Commit 5f1312b5 authored by Wim Taymans's avatar Wim Taymans
Browse files

rename files to match object names

parent ccf511a5
......@@ -1123,8 +1123,8 @@ gst_riff_strh
# rtp
<SECTION>
<FILE>gstbasertpaudiopayload</FILE>
<INCLUDE>gst/rtp/gstbasertpaudiopayload.h</INCLUDE>
<FILE>gstrtpbaseaudiopayload</FILE>
<INCLUDE>gst/rtp/gstrtpbaseaudiopayload.h</INCLUDE>
GstRTPBaseAudioPayload
GstRTPBaseAudioPayloadClass
......@@ -1148,8 +1148,8 @@ GstRTPBaseAudioPayloadPrivate
</SECTION>
<SECTION>
<FILE>gstbasertpdepayload</FILE>
<INCLUDE>gst/rtp/gstbasertpdepayload.h</INCLUDE>
<FILE>gstrtpbasedepayload</FILE>
<INCLUDE>gst/rtp/gstrtpbasedepayload.h</INCLUDE>
GstRTPBaseDepayload
GstRTPBaseDepayloadClass
......@@ -1179,8 +1179,8 @@ QUEUE_UNLOCK
</SECTION>
<SECTION>
<FILE>gstbasertppayload</FILE>
<INCLUDE>gst/rtp/gstbasertppayload.h</INCLUDE>
<FILE>gstrtpbasepayload</FILE>
<INCLUDE>gst/rtp/gstrtpbasepayload.h</INCLUDE>
GstRTPBasePayload
GstRTPBasePayloadClass
......
......@@ -48,11 +48,11 @@ gst_video_orientation_get_type
gst_video_overlay_get_type
#include <gst/rtp/gstbasertpdepayload.h>
#include <gst/rtp/gstrtpbasedepayload.h>
gst_rtp_base_depayload_get_type
#include <gst/rtp/gstbasertppayload.h>
#include <gst/rtp/gstrtpbasepayload.h>
gst_rtp_base_payload_get_type
#include <gst/rtp/gstbasertpaudiopayload.h>
#include <gst/rtp/gstrtpbaseaudiopayload.h>
gst_rtp_base_audio_payload_get_type
......
......@@ -3,18 +3,18 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
libgstrtpinclude_HEADERS = gstrtpbuffer.h \
gstrtcpbuffer.h \
gstrtppayloads.h \
gstbasertpaudiopayload.h \
gstbasertppayload.h \
gstbasertpdepayload.h
gstrtpbaseaudiopayload.h \
gstrtpbasepayload.h \
gstrtpbasedepayload.h
lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
gstrtcpbuffer.c \
gstrtppayloads.c \
gstbasertpaudiopayload.c \
gstbasertppayload.c \
gstbasertpdepayload.c
gstrtpbaseaudiopayload.c \
gstrtpbasepayload.c \
gstrtpbasedepayload.c
libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstrtp_@GST_MAJORMINOR@_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
......
......@@ -18,7 +18,7 @@
*/
/**
* SECTION:gstbasertpaudiopayload
* SECTION:gstrtpbaseaudiopayload
* @short_description: Base class for audio RTP payloader
*
* Provides a base class for audio RTP payloaders for frame or sample based
......@@ -63,10 +63,10 @@
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/base/gstadapter.h>
#include "gstbasertpaudiopayload.h"
#include "gstrtpbaseaudiopayload.h"
GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
GST_DEBUG_CATEGORY_STATIC (rtpbaseaudiopayload_debug);
#define GST_CAT_DEFAULT (rtpbaseaudiopayload_debug)
#define DEFAULT_BUFFER_LIST FALSE
......@@ -166,13 +166,13 @@ gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstbasertppayload_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate));
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_base_audio_payload_finalize;
gobject_class->set_property = gst_rtp_base_audio_payload_set_property;
......@@ -186,12 +186,12 @@ gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state);
gstbasertppayload_class->handle_buffer =
gstrtpbasepayload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer);
gstbasertppayload_class->handle_event =
gstrtpbasepayload_class->handle_event =
GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_handle_event);
GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
GST_DEBUG_CATEGORY_INIT (rtpbaseaudiopayload_debug, "rtpbaseaudiopayload", 0,
"base audio RTP payloader");
}
......@@ -262,55 +262,55 @@ gst_rtp_base_audio_payload_get_property (GObject * object,
/**
* gst_rtp_base_audio_payload_set_frame_based:
* @basertpaudiopayload: a pointer to the element.
* @rtpbaseaudiopayload: a pointer to the element.
*
* Tells #GstRTPBaseAudioPayload that the child element is for a frame based
* audio codec
*/
void
gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *
basertpaudiopayload)
rtpbaseaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
g_return_if_fail (rtpbaseaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
basertpaudiopayload->priv->bytes_to_time =
rtpbaseaudiopayload->priv->bytes_to_time =
gst_rtp_base_audio_payload_frame_bytes_to_time;
basertpaudiopayload->priv->bytes_to_rtptime =
rtpbaseaudiopayload->priv->bytes_to_rtptime =
gst_rtp_base_audio_payload_frame_bytes_to_rtptime;
basertpaudiopayload->priv->time_to_bytes =
rtpbaseaudiopayload->priv->time_to_bytes =
gst_rtp_base_audio_payload_frame_time_to_bytes;
}
/**
* gst_rtp_base_audio_payload_set_sample_based:
* @basertpaudiopayload: a pointer to the element.
* @rtpbaseaudiopayload: a pointer to the element.
*
* Tells #GstRTPBaseAudioPayload that the child element is for a sample based
* audio codec
*/
void
gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *
basertpaudiopayload)
rtpbaseaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL);
g_return_if_fail (rtpbaseaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
basertpaudiopayload->priv->bytes_to_time =
rtpbaseaudiopayload->priv->bytes_to_time =
gst_rtp_base_audio_payload_sample_bytes_to_time;
basertpaudiopayload->priv->bytes_to_rtptime =
rtpbaseaudiopayload->priv->bytes_to_rtptime =
gst_rtp_base_audio_payload_sample_bytes_to_rtptime;
basertpaudiopayload->priv->time_to_bytes =
rtpbaseaudiopayload->priv->time_to_bytes =
gst_rtp_base_audio_payload_sample_time_to_bytes;
}
/**
* gst_rtp_base_audio_payload_set_frame_options:
* @basertpaudiopayload: a pointer to the element.
* @rtpbaseaudiopayload: 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.
*
......@@ -319,46 +319,46 @@ gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *
*/
void
gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint frame_duration, gint frame_size)
* rtpbaseaudiopayload, gint frame_duration, gint frame_size)
{
GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload != NULL);
priv = basertpaudiopayload->priv;
priv = rtpbaseaudiopayload->priv;
basertpaudiopayload->frame_duration = frame_duration;
rtpbaseaudiopayload->frame_duration = frame_duration;
priv->frame_duration_ns = frame_duration * GST_MSECOND;
basertpaudiopayload->frame_size = frame_size;
rtpbaseaudiopayload->frame_size = frame_size;
priv->align = frame_size;
gst_adapter_clear (priv->adapter);
GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d",
GST_DEBUG_OBJECT (rtpbaseaudiopayload, "frame set to %d ms and size %d",
frame_duration, frame_size);
}
/**
* gst_rtp_base_audio_payload_set_sample_options:
* @basertpaudiopayload: a pointer to the element.
* @rtpbaseaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bytes.
*
* Sets the options for sample based audio codecs.
*/
void
gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint sample_size)
* rtpbaseaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload != NULL);
/* sample_size is in bits internally */
gst_rtp_base_audio_payload_set_samplebits_options (basertpaudiopayload,
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
sample_size * 8);
}
/**
* gst_rtp_base_audio_payload_set_samplebits_options:
* @basertpaudiopayload: a pointer to the element.
* @rtpbaseaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bits.
*
* Sets the options for sample based audio codecs.
......@@ -367,16 +367,16 @@ gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload
*/
void
gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload
* basertpaudiopayload, gint sample_size)
* rtpbaseaudiopayload, gint sample_size)
{
guint fragment_size;
GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload != NULL);
priv = basertpaudiopayload->priv;
priv = rtpbaseaudiopayload->priv;
basertpaudiopayload->sample_size = sample_size;
rtpbaseaudiopayload->sample_size = sample_size;
/* sample_size is in bits and is converted into multiple bytes */
fragment_size = sample_size;
......@@ -387,7 +387,7 @@ gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload
gst_adapter_clear (priv->adapter);
GST_DEBUG_OBJECT (basertpaudiopayload,
GST_DEBUG_OBJECT (rtpbaseaudiopayload,
"Samplebits set to sample size %d bits", sample_size);
}
......@@ -922,16 +922,16 @@ static GstStateChangeReturn
gst_rtp_base_payload_audio_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBaseAudioPayload *basertppayload;
GstRTPBaseAudioPayload *rtpbasepayload;
GstStateChangeReturn ret;
basertppayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
basertppayload->priv->cached_mtu = -1;
basertppayload->priv->last_rtptime = -1;
basertppayload->priv->last_timestamp = -1;
rtpbasepayload->priv->cached_mtu = -1;
rtpbasepayload->priv->last_rtptime = -1;
rtpbasepayload->priv->last_timestamp = -1;
break;
default:
break;
......@@ -941,7 +941,7 @@ gst_rtp_base_payload_audio_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_adapter_clear (basertppayload->priv->adapter);
gst_adapter_clear (rtpbasepayload->priv->adapter);
break;
default:
break;
......@@ -979,7 +979,7 @@ gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep,
/**
* gst_rtp_base_audio_payload_get_adapter:
* @basertpaudiopayload: a #GstRTPBaseAudioPayload
* @rtpbaseaudiopayload: a #GstRTPBaseAudioPayload
*
* Gets the internal adapter used by the depayloader.
*
......@@ -989,11 +989,11 @@ gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep,
*/
GstAdapter *
gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload
* basertpaudiopayload)
* rtpbaseaudiopayload)
{
GstAdapter *adapter;
if ((adapter = basertpaudiopayload->priv->adapter))
if ((adapter = rtpbaseaudiopayload->priv->adapter))
g_object_ref (adapter);
return adapter;
......
......@@ -21,7 +21,7 @@
#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
......@@ -78,20 +78,20 @@ struct _GstRTPBaseAudioPayloadClass
GType gst_rtp_base_audio_payload_get_type (void);
/* configure frame based */
void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *basertpaudiopayload);
void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *basertpaudiopayload,
void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
gint frame_duration, gint frame_size);
/* configure sample based */
void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *basertpaudiopayload);
void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *basertpaudiopayload,
void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
gint sample_size);
void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *basertpaudiopayload,
void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
gint sample_size);
/* get the internal adapter */
GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *basertpaudiopayload);
GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
/* push and flushing data */
GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
......
......@@ -19,16 +19,16 @@
*/
/**
* SECTION:gstbasertpdepayload
* SECTION:gstrtpbasedepayload
* @short_description: Base class for RTP depayloader
*
* Provides a base class for RTP depayloaders
*/
#include "gstbasertpdepayload.h"
#include "gstrtpbasedepayload.h"
GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
#define GST_CAT_DEFAULT (basertpdepayload_debug)
GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
#define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate))
......@@ -85,7 +85,7 @@ static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
static GstElementClass *parent_class = NULL;
static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
klass);
static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * basertppayload,
static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
GstRTPBaseDepayloadClass * klass);
GType
......@@ -134,7 +134,7 @@ gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
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,
GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0,
"Base class for RTP Depayloaders");
}
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment