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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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459f5c944e
Added a queue based system for the rtpdtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence. 20070822175533-4f0f6-f27414c406f1f7b00c9a9084a988cf3a7930fe5c.gz
119 lines
3.7 KiB
C
119 lines
3.7 KiB
C
/* GStreamer RTP DTMF source
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*
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* gstrtpdtmfsrc.h:
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*
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* Copyright (C) <2007> Nokia Corporation.
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* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
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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_RTP_DTMF_SRC_H__
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#define __GST_RTP_DTMF_SRC_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstrtpbuffer.h>
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G_BEGIN_DECLS
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#define GST_TYPE_RTP_DTMF_SRC (gst_rtp_dtmf_src_get_type())
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#define GST_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrc))
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#define GST_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrcClass))
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#define GST_RTP_DTMF_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_DTMF_SRC, GstRTPDTMFSrcClass))
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#define GST_IS_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DTMF_SRC))
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#define GST_IS_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DTMF_SRC))
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#define GST_RTP_DTMF_SRC_CAST(obj) ((GstRTPDTMFSrc *)(obj))
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typedef struct {
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unsigned event:8; /* Current DTMF event */
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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unsigned volume:6; /* power level of the tone, in dBm0 */
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unsigned r:1; /* Reserved-bit */
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unsigned e:1; /* End-bit */
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#elif G_BYTE_ORDER == G_BIG_ENDIAN
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unsigned e:1; /* End-bit */
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unsigned r:1; /* Reserved-bit */
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unsigned volume:6; /* power level of the tone, in dBm0 */
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#else
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#error "G_BYTE_ORDER should be big or little endian."
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#endif
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unsigned duration:16; /* Duration of digit, in timestamp units */
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} GstRTPDTMFPayload;
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typedef struct _GstRTPDTMFSrc GstRTPDTMFSrc;
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typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass;
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static enum _GstRTPDTMFEventType {
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RTP_DTMF_EVENT_TYPE_START,
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RTP_DTMF_EVENT_TYPE_STOP
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};
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typedef enum _GstRTPDTMFEventType GstRTPDTMFEventType;
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struct _GstRTPDTMFSrcEvent {
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GstRTPDTMFEventType event_type;
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GstRTPDTMFPayload* payload;
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guint32 sent_packets;
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};
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typedef struct _GstRTPDTMFSrcEvent GstRTPDTMFSrcEvent;
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/**
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* GstRTPDTMFSrc:
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* @element: the parent element.
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*
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* The opaque #GstRTPDTMFSrc data structure.
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*/
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struct _GstRTPDTMFSrc {
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GstElement element;
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GstPad* srcpad;
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GstSegment segment;
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GAsyncQueue* event_queue;
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GstRTPDTMFSrcEvent* last_event;
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GstClockTime timestamp;
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gboolean first_packet;
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gboolean last_packet;
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guint32 ts_base;
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guint16 seqnum_base;
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gint16 seqnum_offset;
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guint16 seqnum;
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gint32 ts_offset;
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guint32 rtp_timestamp;
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guint pt;
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guint ssrc;
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guint current_ssrc;
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guint16 interval;
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guint16 packet_redundancy;
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guint32 clock_rate;
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};
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struct _GstRTPDTMFSrcClass {
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GstElementClass parent_class;
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};
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GType gst_rtp_dtmf_src_get_type (void);
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gboolean gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin);
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G_END_DECLS
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#endif /* __GST_RTP_DTMF_SRC_H__ */
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