mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 01:02:19 +00:00
478 lines
12 KiB
C
478 lines
12 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.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;
|
||
|
}
|