leio 14/12/30 21:46:43 Added: gst-plugins-good-1.4.5-rtp-test-fixes.patch Log: Version bump. Many updates since 1.2.x series as this is a many months overdue next stable cycle upgrade. Includes new rtpstreampay, rtpstreamdepay and rtprtx* RTP elements. (Portage version: 2.2.15/cvs/Linux x86_64, unsigned Manifest commit)
Revision Changes Path 1.1 media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch file : http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch?rev=1.1&view=markup plain: http://sources.gentoo.org/viewvc.cgi/gentoo-x86/media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch?rev=1.1&content-type=text/plain Index: gst-plugins-good-1.4.5-rtp-test-fixes.patch =================================================================== Upstream commits d416336 and d67da4c diff --git a/tests/check/elements/rtpaux.c b/tests/check/elements/rtpaux.c index 1f410bf..729604a 100644 --- a/tests/check/elements/rtpaux.c +++ b/tests/check/elements/rtpaux.c @@ -218,8 +218,8 @@ GST_START_TEST (test_simple_rtpbin_aux) 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"); + 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); @@ -238,7 +238,7 @@ GST_START_TEST (test_simple_rtpbin_aux) 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"); + ("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8"); g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL); gst_caps_unref (rtpcaps); recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc"); @@ -249,8 +249,8 @@ GST_START_TEST (test_simple_rtpbin_aux) 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"); + 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); diff --git a/tests/check/elements/rtpcollision.c b/tests/check/elements/rtpcollision.c index e9528f9..16f665f 100644 --- a/tests/check/elements/rtpcollision.c +++ b/tests/check/elements/rtpcollision.c @@ -156,7 +156,7 @@ fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer) return GST_FLOW_OK; } -/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ +/* This test build the pipeline audiotestsrc ! alawenc ! rtppcmapay ! \ * rtpsession ! fakesink * It manually pushs buffer into rtpsession with same ssrc but different * ip so that collision can be detected @@ -186,9 +186,9 @@ GST_START_TEST (test_master_ssrc_collision) src = gst_element_factory_make ("audiotestsrc", "src"); g_object_set (src, "num-buffers", 5, NULL); - encoder = gst_element_factory_make ("speexenc", NULL); - rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); - g_object_set (rtppayloader, "pt", 96, NULL); + encoder = gst_element_factory_make ("alawenc", NULL); + rtppayloader = gst_element_factory_make ("rtppcmapay", NULL); + g_object_set (rtppayloader, "pt", 8, NULL); rtpsession = gst_element_factory_make ("rtpsession", NULL); sink = gst_element_factory_make ("fakesink", "sink"); gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader, @@ -261,7 +261,7 @@ GST_START_TEST (test_master_ssrc_collision) gst_object_unref (bin); /* check results */ - fail_unless_equals_int (nb_ssrc_changes, 7); + fail_unless_equals_int (nb_ssrc_changes, 4); } GST_END_TEST; @@ -325,7 +325,7 @@ rtpsession_sinkpad_probe2 (GstPad * pad, GstPadProbeInfo * info, return ret; } -/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ +/* This test build the pipeline audiotestsrc ! alawenc ! rtppcmapay ! \ * rtprtxsend ! rtpsession ! fakesink * It manually pushs buffer into rtpsession with same ssrc than rtx stream * but different ip so that collision can be detected @@ -355,12 +355,12 @@ GST_START_TEST (test_rtx_ssrc_collision) src = gst_element_factory_make ("audiotestsrc", "src"); g_object_set (src, "num-buffers", 5, NULL); - encoder = gst_element_factory_make ("speexenc", NULL); - rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); - g_object_set (rtppayloader, "pt", 96, NULL); + encoder = gst_element_factory_make ("alawenc", NULL); + rtppayloader = gst_element_factory_make ("rtppcmapay", NULL); + g_object_set (rtppayloader, "pt", 8, NULL); rtprtxsend = gst_element_factory_make ("rtprtxsend", NULL); pt_map = gst_structure_new ("application/x-rtp-pt-map", - "96", G_TYPE_UINT, 99, NULL); + "8", G_TYPE_UINT, 99, NULL); g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL); gst_structure_free (pt_map); rtpsession = gst_element_factory_make ("rtpsession", NULL);
