mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-26 11:41:09 +00:00
srtpdec: add counts in stats
In order to count the buffers which have been received and dropped for decryption reason, add a stats to track it. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
This commit is contained in:
parent
9f19ca68b3
commit
c77d07752a
3 changed files with 34 additions and 18 deletions
|
@ -225339,7 +225339,7 @@
|
|||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"default": "application/x-srtp-decoder-stats, streams=(int)< >;",
|
||||
"default": "application/x-srtp-decoder-stats, streams=(int)< >, recv-count=(uint)0, recv-drop-count=(uint)0;",
|
||||
"mutable": "null",
|
||||
"readable": true,
|
||||
"type": "GstStructure",
|
||||
|
|
|
@ -229,6 +229,8 @@ struct _GstSrtpDecSsrcStream
|
|||
GstSrtpCipherType rtcp_cipher;
|
||||
GstSrtpAuthType rtcp_auth;
|
||||
GArray *keys;
|
||||
guint recv_count;
|
||||
guint recv_drop_count;
|
||||
};
|
||||
|
||||
#ifdef HAVE_SRTP2
|
||||
|
@ -435,10 +437,11 @@ gst_srtp_dec_create_stats (GstSrtpDec * filter)
|
|||
|
||||
if (filter->session) {
|
||||
GHashTableIter iter;
|
||||
gpointer key;
|
||||
gpointer key, value;
|
||||
|
||||
g_hash_table_iter_init (&iter, filter->streams);
|
||||
while (g_hash_table_iter_next (&iter, &key, NULL)) {
|
||||
while (g_hash_table_iter_next (&iter, &key, &value)) {
|
||||
GstSrtpDecSsrcStream *stream = value;
|
||||
GstStructure *ss;
|
||||
guint32 ssrc = GPOINTER_TO_UINT (key);
|
||||
srtp_err_status_t status;
|
||||
|
@ -450,7 +453,9 @@ gst_srtp_dec_create_stats (GstSrtpDec * filter)
|
|||
}
|
||||
|
||||
ss = gst_structure_new ("application/x-srtp-stream",
|
||||
"ssrc", G_TYPE_UINT, ssrc, "roc", G_TYPE_UINT, roc, NULL);
|
||||
"ssrc", G_TYPE_UINT, ssrc, "roc", G_TYPE_UINT, roc, "recv-count",
|
||||
G_TYPE_UINT, stream->recv_count, "recv-drop-count", G_TYPE_UINT,
|
||||
stream->recv_drop_count, NULL);
|
||||
|
||||
g_value_take_boxed (&v, ss);
|
||||
gst_value_array_append_value (&va, &v);
|
||||
|
@ -458,6 +463,11 @@ gst_srtp_dec_create_stats (GstSrtpDec * filter)
|
|||
}
|
||||
|
||||
gst_structure_take_value (s, "streams", &va);
|
||||
gst_structure_set (s, "recv-count", G_TYPE_UINT, filter->recv_count, NULL);
|
||||
gst_structure_set (s, "recv-drop-count", G_TYPE_UINT,
|
||||
filter->recv_drop_count, NULL);
|
||||
GST_LOG_OBJECT (filter, "stats: recv-count %u recv-drop-count %u",
|
||||
filter->recv_count, filter->recv_drop_count);
|
||||
g_value_unset (&v);
|
||||
|
||||
return s;
|
||||
|
@ -1325,11 +1335,12 @@ gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf,
|
|||
GstMapInfo map;
|
||||
srtp_err_status_t err;
|
||||
gint size;
|
||||
GstSrtpDecSsrcStream *stream;
|
||||
|
||||
GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT
|
||||
" with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf),
|
||||
ssrc);
|
||||
|
||||
filter->recv_count++;
|
||||
/* Change buffer to remove protection */
|
||||
buf = gst_buffer_make_writable (buf);
|
||||
|
||||
|
@ -1342,7 +1353,7 @@ unprotect:
|
|||
|
||||
if (is_rtcp) {
|
||||
#ifdef HAVE_SRTP2
|
||||
GstSrtpDecSsrcStream *stream = find_stream_by_ssrc (filter, ssrc);
|
||||
stream = find_stream_by_ssrc (filter, ssrc);
|
||||
|
||||
err = srtp_unprotect_rtcp_mki (filter->session, map.data, &size,
|
||||
stream && stream->keys);
|
||||
|
@ -1381,7 +1392,7 @@ unprotect:
|
|||
|
||||
#ifdef HAVE_SRTP2
|
||||
{
|
||||
GstSrtpDecSsrcStream *stream = find_stream_by_ssrc (filter, ssrc);
|
||||
stream = find_stream_by_ssrc (filter, ssrc);
|
||||
|
||||
err = srtp_unprotect_mki (filter->session, map.data, &size,
|
||||
stream && stream->keys);
|
||||
|
@ -1390,7 +1401,12 @@ unprotect:
|
|||
err = srtp_unprotect (filter->session, map.data, &size);
|
||||
#endif
|
||||
}
|
||||
|
||||
stream = find_stream_by_ssrc (filter, ssrc);
|
||||
if (stream == NULL) {
|
||||
GST_WARNING_OBJECT (filter, "Could not find matching stream, dropping");
|
||||
goto err;
|
||||
}
|
||||
stream->recv_count++;
|
||||
/* Signal user depending on type of error */
|
||||
switch (err) {
|
||||
case srtp_err_status_ok:
|
||||
|
@ -1399,20 +1415,14 @@ unprotect:
|
|||
case srtp_err_status_replay_fail:
|
||||
GST_DEBUG_OBJECT (filter,
|
||||
"Dropping replayed packet, probably retransmission");
|
||||
stream->recv_drop_count++;
|
||||
goto err;
|
||||
case srtp_err_status_replay_old:
|
||||
GST_DEBUG_OBJECT (filter,
|
||||
"Dropping replayed old packet, probably retransmission");
|
||||
stream->recv_drop_count++;
|
||||
goto err;
|
||||
case srtp_err_status_key_expired:{
|
||||
GstSrtpDecSsrcStream *stream;
|
||||
|
||||
/* Check we have an existing stream to rekey */
|
||||
stream = find_stream_by_ssrc (filter, ssrc);
|
||||
if (stream == NULL) {
|
||||
GST_WARNING_OBJECT (filter, "Could not find matching stream, dropping");
|
||||
goto err;
|
||||
}
|
||||
|
||||
GST_OBJECT_UNLOCK (filter);
|
||||
stream = request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT);
|
||||
|
@ -1428,21 +1438,24 @@ unprotect:
|
|||
}
|
||||
case srtp_err_status_auth_fail:
|
||||
GST_WARNING_OBJECT (filter, "Error authentication packet, dropping");
|
||||
stream->recv_drop_count++;
|
||||
goto err;
|
||||
case srtp_err_status_cipher_fail:
|
||||
GST_WARNING_OBJECT (filter, "Error while decrypting packet, dropping");
|
||||
stream->recv_drop_count++;
|
||||
goto err;
|
||||
default:
|
||||
GST_WARNING_OBJECT (pad,
|
||||
"Unable to unprotect buffer (unprotect failed code %d)", err);
|
||||
stream->recv_drop_count++;
|
||||
goto err;
|
||||
}
|
||||
|
||||
gst_buffer_unmap (buf, &map);
|
||||
gst_buffer_set_size (buf, size);
|
||||
return TRUE;
|
||||
|
||||
err:
|
||||
filter->recv_drop_count++;
|
||||
gst_buffer_unmap (buf, &map);
|
||||
return FALSE;
|
||||
}
|
||||
|
@ -1541,6 +1554,8 @@ gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
|
|||
|
||||
filter->rtp_has_segment = FALSE;
|
||||
filter->rtcp_has_segment = FALSE;
|
||||
filter->recv_count = 0;
|
||||
filter->recv_drop_count = 0;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
break;
|
||||
|
@ -1560,7 +1575,6 @@ gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
|
|||
gst_srtp_dec_clear_streams (filter);
|
||||
g_hash_table_unref (filter->streams);
|
||||
filter->streams = NULL;
|
||||
|
||||
#ifndef HAVE_SRTP2
|
||||
g_hash_table_unref (filter->streams_roc_changed);
|
||||
filter->streams_roc_changed = NULL;
|
||||
|
|
|
@ -82,6 +82,8 @@ struct _GstSrtpDec
|
|||
|
||||
gboolean rtp_has_segment;
|
||||
gboolean rtcp_has_segment;
|
||||
guint recv_count;
|
||||
guint recv_drop_count;
|
||||
|
||||
#ifndef HAVE_SRTP2
|
||||
GHashTable *streams_roc_changed;
|
||||
|
|
Loading…
Reference in a new issue