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126 lines
4.3 KiB
C
126 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_EXPORT
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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_EXPORT
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void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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GST_EXPORT
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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_EXPORT
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void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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GST_EXPORT
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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_EXPORT
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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_EXPORT
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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_EXPORT
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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_EXPORT
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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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