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gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.

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Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
This commit is contained in:
Wim Taymans 2007-04-30 13:41:30 +00:00
parent f4508d302c
commit 5171199836
8 changed files with 151 additions and 43 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

35
ChangeLog
View file

@ -1,3 +1,38 @@
2007-04-30 Wim Taymans <wim@fluendo.com>
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
2007-04-29 Wim Taymans <wim@fluendo.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):

29
gst/rtpmanager/async_jitter_queue.c
View file

@ -100,6 +100,7 @@ signal_waiting_threads (AsyncJitterQueue * queue)
{
if (async_jitter_queue_length_ts_units_unlocked (queue) >=
queue->high_threshold * queue->max_queue_length) {
GST_DEBUG ("stop buffering");
queue->buffering = FALSE;
}
@ -473,6 +474,7 @@ async_jitter_queue_pop_intern_unlocked (AsyncJitterQueue * queue)
{
gpointer retval;
GstBuffer *tail_buffer = NULL;
guint tsunits;
if (queue->pop_flushing)
return NULL;
@ -485,20 +487,27 @@ async_jitter_queue_pop_intern_unlocked (AsyncJitterQueue * queue)
return NULL;
}
if (async_jitter_queue_length_ts_units_unlocked (queue) <=
queue->low_threshold * queue->max_queue_length
tsunits = async_jitter_queue_length_ts_units_unlocked (queue);
GST_DEBUG ("tsunits %u, pops: %u, limit %d", tsunits, queue->pops_remaining,
queue->low_threshold * queue->max_queue_length);
if (tsunits <= queue->low_threshold * queue->max_queue_length
&& queue->pops_remaining == 0) {
if (!queue->buffering) {
GST_DEBUG ("start buffering");
queue->buffering = TRUE;
queue->pops_remaining = queue->queue->length;
} else {
while (!g_queue_peek_tail (queue->queue) || queue->pop_blocking) {
queue->waiting_threads++;
g_cond_wait (queue->cond, queue->mutex);
queue->waiting_threads--;
if (queue->pop_flushing)
return NULL;
}
}
GST_DEBUG ("wait for data");
while (!g_queue_peek_tail (queue->queue) || queue->pop_blocking) {
queue->waiting_threads++;
g_cond_wait (queue->cond, queue->mutex);
queue->waiting_threads--;
if (queue->pop_flushing)
return NULL;
}
}

51
gst/rtpmanager/gstrtpbin.c
View file

@ -84,8 +84,8 @@ GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
@ -195,7 +195,7 @@ struct _GstRTPBinSession
GstPad *recv_rtcp_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_src;
GstPad *rtcp_src;
GstPad *send_rtcp_src;
};
/* find a session with the given id. Must be called with RTP_BIN_LOCK */
@ -432,7 +432,7 @@ gst_rtp_bin_base_init (gpointer klass)
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_rtcp_src_template));
gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
@ -795,10 +795,15 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* get the session, it must exist or we error */
/* get or create the session */
session = find_session_by_id (rtpbin, sessid);
if (!session)
goto no_session;
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->recv_rtcp_sink != NULL)
@ -841,9 +846,9 @@ no_name:
g_warning ("rtpbin: invalid name given");
return NULL;
}
no_session:
create_error:
{
g_warning ("rtpbin: no session with id %d", sessid);
/* create_session already warned */
return NULL;
}
existed:
@ -872,7 +877,7 @@ link_failed:
static GstPad *
create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result, *srcpad, *srcghost;
GstPad *result, *srcghost;
gchar *gname;
guint sessid;
GstRTPBinSession *session;
@ -895,7 +900,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
if (session->send_rtp_sink != NULL)
goto existed;
/* get recv_rtp pad and store */
/* get send_rtp pad and store */
session->send_rtp_sink =
gst_element_get_request_pad (session->session, "send_rtp_sink");
if (session->send_rtp_sink == NULL)
@ -907,8 +912,9 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
/* get srcpad */
srcpad = gst_element_get_pad (session->session, "send_rtp_src");
if (srcpad == NULL)
session->send_rtp_src =
gst_element_get_static_pad (session->session, "send_rtp_src");
if (session->send_rtp_src == NULL)
goto no_srcpad;
/* ghost the new source pad */
@ -916,7 +922,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
gname = g_strdup_printf ("send_rtp_src_%d", sessid);
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
srcghost =
gst_ghost_pad_new_from_template (gname, session->send_rtp_sink, templ);
gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
gst_pad_set_active (srcghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
g_free (gname);
@ -962,7 +968,7 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
GstRTPBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "rtcp_src_%d", &sessid) != 1)
if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
goto no_name;
/* get or create session */
@ -971,16 +977,17 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
goto no_session;
/* check if pad was requested */
if (session->rtcp_src != NULL)
if (session->send_rtcp_src != NULL)
goto existed;
/* get rtcp_src pad and store */
session->rtcp_src =
session->send_rtcp_src =
gst_element_get_request_pad (session->session, "send_rtcp_src");
if (session->rtcp_src == NULL)
if (session->send_rtcp_src == NULL)
goto pad_failed;
result = gst_ghost_pad_new_from_template (name, session->rtcp_src, templ);
result =
gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
@ -999,7 +1006,8 @@ no_session:
}
existed:
{
g_warning ("rtpbin: rtcp_src pad already requested for session %d", sessid);
g_warning ("rtpbin: send_rtcp_src pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
@ -1036,7 +1044,8 @@ gst_rtp_bin_request_new_pad (GstElement * element,
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink_%d")) {
result = create_send_rtp (rtpbin, templ, name);
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%d")) {
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src_%d")) {
result = create_rtcp (rtpbin, templ, name);
} else
goto wrong_template;

2
gst/rtpmanager/gstrtpptdemux.c
View file

@ -258,6 +258,8 @@ gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
&ret);
caps = g_value_get_boxed (&ret);
if (caps == NULL)
caps = GST_PAD_CAPS (rtpdemux->sink);
if (!caps)
goto no_caps;

16
gst/rtpmanager/gstrtpsession.c
View file

@ -451,6 +451,8 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "reading receiving RTP packet");
if (rtpsession->recv_rtp_src) {
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} else {
@ -473,6 +475,8 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "sending RTP packet");
if (rtpsession->send_rtp_src) {
result = gst_pad_push (rtpsession->send_rtp_src, buffer);
} else {
@ -737,7 +741,7 @@ create_recv_rtp_sink (GstRTPSession * rtpsession)
rtpsession->recv_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
NULL);
"recv_rtp_sink");
gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
gst_rtp_session_chain_recv_rtp);
gst_pad_set_event_function (rtpsession->recv_rtp_sink,
@ -766,7 +770,7 @@ create_recv_rtcp_sink (GstRTPSession * rtpsession)
rtpsession->recv_rtcp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
NULL);
"recv_rtcp_sink");
gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_chain_recv_rtcp);
gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
@ -795,18 +799,18 @@ create_send_rtp_sink (GstRTPSession * rtpsession)
rtpsession->send_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
NULL);
"send_rtp_sink");
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
gst_rtp_session_event_send_rtp_sink);
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtcp_sink);
rtpsession->send_rtp_sink);
rtpsession->send_rtp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
NULL);
"send_rtp_src");
gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
@ -824,7 +828,7 @@ create_send_rtcp_src (GstRTPSession * rtpsession)
rtpsession->send_rtcp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
NULL);
"send_rtcp_src");
gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtcp_src);

43
gst/rtpmanager/rtpsession.c
View file

@ -622,6 +622,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
if (source == session->source) {
GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.send_rtp)
result =
@ -629,8 +630,11 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
session->user_data);
else
gst_buffer_unref (buffer);
} else {
GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
result =
session->callbacks.process_rtp (session, source, buffer,
@ -638,6 +642,8 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
else
gst_buffer_unref (buffer);
}
RTP_SESSION_LOCK (session);
return result;
}
@ -877,6 +883,7 @@ rtp_session_create_source (RTPSession * sess)
}
source = rtp_source_new (ssrc);
g_object_ref (source);
rtp_source_set_callbacks (source, &callbacks, sess);
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
source);
/* we have one more source now */
@ -1080,6 +1087,8 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
GST_DEBUG ("RB %d: %08x, %u", i, ssrc, jitter);
if (ssrc == sess->source->ssrc) {
/* only deal with report blocks for our session, we update the stats of
* the sender of the RTCP message. We could also compare our stats against
@ -1361,7 +1370,8 @@ ignore:
* @sess: an #RTPSession
* @buffer: an RTP buffer
*
* Send the RTP buffer in the session manager.
* Send the RTP buffer in the session manager. This function takes ownership of
* @buffer.
*
* Returns: a #GstFlowReturn.
*/
@ -1375,9 +1385,19 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
if (!gst_rtp_buffer_validate (buffer))
goto invalid_packet;
GST_DEBUG ("received RTP packet for sending");
RTP_SESSION_LOCK (sess);
source = sess->source;
/* update last activity */
if (sess->callbacks.get_time)
source->last_rtp_activity =
sess->callbacks.get_time (sess, sess->user_data);
prevsender = RTP_SOURCE_IS_SENDER (source);
/* we use our own source to send */
@ -1388,6 +1408,14 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
RTP_SESSION_UNLOCK (sess);
return result;
/* ERRORS */
invalid_packet:
{
gst_buffer_unref (buffer);
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}
}
static GstClockTime
@ -1534,13 +1562,22 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
if (RTP_SOURCE_IS_SENDER (own)) {
guint64 ntptime;
guint32 rtptime;
/* we are a sender, create SR */
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
/* fill in sender report info, FIXME NTP and RTP timestamps missing */
/* convert clock time to NTP time */
ntptime = gst_util_uint64_scale (data->time, (1LL << 32), GST_SECOND);
ntptime += (2208988800LL << 32);
rtptime = 0;
/* fill in sender report info, FIXME RTP timestamps missing */
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
0, 0, own->stats.packets_sent, own->stats.octets_sent);
ntptime, rtptime, own->stats.packets_sent, own->stats.octets_sent);
} else {
/* we are only receiver, create RR */
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);

3
gst/rtpmanager/rtpsession.h
View file

@ -110,7 +110,7 @@ typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data)
* @sess: an #RTPSession
* @user_data: user data specified when registering
*
* This callback will be called when @sess needs to cancel the previous timeout.
* 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.
*/
@ -122,6 +122,7 @@ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
* @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

15
gst/rtpmanager/rtpsource.c
View file

@ -453,18 +453,29 @@ GstFlowReturn
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
{
GstFlowReturn result = GST_FLOW_OK;
guint len;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
len = gst_rtp_buffer_get_payload_len (buffer);
/* we are a sender now */
src->is_sender = TRUE;
/* update stats for the SR */
src->stats.packets_sent++;
src->stats.octets_sent += len;
/* push packet */
if (src->callbacks.push_rtp)
if (src->callbacks.push_rtp) {
GST_DEBUG ("pushing RTP packet %u", src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
else
} else {
GST_DEBUG ("no callback installed");
gst_buffer_unref (buffer);
}
return result;
}