Commit b63560e0 authored by Sebastian Dröge's avatar Sebastian Dröge

Release 1.3.3

parent 2226633c
=== release 1.3.3 ===
2014-06-22 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.3.3
2014-06-22 14:24:24 +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/mt.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/zh_HK.po:
* po/zh_TW.po:
po: Update translations
2014-06-21 01:32:03 +0100 Tim-Philipp Müller <tim@centricular.com>
* ext/pulse/pulsedevicemonitor.c:
* sys/v4l2/gstv4l2devicemonitor.c:
pulse, v4l2: update for device "klass" -> "device-class" rename
2014-06-20 12:21:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstmultiudpsink.c:
multiudpsink: optimisation: avoid unnecessary memory ref/unrefs
We know the buffer will stay valid and we will also not
modify the buffer, we just want to send out the data.
2014-06-19 14:59:48 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstmultiudpsink.h:
multiudpsink: avoid some unnecessary run-time type checks
2014-06-19 16:17:23 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: pass the stream id when asking for crypto params
This way the app can choose different parameters for each stream.
2014-05-20 14:58:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: add support for key length parameters
This patch adds supports for the incoming key management parameters for
encryption and authentication key lengths.
It also adds a new signal request-rtcp-key that allows the user to
provide the crypto parameters and key for the RTCP stream.
https://bugzilla.gnome.org/show_bug.cgi?id=730473
2014-06-19 15:25:01 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtp/gstrtpvp8depay.c:
vp8depay: fix header size checking
Use a different variable name to make it clear that we are calculating
the header size.
Correctly check that we have enough bytes to read the header bits. We
were checking if there were 5 bytes available in the header while we
only needed 3, causing the packet to be discarded as too small.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595
2014-05-20 12:39:31 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-05-06 17:42:14 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).
We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-05-20 13:58:20 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-05-20 13:58:20 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/rtp/gstrtpjpegpay.c:
rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 15:03:25 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/avi/gstavidemux.c:
avidemux: don't leak flow combiner
2014-06-18 14:38:55 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpj2kpay.c:
rtpjp2kpay: pre-allocate buffer-list of the right size
2014-06-18 14:34:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpjpegpay.c:
rtpjpegpay: pre-allocate buffer list of the right size
2014-06-18 14:19:28 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpmp4vpay.c:
rtpmp4vpay: pre-allocate buffer list of the right size
2014-06-18 13:44:31 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvp8pay.c:
rtpvp8pay: allocate bitreader on the stack
2014-06-18 13:29:47 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvp8pay.c:
rtpvp8pay: post error message on bus on error and don't use g_message()
2014-06-18 13:20:44 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvp8pay.c:
rtpvp8pay: couple of minor optimisations
Pre-allocate buffer list of the right size to avoid re-allocs.
Avoid plenty of double runtime cast checks and re-doing the
same calculation over and over again in rtp_vp8_calc_payload_len().
Only call gst_buffer_get_size() once.
2014-06-18 08:10:03 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpgstpay.c:
rtpgstpay: pre-allocate buffer list of the right size
To avoid re-allocs.
2014-06-18 07:52:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtph264pay.c:
rtph264pay: pre-allocate bufferlist of the right size
To avoid unnecessary re-allocs.
2014-06-16 20:15:43 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtph264pay.c:
* tests/check/elements/rtp-payloading.c:
rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-16 15:35:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvrawpay.c:
* gst/rtp/gstrtpvrawpay.h:
rtpvrawpay: make chunks per frame configurable
Bit of a misnomer because it's really chunks per field
and not per frame, but we're going to ignore that for
the time being.
2014-06-16 14:52:16 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvrawpay.c:
* gst/rtp/gstrtpvrawpay.h:
rtpvrawpay: remove unused variables
2014-06-16 14:44:27 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvrawpay.c:
rtpvrawpay: pre-allocate buffer lists of sufficient size
Avoids unnecessary reallocs when appending buffers
to the bufferlist.
2014-06-16 13:51:03 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvrawpay.c:
rtpvrawpay: micro-optimise variable access in inner loop
Store some values that don't change during the execution
of the inner loops locally, so the compiler knows that too.
2014-06-16 13:38:47 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpvrawpay.c:
rtpvrawpay: use buffer lists
Collect buffers to send out in buffer lists instead of
pushing out single buffers one at a time. For HD video
each frame might easily add up to a couple of thousand
packets, multiply that by the frame rate and that's a
lot of push() and sendmsg() calls per second.
A good reason to push out buffers as early as possible is
latency, so we don't accumulate the whole frame in a single
buffer list, but instead push it out in a few chunks, which
is hopefully a reasonable compromise.
2014-06-16 16:40:07 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
udp: improve element descriptions for dynudpsink and multiudpsink
2014-06-16 16:17:16 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
udp: remove suppression of compiler warnings for deprecated GLib API
Not needed any more.
2014-06-17 13:16:27 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
* gst/videobox/gstvideobox.c:
videobox: Fix caps negotiation issue
Make sure that if AYUV is received it will detect that it can produce
both RGB and YUV formats
Signed-off-by: Ravi Kiran K N <ravi.kiran@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=725248
2014-06-16 12:02:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtptheoradepay.c:
rtptheoradepay: fix double frees
Fix double-frees introduced to fix another coverity report.
CID 1223053
2014-06-13 10:12:07 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstdynudpsink.c:
dynudpsink: return FLUSHING when sendto got canceled, not an error
2014-06-13 09:52:03 +0100 Tim-Philipp Müller <tim@centricular.com>
* sys/oss/gstosshelper.c:
oss: simplify probed caps before returning them
Exposes all formats in the first structure if the
rest is the same for all of them.
2014-06-13 09:45:28 +0100 Tim-Philipp Müller <tim@centricular.com>
* sys/oss/gstosshelper.c:
oss: make sure 16-bit formats are before 8-bit formats in probed caps
Probe supported formats in order of desirability rather than in
what order they may happen to be in the formats bitmask. Fixes
accidentally exposure of 8-bit formats in caps before 16-bit formats
(in case where U16 was not supported S8 might be listed before S16).
https://bugzilla.gnome.org/show_bug.cgi?id=706884
2014-06-12 16:36:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Cleanly handle v4l2_allocator_new failure
2014-06-12 11:24:15 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtp/gstrtptheoradepay.c:
rtptheordepay: fix leaks
Coverity 1212163
2014-06-12 11:16:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtp/gstrtpg729pay.c:
rtpg729pay: leak fixes
Coverity 1212159
2014-06-12 11:11:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtp/gstrtph263pay.c:
rtph263pay: fix leak
Coverity 1212157
2014-06-12 10:43:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtp/gstrtph263pay.c:
rtph263pay: fix leaks
Coverity 1212149
2014-06-12 10:31:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtp/gstrtpdvpay.c:
rtpdvpay: catch failures to map buffer
Coverity 1139741
2014-06-11 17:43:42 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/multipart/multipartdemux.c:
multipartdemux: guard against having no MIME type
The code would previously crash trying to insert a NULL string
into a hash table.
It does seem a little broken that indexing is done by MIME type
and not by index though, unless the spec says there cannot be
two parts with the same MIME type.
https://bugzilla.gnome.org/show_bug.cgi?id=659573
2014-06-10 15:42:14 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartdemux.h:
multipartdemux: Send stream-start event
This event was not sent. Send it before caps, this requires the pad to
be parented. This removes warning like: "Got data flow before
stream-start event".
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475
2014-06-10 15:33:33 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: avoid looping indefinitely in broken svq3 files
Abort if an atom with size 0 is read from within the svq3 stsd
atoms
https://bugzilla.gnome.org/show_bug.cgi?id=726512
2014-06-10 10:52:23 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/flac/gstflacdec.c:
flacdec: add const where appropriate
2014-06-09 10:39:20 +0200 Edward Hervey <bilboed@bilboed.com>
* ext/speex/gstspeexenc.c:
speexenc: add missing va_end in variadic function
Coverity 1139944
2014-06-09 10:04:38 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/flv/gstflvdemux.c:
flvdemux: Attempt upstream seek first
If we have an upstream element that can handle the seek (such as
rtmpsrc), try to do that first before attempting it ourself.
2014-06-04 11:34:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/wavparse/gstwavparse.c:
wavparse: do not include codec_data on raw audio caps
If the wav header contains an extended chunk, we want to keep
the codec_data field, but not for raw audio.
This fixes some elements (such as adder) from failing to intersect
raw audio caps which would otherwise be intersectable.
2014-06-05 09:38:29 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/flv/gstflvdemux.c:
flvdemux: Query duration upstream first
Upstream elements (like rtmpsrc) might be able to provide the duration
more accurately than flvdemux. Especially with index-less vod files
2014-05-30 19:37:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Cleanup poll method and retry on EINTR/EAGAIN
https://bugzilla.gnome.org/show_bug.cgi?id=731015
2014-03-06 16:37:51 +0100 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
* gst/flv/gstflvdemux.c:
flvdemux: set RESYNC buffer flag when bridging large PTS gaps
So downstream gets notified when this happens.
https://bugzilla.gnome.org/show_bug.cgi?id=725903
2014-06-03 17:59:32 -0400 Olivier Crête <olivier.crete@collabora.com>
* tests/check/elements/rtprtx.c:
rtprtx: Reset state on each iteration
Otherwise it didn't wait for the test to finish before checking the results.
https://bugzilla.gnome.org/show_bug.cgi?id=728501
2014-05-09 14:22:42 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/matroska/matroska-read-common.c:
matroskademux: don't leak doctype string in error code path
CID 1212145.
2014-05-20 08:20:42 +0200 Edward Hervey <edward@collabora.com>
* ext/vpx/gstvp9enc.c:
vp9enc: Don't dereference NULL checks
CID #1197703
2014-05-20 08:23:06 +0200 Edward Hervey <edward@collabora.com>
* ext/vpx/gstvp8enc.c:
vp8enc: Don't dereference NULL variable
CID #1139838
2014-05-30 14:32:42 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: upstream handles seek if fragmented and on time segment
Otherwise we can reject seeks on local files that contain fragmented-like
atoms like 'mvex'. Also improve a message log
https://bugzilla.gnome.org/show_bug.cgi?id=730722
2014-05-30 16:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtp/gstrtph264depay.c:
h264depay: make sure we call handle_nal for each NAL
Call handle_nal for each NAL in the STAP-A RTP packet. This makes
sure we correctly extract the SPS and PPS.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999
2014-05-07 14:09:06 +0200 Sebastian Dröge <sebastian@centricular.com>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: Add custom sticky event to contain the HTTP request and response headers
This can be useful to e.g. get cookie information downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=729707
2014-05-26 19:47:39 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: remove stream last flow return
GstPad already stores that information
https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:37:46 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: remove last flow return from stream struct
It is already stored on GstPad on core
https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:19:45 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvdemux.h:
flvdemux: Use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code
https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 13:21:25 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-parse.c:
* gst/matroska/matroska-read-common.c:
matroskademux: use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code
https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 13:04:10 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: use GstFlowCombiner
Removes flow return combination code to use the newly added GstFlowCombiner
2014-05-23 17:53:00 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: use GstFlowCombiner
Removes the common code to combining flow returns to let it be
handled by core gstutils' GstFlowCombiner
https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 10:59:55 -0400 Julien Isorce <julien.isorce@collabora.co.uk>
* sys/v4l2/gstv4l2sink.c:
v4l2sink: implement gstvideosink.show_frame instead of gstbasesink.render
It allows to show preroll frame. Especially it allows to update the
frame when seeking in PAUSED state.
https://bugzilla.gnome.org/show_bug.cgi?id=722303
2014-05-26 10:59:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2sink.c:
v4l2sink: Cleanup old pad alloc declaration
2014-05-26 12:34:42 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2sink.c:
v4l2bufferpool: Copy already queued buffer
This is required as during preroll we pass the first buffer twice, hence already
queued. It is also useful, to allow filters replaying a previous rendered buffers.
This will require 1 more buffer in sink if last-sample is enabled, since the last
sample will not be the same as the currently queued buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=722303
2014-05-24 20:20:07 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2allocator.c:
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2bufferpool.h:
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2transform.c:
* sys/v4l2/gstv4l2videodec.c:
* sys/v4l2/v4l2_calls.c:
v4l2bufferpool: Port to bufferpool flush_start/stop method
Port the buffer pool to use the new flush_start/flush_stop virtual
methods added to GstBufferPool.
https://bugzilla.gnome.org/show_bug.cgi?id=727611
2014-05-25 17:40:58 +0100 Tim-Philipp Müller <tim@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/mt.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/zh_HK.po:
* po/zh_TW.po:
po: update
2014-05-25 16:54:18 +0200 Piotr Drąg <piotrdrag@gmail.com>
* po/POTFILES.in:
po: update POTFILES
https://bugzilla.gnome.org/show_bug.cgi?id=726556
2014-05-24 23:51:58 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Don't queue all the buffers before dequeueing first
For output device, we where queuing all the buffers, and then we would
dequeue one. This means we only have 1 buffer for the pipeline, no matter
the size of the queue. Instead, start dequeued when min_latency is reached.
Eventually, this the min_latency should also be affected by control
MIN_BUFFERS_FOR_OUTPUT (use by encoders).
2014-05-24 23:49:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2object.c:
v4l2object: Simply read back the config to update the query
It's easy to get the min/max outdate when hacking decide allocation. In
order to avoid this, simply read back the choosen value from the config.
2014-05-24 23:31:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2bufferpool.h:
* sys/v4l2/gstv4l2src.c:
v4l2: Cleanup and fix calculation of latency
Calculation of num_buffers (the max latency in buffers) was
up-side-down. If we can allcoate, then our maximum latency match
pool maximum number of buffers. Also renamed it to max latency. Finally
introduced a min_latency for clarity.
2014-05-24 20:00:14 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2allocator.c:
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2bufferpool.h:
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2transform.c:
* sys/v4l2/gstv4l2videodec.c:
* sys/v4l2/v4l2_calls.c:
Revert "v4l2bufferpool: Port to bufferpool flush_start/stop method"
This reverts commit 2e0fb42e868fc9f6d98b028def80a3e953527307.
Conflicts:
sys/v4l2/gstv4l2allocator.c
sys/v4l2/gstv4l2bufferpool.c
sys/v4l2/gstv4l2videodec.c
2014-05-24 18:56:32 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2object.c:
v4l2object: Fix configuration of other_pool and importation case
Fix the choice of min/max, don't override the min/max with own pool selected
size, correct other_pool is_active check, start from other_pool config when
configuring the other pool and finally validate the configuration.
2014-05-24 18:45:30 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2object.c:
v4l2object: Use proposed allocator as default
2014-05-24 18:43:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Fix USERPTR map flags
We need to map READ only for output and write only for capture, we where
doing the opposite. This fixing USERPTR with glimagesink
https://bugzilla.gnome.org/show_bug.cgi?id=730698
2014-05-24 11:16:35 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: parse tkhd transformation matrix and add tags if appropriate
Handle the transformation matrix cases where there are only simple rotations
(90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
when recording with mobile devices
https://bugzilla.gnome.org/show_bug.cgi?id=679522
2014-05-23 19:10:21 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Prevent num_queued from going negative
2014-05-23 18:25:49 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: don't stop if loop returned FLUSHING
The decodeing thread returning flushing isn't an error, we should simply
try starting the task again. If it's actually flushing, it will stop again by itself.
2014-05-23 17:54:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Handle early task stop
2014-05-23 17:28:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Handle gst_pad_start_task() failure
2014-05-23 17:19:07 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Add trace for FLUSH_START/STOP handling
2014-05-23 17:18:16 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Fix use of atomic value
2014-05-23 17:01:53 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Improve debugging
No need to use obj->element, the pool now have a significant name. Also don't
warn if flushing.
2014-05-23 17:01:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Fix handle_frame error handling
2014-05-23 15:56:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Add a trace when _start() is called