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Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 402 |
1 files changed, 400 insertions, 2 deletions
@@ -1,9 +1,407 @@ +=== release 1.3.3 === + +2014-06-22 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.3.3 + +2014-06-22 14:23:32 +0200 Sebastian Dröge <sebastian@centricular.com> + + * po/da.po: + * po/de.po: + * po/hu.po: + * po/id.po: + * po/nl.po: + * po/pl.po: + * po/ru.po: + * po/sr.po: + * po/uk.po: + po: Update translations + +2014-06-20 11:00:14 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * tests/check/libs/audiodecoder.c: + audiodecoder: Don't be too picky about the output frame counter + With most decoder libraries, and especially when accessing codecs via + OpenMAX or similar APIs, we don't have the ability to properly related + the output buffers to a number of input samples. And could e.g. get + a fractional number of input buffers decoded at a time. + Previously this would in the end lead to an error message and stopped + playback. Change it to a warning message instead and try to handle it + gracefully. In theory the subclass can now get timestamp tracking + wrong if it completely misuses the API, but if on average it behaves + correct (and gst-omx and others do) it will continue to work properly. + Also add a test for the new behaviour. + We don't change it in the encoder yet as that requires more internal logic + changes AFAIU and I'm not aware of a case where this was a problem so far. + +2014-06-12 12:36:26 +0200 Michael Olbrich <m.olbrich@pengutronix.de> + + * gst/tcp/gsttcpserversrc.c: + tcpserversrc: close the server socket after accepting a connection + g_socket_accept() is only called once for a server socket. So + keeping the socket open ist just confusing possible clients. + https://bugzilla.gnome.org/show_bug.cgi?id=731566 + +2014-06-13 10:04:47 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/tcp/gsttcpclientsrc.c: + tcpclientsrc: return FLUSHING when select() is canceled + https://bugzilla.gnome.org/show_bug.cgi?id=731567 + +2014-06-12 13:23:29 +0200 Michael Olbrich <m.olbrich@pengutronix.de> + + * gst/tcp/gsttcpserversrc.c: + tcpserversrc: return FLOW_FLUSHING instead of an error when accept/select is canceled + Canceling the accept/select happens when the source is shut down. This is + not an error and the GST_FLOW_ERROR causes problems when only part of the + pipeline is shut down. + https://bugzilla.gnome.org/show_bug.cgi?id=731567 + +2014-06-12 11:55:59 +0200 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/sdp/gstmikey.c: + mikey: Fix Wall to NTP conversion + We are scaling from a unit in microseconds to a unit in ((1 << 32) per seconds). + We therefore scale the microseconds values by: + value of a second in the target unit (1 << 32) + -------------------------------------------------------------- + value of a second in the origin format (1 000 000 microsecond) + +2014-06-06 12:18:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggdemux.c: + oggdemux: allow unset seek stop time in push mode + +2014-06-11 12:50:23 +0100 Tim-Philipp Müller <tim@centricular.com> + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + * docs/plugins/gst-plugins-base-plugins-sections.txt: + docs: add streamsynchronizer to documentation + +2014-06-11 12:43:35 +0100 Tim-Philipp Müller <tim@centricular.com> + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + * docs/plugins/gst-plugins-base-plugins-sections.txt: + docs: add playsink element to documentation + +2014-06-11 10:53:50 +0100 Tim-Philipp Müller <tim@centricular.com> + + * docs/libs/gst-plugins-base-libs-docs.sgml: + docs: add navigation interface to docs + +2014-06-10 12:59:53 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/app/gstappsrc.c: + appsrc: add send_event handler for flushing + Adds a send_event handling for allowing appsrc to flush its internal + data, allowing users to flush the pipeline without setting it to null. + https://bugzilla.gnome.org/show_bug.cgi?id=724231 + +2014-06-09 21:05:00 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst/videoscale/vs_fill_borders.c: + * gst/videoscale/vs_image.h: + videoscale: vs_image: strides are a gsize + The strides that are set from the GstVideoInfo structs are + a gsize. Using an int can cause overflows when dealing with large + enough images + https://bugzilla.gnome.org/show_bug.cgi?id=731195 + +2014-06-09 19:44:56 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/video-info.c: + * tests/check/libs/video.c: + video: avoid overflows when doing int operations for size + size is a gsize, so cast the operands to it to avoid overflows + and setting wrong value to the video size. + Includes tests. + https://bugzilla.gnome.org/show_bug.cgi?id=731195 + +2014-06-09 10:53:03 +0200 Edward Hervey <bilboed@bilboed.com> + + * ext/theora/gsttheoraenc.c: + theoraenc: Remove unneeded check + running timestamps are guaranteed to be positive and valid since the + GstVideoEncoder base class will clip incoming buffers + CID #1139797 + +2014-06-09 10:38:53 +0200 Edward Hervey <bilboed@bilboed.com> + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: add missing va_end in variadic function + Coverity 1139944 + +2014-06-06 10:35:31 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * tests/check/libs/videodecoder.c: + tests: fix uninitialized variable use in video decoder test + +2014-06-05 15:35:31 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Also catch CODEC_NOT_FOUND errors and delay them until all decodebins are done + +2014-06-04 17:00:34 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Ignore missing-plugin messages unless all decodebins post one + When playing RTSP streams there will be one decodebin per stream. If some of + them fail because of a missing plugin we should not fail completely but play + the supported streams at least. + https://bugzilla.gnome.org/show_bug.cgi?id=730868 + +2014-06-04 14:14:14 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Do async-done on expose errors too + +2014-05-20 12:28:15 +0200 Michael Olbrich <m.olbrich@pengutronix.de> + + * gst-libs/gst/allocators/gstdmabuf.c: + dmabuf: fix checking mmap flags + A simple '&' is not sufficiant. With mmapping_flags == PROT_READ and + prot == PROT_READ|PROT_WRITE the check produces the wrong result. + Change the check to make sure that prot is a subset of mmapping_flags. + https://bugzilla.gnome.org/show_bug.cgi?id=730559 + +2014-06-03 15:16:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/alsa/gstalsasink.c: + alsasink: make gst-ident happy + +2014-06-03 15:10:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/alsa/gstalsasink.c: + alsasink: fix occasional crash intersecting invalid values + When a pipeline using alsasink and push mode upstream fails + to preroll, the following state will be the case: + - A loop upstream will be PAUSED, pushing a first buffer + - alsasink will be READY, pending PAUSED, because async + On error, the pipeline will switch to NULL. alsasink is in + READY, so goes to NULL immediately. It zeroes its cached + caps. Meanwhile, the upstream loop can cause a caps query, + conccurent with the state change. This will use those cached + caps. If the zeroing happens between the NULL test and the + dereferencing, GStreamer will critical down in the GstValue + code. + Since it appears that such a gap between states (PAUSED + and pushing upstream, and NULL downstream) is expected, we + need to protect the read/write access to the cached caps. + This fixes the critical. + See https://bugzilla.gnome.org/show_bug.cgi?id=731121 + +2013-10-14 18:56:55 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * tests/check/libs/videodecoder.c: + videodecoder: Keep still meaningfull pending events on FLUSH_STOP + Only EOS and segment should be deleted in that case. + + Add a testcase + https://bugzilla.gnome.org/show_bug.cgi?id=709868 + +2013-10-14 18:48:08 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * tests/check/libs/audiodecoder.c: + audiodecoder: Keep still meaningfull pending events on FLUSH_STOP + Only EOS and segment should be deleted in that case. + https://bugzilla.gnome.org/show_bug.cgi?id=709868 + +2013-10-14 18:45:10 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst-libs/gst/video/gstvideoencoder.c: + * tests/check/libs/videoencoder.c: + videoencoder: Keep still meaningfull pending events on FLUSH_STOP + Only EOS and segment should be deleted in that case. + https://bugzilla.gnome.org/show_bug.cgi?id=709868 + +2013-10-10 18:50:17 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst/encoding/gststreamsplitter.c: + streamsplitter: Keep still meaningfull pending events on FLUSH_STOP + Only EOS and segment should be deleted in that case. + https://bugzilla.gnome.org/show_bug.cgi?id=709868 + +2013-10-10 18:48:47 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + * tests/check/libs/audioencoder.c: + audioencoder: Keep still meaningfull pending events on FLUSH_STOP + Only EOS and segment should be deleted in that case. + https://bugzilla.gnome.org/show_bug.cgi?id=709868 + +2014-06-02 12:40:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggstream.c: + oggstream: consider all opus packets as "keyframes" + This lets oggdemux determine they are not delta units, and removes + spurious per packet warnings about being unable to determine the + packet's keyframeness. + +2014-05-12 17:13:50 +0200 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/sdp/gstmikey.c: + mikey: Free MikeyPayload in error cases + CID #1212136 + +2014-03-16 14:27:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst/playback/gstdecodebin2.c: + * tests/check/elements/decodebin.c: + decodebin: aggregate buffering messages + Aggregate buffering messages to only post the lower value + to avoid setting pipeline to playing while any multiqueue + is still buffering. + There are 3 scenarios where the entries should be removed from + the list: + 1) When decodebin is set to READY + 2) When an element posts a 100% buffering (already implemented) + 3) When a multiqueue is removed from decodebin. + For item 3 we don't need to handle it because this should only + happen when either 1 is hapenning or when it is playing a + chained file, for which number 2 should have happened for the + previous stream to finish + https://bugzilla.gnome.org/show_bug.cgi?id=726423 + +2014-05-28 10:23:24 +0100 Philip Withnall <philip.withnall@collabora.co.uk> + + * gst-libs/gst/audio/audio-format.c: + audio: Add a missing precondition to gst_audio_format_from_string() + https://bugzilla.gnome.org/show_bug.cgi?id=730874 + +2014-05-26 20:57:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/audiodecoder.c: + * tests/check/libs/videodecoder.c: + tests: videodecoder: audiodecoder: add tests for eos after segment + Tests that pushing a buffer after the segment returns EOS + +2014-05-26 21:24:07 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: actually return the push result in backwards playback + It was always returning _OK regardless of what downstream returned + +2014-05-26 12:44:48 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: return EOS when segment is over + if a buffer is clipped by being completely out of segment, check if this + buffer is after the end of the segment and return EOS upstream + https://bugzilla.gnome.org/show_bug.cgi?id=709224 + +2014-05-26 12:44:38 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: return EOS when segment is over + if a buffer is clipped by being completely out of segment, check if this + buffer is after the end of the segment and return EOS upstream + https://bugzilla.gnome.org/show_bug.cgi?id=709224 + +2014-05-26 11:45:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggdemux.h: + oggdemux: use new gstutils helper GstFlowCombiner + Fixes the handling of GST_FLOW_EOS by using the helper object + from gstutils that does the correct combination of flow returns. + https://bugzilla.gnome.org/show_bug.cgi?id=709224 + +2014-05-23 19:21:35 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: play: use cubic volume factor when adjusting volume + This is more natural and better-suited for a playback application. + +2014-05-21 13:23:24 +0200 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.3.2 === -2014-05-21 Sebastian Dröge <slomo@coaxion.net> +2014-05-21 13:06:34 +0200 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: + * common: * configure.ac: - releasing 1.3.2 + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-alsa.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-ivorbisdec.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-videoconvert.xml: + * docs/plugins/inspect/plugin-videorate.xml: + * docs/plugins/inspect/plugin-videoscale.xml: + * docs/plugins/inspect/plugin-videotestsrc.xml: + * docs/plugins/inspect/plugin-volume.xml: + * docs/plugins/inspect/plugin-vorbis.xml: + * docs/plugins/inspect/plugin-ximagesink.xml: + * docs/plugins/inspect/plugin-xvimagesink.xml: + * gst-plugins-base.doap: + * win32/common/_stdint.h: + * win32/common/config.h: + Release 1.3.2 + +2014-05-21 12:01:15 +0200 Sebastian Dröge <sebastian@centricular.com> + + * po/af.po: + * po/az.po: + * po/bg.po: + * po/ca.po: + * po/cs.po: + * po/da.po: + * po/de.po: + * po/el.po: + * po/en_GB.po: + * po/eo.po: + * po/es.po: + * po/eu.po: + * po/fi.po: + * po/fr.po: + * po/gl.po: + * po/hr.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.po: + * po/nb.po: + * po/nl.po: + * po/or.po: + * po/pl.po: + * po/pt_BR.po: + * po/ro.po: + * po/ru.po: + * po/sk.po: + * po/sl.po: + * po/sq.po: + * po/sr.po: + * po/sv.po: + * po/tr.po: + * po/uk.po: + * po/vi.po: + * po/zh_CN.po: + Update .po files 2014-05-21 10:50:56 +0200 Sebastian Dröge <sebastian@centricular.com> |