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