gst-plugins-good 1.4.4


2014-11-06  Sebastian Dröge <slomo coaxion net>

          releasing 1.4.4

2014-11-01 12:18:02 +0100  Aurélien Zanelli <aurelien zanelli darkosphere fr>

        * ext/vpx/gstvp8utils.h:
          vpx: remove compatibility defines
          We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y,
          VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0.

2014-11-01 11:59:26 +0000  Tim-Philipp Müller <tim centricular com>

        * gst/rtp/gstrtpmp2tpay.c:
          rtpmp2tpay: fix up template caps so we can output the default pt 33
          Add fixed payload type for mp2t to template caps as well, so
          our output caps match the advertised default pt. Fixes a
          regression from 1.2.
          There's still something wrong with caps negotiation though,
          rtpmp2tpay payload=96 ! fakesink will not output caps with

2014-10-27 11:08:20 +0100  Sebastian Dröge <sebastian centricular com>

        * tests/check/elements/aacparse.c:
          aacparse: Fix unit test now that we always have profile/level in the caps

2014-10-26 11:47:25 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/audioparsers/gstaacparse.c:
          aacparse: Always set profile/level on the caps
          We have the information already, so why not use it?

2014-10-30 15:37:36 -0700  Aleix Conchillo Flaqué <aleix oblong com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: mikey related memory leaks

2014-10-28 21:32:06 +0000  Tim-Philipp Müller <tim centricular com>

        * ext/pulse/pulsedeviceprovider.h:
        * sys/v4l2/gstv4l2deviceprovider.h:
        * sys/v4l2/gstv4l2tuner.h:
          pulse, v4l2: add missing G_END_DECLS in some places

2014-10-22 22:50:54 +0530  Arun Raghavan <arun accosted net>

        * ext/pulse/pulsesink.c:
          pulsesink: Temporarily disable stream status posting
          We need a mechanism in PulseAudio to allow running code outside the
          mainloop lock. Then we'd be able to post to the bus (taking the
          GST_OBJECT_LOCK), without worrying about locking order with the mainloop
          lock, which is the current cause of deadlocks while trying to post the
          stream status messages.

2014-10-07 15:29:33 +0200  Aurélien Zanelli <aurelien zanelli parrot com>

        * sys/v4l2/gstv4l2bufferpool.c:
          v4l2bufferpool: cleanly handle streamon failure for output device
          On streamon failure, the queued buffer is not released from the
          bufferpool class point of view because it is queued to the driver and
          the flush logic is not performed since we are not in streaming state.
          It causes the v4l2 bufferpool to always return that stop method failed
          and to leak v4l2 objects and buffers.
          This commit solve this by performing the flush logic in error case, ie
          flushing the allocator and restoring queued buffer state to non-queued.

2014-10-08 10:31:21 +0200  Aurélien Zanelli <aurelien zanelli parrot com>

        * sys/v4l2/gstv4l2bufferpool.c:
          v4l2bufferpool: implement dispose method
          Unref objects in dispose method rather than in finalize in order to
          prevent circular reference.

2014-10-08 10:35:14 +0200  Aurélien Zanelli <aurelien zanelli parrot com>

        * sys/v4l2/gstv4l2bufferpool.c:
          v4l2bufferpool: check that allocator is non null when stopping pool
          Otherwise, we could dereference NULL allocator when the stop method is
          called by the GstBufferPool's finalize method.

2014-10-09 12:15:05 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/v4l2/gstv4l2sink.c:
          v4l2sink: Implement unlock/unlock_stop
          This will prevent deadlocks, but will also properly flush the pool and allocator
          when going to READY state. It should also fix issues reported on mailing list
          when seeking is performed.

2014-10-25 12:36:02 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: fix crash on some 32-bit systems
          Make sure to pass right number of bits to gst_structure_new()
          which is a vararg function.
          Fixes elements/rtpaux unit test on ppc32.

2014-10-24 23:48:30 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/interleave/interleave.c:
          interleave: intersect result with filter caps in caps query
          Fixes crash in audiotestsrc because of an unsupported format
          getting negotiated on big-endian systems with
          audiotestsrc ! interleave ! audioconvert ! wavenc

2014-10-22 15:28:44 +0200  Ananda <ananda latelier23 com>

        * ext/speex/gstspeexdec.c:
        * ext/speex/gstspeexenc.c:
          speex: Fix segfault when resetting the codecs multiple times

2014-10-21 13:10:24 +0200  Wim Taymans <wtaymans redhat com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: make debug line less confusing

2014-10-03 17:28:06 -0700  Aleix Conchillo Flaqué <aleix oblong com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: set full stream caps on internal src TCP pads
          Set the complete stream caps on the TCP internal src pads. Otherwise,
          ptdemux will not properly detect the caps change.

2014-10-17 22:23:27 +0200  Sjoerd Simons <sjoerd luon net>

        * gst/rtpmanager/gstrtpmux.c:
        * tests/check/elements/rtpmux.c:
          rtpmux: Don't set PROXY_CAPS flag on the src pad
          rtpmux behaves like a funnel in that it forwards whatever upstream is
          sending buffers. So setting proxy caps doesn't make sense as the
          upstream don't have to have compatible caps, thus resulting in an empty
          caps set as a result of a caps query. Instead set fixed caps just
          as funnel does.

2014-10-20 11:57:38 +0530  Vineeth T M <vineeth tm samsung com>

        * gst/videobox/gstvideobox.c:
          videobox: critical error when element properties set as max/min
          left, right, top, bottom can be set from range of -2147483648 to 2147483647
          when i launch the videobox element with that values, it gives a critical error
          (gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' 
          This happens because min cannot be equal to max.

2014-10-11 11:18:42 +1100  David Sansome <me davidsansome com>

        * gst/equalizer/gstiirequalizer.c:
          equalizer: Don't call iirequalizer's transform_ip in passthrough mode
          It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.

2014-10-02 14:26:08 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * ext/soup/gstsouphttpclientsink.c:
          souphttpclientsink: Fix lifetime of stream headers and queued buffers
          Stream headers are updated whenever ::set_caps is called, so we can't assume
          they'll be valid before the message body is written out. We *can* assume that
          for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.
          Also, add some debug logging for stream header interactions.

2014-10-02 03:26:22 +0200  Matej Knopp <matej knopp gmail com>

        * gst/audioparsers/gstaacparse.c:
          aacparse: fix memory leak when prepending ADTS headers

2014-10-02 10:10:11 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/wavenc/gstwavenc.c:
          wavenc: Send CAPS event after the pad was activated
          Otherwise the CAPS event will be dropped and we never configure any caps at
          all, leading to weird behaviour in many situations. Especially header
          rewriting is not going to work if a capsfilter is after wavenc.

2014-10-01 23:12:30 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * ext/soup/gstsouphttpclientsink.c:
          souphttpclientsink: Add some more useful debug logging

2014-10-01 23:05:03 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * ext/soup/gstsouphttpclientsink.c:
          souphttpclientsink: Free queued buffers in ::reset
          ::render sets a new callback for writing out new buffers only if there aren't
          already buffers queued for writing with a previously-scheduled callback.
          However, if the previously-scheduled callback is interrupted by a state change
          (either manually or due to an error) and there are still buffers in the queue,
          restarting the pipeline will result in buffers being queued forever, and no
          callbacks will ever be scheduled, and no buffers will be written out.

2014-09-30 11:28:39 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/vpx/gstvp8enc.c:
          vp8enc: finish() and drain() should return a GstFlowReturn

2014-09-30 11:35:12 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/vpx/gstvp8enc.c:
        * ext/vpx/gstvp9enc.c:
          vp8enc/vp9enc: Protect the encoder with a mutex in all situations

2014-09-30 11:31:43 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/vpx/gstvp9enc.c:
          vp9enc: Allow caps renegotiation

2014-03-14 12:59:02 +0100  Jose Antonio Santos Cadenas <santoscadenas gmail com>

        * ext/vpx/gstvp8enc.c:
          vp8enc: Allow caps renegotiation

2014-09-29 22:48:16 +0530  Arun Raghavan <arun accosted net>

        * ext/pulse/pulsesink.c:
        * ext/pulse/pulsesrc.c:
          pulse: Add some documentation about threading and synchronisation
          This gives a quick introduction to how the pulsesink/pulsesrc code
          interacts with the pa_threaded_mainloop that we start up to communicate
          with the server.

2014-09-29 20:18:08 +0530  Arun Raghavan <arun accosted net>

        * ext/pulse/pulsesink.c:
          pulsesink: Make emitting stream status messages synchronous
          The stream status messages are emitted in the PA mainloop thread, which
          means the mainloop lock is taken, followed by the Gst object lock (by
          gst_element_post_message()). In all other locations, the order of
          locking is reversed (this is unavoidable in a bunch of cases where the
          object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
          control to take the mainloop lock).
          The only way to guarantee that the defer callback for stream status
          messages doesn't deadlock is to either stop posting those messages, or
          make sure that the message emission is completed before we proceed to
          any point that might take the object lock before the mainloop lock
          (which is what we do after this patch).

2014-10-10 18:30:07 -0400  Olivier Crête <olivier crete ocrete ca>

        * gst/rtpmanager/rtpsource.c:
        * gst/rtpmanager/rtpsource.h:
          rtpsource: Rename seqnum-base to seqnum-offset in caps
          This was modified back in 1.0 in GstRtpBasePayload

2014-10-10 17:30:24 -0400  Olivier Crête <olivier crete ocrete ca>

        * gst/rtpmanager/gstrtpmux.c:
        * gst/rtpmanager/gstrtpmux.h:
        * tests/check/elements/rtpmux.c:
          rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
          These were renamed in GstRTPBasePayload in 1.0

2014-10-10 18:11:19 -0400  Olivier Crête <olivier crete ocrete ca>

        * gst/dtmf/gstrtpdtmfsrc.c:
        * tests/check/elements/dtmf.c:
          rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
          These were renamed in GstRTPBasePayload in 1.0

======== (2.89M)
  sha256sum: 2df90e99da45211c7b2525ae4ac34830a9e7784bd48c072c406c0cf014bdb277

[Date Prev][Date Next]   [Thread Prev][Thread Next]   [Thread Index] [Date Index] [Author Index]