gst-plugins-base 1.7.2

From: Sebastian Dröge <install-module master gnome org>
To: FTP Releases <ftp-release-list gnome org>
Subject: gst-plugins-base 1.7.2
Date: Fri, 19 Feb 2016 12:17:00 +0000 (UTC)
ChangeLog
=========

2016-02-19  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.7.2

2016-02-19 10:31:05 +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:
          po: Update translations

2016-02-18 14:31:28 +0000  Julien Isorce <j isorce samsung com>

        * pkgconfig/gstreamer-allocators-uninstalled.pc.in:
        * pkgconfig/gstreamer-app-uninstalled.pc.in:
        * pkgconfig/gstreamer-audio-uninstalled.pc.in:
        * pkgconfig/gstreamer-fft-uninstalled.pc.in:
        * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
        * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
        * pkgconfig/gstreamer-riff-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
        * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
        * pkgconfig/gstreamer-tag-uninstalled.pc.in:
        * pkgconfig/gstreamer-video-uninstalled.pc.in:
          uninstalled.pc: add support for non libtool build systems
          Currently the .la path is provided which requires to use libtool as
          mentioned in the GStreamer manual section-helloworld-compilerun.html.
          It is fine as long as the application is built using libtool.
          So currently it is not possible to compile a GStreamer application
          within gst-uninstalled with CMake or other build system different
          than autotools.
          This patch allows to do the following in gst-uninstalled env:
          gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
          gstreamer-video-1.0)
          Previously it required to prepend libtool --mode=link
          https://bugzilla.gnome.org/show_bug.cgi?id=720778

2016-01-22 18:26:01 -0800  Reynaldo H. Verdejo Pinochet <reynaldo osg samsung com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: strengthen check for valid H.263 picture layer
          Avoids some false positives leading to miss identification:
          * Prevent picture start code emulation for the first 2 bytes read
          * Add check for valid "picture coding type" and "PB-frames mode" combination
          Additionally, change name on confusingly named TR var to what
          it is, the layer's PTYPE.
          https://bugzilla.gnome.org/show_bug.cgi?id=693263

2015-11-23 15:06:02 +0900  Vineeth T M <vineeth tm samsung com>

        * gst/playback/gstdecodebin2.c:
          decodebin: return incomplete topology if decode chains' cap could not be obtained
          When getting caps of the decode chain, in get_topology, the caps are being
          checked if fixed or not. But get_topology will be called when the decode is
          chain is being exposed and hence it will always be fixed. Hence removing the
          check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
          get_pad_caps will again call the same api.
          And get_topology can return NULL value if currently shutting down the
          pipeline, which on being passed to create message will result in assertion
          error. Check if topology is valid before using it
          https://bugzilla.gnome.org/show_bug.cgi?id=755918

2016-02-05 10:10:40 +0100  Havard Graff <havard graff gmail com>

        * gst-libs/gst/Makefile.am:
          rtp: build audio library before rtp
          Because audio-enumtypes.h needs to be available for
          gstrtpbaseaudiopayload.c
          https://bugzilla.gnome.org/show_bug.cgi?id=761949

2016-02-15 21:28:33 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstdecodebin2.c:
          decodebin: Fix documentation of the autoplug-query signal

2016-01-26 13:54:46 +0100  Stian Selnes <stian pexip com>

        * gst-libs/gst/video/gstvideoencoder.c:
        * tests/check/libs/videoencoder.c:
          videoencoder: Fix leak when pre_push does not return OK
          https://bugzilla.gnome.org/show_bug.cgi?id=761951

2016-02-11 19:47:04 +0100  Wim Taymans <wtaymans redhat com>

        * gst/audioresample/resample.c:
          resample: avoid overflows
          Avoid overflow in rate calculation. This can cause the resampler to
          start on the wrong phase after a rate change.
          Avoid overflow in cubic fraction calculation. This can cause noise when
          dealing with higher samplerates.

2016-02-11 18:01:40 +0100  Wim Taymans <wtaymans redhat com>

        * gst/audioresample/resample_sse.h:
          resample: fix double interpolation sse code
          We were only reading 2 filter taps and we need to read 4 to do cubic
          interpolation.

2016-02-10 12:48:15 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: make a copy if we can't write in unpack
          If we don't have writable memory, make sure to make a copy of the input
          samples into a temporary (writable) buffer, even if we are dealing with
          a native intermediate format that we don't need to call the unpack
          function for.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655

2016-02-05 19:15:16 -0300  Thiago Santos <thiagoss osg samsung com>

        * tests/check/Makefile.am:
          tests: extend the AM_TESTS_ENVIRONMENT from check.mak
          To get the CK_DEFAULT_TIMEOUT defined for all tests.
          Also replaces a 120 timeout that was set.
          https://bugzilla.gnome.org/show_bug.cgi?id=761472

2016-02-05 18:03:07 -0300  Thiago Santos <thiagoss osg samsung com>

        * autogen.sh:
        * common:
          Automatic update of common submodule
          From 86e4663 to b64f03f

2016-01-21 09:43:35 +0100  Lubosz Sarnecki <lubosz sarnecki collabora co uk>

        * ext/pango/gstbasetextoverlay.c:
        * ext/pango/gstbasetextoverlay.h:
          textoverlay: Expose rendering dimensions as properties.
          In order to detect graphical user input on the
          textoverlay, the resulting rendering properties
          need to be exposed to applications.
          Fixes delayx property declaration.
          https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-01-20 15:37:44 +0100  Lubosz Sarnecki <lubosz sarnecki collabora co uk>

        * ext/pango/gstbasetextoverlay.c:
          textoverlay: Do not limit positioning to video area.
          The current position property is limited to X,Y positions
          in the range of [0, 1]. This patch allows full control
          over the overlay position, including partially outside
          of the video area.
          https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-01-28 13:29:39 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/audiorate/gstaudiorate.c:
          audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
          For unsigned formats, silence is not all bits 0.

2016-01-28 13:21:33 +0100  HoonHee Lee <hoonhee lee lge com>

        * gst-libs/gst/audio/gstaudiodecoder.c:
        * gst-libs/gst/video/gstvideodecoder.c:
          audio/videodecoder: Minor cleanup of last commit
          https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-28 18:06:44 +0900  HoonHee Lee <hoonhee lee lge com>

        * gst-libs/gst/audio/gstaudiodecoder.c:
        * gst-libs/gst/video/gstvideodecoder.c:
          audio/videodecoder: use gst_pad_peer_query_caps to make output caps
          gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
          In that case, use gst_pad_peer_query_caps() with template caps as filter
          to have negotiated output caps properly before forwarding GAP event.
          https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-26 19:23:04 +0100  Thibault Saunier <tsaunier gnome org>

        * gst/encoding/gstencodebin.c:
          encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
          Some encoders can update the stream header through time (for example
          vp8 might do that) but it does not strictly changes the output format.

2016-01-26 14:09:42 +0100  Aurélien Zanelli <aurelien zanelli parrot com>

        * gst-libs/gst/video/video-format.h:
          video-format: fix GstVideoFormatInfo documentation warnings
          Add missing ':' to tile_ws and tile_hs fields documentation to avoid
          bad render of these two fields, mark reserved bytes as private to hide
          field and avoid gtkdoc warning and add parameters description to
          documented macro to avoid gtkdoc warnings.
          https://bugzilla.gnome.org/show_bug.cgi?id=761132

2016-01-26 16:56:57 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-converter.h:
        * win32/common/libgstaudio.def:
          audio-converter: add reset function

2016-01-26 16:36:41 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: handle NULL input
          Allow NULL as input to mean silence samples.

2016-01-26 17:16:52 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: improve _update_config
          Allow NULL config to keep the existing parameters.
          Fix the docs.

2016-01-26 17:14:20 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-converter.h:
          audio-converter: audio-converter: make some optimized functions
          Make optimized functions for generic and passthrough conversion.

2016-01-26 16:34:35 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-quantize.c:
        * gst-libs/gst/audio/audio-quantize.h:
          audio-quantize: add _reset function
          Add a reset function that clears any history.

2016-01-25 17:40:23 +0000  Tim-Philipp Müller <tim centricular com>

        * configure.ac:
        * m4/Makefile.am:
        * m4/freetype2.m4:
        * tests/examples/Makefile.am:
          build: remove nonsensical check for freetype
          The examples need Gtk+, nothing uses freetype directly.

2016-01-25 16:22:17 +0000  Tim-Philipp Müller <tim centricular com>

        * tests/check/elements/libvisual.c:
          tests: libvisual: make run faster
          Reduce resolution, which shouldn't make any difference
          to what's tested here. Makes test finish in less than
          half the time it took before (8s vs. 21s).

2016-01-25 18:30:30 +0530  Arun Raghavan <git arunraghavan net>

        * ext/alsa/gstalsasink.c:
          alsa: Trivial doc update
          alsasink now does more than just raw audio.

2016-01-21 18:30:40 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstdecodebin2.c:
          decodebin: Correctly expose pads from elements that have directly exposable pads
          analyze_new_pad() can return a new decode chain, which might have a new
          GstDecodePad in the end. We should use those two for expose_pad() and not the
          original ones that were passed to analyze_new_pad().
          This fails when having a demuxer element that has raw pads immediately or
          if a decoder with raw caps is after an adaptive demuxer.
          https://bugzilla.gnome.org/show_bug.cgi?id=760949

2016-01-21 16:08:46 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: ensure correct alignment of samples
          Make sure that the data we allocate for our temporary buffers is
          properly aligned.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938

2016-01-21 10:45:40 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/video/video-color.c:
        * gst-libs/gst/video/video-color.h:
          video-color: add Adobe RGB primaries and transfer function

2016-01-20 10:19:34 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/video/video-info.c:
          video-info: enfore RGB matrix for RGB formats
          In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
          RGB formats and warn when the GstVideoInfo colorimetry is wrong.
          In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
          for RGB formats and warn about inconsistent caps.
          See https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-20 10:02:20 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/video/video-converter.c:
          video-converter: ignore matrix for RGB formats
          For RGB formats, the matrix in the colorimetry (conversion from YUV to
          RGB) is irrelevant and we should ignore it and assume the identity
          transform for everything we do.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 23:26:57 +0100  Thibault Saunier <tsaunier gnome org>

        * gst-libs/gst/video/gstvideoencoder.h:
          videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
          It was never actually supported or used
          https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 23:22:35 +0100  Thibault Saunier <tsaunier gnome org>

        * gst-libs/gst/video/gstvideoencoder.c:
          Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
          This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
          It was wrong ref counting wise and we decided to deprecated DROPPED
          return value
          https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-18 11:40:36 +0900  Vineeth TM <vineeth tm samsung com>

        * tests/check/elements/audioconvert.c:
          tests:audioconvert: Fix integer overflow build error
          value of 32768L << 16 and 1L << 31 is 2147483648
          but it exceeds the positive range of int which is 2147483647
          resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
          https://bugzilla.gnome.org/show_bug.cgi?id=760769

2016-01-19 12:39:22 +0530  Arun Raghavan <git arunraghavan net>

        * gst-libs/gst/app/gstappsrc.c:
          appsrc: Minor documentation cleanup

2016-01-14 23:14:27 +0000  Tim-Philipp Müller <tim centricular com>

        * tools/gst-play.c:
          tools: gst-play: allow setting of flags in serialized foo+bar format
          https://bugzilla.gnome.org/show_bug.cgi?id=751901

2015-07-02 17:58:00 +0200  Hugues Fruchet <hugues fruchet st com>

        * tools/gst-play.c:
          tools: gst-play: add command line options for verbose output and playbin flags
          https://bugzilla.gnome.org/show_bug.cgi?id=751901

2016-01-18 15:51:16 +0200  Sebastian Dröge <sebastian centricular com>

        * win32/common/libgstapp.def:
          win32: Update exports

2015-10-15 10:38:16 -0400  Evan Callaway <evan callaway ipconfigure com>

        * gst-libs/gst/app/gstappsink.c:
        * gst-libs/gst/app/gstappsink.h:
          Add WAIT_ON_EOS flag to gstappsink.
          If set, an appsink that receives an EOS will wait until all of its buffers have been processed 
before continuing.
          https://bugzilla.gnome.org/show_bug.cgi?id=756187

2016-01-16 10:17:50 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/audio/gstaudioencoder.c:
          audioencoder: Add note to the documentation about various settings being reset before set_format()
          It's quite unexpected behaviour that various subclass settings are just
          reset before set_format(). Unfortunately changing this now has the risk
          of breaking existing code but we should reconsider this for 2.0.

2016-01-09 04:35:23 +0100  Mathieu Duponchelle <mathieu duponchelle opencreed com>

        * gst/playback/gststreamsynchronizer.c:
          streamsynchronizer: Ignore flushing streams [..]
          [..] when resetting group start time. In GES, we are usually connected
          to the streamsynchronizer on one audio and one video pad.
          When seeking the timeline, both nlecompositions often output their flush_start
          before any of them has output its flush_stop.
          The current code, when receiving the first flush stop was using the
          running time of the start of the second composition, which could
          be pretty much anything, and means nothing at that point.
          This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
          both when setting flushing and when checking it.
          https://bugzilla.gnome.org/show_bug.cgi?id=750013

2016-01-08 18:53:52 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaybin2.c:
          playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
          Otherwise a decoder supporting GL memory will think that all downstream can
          support GL memory because of seeing its own template caps.
          https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-08 18:37:16 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaybin2.c:
          Revert "playbin: only add the template caps when the result is empty"
          This reverts commit 023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
          https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-15 13:35:22 +0000  Thibault Saunier <tsaunier gnome org>

        * gst-libs/gst/video/gstvideoencoder.c:
          videoencoder: Release video frame when ->handle return ERROR or DROPPED
          https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-15 09:50:29 +0100  Edward Hervey <edward centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Properly mark pending blocked pads
          When blocking input pads, we also need to properly set the appropriate
          pending flag.
          Without this, when switching stream types after initial configuration
          (like going from Audio+Video to Audio+Video+Sub) playsink would never
          wait for *all* input streams to be blocked (it would just wait for the
          new input pad (text in this case) to be blocked).
          Since the reconfiguration might introduce unlinking/relinking of elements,
          we need to ensure that *ALL* input streams are blocked.
          Failure to do so would result in having some input streams pushing data
          to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
          (returning GST_FLOW_NOT_LINKED).
          A later optimization could involve only blocking the input pads that
          might be involved in reconfiguration. But better be safe than sorry for
          now :)

2016-01-06 10:12:43 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * tools/gst-device-monitor.c:
          gst-device-monitor: Use g_printerr instead of g_error
          g_error is meant to be used for programmer errors (causes an abort),
          not for expected runtime errors.

2016-01-13 16:32:25 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/playback/gstsubtitleoverlay.c:
          subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
          Subset check verifies also that all required fields are present
          and is mostly commonly used when checking if an element accepts
          a certain caps

2016-01-12 11:31:50 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/playback/gstplaybin2.c:
          playbin: use subset check instead of intersect
          Elements usually require that all fields on their caps are present
          on the fixed caps they receive. Using intersection won't verify it,
          resort to using is_subset() checks.
          https://bugzilla.gnome.org/show_bug.cgi?id=760477

2016-01-12 15:56:36 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-channel-mixer.c:
          audio-channel-mixer: round before truncating
          Round the result before truncating for int channel mixing.

2016-01-12 15:27:16 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: Avoid conversion when possible
          When the input and output formats are the same and in a possible
          intermediate format, avoid unpack and pack.
          Never do passthrough channel mixing.
          Only do dithering and noise shaping in S32 format

2016-01-12 11:43:20 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-channel-mixer.c:
          audio-channel-mixer: add more formats
          Add support for float and int16 mixing
          Remove in-place processing, this simplifies things as we won't be using it.
          Don't do clipping for float audio formats

2016-01-12 11:37:17 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: improve processing loop
          Process as many samples as we can from the input and return the number
          of processed samples from the chain. This simplifies some code.
          Fix the IN_WRITABLE handling, don't overwrite the flags.

2016-01-11 18:24:48 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/playback/gstsubtitleoverlay.c:
          subtitleoverlay: replace accept-caps with caps query
          Those accept caps are actually checking if downstream supports
          some particular caps to check if it need to negotiate a different
          format. Checking only the next element with accept-caps is not enough
          to guarantee that it is supported.
          Using a caps query makes it obtain the supported caps for downstream
          as a whole instead of only the next element.

2016-01-08 21:27:16 +0200  Sebastian Dröge <sebastian centricular com>

        * win32/common/libgstaudio.def:
          audio: Update exported symbols list

2016-01-08 15:05:38 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/videorate/gstvideorate.c:
          videorate: replace accept-caps with a caps query
          accept-caps is only a shallow check, it needs to know
          whether downstream as a whole accepts the framerate

2016-01-08 16:08:47 +0000  Tim-Philipp Müller <tim centricular com>

        * docs/libs/gst-plugins-base-libs-sections.txt:
          docs: fix up for GstAudioChannelMix rename as well

2016-01-08 17:34:50 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-converter.h:
        * gst/audioconvert/gstaudioconvert.c:
          audio-converter: small API tweaks
          Pass flags in _converter_new() so that we can configure ourselves
          differently depending on some options.
          SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'

2016-01-08 17:28:31 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-converter.h:
          audio-converter: prepare API for rate changes
          Use the update function to update the sample rates along with the config
          once we implement resampling.

2016-01-08 17:17:44 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-converter.h:
        * gst/audioconvert/gstaudioconvert.c:
          audio-convert: simplify API
          Simplify the API, we don't need the consumed and produced output
          arguments. The caller needs to use the _get_in_frames/get_out_frames API
          to check how much input is needed and how much output will be produced.

2016-01-08 17:50:21 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/audio/gstaudioutilsprivate.h:
        * gst-libs/gst/video/gstvideoutilsprivate.h:
          audio/video: Use G_GNUC_INTERNAL for internal functions

2016-01-08 16:22:25 +0100  Wim Taymans <wtaymans redhat com>

        * gst-libs/gst/audio/Makefile.am:
        * gst-libs/gst/audio/audio-channel-mix.c:
        * gst-libs/gst/audio/audio-channel-mix.h:
        * gst-libs/gst/audio/audio-channel-mixer.c:
        * gst-libs/gst/audio/audio-channel-mixer.h:
        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio.h:
        * win32/common/libgstaudio.def:
          audio: GstAudioChannelMix -> GstAudioChannelMixer
          Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
          looks better and to avoid a conflict with a library in -bad.

2016-01-07 15:24:25 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaybin2.c:
          playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
          accept-caps is only for one element, caps query is recursive. Fixes playback
          with totem and other situations.
          https://bugzilla.gnome.org/show_bug.cgi?id=760234

2016-01-06 15:49:59 +0100  Aurélien Zanelli <aurelien zanelli parrot com>

        * gst-libs/gst/video/gstvideopool.c:
          videopool: store videoinfo after choosing the biggest buffer size
          Otherwise, pool could be negotiated with a size which will be different
          from the one used in allocation which is the GstVideoInfo.
          https://bugzilla.gnome.org/show_bug.cgi?id=760222

2016-01-06 12:14:39 +0100  Aurélien Zanelli <aurelien zanelli parrot com>

        * gst/videotestsrc/gstvideotestsrc.c:
          videotestsrc: add missing break in set_property switch case
          To avoid future issue when adding new properties.
          https://bugzilla.gnome.org/show_bug.cgi?id=760204

2016-01-06 01:04:31 +0000  Koop Mast <kwm FreeBSD org>

        * tests/check/elements/audioconvert.c:
          tests: audioconvert: fix test compilation with clang
          With clang 3.7.1 on FreeBSD:
          elements/audioconvert.c:650:12: error: shifting a negative signed value is
          undefined [-Werror,-Wshift-negative-value]
          (-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
          ~~~ ^
          https://bugzilla.gnome.org/show_bug.cgi?id=760134

2016-01-06 01:06:10 +0000  Tim-Philipp Müller <tim centricular com>

        * tests/check/libs/audiodecoder.c:
        * tests/check/libs/audioencoder.c:
        * tests/check/libs/rtp.c:
        * tests/check/libs/rtpbasepayload.c:
          tests: fix indentation of various unit tests

2016-01-05 22:52:34 +0000  Tim-Philipp Müller <tim centricular com>

        * docs/libs/gst-plugins-base-libs-docs.sgml:
        * docs/libs/gst-plugins-base-libs-sections.txt:
          docs: add new audio API

2016-01-03 17:21:18 +0000  Tim-Philipp Müller <tim centricular com>

        * gst-libs/gst/sdp/gstmikey.h:
        * gst-libs/gst/video/video-overlay-composition.h:
          docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
          gtk-doc can handle static inline functions just fine these days,
          there's no need for this stuff any more.

2016-01-03 10:33:53 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/riff/riff-ids.h:
          riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
          Apparently this #define is unused.

2016-01-02 23:29:22 +0100  Stefan Sauer <ensonic users sf net>

        * gst-libs/gst/riff/riff-ids.h:
          riff-ids: remove trailing whitespace

2016-01-02 23:27:44 +0100  Stefan Sauer <ensonic users sf net>

        * gst-libs/gst/riff/riff-ids.h:
          riff-ids: fix two swapped ids
          For these fourcc ids the name and value is swapped. This was causing a warning
          when registering the avi ids.

2015-12-31 20:43:28 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/Makefile.am:
          sdp: Also reorder SUBDIRS to try even harder to build the RTP library first

2015-12-31 20:41:38 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/Makefile.am:
          sdp: The SDP library depends on the RTP library now and is not independent anymore
          Fix up the build dependencies.

2015-10-07 18:50:18 +0900  Hyunjun Ko <zzoon ko samsung com>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/sdp/Makefile.am:
        * gst-libs/gst/sdp/gstmikey.c:
        * gst-libs/gst/sdp/gstmikey.h:
        * gst-libs/gst/sdp/gstsdpmessage.c:
        * gst-libs/gst/sdp/gstsdpmessage.h:
        * tests/check/libs/sdp.c:
        * win32/common/libgstsdp.def:
          sdp: add helper fuctions from/to sdp from/to caps
          <gstsdpmessage.h>
          GstCaps*       gst_sdp_media_get_caps_from_media   (const GstSDPMedia *media, gint pt);
          GstSDPResult   gst_sdp_media_set_media_from_caps   (const GstCaps* caps, GstSDPMedia *media);
          gchar *        gst_sdp_make_keymgmt                (const gchar *uri, const gchar *base64);
          GstSDPResult   gst_sdp_message_attributes_to_caps  (GstSDPMessage *msg, GstCaps *caps);
          GstSDPResult   gst_sdp_media_attributes_to_caps    (GstSDPMedia *media, GstCaps *caps);
          <gstmikey.h>
          GstMIKEYMessage * gst_mikey_message_new_from_caps  (GstCaps *caps);
          gchar *           gst_mikey_message_base64_encode  (GstMIKEYMessage* msg);
          https://bugzilla.gnome.org/show_bug.cgi?id=745880

2015-12-29 18:14:54 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/audioconvert/gstaudioconvert.c:
          audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
          In this specific case it wouldn't cause problems as we only ever access the
          first array element, but let's make explicit what is happening here.
          CID 1346530 and 1346529

2015-12-29 17:56:21 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/pbutils/encoding-profile.c:
          encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE

2015-12-29 17:54:44 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/pbutils/encoding-profile.c:
          encoding-profile: Don't use preset_name string after free
          When we run the loop for another time and do not have a preset name, we would
          try to print the preset name of a previous iteration that is already freed.
          Also move some other variables into the block where they are actually used
          to prevent similar mistakes in the future.
          CID 1346536

2015-12-29 14:40:04 +0100  Stefan Sauer <ensonic users sf net>

        * tests/check/elements/audioconvert.c:
          audioconvert: add a test for gap handling

2015-12-29 14:23:59 +0100  Stefan Sauer <ensonic users sf net>

        * gst-libs/gst/audio/audio-converter.c:
        * tests/check/elements/audioconvert.c:
          audioconvert: fix passthrough operation
          We did not take the sample size into account. Rearrange the tests to have more
          conversion test and an extra test case for passthrough operations.
          Fixes #759890

2015-12-29 11:29:31 +0000  Tim-Philipp Müller <tim centricular com>

        * tools/gst-device-monitor.c:
          tools: gst-device-monitor: print uint properties in both decimal and hex
          Some values are easier to read and make sense of in hex.
          https://bugzilla.gnome.org//show_bug.cgi?id=759780

2015-11-12 14:01:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo osg samsung com>

        * gst-libs/gst/video/video-blend.c:
          videoblend: special case 1x1 src dims on increment computation
          Fix crash with 1x1 overlay pixmap
          https://bugzilla.gnome.org/show_bug.cgi?id=757290

2015-12-28 12:28:26 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/typefind/gsttypefindfunctions.c:
          typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
          We would otherwise read beyond the array bounds and crash every now and then.
          This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
          https://bugzilla.gnome.org/show_bug.cgi?id=759910

2015-12-27 19:41:43 +0100  Stefan Sauer <ensonic users sf net>

        * tests/check/elements/audioconvert.c:
          tests: remove commented code from audioconvert test
          This is just what we have in gst_check_buffer_data().

2015-12-27 19:25:20 +0100  Stefan Sauer <ensonic users sf net>

        * gst-libs/gst/audio/audio-converter.c:
          audio-converter: code cleanup
          Rename samples to num_samples, since we also have samples in chain, but that is
          the data pointer. Always use gzize for num_samples. Make the log output a bit
          more homogenous.

2015-12-26 11:34:47 +0000  Tim-Philipp Müller <tim centricular com>

        * tools/gst-device-monitor.c:
          tools: gst-device-monitor: print non-string device properties too

2015-12-26 09:43:56 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/audio/audio-channel-mix.c:
        * gst-libs/gst/audio/audio-converter.c:
        * gst-libs/gst/audio/audio-quantize.c:
          audio: Fix some documentation warnings
          Remove/rename function parameters and skip some functions that can't
          be used by bindings as they are now.

2015-12-26 09:43:51 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
          videoaffinetransformmeta: Add (transfer none) annotation for return value

2015-12-25 11:34:10 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Don't leak audio/video filters due to floating references weirdness
          The filters' floating references are sinked during set_property() already,
          which means that GstBin takes a new reference when adding the filter to it.
          Get rid of the additional reference after adding the filter to the bin.

2015-12-25 10:36:44 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Allow reuse of audio/video filters by unparenting them from their bins
          And also recreate the chains if the filter is changing.

2015-12-25 10:28:02 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Don't leak audio/video filters when using non-raw media

2015-12-24 15:27:43 +0100  Sebastian Dröge <sebastian centricular com>

        * configure.ac:
          Back to development

2015-12-24 13:59:52 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/pbutils/Makefile.am:
          pbutils: Link to libgstbase for bytewriter and adapter



Download
========
https://download.gnome.org/sources/gst-plugins-base/1.7/gst-plugins-base-1.7.2.tar.xz (2.76M)
  sha256sum: 1589ab66e1ed0e8881429d690abccb34a46a6d27cb74a436e9c6bdbd1c85b543
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