configure.ac: Disable rtpmanager for now because it depends on CVS -base.

Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.

1 files changed, 206 insertions, 0 deletions

diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h
new file mode 100644
index 00000000..46062c99
--- /dev/null
+++ b/gst/rtpmanager/rtpsession.h
@@ -0,0 +1,206 @@
+/* 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);
+
+/**
+ * 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
+ *
+ * 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;
+} 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;
+
+  RTPSource    *source;
+  GHashTable   *ssrcs;
+  GHashTable   *cnames;
+  guint         total_sources;
+
+  RTPSessionCallbacks callbacks;
+  gpointer            user_data;
+
+  RTPSessionStats stats;
+};
+
+/**
+ * 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);
+};
+
+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);
+
+/* handling sources */
+gboolean        rtp_session_add_source             (RTPSession *sess, RTPSource *src);
+gint            rtp_session_get_num_sources        (RTPSession *sess);
+gint            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);
+
+/* get interval for next RTCP interval */
+gdouble         rtp_session_get_rtcp_interval      (RTPSession *sess);
+GstFlowReturn   rtp_session_produce_rtcp           (RTPSession *sess);
+
+#endif /* __RTP_SESSION_H__ */