPrivate
+GST_TYPE_RTP_BASE_AUDIO_PAYLOAD
+GST_RTP_BASE_AUDIO_PAYLOAD
+GST_RTP_BASE_AUDIO_PAYLOAD_CLASS
+GST_IS_RTP_BASE_AUDIO_PAYLOAD
+GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS
+GST_RTP_BASE_AUDIO_PAYLOAD_CAST
+gst_rtp_base_audio_payload_get_type
+GstRTPBaseAudioPayloadPrivate
</SECTION>
<SECTION>
<FILE>gstbasertpdepayload</FILE>
<INCLUDE>gst/rtp/gstbasertpdepayload.h</INCLUDE>
-GstBaseRTPDepayload
-GstBaseRTPDepayloadClass
+GstRTPBaseDepayload
+GstRTPBaseDepayloadClass
-GST_BASE_RTP_DEPAYLOAD_SINKPAD
-GST_BASE_RTP_DEPAYLOAD_SRCPAD
+GST_RTP_BASE_DEPAYLOAD_SINKPAD
+GST_RTP_BASE_DEPAYLOAD_SRCPAD
-gst_base_rtp_depayload_push
-gst_base_rtp_depayload_push_ts
-gst_base_rtp_depayload_push_list
+gst_rtp_base_depayload_push
+gst_rtp_base_depayload_push_list
<SUBSECTION Standard>
-GstBaseRTPDepayloadPrivate
-GST_TYPE_BASE_RTP_DEPAYLOAD
-GST_BASE_RTP_DEPAYLOAD
-GST_BASE_RTP_DEPAYLOAD_CLASS
-GST_BASE_RTP_DEPAYLOAD_GET_CLASS
-GST_IS_BASE_RTP_DEPAYLOAD
-GST_IS_BASE_RTP_DEPAYLOAD_CLASS
-GST_BASE_RTP_PAYLOAD_CAST
-gst_base_rtp_depayload_get_type
-GstBaseRTPPayloadPrivate
+GstRTPBaseDepayloadPrivate
+GST_TYPE_RTP_BASE_DEPAYLOAD
+GST_RTP_BASE_DEPAYLOAD
+GST_RTP_BASE_DEPAYLOAD_CLASS
+GST_RTP_BASE_DEPAYLOAD_GET_CLASS
+GST_IS_RTP_BASE_DEPAYLOAD
+GST_IS_RTP_BASE_DEPAYLOAD_CLASS
+GST_RTP_BASE_PAYLOAD_CAST
+gst_rtp_base_depayload_get_type
+GstRTPBasePayloadPrivate
<SUBSECTION Private>
QUEUE_LOCK_INIT
<SECTION>
<FILE>gstbasertppayload</FILE>
<INCLUDE>gst/rtp/gstbasertppayload.h</INCLUDE>
-GstBaseRTPPayload
-GstBaseRTPPayloadClass
-
-GST_BASE_RTP_PAYLOAD_MTU
-GST_BASE_RTP_PAYLOAD_PT
-GST_BASE_RTP_PAYLOAD_SINKPAD
-GST_BASE_RTP_PAYLOAD_SRCPAD
-
-gst_basertppayload_is_filled
-gst_basertppayload_push
-gst_basertppayload_push_list
-gst_basertppayload_set_options
-gst_basertppayload_set_outcaps
+GstRTPBasePayload
+GstRTPBasePayloadClass
+
+GST_RTP_BASE_PAYLOAD_MTU
+GST_RTP_BASE_PAYLOAD_PT
+GST_RTP_BASE_PAYLOAD_SINKPAD
+GST_RTP_BASE_PAYLOAD_SRCPAD
+
+gst_rtp_base_payload_is_filled
+gst_rtp_base_payload_push
+gst_rtp_base_payload_push_list
+gst_rtp_base_payload_set_options
+gst_rtp_base_payload_set_outcaps
<SUBSECTION Standard>
-GST_TYPE_BASE_RTP_PAYLOAD
-GST_BASE_RTP_PAYLOAD
-GST_BASE_RTP_PAYLOAD_CLASS
-GST_BASE_RTP_PAYLOAD_GET_CLASS
-GST_IS_BASE_RTP_PAYLOAD
-GST_IS_BASE_RTP_PAYLOAD_CLASS
-gst_basertppayload_get_type
+GST_TYPE_RTP_BASE_PAYLOAD
+GST_RTP_BASE_PAYLOAD
+GST_RTP_BASE_PAYLOAD_CLASS
+GST_RTP_BASE_PAYLOAD_GET_CLASS
+GST_IS_RTP_BASE_PAYLOAD
+GST_IS_RTP_BASE_PAYLOAD_CLASS
+gst_rtp_base_payload_get_type
</SECTION>
<SECTION>
#include <gst/rtp/gstbasertpdepayload.h>
-gst_base_rtp_depayload_get_type
+gst_rtp_base_depayload_get_type
#include <gst/rtp/gstbasertppayload.h>
-gst_base_rtp_payload_get_type
+gst_rtp_base_payload_get_type
#include <gst/rtp/gstbasertpaudiopayload.h>
-gst_base_rtp_audio_payload_get_type
+gst_rtp_base_audio_payload_get_type
#include <gst/video/gstvideofilter.h>
* Provides a base class for audio RTP payloaders for frame or sample based
* audio codecs (constant bitrate)
*
- * This class derives from GstBaseRTPPayload. It can be used for payloading
+ * This class derives from GstRTPBasePayload. It can be used for payloading
* audio codecs. It will only work with constant bitrate codecs. It supports
* both frame based and sample based codecs. It takes care of packing up the
* audio data into RTP packets and filling up the headers accordingly. The
* <title>Usage</title>
* <para>
* To use this base class, your child element needs to call either
- * gst_base_rtp_audio_payload_set_frame_based() or
- * gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the
+ * gst_rtp_base_audio_payload_set_frame_based() or
+ * gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
* element's _init() function. Then, the child element must call either
- * gst_base_rtp_audio_payload_set_frame_options(),
- * gst_base_rtp_audio_payload_set_sample_options() or
- * gst_base_rtp_audio_payload_set_samplebits_options. Since
- * GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element
+ * gst_rtp_base_audio_payload_set_frame_options(),
+ * gst_rtp_base_audio_payload_set_sample_options() or
+ * gst_rtp_base_audio_payload_set_samplebits_options. Since
+ * GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element
* must set any variables or call/override any functions required by that base
* class. The child element does not need to override any other functions
- * specific to GstBaseRTPAudioPayload.
+ * specific to GstRTPBaseAudioPayload.
* </para>
* </refsect2>
*/
};
/* function to convert bytes to a time */
-typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload,
+typedef GstClockTime (*GetBytesToTimeFunc) (GstRTPBaseAudioPayload * payload,
guint64 bytes);
/* function to convert bytes to a RTP time */
-typedef guint32 (*GetBytesToRTPTimeFunc) (GstBaseRTPAudioPayload * payload,
+typedef guint32 (*GetBytesToRTPTimeFunc) (GstRTPBaseAudioPayload * payload,
guint64 bytes);
/* function to convert time to bytes */
-typedef guint64 (*GetTimeToBytesFunc) (GstBaseRTPAudioPayload * payload,
+typedef guint64 (*GetTimeToBytesFunc) (GstRTPBaseAudioPayload * payload,
GstClockTime time);
-struct _GstBaseRTPAudioPayloadPrivate
+struct _GstRTPBaseAudioPayloadPrivate
{
GetBytesToTimeFunc bytes_to_time;
GetBytesToRTPTimeFunc bytes_to_rtptime;
};
-#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \
- GstBaseRTPAudioPayloadPrivate))
+#define GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE(o) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_BASE_AUDIO_PAYLOAD, \
+ GstRTPBaseAudioPayloadPrivate))
-static void gst_base_rtp_audio_payload_finalize (GObject * object);
+static void gst_rtp_base_audio_payload_finalize (GObject * object);
-static void gst_base_rtp_audio_payload_set_property (GObject * object,
+static void gst_rtp_base_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_base_rtp_audio_payload_get_property (GObject * object,
+static void gst_rtp_base_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* bytes to time functions */
static GstClockTime
-gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes);
static GstClockTime
-gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes);
/* bytes to RTP time functions */
static guint32
-gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes);
static guint32
-gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes);
/* time to bytes functions */
static guint64
-gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time);
static guint64
-gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time);
-static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
+static GstFlowReturn gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload
* payload, GstBuffer * buffer);
-static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement
+static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement
* element, GstStateChange transition);
-static gboolean gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload
+static gboolean gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload
* payload, GstEvent * event);
-#define gst_base_rtp_audio_payload_parent_class parent_class
-G_DEFINE_TYPE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
- GST_TYPE_BASE_RTP_PAYLOAD);
+#define gst_rtp_base_audio_payload_parent_class parent_class
+G_DEFINE_TYPE (GstRTPBaseAudioPayload, gst_rtp_base_audio_payload,
+ GST_TYPE_RTP_BASE_PAYLOAD);
static void
-gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)
+gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstbasertppayload_class;
- g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate));
+ g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate));
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
- gobject_class->finalize = gst_base_rtp_audio_payload_finalize;
- gobject_class->set_property = gst_base_rtp_audio_payload_set_property;
- gobject_class->get_property = gst_base_rtp_audio_payload_get_property;
+ gobject_class->finalize = gst_rtp_base_audio_payload_finalize;
+ gobject_class->set_property = gst_rtp_base_audio_payload_set_property;
+ gobject_class->get_property = gst_rtp_base_audio_payload_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
g_param_spec_boolean ("buffer-list", "Buffer List",
DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state);
+ GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state);
gstbasertppayload_class->handle_buffer =
- GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer);
+ GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer);
gstbasertppayload_class->handle_event =
- GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event);
+ GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_handle_event);
GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
"base audio RTP payloader");
}
static void
-gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * payload)
+gst_rtp_base_audio_payload_init (GstRTPBaseAudioPayload * payload)
{
- payload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (payload);
+ payload->priv = GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE (payload);
/* these need to be set by child object if frame based */
payload->frame_size = 0;
}
static void
-gst_base_rtp_audio_payload_finalize (GObject * object)
+gst_rtp_base_audio_payload_finalize (GObject * object)
{
- GstBaseRTPAudioPayload *payload;
+ GstRTPBaseAudioPayload *payload;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
g_object_unref (payload->priv->adapter);
}
static void
-gst_base_rtp_audio_payload_set_property (GObject * object,
+gst_rtp_base_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
- GstBaseRTPAudioPayload *payload;
+ GstRTPBaseAudioPayload *payload;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
switch (prop_id) {
case PROP_BUFFER_LIST:
}
static void
-gst_base_rtp_audio_payload_get_property (GObject * object,
+gst_rtp_base_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
- GstBaseRTPAudioPayload *payload;
+ GstRTPBaseAudioPayload *payload;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
switch (prop_id) {
case PROP_BUFFER_LIST:
}
/**
- * gst_base_rtp_audio_payload_set_frame_based:
+ * gst_rtp_base_audio_payload_set_frame_based:
* @basertpaudiopayload: a pointer to the element.
*
- * Tells #GstBaseRTPAudioPayload that the child element is for a frame based
+ * Tells #GstRTPBaseAudioPayload that the child element is for a frame based
* audio codec
*/
void
-gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
basertpaudiopayload->priv->bytes_to_time =
- gst_base_rtp_audio_payload_frame_bytes_to_time;
+ gst_rtp_base_audio_payload_frame_bytes_to_time;
basertpaudiopayload->priv->bytes_to_rtptime =
- gst_base_rtp_audio_payload_frame_bytes_to_rtptime;
+ gst_rtp_base_audio_payload_frame_bytes_to_rtptime;
basertpaudiopayload->priv->time_to_bytes =
- gst_base_rtp_audio_payload_frame_time_to_bytes;
+ gst_rtp_base_audio_payload_frame_time_to_bytes;
}
/**
- * gst_base_rtp_audio_payload_set_sample_based:
+ * gst_rtp_base_audio_payload_set_sample_based:
* @basertpaudiopayload: a pointer to the element.
*
- * Tells #GstBaseRTPAudioPayload that the child element is for a sample based
+ * Tells #GstRTPBaseAudioPayload that the child element is for a sample based
* audio codec
*/
void
-gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
basertpaudiopayload->priv->bytes_to_time =
- gst_base_rtp_audio_payload_sample_bytes_to_time;
+ gst_rtp_base_audio_payload_sample_bytes_to_time;
basertpaudiopayload->priv->bytes_to_rtptime =
- gst_base_rtp_audio_payload_sample_bytes_to_rtptime;
+ gst_rtp_base_audio_payload_sample_bytes_to_rtptime;
basertpaudiopayload->priv->time_to_bytes =
- gst_base_rtp_audio_payload_sample_time_to_bytes;
+ gst_rtp_base_audio_payload_sample_time_to_bytes;
}
/**
- * gst_base_rtp_audio_payload_set_frame_options:
+ * gst_rtp_base_audio_payload_set_frame_options:
* @basertpaudiopayload: a pointer to the element.
* @frame_duration: The duraction of an audio frame in milliseconds.
* @frame_size: The size of an audio frame in bytes.
*
*/
void
-gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
+gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint frame_duration, gint frame_size)
{
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (basertpaudiopayload != NULL);
}
/**
- * gst_base_rtp_audio_payload_set_sample_options:
+ * gst_rtp_base_audio_payload_set_sample_options:
* @basertpaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bytes.
*
* Sets the options for sample based audio codecs.
*/
void
-gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
+gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
/* sample_size is in bits internally */
- gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ gst_rtp_base_audio_payload_set_samplebits_options (basertpaudiopayload,
sample_size * 8);
}
/**
- * gst_base_rtp_audio_payload_set_samplebits_options:
+ * gst_rtp_base_audio_payload_set_samplebits_options:
* @basertpaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bits.
*
* Since: 0.10.18
*/
void
-gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
+gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint sample_size)
{
guint fragment_size;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (basertpaudiopayload != NULL);
}
static void
-gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload,
+gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload,
GstBuffer * buffer, guint payload_len, GstClockTime timestamp)
{
- GstBaseRTPPayload *basepayload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBasePayload *basepayload;
+ GstRTPBaseAudioPayloadPrivate *priv;
GstRTPBuffer rtp;
- basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload);
+ basepayload = GST_RTP_BASE_PAYLOAD_CAST (payload);
priv = payload->priv;
/* set payload type */
}
/**
- * gst_base_rtp_audio_payload_push:
- * @baseaudiopayload: a #GstBaseRTPPayload
+ * gst_rtp_base_audio_payload_push:
+ * @baseaudiopayload: a #GstRTPBasePayload
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
* Since: 0.10.13
*/
GstFlowReturn
-gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
+gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
- GstBaseRTPPayload *basepayload;
+ GstRTPBasePayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GstRTPBuffer rtp;
- basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+ basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
gst_rtp_buffer_unmap (&rtp);
/* set metadata */
- gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
+ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
- ret = gst_base_rtp_payload_push (basepayload, outbuf);
+ ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
}
static GstFlowReturn
-gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
- GstBaseRTPPayload *basepayload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBasePayload *basepayload;
+ GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
- basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+ basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
}
/* set metadata */
- gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
+ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
- ret = gst_base_rtp_payload_push_list (basepayload, list);
+ ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
GstRTPBuffer rtp;
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
- ret = gst_base_rtp_payload_push (basepayload, outbuf);
+ ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
/**
- * gst_base_rtp_audio_payload_flush:
- * @baseaudiopayload: a #GstBaseRTPPayload
+ * gst_rtp_base_audio_payload_flush:
+ * @baseaudiopayload: a #GstRTPBasePayload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Since: 0.10.25
*/
GstFlowReturn
-gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
+gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
- GstBaseRTPPayload *basepayload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBasePayload *basepayload;
+ GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
- basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+ basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
if (payload_len == -1)
payload_len = gst_adapter_available (adapter);
buffer = gst_adapter_take_buffer (adapter, payload_len);
ret =
- gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer,
+ gst_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,
timestamp);
} else {
GstRTPBuffer rtp;
gst_rtp_buffer_unmap (&rtp);
/* set metadata */
- gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
+ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
- ret = gst_base_rtp_payload_push (basepayload, outbuf);
+ ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
* mtu and min/max_ptime values. We cache those so that we don't have to redo
* all the calculations */
static gboolean
-gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload *
+gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
basepayload, guint * min_payload_len, guint * max_payload_len,
guint * align)
{
- GstBaseRTPAudioPayload *payload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBaseAudioPayload *payload;
+ GstRTPBaseAudioPayloadPrivate *priv;
guint max_mtu, mtu;
guint maxptime_octets;
guint minptime_octets;
guint ptime_mult_octets;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
if (priv->align == 0)
return FALSE;
- mtu = GST_BASE_RTP_PAYLOAD_MTU (payload);
+ mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
/* check cached values */
if (G_LIKELY (priv->cached_mtu == mtu
/* frame conversions functions */
static GstClockTime
-gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes)
{
guint64 framecount;
}
static guint32
-gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes)
{
guint64 framecount;
time = framecount * payload->priv->frame_duration_ns;
return gst_util_uint64_scale_int (time,
- GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
+ GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);
}
static guint64
-gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time)
{
return gst_util_uint64_scale (time, payload->frame_size,
/* sample conversion functions */
static GstClockTime
-gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes)
{
guint64 rtptime;
rtptime = bytes;
return gst_util_uint64_scale_int (rtptime, GST_SECOND,
- GST_BASE_RTP_PAYLOAD (payload)->clock_rate);
+ GST_RTP_BASE_PAYLOAD (payload)->clock_rate);
}
static guint32
-gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes)
{
/* avoid division when we can */
}
static guint64
-gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload *
+gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
payload, guint64 time)
{
guint64 samples;
samples = gst_util_uint64_scale_int (time,
- GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND);
+ GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);
/* avoid multiplication when we can */
if (G_LIKELY (payload->sample_size != 8))
}
static GstFlowReturn
-gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
+gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
basepayload, GstBuffer * buffer)
{
- GstBaseRTPAudioPayload *payload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBaseAudioPayload *payload;
+ GstRTPBaseAudioPayloadPrivate *priv;
guint payload_len;
GstFlowReturn ret;
guint available;
ret = GST_FLOW_OK;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (payload, "Got DISCONT");
/* flush everything out of the adapter, mark DISCONT */
- ret = gst_base_rtp_audio_payload_flush (payload, -1, -1);
+ ret = gst_rtp_base_audio_payload_flush (payload, -1, -1);
priv->discont = TRUE;
/* get the distance between the timestamp gap and produce the same gap in
}
}
- if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len,
+ if (!gst_rtp_base_audio_payload_get_lengths (basepayload, &min_payload_len,
&max_payload_len, &align))
goto config_error;
/* If buffer fits on an RTP packet, let's just push it through
* this will check against max_ptime and max_mtu */
GST_DEBUG_OBJECT (payload, "Fast packet push");
- ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer, timestamp);
+ ret = gst_rtp_base_audio_payload_push_buffer (payload, buffer, timestamp);
} else {
/* push the buffer in the adapter */
gst_adapter_push (priv->adapter, buffer);
/* and flush out the bytes from the adapter, automatically set the
* timestamp. */
- ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1);
+ ret = gst_rtp_base_audio_payload_flush (payload, payload_len, -1);
available -= payload_len;
GST_DEBUG_OBJECT (payload, "available after push %u", available);
}
static GstStateChangeReturn
-gst_base_rtp_payload_audio_change_state (GstElement * element,
+gst_rtp_base_payload_audio_change_state (GstElement * element,
GstStateChange transition)
{
- GstBaseRTPAudioPayload *basertppayload;
+ GstRTPBaseAudioPayload *basertppayload;
GstStateChangeReturn ret;
- basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element);
+ basertppayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
}
static gboolean
-gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload * basep,
+gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep,
GstEvent * event)
{
- GstBaseRTPAudioPayload *payload;
+ GstRTPBaseAudioPayload *payload;
gboolean res = FALSE;
- payload = GST_BASE_RTP_AUDIO_PAYLOAD (basep);
+ payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* flush remaining bytes in the adapter */
- gst_base_rtp_audio_payload_flush (payload, -1, -1);
+ gst_rtp_base_audio_payload_flush (payload, -1, -1);
break;
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (payload->priv->adapter);
}
/* let parent handle the remainder of the event */
- res = GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_event (basep, event);
+ res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_event (basep, event);
return res;
}
/**
- * gst_base_rtp_audio_payload_get_adapter:
- * @basertpaudiopayload: a #GstBaseRTPAudioPayload
+ * gst_rtp_base_audio_payload_get_adapter:
+ * @basertpaudiopayload: a #GstRTPBaseAudioPayload
*
* Gets the internal adapter used by the depayloader.
*
* Since: 0.10.13
*/
GstAdapter *
-gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
+gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload
* basertpaudiopayload)
{
GstAdapter *adapter;
* Boston, MA 02111-1307, USA.
*/
-#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__
-#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__
+#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
+#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
G_BEGIN_DECLS
-typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
-typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
+typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
+typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;
-typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate;
+typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;
-#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
- (gst_base_rtp_audio_payload_get_type())
-#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
+#define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \
+ (gst_rtp_base_audio_payload_get_type())
+#define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), \
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
-#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
+ GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
+#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass), \
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayloadClass))
-#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
-#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
-#define GST_BASE_RTP_AUDIO_PAYLOAD_CAST(obj) \
- ((GstBaseRTPAudioPayload *) (obj))
-
-struct _GstBaseRTPAudioPayload
+ GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass))
+#define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
+#define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
+#define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \
+ ((GstRTPBaseAudioPayload *) (obj))
+
+struct _GstRTPBaseAudioPayload
{
- GstBaseRTPPayload payload;
+ GstRTPBasePayload payload;
- GstBaseRTPAudioPayloadPrivate *priv;
+ GstRTPBaseAudioPayloadPrivate *priv;
GstClockTime base_ts;
gint frame_size;
};
/**
- * GstBaseRTPAudioPayloadClass:
+ * GstRTPBaseAudioPayloadClass:
* @parent_class: the parent class
*
* Base class for audio RTP payloader.
*/
-struct _GstBaseRTPAudioPayloadClass
+struct _GstRTPBaseAudioPayloadClass
{
- GstBaseRTPPayloadClass parent_class;
+ GstRTPBasePayloadClass parent_class;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
-GType gst_base_rtp_audio_payload_get_type (void);
+GType gst_rtp_base_audio_payload_get_type (void);
/* configure frame based */
-void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
+void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *basertpaudiopayload);
-void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload,
+void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *basertpaudiopayload,
gint frame_duration, gint frame_size);
/* configure sample based */
-void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
-void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload,
+void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *basertpaudiopayload);
+void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *basertpaudiopayload,
gint sample_size);
-void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload,
+void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *basertpaudiopayload,
gint sample_size);
/* get the internal adapter */
-GstAdapter* gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload *basertpaudiopayload);
+GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *basertpaudiopayload);
/* push and flushing data */
-GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
+GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len,
GstClockTime timestamp);
-GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
+GstFlowReturn gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp);
G_END_DECLS
-#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
+#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
#define GST_CAT_DEFAULT (basertpdepayload_debug)
-#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
+#define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate))
-struct _GstBaseRTPDepayloadPrivate
+struct _GstRTPBaseDepayloadPrivate
{
GstClockTime npt_start;
GstClockTime npt_stop;
PROP_LAST
};
-static void gst_base_rtp_depayload_finalize (GObject * object);
-static void gst_base_rtp_depayload_set_property (GObject * object,
+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_base_rtp_depayload_get_property (GObject * object,
+static void gst_rtp_base_depayload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
-static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
+static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
GstBuffer * in);
-static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
+static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
GstEvent * event);
-static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
+static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
element, GstStateChange transition);
-static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload *
+static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
filter, GstEvent * event);
-static gboolean gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload *
+static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
filter, GstEvent * event);
static GstElementClass *parent_class = NULL;
-static void gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass *
+static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
klass);
-static void gst_base_rtp_depayload_init (GstBaseRTPDepayload * basertppayload,
- GstBaseRTPDepayloadClass * klass);
+static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * basertppayload,
+ GstRTPBaseDepayloadClass * klass);
GType
-gst_base_rtp_depayload_get_type (void)
+gst_rtp_base_depayload_get_type (void)
{
- static GType base_rtp_depayload_type = 0;
+ static GType rtp_base_depayload_type = 0;
- if (g_once_init_enter ((gsize *) & base_rtp_depayload_type)) {
- static const GTypeInfo base_rtp_depayload_info = {
- sizeof (GstBaseRTPDepayloadClass),
+ if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
+ static const GTypeInfo rtp_base_depayload_info = {
+ sizeof (GstRTPBaseDepayloadClass),
NULL,
NULL,
- (GClassInitFunc) gst_base_rtp_depayload_class_init,
+ (GClassInitFunc) gst_rtp_base_depayload_class_init,
NULL,
NULL,
- sizeof (GstBaseRTPDepayload),
+ sizeof (GstRTPBaseDepayload),
0,
- (GInstanceInitFunc) gst_base_rtp_depayload_init,
+ (GInstanceInitFunc) gst_rtp_base_depayload_init,
};
- g_once_init_leave ((gsize *) & base_rtp_depayload_type,
- g_type_register_static (GST_TYPE_ELEMENT, "GstBaseRTPDepayload",
- &base_rtp_depayload_info, G_TYPE_FLAG_ABSTRACT));
+ g_once_init_leave ((gsize *) & rtp_base_depayload_type,
+ g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload",
+ &rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT));
}
- return base_rtp_depayload_type;
+ return rtp_base_depayload_type;
}
static void
-gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
+gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
- g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
+ g_type_class_add_private (klass, sizeof (GstRTPBaseDepayloadPrivate));
- 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;
+ 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;
- gstelement_class->change_state = gst_base_rtp_depayload_change_state;
+ gstelement_class->change_state = gst_rtp_base_depayload_change_state;
- klass->packet_lost = gst_base_rtp_depayload_packet_lost;
- klass->handle_event = gst_base_rtp_depayload_handle_event;
+ klass->packet_lost = gst_rtp_base_depayload_packet_lost;
+ klass->handle_event = gst_rtp_base_depayload_handle_event;
GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
"Base class for RTP Depayloaders");
}
static void
-gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
- GstBaseRTPDepayloadClass * klass)
+gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
+ GstRTPBaseDepayloadClass * klass)
{
GstPadTemplate *pad_template;
- GstBaseRTPDepayloadPrivate *priv;
+ GstRTPBaseDepayloadPrivate *priv;
- priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
+ priv = GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE (filter);
filter->priv = priv;
GST_DEBUG_OBJECT (filter, "init");
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_base_rtp_depayload_chain);
+ gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
gst_pad_set_event_function (filter->sinkpad,
- gst_base_rtp_depayload_handle_sink_event);
+ gst_rtp_base_depayload_handle_sink_event);
gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
pad_template =
}
static void
-gst_base_rtp_depayload_finalize (GObject * object)
+gst_rtp_base_depayload_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
-gst_base_rtp_depayload_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps)
+gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
- GstBaseRTPDepayloadClass *bclass;
- GstBaseRTPDepayloadPrivate *priv;
+ GstRTPBaseDepayloadClass *bclass;
+ GstRTPBaseDepayloadPrivate *priv;
gboolean res;
GstStructure *caps_struct;
const GValue *value;
priv = filter->priv;
- bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
GST_DEBUG_OBJECT (filter, "Set caps");
}
static GstFlowReturn
-gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
+gst_rtp_base_depayload_chain (GstPad * pad, GstBuffer * in)
{
- GstBaseRTPDepayload *filter;
- GstBaseRTPDepayloadPrivate *priv;
- GstBaseRTPDepayloadClass *bclass;
+ GstRTPBaseDepayload *filter;
+ GstRTPBaseDepayloadPrivate *priv;
+ GstRTPBaseDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime pts, dts;
gint gap;
GstRTPBuffer rtp;
- filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
+ filter = GST_RTP_BASE_DEPAYLOAD (GST_OBJECT_PARENT (pad));
priv = filter->priv;
/* we must have a setcaps first */
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
- bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ bclass = GST_RTP_BASE_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) {
- ret = gst_base_rtp_depayload_push (filter, out_buf);
+ ret = gst_rtp_base_depayload_push (filter, out_buf);
}
gst_buffer_unref (in);
}
static gboolean
-gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter,
+gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter,
GstEvent * event)
{
gboolean res = TRUE;
gst_event_parse_caps (event, &caps);
- res = gst_base_rtp_depayload_setcaps (filter, caps);
+ res = gst_rtp_base_depayload_setcaps (filter, caps);
forward = FALSE;
break;
}
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
- GstBaseRTPDepayloadClass *bclass;
+ GstRTPBaseDepayloadClass *bclass;
- bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ 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
}
static gboolean
-gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
+gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res = FALSE;
- GstBaseRTPDepayload *filter;
- GstBaseRTPDepayloadClass *bclass;
+ GstRTPBaseDepayload *filter;
+ GstRTPBaseDepayloadClass *bclass;
- filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
+ filter = GST_RTP_BASE_DEPAYLOAD (gst_pad_get_parent (pad));
if (G_UNLIKELY (filter == NULL)) {
gst_event_unref (event);
return FALSE;
}
- bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
if (bclass->handle_event)
res = bclass->handle_event (filter, event);
else
}
static GstEvent *
-create_segment_event (GstBaseRTPDepayload * filter, gboolean update,
+create_segment_event (GstRTPBaseDepayload * filter, gboolean update,
GstClockTime position)
{
GstEvent *event;
GstClockTime stop;
- GstBaseRTPDepayloadPrivate *priv;
+ GstRTPBaseDepayloadPrivate *priv;
GstSegment segment;
priv = filter->priv;
typedef struct
{
- GstBaseRTPDepayload *depayload;
- GstBaseRTPDepayloadClass *bclass;
+ GstRTPBaseDepayload *depayload;
+ GstRTPBaseDepayloadClass *bclass;
} HeaderData;
static gboolean
set_headers (GstBuffer ** buffer, guint idx, HeaderData * data)
{
- GstBaseRTPDepayload *depayload = data->depayload;
- GstBaseRTPDepayloadPrivate *priv = depayload->priv;
+ GstRTPBaseDepayload *depayload = data->depayload;
+ GstRTPBaseDepayloadPrivate *priv = depayload->priv;
GstClockTime pts, dts, duration;
*buffer = gst_buffer_make_writable (*buffer);
}
static GstFlowReturn
-gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter,
+gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter,
gboolean is_list, gpointer obj)
{
HeaderData data;
data.depayload = filter;
- data.bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ data.bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
if (is_list) {
GstBufferList **blist = obj;
}
/**
- * gst_base_rtp_depayload_push:
- * @filter: a #GstBaseRTPDepayload
+ * gst_rtp_base_depayload_push:
+ * @filter: a #GstRTPBaseDepayload
* @out_buf: a #GstBuffer
*
* Push @out_buf to the peer of @filter. This function takes ownership of
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
+gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
{
GstFlowReturn res;
- res = gst_base_rtp_depayload_prepare_push (filter, FALSE, &out_buf);
+ res = gst_rtp_base_depayload_prepare_push (filter, FALSE, &out_buf);
if (G_LIKELY (res == GST_FLOW_OK))
res = gst_pad_push (filter->srcpad, out_buf);
}
/**
- * gst_base_rtp_depayload_push_list:
- * @filter: a #GstBaseRTPDepayload
+ * gst_rtp_base_depayload_push_list:
+ * @filter: a #GstRTPBaseDepayload
* @out_list: a #GstBufferList
*
* Push @out_list to the peer of @filter. This function takes ownership of
* Since: 0.10.32
*/
GstFlowReturn
-gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter,
+gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
GstBufferList * out_list)
{
GstFlowReturn res;
- res = gst_base_rtp_depayload_prepare_push (filter, TRUE, &out_list);
+ res = gst_rtp_base_depayload_prepare_push (filter, TRUE, &out_list);
if (G_LIKELY (res == GST_FLOW_OK))
res = gst_pad_push_list (filter->srcpad, out_list);
/* 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,
+gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter,
GstEvent * event)
{
GstClockTime timestamp, duration, position;
}
static GstStateChangeReturn
-gst_base_rtp_depayload_change_state (GstElement * element,
+gst_rtp_base_depayload_change_state (GstElement * element,
GstStateChange transition)
{
- GstBaseRTPDepayload *filter;
- GstBaseRTPDepayloadPrivate *priv;
+ GstRTPBaseDepayload *filter;
+ GstRTPBaseDepayloadPrivate *priv;
GstStateChangeReturn ret;
- filter = GST_BASE_RTP_DEPAYLOAD (element);
+ filter = GST_RTP_BASE_DEPAYLOAD (element);
priv = filter->priv;
switch (transition) {
}
static void
-gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
+gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
}
static void
-gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
+gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
* Boston, MA 02111-1307, USA.
*/
-#ifndef __GST_BASE_RTP_DEPAYLOAD_H__
-#define __GST_BASE_RTP_DEPAYLOAD_H__
+#ifndef __GST_RTP_BASE_DEPAYLOAD_H__
+#define __GST_RTP_BASE_DEPAYLOAD_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
G_BEGIN_DECLS
-#define GST_TYPE_BASE_RTP_DEPAYLOAD (gst_base_rtp_depayload_get_type())
-#define GST_BASE_RTP_DEPAYLOAD(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayload))
-#define GST_BASE_RTP_DEPAYLOAD_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayloadClass))
-#define GST_BASE_RTP_DEPAYLOAD_GET_CLASS(obj) \
- (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayloadClass))
-#define GST_IS_BASE_RTP_DEPAYLOAD(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_DEPAYLOAD))
-#define GST_IS_BASE_RTP_DEPAYLOAD_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_DEPAYLOAD))
-
-#define GST_BASE_RTP_DEPAYLOAD_SINKPAD(depayload) (GST_BASE_RTP_DEPAYLOAD (depayload)->sinkpad)
-#define GST_BASE_RTP_DEPAYLOAD_SRCPAD(depayload) (GST_BASE_RTP_DEPAYLOAD (depayload)->srcpad)
-
-typedef struct _GstBaseRTPDepayload GstBaseRTPDepayload;
-typedef struct _GstBaseRTPDepayloadClass GstBaseRTPDepayloadClass;
-typedef struct _GstBaseRTPDepayloadPrivate GstBaseRTPDepayloadPrivate;
-
-struct _GstBaseRTPDepayload
+#define GST_TYPE_RTP_BASE_DEPAYLOAD (gst_rtp_base_depayload_get_type())
+#define GST_RTP_BASE_DEPAYLOAD(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayload))
+#define GST_RTP_BASE_DEPAYLOAD_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass))
+#define GST_RTP_BASE_DEPAYLOAD_GET_CLASS(obj) \
+ (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass))
+#define GST_IS_RTP_BASE_DEPAYLOAD(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_DEPAYLOAD))
+#define GST_IS_RTP_BASE_DEPAYLOAD_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_DEPAYLOAD))
+
+#define GST_RTP_BASE_DEPAYLOAD_SINKPAD(depayload) (GST_RTP_BASE_DEPAYLOAD (depayload)->sinkpad)
+#define GST_RTP_BASE_DEPAYLOAD_SRCPAD(depayload) (GST_RTP_BASE_DEPAYLOAD (depayload)->srcpad)
+
+typedef struct _GstRTPBaseDepayload GstRTPBaseDepayload;
+typedef struct _GstRTPBaseDepayloadClass GstRTPBaseDepayloadClass;
+typedef struct _GstRTPBaseDepayloadPrivate GstRTPBaseDepayloadPrivate;
+
+struct _GstRTPBaseDepayload
{
GstElement parent;
gboolean need_newsegment;
/*< private >*/
- GstBaseRTPDepayloadPrivate *priv;
+ GstRTPBaseDepayloadPrivate *priv;
gpointer _gst_reserved[GST_PADDING-1];
};
/**
- * GstBaseRTPDepayloadClass:
+ * GstRTPBaseDepayloadClass:
* @parent_class: the parent class
* @set_caps: configure the depayloader
- * @add_to_queue: (deprecated)
* @process: process incoming rtp packets
- * @set_gst_timestamp: convert from RTP timestamp to GST timestamp
* @packet_lost: signal the depayloader about packet loss
* @handle_event: custom event handling
*
* Base class for audio RTP payloader.
*/
-struct _GstBaseRTPDepayloadClass
+struct _GstRTPBaseDepayloadClass
{
GstElementClass parent_class;
/* virtuals, inform the subclass of the caps. */
- gboolean (*set_caps) (GstBaseRTPDepayload *filter, GstCaps *caps);
+ gboolean (*set_caps) (GstRTPBaseDepayload *filter, GstCaps *caps);
/* pure virtual function, child must use this to process incoming
* rtp packets. If the child returns a buffer without a valid timestamp,
* the timestamp of @in will be applied to the result buffer and the
* buffer will be pushed. If this function returns %NULL, nothing is
* pushed. */
- GstBuffer * (*process) (GstBaseRTPDepayload *base, GstBuffer *in);
+ GstBuffer * (*process) (GstRTPBaseDepayload *base, GstBuffer *in);
/* non-pure function used to to signal the depayloader about packet loss. the
* timestamp and duration are the estimated values of the lost packet.
* The default implementation of this message pushes a segment update. */
- gboolean (*packet_lost) (GstBaseRTPDepayload *filter, GstEvent *event);
+ gboolean (*packet_lost) (GstRTPBaseDepayload *filter, GstEvent *event);
/* the default implementation does the default actions for events but
* implementation can override.
* Since: 0.10.32 */
- gboolean (*handle_event) (GstBaseRTPDepayload * filter, GstEvent * event);
+ gboolean (*handle_event) (GstRTPBaseDepayload * filter, GstEvent * event);
/*< private >*/
gpointer _gst_reserved[GST_PADDING-2];
};
-GType gst_base_rtp_depayload_get_type (void);
+GType gst_rtp_base_depayload_get_type (void);
-GstFlowReturn gst_base_rtp_depayload_push (GstBaseRTPDepayload *filter, GstBuffer *out_buf);
-GstFlowReturn gst_base_rtp_depayload_push_list (GstBaseRTPDepayload *filter, GstBufferList *out_list);
+GstFlowReturn gst_rtp_base_depayload_push (GstRTPBaseDepayload *filter, GstBuffer *out_buf);
+GstFlowReturn gst_rtp_base_depayload_push_list (GstRTPBaseDepayload *filter, GstBufferList *out_list);
G_END_DECLS
-#endif /* __GST_BASE_RTP_DEPAYLOAD_H__ */
+#endif /* __GST_RTP_BASE_DEPAYLOAD_H__ */
GST_DEBUG_CATEGORY_STATIC (basertppayload_debug);
#define GST_CAT_DEFAULT (basertppayload_debug)
-#define GST_BASE_RTP_PAYLOAD_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_PAYLOAD, GstBaseRTPPayloadPrivate))
+#define GST_RTP_BASE_PAYLOAD_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadPrivate))
-struct _GstBaseRTPPayloadPrivate
+struct _GstRTPBasePayloadPrivate
{
gboolean ts_offset_random;
gboolean seqnum_offset_random;
gint64 caps_max_ptime;
};
-/* BaseRTPPayload signals and args */
+/* RTPBasePayload signals and args */
enum
{
/* FILL ME */
PROP_LAST
};
-static void gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass);
-static void gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload,
+static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass);
+static void gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload,
gpointer g_class);
-static void gst_base_rtp_payload_finalize (GObject * object);
+static void gst_rtp_base_payload_finalize (GObject * object);
-static GstCaps *gst_base_rtp_payload_sink_getcaps (GstPad * pad,
+static GstCaps *gst_rtp_base_payload_sink_getcaps (GstPad * pad,
GstCaps * filter);
-static gboolean gst_base_rtp_payload_event_default (GstBaseRTPPayload *
+static gboolean gst_rtp_base_payload_event_default (GstRTPBasePayload *
basertppayload, GstEvent * event);
-static gboolean gst_base_rtp_payload_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn gst_base_rtp_payload_chain (GstPad * pad,
+static gboolean gst_rtp_base_payload_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad,
GstBuffer * buffer);
-static GstCaps *gst_base_rtp_payload_getcaps_default (GstBaseRTPPayload *
+static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload *
basertppayload, GstPad * pad, GstCaps * filter);
-static void gst_base_rtp_payload_set_property (GObject * object, guint prop_id,
+static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void gst_base_rtp_payload_get_property (GObject * object, guint prop_id,
+static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static GstStateChangeReturn gst_base_rtp_payload_change_state (GstElement *
+static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement *
element, GstStateChange transition);
static GstElementClass *parent_class = NULL;
/* FIXME 0.11: API should be changed to gst_base_typ_payload_xyz */
GType
-gst_base_rtp_payload_get_type (void)
+gst_rtp_base_payload_get_type (void)
{
static GType basertppayload_type = 0;
if (g_once_init_enter ((gsize *) & basertppayload_type)) {
static const GTypeInfo basertppayload_info = {
- sizeof (GstBaseRTPPayloadClass),
+ sizeof (GstRTPBasePayloadClass),
NULL,
NULL,
- (GClassInitFunc) gst_base_rtp_payload_class_init,
+ (GClassInitFunc) gst_rtp_base_payload_class_init,
NULL,
NULL,
- sizeof (GstBaseRTPPayload),
+ sizeof (GstRTPBasePayload),
0,
- (GInstanceInitFunc) gst_base_rtp_payload_init,
+ (GInstanceInitFunc) gst_rtp_base_payload_init,
};
g_once_init_leave ((gsize *) & basertppayload_type,
- g_type_register_static (GST_TYPE_ELEMENT, "GstBaseRTPPayload",
+ g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload",
&basertppayload_info, G_TYPE_FLAG_ABSTRACT));
}
return basertppayload_type;
}
static void
-gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass)
+gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- g_type_class_add_private (klass, sizeof (GstBaseRTPPayloadPrivate));
+ g_type_class_add_private (klass, sizeof (GstRTPBasePayloadPrivate));
parent_class = g_type_class_peek_parent (klass);
- gobject_class->finalize = gst_base_rtp_payload_finalize;
+ gobject_class->finalize = gst_rtp_base_payload_finalize;
- gobject_class->set_property = gst_base_rtp_payload_set_property;
- gobject_class->get_property = gst_base_rtp_payload_get_property;
+ gobject_class->set_property = gst_rtp_base_payload_set_property;
+ gobject_class->get_property = gst_rtp_base_payload_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MTU,
g_param_spec_uint ("mtu", "MTU",
-1, G_MAXINT64, DEFAULT_MAX_PTIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseRTPAudioPayload:min-ptime:
+ * GstRTPBaseAudioPayload:min-ptime:
*
* Minimum duration of the packet data in ns (can't go above MTU)
*
0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseRTPAudioPayload:perfect-rtptime:
+ * GstRTPBaseAudioPayload:perfect-rtptime:
*
* Try to use the offset fields to generate perfect RTP timestamps. when this
* option is disabled, RTP timestamps are generated from the GStreamer
"Generate perfect RTP timestamps when possible",
DEFAULT_PERFECT_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseRTPAudioPayload:ptime-multiple:
+ * GstRTPBaseAudioPayload:ptime-multiple:
*
* Force buffers to be multiples of this duration in ns (0 disables)
*
0, G_MAXINT64, DEFAULT_PTIME_MULTIPLE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- gstelement_class->change_state = gst_base_rtp_payload_change_state;
+ gstelement_class->change_state = gst_rtp_base_payload_change_state;
- klass->get_caps = gst_base_rtp_payload_getcaps_default;
- klass->handle_event = gst_base_rtp_payload_event_default;
+ klass->get_caps = gst_rtp_base_payload_getcaps_default;
+ klass->handle_event = gst_rtp_base_payload_event_default;
GST_DEBUG_CATEGORY_INIT (basertppayload_debug, "basertppayload", 0,
"Base class for RTP Payloaders");
}
static void
-gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload, gpointer g_class)
+gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload, gpointer g_class)
{
GstPadTemplate *templ;
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayloadPrivate *priv;
basertppayload->priv = priv =
- GST_BASE_RTP_PAYLOAD_GET_PRIVATE (basertppayload);
+ GST_RTP_BASE_PAYLOAD_GET_PRIVATE (basertppayload);
templ =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
basertppayload->sinkpad = gst_pad_new_from_template (templ, "sink");
gst_pad_set_getcaps_function (basertppayload->sinkpad,
- gst_base_rtp_payload_sink_getcaps);
+ gst_rtp_base_payload_sink_getcaps);
gst_pad_set_event_function (basertppayload->sinkpad,
- gst_base_rtp_payload_event);
+ gst_rtp_base_payload_event);
gst_pad_set_chain_function (basertppayload->sinkpad,
- gst_base_rtp_payload_chain);
+ gst_rtp_base_payload_chain);
gst_element_add_pad (GST_ELEMENT (basertppayload), basertppayload->sinkpad);
basertppayload->mtu = DEFAULT_MTU;
}
static void
-gst_base_rtp_payload_finalize (GObject * object)
+gst_rtp_base_payload_finalize (GObject * object)
{
- GstBaseRTPPayload *basertppayload;
+ GstRTPBasePayload *basertppayload;
- basertppayload = GST_BASE_RTP_PAYLOAD (object);
+ basertppayload = GST_RTP_BASE_PAYLOAD (object);
g_free (basertppayload->media);
basertppayload->media = NULL;
}
static GstCaps *
-gst_base_rtp_payload_getcaps_default (GstBaseRTPPayload * basertppayload,
+gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * basertppayload,
GstPad * pad, GstCaps * filter)
{
GstCaps *caps;
}
static GstCaps *
-gst_base_rtp_payload_sink_getcaps (GstPad * pad, GstCaps * filter)
+gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadClass *basertppayload_class;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadClass *basertppayload_class;
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (pad, "getting caps");
- basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
- basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
+ basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
+ basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->get_caps)
caps = basertppayload_class->get_caps (basertppayload, pad, filter);
}
static gboolean
-gst_base_rtp_payload_event_default (GstBaseRTPPayload * basertppayload,
+gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload,
GstEvent * event)
{
gboolean res = FALSE;
break;
case GST_EVENT_CAPS:
{
- GstBaseRTPPayloadClass *basertppayload_class;
+ GstRTPBasePayloadClass *basertppayload_class;
GstCaps *caps;
gst_event_parse_caps (event, &caps);
GST_DEBUG_OBJECT (basertppayload, "setting caps %" GST_PTR_FORMAT, caps);
- basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
+ basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->set_caps)
res = basertppayload_class->set_caps (basertppayload, caps);
}
static gboolean
-gst_base_rtp_payload_event (GstPad * pad, GstEvent * event)
+gst_rtp_base_payload_event (GstPad * pad, GstEvent * event)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadClass *basertppayload_class;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadClass *basertppayload_class;
gboolean res = FALSE;
- basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
+ basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
if (G_UNLIKELY (basertppayload == NULL)) {
gst_event_unref (event);
return FALSE;
}
- basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
+ basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload);
if (basertppayload_class->handle_event)
res = basertppayload_class->handle_event (basertppayload, event);
static GstFlowReturn
-gst_base_rtp_payload_chain (GstPad * pad, GstBuffer * buffer)
+gst_rtp_base_payload_chain (GstPad * pad, GstBuffer * buffer)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadClass *basertppayload_class;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadClass *basertppayload_class;
GstFlowReturn ret;
- basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad));
- basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload);
+ basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad));
+ basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload);
if (!basertppayload_class->handle_buffer)
goto no_function;
}
/**
- * gst_base_rtp_payload_set_options:
- * @payload: a #GstBaseRTPPayload
+ * gst_rtp_base_payload_set_options:
+ * @payload: a #GstRTPBasePayload
* @media: the media type (typically "audio" or "video")
* @dynamic: if the payload type is dynamic
* @encoding_name: the encoding name
*
* Set the rtp options of the payloader. These options will be set in the caps
* of the payloader. Subclasses must call this method before calling
- * gst_base_rtp_payload_push() or gst_base_rtp_payload_set_outcaps().
+ * gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps().
*/
void
-gst_base_rtp_payload_set_options (GstBaseRTPPayload * payload,
+gst_rtp_base_payload_set_options (GstRTPBasePayload * payload,
const gchar * media, gboolean dynamic, const gchar * encoding_name,
guint32 clock_rate)
{
}
static void
-update_max_ptime (GstBaseRTPPayload * basertppayload)
+update_max_ptime (GstRTPBasePayload * basertppayload)
{
if (basertppayload->priv->caps_max_ptime != -1 &&
basertppayload->priv->prop_max_ptime != -1)
}
/**
- * gst_base_rtp_payload_set_outcaps:
- * @payload: a #GstBaseRTPPayload
+ * gst_rtp_base_payload_set_outcaps:
+ * @payload: a #GstRTPBasePayload
* @fieldname: the first field name or %NULL
* @...: field values
*
* Returns: %TRUE if the caps could be set.
*/
gboolean
-gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload,
+gst_rtp_base_payload_set_outcaps (GstRTPBasePayload * payload,
const gchar * fieldname, ...)
{
GstCaps *srccaps, *peercaps;
if (peercaps == NULL) {
/* no peer caps, just add the other properties */
gst_caps_set_simple (srccaps,
- "payload", G_TYPE_INT, GST_BASE_RTP_PAYLOAD_PT (payload),
+ "payload", G_TYPE_INT, GST_RTP_BASE_PAYLOAD_PT (payload),
"ssrc", G_TYPE_UINT, payload->current_ssrc,
"timestamp-offset", G_TYPE_UINT, payload->ts_base,
"seqnum-offset", G_TYPE_UINT, payload->seqnum_base, NULL);
if (gst_structure_get_int (s, "payload", &pt)) {
/* use peer pt */
- GST_BASE_RTP_PAYLOAD_PT (payload) = pt;
+ GST_RTP_BASE_PAYLOAD_PT (payload) = pt;
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
} else {
if (gst_structure_has_field (s, "payload")) {
/* can only fixate if there is a field */
gst_structure_fixate_field_nearest_int (s, "payload",
- GST_BASE_RTP_PAYLOAD_PT (payload));
+ GST_RTP_BASE_PAYLOAD_PT (payload));
gst_structure_get_int (s, "payload", &pt);
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
} else {
/* no pt field, use the internal pt */
- pt = GST_BASE_RTP_PAYLOAD_PT (payload);
+ pt = GST_RTP_BASE_PAYLOAD_PT (payload);
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
GST_LOG_OBJECT (payload, "using internal pt %d", pt);
}
update_max_ptime (payload);
- res = gst_pad_set_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (payload), srccaps);
+ res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps);
gst_caps_unref (srccaps);
return res;
}
/**
- * gst_base_rtp_payload_is_filled:
- * @payload: a #GstBaseRTPPayload
+ * gst_rtp_base_payload_is_filled:
+ * @payload: a #GstRTPBasePayload
* @size: the size of the packet
* @duration: the duration of the packet
*
* configured MTU or max_ptime.
*/
gboolean
-gst_base_rtp_payload_is_filled (GstBaseRTPPayload * payload,
+gst_rtp_base_payload_is_filled (GstRTPBasePayload * payload,
guint size, GstClockTime duration)
{
if (size > payload->mtu)
typedef struct
{
- GstBaseRTPPayload *payload;
+ GstRTPBasePayload *payload;
guint32 ssrc;
guint16 seqnum;
guint8 pt;
/* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
* before the buffer is pushed. */
static GstFlowReturn
-gst_base_rtp_payload_prepare_push (GstBaseRTPPayload * payload,
+gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
gpointer obj, gboolean is_list)
{
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayloadPrivate *priv;
HeaderData data;
if (payload->clock_rate == 0)
}
/**
- * gst_base_rtp_payload_push_list:
- * @payload: a #GstBaseRTPPayload
+ * gst_rtp_base_payload_push_list:
+ * @payload: a #GstRTPBasePayload
* @list: a #GstBufferList
*
* Push @list to the peer element of the payloader. The SSRC, payload type,
* Since: 0.10.24
*/
GstFlowReturn
-gst_base_rtp_payload_push_list (GstBaseRTPPayload * payload,
+gst_rtp_base_payload_push_list (GstRTPBasePayload * payload,
GstBufferList * list)
{
GstFlowReturn res;
- res = gst_base_rtp_payload_prepare_push (payload, list, TRUE);
+ res = gst_rtp_base_payload_prepare_push (payload, list, TRUE);
if (G_LIKELY (res == GST_FLOW_OK))
res = gst_pad_push_list (payload->srcpad, list);
}
/**
- * gst_base_rtp_payload_push:
- * @payload: a #GstBaseRTPPayload
+ * gst_rtp_base_payload_push:
+ * @payload: a #GstRTPBasePayload
* @buffer: a #GstBuffer
*
* Push @buffer to the peer element of the payloader. The SSRC, payload type,
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-gst_base_rtp_payload_push (GstBaseRTPPayload * payload, GstBuffer * buffer)
+gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstFlowReturn res;
- res = gst_base_rtp_payload_prepare_push (payload, buffer, FALSE);
+ res = gst_rtp_base_payload_prepare_push (payload, buffer, FALSE);
if (G_LIKELY (res == GST_FLOW_OK))
res = gst_pad_push (payload->srcpad, buffer);
}
static void
-gst_base_rtp_payload_set_property (GObject * object, guint prop_id,
+gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadPrivate *priv;
gint64 val;
- basertppayload = GST_BASE_RTP_PAYLOAD (object);
+ basertppayload = GST_RTP_BASE_PAYLOAD (object);
priv = basertppayload->priv;
switch (prop_id) {
}
static void
-gst_base_rtp_payload_get_property (GObject * object, guint prop_id,
+gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadPrivate *priv;
- basertppayload = GST_BASE_RTP_PAYLOAD (object);
+ basertppayload = GST_RTP_BASE_PAYLOAD (object);
priv = basertppayload->priv;
switch (prop_id) {
}
static GstStateChangeReturn
-gst_base_rtp_payload_change_state (GstElement * element,
+gst_rtp_base_payload_change_state (GstElement * element,
GstStateChange transition)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayload *basertppayload;
+ GstRTPBasePayloadPrivate *priv;
GstStateChangeReturn ret;
- basertppayload = GST_BASE_RTP_PAYLOAD (element);
+ basertppayload = GST_RTP_BASE_PAYLOAD (element);
priv = basertppayload->priv;
switch (transition) {
* Boston, MA 02111-1307, USA.
*/
-#ifndef __GST_BASE_RTP_PAYLOAD_H__
-#define __GST_BASE_RTP_PAYLOAD_H__
+#ifndef __GST_RTP_BASE_PAYLOAD_H__
+#define __GST_RTP_BASE_PAYLOAD_H__
#include <gst/gst.h>
G_BEGIN_DECLS
-#define GST_TYPE_BASE_RTP_PAYLOAD \
- (gst_base_rtp_payload_get_type())
-#define GST_BASE_RTP_PAYLOAD(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_PAYLOAD,GstBaseRTPPayload))
-#define GST_BASE_RTP_PAYLOAD_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_PAYLOAD,GstBaseRTPPayloadClass))
-#define GST_BASE_RTP_PAYLOAD_GET_CLASS(obj) \
- (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_RTP_PAYLOAD, GstBaseRTPPayloadClass))
-#define GST_IS_BASE_RTP_PAYLOAD(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_PAYLOAD))
-#define GST_IS_BASE_RTP_PAYLOAD_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_PAYLOAD))
-#define GST_BASE_RTP_PAYLOAD_CAST(obj) \
- ((GstBaseRTPPayload*)(obj))
-
-typedef struct _GstBaseRTPPayload GstBaseRTPPayload;
-typedef struct _GstBaseRTPPayloadPrivate GstBaseRTPPayloadPrivate;
-typedef struct _GstBaseRTPPayloadClass GstBaseRTPPayloadClass;
+#define GST_TYPE_RTP_BASE_PAYLOAD \
+ (gst_rtp_base_payload_get_type())
+#define GST_RTP_BASE_PAYLOAD(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayload))
+#define GST_RTP_BASE_PAYLOAD_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayloadClass))
+#define GST_RTP_BASE_PAYLOAD_GET_CLASS(obj) \
+ (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadClass))
+#define GST_IS_RTP_BASE_PAYLOAD(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_PAYLOAD))
+#define GST_IS_RTP_BASE_PAYLOAD_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_PAYLOAD))
+#define GST_RTP_BASE_PAYLOAD_CAST(obj) \
+ ((GstRTPBasePayload*)(obj))
+
+typedef struct _GstRTPBasePayload GstRTPBasePayload;
+typedef struct _GstRTPBasePayloadPrivate GstRTPBasePayloadPrivate;
+typedef struct _GstRTPBasePayloadClass GstRTPBasePayloadClass;
/**
- * GST_BASE_RTP_PAYLOAD_SINKPAD:
- * @payload: a #GstBaseRTPPayload
+ * GST_RTP_BASE_PAYLOAD_SINKPAD:
+ * @payload: a #GstRTPBasePayload
*
* Get access to the sinkpad of @payload.
*/
-#define GST_BASE_RTP_PAYLOAD_SINKPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->sinkpad)
+#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad)
/**
- * GST_BASE_RTP_PAYLOAD_SRCPAD:
- * @payload: a #GstBaseRTPPayload
+ * GST_RTP_BASE_PAYLOAD_SRCPAD:
+ * @payload: a #GstRTPBasePayload
*
* Get access to the srcpad of @payload.
*/
-#define GST_BASE_RTP_PAYLOAD_SRCPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->srcpad)
+#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->srcpad)
/**
- * GST_BASE_RTP_PAYLOAD_PT:
- * @payload: a #GstBaseRTPPayload
+ * GST_RTP_BASE_PAYLOAD_PT:
+ * @payload: a #GstRTPBasePayload
*
* Get access to the configured payload type of @payload.
*/
-#define GST_BASE_RTP_PAYLOAD_PT(payload) (GST_BASE_RTP_PAYLOAD (payload)->pt)
+#define GST_RTP_BASE_PAYLOAD_PT(payload) (GST_RTP_BASE_PAYLOAD (payload)->pt)
/**
- * GST_BASE_RTP_PAYLOAD_MTU:
- * @payload: a #GstBaseRTPPayload
+ * GST_RTP_BASE_PAYLOAD_MTU:
+ * @payload: a #GstRTPBasePayload
*
* Get access to the configured MTU of @payload.
*/
-#define GST_BASE_RTP_PAYLOAD_MTU(payload) (GST_BASE_RTP_PAYLOAD (payload)->mtu)
+#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu)
-struct _GstBaseRTPPayload
+struct _GstRTPBasePayload
{
GstElement element;
guint64 min_ptime;
/*< private >*/
- GstBaseRTPPayloadPrivate *priv;
+ GstRTPBasePayloadPrivate *priv;
union {
struct {
};
/**
- * GstBaseRTPPayloadClass:
+ * GstRTPBasePayloadClass:
* @parent_class: the parent class
* @set_caps: configure the payloader
* @handle_buffer: process data
*
* Base class for audio RTP payloader.
*/
-struct _GstBaseRTPPayloadClass
+struct _GstRTPBasePayloadClass
{
GstElementClass parent_class;
/* query accepted caps */
- GstCaps * (*get_caps) (GstBaseRTPPayload *payload, GstPad * pad, GstCaps * filter);
+ GstCaps * (*get_caps) (GstRTPBasePayload *payload, GstPad * pad, GstCaps * filter);
/* receive caps on the sink pad, configure the payloader. */
- gboolean (*set_caps) (GstBaseRTPPayload *payload, GstCaps *caps);
+ gboolean (*set_caps) (GstRTPBasePayload *payload, GstCaps *caps);
- /* handle a buffer, perform 0 or more gst_base_rtp_payload_push() on
+ /* handle a buffer, perform 0 or more gst_rtp_base_payload_push() on
* the RTP buffers. This function takes ownership of the buffer. */
- GstFlowReturn (*handle_buffer) (GstBaseRTPPayload *payload,
+ GstFlowReturn (*handle_buffer) (GstRTPBasePayload *payload,
GstBuffer *buffer);
- gboolean (*handle_event) (GstBaseRTPPayload *payload, GstEvent * event);
+ gboolean (*handle_event) (GstRTPBasePayload *payload, GstEvent * event);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
-GType gst_base_rtp_payload_get_type (void);
+GType gst_rtp_base_payload_get_type (void);
-void gst_base_rtp_payload_set_options (GstBaseRTPPayload *payload,
+void gst_rtp_base_payload_set_options (GstRTPBasePayload *payload,
const gchar *media,
gboolean dynamic,
const gchar *encoding_name,
guint32 clock_rate);
-gboolean gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload *payload,
+gboolean gst_rtp_base_payload_set_outcaps (GstRTPBasePayload *payload,
const gchar *fieldname, ...);
-gboolean gst_base_rtp_payload_is_filled (GstBaseRTPPayload *payload,
+gboolean gst_rtp_base_payload_is_filled (GstRTPBasePayload *payload,
guint size, GstClockTime duration);
-GstFlowReturn gst_base_rtp_payload_push (GstBaseRTPPayload *payload,
+GstFlowReturn gst_rtp_base_payload_push (GstRTPBasePayload *payload,
GstBuffer *buffer);
-GstFlowReturn gst_base_rtp_payload_push_list (GstBaseRTPPayload *payload,
+GstFlowReturn gst_rtp_base_payload_push_list (GstRTPBasePayload *payload,
GstBufferList *list);
G_END_DECLS
-#endif /* __GST_BASE_RTP_PAYLOAD_H__ */
+#endif /* __GST_RTP_BASE_PAYLOAD_H__ */
/**
* SECTION:gstrtcpbuffer
* @short_description: Helper methods for dealing with RTCP buffers
- * @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, #gstrtpbuffer
+ * @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, #gstrtpbuffer
*
* Note: The API in this module is not yet declared stable.
*
/**
* SECTION:gstrtpbuffer
* @short_description: Helper methods for dealing with RTP buffers
- * @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, gstrtcpbuffer
+ * @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, gstrtcpbuffer
*
* <refsect2>
* <para>
projects / platform / upstream / gstreamer.git / commitdiff