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363b790d38
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
127 lines
4.3 KiB
C
127 lines
4.3 KiB
C
/* GStreamer
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* Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
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#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstrtpbasepayload.h>
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#include <gst/base/gstadapter.h>
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G_BEGIN_DECLS
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typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
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typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;
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typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;
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#define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \
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(gst_rtp_base_audio_payload_get_type())
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#define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj), \
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GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
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#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass), \
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GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass))
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#define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
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#define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
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#define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \
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((GstRTPBaseAudioPayload *) (obj))
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struct _GstRTPBaseAudioPayload
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{
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GstRTPBasePayload payload;
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GstRTPBaseAudioPayloadPrivate *priv;
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GstClockTime base_ts;
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gint frame_size;
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gint frame_duration;
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gint sample_size;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstRTPBaseAudioPayloadClass:
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* @parent_class: the parent class
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*
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* Base class for audio RTP payloader.
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*/
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struct _GstRTPBaseAudioPayloadClass
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{
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GstRTPBasePayloadClass parent_class;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_RTP_API
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GType gst_rtp_base_audio_payload_get_type (void);
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/* configure frame based */
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GST_RTP_API
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void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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GST_RTP_API
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void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint frame_duration, gint frame_size);
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/* configure sample based */
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GST_RTP_API
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void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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GST_RTP_API
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void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint sample_size);
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GST_RTP_API
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void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint sample_size);
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/* get the internal adapter */
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GST_RTP_API
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GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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/* push and flushing data */
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GST_RTP_API
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GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
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const guint8 * data, guint payload_len,
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GstClockTime timestamp);
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GST_RTP_API
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GstFlowReturn gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
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guint payload_len, GstClockTime timestamp);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseAudioPayload, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
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