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)

1 files changed, 98 insertions, 0 deletions

diff --git a/media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch b/media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch
new file mode 100644
index 000000000000..08f49f396476
--- /dev/null
+++ b/media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch
@@ -0,0 +1,98 @@
+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);