diff options
| author | Sebastian Dröge <sebastian@centricular.com> | 2015-08-19 13:29:53 +0300 |
|---|---|---|
| committer | Sebastian Dröge <sebastian@centricular.com> | 2015-08-19 13:29:53 +0300 |
| commit | ec0926144f07aef493e31b53231a9c7e75aa03c7 (patch) | |
| tree | 53394ca64ad8cbbf7daa49a4ac1e3c5b6015518c /ChangeLog | |
| parent | 5bb480485b187c0f98aff6ec07358f87e0c30730 (diff) | |
Release 1.5.90
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 1775 |
1 files changed, 1773 insertions, 2 deletions
@@ -1,9 +1,1780 @@ +=== release 1.5.90 === + +2015-08-19 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.5.90 + +2015-08-19 11:29:55 +0300 Sebastian Dröge <sebastian@centricular.com> + + * po/el.po: + * po/zh_CN.po: + po: Update translations + +2015-08-13 17:29:58 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/multifile/gstmultifilesrc.c: + multifilesrc: fix regression with starting from index set via index property + When we haven't started yet, set the start_index when we set the index property, + so that we start at the right index position after the initial seek. The index + property was never really meant to be for writing, but it used to work, so let's + support it for backwards compatibility. + https://bugzilla.gnome.org/show_bug.cgi?id=739472 + +2015-08-18 10:52:11 +0100 Alex Ashley <bugzilla@ashley-family.net> + + * gst/isomp4/qtdemux.c: + qtdemux: fix offset calculation when parsing CENC aux info + Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for + DASH common encryption, however commit + bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master + shortly before the CENC commit caused the calculation of the CENC + aux info offset to be incorrect. + The base_offset was being added if present, but if the base_offset + is relative to the start of the moof, the offset was being added twice. + The correct approach is to calculate the offset from the start of the + moof and use that offset when parsing the CENC aux info. + +2015-08-17 14:28:24 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/flac/gstflacenc.c: + flacenc: actually return true for accept-caps query handling + +2015-08-17 14:07:10 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtp/gstrtpg723pay.c: + * gst/rtp/gstrtpgsmpay.c: + * gst/rtp/gstrtpklvpay.c: + rtp: copy metadata in the (de)payloaders which is missed before + https://bugzilla.gnome.org/show_bug.cgi?id=753706 + +2015-08-16 15:21:51 -0400 Dustin Spicuzza <dustin@virtualroadside.com> + + * configure.ac: + * sys/directsound/gstdirectsoundsink.c: + * sys/directsound/gstdirectsoundsink.h: + directsoundsink: allow specifying audio playback device + https://bugzilla.gnome.org/show_bug.cgi?id=753670 + +2015-08-16 13:51:47 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/flac/gstflacenc.c: + flacenc: remove single entry if from loop + Iterate from the 2nd channel on and create the 1 channel struct + outside to make loop structure simpler and only slightly faster. + +2015-08-16 13:21:41 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/flac/gstflacenc.c: + flacenc: implement proper accept-caps + Should just compare with what can be immediatelly accepted by + the element. flacenc can't renegotiate so if it has a caps already + it should only accept if it is that caps otherwise just use the + template caps + +2015-08-16 13:03:36 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/flac/gstflacenc.c: + flacenc: improve sink pad template caps + Removes the need for custom caps query handling and makes it more + correct from the beginning on the template. It is a bit uglier + to read because there is 1 entry per channel but makes code easier + to maintain. + +2015-08-16 12:41:56 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/y4m/gsty4mencode.c: + y4mencode: fix gst-launch version in documentation + +2015-08-15 22:32:21 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/speex/gstspeexenc.c: + * ext/wavpack/gstwavpackenc.c: + * gst/law/alaw-encode.c: + * gst/law/mulaw-encode.c: + audioencoders: use template subset check for accept-caps + It is faster than doing a query that propagates downstream and + should be enough + Elements: speexenc, wavpackenc, mulawenc, alawenc + +2015-08-15 22:29:41 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/jpeg/gstjpegenc.c: + * ext/libpng/gstpngenc.c: + * ext/vpx/gstvp8enc.c: + * ext/vpx/gstvp9enc.c: + * gst/y4m/gsty4mencode.c: + videoencoders: use template subset check for accept-caps + It is faster than doing a query that propagates downstream and + should be enough + Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc + +2015-08-16 17:21:24 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/audioparsers/gstmpegaudioparse.c: + mpegaudioparse: use new baseparse API to fix tag handling + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-03-17 17:50:37 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst/audioparsers/gstaacparse.c: + * gst/audioparsers/gstac3parse.c: + * gst/audioparsers/gstamrparse.c: + * gst/audioparsers/gstdcaparse.c: + * gst/audioparsers/gstsbcparse.c: + * gst/audioparsers/gstwavpackparse.c: + audioparsers: use new base parse API to fix tag handling + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-16 14:37:53 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: use new baseparse API and fix tag handling + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-16 13:04:02 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/isomp4/qtdemux.c: + qtdemux: Use signed integer type to be able to check for negative subtraction results + CID 1315829 + +2015-08-16 11:50:34 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtpvorbisdepay.c: + rtpvorbisdepay: remove dead code + payload_buffer must be NULL in ignore_reserved. Check will always be false. + Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf + CID #1316476 + +2015-08-15 22:45:53 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/law/alaw-encode.c: + * gst/law/alaw-encode.h: + alawenc: port to AudioEncoder base class + +2015-08-15 09:16:23 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/flac/gstflacdec.c: + * ext/speex/gstspeexdec.c: + * ext/wavpack/gstwavpackdec.c: + * gst/law/alaw-decode.c: + * gst/law/mulaw-decode.c: + audiodecoders: use default pad accept-caps handling + Avoids useless check of downstream caps when handling an + accept-caps query + Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec + +2015-08-15 08:49:57 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/jpeg/gstjpegdec.c: + * ext/libpng/gstpngdec.c: + * ext/vpx/gstvp8dec.c: + * ext/vpx/gstvp9dec.c: + videodecoders: use default pad accept-caps handling + Avoids useless check of downstream caps when handling an + accept-caps query + Elements: jpegdec, pngdec, vp8dec, vp9dec + +2015-08-15 11:31:04 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/law/alaw-decode.c: + alawdec: make error handling a bit nicer + Print the element along with the debug to make it easier to trace + the failures + +2015-08-15 11:04:16 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/law/alaw-decode.c: + * gst/law/alaw-decode.h: + alawdec: port to audiodecoder base class + mulawdec was already ported, alawdec was left behind. + +2015-08-15 10:34:14 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: only look for more samples in moofs in pull-mode + For playback of some fragmented formats with qtdemux it will + try to look for the next moof after finishing one but it is only + possible for pull-mode. For playback of streaming fragmented formats + such as DASH it should just not try to look for another moof but + instead wait for more data. + https://bugzilla.gnome.org/show_bug.cgi?id=752602 + https://bugzilla.gnome.org/show_bug.cgi?id=752603 + +2015-08-15 12:58:50 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioparsers/gstdcaparse.c: + dcaparse: Don't look for a second syncword + There are streams out there that consistently contain garbage between + every frame so we never ever find a second consecutive syncword. + See https://bugzilla.gnome.org/show_bug.cgi?id=738237 + +2015-08-15 11:12:05 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/vpx/gstvp8enc.c: + * ext/vpx/gstvp9enc.c: + vp8enc, vp9enc: reset multipass file index when stopping encoder + Fixes multipass encoding when re-using the same element/pipeline + for subsequent encoding runs. + https://bugzilla.gnome.org/show_bug.cgi?id=747728 + +2015-08-15 11:09:42 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/vpx/gstvp9enc.c: + * ext/vpx/gstvp9enc.h: + vp9enc: provide support for multiple pass cache files + Some files may provide different caps insight of one stream. Since + vp9enc support caps reinit, we should support cache reinit too. + If more then file cache file will be created, the naming will be: + cache cache.1 cache.2 ... + Based on patch by: Oleksij Rempel <linux@rempel-privat.de> + https://bugzilla.gnome.org/show_bug.cgi?id=747728 + +2015-08-14 11:41:42 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/aacparse.c: + tests: aacparse: use caps query instead of accept-caps + The accept-caps query just does a shallow check at the current + element while at this test we want it to also look at downstream. + So use caps query there. + https://bugzilla.gnome.org/show_bug.cgi?id=753623 + +2015-08-14 11:40:22 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/audioparsers/gstaacparse.c: + * gst/audioparsers/gstac3parse.c: + * gst/audioparsers/gstamrparse.c: + * gst/audioparsers/gstdcaparse.c: + * gst/audioparsers/gstflacparse.c: + * gst/audioparsers/gstmpegaudioparse.c: + * gst/audioparsers/gstsbcparse.c: + * gst/audioparsers/gstwavpackparse.c: + audioparsers: enable accept-template flag + Do a quick check with the pad template caps as it is enough. Users + should have figured the appropriate full caps on a previous caps query + https://bugzilla.gnome.org/show_bug.cgi?id=753623 + +2015-08-14 15:46:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst/rtsp/gstrtspsrc.c: + * gst/rtsp/gstrtspsrc.h: + rtspsrc: send the User-Agent header + Sometimes it is useful to know this information on the + server side. Other popular implementations (vlc, ffmpeg, ...) + also send this header on every message. + This includes a new "user-agent" property that the user + can set to use a custom User-Agent string. The default + is "GStreamer/<version>" + https://bugzilla.gnome.org/show_bug.cgi?id=750101 + +2015-08-14 15:42:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: wrap gst_rtsp_message_init_request in a local function + This will allow adding common request initialization, like the + user agent string, in just one place. + +2015-08-14 09:36:09 +0530 Prashant Gotarne <ps.gotarne@samsung.com> + + * gst/audiofx/audioecho.c: + audioecho: make sure buffer gets reallocated if max_delay changes + https://bugzilla.gnome.org/show_bug.cgi?id=753490 + +2015-07-09 09:51:26 +0200 Oleksij Rempel <linux@rempel-privat.de> + + * ext/vpx/gstvp8enc.c: + * ext/vpx/gstvp8enc.h: + vp8enc: provide support for multiple pass cache files + Some files may provide different caps insight of one stream. Since vp8enc + support caps reinit, we should support cache reinit too. + If more then file cache file will be created, the naming will be: + cache + cache.1 + cache.2 + ... + https://bugzilla.gnome.org/show_bug.cgi?id=747728 + +2015-04-15 22:51:51 +0200 Ramiro Polla <ramiro.polla@collabora.co.uk> + + * gst/rtp/gstrtpmp4gdepay.c: + rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs + Use constantDuration to calculate the timestamp of non-first AU in the + RTP packet. + If constantDuration is not present in the MIME parameters, its value + must be calculated based on the timing information from two consecutive + RTP packets with AU-Index equal to 0. + https://bugzilla.gnome.org/show_bug.cgi?id=747881 + +2015-08-14 06:43:13 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/soup/gstsouphttpsrc.c: + souphttpsrc: remove unnecessary if, g_free is null safe + +2015-08-14 08:33:56 +0100 Alex Ashley <bugzilla@ashley-family.net> + + * ext/soup/gstsouphttpsrc.c: + * ext/soup/gstsouphttpsrc.h: + souphttpsrc: add property to set HTTP method + To allow souphttpsrc to be use HTTP methods other than GET + (e.g. HEAD), add a "method" property that is a string. If this + property is not set, GET is used. + https://bugzilla.gnome.org/show_bug.cgi?id=752413 + +2015-08-14 11:13:01 +0200 Edward Hervey <bilboed@bilboed.com> + + * tests/check/generic/states.c: + check: Rename states unit test + Makes it easier to differentiate from other modules states unit test + +2015-08-14 09:21:25 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/goom/gstaudiovisualizer.c: + * gst/goom/gstaudiovisualizer.h: + * gst/goom2k1/gstaudiovisualizer.c: + * gst/goom2k1/gstaudiovisualizer.h: + goom: Rename get_type() function of base class to prevent symbol conflicts + This is a problem when statically linking. + +2015-08-13 16:32:55 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset + Otherwise we will just output buffers without timestamps after a reset if no + timestamps are provided by upstream, e.g. when using RTSP over TCP. + https://bugzilla.gnome.org/show_bug.cgi?id=749536 + +2015-08-12 17:16:01 +0530 Ravi Kiran K N <ravi.kiran@samsung.com> + + * gst/matroska/matroska-demux.h: + * gst/matroska/matroska-parse.h: + matroska: Remove unused variable + https://bugzilla.gnome.org/show_bug.cgi?id=753556 + +2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtpL16depay.c: + * gst/rtp/gstrtpL24depay.c: + * gst/rtp/gstrtpac3depay.c: + * gst/rtp/gstrtpac3pay.c: + * gst/rtp/gstrtpamrdepay.c: + * gst/rtp/gstrtpamrpay.c: + * gst/rtp/gstrtpbvdepay.c: + * gst/rtp/gstrtpceltdepay.c: + * gst/rtp/gstrtpceltpay.c: + * gst/rtp/gstrtpdvdepay.c: + * gst/rtp/gstrtpdvpay.c: + * gst/rtp/gstrtpg722depay.c: + * gst/rtp/gstrtpg723pay.c: + * gst/rtp/gstrtpg726depay.c: + * gst/rtp/gstrtpg729depay.c: + * gst/rtp/gstrtpg729pay.c: + * gst/rtp/gstrtpgsmdepay.c: + * gst/rtp/gstrtpgsmpay.c: + * gst/rtp/gstrtpgstdepay.c: + * gst/rtp/gstrtpgstpay.c: + * gst/rtp/gstrtph261depay.c: + * gst/rtp/gstrtph261pay.c: + * gst/rtp/gstrtph263depay.c: + * gst/rtp/gstrtph263pay.c: + * gst/rtp/gstrtph263pdepay.c: + * gst/rtp/gstrtph263ppay.c: + * gst/rtp/gstrtph264depay.c: + * gst/rtp/gstrtph264pay.c: + * gst/rtp/gstrtpilbcdepay.c: + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpj2kpay.c: + * gst/rtp/gstrtpjpegdepay.c: + * gst/rtp/gstrtpjpegpay.c: + * gst/rtp/gstrtpmp1sdepay.c: + * gst/rtp/gstrtpmp2tdepay.c: + * gst/rtp/gstrtpmp2tpay.c: + * gst/rtp/gstrtpmp4adepay.c: + * gst/rtp/gstrtpmp4apay.c: + * gst/rtp/gstrtpmp4gdepay.c: + * gst/rtp/gstrtpmp4gpay.c: + * gst/rtp/gstrtpmp4vdepay.c: + * gst/rtp/gstrtpmp4vpay.c: + * gst/rtp/gstrtpmpadepay.c: + * gst/rtp/gstrtpmpapay.c: + * gst/rtp/gstrtpmpvdepay.c: + * gst/rtp/gstrtpmpvpay.c: + * gst/rtp/gstrtppcmadepay.c: + * gst/rtp/gstrtppcmudepay.c: + * gst/rtp/gstrtpqcelpdepay.c: + * gst/rtp/gstrtpqdmdepay.c: + * gst/rtp/gstrtpsbcdepay.c: + * gst/rtp/gstrtpsbcpay.c: + * gst/rtp/gstrtpsirendepay.c: + * gst/rtp/gstrtpspeexdepay.c: + * gst/rtp/gstrtpspeexpay.c: + * gst/rtp/gstrtpsv3vdepay.c: + * gst/rtp/gstrtptheoradepay.c: + * gst/rtp/gstrtptheorapay.c: + * gst/rtp/gstrtptheorapay.h: + * gst/rtp/gstrtputils.c: + * gst/rtp/gstrtputils.h: + * gst/rtp/gstrtpvorbisdepay.c: + * gst/rtp/gstrtpvorbispay.c: + * gst/rtp/gstrtpvorbispay.h: + * gst/rtp/gstrtpvp8depay.c: + * gst/rtp/gstrtpvp8pay.c: + * gst/rtp/gstrtpvrawdepay.c: + * gst/rtp/gstrtpvrawpay.c: + rtp: Copy metadata in the (de)payloader, but only the relevant ones + The payloader didn't copy anything so far, the depayloader copied every + possible meta. Let's make it consistent and just copy all metas without + tags or with only the video tag. + https://bugzilla.gnome.org/show_bug.cgi?id=751774 + +2015-08-10 18:20:15 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix small typo in comment + +2015-08-10 16:19:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst/goom2k1/gstgoom.c: + goom2k1/doc: Fixup previous commit + +2015-08-10 15:55:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * docs/plugins/gst-plugins-good-plugins-sections.txt: + * gst/goom2k1/gstgoom.c: + * gst/goom2k1/gstgoom.h: + goom2k1/doc: Use GstGoom2k1 namespace + The doc generator isn't happy when we have class name clash. Simply + use it's own namespace. + +2015-08-10 17:10:42 +0530 Prashant Gotarne <ps.gotarne@samsung.com> + + * gst/audiofx/audioecho.c: + audioecho: removed unused variable in set_property + unused local variable 'delay' is removed. + https://bugzilla.gnome.org/show_bug.cgi?id=753450 + +2015-08-10 12:45:27 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix suboptimal queue iteration code + +2015-08-09 17:25:45 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/isomp4/qtdemux.c: + qtdemux: don't use glib 2.44-only API + +2015-07-29 14:14:50 +0100 Alex Ashley <bugzilla@ashley-family.net> + + * gst/isomp4/fourcc.h: + * gst/isomp4/qtdemux.c: + * gst/isomp4/qtdemux.h: + * gst/isomp4/qtdemux_types.c: + qtdemux: add support for ISOBMFF Common Encryption + This commit adds support for ISOBMFF Common Encryption (cenc), as + defined in ISO/IEC 23001-7. It uses a GstProtection event to + pass the contents of PSSH boxes to downstream decryptor elements + and attached GstProtectionMeta to each sample. + https://bugzilla.gnome.org/show_bug.cgi?id=705991 + +2015-08-10 14:13:50 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: checking if depay has sps/pps nals before insertion + https://bugzilla.gnome.org/show_bug.cgi?id=753430 + +2015-08-08 16:44:49 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/matroska/matroska-mux.c: + matroskamux: fix outdated comment + The default behaviour was changed in the 0.10 -> 1.x + transition, but the comment was not updated. + +2015-08-08 17:42:22 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtptheorapay.c: + rtptheorapay: If flushing a packet failed, go out of the loop immediately + +2015-08-08 17:41:02 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpvorbispay.c: + rtpvorbispay: If flushing a packet failed, go out of the loop immediately + +2015-08-08 17:34:50 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtptheorapay.c: + * gst/rtp/gstrtptheorapay.h: + rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps + We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2 + and 4:4:4 formats. + +2015-08-06 17:46:13 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst/rtp/gstrtpklvdepay.c: + * gst/rtp/gstrtpklvpay.c: + rtpklv(de)pay: add "RTP" in the klass string + GstRTSPMedia uses this classification to detect the real payloader + inside a dynpay bin and asserts if it doesn't find it, therefore + it is required + https://bugzilla.gnome.org/show_bug.cgi?id=753325 + +2015-08-05 11:13:09 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/rtpaux.c: + tests: rtpaux: use a dynamic pt in the test + 1) Tests that using dynamic PT instead of the default ones work + 2) If we ever decide to change the codec here we don't need to + worry about change the PT for the default one of the new codec + in the test + https://bugzilla.gnome.org/show_bug.cgi?id=746445 + +2015-08-05 10:53:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtpmanager/gstrtprtxsend.c: + rtprtxsend: print valid type where guint32 is expected + https://bugzilla.gnome.org/show_bug.cgi?id=746445 + +2015-08-06 11:33:37 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtp/gstrtpL16pay.c: + * gst/rtp/gstrtpg722pay.c: + * gst/rtp/gstrtpg723pay.c: + * gst/rtp/gstrtpg729pay.c: + * gst/rtp/gstrtpgsmpay.c: + * gst/rtp/gstrtph261pay.c: + * gst/rtp/gstrtph263pay.c: + * gst/rtp/gstrtpjpegpay.c: + * gst/rtp/gstrtpmp2tpay.c: + * gst/rtp/gstrtpmpapay.c: + * gst/rtp/gstrtpmpvpay.c: + * gst/rtp/gstrtppcmapay.c: + * gst/rtp/gstrtppcmupay.c: + rtppayload: set standard payload type as default + Initialize the PT to the default value of the codec and check if + it is still the default before declaring the pt to be dynamic or + not when setting the caps. + Also use the PT constants from the rtp lib when possible + https://bugzilla.gnome.org/show_bug.cgi?id=747965 + +2015-07-26 12:07:56 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: store the moof-offset also for push mode + It will be used in some cases for getting the correct offsets + from trun atoms. + https://bugzilla.gnome.org/show_bug.cgi?id=752603 + +2015-07-26 02:09:24 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/atoms.h: + * gst/isomp4/qtdemux.c: + * gst/isomp4/qtdemux_types.h: + qtdemux: handle default-base-is-moof flag + Handle the flag from the tfhd that signals the base offset to + start from the moof atom + https://bugzilla.gnome.org/show_bug.cgi?id=752603 + +2015-07-29 18:54:35 -0600 Glen Diener <grd@loganmill.net> + + * gst/matroska/matroska-demux.c: + * gst/matroska/matroska-read-common.c: + * gst/matroska/matroska-read-common.h: + matroskademux: Preserve forward referenced track tags + https://bugzilla.gnome.org/show_bug.cgi?id=752850 + +2015-08-04 18:07:35 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/rtpaux.c: + tests: rtpaux: fix test failure + The RTP PT for alaw is 8. + Less than 50 packets are received in the length of this test so it + would never drop a buffer or would drop only the last buffer and + it would fail sometimes when the received wouldn't receive the + retransmission packet in time. + https://bugzilla.gnome.org/show_bug.cgi?id=746445 + +2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpstreamdepay.c: + rtpstreamdepay: Only allow activation in push mode + We need a proper caps event from upstream with the full RTP caps as we can't + create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g. + a filesrc or any other element that supports pull mode. + https://bugzilla.gnome.org/show_bug.cgi?id=753066 + +2015-08-04 16:28:17 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/soup/gstsouphttpsrc.c: + soup: fix typo in translated string + https://bugzilla.gnome.org/show_bug.cgi?id=753240 + +2015-08-04 12:25:46 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: Put the profile and level into the caps + +2015-08-04 12:09:12 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: Only update the srcpad caps if something else than the codec_data changed + h264parse does the same, let's keep the behaviour consistent. As we now + include the codec_data inside the stream too here, this causes less caps + renegotiation. + +2015-08-04 11:48:27 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id + The spec says: + When a picture parameter set NAL unit with a particular value of + pic_parameter_set_id is received, its content replaces the content of the + previous picture parameter set NAL unit, in decoding order, with the same + value of pic_parameter_set_id (when a previous picture parameter set NAL unit + with the same value of pic_parameter_set_id was present in the bitstream). + +2015-08-03 13:45:59 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/multifile/gstsplitmuxsink.c: + splitmuxsink: remove extra \n at debug message + +2015-08-03 13:42:20 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/multifile/gstsplitmuxsink.c: + splitmuxsink: prevent deadlock when states change too fast + If the GOP is completed, pads have to start gathering for the + next one but it is possible that the the state might go to + COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the + thread has a chance to wake up and proceed, leaving it trapped in + the check_completed_gop loop and deadlocking the other threads + waiting for it to advance. + To solve it, this patch also checks that tha input running time + hasn't changed to prevent this scenario. + +2015-08-03 17:55:01 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: Insert SPS/PPS NALs into the stream + h264parse does the same and this fixes decoding of some streams with 32 SPS + (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but + the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit. + As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere. + This looks like a mistake in the part of the spec about the codec_data. + +2015-07-30 11:29:27 +0900 Eunhae Choi <eunhae1.choi@samsung.com> + + * ext/soup/gstsouphttpsrc.c: + souphttpsrc: handle empty http proxy string + 1) If the system http_proxy environment variable is not set + or set to an empty string, we must not set proxy to avoid + http connection error. + 2) In case of proxy property setting, if user want to clear + the proxy setting, they should be able to set it to NULL or + an empty string again, so this is fixed too. + 3) Check if the proxy string was parsed correctly. + https://bugzilla.gnome.org/show_bug.cgi?id=752866 + +2015-07-29 15:46:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com> + + * ext/dv/gstdvdemux.c: + * ext/dv/gstdvdemux.h: + dvdemux: remove unused variable + Remove unused variable 'framecount' from dvdemux + https://bugzilla.gnome.org/show_bug.cgi?id=753008 + +2015-07-30 15:32:09 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: assertion error due to wrong condition check + In media to caps function, reserved_keys array is being used for variable i, + leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed + changed it to variable j + https://bugzilla.gnome.org/show_bug.cgi?id=753009 + +2015-07-30 15:21:20 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/rtp/gstrtpmp4vdepay.c: + rtpmp4vdepay: rtpbuffer is being unref'ed twice + process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay + the refernce should not be removed here + https://bugzilla.gnome.org/show_bug.cgi?id=753042 + +2015-07-29 11:26:46 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Strip keys from the fmtp that we use internally in our caps + Skip keys from the fmtp, which we already use ourselves for the + caps. Some software is adding random things like clock-rate into + the fmtp, and we would otherwise here set a string-typed clock-rate + in the caps... and thus fail to create valid RTP caps + https://bugzilla.gnome.org/show_bug.cgi?id=753009 + +2015-07-29 19:28:33 +1000 Jan Schmidt <jan@centricular.com> + + * gst/multifile/gstsplitmuxsink.c: + splitmuxsink: Support mpegtsmux as a muxer. + As a fallback, look for a pad template sink_%d on + the muxer when requesting pads, to support mpegtsmux + https://bugzilla.gnome.org/show_bug.cgi?id=752999 + +2015-06-25 01:35:27 +1000 Jan Schmidt <jan@centricular.com> + + * gst/multifile/gstsplitmuxpartreader.c: + * gst/multifile/gstsplitmuxpartreader.h: + splitmuxsrc: Use a separate lock to delay typefind. + Don't hold the main splitmux part lock over + the parent state change function, as it prevents + posting error messages that happen. Since the purpose + is to prevent typefinding from proceeding, use a + separate mutex just for that. + +2015-07-29 13:43:50 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/matroska/matroska-read-common.c: + matroska: fix memory leak + After adding to tag list, key_val is not being free'd + resulting in memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=752992 + +2015-07-27 13:34:14 +0900 Manasa Athreya <manasa.athreya@lge.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc + 'NONE' and 'raw ' fourcc don't always contain U8 audio, it can + be more bits as well, in which case it's just like 'twos'. + https://bugzilla.gnome.org/show_bug.cgi?id=752613 + +2015-07-24 15:10:05 +0200 Dimitrios Katsaros <patcherwork@gmail.com> + + * sys/v4l2/gstv4l2object.c: + * sys/v4l2/gstv4l2src.c: + v4l2: Allow framerate to be large then 100pfs + This limit was arbitrary. We still fixate near 100pfs for compatibility. + https://bugzilla.gnome.org/show_bug.cgi?id=752825 + +2015-07-25 03:25:28 -0400 Olivier Crête <olivier.crete@ocrete.ca> + + * gst/avi/gstavidemux.c: + avidemux: Stop without posting error on flushing + This could just be a normal pipeline shutdown. + +2015-07-23 15:00:08 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also + https://bugzilla.gnome.org/show_bug.cgi?id=752618 + +2015-07-16 18:09:30 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/Makefile.am: + * tests/check/elements/.gitignore: + * tests/check/elements/matroskademux.c: + tests: add minmal matroskademux test for subtitle output + Some of the subtitle chunks will have embedded + NUL-terminators (last three), some don't (first three), + some will have markup, some won't, some will be valid + UTF-8 (all but last), some won't (last stanza). + https://bugzilla.gnome.org/show_bug.cgi?id=752421 + +2015-07-16 18:49:26 +0300 Dimitrios Christidis <dchristidis@mykolab.com> + + * gst/matroska/matroska-demux.c: + matroskademux: fix for subtitle buffers with NUL terminators + Commit 45892ec8 created a regression where g_utf8_validate() would fail + if the subtitle buffer had a NUL terminator as part of the data. + https://bugzilla.gnome.org/show_bug.cgi?id=752421 + +2015-07-21 13:31:05 +0200 Stian Selnes <stian@pexip.com> + + * gst/rtp/gstrtpvp8depay.c: + rtpvp8depay: Check available bytes before copy + Need to check that the number of bytes we want to copy from the adapter + actually is available and handle the error case gracefully. This error + may happen if malformed packets are received and we don't have a + complete frame. + https://bugzilla.gnome.org/show_bug.cgi?id=752663 + +2015-07-16 09:32:36 +0900 Paul Hyunil <paul.hyunil@lge.com> + + * gst/isomp4/fourcc.h: + * gst/isomp4/qtdemux.c: + qtdemux: Support subtitle when track subtype is fourcc_subt + https://bugzilla.gnome.org/show_bug.cgi?id=752655 + +2015-07-20 16:59:40 +0800 Song Bing <b06498@freescale.com> + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2bufferpool: Set timestamp when queue buffer. + Should set timestamp when queue buffer. + https://bugzilla.gnome.org/show_bug.cgi?id=752618 + +2015-07-16 15:12:17 +0200 Havard Graff <havard.graff@gmail.com> + + * gst/rtpmanager/gstrtpmux.c: + * tests/check/elements/rtpmux.c: + rtpmux: handle different ssrc's on sinkpads + Do this by not putting the ssrc from the src pads in the caps used to + probe other sinkpads, and then intersecting with it later. + https://bugzilla.gnome.org/show_bug.cgi?id=752491 + +2015-07-16 17:19:03 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/avi/gstavimux.c: + * gst/matroska/matroska-demux.c: + * gst/matroska/matroska-mux.c: + * gst/matroska/matroska-parse.c: + * gst/matroska/webm-mux.c: + Update mailing list address from sourceforge to freedesktop + +2015-07-15 13:44:52 +0300 Dimitrios Christidis <dchristidis@mykolab.com> + + * gst/matroska/matroska-demux.c: + matroskademux: fix trailing '*' displayed with some text subtitles + The subtitle buffer we push out should not include a NUL terminator + as part of the data, we just add such a terminator for safety, but + it should not be included in the buffer size. + A NUL terminator is not valid UTF-8, so checks will fail if it's + included in the size, and the NUL will be replaced by the fallback + character specified when converting, i.e. '*'. + https://bugzilla.gnome.org/show_bug.cgi?id=752421 + +2015-07-15 18:23:05 +0200 Wim Taymans <wtaymans@redhat.com> + + * ext/pulse/pulsedeviceprovider.c: + * ext/pulse/pulseutil.c: + * ext/pulse/pulseutil.h: + pulse: add properties to GstDevice + Add the extra properties we get from pulse to the GstDevice we expose + with the device monitor + +2015-07-15 17:20:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com> + + * gst/audiofx/audioinvert.c: + * gst/audiofx/audiowsincband.c: + audiofx: Fix typo in example pipelines + Fix typo in example pipelines of audiowsincband and audioinvert. + https://bugzilla.gnome.org/show_bug.cgi?id=752416 + +2015-04-15 18:27:04 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst/multifile/gstsplitmuxsink.c: + splitmuxsink: add a "format-location" signal that allows better control over filenames + In certain applications, splitting into files named after a base + location template and an incremental sequence number is not enough. + This signal gives more fine-grained control to the application to + decide how to name the files. + https://bugzilla.gnome.org/show_bug.cgi?id=750106 + +2015-04-15 20:13:27 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxcoreaudio.c: + osxaudiosrc: no resampling on OS X + Unlike Remote IO, AUHAL doesn't have built-in resampling + for sources -- confirmed by Core Audio engineer Doug Wyatt: + http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html + https://bugzilla.gnome.org/show_bug.cgi?id=743758 + +2015-04-15 18:29:14 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxcoreaudio.c: + osxaudiosrc: avoid get_channel_layout + This only produces a warning and serves no purpose. + https://bugzilla.gnome.org/show_bug.cgi?id=743758 + +2015-04-07 15:40:14 +0530 Arun Raghavan <arun@centricular.com> + + * sys/osxaudio/gstosxcoreaudio.c: + osxaudio: Avoid making a duplicate structure in caps for mono/stereo case + For 1ch or 2ch devices, we just need to set the caps to allow both + options since CoreAudio will up/downmix appropriately. + Also fixes the condition for the 2ch case to be exact, rather than at + least 2 channels since the downmix will not take place in the >stereo + case. + +2015-04-06 16:22:34 +0530 Arun Raghavan <arun@centricular.com> + + * sys/osxaudio/gstosxcoreaudio.c: + * sys/osxaudio/gstosxcoreaudiocommon.c: + * sys/osxaudio/gstosxcoreaudiohal.c: + * sys/osxaudio/gstosxcoreaudioremoteio.c: + osxaudio: Don't set the format on an initialized AudioUnit + We need to initialize the AudioUnit early to be able to probe the + underlying device, but according to the AudioUnitInitialize() and + AudioUnitUninitialize() documentation, format changes should be done + while the AudioUnit is uninitialized. So we explicitly uninitialize the + AudioUnit during a format change and reinitialize it when we're done. + +2015-04-06 15:55:59 +0530 Arun Raghavan <arun@centricular.com> + + * sys/osxaudio/gstosxaudioringbuffer.c: + * sys/osxaudio/gstosxcoreaudio.c: + * sys/osxaudio/gstosxcoreaudio.h: + osxaudio: Minor spelling fix (unitialize -> uninitialize) + +2015-03-21 20:34:25 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxaudiosink.c: + * sys/osxaudio/gstosxaudiosrc.c: + * sys/osxaudio/gstosxcoreaudio.c: + * sys/osxaudio/gstosxcoreaudio.h: + osxaudio: Fix lockup in _audio_unit_property_listener + _audio_unit_property_listener is called either from a Core Audio thread + or as a result of a Core Audio API (e.g. AudioUnitInitialize) + from our own thread. In the latter case, osxbuf can be already locked + (GStreamer's mutex is not recursive). + We introduce the flag cached_caps_valid and use it instead of nullifying + cached_caps when we cannot lock on osxbuf. + https://bugzilla.gnome.org/show_bug.cgi?id=743758 + +2015-03-12 12:15:12 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxcoreaudio.c: + osxaudio: Invalidate cached caps on format change + Listen for changes in hardware stream format and channel layout, and + invalidate cached caps (since they contain the preferred caps). + https://bugzilla.gnome.org/show_bug.cgi?id=743758 + +2015-03-09 23:34:06 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxaudioringbuffer.c: + * sys/osxaudio/gstosxaudiosink.c: + * sys/osxaudio/gstosxaudiosink.h: + * sys/osxaudio/gstosxaudiosrc.c: + * sys/osxaudio/gstosxaudiosrc.h: + * sys/osxaudio/gstosxcoreaudio.c: + * sys/osxaudio/gstosxcoreaudio.h: + * sys/osxaudio/gstosxcoreaudiocommon.c: + * sys/osxaudio/gstosxcoreaudiocommon.h: + * sys/osxaudio/gstosxcoreaudiohal.c: + * sys/osxaudio/gstosxcoreaudioremoteio.c: + osxaudio: Overhaul of probing caps + - Probing caps is unified between source and sink + - Hardware stream format is now reported as preferred capabilities + (dynamically updated when hardware configuration changes) + - Get hardware channel layout from Remote IO just like from HAL + - More comprehensive mapping between AudioChannelLabel and + GstAudioChannelPosition + - Support for unpositioned channel layouts + - Announce stereo-mono upmixing/downmixing in caps + https://bugzilla.gnome.org/show_bug.cgi?id=743758 + +2015-03-09 23:15:56 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com> + + * sys/osxaudio/gstosxcoreaudio.c: + osxaudio: AudioUnitInitialize on open + Call AudioUnitInitialize upon open. Otherwise, we cannot get + (hardware) stream format nor channel layout from the outer scope. + +2015-07-12 14:27:15 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpL16depay.c: + * gst/rtp/gstrtpL24depay.c: + * gst/rtp/gstrtpac3depay.c: + * gst/rtp/gstrtpamrdepay.c: + * gst/rtp/gstrtpbvdepay.c: + * gst/rtp/gstrtpceltdepay.c: + * gst/rtp/gstrtpdvdepay.c: + * gst/rtp/gstrtpg722depay.c: + * gst/rtp/gstrtpg723depay.c: + * gst/rtp/gstrtpg726depay.c: + * gst/rtp/gstrtpg729depay.c: + * gst/rtp/gstrtpgsmdepay.c: + * gst/rtp/gstrtpgstdepay.c: + * gst/rtp/gstrtph261depay.c: + * gst/rtp/gstrtph263depay.c: + * gst/rtp/gstrtph263pdepay.c: + * gst/rtp/gstrtph264depay.c: + * gst/rtp/gstrtpilbcdepay.c: + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpjpegdepay.c: + * gst/rtp/gstrtpklvdepay.c: + * gst/rtp/gstrtpmp1sdepay.c: + * gst/rtp/gstrtpmp2tdepay.c: + * gst/rtp/gstrtpmp4adepay.c: + * gst/rtp/gstrtpmp4gdepay.c: + * gst/rtp/gstrtpmp4vdepay.c: + * gst/rtp/gstrtpmpadepay.c: + * gst/rtp/gstrtpmparobustdepay.c: + * gst/rtp/gstrtpmpvdepay.c: + * gst/rtp/gstrtppcmadepay.c: + * gst/rtp/gstrtppcmudepay.c: + * gst/rtp/gstrtpqcelpdepay.c: + * gst/rtp/gstrtpqdmdepay.c: + * gst/rtp/gstrtpsbcdepay.c: + * gst/rtp/gstrtpsirendepay.c: + * gst/rtp/gstrtpspeexdepay.c: + * gst/rtp/gstrtpsv3vdepay.c: + * gst/rtp/gstrtptheoradepay.c: + * gst/rtp/gstrtpvorbisdepay.c: + * gst/rtp/gstrtpvp8depay.c: + rtp: depayloaders: implement process_rtp_packet() vfunc + For more optimised RTP packet handling: means we don't + need to map the input buffer again but can just re-use + the mapping the base class has already done. + https://bugzilla.gnome.org/show_bug.cgi?id=750235 + +2015-05-27 19:19:27 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpvrawdepay.c: + rtpvrawdepay: implement process_rtp_packet() vfunc + For more optimised RTP packet handling: means we don't + need to map the input buffer again but can just re-use + the map the base class has already done. + https://bugzilla.gnome.org/show_bug.cgi?id=750235 + +2015-07-10 00:13:32 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Fix indention + +2015-07-09 23:59:10 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Always estimate DTS from the current clock time + Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS + we would produce wrong DTS. As now the estimated DTS is based on the clock, + don't store it in the jitterbuffer items as it would otherwise be used in the + skew calculations and would influence the results. We only really need the DTS + for timer calculations. + https://bugzilla.gnome.org/show_bug.cgi?id=749536 + +2015-07-09 09:26:09 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/.gitignore: + gitignore: ignore rtph263 test + +2015-07-08 23:47:44 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/rtpjitterbuffer.c: + rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2 + Replace static constants with macros to make gcc happy + CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o + elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant + static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND; + ^ + elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant + static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000; + ^ + elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant + PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000; + +2015-07-08 23:40:45 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/elements/rtpjitterbuffer.c: + rtpjitterbuffer: run indent and fix some comments + Fix indent on this file and break some comment lines into two to make + it fit 80 chars per line + +2015-07-08 15:02:24 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: rework segment event handling for adaptive streaming + When a new time segment is received upstream is going to restart + with a new atom. Make the neededbytes and todrop variables + reflect that to avoid waiting too much or dropping the + initial bytes that contain the header. + +2015-07-08 12:35:55 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: push data from adapter before starting new segment + The adapter might have data remaining from the previous segment, + push it all before clearing the adapter and starting a new segment. + It can accumulate data if it had pushed and got not-linked, returning + immediately without processing all the data. Before starting a new + segment this data should be handled. + +2015-07-08 19:59:13 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset + https://bugzilla.gnome.org/show_bug.cgi?id=749536 + +2015-07-08 21:08:36 +0200 Havard Graff <havard.graff@gmail.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + * tests/check/elements/rtpjitterbuffer.c: + rtpjitterbuffer: fix gap-time calculation and remove "late" + The amount of time that is completely expired and not worth waiting for, + is the duration of the packets in the gap (gap * duration) - the + latency (size) of the jitterbuffer (priv->latency_ns). This is the duration + that we make a "multi-lost" packet for. + The "late" concept made some sense in 0.10 as it reflected that a buffer + coming in had not been waited for at all, but had a timestamp that was + outside the jitterbuffer to wait for. With the rewrite of the waiting + (timeout) mechanism in 1.0, this no longer makes any sense, and the + variable no longer reflects anything meaningful (num > 0 is useless, + the duration is what matters) + Fixed up the tests that had been slightly modified in 1.0 to allow faulty + behavior to sneak in, and port some of them to use GstHarness. + https://bugzilla.gnome.org/show_bug.cgi?id=738363 + +2015-06-30 11:21:31 +0200 Stian Selnes <stian@pexip.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected" + This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0. + The reverted patch is wrong and introduces a regression because there + may still be time to receive some of the packets included in the gap + if they are reordered. + +2015-07-07 23:53:02 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: flush samples before adding more from moof + Avoids accumulating all samples from a fragmented stream that could + lead to a 'index-too-big' error once it goes over 50MB of data. It + could reach that before 2h of playback so it doesn't take that long. + As upstream elements are providing data in time format they should + be the ones that have more information about the full media index + and should be able to seek if possible. + +2015-07-07 23:56:12 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + * gst/isomp4/qtdemux.h: + qtdemux: rename upstream_newsegment to upstream_format_is_time + upstream_newsegment isn't really clear on what it means, it is set + to TRUE when the upstream element sends a segment in TIME format, so + rename it to be more clear about it. + It is important to know this because it means that upstream has + a notion of time and qtdemux is likely being driven by an upstream + element that is reading from a higher level abstraction than a file, + such as a DASH, MSS or DLNA element. + +2015-07-07 21:31:08 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix leak by flushing previous sample info from trak + In fragmented streaming, multiple moov/moof will be parsed and their + previously stored samples array might leak when new values are parsed. + The parse_trak and callees won't free the previously stored values + before parsing the new ones. + In step-by-step, this is what happens: + 1) initial moov is parsed, traks as well, streams are created. The + trak doesn't contain samples because they are in the moof's trun + boxes. n_samples is set to 0 while parsing the trak and the samples + array is still NULL. + 2) moofs are parsed, and their trun boxes will increase n_samples and + create/extend the samples array + 3) At some point a new moov might be sent (bitrate switching, for example) + and parsing the trak will overwrite n_samples with the values from + this trak. If the n_samples is set to 0 qtdemux will assume that + the samples array is NULL and will leak it when a new one is + created for the subsequent moofs. + This patch makes qtdemux properly free previous sample data before + creating new ones and adds an assert to catch future occurrences of + this issue when the code changes. + +2015-07-07 16:46:33 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix index size check and debug message + It is allocating samples_count + n_samples, not only n_samples + +2015-07-08 17:02:05 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Calculate receive time if we don't have any + This is required to properly schedule packet loss timers and make + sure all our calculations work properly. + https://bugzilla.gnome.org/show_bug.cgi?id=749536 + +2015-07-08 15:13:17 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations + That is, handle DTS==GST_CLOCK_TIME_NONE correctly. + https://bugzilla.gnome.org/show_bug.cgi?id=749536 + +2015-07-08 20:31:42 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * gst/avi/gstavidemux.c: + avidemux: fix event leak + when seek fails in avidemux, event is not being freed. + https://bugzilla.gnome.org/show_bug.cgi?id=752117 + +2015-07-08 12:02:22 +0200 Stian Selnes <stian@pexip.com> + + * gst/rtp/gstrtph263depay.c: + * tests/check/Makefile.am: + * tests/check/elements/rtph263.c: + rtph263depay: Make sure payload is large enough + Plus new unit test. + https://bugzilla.gnome.org/show_bug.cgi?id=752112 + +2015-07-08 08:59:49 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/rtp/gstrtpklvdepay.c: + rtpklvdepay: fix printf format compiler warning + v_len is of type guint64, but while print the value(16 + len_size + v_len) + G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT + https://bugzilla.gnome.org/show_bug.cgi?id=752100 + +2015-07-07 20:25:47 +0100 Tim-Philipp Müller <tim@centricular.com> + + * docs/plugins/gst-plugins-good-plugins-docs.sgml: + * docs/plugins/gst-plugins-good-plugins-sections.txt: + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/inspect/plugin-rtp.xml: + docs: add new RTP elements to docs + +2015-07-07 20:07:31 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/elements/rtp-payloading.c: + tests: rtp-payloading: add basic unit test for KLV payloading + Also make it so that the mtu is always set if specified, not + only in case of the rather weird bufferlist test code path. + This allows us to easily make the payloader fragment a payload + across multiple output packets by setting a small MTU on it. + +2015-07-07 19:58:42 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpklvdepay.c: + * gst/rtp/gstrtpklvdepay.h: + rtpklvdepay: improve start detection and handle fragmented KLV units + +2015-07-05 20:25:10 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtpklvdepay.c: + * gst/rtp/gstrtpklvdepay.h: + rtp: add SMPTE 336M KLV metadata depayloader + http://tools.ietf.org/html/rfc6597 + +2014-08-09 10:08:42 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtpklvpay.c: + * gst/rtp/gstrtpklvpay.h: + rtp: add SMPTE 336M KLV metadata payloader + http://tools.ietf.org/html/rfc6597 + +2015-07-07 16:59:20 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst/isomp4/atoms.c: + * gst/isomp4/atoms.h: + * gst/isomp4/atomsrecovery.c: + * gst/isomp4/properties.h: + * gst/matroska/matroska-mux.c: + * gst/rtpmanager/rtpsource.c: + docs: fix "Symbol name not found at the start of the comment block" + Add symbols or change comment into a regular comment. + +2015-07-07 16:58:53 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst/audioparsers/gstamrparse.h: + docs: remove outdated doc strings + +2015-07-03 23:10:40 +0200 Stefan Sauer <ensonic@users.sf.net> + + * docs/plugins/gst-plugins-good-plugins-docs.sgml: + docs: add missing plugins and ensure master doc is sorted + +2015-07-07 15:54:41 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/imagefreeze/gstimagefreeze.c: + Revert "imagefreeze: Remove impossible error condition" + This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94. + pad only handle EOS events but not EOS flow, and will push the buffer again + resulting in an assertion error. So we should not handle the buffer + and return EOS flow. + +2015-07-07 15:50:50 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpg729depay.c: + rtpg729depay: unmap rtp buffer in error path + +2015-07-07 15:48:40 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpg729pay.c: + rtpg729pay: fix buffer leak + The handle_buffer vfunc takes ownership of the input buffer. + Fixes elements/rtp-payloading under valgrind. + +2015-07-02 08:52:43 +0200 Tobias Mueller <muelli@cryptobitch.de> + + * gst/goom/goom_core.c: + goom: Initialised variables to remove compiler warnings + goom_core.c: In function 'goom_update': + goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized] + goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur); + ^ + goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized] + goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur); + ^ + https://bugzilla.gnome.org/show_bug.cgi?id=752053 + +2015-07-07 09:18:39 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtph261pay.c: + rtph261pay: fix indentation + +2015-07-06 19:11:00 +0900 Jimmy Ohn <yongjin.ohn@lge.com> + + * gst/rtp/gstrtph261pay.c: + rtph261pay: Fix uninitialized variable compiler error + endpos variable does not correctly understand in the + 4.6.3 GCC version. So compile error appears when we do + compile rtph261pay using jhbuild. + This patch is fixed the compile error in 4.6.3 GCC version. + https://bugzilla.gnome.org/show_bug.cgi?id=751985 + +2014-11-12 12:08:58 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv> + + * gst/flv/gstflvdemux.c: + flvdemux: Handle seek flags properly + Allows for non-keyframe seeks. + https://bugzilla.gnome.org/show_bug.cgi?id=738570 + +2015-02-24 10:50:52 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: avoid looping reading the 'moof' atom forever + It gets stuck if it only finds a moof and no mfra/mfro or moov + atoms. Skip the moof to continue the parsing to have it either + play or error out. + https://bugzilla.gnome.org/show_bug.cgi?id=745089 + +2015-06-26 13:24:17 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * ext/flac/gstflacdec.c: + flacdec: improve error handling + for files which have corrupted header, libflac is not able to + process the metadata properly. We just try to ignore the error + and continue with the processing, since metadata parsing is not + making much of a difference to libflac + https://bugzilla.gnome.org/show_bug.cgi?id=751334 + +2015-07-06 20:16:38 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * sys/ximage/ximageutil.c: + ximagesrc: add meta transform function + ximage metadata can't be transformed or copied, but provide an empty + transformation function instead of NULL to allow unconditional calling + of metas' transform functions. + https://bugzilla.gnome.org/show_bug.cgi?id=751778 + +2014-06-16 16:14:28 +0200 Stian Selnes <stian.selnes@gmail.com> + + * gst/rtp/gstrtph263pdepay.c: + rtph263pdepay: init debug category + https://bugzilla.gnome.org/show_bug.cgi?id=752012 + +2014-06-20 10:59:14 +0200 Stian Selnes <stian@pexip.com> + + * gst/rtp/gstrtpvp8depay.c: + rtpv8depay: ignore reserved bit in payload descriptor + Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that: + R: Bit reserved for future use. MUST be set to zero and MUST be + ignored by the receiver. + https://bugzilla.gnome.org/show_bug.cgi?id=751929 + +2015-07-04 20:56:42 +0200 Stian Selnes <stian@pexip.com> + + * docs/plugins/gst-plugins-good-plugins-docs.sgml: + * docs/plugins/gst-plugins-good-plugins-sections.txt: + * gst/rtp/gstrtph261depay.c: + * gst/rtp/gstrtph261pay.c: + rtph261pay: rtph261depay: Add documentation + https://bugzilla.gnome.org/show_bug.cgi?id=751982 + +2015-07-03 21:58:14 +0200 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From f74b2df to 9aed1d7 + +2015-07-03 14:29:16 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph261pay.c: + rtph261pay: Fix compiler warning + gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init': + gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable] + GObjectClass *gobject_class; + +2015-07-03 14:03:05 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph261depay.c: + rtph261depay: Let the base class push the buffer so it can deal with the flow return + +2015-07-03 14:11:35 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph261pay.c: + rtph261pay: Remove unused adapter + +2015-07-03 13:17:24 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpspeexpay.c: + speexpay: Directly attach payload to the output buffer instead of copying it + +2015-07-03 13:07:20 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpsbcpay.c: + sbcpay: Attach payload directly to the output instead of copying + +2014-12-01 14:18:40 +0100 Stian Selnes <stian@pexip.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtph261depay.c: + * gst/rtp/gstrtph261depay.h: + * gst/rtp/gstrtph261pay.c: + * gst/rtp/gstrtph261pay.h: + * tests/check/elements/rtp-payloading.c: + rtp: add H.261 RTP payloader and depayloader + Implementation according to RFC 4587. + Payloader create fragments on MB boundaries in order to match MTU size + the best it can. Some decoders/depayloaders in the wild are very strict + about receiving a continuous bit-stream (e.g. no no-op bits between + frames), so the payloader will shift the compressed bit-stream of a + frame to align with the last significant bit of the previous frame. + Depayloader does not try to be fancy in case of packet loss. It simply + drops all packets for a frame if there is a loss, keeping it simple. + https://bugzilla.gnome.org/show_bug.cgi?id=751886 + +2015-07-03 12:18:52 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpmpvdepay.c: + rtpmpvdepay: Don't forget to unmap the input buffer + +2015-07-03 12:14:47 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpmpvpay.c: + rtpmpvpay: Create buffer lists instead of pushing each buffer individually + +2015-07-03 12:03:59 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpmpapay.c: + rtpmpapay: Use buffer lists instead of pushing each fragment individually + +2015-07-03 10:51:57 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpmp4apay.c: + rtpmp4apay: Create buffer lists and don't copy payload memory + +2015-06-29 16:14:18 +0200 Miguel París Díaz <mparisdiaz@gmail.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT + When there are a lot of small gaps, we can consider that there is + a big gap (too losses) to reset the buffer. + https://bugzilla.gnome.org/show_bug.cgi?id=751636 + +2015-06-29 15:53:52 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + * tests/check/elements/rtpjitterbuffer.c: + rtpjitterbuffer: If possible, always update the current time before looping over all timers + If we have a clock, update "now" now with the very latest running time we have. + If timers are unscheduled below we otherwise wouldn't update now (it's only updated + when timers expire), and also for the very first loop iteration now would otherwise + always be 0. + Also the time is used for the timeout functions, e.g. to calculate any times + for the next timeouts and we would otherwise pass too old times there. + https://bugzilla.gnome.org/show_bug.cgi?id=751636 + +2015-07-02 14:34:57 +0100 Luis de Bethencourt <luis.bg@samsung.com> + + * sys/v4l2/gstv4l2transform.c: + v4l2transform: fix memory leak + tmp needs to be freed before going out of scope in 'done'. + CID #1308954 + +2015-07-02 12:23:45 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph263ppay.c: + rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it + +2015-07-02 09:48:02 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph263pdepay.c: + rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us + +2015-07-02 09:17:59 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph263pay.c: + * gst/rtp/gstrtph263pay.h: + rtph263pay: Stop using an adapter and directly use the buffer + We always pushed one buffer into the adapter, then handled exactly that one + buffer and flushed it from the adapter. Now also don't memcpy() the actual + payload but just attach the input buffer's data to the output buffer. + This code still needs some serious refactoring/rewriting. + +2015-07-01 21:57:28 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpgsmpay.c: + rtpgsmpay: Remove non-existing includes for now + git add -p mistake. + +2015-07-01 19:29:07 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpgstpay.c: + rtpgstpay: Use the return value of gst_buffer_append() + +2015-07-01 19:19:13 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpgsmpay.c: + rtpgsmpay: Attach payload to the output buffer instead of copying it + +2015-07-01 17:58:56 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpg729pay.c: + rtpg729pay: Attach payload directly to output buffers instead of copying + +2015-07-01 17:43:51 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpg723pay.c: + rtpg723pay: Attach payload buffer to the output instead of copying + +2015-07-01 17:30:39 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpdvdepay.c: + rtpdvdepay: Map the output buffer once instead of once every 80 bytes + +2015-07-01 21:46:46 +0900 Jimmy Ohn <yongjin.ohn@lge.com> + + * gst/avi/gstavidemux.c: + avidemux: fix return type of index_entry_offset_search() + It's a compare function and may return a negative value, + so should for correctness and consistency return a signed + integer. + https://bugzilla.gnome.org/show_bug.cgi?id=751780 + +2015-07-01 14:12:57 +0200 Miguel París Díaz <mparisdiaz@gmail.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: refactor handle_next_buffer + The goal of this patch is making handle_next_buffer function + more readable avoiding unnecesary gotos and adding other + cosmetic changes. + +2015-07-01 15:40:25 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpac3pay.c: + rtpac3pay: Attach the payload to the output buffer instead of copying it + Might also want to produce buffer lists here if needed. + +2015-07-01 15:38:47 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpilbcdepay.c: + * gst/rtp/gstrtpsirendepay.c: + rtp: Fix indention + +2015-07-01 12:37:11 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/examples/rtp/Makefile.am: + * tests/examples/rtp/client-VP8-OPUS.sh: + * tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh: + rtp: Add examples with VTS/ATS for VP8/OPUS + Let's have an example with modern codecs. + +2015-06-30 18:11:33 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtph264pay.c: + rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() + +2015-06-30 14:06:20 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpvp8depay.c: + vp8depay: Don't lock/map every non-keyframe buffer twice + Just copy the complete header instead of first looking at the first byte + and then at the remaining 10 bytes. + +2015-06-29 16:05:44 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * sys/v4l2/gstv4l2object.c: + v4l2: document fallthrough cases + Pacify coverity and document fallthrough cases in switch statements. + CID #1308948, #1308947, #1308946 + +2015-06-29 10:36:58 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout" + This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182. + If we have multiple immediate timers, we want to first handle the one with the + lowest sequence number... which would be broken now. + Instead of this we should just use a GSequence for the timers, and have them + sorted first by timestamp, and for equal timestamps by sequence number. Then + we would always only have to take the very first timer from the list and never + have to look at any others. + +2015-06-29 10:14:05 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout + If we have lots of such immediate timeouts, we would otherwise have quadratic + runtime in the number of timeouts. + +2015-06-19 18:01:03 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/multifile/gstsplitmuxsrc.c: + splitmuxsrc: sticky events are sent automatically from the pad + No need to send them explicitly from the element + https://bugzilla.gnome.org/show_bug.cgi?id=751240 + +2015-06-19 18:00:40 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/multifile/gstsplitmuxsrc.c: + splitmuxsrc: make sure to push sticky events before adding pad + It allows the caps to be set on the pad before being added for + dynamic autoplugging to work. + https://bugzilla.gnome.org/show_bug.cgi?id=751240 + +2015-06-26 00:05:29 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtsp/gstrtspsrc.c: + * gst/rtsp/gstrtspsrc.h: + rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property + Enable to use new ntp-time-source property of rtpbin + https://bugzilla.gnome.org/show_bug.cgi?id=751496 + +2015-06-25 23:19:58 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtpmanager/gstrtpbin.c: + * gst/rtpmanager/gstrtpsession.c: + rtpbin/session: fix description + https://bugzilla.gnome.org/show_bug.cgi?id=751496 + +2015-06-25 10:57:25 +0100 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/imagefreeze/gstimagefreeze.c: + * gst/matroska/matroska-demux.c: + * tests/examples/shapewipe/shapewipe-example.c: + docs: decodebin2 -> decodebin + +2015-06-25 10:47:06 +0100 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: update example pipeline + Update reference to decodebin2 to decodebin + +2015-06-25 10:45:35 +0100 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: remove dead assignments + Values in fields_required and same_buffer are overwritten before used. Removing + assignment + +2015-06-25 10:06:07 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/Makefile.am: + * ext/mikmod/Makefile.am: + * ext/mikmod/README: + * ext/mikmod/drv_gst.c: + * ext/mikmod/gstmikmod.c: + * ext/mikmod/gstmikmod.h: + * ext/mikmod/mikmod_reader.c: + * ext/mikmod/mikmod_types.c: + * ext/mikmod/mikmod_types.h: + * m4/Makefile.am: + * m4/libmikmod.m4: + * win32/MANIFEST: + * win32/vs8/libgstmikmod.vcproj: + mikmod: remove ancient unported plugin + This hasn't been touched in 11 years, and + clearly no one's been missing it. + +2015-06-23 20:15:13 +0900 Gilbok Lee <gilbok.lee@samsung.com> + + * gst/isomp4/qtdemux.c: + qtdemux: does not detect orientation + Most files don't contain the values for transposing the coordinates + back to the positive quadrant so qtdemux was ignoring the rotation + tag. To be able to properly handle those files qtdemux will also ignore + the transposing values to only detect the rotation using the values + abde from the transformation matrix: + [a b c] + [d e f] + [g h i] + https://bugzilla.gnome.org/show_bug.cgi?id=738681 + +2015-06-25 00:04:16 +0200 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.5.2 === -2015-06-24 Sebastian Dröge <slomo@coaxion.net> +2015-06-24 23:30:41 +0200 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.5.2 + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-aasink.xml: + * docs/plugins/inspect/plugin-alaw.xml: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-alphacolor.xml: + * docs/plugins/inspect/plugin-apetag.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-audioparsers.xml: + * docs/plugins/inspect/plugin-auparse.xml: + * docs/plugins/inspect/plugin-autodetect.xml: + * docs/plugins/inspect/plugin-avi.xml: + * docs/plugins/inspect/plugin-cacasink.xml: + * docs/plugins/inspect/plugin-cairo.xml: + * docs/plugins/inspect/plugin-cutter.xml: + * docs/plugins/inspect/plugin-debug.xml: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-dtmf.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-effectv.xml: + * docs/plugins/inspect/plugin-equalizer.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-flv.xml: + * docs/plugins/inspect/plugin-flxdec.xml: + * docs/plugins/inspect/plugin-gdkpixbuf.xml: + * docs/plugins/inspect/plugin-goom.xml: + * docs/plugins/inspect/plugin-goom2k1.xml: + * docs/plugins/inspect/plugin-icydemux.xml: + * docs/plugins/inspect/plugin-id3demux.xml: + * docs/plugins/inspect/plugin-imagefreeze.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-isomp4.xml: + * docs/plugins/inspect/plugin-jack.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-level.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-mulaw.xml: + * docs/plugins/inspect/plugin-multifile.xml: + * docs/plugins/inspect/plugin-multipart.xml: + * docs/plugins/inspect/plugin-navigationtest.xml: + * docs/plugins/inspect/plugin-oss4.xml: + * docs/plugins/inspect/plugin-ossaudio.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-pulseaudio.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-rtpmanager.xml: + * docs/plugins/inspect/plugin-rtsp.xml: + * docs/plugins/inspect/plugin-shapewipe.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-smpte.xml: + * docs/plugins/inspect/plugin-soup.xml: + * docs/plugins/inspect/plugin-spectrum.xml: + * docs/plugins/inspect/plugin-speex.xml: + * docs/plugins/inspect/plugin-taglib.xml: + * docs/plugins/inspect/plugin-udp.xml: + * docs/plugins/inspect/plugin-video4linux2.xml: + * docs/plugins/inspect/plugin-videobox.xml: + * docs/plugins/inspect/plugin-videocrop.xml: + * docs/plugins/inspect/plugin-videofilter.xml: + * docs/plugins/inspect/plugin-videomixer.xml: + * docs/plugins/inspect/plugin-vpx.xml: + * docs/plugins/inspect/plugin-wavenc.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + * docs/plugins/inspect/plugin-wavparse.xml: + * docs/plugins/inspect/plugin-ximagesrc.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + * gst-plugins-good.doap: + * win32/common/config.h: + Release 1.5.2 2015-06-24 22:56:12 +0200 Sebastian Dröge <sebastian@centricular.com> |