mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 13:21:28 +00:00
985897d8d9
Fix the raciness by iterating on a condition instead of using the gmainloop. Don't use the EOS as the target, otherwise the retransmission of the last packets are lost. Also count the retranmissions requests that are dropped. Check the condition before blocking on the GCond https://bugzilla.gnome.org/show_bug.cgi?id=728501
418 lines
14 KiB
C
418 lines
14 KiB
C
/* GStreamer
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*
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* Copyright (C) 2013 Collabora Ltd.
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* @author Julien Isorce <julien.isorce@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstconsistencychecker.h>
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#include <gst/check/gsttestclock.h>
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#include <gst/rtp/gstrtpbuffer.h>
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static gboolean send_pipeline_eos = FALSE;
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static gboolean receive_pipeline_eos = FALSE;
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static void
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message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_EOS:
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if (!strcmp ("pipeline_send",
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GST_OBJECT_NAME (GST_MESSAGE_SRC (message))))
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send_pipeline_eos = TRUE;
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else if (!strcmp ("pipeline_receive",
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GST_OBJECT_NAME (GST_MESSAGE_SRC (message))))
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receive_pipeline_eos = TRUE;
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else
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fail ("Unknown pipeline: %s",
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GST_OBJECT_NAME (GST_MESSAGE_SRC (message)));
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break;
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case GST_MESSAGE_WARNING:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_warning (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_ERROR:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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fail ("Error!");
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break;
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}
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default:
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break;
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}
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}
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typedef struct
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{
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guint count;
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guint nb_packets;
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guint drop_every_n_packets;
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} RTXSendData;
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static GstPadProbeReturn
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rtprtxsend_srcpad_probe (GstPad * pad, GstPadProbeInfo * info,
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gpointer user_data)
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{
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GstPadProbeReturn ret = GST_PAD_PROBE_OK;
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if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
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GstBuffer *buffer = GST_BUFFER (info->data);
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RTXSendData *rtxdata = (RTXSendData *) user_data;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint payload_type = 0;
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gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
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payload_type = gst_rtp_buffer_get_payload_type (&rtp);
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/* main stream packets */
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if (payload_type == 96) {
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/* count packets of the main stream */
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++rtxdata->nb_packets;
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/* drop some packets */
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if (rtxdata->count < rtxdata->drop_every_n_packets) {
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++rtxdata->count;
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} else {
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/* drop a packet every 'rtxdata->count' packets */
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rtxdata->count = 1;
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ret = GST_PAD_PROBE_DROP;
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}
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} else {
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/* retransmission packets */
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}
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gst_rtp_buffer_unmap (&rtp);
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}
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return ret;
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}
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static void
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on_rtpbinreceive_pad_added (GstElement * element, GstPad * newPad,
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gpointer data)
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{
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GstElement *rtpdepayloader = GST_ELEMENT (data);
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gchar *padName = gst_pad_get_name (newPad);
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if (g_str_has_prefix (padName, "recv_rtp_src_")) {
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GstPad *sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
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gst_pad_link (newPad, sinkpad);
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gst_object_unref (sinkpad);
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}
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g_free (padName);
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}
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static gboolean
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on_timeout (gpointer data)
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{
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GstEvent *eos = gst_event_new_eos ();
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if (!gst_element_send_event (GST_ELEMENT (data), eos)) {
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GST_ERROR ("failed to send end of stream event");
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gst_event_unref (eos);
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}
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return FALSE;
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}
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static GstElement *
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request_aux_receive (GstElement * rtpbin, guint sessid, GstElement * receive)
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{
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GstElement *bin;
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GstPad *pad;
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GST_INFO ("creating AUX receiver");
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bin = gst_bin_new (NULL);
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gst_bin_add (GST_BIN (bin), receive);
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pad = gst_element_get_static_pad (receive, "src");
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gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
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gst_object_unref (pad);
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pad = gst_element_get_static_pad (receive, "sink");
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gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
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gst_object_unref (pad);
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return bin;
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}
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static GstElement *
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request_aux_send (GstElement * rtpbin, guint sessid, GstElement * send)
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{
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GstElement *bin;
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GstPad *pad;
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GST_INFO ("creating AUX sender");
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bin = gst_bin_new (NULL);
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gst_bin_add (GST_BIN (bin), send);
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pad = gst_element_get_static_pad (send, "src");
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gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
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gst_object_unref (pad);
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pad = gst_element_get_static_pad (send, "sink");
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gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
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gst_object_unref (pad);
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return bin;
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}
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GST_START_TEST (test_simple_rtpbin_aux)
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{
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GstElement *binsend, *rtpbinsend, *src, *encoder, *rtppayloader,
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*rtprtxsend, *sendrtp_udpsink, *sendrtcp_udpsink, *sendrtcp_udpsrc;
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GstElement *binreceive, *rtpbinreceive, *recvrtp_udpsrc, *recvrtcp_udpsrc,
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*recvrtcp_udpsink, *rtprtxreceive, *rtpdepayloader, *decoder, *converter,
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*sink;
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GstBus *bussend;
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GstBus *busreceive;
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gboolean res;
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GstCaps *rtpcaps = NULL;
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GstStructure *pt_map;
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GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
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GstPad *srcpad = NULL;
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guint nb_rtx_send_packets = 0;
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guint nb_rtx_recv_packets = 0;
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RTXSendData send_rtxdata;
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send_rtxdata.count = 1;
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send_rtxdata.nb_packets = 0;
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send_rtxdata.drop_every_n_packets = 50;
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GST_INFO ("preparing test");
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/* build pipeline */
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binsend = gst_pipeline_new ("pipeline_send");
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bussend = gst_element_get_bus (binsend);
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gst_bus_add_signal_watch_full (bussend, G_PRIORITY_HIGH);
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binreceive = gst_pipeline_new ("pipeline_receive");
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busreceive = gst_element_get_bus (binreceive);
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gst_bus_add_signal_watch_full (busreceive, G_PRIORITY_HIGH);
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rtpbinsend = gst_element_factory_make ("rtpbin", "rtpbinsend");
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g_object_set (rtpbinsend, "latency", 200, "do-retransmission", TRUE, NULL);
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src = gst_element_factory_make ("audiotestsrc", "src");
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encoder = gst_element_factory_make ("speexenc", "encoder");
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rtppayloader = gst_element_factory_make ("rtpspeexpay", "rtppayloader");
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rtprtxsend = gst_element_factory_make ("rtprtxsend", "rtprtxsend");
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sendrtp_udpsink = gst_element_factory_make ("udpsink", "sendrtp_udpsink");
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g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
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g_object_set (sendrtp_udpsink, "port", 5006, NULL);
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sendrtcp_udpsink = gst_element_factory_make ("udpsink", "sendrtcp_udpsink");
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g_object_set (sendrtcp_udpsink, "host", "127.0.0.1", NULL);
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g_object_set (sendrtcp_udpsink, "port", 5007, NULL);
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g_object_set (sendrtcp_udpsink, "sync", FALSE, NULL);
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g_object_set (sendrtcp_udpsink, "async", FALSE, NULL);
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sendrtcp_udpsrc = gst_element_factory_make ("udpsrc", "sendrtcp_udpsrc");
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g_object_set (sendrtcp_udpsrc, "port", 5009, NULL);
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rtpbinreceive = gst_element_factory_make ("rtpbin", "rtpbinreceive");
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g_object_set (rtpbinreceive, "latency", 200, "do-retransmission", TRUE, NULL);
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recvrtp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtp_udpsrc");
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g_object_set (recvrtp_udpsrc, "port", 5006, NULL);
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rtpcaps =
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gst_caps_from_string
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("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1");
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g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL);
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gst_caps_unref (rtpcaps);
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recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc");
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g_object_set (recvrtcp_udpsrc, "port", 5007, NULL);
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recvrtcp_udpsink = gst_element_factory_make ("udpsink", "recvrtcp_udpsink");
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g_object_set (recvrtcp_udpsink, "host", "127.0.0.1", NULL);
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g_object_set (recvrtcp_udpsink, "port", 5009, NULL);
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g_object_set (recvrtcp_udpsink, "sync", FALSE, NULL);
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g_object_set (recvrtcp_udpsink, "async", FALSE, NULL);
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rtprtxreceive = gst_element_factory_make ("rtprtxreceive", "rtprtxreceive");
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rtpdepayloader = gst_element_factory_make ("rtpspeexdepay", "rtpdepayloader");
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decoder = gst_element_factory_make ("speexdec", "decoder");
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converter = gst_element_factory_make ("identity", "converter");
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sink = gst_element_factory_make ("fakesink", "sink");
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g_object_set (sink, "sync", TRUE, NULL);
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gst_bin_add_many (GST_BIN (binsend), rtpbinsend, src, encoder, rtppayloader,
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sendrtp_udpsink, sendrtcp_udpsink, sendrtcp_udpsrc, NULL);
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gst_bin_add_many (GST_BIN (binreceive), rtpbinreceive,
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recvrtp_udpsrc, recvrtcp_udpsrc, recvrtcp_udpsink,
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rtpdepayloader, decoder, converter, sink, NULL);
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g_signal_connect (rtpbinreceive, "pad-added",
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G_CALLBACK (on_rtpbinreceive_pad_added), rtpdepayloader);
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pt_map = gst_structure_new ("application/x-rtp-pt-map",
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"96", G_TYPE_UINT, 99, NULL);
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g_object_set (rtppayloader, "pt", 96, NULL);
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g_object_set (rtppayloader, "seqnum-offset", 1, NULL);
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g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL);
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g_object_set (rtprtxreceive, "payload-type-map", pt_map, NULL);
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gst_structure_free (pt_map);
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/* set rtp aux receive */
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g_signal_connect (rtpbinreceive, "request-aux-receiver", (GCallback)
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request_aux_receive, rtprtxreceive);
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/* set rtp aux send */
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g_signal_connect (rtpbinsend, "request-aux-sender", (GCallback)
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request_aux_send, rtprtxsend);
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/* gst-launch-1.0 rtpbin name=rtpbin audiotestsrc ! amrnbenc ! rtpamrpay ! \
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* rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host="127.0.0.1" \
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* port=5002 rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5003 \
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* sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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*/
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res = gst_element_link (src, encoder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (encoder, rtppayloader);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (rtppayloader, "src", rtpbinsend,
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"send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (rtpbinsend, "send_rtp_src_0", sendrtp_udpsink,
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"sink", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (rtpbinsend, "send_rtcp_src_0",
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sendrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (sendrtcp_udpsrc, "src", rtpbinsend,
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"recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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srcpad = gst_element_get_static_pad (rtpbinsend, "send_rtp_src_0");
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gst_pad_add_probe (srcpad,
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(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
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(GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL);
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gst_object_unref (srcpad);
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/* gst-launch-1.0 rtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio, \
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* clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,o
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* ctet-align=(string)1" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! \
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* amrnbdec ! fakesink sync=True udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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* rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5007 sync=false async=false
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*/
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res =
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gst_element_link_pads_full (recvrtp_udpsrc, "src", rtpbinreceive,
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"recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (rtpdepayloader, "src", decoder, "sink",
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GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (decoder, converter);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (converter, "src", sink, "sink",
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GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (recvrtcp_udpsrc, "src", rtpbinreceive,
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"recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res =
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gst_element_link_pads_full (rtpbinreceive, "send_rtcp_src_0",
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recvrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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g_signal_connect (bussend, "message::error", (GCallback) message_received,
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binsend);
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g_signal_connect (bussend, "message::warning", (GCallback) message_received,
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binsend);
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g_signal_connect (bussend, "message::eos", (GCallback) message_received,
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binsend);
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g_signal_connect (busreceive, "message::error", (GCallback) message_received,
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binreceive);
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g_signal_connect (busreceive, "message::warning",
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(GCallback) message_received, binreceive);
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g_signal_connect (busreceive, "message::eos", (GCallback) message_received,
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binreceive);
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state_res = gst_element_set_state (binreceive, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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state_res = gst_element_set_state (binsend, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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g_timeout_add (5000, on_timeout, binsend);
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g_timeout_add (5000, on_timeout, binreceive);
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GST_INFO ("enter mainloop");
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while (!send_pipeline_eos && !receive_pipeline_eos)
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g_main_context_iteration (NULL, TRUE);
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GST_INFO ("exit mainloop");
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/* check that FB NACK is working */
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g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nb_rtx_send_packets,
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NULL);
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g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests",
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&nb_rtx_recv_packets, NULL);
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state_res = gst_element_set_state (binsend, GST_STATE_NULL);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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state_res = gst_element_set_state (binreceive, GST_STATE_NULL);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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GST_INFO ("nb_rtx_send_packets %d", nb_rtx_send_packets);
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GST_INFO ("nb_rtx_recv_packets %d", nb_rtx_recv_packets);
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fail_if (nb_rtx_send_packets < 1);
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fail_if (nb_rtx_recv_packets < 1);
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/* cleanup */
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gst_bus_remove_signal_watch (bussend);
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gst_object_unref (bussend);
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gst_object_unref (binsend);
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gst_bus_remove_signal_watch (busreceive);
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gst_object_unref (busreceive);
|
|
gst_object_unref (binreceive);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtpaux_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtpaux");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
tcase_set_timeout (tc_chain, 10000);
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
|
|
tcase_add_test (tc_chain, test_simple_rtpbin_aux);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtpaux);
|