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Original commit message from CVS: 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk> * gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children * gst-libs/gst/rtp/gstbasertpaudiopayload.c: * gst-libs/gst/rtp/gstbasertpaudiopayload.h: New RTP audio base payloader class. Supports frame or sample based codecs
88 lines
2.7 KiB
C
88 lines
2.7 KiB
C
/* GStreamer
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* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__
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#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertppayload.h>
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#include <gst/base/gstadapter.h>
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G_BEGIN_DECLS
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typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
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typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
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#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
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(gst_basertpaudiopayload_get_type())
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#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
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#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
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#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
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#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
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typedef enum {
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AUDIO_CODEC_TYPE_NONE,
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AUDIO_CODEC_TYPE_FRAME_BASED,
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AUDIO_CODEC_TYPE_SAMPLE_BASED
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} AudioCodecType;
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struct _GstBaseRTPAudioPayload
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{
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GstBaseRTPPayload payload;
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GstClockTime adapter_base_ts;
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GstAdapter *adapter;
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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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AudioCodecType type;
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};
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struct _GstBaseRTPAudioPayloadClass
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{
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GstBaseRTPPayloadClass parent_class;
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};
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gboolean gst_basertpaudiopayload_plugin_init (GstPlugin * plugin);
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GType gst_basertpaudiopayload_get_type (void);
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void
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gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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void
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gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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void
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gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
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*basertpaudiopayload, gint frame_duration, gint frame_size);
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void
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gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
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*basertpaudiopayload, gint sample_size);
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G_END_DECLS
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#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
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