[Bug 771525] gst-rtsp-server: Poor performance with interleaved RTSP due to missing buffer list support and merging of all memories in buffers
GStreamer (GNOME Bugzilla)
bugzilla at gnome.org
Thu Oct 11 16:40:40 UTC 2018
https://bugzilla.gnome.org/show_bug.cgi?id=771525
Mathieu Duponchelle <mduponchelle1 at gmail.com> changed:
What |Removed |Added
----------------------------------------------------------------------------
Attachment #373684|reviewed |needs-work
status| |
Attachment #373684|reviewed |needs-work
status| |
Attachment #373684|reviewed reviewed |needs-work needs-work
status| |
--- Comment #31 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #32 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #33 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #34 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #35 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #36 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
--- Comment #37 from Mathieu Duponchelle <mduponchelle1 at gmail.com> ---
Review of attachment 373684:
--> (https://bugzilla.gnome.org/review?bug=771525&attachment=373684)
This will require updating sections.txt, lgtm apart from a few comments
::: gst/rtsp-server/rtsp-client.c
@@ +1187,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
The implementation seems similar to what rtsp-stream-transport will do if not
callback was provided, is this mostly a placeholder for future extension?
::: gst/rtsp-server/rtsp-media.c
@@ +2187,3 @@
g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
+ TRUE, "emit-signals", FALSE, NULL);
+ g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
emit-signals is false by default for appsink
::: gst/rtsp-server/rtsp-stream-transport.c
@@ +219,3 @@
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
Since:
@@ -531,0 +566,7 @@
+/**
+ * gst_rtsp_stream_transport_send_rtp_list:
+ * @trans: a #GstRTSPStreamTransport
... 4 more ...
Since:
@@ +614,3 @@
+ *
+ * Send @buffer_list to the installed RTCP callback for @trans.
+ *
Since:
::: gst/rtsp-server/rtsp-stream.c
@@ +2474,3 @@
+ send_ret = gst_rtsp_stream_transport_send_rtp (tr, buffer);
+ if (buffer_list)
+ send_ret = gst_rtsp_stream_transport_send_rtp_list (tr, buffer_list);
this will ignore any potential error if the sample contained both a buffer and
a buffer list, not sure if that is allowed to happen in practice
@@ +3332,3 @@
/* make appsink */
priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, "buffer-list",
emit-signals is FALSE by default in appsink
::: gst/rtsp-sink/gstrtspclientsink.c
@@ +3853,3 @@
+static gboolean
+do_send_data_list (GstBufferList * buffer_list, guint8 channel,
+ GstRTSPStreamContext * context)
Same comment as the do_send_list in rtsp-client
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