| /* GStreamer |
| * Copyright (C) <2007> Wim Taymans <wim@fluendo.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 59 Temple Place - Suite 330, |
| * Boston, MA 02111-1307, USA. |
| */ |
| |
| #ifndef __RTP_SOURCE_H__ |
| #define __RTP_SOURCE_H__ |
| |
| #include <gst/gst.h> |
| #include <gst/rtp/gstrtcpbuffer.h> |
| #include <gst/netbuffer/gstnetbuffer.h> |
| |
| #include "rtpstats.h" |
| |
| /* the default number of consecutive RTP packets we need to receive before the |
| * source is considered valid */ |
| #define RTP_NO_PROBATION 0 |
| #define RTP_DEFAULT_PROBATION 2 |
| |
| #define RTP_SEQ_MOD (1 << 16) |
| #define RTP_MAX_DROPOUT 3000 |
| #define RTP_MAX_MISORDER 100 |
| |
| typedef struct _RTPSource RTPSource; |
| typedef struct _RTPSourceClass RTPSourceClass; |
| |
| #define RTP_TYPE_SOURCE (rtp_source_get_type()) |
| #define RTP_SOURCE(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource)) |
| #define RTP_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass)) |
| #define RTP_IS_SOURCE(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE)) |
| #define RTP_IS_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE)) |
| #define RTP_SOURCE_CAST(src) ((RTPSource *)(src)) |
| |
| /** |
| * RTP_SOURCE_IS_ACTIVE: |
| * @src: an #RTPSource |
| * |
| * Check if @src is active. A source is active when it has been validated |
| * and has not yet received a BYE packet. |
| */ |
| #define RTP_SOURCE_IS_ACTIVE(src) (src->validated && !src->received_bye) |
| |
| /** |
| * RTP_SOURCE_IS_SENDER: |
| * @src: an #RTPSource |
| * |
| * Check if @src is a sender. |
| */ |
| #define RTP_SOURCE_IS_SENDER(src) (src->is_sender) |
| |
| /** |
| * RTPSourcePushRTP: |
| * @src: an #RTPSource |
| * @buffer: the RTP buffer ready for processing |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @src has @buffer ready for further |
| * processing. |
| * |
| * Returns: a #GstFlowReturn. |
| */ |
| typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, |
| gpointer user_data); |
| |
| /** |
| * RTPSourceClockRate: |
| * @src: an #RTPSource |
| * @payload: a payload type |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @src needs the clock-rate of the |
| * @payload. |
| * |
| * Returns: a clock-rate for @payload. |
| */ |
| typedef gint (*RTPSourceClockRate) (RTPSource *src, guint8 payload, gpointer user_data); |
| |
| /** |
| * RTPSourceCallbacks: |
| * @push_rtp: a packet becomes available for handling |
| * @clock_rate: a clock-rate is requested |
| * @get_time: the current clock time is requested |
| * |
| * Callbacks performed by #RTPSource when actions need to be performed. |
| */ |
| typedef struct { |
| RTPSourcePushRTP push_rtp; |
| RTPSourceClockRate clock_rate; |
| } RTPSourceCallbacks; |
| |
| /** |
| * RTPSource: |
| * |
| * A source in the #RTPSession |
| */ |
| struct _RTPSource { |
| GObject object; |
| |
| /*< private >*/ |
| guint32 ssrc; |
| |
| gint probation; |
| gboolean validated; |
| gboolean is_csrc; |
| gboolean is_sender; |
| |
| gchar *cname; |
| gchar *name; |
| gchar *email; |
| gchar *phone; |
| gchar *location; |
| gchar *tool; |
| gchar *note; |
| gboolean received_bye; |
| gchar *bye_reason; |
| |
| gboolean have_rtp_from; |
| GstNetAddress rtp_from; |
| gboolean have_rtcp_from; |
| GstNetAddress rtcp_from; |
| |
| guint8 payload; |
| gint clock_rate; |
| |
| GstClockTime bye_time; |
| GstClockTime last_activity; |
| GstClockTime last_rtp_activity; |
| |
| GQueue *packets; |
| |
| RTPSourceCallbacks callbacks; |
| gpointer user_data; |
| |
| RTPSourceStats stats; |
| }; |
| |
| struct _RTPSourceClass { |
| GObjectClass parent_class; |
| }; |
| |
| GType rtp_source_get_type (void); |
| |
| /* managing lifetime of sources */ |
| RTPSource* rtp_source_new (guint32 ssrc); |
| |
| void rtp_source_set_callbacks (RTPSource *src, RTPSourceCallbacks *cb, gpointer data); |
| void rtp_source_set_as_csrc (RTPSource *src); |
| |
| void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address); |
| void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address); |
| |
| /* handling RTP */ |
| GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival); |
| |
| GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer); |
| |
| /* RTCP messages */ |
| void rtp_source_process_bye (RTPSource *src, const gchar *reason); |
| void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime, |
| guint32 packet_count, guint32 octet_count, GstClockTime time); |
| void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost, |
| guint32 exthighestseq, guint32 jitter, |
| guint32 lsr, guint32 dlsr); |
| |
| gboolean rtp_source_get_last_sr (RTPSource *src, guint64 *ntptime, guint32 *rtptime, |
| guint32 *packet_count, guint32 *octet_count, GstClockTime *time); |
| gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost, |
| guint32 *exthighestseq, guint32 *jitter, |
| guint32 *lsr, guint32 *dlsr); |
| |
| #endif /* __RTP_SOURCE_H__ */ |