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author | Robin H. Johnson <robbat2@gentoo.org> | 2015-08-08 13:49:04 -0700 |
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committer | Robin H. Johnson <robbat2@gentoo.org> | 2015-08-08 17:38:18 -0700 |
commit | 56bd759df1d0c750a065b8c845e93d5dfa6b549d (patch) | |
tree | 3f91093cdb475e565ae857f1c5a7fd339e2d781e /media-libs/gst-plugins-good/files | |
download | gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.tar.gz gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.tar.bz2 gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.zip |
proj/gentoo: Initial commit
This commit represents a new era for Gentoo:
Storing the gentoo-x86 tree in Git, as converted from CVS.
This commit is the start of the NEW history.
Any historical data is intended to be grafted onto this point.
Creation process:
1. Take final CVS checkout snapshot
2. Remove ALL ChangeLog* files
3. Transform all Manifests to thin
4. Remove empty Manifests
5. Convert all stale $Header$/$Id$ CVS keywords to non-expanded Git $Id$
5.1. Do not touch files with -kb/-ko keyword flags.
Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
X-Thanks: Alec Warner <antarus@gentoo.org> - did the GSoC 2006 migration tests
X-Thanks: Robin H. Johnson <robbat2@gentoo.org> - infra guy, herding this project
X-Thanks: Nguyen Thai Ngoc Duy <pclouds@gentoo.org> - Former Gentoo developer, wrote Git features for the migration
X-Thanks: Brian Harring <ferringb@gentoo.org> - wrote much python to improve cvs2svn
X-Thanks: Rich Freeman <rich0@gentoo.org> - validation scripts
X-Thanks: Patrick Lauer <patrick@gentoo.org> - Gentoo dev, running new 2014 work in migration
X-Thanks: Michał Górny <mgorny@gentoo.org> - scripts, QA, nagging
X-Thanks: All of other Gentoo developers - many ideas and lots of paint on the bikeshed
Diffstat (limited to 'media-libs/gst-plugins-good/files')
-rw-r--r-- | media-libs/gst-plugins-good/files/gst-plugins-good-1.4.5-rtp-test-fixes.patch | 98 |
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); |