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leio 14/12/30 21:46:43 |
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Added: gst-plugins-good-1.4.5-rtp-test-fixes.patch |
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Log: |
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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. |
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(Portage version: 2.2.15/cvs/Linux x86_64, unsigned Manifest commit) |
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Revision Changes Path |
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1.1 media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch |
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|
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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 |
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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 |
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Index: gst-plugins-good-1.4.5-rtp-test-fixes.patch |
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=================================================================== |
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Upstream commits d416336 and d67da4c |
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|
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diff --git a/tests/check/elements/rtpaux.c b/tests/check/elements/rtpaux.c |
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index 1f410bf..729604a 100644 |
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--- a/tests/check/elements/rtpaux.c |
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+++ b/tests/check/elements/rtpaux.c |
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@@ -218,8 +218,8 @@ GST_START_TEST (test_simple_rtpbin_aux) |
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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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+ encoder = gst_element_factory_make ("alawenc", "encoder"); |
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+ rtppayloader = gst_element_factory_make ("rtppcmapay", "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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@@ -238,7 +238,7 @@ GST_START_TEST (test_simple_rtpbin_aux) |
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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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+ ("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA,payload=(int)8"); |
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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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@@ -249,8 +249,8 @@ GST_START_TEST (test_simple_rtpbin_aux) |
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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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+ rtpdepayloader = gst_element_factory_make ("rtppcmadepay", "rtpdepayloader"); |
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+ decoder = gst_element_factory_make ("alawdec", "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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diff --git a/tests/check/elements/rtpcollision.c b/tests/check/elements/rtpcollision.c |
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index e9528f9..16f665f 100644 |
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--- a/tests/check/elements/rtpcollision.c |
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+++ b/tests/check/elements/rtpcollision.c |
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@@ -156,7 +156,7 @@ fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer) |
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return GST_FLOW_OK; |
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} |
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|
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-/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ |
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+/* This test build the pipeline audiotestsrc ! alawenc ! rtppcmapay ! \ |
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* rtpsession ! fakesink |
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* It manually pushs buffer into rtpsession with same ssrc but different |
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* ip so that collision can be detected |
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@@ -186,9 +186,9 @@ GST_START_TEST (test_master_ssrc_collision) |
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|
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src = gst_element_factory_make ("audiotestsrc", "src"); |
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g_object_set (src, "num-buffers", 5, NULL); |
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- encoder = gst_element_factory_make ("speexenc", NULL); |
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- rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); |
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- g_object_set (rtppayloader, "pt", 96, NULL); |
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+ encoder = gst_element_factory_make ("alawenc", NULL); |
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+ rtppayloader = gst_element_factory_make ("rtppcmapay", NULL); |
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+ g_object_set (rtppayloader, "pt", 8, NULL); |
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rtpsession = gst_element_factory_make ("rtpsession", NULL); |
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sink = gst_element_factory_make ("fakesink", "sink"); |
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gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader, |
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@@ -261,7 +261,7 @@ GST_START_TEST (test_master_ssrc_collision) |
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gst_object_unref (bin); |
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|
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/* check results */ |
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- fail_unless_equals_int (nb_ssrc_changes, 7); |
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+ fail_unless_equals_int (nb_ssrc_changes, 4); |
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} |
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|
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GST_END_TEST; |
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@@ -325,7 +325,7 @@ rtpsession_sinkpad_probe2 (GstPad * pad, GstPadProbeInfo * info, |
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return ret; |
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} |
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|
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-/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \ |
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+/* This test build the pipeline audiotestsrc ! alawenc ! rtppcmapay ! \ |
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* rtprtxsend ! rtpsession ! fakesink |
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* It manually pushs buffer into rtpsession with same ssrc than rtx stream |
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* but different ip so that collision can be detected |
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@@ -355,12 +355,12 @@ GST_START_TEST (test_rtx_ssrc_collision) |
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|
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src = gst_element_factory_make ("audiotestsrc", "src"); |
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g_object_set (src, "num-buffers", 5, NULL); |
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- encoder = gst_element_factory_make ("speexenc", NULL); |
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- rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL); |
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- g_object_set (rtppayloader, "pt", 96, NULL); |
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+ encoder = gst_element_factory_make ("alawenc", NULL); |
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+ rtppayloader = gst_element_factory_make ("rtppcmapay", NULL); |
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+ g_object_set (rtppayloader, "pt", 8, NULL); |
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rtprtxsend = gst_element_factory_make ("rtprtxsend", NULL); |
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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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+ "8", G_TYPE_UINT, 99, NULL); |
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g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL); |
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gst_structure_free (pt_map); |
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rtpsession = gst_element_factory_make ("rtpsession", NULL); |