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, 477 insertions, 0 deletions

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
new file mode 100644
index 00000000..36f54381
--- /dev/null
+++ b/gst/rtpmanager/rtpsource.c
@@ -0,0 +1,477 @@
+/* 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.
+ */
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include "rtpsource.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
+#define GST_CAT_DEFAULT rtp_source_debug
+
+#define RTP_MAX_PROBATION_LEN	32
+
+/* signals and args */
+enum
+{
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0
+};
+
+/* GObject vmethods */
+static void rtp_source_finalize (GObject * object);
+
+/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
+
+G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
+
+static void
+rtp_source_class_init (RTPSourceClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->finalize = rtp_source_finalize;
+
+  GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
+}
+
+static void
+rtp_source_init (RTPSource * src)
+{
+  /* sources are initialy on probation until we receive enough valid RTP
+   * packets or a valid RTCP packet */
+  src->validated = FALSE;
+  src->probation = RTP_DEFAULT_PROBATION;
+
+  src->payload = 0;
+  src->clock_rate = -1;
+  src->packets = g_queue_new ();
+
+  src->stats.jitter = 0;
+  src->stats.transit = -1;
+  src->stats.curr_sr = 0;
+  src->stats.curr_rr = 0;
+}
+
+static void
+rtp_source_finalize (GObject * object)
+{
+  RTPSource *src;
+  GstBuffer *buffer;
+
+  src = RTP_SOURCE_CAST (object);
+
+  while ((buffer = g_queue_pop_head (src->packets)))
+    gst_buffer_unref (buffer);
+  g_queue_free (src->packets);
+
+  G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
+}
+
+/**
+ * rtp_source_new:
+ * @ssrc: an SSRC
+ *
+ * Create a #RTPSource with @ssrc.
+ *
+ * Returns: a new #RTPSource. Use g_object_unref() after usage.
+ */
+RTPSource *
+rtp_source_new (guint32 ssrc)
+{
+  RTPSource *src;
+
+  src = g_object_new (RTP_TYPE_SOURCE, NULL);
+  src->ssrc = ssrc;
+
+  return src;
+}
+
+/**
+ * rtp_source_set_callbacks:
+ * @src: an #RTPSource
+ * @cb: callback functions
+ * @user_data: user data
+ *
+ * Set the callbacks for the source.
+ */
+void
+rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
+    gpointer user_data)
+{
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  src->callbacks.push_rtp = cb->push_rtp;
+  src->callbacks.clock_rate = cb->clock_rate;
+  src->user_data = user_data;
+}
+
+/**
+ * rtp_source_set_as_csrc:
+ * @src: an #RTPSource
+ *
+ * Configure @src as a CSRC, this will validate the RTpSource.
+ */
+void
+rtp_source_set_as_csrc (RTPSource * src)
+{
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  src->validated = TRUE;
+  src->is_csrc = TRUE;
+}
+
+/**
+ * rtp_source_set_rtp_from:
+ * @src: an #RTPSource
+ * @address: the RTP address to set
+ *
+ * Set that @src is receiving RTP packets from @address. This is used for
+ * collistion checking.
+ */
+void
+rtp_source_set_rtp_from (RTPSource * src, GstNetAddress * address)
+{
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  src->have_rtp_from = TRUE;
+  memcpy (&src->rtp_from, address, sizeof (GstNetAddress));
+}
+
+/**
+ * rtp_source_set_rtcp_from:
+ * @src: an #RTPSource
+ * @address: the RTCP address to set
+ *
+ * Set that @src is receiving RTCP packets from @address. This is used for
+ * collistion checking.
+ */
+void
+rtp_source_set_rtcp_from (RTPSource * src, GstNetAddress * address)
+{
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  src->have_rtcp_from = TRUE;
+  memcpy (&src->rtcp_from, address, sizeof (GstNetAddress));
+}
+
+static GstFlowReturn
+push_packet (RTPSource * src, GstBuffer * buffer)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  /* push queued packets first if any */
+  while (!g_queue_is_empty (src->packets)) {
+    GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
+
+    GST_DEBUG ("pushing queued packet");
+    if (src->callbacks.push_rtp)
+      src->callbacks.push_rtp (src, buffer, src->user_data);
+    else
+      gst_buffer_unref (buffer);
+  }
+  GST_DEBUG ("pushing new packet");
+  /* push packet */
+  if (src->callbacks.push_rtp)
+    ret = src->callbacks.push_rtp (src, buffer, src->user_data);
+  else
+    gst_buffer_unref (buffer);
+
+  return ret;
+}
+
+static gint
+get_clock_rate (RTPSource * src, guint8 payload)
+{
+  if (payload != src->payload) {
+    gint clock_rate = -1;
+
+    if (src->callbacks.clock_rate)
+      clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
+
+    GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
+
+    src->clock_rate = clock_rate;
+    src->payload = payload;
+  }
+  return src->clock_rate;
+}
+
+static void
+calculate_jitter (RTPSource * src, GstBuffer * buffer,
+    RTPArrivalStats * arrival)
+{
+  GstClockTime current;
+  guint32 rtparrival, transit, rtptime;
+  gint32 diff;
+  gint clock_rate;
+  guint8 pt;
+
+  /* get arrival time */
+  if ((current = arrival->time) == GST_CLOCK_TIME_NONE)
+    goto no_time;
+
+  pt = gst_rtp_buffer_get_payload_type (buffer);
+
+  /* get clockrate */
+  if ((clock_rate = get_clock_rate (src, pt)) == -1)
+    goto no_clock_rate;
+
+  rtptime = gst_rtp_buffer_get_timestamp (buffer);
+
+  /* convert arrival time to RTP timestamp units */
+  rtparrival = gst_util_uint64_scale_int (current, clock_rate, GST_SECOND);
+
+  /* transit time is difference with RTP timestamp */
+  transit = rtparrival - rtptime;
+  /* get diff with previous transit time */
+  if (src->stats.transit != -1)
+    diff = transit - src->stats.transit;
+  else
+    diff = 0;
+  src->stats.transit = transit;
+  if (diff < 0)
+    diff = -diff;
+  /* update jitter */
+  src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
+
+  src->stats.prev_rtptime = src->stats.last_rtptime;
+  src->stats.last_rtptime = rtparrival;
+
+  GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %u",
+      rtparrival, rtptime, clock_rate, diff, src->stats.jitter);
+
+  return;
+
+  /* ERRORS */
+no_time:
+  {
+    GST_WARNING ("cannot get current time");
+    return;
+  }
+no_clock_rate:
+  {
+    GST_WARNING ("cannot get clock-rate for pt %d", pt);
+    return;
+  }
+}
+
+/**
+ * rtp_source_process_rtp:
+ * @src: an #RTPSource
+ * @buffer: an RTP buffer
+ *
+ * Let @src handle the incomming RTP @buffer.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+GstFlowReturn
+rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
+    RTPArrivalStats * arrival)
+{
+  GstFlowReturn result = GST_FLOW_OK;
+
+  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
+  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+
+  /* if we are still on probation, check seqnum */
+  if (src->probation) {
+    guint16 seqnr, expected;
+
+    expected = src->stats.max_seqnr + 1;
+
+    /* when in probation, we require consecutive seqnums */
+    seqnr = gst_rtp_buffer_get_seq (buffer);
+    if (seqnr == expected) {
+      /* expected packet */
+      src->probation--;
+      src->stats.max_seqnr = seqnr;
+
+      GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
+    } else {
+      GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
+      src->probation = RTP_DEFAULT_PROBATION;
+      src->stats.max_seqnr = seqnr;
+    }
+  }
+  if (src->probation) {
+    GstBuffer *q;
+
+    GST_DEBUG ("probation %d: queue buffer", src->probation);
+    /* when still in probation, keep packets in a list. */
+    g_queue_push_tail (src->packets, buffer);
+    /* remove packets from queue if there are too many */
+    while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
+      q = g_queue_pop_head (src->packets);
+      gst_object_unref (q);
+    }
+  } else {
+    /* we are not in probation */
+    src->stats.octetsreceived += arrival->payload_len;
+    src->stats.bytesreceived += arrival->bytes;
+    src->stats.packetsreceived++;
+    src->is_sender = TRUE;
+
+    GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
+        src->stats.packetsreceived, src->stats.octetsreceived);
+
+    /* calculate jitter */
+    calculate_jitter (src, buffer, arrival);
+
+    /* we're ready to push the RTP packet now */
+    result = push_packet (src, buffer);
+  }
+  return result;
+}
+
+/**
+ * rtp_source_process_bye:
+ * @src: an #RTPSource
+ * @reason: the reason for leaving
+ *
+ * Notify @src that a BYE packet has been received. This will make the source
+ * inactive.
+ */
+void
+rtp_source_process_bye (RTPSource * src, const gchar * reason)
+{
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
+      GST_STR_NULL (reason));
+
+  /* copy the reason and mark as received_bye */
+  g_free (src->bye_reason);
+  src->bye_reason = g_strdup (reason);
+  src->received_bye = TRUE;
+}
+
+/**
+ * rtp_source_send_rtp:
+ * @src: an #RTPSource
+ * @buffer: an RTP buffer
+ *
+ * Send an RTP @buffer originating from @src. This will make @src a sender.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+GstFlowReturn
+rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
+{
+  GstFlowReturn result = GST_FLOW_OK;
+
+  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
+  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+
+  /* we are a sender now */
+  src->is_sender = TRUE;
+
+  /* push packet */
+  if (src->callbacks.push_rtp)
+    result = src->callbacks.push_rtp (src, buffer, src->user_data);
+  else
+    gst_buffer_unref (buffer);
+
+  return result;
+}
+
+/**
+ * rtp_source_process_sr:
+ * @src: an #RTPSource
+ * @ntptime: the NTP time
+ * @rtptime: the RTP time
+ * @packet_count: the packet count
+ * @octet_count: the octect count
+ *
+ * Update the sender report in @src.
+ */
+void
+rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
+    guint32 packet_count, guint32 octet_count)
+{
+  RTPSenderReport *curr;
+  gint curridx;
+
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  GST_DEBUG ("got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
+      ", RTP %u, PC %u, OC %u", src->ssrc, ntptime, rtptime, packet_count,
+      octet_count);
+
+  curridx = src->stats.curr_sr ^ 1;
+  curr = &src->stats.sr[curridx];
+
+  /* update current */
+  curr->is_valid = TRUE;
+  curr->ntptime = ntptime;
+  curr->rtptime = rtptime;
+  curr->packet_count = packet_count;
+  curr->octet_count = octet_count;
+
+  /* make current */
+  src->stats.curr_sr = curridx;
+}
+
+/**
+ * rtp_source_process_rb:
+ * @src: an #RTPSource
+ * @fractionlost: fraction lost since last SR/RR
+ * @packetslost: the cumululative number of packets lost
+ * @exthighestseq: the extended last sequence number received
+ * @jitter: the interarrival jitter
+ * @lsr: the last SR packet from this source
+ * @dlsr: the delay since last SR packet
+ *
+ * Update the report block in @src.
+ */
+void
+rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
+    guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr)
+{
+  RTPReceiverReport *curr;
+  gint curridx;
+
+  g_return_if_fail (RTP_IS_SOURCE (src));
+
+  GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
+      ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", src->ssrc, fractionlost,
+      packetslost, exthighestseq, jitter, lsr, dlsr);
+
+  curridx = src->stats.curr_rr ^ 1;
+  curr = &src->stats.rr[curridx];
+
+  /* update current */
+  curr->is_valid = TRUE;
+  curr->fractionlost = fractionlost;
+  curr->packetslost = packetslost;
+  curr->exthighestseq = exthighestseq;
+  curr->jitter = jitter;
+  curr->lsr = lsr;
+  curr->dlsr = dlsr;
+
+  /* make current */
+  src->stats.curr_rr = curridx;
+}