--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2013 Collabora Ltd.
+ * @author Julien Isorce <julien.isorce@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+static GMainLoop *main_loop;
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+ GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+ GST_MESSAGE_SRC (message), message);
+
+ switch (message->type) {
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (main_loop);
+ break;
+ case GST_MESSAGE_WARNING:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ERROR:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ g_main_loop_quit (main_loop);
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+typedef struct
+{
+ guint count;
+ guint nb_packets;
+ guint drop_every_n_packets;
+} RTXSendData;
+
+static GstPadProbeReturn
+rtprtxsend_srcpad_probe (GstPad * pad, GstPadProbeInfo * info,
+ gpointer user_data)
+{
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
+ GstBuffer *buffer = GST_BUFFER (info->data);
+ RTXSendData *rtxdata = (RTXSendData *) user_data;
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+ guint payload_type = 0;
+
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ payload_type = gst_rtp_buffer_get_payload_type (&rtp);
+
+ /* main stream packets */
+ if (payload_type == 96) {
+ /* count packets of the main stream */
+ ++rtxdata->nb_packets;
+ /* drop some packets */
+ if (rtxdata->count < rtxdata->drop_every_n_packets) {
+ ++rtxdata->count;
+ } else {
+ /* drop a packet every 'rtxdata->count' packets */
+ rtxdata->count = 1;
+ ret = GST_PAD_PROBE_DROP;
+ }
+ } else {
+ /* retransmission packets */
+ }
+
+ gst_rtp_buffer_unmap (&rtp);
+ }
+
+ return ret;
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * element, GstPad * newPad,
+ gpointer data)
+{
+ GstElement *rtpdepayloader = GST_ELEMENT (data);
+
+ gchar *padName = gst_pad_get_name (newPad);
+ if (g_str_has_prefix (padName, "recv_rtp_src_")) {
+ GstPad *sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+ gst_pad_link (newPad, sinkpad);
+ gst_object_unref (sinkpad);
+ }
+ g_free (padName);
+}
+
+static gboolean
+on_timeout (gpointer data)
+{
+ GstEvent *eos = gst_event_new_eos ();
+ if (!gst_element_send_event (GST_ELEMENT (data), eos)) {
+ GST_ERROR ("failed to send end of stream event");
+ gst_event_unref (eos);
+ }
+
+ return FALSE;
+}
+
+static GstElement *
+request_aux_receive (GstElement * rtpbin, guint sessid, GstElement * receive)
+{
+ GstElement *bin;
+ GstPad *pad;
+
+ GST_INFO ("creating AUX receiver");
+ bin = gst_bin_new (NULL);
+ gst_bin_add (GST_BIN (bin), receive);
+
+ pad = gst_element_get_static_pad (receive, "src");
+ gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
+ gst_object_unref (pad);
+ pad = gst_element_get_static_pad (receive, "sink");
+ gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
+ gst_object_unref (pad);
+
+ return bin;
+}
+
+static GstElement *
+request_aux_send (GstElement * rtpbin, guint sessid, GstElement * send)
+{
+ GstElement *bin;
+ GstPad *pad;
+
+ GST_INFO ("creating AUX sender");
+ bin = gst_bin_new (NULL);
+ gst_bin_add (GST_BIN (bin), send);
+
+ pad = gst_element_get_static_pad (send, "src");
+ gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
+ gst_object_unref (pad);
+ pad = gst_element_get_static_pad (send, "sink");
+ gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
+ gst_object_unref (pad);
+
+ return bin;
+}
+
+
+GST_START_TEST (test_simple_rtpbin_aux)
+{
+ GstElement *binsend, *rtpbinsend, *src, *encoder, *rtppayloader,
+ *rtprtxsend, *sendrtp_udpsink, *sendrtcp_udpsink, *sendrtcp_udpsrc;
+ GstElement *binreceive, *rtpbinreceive, *recvrtp_udpsrc, *recvrtcp_udpsrc,
+ *recvrtcp_udpsink, *rtprtxreceive, *rtpdepayloader, *decoder, *converter,
+ *sink;
+ GstBus *bussend;
+ GstBus *busreceive;
+ gboolean res;
+ GstCaps *rtpcaps = NULL;
+ GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+ GstPad *srcpad = NULL;
+ guint nb_rtx_send_packets = 0;
+ guint nb_rtx_recv_packets = 0;
+ RTXSendData send_rtxdata;
+ send_rtxdata.count = 1;
+ send_rtxdata.nb_packets = 0;
+ send_rtxdata.drop_every_n_packets = 50;
+
+ GST_INFO ("preparing test");
+
+ /* build pipeline */
+ binsend = gst_pipeline_new ("pipeline_send");
+ bussend = gst_element_get_bus (binsend);
+ gst_bus_add_signal_watch_full (bussend, G_PRIORITY_HIGH);
+
+ binreceive = gst_pipeline_new ("pipeline_receive");
+ busreceive = gst_element_get_bus (binreceive);
+ gst_bus_add_signal_watch_full (busreceive, G_PRIORITY_HIGH);
+
+ rtpbinsend = gst_element_factory_make ("rtpbin", "rtpbinsend");
+ g_object_set (rtpbinsend, "latency", 200, "do-retransmission", TRUE, NULL);
+ src = gst_element_factory_make ("audiotestsrc", "src");
+ encoder = gst_element_factory_make ("speexenc", "encoder");
+ rtppayloader = gst_element_factory_make ("rtpspeexpay", "rtppayloader");
+ rtprtxsend = gst_element_factory_make ("rtprtxsend", "rtprtxsend");
+ sendrtp_udpsink = gst_element_factory_make ("udpsink", "sendrtp_udpsink");
+ g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (sendrtp_udpsink, "port", 5006, NULL);
+ sendrtcp_udpsink = gst_element_factory_make ("udpsink", "sendrtcp_udpsink");
+ g_object_set (sendrtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (sendrtcp_udpsink, "port", 5007, NULL);
+ g_object_set (sendrtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (sendrtcp_udpsink, "async", FALSE, NULL);
+ sendrtcp_udpsrc = gst_element_factory_make ("udpsrc", "sendrtcp_udpsrc");
+ g_object_set (sendrtcp_udpsrc, "port", 5009, NULL);
+
+ rtpbinreceive = gst_element_factory_make ("rtpbin", "rtpbinreceive");
+ g_object_set (rtpbinreceive, "latency", 200, "do-retransmission", TRUE, NULL);
+ recvrtp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtp_udpsrc");
+ g_object_set (recvrtp_udpsrc, "port", 5006, NULL);
+ rtpcaps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1");
+ g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL);
+ gst_caps_unref (rtpcaps);
+ recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc");
+ g_object_set (recvrtcp_udpsrc, "port", 5007, NULL);
+ recvrtcp_udpsink = gst_element_factory_make ("udpsink", "recvrtcp_udpsink");
+ g_object_set (recvrtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (recvrtcp_udpsink, "port", 5009, NULL);
+ g_object_set (recvrtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (recvrtcp_udpsink, "async", FALSE, NULL);
+ rtprtxreceive = gst_element_factory_make ("rtprtxreceive", "rtprtxreceive");
+ rtpdepayloader = gst_element_factory_make ("rtpspeexdepay", "rtpdepayloader");
+ decoder = gst_element_factory_make ("speexdec", "decoder");
+ converter = gst_element_factory_make ("identity", "converter");
+ sink = gst_element_factory_make ("alsasink", "sink");
+
+ gst_bin_add_many (GST_BIN (binsend), rtpbinsend, src, encoder, rtppayloader,
+ sendrtp_udpsink, sendrtcp_udpsink, sendrtcp_udpsrc, NULL);
+
+ gst_bin_add_many (GST_BIN (binreceive), rtpbinreceive,
+ recvrtp_udpsrc, recvrtcp_udpsrc, recvrtcp_udpsink,
+ rtpdepayloader, decoder, converter, sink, NULL);
+
+ g_signal_connect (rtpbinreceive, "pad-added",
+ G_CALLBACK (on_rtpbinreceive_pad_added), rtpdepayloader);
+
+ g_object_set (rtppayloader, "pt", 96, NULL);
+ g_object_set (rtppayloader, "seqnum-offset", 1, NULL);
+ g_object_set (rtprtxsend, "rtx-payload-type", 99, NULL);
+ g_object_set (rtprtxreceive, "rtx-payload-types", "99:111:125", NULL);
+
+ /* set rtp aux receive */
+ g_signal_connect (rtpbinreceive, "request-aux-receiver", (GCallback)
+ request_aux_receive, rtprtxreceive);
+ /* set rtp aux send */
+ g_signal_connect (rtpbinsend, "request-aux-sender", (GCallback)
+ request_aux_send, rtprtxsend);
+
+ /* gst-launch-1.0 rtpbin name=rtpbin audiotestsrc ! amrnbenc ! rtpamrpay ! \
+ * rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host="127.0.0.1" \
+ * port=5002 rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5003 \
+ * sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
+ */
+
+ res = gst_element_link (src, encoder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (encoder, rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (rtppayloader, "src", rtpbinsend,
+ "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (rtpbinsend, "send_rtp_src_0", sendrtp_udpsink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (rtpbinsend, "send_rtcp_src_0",
+ sendrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (sendrtcp_udpsrc, "src", rtpbinsend,
+ "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ srcpad = gst_element_get_static_pad (rtpbinsend, "send_rtp_src_0");
+ gst_pad_add_probe (srcpad,
+ (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
+ (GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL);
+ gst_object_unref (srcpad);
+
+ /* gst-launch-1.0 rtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio, \
+ * clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,o
+ * ctet-align=(string)1" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! \
+ * amrnbdec ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
+ * rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5007 sync=false async=false
+ */
+
+ res =
+ gst_element_link_pads_full (recvrtp_udpsrc, "src", rtpbinreceive,
+ "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (rtpdepayloader, "src", decoder, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (decoder, converter);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (converter, "src", sink, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (recvrtcp_udpsrc, "src", rtpbinreceive,
+ "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (rtpbinreceive, "send_rtcp_src_0",
+ recvrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ main_loop = g_main_loop_new (NULL, FALSE);
+ g_signal_connect (bussend, "message::error", (GCallback) message_received,
+ binsend);
+ g_signal_connect (bussend, "message::warning", (GCallback) message_received,
+ binsend);
+ g_signal_connect (bussend, "message::eos", (GCallback) message_received,
+ binsend);
+
+ g_signal_connect (busreceive, "message::error", (GCallback) message_received,
+ binreceive);
+ g_signal_connect (busreceive, "message::warning",
+ (GCallback) message_received, binreceive);
+ g_signal_connect (busreceive, "message::eos", (GCallback) message_received,
+ binreceive);
+
+ state_res = gst_element_set_state (binreceive, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (binsend, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ g_timeout_add (5000, on_timeout, binsend);
+ g_timeout_add (5000, on_timeout, binreceive);
+
+ GST_INFO ("enter mainloop");
+ g_main_loop_run (main_loop);
+ g_main_loop_run (main_loop);
+ GST_INFO ("exit mainloop");
+
+ /* check that FB NACK is working */
+ g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nb_rtx_send_packets,
+ NULL);
+ g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests",
+ &nb_rtx_recv_packets, NULL);
+
+ state_res = gst_element_set_state (binsend, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (binreceive, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ GST_INFO ("nb_rtx_send_packets %d", nb_rtx_send_packets);
+ GST_INFO ("nb_rtx_recv_packets %d", nb_rtx_recv_packets);
+ fail_if (nb_rtx_send_packets < 1);
+ fail_if (nb_rtx_recv_packets < 1);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+
+ gst_bus_remove_signal_watch (bussend);
+ gst_object_unref (bussend);
+ gst_object_unref (binsend);
+
+ gst_bus_remove_signal_watch (busreceive);
+ 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);
projects / platform / upstream / gstreamer.git / commitdiff