[Bug 760532] New: Seeking continuously cause rtsp playback (tcp) to stall

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=760532

            Bug ID: 760532
           Summary: Seeking continuously cause rtsp playback (tcp) to
                    stall
    Classification: Platform
           Product: GStreamer
           Version: 1.7.1
                OS: Windows
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-rtsp-server
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: stevenhoving at hotmail.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

I'm having a issue at the moment with continuously seeks with RTSP playback.
The rtsp client is connected with TCP interleave and I am seeking continuously,
lets say every second or so I do a new seek. Then sometimes after seeking 3-5
times the video playback stalls for 20 seconds or so, but sometimes I can seek
for 30 times before it happens.

Server:
rtsp server serving a mkv file, the pipeline looks something like:
.---------.  .---------------.  .------------.
| filesrc |->| matroskademux |->| rtph264pay |
'---------'  '---------------'  '------------'

But I have also tried adding queues in between the elements.

The video file:
  resolution: 1280x720
  bitrate: 4096kb
  framerate: 30fps
  duration: 60 min
  gov length: 10

Client:
rtspsrc location=rtsp://127.0.0.1:8554/something protocols=0x4 ! queue !
rtph264depay! avdec_h264 ! autovideosink


Test:
I have tried to create a test application for this, but it seems to not always
happen.

#include <gst/gst.h>

void do_seek(GstElement *pipeline, gint64 timestamp)
{
    unsigned int seek_flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT |
GST_SEEK_FLAG_SNAP_NEAREST;
    gst_element_seek_simple(pipeline, GST_FORMAT_TIME, seek_flags, timestamp);
}

int main()
{
    GstElement *pipeline;
    GstBus *bus;
    GstMessage *msg;
    gint64 position = GST_SECOND * 5;

    // initialize GStreamer
    gst_init(NULL, NULL);

    pipeline = gst_parse_launch("rtspsrc name=rtspsrc ! queue ! rtph264depay !
fakesink", NULL);

    GstElement *rtspsrc = gst_bin_get_by_name(GST_BIN(pipeline), "rtspsrc");
    g_object_set(rtspsrc,
        "location", "rtsp://localhost:8554/something",
        "protocols", 4, // 4 = GST_RTSP_LOWER_TRANS_TCP
        NULL);
    g_object_unref(rtspsrc);

    gst_element_set_state(pipeline, GST_STATE_PLAYING);
    gst_element_get_state(pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);

    // Wait until error, eos or async done
    bus = gst_element_get_bus(pipeline);

    unsigned int bus_flags = GST_MESSAGE_ERROR | GST_MESSAGE_EOS |
GST_MESSAGE_ASYNC_DONE;
    msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,
(GstMessageType)(bus_flags));

    unsigned int seek_count = 0;
    while(GST_MESSAGE_TYPE(msg) != GST_MESSAGE_EOS)
    {
        g_print("seek (%u) again... :D\n", seek_count++);
        position += GST_SECOND;
        do_seek(pipeline, position);

        gst_message_unref(msg);
        msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,
(GstMessageType)(bus_flags));
    }

    // Free resources
    gst_message_unref(msg);
    gst_object_unref(bus);
    gst_element_set_state(pipeline, GST_STATE_NULL);
    gst_object_unref(pipeline);

    g_print("done");

    return 0;
}

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