gstreamer/gst/rtpmanager/rtpsource.c
Wim Taymans 67c69ca0ea gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
2007-04-25 13:19:36 +00:00

635 lines
16 KiB
C

/* 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.cycles = -1;
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 ABS diff with previous transit time */
if (src->stats.transit != -1) {
if (transit > src->stats.transit)
diff = transit - src->stats.transit;
else
diff = src->stats.transit - transit;
} else
diff = 0;
src->stats.transit = transit;
/* update jitter, the value we store is scaled up so we can keep precision. */
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;
}
}
static void
init_seq (RTPSource * src, guint16 seq)
{
src->stats.base_seq = seq;
src->stats.max_seq = seq;
src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
src->stats.cycles = 0;
src->stats.packets_received = 0;
src->stats.octets_received = 0;
src->stats.bytes_received = 0;
src->stats.prev_received = 0;
src->stats.prev_expected = 0;
GST_DEBUG ("base_seq %d", seq);
}
/**
* 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;
guint16 seqnr, udelta;
RTPSourceStats *stats;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
stats = &src->stats;
seqnr = gst_rtp_buffer_get_seq (buffer);
if (stats->cycles == -1) {
GST_DEBUG ("received first buffer");
/* first time we heard of this source */
init_seq (src, seqnr);
src->stats.max_seq = seqnr - 1;
src->probation = RTP_DEFAULT_PROBATION;
}
udelta = seqnr - stats->max_seq;
/* if we are still on probation, check seqnum */
if (src->probation) {
guint16 expected;
expected = src->stats.max_seq + 1;
/* when in probation, we require consecutive seqnums */
if (seqnr == expected) {
/* expected packet */
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
src->probation--;
src->stats.max_seq = seqnr;
if (src->probation == 0) {
GST_DEBUG ("probation done!", src->probation);
init_seq (src, seqnr);
} else {
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);
}
goto done;
}
} else {
GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
src->probation = RTP_DEFAULT_PROBATION;
src->stats.max_seq = seqnr;
goto done;
}
} else if (udelta < RTP_MAX_DROPOUT) {
/* in order, with permissible gap */
if (seqnr < stats->max_seq) {
/* sequence number wrapped - count another 64K cycle. */
stats->cycles += RTP_SEQ_MOD;
}
stats->max_seq = seqnr;
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
/* the sequence number made a very large jump */
if (seqnr == stats->bad_seq) {
/* two sequential packets -- assume that the other side
* restarted without telling us so just re-sync
* (i.e., pretend this was the first packet). */
init_seq (src, seqnr);
} else {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
goto bad_sequence;
}
} else {
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
}
src->stats.octets_received += arrival->payload_len;
src->stats.bytes_received += arrival->bytes;
src->stats.packets_received++;
/* the source that sent the packet must be a sender */
src->is_sender = TRUE;
src->validated = TRUE;
GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
/* calculate jitter */
calculate_jitter (src, buffer, arrival);
/* we're ready to push the RTP packet now */
result = push_packet (src, buffer);
done:
return result;
/* ERRORS */
bad_sequence:
{
GST_WARNING ("unacceptable seqnum received");
return GST_FLOW_OK;
}
}
/**
* 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
* @time: time of packet arrival
*
* Update the sender report in @src.
*/
void
rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count, GstClockTime time)
{
RTPSenderReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %u, PC %u, OC %u",
src->ssrc, ntptime >> 32, ntptime & 0xffffffff, rtptime, packet_count,
octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
/* this is a sender now */
src->is_sender = TRUE;
/* update current */
curr->is_valid = TRUE;
curr->ntptime = ntptime;
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
curr->time = time;
/* 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 %08x, DLSR %08x", 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;
}
/**
* rtp_source_get_last_sr:
* @src: an #RTPSource
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
* @time: time of packet arrival
*
* Get the values of the last sender report as set with rtp_source_process_sr().
*
* Returns: %TRUE if there was a valid SR report.
*/
gboolean
rtp_source_get_last_sr (RTPSource * src, guint64 * ntptime, guint32 * rtptime,
guint32 * packet_count, guint32 * octet_count, GstClockTime * time)
{
RTPSenderReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.sr[src->stats.curr_sr];
if (!curr->is_valid)
return FALSE;
if (ntptime)
*ntptime = curr->ntptime;
if (rtptime)
*rtptime = curr->rtptime;
if (packet_count)
*packet_count = curr->packet_count;
if (octet_count)
*octet_count = curr->octet_count;
if (time)
*time = curr->time;
return TRUE;
}
/**
* rtp_source_get_last_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
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
* Returns: %TRUE if there was a valid SB report.
*/
gboolean
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
guint32 * lsr, guint32 * dlsr)
{
RTPReceiverReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.rr[src->stats.curr_rr];
if (!curr->is_valid)
return FALSE;
if (fractionlost)
*fractionlost = curr->fractionlost;
if (packetslost)
*packetslost = curr->packetslost;
if (exthighestseq)
*exthighestseq = curr->exthighestseq;
if (jitter)
*jitter = curr->jitter;
if (lsr)
*lsr = curr->lsr;
if (dlsr)
*dlsr = curr->dlsr;
return TRUE;
}