mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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fb5037f727
Refactor getting the packet min/max size and alignment code. Refactor converting bytes to time. change some variable to something shorter.
98 lines
3.9 KiB
C
98 lines
3.9 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., 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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typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate;
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#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
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(gst_base_rtp_audio_payload_get_type())
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#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj), \
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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), \
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayloadClass))
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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(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
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#define GST_BASE_RTP_AUDIO_PAYLOAD_CAST(obj) \
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((GstBaseRTPAudioPayload *) (obj))
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struct _GstBaseRTPAudioPayload
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{
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GstBaseRTPPayload payload;
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GstBaseRTPAudioPayloadPrivate *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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gpointer _gst_reserved[GST_PADDING];
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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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gpointer _gst_reserved[GST_PADDING];
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};
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GType gst_base_rtp_audio_payload_get_type (void);
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/* configure frame based */
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void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload,
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gint frame_duration, gint frame_size);
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/* configure sample based */
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void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
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void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload,
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gint sample_size);
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void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload,
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gint sample_size);
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/* get the internal adapter */
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GstAdapter* gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload *basertpaudiopayload);
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/* push and flushing data */
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GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
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const guint8 * data, guint payload_len,
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GstClockTime timestamp);
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GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
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guint payload_len, GstClockTime timestamp);
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G_END_DECLS
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#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
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