| /* 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_SESSION_H__ |
| #define __RTP_SESSION_H__ |
| |
| #include <gst/gst.h> |
| #include <gst/netbuffer/gstnetbuffer.h> |
| |
| #include "rtpsource.h" |
| |
| typedef struct _RTPSession RTPSession; |
| typedef struct _RTPSessionClass RTPSessionClass; |
| |
| #define RTP_TYPE_SESSION (rtp_session_get_type()) |
| #define RTP_SESSION(sess) (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession)) |
| #define RTP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass)) |
| #define RTP_IS_SESSION(sess) (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION)) |
| #define RTP_IS_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION)) |
| #define RTP_SESSION_CAST(sess) ((RTPSession *)(sess)) |
| |
| #define RTP_SESSION_LOCK(sess) (g_mutex_lock ((sess)->lock)) |
| #define RTP_SESSION_UNLOCK(sess) (g_mutex_unlock ((sess)->lock)) |
| |
| /** |
| * RTPSessionProcessRTP: |
| * @sess: an #RTPSession |
| * @src: the #RTPSource |
| * @buffer: the RTP buffer ready for processing |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess has @buffer ready for further |
| * processing. Processing the buffer typically includes decoding and displaying |
| * the buffer. |
| * |
| * Returns: a #GstFlowReturn. |
| */ |
| typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data); |
| |
| /** |
| * RTPSessionSendRTP: |
| * @sess: an #RTPSession |
| * @src: the #RTPSource |
| * @buffer: the RTP buffer ready for sending |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess has @buffer ready for sending to |
| * all listening participants in this session. |
| * |
| * Returns: a #GstFlowReturn. |
| */ |
| typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data); |
| |
| /** |
| * RTPSessionSendRTCP: |
| * @sess: an #RTPSession |
| * @src: the #RTPSource |
| * @buffer: the RTCP buffer ready for sending |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess has @buffer ready for sending to |
| * all listening participants in this session. |
| * |
| * Returns: a #GstFlowReturn. |
| */ |
| typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data); |
| |
| /** |
| * RTPSessionClockRate: |
| * @sess: an #RTPSession |
| * @payload: the payload |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess needs the clock-rate of @payload. |
| * |
| * Returns: the clock-rate of @pt. |
| */ |
| typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data); |
| |
| /** |
| * RTPSessionGetTime: |
| * @sess: an #RTPSession |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess needs the current time in |
| * nanoseconds. |
| * |
| * Returns: a #GstClockTime with the current time in nanoseconds. |
| */ |
| typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data); |
| |
| /** |
| * RTPSessionReconsider: |
| * @sess: an #RTPSession |
| * @user_data: user data specified when registering |
| * |
| * This callback will be called when @sess needs to cancel the current timeout. |
| * The currently running timeout should be canceled and a new reporting interval |
| * should be requested from @sess. |
| */ |
| typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data); |
| |
| /** |
| * RTPSessionCallbacks: |
| * @RTPSessionProcessRTP: callback to process RTP packets |
| * @RTPSessionSendRTP: callback for sending RTP packets |
| * @RTPSessionSendRTCP: callback for sending RTCP packets |
| * @RTPSessionGetTime: callback for returning the current time |
| * @RTPSessionReconsider: callback for reconsidering the timeout |
| * |
| * These callbacks can be installed on the session manager to get notification |
| * when RTP and RTCP packets are ready for further processing. These callbacks |
| * are not implemented with signals for performance reasons. |
| */ |
| typedef struct { |
| RTPSessionProcessRTP process_rtp; |
| RTPSessionSendRTP send_rtp; |
| RTPSessionSendRTCP send_rtcp; |
| RTPSessionClockRate clock_rate; |
| RTPSessionGetTime get_time; |
| RTPSessionReconsider reconsider; |
| } RTPSessionCallbacks; |
| |
| /** |
| * RTPSession: |
| * @lock: lock to protect the session |
| * @source: the source of this session |
| * @ssrcs: Hashtable of sources indexed by SSRC |
| * @cnames: Hashtable of sources indexed by CNAME |
| * @num_sources: the number of sources |
| * @activecount: the number of active sources |
| * @callbacks: callbacks |
| * @user_data: user data passed in callbacks |
| * |
| * The RTP session manager object |
| */ |
| struct _RTPSession { |
| GObject object; |
| |
| GMutex *lock; |
| |
| guint header_len; |
| guint mtu; |
| |
| RTPSource *source; |
| |
| /* info for creating reports */ |
| gchar *cname; |
| gchar *name; |
| gchar *email; |
| gchar *phone; |
| gchar *location; |
| gchar *tool; |
| gchar *note; |
| |
| /* for sender/receiver counting */ |
| guint32 key; |
| guint32 mask_idx; |
| guint32 mask; |
| GHashTable *ssrcs[32]; |
| GHashTable *cnames; |
| guint total_sources; |
| |
| GstClockTime next_rtcp_check_time; |
| GstClockTime last_rtcp_send_time; |
| gboolean first_rtcp; |
| |
| GstBuffer *bye_packet; |
| gchar *bye_reason; |
| gboolean sent_bye; |
| |
| RTPSessionCallbacks callbacks; |
| gpointer user_data; |
| |
| RTPSessionStats stats; |
| |
| /* for mapping RTP time to NTP time */ |
| GstClockTime start_timestamp; |
| GstClockTime base_time; |
| }; |
| |
| /** |
| * RTPSessionClass: |
| * @on_new_ssrc: emited when a new source is found |
| * @on_bye_ssrc: emited when a source is gone |
| * |
| * The session class. |
| */ |
| struct _RTPSessionClass { |
| GObjectClass parent_class; |
| |
| /* signals */ |
| void (*on_new_ssrc) (RTPSession *sess, RTPSource *source); |
| void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source); |
| void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source); |
| void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source); |
| void (*on_bye_timeout) (RTPSession *sess, RTPSource *source); |
| void (*on_timeout) (RTPSession *sess, RTPSource *source); |
| }; |
| |
| GType rtp_session_get_type (void); |
| |
| /* create and configure */ |
| RTPSession* rtp_session_new (void); |
| void rtp_session_set_callbacks (RTPSession *sess, |
| RTPSessionCallbacks *callbacks, |
| gpointer user_data); |
| void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth); |
| gdouble rtp_session_get_bandwidth (RTPSession *sess); |
| void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction); |
| gdouble rtp_session_get_rtcp_fraction (RTPSession *sess); |
| |
| void rtp_session_set_cname (RTPSession *sess, const gchar *cname); |
| gchar* rtp_session_get_cname (RTPSession *sess); |
| void rtp_session_set_name (RTPSession *sess, const gchar *name); |
| gchar* rtp_session_get_name (RTPSession *sess); |
| void rtp_session_set_email (RTPSession *sess, const gchar *email); |
| gchar* rtp_session_get_email (RTPSession *sess); |
| void rtp_session_set_phone (RTPSession *sess, const gchar *phone); |
| gchar* rtp_session_get_phone (RTPSession *sess); |
| void rtp_session_set_location (RTPSession *sess, const gchar *location); |
| gchar* rtp_session_get_location (RTPSession *sess); |
| void rtp_session_set_tool (RTPSession *sess, const gchar *tool); |
| gchar* rtp_session_get_tool (RTPSession *sess); |
| void rtp_session_set_note (RTPSession *sess, const gchar *note); |
| gchar* rtp_session_get_note (RTPSession *sess); |
| |
| /* handling sources */ |
| gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src); |
| guint rtp_session_get_num_sources (RTPSession *sess); |
| guint rtp_session_get_num_active_sources (RTPSession *sess); |
| RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc); |
| RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname); |
| RTPSource* rtp_session_create_source (RTPSession *sess); |
| |
| /* processing packets from receivers */ |
| GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer); |
| GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer); |
| |
| /* processing packets for sending */ |
| GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer); |
| void rtp_session_set_base_time (RTPSession *sess, GstClockTime base_time); |
| void rtp_session_set_timestamp_sync (RTPSession *sess, GstClockTime start_timestamp); |
| /* stopping the session */ |
| GstFlowReturn rtp_session_send_bye (RTPSession *sess, const gchar *reason); |
| |
| /* get interval for next RTCP interval */ |
| GstClockTime rtp_session_next_timeout (RTPSession *sess, GstClockTime time); |
| GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTime time); |
| |
| #endif /* __RTP_SESSION_H__ */ |