gst-plugins-good 1.10.3



ChangeLog
=========

2017-01-30  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.10.3

2017-01-30 13:33:23 +0200  Sebastian Dröge <sebastian centricular com>

        * po/el.po:
          po: Update translations

2017-01-27 16:14:16 +0200  Vivia Nikolaidou <vivia toolsonair com>

        * gst/isomp4/atoms.c:
          qtmux: Timecode track fixes for STSD entry
          The n_frames field (frames per second) should follow the nominal frame
          rate for drop-frame timecodes.
          Also, the trak's timescale (and duration, accordingly) should follow the
          STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
          not the other way around.
          https://bugzilla.gnome.org/show_bug.cgi?id=777832

2017-01-19 11:08:11 +0100  Arnaud Vrac <avrac freebox fr>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: retry request on early termination from the server
          Fix a regression introduced by commit 183695c61a54f1 (refactor to use
          Soup's sync API). The code previously attempted to reconnect when the
          server closed the connection early, for example when the stream was put
          in pause for some time.
          Reintroduce this feature by checking if EOS is received before the
          expected content size is downloaded. In this case, do the request
          starting at the previous read position.
          https://bugzilla.gnome.org/show_bug.cgi?id=776720

2017-01-10 09:40:56 -0700  Matt Staples <staples255 gmail com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: find_stream_by_channel should ignore unconfigured streams
          https://bugzilla.gnome.org/show_bug.cgi?id=777101

2017-01-25 18:43:00 +0000  Brendan Shanks <brendan shanks teradek com>

        * gst/isomp4/gstqtmux.c:
          qtmux: Fix debug typo and remove misleading warning
          https://bugzilla.gnome.org/show_bug.cgi?id=777362

2017-01-26 13:54:14 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/autodetect/gstautodetect.c:
          Revert "autodetect: bring the element state down after success"
          This reverts commit 67f6d3358e4620319335065db25edaaba1f5ae0a.
          It causes problems in certain scenarios and needs further investigation
          https://bugzilla.gnome.org/show_bug.cgi?id=764947#c9

2017-01-09 11:32:35 +0530  Rahul Bedarkar <rahul bedarkar imgtec com>

        * gst/wavparse/gstwavparse.c:
          wavparse: check for not NULL before clearing adapter
          In case wavparse receives a manually injected FLUSH_STOP event
          while operating in pull mode we get criticals because we'd try
          to clear a NULL adapter.
          https://bugzilla.gnome.org/show_bug.cgi?id=777123

2017-01-20 17:16:10 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/avi/gstavidemux.c:
          avidemux: Stop reading a ncdt sub-tag if it goes behind the surrounding tag
          https://bugzilla.gnome.org/show_bug.cgi?id=777532

2017-01-20 07:58:26 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/avi/gstavidemux.c:
          avidemux: Fix various out of bounds reads when parsing ncdt tags
          https://bugzilla.gnome.org/show_bug.cgi?id=777500

2017-01-19 13:46:58 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Increment current stts index whenever we finished one stts entry
          Otherwise we could read more chunks than there are available, doing an
          out of bounds read and potentially crash.
          https://bugzilla.gnome.org/show_bug.cgi?id=777469

2017-01-19 13:25:53 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          Revert "qtdemux: Increment current stts index in all code paths after reading one chunk"
          This reverts commit 99d5d7570d0b53dad3bc8eb653b1320ee422aace. It broke
          playback of various valid files.

2017-01-19 07:52:33 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Increment current stts index in all code paths after reading one chunk
          Otherwise we could read more chunks than there are available, doing an
          out of bounds read and potentially crash.
          https://bugzilla.gnome.org/show_bug.cgi?id=777469

2017-01-13 16:40:43 +0100  Arnaud Vrac <avrac freebox fr>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: properly track redirections
          The current code configures libsoup to handle redirections
          transparently, without informing the caller, thus preventing the element
          to record the redirect code and location uri.
          Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing
          libsoup from handling the redirection. When we receive a redirection
          request and libsoup can safely handle it, return a custom error which
          triggers a retry with the new URI.
          https://bugzilla.gnome.org/show_bug.cgi?id=777222

2017-01-13 00:01:06 +1100  Jan Schmidt <jan centricular com>

        * gst/isomp4/gstqtmux.c:
          qtmux: Don't reset request pad numbering across uses
          When reset, don't restart request pad numberings, as
          request pads can survive across state changes. Only
          restart at 0 if all request pads are handed back first.
          https://bugzilla.gnome.org/show_bug.cgi?id=777174

2017-01-11 17:53:32 -0800  Andre McCurdy <armccurdy gmail com>

        * gst/isomp4/qtdemux.c:
          qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data()
          The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
          be freed by the caller after use.
          https://bugzilla.gnome.org/show_bug.cgi?id=777157
          Signed-off-by: Andre McCurdy <armccurdy gmail com>

2017-01-16 15:17:15 +0100  Jean-Christophe Trotin <jean-christophe trotin st com>

        * sys/v4l2/gstv4l2allocator.c:
          v4l2allocator: fix memory type in allocator probe
          The buffer memory type provided to the VIDIOC_CREATE_BUFS ioctl shall
          be set with the value ("memory") given as input parameter of the
          gst_v4l2_allocator_probe() function.
          https://bugzilla.gnome.org/show_bug.cgi?id=777327

2016-11-11 14:31:03 +1100  Matthew Waters <matthew centricular com>

        * gst/autodetect/gstautodetect.c:
          autodetect: bring the element state down after success
          Otherwise some messages that are emitted by the element on NULL->READY
          will not reach the application.
          https://bugzilla.gnome.org/show_bug.cgi?id=764947

2016-04-24 21:38:51 +0900  Seungha Yang <sh yang lge com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix key_time in gst_qtdemux_adjust_seek()
          time in segment should be PTS based (not DTS).
          https://bugzilla.gnome.org/show_bug.cgi?id=765498

2017-01-07 23:55:42 +1100  Jan Schmidt <jan centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Don't reset output timestamps when no tfdt
          If a fragmented stream doesn't have a tfdt, don't
          reset the output timestamps at each fragment boundary
          by erroneously using the default value of 0. Introduced
          by commit 69fc48
          https://bugzilla.gnome.org/show_bug.cgi?id=754230

2016-12-14 21:45:15 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Check if we have enough data available when parsing edit lists
          Also consume the data entry by entry to get complicated indexing out of
          the code.
          https://bugzilla.gnome.org/show_bug.cgi?id=776107

2016-12-14 10:15:10 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Check that the XiTh size is big enough
          https://bugzilla.gnome.org/show_bug.cgi?id=775794

2016-12-09 20:27:53 +0900  Heekyoung Seo <heekyoung seo lge com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Check node length of video sample description
          Add check for node length of video sample description and its fields and
          for the XiTh atom.
          Also unify the code a bit.
          https://bugzilla.gnome.org/show_bug.cgi?id=775794

2016-12-11 13:27:27 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/audiofx/gstscaletempo.c:
          scaletempo: Ensure to reinit buffers whenever they were not allocated yet
          That is, whenever we go through start/stop we have to ensure that on the
          next opportunity the buffers are reallocated again. Otherwise the
          buffers might be NULL because the element was reused with the same
          configuration as before (i.e. set_caps() wouldn't have reinited the
          buffers).
          https://bugzilla.gnome.org/show_bug.cgi?id=775898

2016-12-09 17:55:39 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/flx/gstflxdec.c:
        * gst/flx/gstflxdec.h:
          flxdec: Only send SEGMENT events after CAPS
          I.e., don't just forward the event but delay it if we don't have caps on
          the srcpad yet.

2016-12-09 17:49:40 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/flx/gstflxdec.c:
          flxdec: Unref and unmap buffers in all code paths as needed
          https://bugzilla.gnome.org/show_bug.cgi?id=775888

2016-12-06 07:48:47 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/flx/gstflxdec.c:
          flxdec: Allocate 0-initialized memory for the decoded frame
          Otherwise we might leak arbitrary information from the uninitialized
          memory if not every pixel is written.
          https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html

2016-12-05 07:57:19 -0700  Matt Staples <staples255 gmail com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Fix session cleanup when handling redirect on PLAY
          Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
          removing code from gst_rtspsrc_send that changed the state varable upon
          encountering a redirect. Better to let the redirect handlers in
          gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
          state-dependent cleanup.
          https://bugzilla.gnome.org/show_bug.cgi?id=775543

2016-12-01 17:08:09 +0100  Edward Hervey <bilboed bilboed com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * gst/rtpmanager/rtpjitterbuffer.c:
          jitterbuffer: Don't leak duplicate items
          When providing items with a seqnum, there is a (very small) probability
          that an element with the same seqnum already exists. Don't forget
          to free that item if it wasn't inserted.
          And avoid returning undefined values when dealing with duplicate items

2016-11-03 15:03:59 +0100  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: fix timer-reuse bug
          When doing rtx, the jitterbuffer will always add an rtx-timer for the next
          sequence number.
          In the case of the packet corresponding to that sequence number arriving,
          that same timer will be reused, and simply moved on to wait for the
          following sequence number etc.
          Once an rtx-timer expires (after all retries), it will be rescheduled as
          a lost-timer instead for the same sequence number.
          Now, if this particular sequence-number now arrives (after the timer has
          become a lost-timer), the reuse mechanism *should* now set a new
          rtx-timer for the next sequence number, but the bug is that it does
          not change the timer-type, and hence schedules a lost-timer for that
          following sequence number, with the result that you will have a very
          early lost-event for a packet that might still arrive, and you will
          never be able to send any rtx for this packet.
          Found by Erlend Graff - erlend pexip com
          https://bugzilla.gnome.org/show_bug.cgi?id=773891

2016-10-09 15:59:05 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * gst/rtpmanager/rtpjitterbuffer.c:
        * gst/rtpmanager/rtpjitterbuffer.h:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: fix lost-event using dts instead of pts
          The lost-event was using a different time-domain (dts) than the outgoing
          buffers (pts). Given certain network-conditions these two would become
          sufficiently different and the lost-event contained timestamp/duration
          that was really wrong. As an example GstAudioDecoder could produce
          a stream that jumps back and forth in time after receiving a lost-event.
          The previous behavior calculated the pts (based on the rtptime) inside the
          rtp_jitter_buffer_insert function, but now this functionality has been
          refactored into a new function rtp_jitter_buffer_calculate_pts that is
          called much earlier in the _chain function to make pts available to
          various calculations that wrongly used dts previously
          (like the lost-event).
          There are however two calculations where using dts is the right thing to
          do: calculating the receive-jitter and the rtx-round-trip-time, where the
          arrival time of the buffer from the network is the right metric
          (and is what dts in fact is today).
          The patch also adds two tests regarding B-frames or the
          “rtptime-going-backwards”-scenario, as there were some concerns that this
          patch might break this behavior (which the tests shows it does not).

2016-11-03 16:33:53 +0100  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: fix bug in reschedule_timer
          The new timeout is always going to be (timeout + delay), however, the
          old behavior compared the current timeout to just (timeout), basically
          being (delay) off.
          This would happen if rtx-delay == rtx-retry-timeout, with the result that
          a second rtx attempt for any buffers would be scheduled immediately instead
          of after rtx-delay ms.
          Simply calculate (new_timeout = timeout + delay) and then use that instead.
          https://bugzilla.gnome.org/show_bug.cgi?id=773905

2016-12-01 15:06:06 +0530  Garima Gaur <garima g samsung com>

        * gst/rtp/gstrtph264depay.c:
        * gst/rtp/gstrtpsbcdepay.c:
          rtp: Fix some memory leaks in usage of gst_pad_get_current_caps()
          https://bugzilla.gnome.org/show_bug.cgi?id=775071

2016-12-01 11:23:02 +0100  Edward Hervey <edward centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Sanitize unknown codec caps
          We might have non-printable characters in the unknown fourcc, replace
          them with '_', in the same way we do it for unknown tags.

2016-12-01 20:04:28 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/avi/gstavidemux.c:
          avidemux: Free vprp chunk also if it existed but we made no use of it
          https://bugzilla.gnome.org/show_bug.cgi?id=775479

2016-12-01 17:38:33 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/matroska/matroska-read-common.c:
          matroskademux: Fix memory leak when parsing attachments
          gst_tag_image_data_to_image_sample() does not take ownership of the
          passed memory, so don't set it to NULL to allow us to free it later.
          https://bugzilla.gnome.org/show_bug.cgi?id=775472

2016-12-01 14:56:18 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/matroska/matroska-read-common.c:
          matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux
          Especially, simplify the code a bit.

2016-12-01 14:41:48 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Increase inflate buffer in bigger steps
          1024 bytes is quite small, let's do 4096 bytes (or one page).
          Also remove redundant if, we're always in that case when getting here.

2016-12-01 14:30:49 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Ensure that size of the pasp atom is as much as we need
          https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:27:55 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix zlib inflate loop
          Handle errors cleanly, deallocate all memory and return the actual size
          of the inflated data.
          https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:30:10 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Free compressed moov node and it's corresponding decompressed data
          https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:29:21 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Check size of compressed MOOV header against available data
          And actually read the size of the cmvd atom from the right position.
          https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 13:38:16 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/audioparsers/gstaacparse.c:
          aacparse: Make sure we have enough data in the codec_data to be able to parse it
          Also error out cleanly if mapping the buffer failed.
          https://bugzilla.gnome.org/show_bug.cgi?id=775450

2016-12-01 13:32:22 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix out of bounds read in tag parsing code
          We can't simply assume that the length of the tag value as given
          inside the stream is correct but should also check against the amount of
          data we have actually available.
          https://bugzilla.gnome.org/show_bug.cgi?id=775451

2016-10-26 13:21:29 +0200  Alejandro G. Castro <alex igalia com>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          rtpbin: pipeline gets an EOS when any rtpsources byes
          Instead of sending EOS when a source byes we have to wait for
          all the sources to be gone, which means they already sent BYE and
          were removed from the session. We now handle the EOS in the rtcp
          loop checking the amount of sources in the session.
          https://bugzilla.gnome.org/show_bug.cgi?id=773218

2016-10-24 16:56:31 +0000  Enrique Ocaña González <eocanha igalia com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time 
in the last sample
          We consider there's a sifnificant difference when it's larger than on second
          or than half the duration of the last processed fragment in case the latter is
          larger.
          https://bugzilla.gnome.org/show_bug.cgi?id=754230



Download
========
https://download.gnome.org/sources/gst-plugins-good/1.10/gst-plugins-good-1.10.3.tar.xz (3.27M)
  sha256sum: 4e07e93e34d4b93208f1579c21e7d91a236577b36f128a5332ffee85b4465955



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