| /* 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 gboolean send_pipeline_eos = FALSE; |
| static gboolean receive_pipeline_eos = FALSE; |
| |
| 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: |
| if (!strcmp ("pipeline_send", |
| GST_OBJECT_NAME (GST_MESSAGE_SRC (message)))) |
| send_pipeline_eos = TRUE; |
| else if (!strcmp ("pipeline_receive", |
| GST_OBJECT_NAME (GST_MESSAGE_SRC (message)))) |
| receive_pipeline_eos = TRUE; |
| else |
| fail ("Unknown pipeline: %s", |
| GST_OBJECT_NAME (GST_MESSAGE_SRC (message))); |
| 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); |
| fail ("Error!"); |
| 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; |
| GstStructure *pt_map; |
| 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 = 25; |
| |
| 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 ("alawenc", "encoder"); |
| rtppayloader = gst_element_factory_make ("rtppcmapay", "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)PCMA,payload=(int)96"); |
| 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 ("rtppcmadepay", "rtpdepayloader"); |
| decoder = gst_element_factory_make ("alawdec", "decoder"); |
| converter = gst_element_factory_make ("identity", "converter"); |
| sink = gst_element_factory_make ("fakesink", "sink"); |
| g_object_set (sink, "sync", TRUE, NULL); |
| |
| 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); |
| |
| pt_map = gst_structure_new ("application/x-rtp-pt-map", |
| "96", G_TYPE_UINT, 99, NULL); |
| g_object_set (rtppayloader, "pt", 96, NULL); |
| g_object_set (rtppayloader, "seqnum-offset", 1, NULL); |
| g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL); |
| g_object_set (rtprtxreceive, "payload-type-map", pt_map, NULL); |
| gst_structure_free (pt_map); |
| |
| /* 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 ! fakesink sync=True 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); |
| |
| 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"); |
| while (!send_pipeline_eos && !receive_pipeline_eos) |
| g_main_context_iteration (NULL, TRUE); |
| 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 */ |
| 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); |