mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-06 17:39:47 +00:00
124 lines
4.2 KiB
C
124 lines
4.2 KiB
C
/* GStreamer
|
|
* Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
|
|
*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
|
|
#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/rtp/gstrtpbasepayload.h>
|
|
#include <gst/base/gstadapter.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
|
|
typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;
|
|
|
|
typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;
|
|
|
|
#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_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
|
|
#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass), \
|
|
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
|
|
{
|
|
GstRTPBasePayload payload;
|
|
|
|
GstRTPBaseAudioPayloadPrivate *priv;
|
|
|
|
GstClockTime base_ts;
|
|
gint frame_size;
|
|
gint frame_duration;
|
|
|
|
gint sample_size;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
/**
|
|
* GstRTPBaseAudioPayloadClass:
|
|
* @parent_class: the parent class
|
|
*
|
|
* Base class for audio RTP payloader.
|
|
*/
|
|
struct _GstRTPBaseAudioPayloadClass
|
|
{
|
|
GstRTPBasePayloadClass parent_class;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GST_RTP_API
|
|
GType gst_rtp_base_audio_payload_get_type (void);
|
|
|
|
/* configure frame based */
|
|
|
|
GST_RTP_API
|
|
void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
|
|
|
|
GST_RTP_API
|
|
void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
|
|
gint frame_duration, gint frame_size);
|
|
|
|
/* configure sample based */
|
|
|
|
GST_RTP_API
|
|
void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
|
|
|
|
GST_RTP_API
|
|
void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
|
|
gint sample_size);
|
|
|
|
GST_RTP_API
|
|
void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
|
|
gint sample_size);
|
|
|
|
/* get the internal adapter */
|
|
|
|
GST_RTP_API
|
|
GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
|
|
|
|
/* push and flushing data */
|
|
|
|
GST_RTP_API
|
|
GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
|
|
const guint8 * data, guint payload_len,
|
|
GstClockTime timestamp);
|
|
|
|
GST_RTP_API
|
|
GstFlowReturn gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
|
|
guint payload_len, GstClockTime timestamp);
|
|
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseAudioPayload, gst_object_unref)
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
|