2010-02-10 Tim-Philipp Müller <tim.muller@collabora.co.uk>

=== release 0.10.18 ===

2010-02-10 Tim-Philipp Müller <[email protected]>

* configure.ac:
releasing 0.10.18, "Short Circuit"

2010-02-10 20:36:56 +0000 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: temporary safety check to avoid crashes with a certain file
Add temporary check to avoid crashes with a certain file when seeking
until the real cause of this is figured out. See #609405.

2010-02-05 18:05:39 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
qtdemux: skip unknown atoms when looking for moov
Fixes bug #609107

2010-02-05 02:13:33 +0000 Tim-Philipp Müller <[email protected]>

* configure.ac:
* win32/common/config.h:
0.10.17.3 pre-release

2010-02-04 19:10:36 +0000 Tim-Philipp Müller <[email protected]>

* po/bg.po:
* po/hu.po:
po: update translations

2010-02-04 14:46:56 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks

2010-02-04 12:00:03 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Fix time returned for index at a byte offset
The logic for searching forwards/backwards was swapped

2010-02-01 19:22:24 +0100 Mark Nauwelaerts <[email protected]>

* ext/speex/gstspeexdec.c:
speexdec: initialize stereo decoding state

2010-01-28 18:58:08 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: improve stream synchronization
In particular, do not make it send newsegment updates that
sort-of contradict the indented playback segment (e.g. start time).

2010-01-28 18:53:18 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: fix bridging (time) gaps in streams
As a side effect, avoid sending newsegment updates with start times
that go back and forth, which leads to bogus downstream running_time.
Also fixes seeking in bug #606744.

2010-01-28 18:49:57 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: fix stream synchronization
.. by initializing streams starting at 0, as that is basically
where we 'seek to' at the start and assume streams to start elsewhere.
Also enables newsegment update events for subtitle streams.

2010-02-02 13:41:03 +0200 Stefan Kost <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpeg: don't directly access message, some message have args
This caused bogus messages, such as reported in bug #607471.

2010-02-02 00:02:34 +0000 David Hoyt <[email protected]>

* ext/libpng/gstpngdec.c:
png: fix compilation with libpng 1.4
png_set_gray_1_2_4_to_8() has been deprecated for a while and was
finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8()
instead.
Fixes #608629.

2010-02-01 16:46:36 +0100 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: free transports on errors
See #608564

2010-02-01 09:18:53 +0000 Tim-Philipp Müller <[email protected]>

* sys/v4l2/v4l2_calls.c:
v4l2: fix unportable printf format

2010-01-30 15:18:48 +0000 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From 15d47a6 to 96dc793

2010-01-27 17:53:07 +0100 Robert Swain <[email protected]>

* gst/flv/gstflvmux.c:
flvmux: index timestamps should be in seconds, not milliseconds

2010-01-27 15:24:52 +0100 Mark Nauwelaerts <[email protected]>

* ext/speex/gstspeexdec.c:
speexdec: free some more when resetting
Fixes #608255.

2010-01-27 15:24:24 +0100 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpspeexpay.c:
rtpspeexpay: fix occasional buffer leak
Fixes #608255.

2010-01-27 15:22:46 +0100 Mark Nauwelaerts <[email protected]>

* ext/speex/gstspeexenc.c:
speexenc: prevent invalid arithmetic if not setup yet
Fixes #608255.

2010-01-27 16:34:21 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_mmx.h:
videomixer: Fix assembly register constraints
Fixes bug #608209.

2010-01-27 01:56:03 +0000 Tim-Philipp Müller <[email protected]>

* configure.ac:
* win32/common/config.h:
0.10.17.2 pre-release

2010-01-27 01:52:59 +0000 Tim-Philipp Müller <[email protected]>

* po/LINGUAS:
* 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/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.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/ru.po:
* po/sk.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

2010-01-27 01:49:49 +0000 Tim-Philipp Müller <[email protected]>

* tests/check/elements/.gitignore:
checks: ignore deinterlace check binary

2010-01-27 01:18:51 +0000 Tim-Philipp Müller <[email protected]>

* configure.ac:
configure: purge all mention of CVS

2010-01-26 11:18:28 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: ignore streams that finished
When we receive an UNEXPECTED from a stream, move to the next stream and only go
EOS when all streams are EOS. When selecting a stream to push, ignore streams
that went EOS.
Fixes #607949

2010-01-25 17:23:43 +0200 Stefan Kost <[email protected]>

* sys/v4l2/v4l2src_calls.c:
v4l2src: don't deref NULL
Error out when the pool gets shutdown.

2010-01-25 17:21:13 +0200 Stefan Kost <[email protected]>

* ext/jpeg/gstjpegenc.c:
* sys/v4l2/v4l2src_calls.c:
* tests/check/Makefile.am:
Revert "v4l2src: don't deref NULL"
This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba.

2010-01-25 14:16:22 +0200 Stefan Kost <[email protected]>

* ext/jpeg/gstjpegenc.c:
* sys/v4l2/v4l2src_calls.c:
* tests/check/Makefile.am:
v4l2src: don't deref NULL
Error out when the pool gets shutdown.

2010-01-23 15:32:48 -0800 Michael Smith <[email protected]>

* ext/jpeg/gstjpegenc.c:
jpegenc: when creating an overflow buffer, copy timestamps.

2010-01-23 14:47:55 +0100 Edward Hervey <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: dmb1 is a valid fourcc for Motion-JPEG

2010-01-23 14:20:02 +0100 Edward Hervey <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdeux: IV32 is also used for Indeo 3 video streams

2010-01-22 16:48:01 +0200 Stefan Kost <[email protected]>

* tests/icles/ximagesrc-test.c:
build: no unused variables when disabling asserts

2010-01-21 23:17:40 -0300 Roland Krikava <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Avoid negative overflow on keyframe search
Do not overflow negatively when searching a previous
"keyframe" on audio streams. Could cause infinite loops
on backwards playback
Fixes #607718

2010-01-21 17:22:38 -0800 Peter van Hardenberg <[email protected]>

* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstjpegenc.h:
jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes.

2010-01-21 19:24:22 +0100 Alessandro Decina <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix compiler warnings under OS X.

2010-01-21 17:57:36 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: don't parse NULL indexes
for some streams we might fail to fetch the index offsets. Don't try to parse
NULL indexes in those cases.

2010-01-18 21:15:51 -0500 Olivier Crête <[email protected]>

* gst/rtp/gstrtpg729pay.c:
rtpg729pay: ptime should is in nanoseconds
https://bugzilla.gnome.org/show_bug.cgi?id=607403

2010-01-20 15:11:15 -0300 Thiago Santos <[email protected]>

* gst/wavenc/gstwavenc.c:
* gst/wavenc/gstwavenc.h:
wavenc: Post warning if file isnt finished properly
When the pipeline is shut down and the file isn't
finished properly, wavenc should post a warning.
Fixes #607440

2009-05-27 13:51:44 +0200 Arnout Vandecappelle <[email protected]>

* gst/matroska/matroska-mux.c:
* gst/matroska/matroska-mux.h:
matroskamux: make index size configurable.
Added the 'min-index-interval' property to matroskamux,
which determines how much time (nanoseconds) is left
between keyframes stored in the index.
Fixes #583985.

2010-01-20 16:28:31 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: scale spspps_interval to milliseconds
The spspps_interval is kept in seconds. Convert it to milliseconds before
comparing it to another value in milliseconds.

2010-01-20 15:18:47 +0100 Mark Nauwelaerts <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: always keep media segments within total duration
... as opposed to only doing so following a seek.

2010-01-20 15:44:40 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: rename spspps-interval property
Rename the spspps-interval property to config-interval because it is nicer.

2010-01-19 18:37:31 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: skip RIFF and index in push mode
When we are in push mode, we can encounter RIFF and idx tags in the data chunk
when we are dealing with ODML files. In these cases, simply skip the chunks and
continue streaming instead of going EOS.

2010-01-20 11:27:23 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: more DISCONT handling
Add some debug in the DISCONT handling code.
When we receive a DISCONT in push mode, mark all streams as DISCONT.

2010-01-20 11:26:34 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: reset on flush events
When we receive a flush event on the sinkpad, reset the EOS state and the
flowreturn of all streams. Also mark the streams with a DISCONT.

2010-01-20 11:22:04 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: rename some variable
Rename the seek_event variable to seg_event because it really contains the
newsegment event that needs to be pushed.

2010-01-20 00:54:03 +0000 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From 14cec89 to 15d47a6

2010-01-18 14:49:26 -0500 Olivier Crête <[email protected]>

* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
rtph264pay: Don't set profile-level-id in out caps
The profile-level-id represents restrictions on what can be sent, it does not
describe the stream. So it should be reflected in the sink caps of the
payloader, not the src caps.
https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 14:41:10 -0500 Olivier Crête <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: Don't ignore the return value from set_outcaps
https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 17:43:41 +0100 Sebastian Dröge <[email protected]>

* gst/deinterlace/tvtime/greedyhmacros.h:
* gst/deinterlace/tvtime/linear.c:
* gst/deinterlace/tvtime/linearblend.c:
* gst/deinterlace/tvtime/tomsmocomp.c:
* gst/deinterlace/tvtime/weave.c:
* gst/deinterlace/tvtime/weavebff.c:
* gst/deinterlace/tvtime/weavetff.c:
deinterlace: Fix license and copyright headers

2010-01-18 14:57:42 +0200 Stefan Kost <[email protected]>

* sys/v4l2/gstv4l2bufferpool.h:
v4l2: move G_END_DECLS to the end

2010-01-18 14:55:38 +0200 Stefan Kost <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2bufferpool.h:
v4l2: fix bufferpool file names in header comment

2010-01-15 18:15:14 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: avoid some typecasting

2010-01-15 18:13:24 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: avoid some type checks

2010-01-15 18:09:15 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: fallback to avih duration
when we have not yet parsed the indexes (in push mode, for example) use
the duration as given in the avih header instead of -1.

2010-01-15 13:32:32 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: g_free is NULL safe

2010-01-15 13:27:40 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: use DEMUX errors, instead of DECODE
qtdemux should use DEMUX errors, and not DECODE
Conflicts:
gst/qtdemux/qtdemux.c

2010-01-14 19:16:19 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Minor refactor
Replace repeated code with a function call

2010-01-14 17:11:13 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux_fourcc.h:
qtdemux: Handle another kind of redirect trak
Some traks might contain a redirect rtsp uri inside
hndl atom (which is a dref atom entry). This commit makes qtdemux
post a message when it finds one of these traks and there are
no other traks.
Fixes #597497

2010-01-14 16:13:08 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
qtdemux: Post error when reaching EOS without pads
Post an error when EOS is reached and there are no src pads

2010-01-14 14:13:50 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Do not post empty redirect messages
Some misinterpreted data could result in posting redirect messages
with empty redirect strings. It is better not to post them.
An example is the file on bug #597497

2010-01-14 18:19:25 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: polish last buffer end time usage
That is, reset it upon seek, and note that (rarely) last pushed buffer
time might precede segment start.

2010-01-13 16:48:46 +0200 Stefan Kost <[email protected]>

* gst/videomixer/blend_mmx.h:
videomixer: use 'q' constraint instead of 'r'
This avoids the "bad register name `%dil'" compilation errors on 32bit where
because of 'r' gcc puts the value in a general purpose register and then tries
to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests
a-d registers

2010-01-13 16:44:58 +0200 Stefan Kost <[email protected]>

* gst/avi/gstavidemux.c:
avi: add missing include for sscanf

2010-01-13 09:36:03 +0100 Sebastian Dröge <[email protected]>

* gst/equalizer/gstiirequalizer10bands.c:
equalizer: Fix property description for the 3rd band of the 10band equalizer
The frequency is actually 237 Hz, not 227 Hz.
Fixes bug #606692.

2010-01-13 09:22:20 +0100 Kipp Cannon <[email protected]>

* gst/audiofx/audioamplify.c:
audioamplify: Allow negative amplifications
Fixes bug #606807.

2010-01-13 09:17:05 +0100 Sebastian Dröge <[email protected]>

* ext/taglib/gstapev2mux.cc:
apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5

2010-01-12 17:39:05 +0100 Edward Hervey <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier
Fixes build on macosx

2010-01-11 19:02:34 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: refactor eos sending when pausing loop
Also, prevent hanging if no pads yet on which to send eos by
posting a message instead.

2010-01-11 17:50:35 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: standardize seek handling
... which implies fixing some corner cases.

2010-01-11 15:14:06 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: use more generic xiphN_streamheader_to_codecdata helper

2010-01-11 17:50:04 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: reflow audio and video setcaps and improve logging
Also ensure width and height are available as they are mandatory
in matroska specs.

2010-01-11 11:42:43 -0800 Michael Smith <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix offset for type 2 mp4a sound sample descriptions.
Allows us to correctly find the esds (and thus the codec data) for such
mp4a files.

2010-01-11 15:45:49 -0300 Thiago Santos <[email protected]>

* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
rtpmp4g(de)pay: Only handle raw aac
rtpmp4g(de)pay should only handle raw AAC streams

2010-01-11 18:59:43 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
videomixer: Implement basic QoS
This drops frames if they're too late anyway before blending and all
that starts but QoS events are not forwarded upstream. In the future
the QoS events should be transformed somehow and forwarded upstream.

2010-01-11 14:48:26 -0300 Thiago Santos <[email protected]>

* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtpmp4apay.c:
rtpmp4a(de)pay: Only accept raw aac
rtpmp4a(de)pay should only handle raw aac to conform to the RFC

2010-01-11 18:35:47 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend.c:
* gst/videomixer/blend_mmx.h:
videomixer: Add MMX implementations for I420 and all non-alpha RGB formats

2010-01-04 10:24:45 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/Makefile.am:
* gst/videomixer/blend.c:
* gst/videomixer/blend.h:
* gst/videomixer/blend_ayuv.c:
* gst/videomixer/blend_bgra.c:
* gst/videomixer/blend_i420.c:
* gst/videomixer/blend_mmx.h:
* gst/videomixer/blend_rgb.c:
* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
videomixer: Refactor processing functions
This allows easier plugging of optimized processing functions
in the future, like for SSE or AltiVec.

2010-01-11 13:26:32 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavimux.c:
* gst/matroska/matroska-mux.c:
avimux: matroskamux: rename aac's stream-format to raw
AAC's none stream-format has been renamed to raw, rename
on avimux and matroskamux as well

2010-01-11 12:07:29 -0300 Thiago Santos <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: Only accept raw aac
makes matroskamux reject aac streams that are not
in raw format (stream-format=none)
Fixes #598350

2010-01-11 12:08:55 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavimux.c:
avimux: Only accept raw aac
makes avimux reject aac streams that are not
in raw format (stream-format=none)
Fixes #598350

2010-01-11 10:38:10 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements

2010-01-11 10:17:54 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Debug -> info level for a message for benchmarking index parsing
The extra message output at higher levels affects the accuracy of the
benchmark.

2010-01-11 10:05:10 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed

2010-01-08 13:55:05 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed.

2010-01-08 14:32:06 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour

2010-01-11 00:10:34 +0000 Tim-Philipp Müller <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: remove newline at end of debug statement

2010-01-08 19:26:21 +0100 Havard Graff <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: Compiler warning fixes for Windows
Just simple missing casts
Fixes bug #606438.

2010-01-08 18:04:14 +0100 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: fix seekpoints property copy-and-paste documentation

2010-01-06 17:06:53 +0100 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
* ext/flac/gstflacenc.h:
flacenc: optionally add a seek table
API: GstFlacEnc:seekpoints
Fixes #351595.

2010-01-08 11:33:02 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: Use more glib and be safer
Be safer on sscanf by limiting string format sizes.
Remove useless parameter and use g_strndup.

2010-01-08 10:44:44 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: Simplifying code
Greatly simplify the IDIT chunk handling by using sscanf
instead of 'manually' parsing. Also replaces strncasecmp and
is_alpha/is_digit with glib versions.

2010-01-08 10:18:30 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: it's feb for february
Fix typo in last commit.

2010-01-08 09:17:22 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: Parse and post IDIT dates
Parses and post date tags contained in IDIT chunks.
Fixes #503582

2010-01-07 17:25:05 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofirfilter.c:
* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Add property for not draining the history on kernel changes
Currently this only works if the kernel size doesn't change, in the future
it will be possible to change the kernel size too without draining
the complete history and without loosing anything.
Partially based on a patch by
Thiago Santos <[email protected]>

2010-01-07 16:58:55 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: remove weird memcmp code
Use plain memcmp for comparing memory instead of the custom buggy one.
Fixes #606198

2010-01-07 15:38:36 +0100 Edward Hervey <[email protected]>

* gst/level/gstlevel.c:
level: fix typo in 'message' property description

2010-01-06 14:06:14 +0100 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacdec.c:
flacdec: really use upstream timestamp if there is one
See/fixes #603471.

2010-01-06 13:45:59 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtpg729pay.c:
rtpg728pay: remove unused adapter peek

2010-01-05 19:00:35 -0300 Thiago Santos <[email protected]>

* tests/check/elements/deinterlace.c:
deinterlace: Improve passthrough tests
Improve passthrough tests by forcing more specific
interlaced/deinterlaced caps to be tested

2010-01-05 18:22:49 -0300 Thiago Santos <[email protected]>

* tests/check/elements/deinterlace.c:
deinterlace: Adds some docs to the new tests
Adds some docs explaining the utility functions of the check
tests of deinterlace

2010-01-05 18:14:08 -0300 Thiago Santos <[email protected]>

* tests/check/elements/deinterlace.c:
deinterlace: Adds tests for passthrough
Adds tests for checking if the element really does
passthrough in disabled mode and in auto (if the input is
not interlaced)

2010-01-05 07:50:51 -0300 Thiago Santos <[email protected]>

* tests/check/Makefile.am:
* tests/check/elements/deinterlace.c:
deinterlace: Adds tests for caps acceptance
Adds check unit tests for deinterlace for validating
caps accepting and the expected caps output on the
other pad

2010-01-04 13:43:00 -0300 Thiago Santos <[email protected]>

* tests/check/Makefile.am:
* tests/check/elements/deinterlace.c:
deinterlace: Adds basic check test
Adds a basic check test for deinterlace element

2010-01-04 15:44:28 -0800 Michael Smith <[email protected]>

* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c:
qtdemux: Add support for wave-style audio in qt.
Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM
content.

2009-12-31 17:09:03 -0500 Olivier Crête <[email protected]>

* tests/check/elements/rtp-payloading.c:
tests: Add G.729 RTP payloader/depayloader test
https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:52:30 -0500 Olivier Crête <[email protected]>

* gst/rtp/gstrtpg729pay.c:
rtpg729pay: Simplify adapter usage
https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:27:30 -0500 Olivier Crête <[email protected]>

* gst/rtp/gstrtpg729pay.c:
rtpg729pay: Support ptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-02 19:35:21 +0530 Olivier Crête <[email protected]>

* gst/rtp/README:
rtp: Add maxptime to the README
https://bugzilla.gnome.org/show_bug.cgi?id=606050

2010-01-05 19:03:06 +0100 Wim Taymans <[email protected]>

* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpg723depay.c:
* gst/rtp/gstrtpg723depay.h:
rtpg723depay: add G723 depayloader

2010-01-05 19:02:39 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtpg729depay.c:
* gst/rtp/gstrtpg729depay.h:
rtpg729depay: remove unused variable

2010-01-05 18:33:25 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtpg723pay.c:
* gst/rtp/gstrtpg723pay.h:
rtpg723pay: rewrite payloader
Handle all 3 packet sizes according to RFC 3551.
Totally untested, we don't have a G723 encoder.
Fixes #605882

2010-01-05 11:47:20 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix chunk counter

2010-01-04 19:44:53 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: more work at reducing loop overhead
Try to avoid derefs when parsing the index. Save the state into the structures
when we exit the loop instead of for each iteration.

2010-01-04 16:33:30 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: cleanups and make duration more accurate
Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
as their 32 bit values.
Make some macros to calculate PTS, DTS and duration of a sample.
Deref the sample index less often by keeping a ref to the sample we're dealing
with.

2010-01-04 13:41:18 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: simplify logic to calculate duration
Since we no longer store the timestamp and duration in nanoseconds, we can now
simply store the duration as-is.

2010-01-01 16:42:57 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Store timestamps in mov format in the index
This allows faster building of the index upon seeks so that scaling of
timestamps only occurs when actually needed.

2009-12-18 13:54:46 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: make seeking in push mode work
Move sample position checks into qtdemux_parse_samples where we can protect it
with a lock.
Refactor and make an qtdemux_ensure_index function.
Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
with gst_qtdemux_do_push_seek.

2009-12-18 12:44:27 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: move error code out of normal flow

2009-11-24 16:27:26 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
qtdemux: Add push mode seek support for seeking to obtain the moov atom

2010-01-05 12:22:09 +0100 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: fix on-npt-stop signal warnings for RDT
The RDT manager does not implement this signal so we need to check for it before
trying to connect to it.

2010-01-05 09:47:00 +0000 Tim-Philipp Müller <[email protected]>

* sys/v4l2/gstv4l2src.c:
v4l2src: fix memory leak in new uri handler code
Don't leak a string everytime get_uri() is called and a device
has been set. There's a limited number of devices, so just
intern the string instead of doing more elaborate housekeeping
and storing it in the instance struct or so.

2010-01-01 14:10:49 +0200 Stefan Kost <[email protected]>

* gst/avi/gstavimux.c:
avimux: fix typo in warning message

2010-01-04 09:28:36 -0300 Robert Weidlich <[email protected]>

* ext/shout2/gstshout2.c:
* ext/shout2/gstshout2.h:
shout2send: Add 'public' property
Adds a property to set 'public' flag on libshout, making
the stream listed on the server's stream directory.
Fixes #605269

2009-12-30 14:14:55 +0530 Arun Raghavan <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Add tags for average and maximum bitrate
Fixes #599300.

2009-12-26 16:59:14 -0300 Thiago Santos <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: do not try to alloc really large buffers
When nsamples_out is larger than nsamples_in, using unsigned
ints lead to a overflow and the resulting value is wrong and
way too large for allocating a buffer. Use signed integers
and returning immediatelly when that happens.

2009-12-25 12:38:35 +0100 Wim Taymans <[email protected]>

* gst/videomixer/blend_ayuv.c:
videomixer: optimize blend code some more
Use more efficient formula that uses less multiplies.
Reduce the amount of scalar code, use MMX to calculate the desired
alpha value.
Unroll and handle 2 pixels in one iteration for improved pairing.

2009-12-24 22:59:09 +0100 Wim Taymans <[email protected]>

* gst/videomixer/blend_ayuv.c:
* gst/videomixer/blend_bgra.c:
* gst/videomixer/blend_i420.c:
* gst/videomixer/blend_rgb.c:
videomixer: scale and clamp
Scale and clamp to the max alpha values.

2009-12-24 22:50:31 +0100 Wim Taymans <[email protected]>

* gst/alpha/gstalpha.c:
alpha: scale and clamp alpha to its full extend
Convert the alpha value to 0->255 when setting and to 0->256 when using as
a scaling factor. This makes sure we can reach the full opacity value of 0xff in
all cases.

2009-12-24 22:23:01 +0100 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: fix some comments, remove property check
Fix some comments, clarify some FIXMEs
Remove the on-ntp-stop signal check now that the jitterbuffer is in
-good and we know that it supports this signal.

2009-12-24 20:27:57 +0100 Wim Taymans <[email protected]>

* gst/videomixer/videomixer.c:
videomixer: some trivial cleanups

2009-12-24 17:04:28 -0300 Thiago Santos <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: Parse all rtpinfo entries
Do not forget to parse all rtp-info entries, instead of
parsing the first one only.
Fixes #605222

2009-12-22 12:44:50 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: perf tag should map to GST_TAG_ARTIST

2009-12-24 17:03:02 +0100 Wim Taymans <[email protected]>

* gst/interleave/interleave.c:
interleave: fix weird indentation

2009-12-24 17:01:54 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph263ppay.c:
rtph263ppay: use faster _adapter_copy() whem possible

2009-12-24 17:01:15 +0100 Wim Taymans <[email protected]>

* tests/examples/audiofx/firfilter-example.c:
tests: use right type when passing vararg value

2009-12-23 17:50:34 +0100 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacdec.c:
* ext/flac/gstflacdec.h:
flacdec: use a single decoder field for both push and pull mode

2009-12-23 17:03:32 +0100 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacdec.c:
flacdec: fix possible hanging in pull mode seeking
A seek in multi-sink pipeline typically leads to several seek events in a row,
which could lead to sending several newsegments in a row without intermediate
flushing. These would then accumulate, distort rendering times and as such
lead to 'hanging'.

2009-12-23 19:39:05 +0100 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: fix uninitialized variable

2009-12-23 13:09:54 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpac3depay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpbvpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpg729depay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263depay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpjpegdepay.c:
* gst/rtp/gstrtpmp1sdepay.c:
* gst/rtp/gstrtpmp2tdepay.c:
* gst/rtp/gstrtpmp4apay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtpmpvdepay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpqdmdepay.c:
* gst/rtp/gstrtpsirenpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
* gst/rtp/gstrtpvrawdepay.c:
* gst/rtp/gstrtpvrawpay.c:
rtp: use boilerplate

2009-12-23 00:38:05 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpL16pay.h:
rtpL16pay: convert to baseaudiopayload
Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
a bunch of problems that were already solved in the base class.
Fixes #853367

2009-12-23 00:30:49 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtppcmapay.c:
rtppcmapay: the boilerplate macro sets parent_class

2009-12-22 22:27:21 +0100 Wim Taymans <[email protected]>

* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
rtpbin: avoid some structure copies
Don't make copied in the getter and setter for SDES in the RTPSource. This
avoids a couple of copies of the SDES structure when generating RTCP
packets.

2009-08-31 18:42:25 +0200 Pascal Buhler <[email protected]>

* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
rtpmanager: improve SDES handling
Store SDES internally as a struct to support multiple PRIV values.
Include all values set in SDES struct when sending RTCP SDES.

2009-12-22 14:41:35 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph263depay.c:
rtph263depay: add some fixmes

2009-12-22 14:35:13 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph263depay.c:
rtph263depay: baseclass handles timestamps for us

2009-12-22 14:27:40 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph263depay.c:
rtph263depay: reset start variable properly

2009-05-29 15:49:27 +0300 Marco Ballesio <[email protected]>

* gst/rtp/gstrtph263depay.c:
* gst/rtp/gstrtph263depay.h:
Drop the whole frame if a packet is lost.
Fixes #582575

2009-12-21 20:39:53 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
rtph264pay: add option to insert PPS/SPS in streams
Add a new spspps-interval property to instruct the payloader to insert
SPS and PPS at periodic intervals in the stream.
Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
same code paths to handle sprop-parameter-sets. This also allows to have the AVC
code to insert SPS/PPS like the bytestream code.
Fixes #604913

2009-12-21 19:12:22 +0100 Mark Nauwelaerts <[email protected]>

* common:
Automatic update of common submodule
From 47cb23a to 14cec89

2009-12-21 12:01:53 -0300 Jonathan Conder <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
qtdemux: Adds new tags
Adds some new tags mapping to qtdemux.
Fixes #599759

2009-12-21 15:05:09 +0100 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpbin.c:
rtpbin: add property to remove pads automatically
Add a property called autoremove to automatically remove the pads of sources
that timed out.
Fixes #554839

2009-12-21 14:55:16 +0100 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpssrcdemux.c:
ssrcdemux: fix comparison
A NULL means no pad was found.

2009-11-08 11:49:14 +0100 Edward Hervey <[email protected]>

* sys/v4l2/gstv4l2src.c:
v4l2src: Add GstURIHandler interface. Fixes #601143
This allows using v4l2://[<device>]

2009-12-20 17:24:47 -0800 Michael Smith <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: pass length parameter to g_convert

2009-12-18 12:44:50 +0100 Edward Hervey <[email protected]>

* gst/matroska/matroska-demux.c:
matroska: Fix unitialized variable.
Yes, it's stupid, but macosx compilers are even more stupid.

2009-12-17 16:01:25 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_ayuv.c:
videomixer: Fix assembly compilation on x86
Fixes bug #604814.

2009-12-17 17:37:03 +0100 Branko Čibej <brane at xbc.nu>

* gst/replaygain/rganalysis.c:
rganalysis: fix timestamp rounding
Use scaling function to round and avoid overflows.
Fixes #604352

2009-12-17 17:27:42 +0100 Tiago Katcipis <[email protected]>

* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpg723pay.c:
* gst/rtp/gstrtpg723pay.h:
rtp: add G723 payloader
Fixes #597823

2009-12-17 16:22:56 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux_types.c:
qtdemux: Fix ALAC codec_data parsing
Fixes #604611

2009-12-16 17:28:30 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Remove cpp style coments
Removes // comments and replace them with /* */ comments

2009-12-16 12:48:02 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: also consider BlockNumber indicated in index when seeking

2009-12-16 12:43:27 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/ebml-read.c:
* gst/matroska/ebml-read.h:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: support push based mode
Fixes #598610.

2009-12-16 12:44:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/ebml-read.c:
matroskademux: fix ebml read cache usage

2009-12-16 10:50:32 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_ayuv.c:
videomixer: Use movzbl instead of movzxb for moving one byte to a l register
For some reason latest gcc/binutils accept movzxb here while
movzbl would be correct and is the only thing accepted by older
gcc/binutils.
Fixes bug #604679.

2009-12-16 06:59:01 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_ayuv.c:
videomixer: src/dest are input and output of the AYUV blending MMX assembler

2009-12-15 18:18:54 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsincband.c:
audiowsincband: Use the same upper length limit as audiowsinclimit

2009-12-12 17:00:50 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
audiowsinc{limit,band}: Allow much larger filter lengths now

2009-12-11 12:27:32 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Fix frequency response calculation

2009-12-08 14:57:02 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Remove dead assignments

2009-12-06 16:58:51 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Add special processing functions for Mono/Stereo
This provides another 7% speedup for the time domain convolution and 1.5%
speedup for the FFT convolution on Mono input.
This optimization assumes that the compiler simplifies calculations
and conditions on constant numbers and unrolls loops with a constant
number of repeats.

2009-12-04 09:25:49 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Add a "low-latency" mode
This will always use time-domain convolution, which lowers the latency.
With FFT convolution it's always a multiple of the kernel length,
with time domain convolution it's only the pre-latency of the filter kernel.

2009-12-04 09:00:22 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Remove obsolete TODO comments

2009-12-03 20:12:01 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes

2009-12-03 17:27:13 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: FFT convolution implementation
This provides a great speedup, especially the relationship between kernel
length and processing size is now logarithmic instead of linear. Below a
kernel size of 32 it's a bit slower, afterwards it's much faster:
17 0.788000 -> 0.950000
33 1.208000 -> 1.146000
65 2.166000 -> 1.146000
...
4097 107.444000 -> 1.508000
For sizes smaller 32 the normal time-domain convolution is chosen,
for larger sizes the FFT convolution is automatically used.
Fixes bug #594381.

2009-11-27 20:33:14 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm
Only remaining part is the residue pushing, which will be fixed later.

2009-11-26 15:17:27 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Optimize time-domain convolution
Remove some redundant calculations, move comparisions out of
inner loops, etc.
This makes the convolution about 3 (!) times faster but
processing time is of course still proportional to the
filter size.

2009-11-26 10:45:37 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once

2009-11-25 18:12:05 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Rewrite timestamp tracking
It's much simpler now and doesn't introduce accumulating rounding
errors.

2009-11-25 17:39:53 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Rename some variables and change comments

2009-11-24 20:06:25 +0100 Sebastian Dröge <[email protected]>

* gst/audiofx/audiofxbasefirfilter.c:
* gst/audiofx/audiofxbasefirfilter.h:
audiofxbasefirfilter: Add const qualifier to the source data array

2009-12-14 20:08:06 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/Makefile.am:
* gst/videomixer/blend_ayuv.c:
* gst/videomixer/videomixer.c:
videomixer: Add MMX implementations of the AYUV blending and color filling functions
This provides a 20% speedup for blending and 100% for color filling.
The blending can probably be optimized even more.

2009-12-13 13:19:43 +0000 Tim-Philipp Müller <[email protected]>

* gst/id3demux/id3v2frames.c:
id3demux: prefer two letter ISO 639-1 code for extended comment

2009-12-13 13:10:12 +0000 Tim-Philipp Müller <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix up language code extraction some more
Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
is supposed to hold a ISO 639-1 code, so convert as needed using
the new API from -base.
See #602126.

2009-12-13 12:45:22 +0000 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
matroska: fix language code writing and extraction
Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
supposed to contain two-letter ISO 639-1 codes, so use new language
code mapping functions in -base to convert between those two as
needed.
Fixes #505823.

2009-12-07 20:54:07 +0000 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: minor debug message changes
Fix up a few debug messages so that it's clearer what they mean.

2009-12-12 17:44:04 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
Revert "qtdemux: Correctly parse classification tags"
This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde.
Previous code was correct, 4 is due to table and language code,
not only language code

2009-12-12 16:28:36 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Correctly parse classification tags
In clsf atoms, the language code is 2 bytes long, not 4.

2009-12-12 16:55:13 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers
... NULL buffers shouldn't really happen anymore when popping the
buffer from GstCollectPads but better check for this and print a warning.

2009-12-11 13:11:12 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_i420.c:
videomixer: Fix stupid mistake in last commit

2009-12-11 12:35:59 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_i420.c:
videomixer: Don't do floating point math in the inner processing loop for I420 blending

2009-12-10 18:43:44 +0100 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: handle NULL and empty transport strings
When an RTSP extension returns NULL or an empty transport string, just ignore it
and try to get the next possible transport. Fixes playback of RealMedia streams.

2009-12-10 18:42:51 +0100 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: install event function on internal RTCP pad
Install a custom event function on the internal RTCP pad so that we can reply
TRUE to a latency event.

2009-12-10 10:48:49 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/blend_ayuv.c:
* gst/videomixer/blend_bgra.c:
* gst/videomixer/blend_rgb.c:
videomixer: Remove wrong comments, copied from the I420 blend function

2009-12-09 21:15:07 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
videomixer: The queued duration is a signed integer
...and it will really be negative sometimes.

2009-12-09 21:03:57 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
videomixer: Only pop buffers from collectpads after they're fully consumed
This decreases latency and memory usage because new buffers are only
accepted by collectpads if there's no queued buffer.

2009-12-09 20:42:44 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: Clean up position/duration handling
Also use the last end time for closing the segment, not the
start time of the last buffer.

2009-12-09 16:50:02 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Close the segment on EOS if the real duration is known

2009-12-09 16:46:18 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Update duration if current buffer is already after the old duration

2009-12-09 16:43:41 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Drop buffers that are after segment stop
...and if this happened for all streams go EOS.

2009-12-09 16:41:04 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Fix position tracking and sending of filler segments

2009-12-09 16:15:09 +0100 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations

2009-12-08 17:34:15 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Keep the segment stop position for update newsegment events

2009-12-04 14:42:49 +0100 Sebastian Dröge <[email protected]>

* configure.ac:
* ext/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/ladspa/gstladspa.c:
* ext/ladspa/gstladspa.h:
* ext/ladspa/gstsignalprocessor.c:
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/load.c:
* ext/ladspa/search.c:
* ext/ladspa/utils.h:
ladspa: Remove the sources from gst-plugins-good
It's disabled anyway and the latest version of it is in
gst-plugins-bad. Fixes bug #603779.

2009-12-04 13:50:59 +0100 Wim Taymans <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: init current_entry in push mode
Set the current_entry to 0 (instead of -1) in push mode so that we correctly
calculate the current frame number and timestamp.
Add some more debug info and fic the duration debug.

2009-12-04 11:14:03 +0000 Tim-Philipp Müller <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: fix major memory leak when playing back rtsp video streams
Don't forget to unref QoS, navigation and latency events when
dropping them.

2009-12-03 08:58:08 +0000 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: only send pending tags with newsegment events
Send pending tags only from the streaming thread, just after we've sent
the newsegment event, not with e.g. flush-start. This not only does the
right thing, but also makes sure we're not trampling over variables set
up in the streaming thread from the seeking thread in case someone tries
to issue a seek just as the demuxer is parsing the headers.
Fixes #601617. Spotted by Ognyan Tonchev.

2009-12-03 17:49:55 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix debug message printf args
Fixes debug message printf format to make it build in mac's gcc

2009-12-02 13:33:20 -0300 Thiago Santos <[email protected]>

* ext/shout2/gstshout2.c:
shout2: Convert delay correctly
Use GST_MSECOND to convert delay in msecs to nanosecs
Fixes #603547

2009-12-01 19:24:02 +0100 Wim Taymans <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: reset segment info after flush
Reset the segment info after a flush. We use the segment for handling QoS and if
we don't reset the segment, QoS is basically disabled after a flushing seek.

2009-12-01 15:07:06 +0000 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From 87bf428 to 47cb23a

2009-12-01 14:15:46 +0100 Sebastian Dröge <[email protected]>

* common:
Automatic update of common submodule
From da4c75c to 87bf428

2009-11-30 15:59:50 +0100 Aurelien Grimaud <gstelzz at yahoo dot fr>

* gst/rtpmanager/rtpsession.c:
rtpsession: avoid buffer ref/unref pairs for CSRCs
We ref the buffer before pushing it downstream in order to get the CSRCs of it
after pushing. This causes performance problems when downstream elements want to
change the metadata because the buffer needs to be subbuffered.
Instead, read and store the CSRCs of the buffer in an array before pushing it
and process the array after pushing the buffer. This allows us to remove the
ref/unref pair.
Fixes #603376

2009-11-28 19:23:26 +0100 Wim Taymans <[email protected]>

* ext/shout2/gstshout2.c:
* ext/shout2/gstshout2.h:
shout2: use gstpoll for timeouts
Use our own GstPoll based timeout instead of the shout sleep so that we can
interrupt when doing a state change and shutting down.
Fixes #602887

2009-11-28 12:25:06 +0100 Wim Taymans <[email protected]>

* tests/check/elements/rtpjitterbuffer.c:
check: fix jitterbuffer check
Make sure we set a base_time on the element.
Fix the timeout to at least twice the jitterbuffer latency.
Enable previously failing tests.
Remove impossible checks.

2009-11-27 18:55:20 +0100 Edward Hervey <[email protected]>

* common:
Automatic update of common submodule
From 53a2485 to da4c75c

2009-11-26 16:14:30 +0100 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph264depay.h:
rtph264depay: optionally merge NALUs into Access Units
... which may be expected/desired by some downstream decoders
(and spec-wise highly recommended for at least non-bytestream mode).

2009-11-26 17:29:03 +0100 Mark Nauwelaerts <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix timestamp datatype

2009-11-25 10:38:23 -0600 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: avoid using wrong clock-rate
Check for a valid clock-rate before attempting to estimate the npt
stop time.

2009-11-25 10:37:30 -0600 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpbin.c:
rtpbin: fix typo in comments

2009-11-25 16:05:10 +0200 Stefan Kost <[email protected]>

* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffertest: add one more test and file a bug now
CHange the backwards test to always send first buffer first to have a define
basetime. Add another test that sends buffers backwards to assert that only
first sent buffer is keep and used as basetime. Disabled those tests still,
as its not passing/failing consitently and file a bug for jitterbuffer.

2009-11-25 10:17:34 +0200 Stefan Kost <[email protected]>

* tests/check/elements/rtpjitterbuffer.c:
jitterbuffertest: improve the test
the tests are a bit more solid now but still not produce reliable results.
Wonder if they are still flawky or if its a bug in jitterbuffer.

2009-11-24 11:13:06 -0800 Michael Smith <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: return error message on windows too.

2009-11-24 10:58:49 -0800 Michael Smith <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: first phase of fixing up error reporting for windows.

2009-10-30 03:13:54 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavimux.c:
avimux: also set the suggested buf size for audio
We were only setting the suggested buf size for video,
we can set it for audio as well.
This and 195e14529d80ef318ce3a778c1995efb11f266cd
fix an issue that prevented seeking on large avi files
on WMP (non-recent versions).

2009-11-04 16:10:23 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavimux.c:
* gst/avi/gstavimux.h:
avimux: fix indx duration for PCM audio
GstBuffers for PCM audio usually contains more than
1 sample, we need to get the total number of samples to set
the indx duration.

2009-11-04 16:04:10 -0300 Thiago Santos <[email protected]>

* gst/avi/gstavimux.c:
avimux: Audio buffers should be picked earlier
Adds a 0.5s advantage for audio buffers to being
picked earlier for muxing.

2009-11-24 16:40:19 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Fix push mode by making sure stbl information is available in next_entry_size ()

2009-11-24 16:35:20 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Fix order of arguments in log message

2009-11-24 15:51:21 +0200 Stefan Kost <[email protected]>

* ext/jpeg/gstjpegenc.c:
jpegenc: fix spelling in comment

2009-11-23 17:58:17 +0100 Robert Swain <[email protected]>

* common:
build system: Fix wrongly committed change to common/

2009-11-10 10:26:07 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Ease debugging by removing a goto for an error message

2009-11-14 15:52:09 +0100 Robert Swain <[email protected]>

* common:
* gst/qtdemux/qtdemux.c:
qtdemux: Parse per sample rather than all at once but build complete index when seeking

2009-11-04 17:31:15 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Save atom data for later use so it doesn't get freed after initial parsing

2009-11-06 11:00:04 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Parse from the previously parsed sample up to sample n

2009-11-04 17:04:22 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Make qtdemux_parse_samples () parse up to n samples

2009-10-28 17:49:02 +0000 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Separate off stbl sub-atom initialisation

2009-10-26 22:42:36 +0000 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Move variables into context in preparation for refactorisation

2009-10-26 20:36:08 +0000 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Fix bug where stps is never parsed due to logic error

2009-11-04 17:31:15 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Port ctts from Gnode * to GstByteReader

2009-10-23 13:06:44 +0100 Robert Swain <[email protected]>

* gst/qtdemux/qtatomparser.h:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux_dump.c:
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_types.h:
qtdemux: Switch from QtAtomParser to GstByteReader

2009-11-23 12:53:50 +0100 Wim Taymans <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix typo and grammar

2009-11-20 10:30:00 +0000 Tim-Philipp Müller <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: fix typo in mode enum description

2009-11-20 11:25:49 +0200 Stefan Kost <[email protected]>

* gst/rtpmanager/gstrtpbin.c:
docs: more links and better short description
Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
the short description to be more meaningful.

2009-11-20 09:58:26 +0100 Sebastian Dröge <[email protected]>

* tests/check/elements/wavpackparse.c:
wavpackparse: Fix unit test for recent position reporting changes

2009-11-19 16:09:38 +0100 Sebastian Dröge <[email protected]>

* ext/wavpack/gstwavpackparse.c:
wavpackparse: After pushing a frame, update last_stop to the end of the frame
This improves position reporting, especially because of the fact that
WavPack frames are usually 0.5-1.0 seconds long.

2009-11-19 16:08:33 +0100 Sebastian Dröge <[email protected]>

* ext/wavpack/gstwavpackparse.c:
wavpackparse: Allow pulling the last WavPack frame of a file
Because of a >= instead of a >, that last frame of a WavPack file
would never be parsed in pull mode.

2009-11-19 10:30:43 +0000 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From 0702fe1 to 53a2485

2009-10-29 08:29:38 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux_fourcc.h:
qtdemux: Add more fields to SVQ3 caps
qtdemux only added the whole stsd atom as 'codec_data'
in its output caps for SVQ3. This patch makes it add
the SEQH (inside a SMI atom) and a gamma field (taken
from the gama atom) if available.
Fixes #587922

2009-11-18 17:55:42 +0100 Edward Hervey <[email protected]>

* gst/wavenc/gstwavenc.c:
wavenc: Raise rank of muxer to PRIMARY

2009-11-18 17:54:16 +0100 Edward Hervey <[email protected]>

* gst/y4m/gsty4mencode.c:
y4m: Raise rank of encoder to PRIMARY

2009-11-18 17:54:02 +0100 Edward Hervey <[email protected]>

* gst/law/alaw.c:
* gst/law/mulaw.c:
law: Raise rank of encoders to PRIMARY

2009-11-12 19:11:18 +0000 Bastien Nocera <[email protected]>

* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
Add user-id and user-pw properties
So that one doesn't need to modify the URL to have access
to authenticated RTSP streams.
fixes #601728

2009-11-18 12:22:10 +0100 Wim Taymans <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: use acquired flag when checking valid state
Use the acquired field of the ringbuffer in get_time to know when we are in an
invalid state. We don't clear the rate flag when releasing the ringbuffer so
this values is not usable.
Avoids some error messages being posted because the pulseaudio connection is
down.

2009-11-18 10:17:02 +0000 Tim-Philipp Müller <[email protected]>

* configure.ac:
configure: bump core requirement to 0.10.25.1 as well
Make implicit requirement explicit.

2009-11-18 12:53:44 +0100 Mark Nauwelaerts <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: fix bogus memory chunk size check

2009-11-18 12:01:52 +0100 Wim Taymans <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: implement some more callbacks
Implement some more callbacks for debugging purposes.

2009-11-11 15:50:19 +0100 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: release lock before emiting signals
Release the jbuf lock before emiting the request-pt-map signal to avoid
deadlocks. We also need to catch the shutdown case when locking again.
Fixes #593354

2009-11-11 11:59:16 +0100 Wim Taymans <[email protected]>

* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpbvdepay.c:
* gst/rtp/gstrtpbvdepay.h:
rtp: add BroadcomVoice depayloader

2009-11-11 11:38:36 +0100 Wim Taymans <[email protected]>

* gst/rtp/gstrtpbvpay.c:
rtpbvpay: add rfc reference

2009-11-11 11:37:07 +0100 Wim Taymans <[email protected]>

* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpbvpay.c:
* gst/rtp/gstrtpbvpay.h:
rtp: add BroadcomVoice payloader

2009-11-09 12:17:34 +0100 Jan Urbański <[email protected]>

* gst/flv/gstflvmux.c:
flvmux: properly finish the ECMA array
The ECMA array with the file index was missing a mandatory end marker.
Fixes bug #601242.

2009-11-18 02:15:15 +0000 Jan Schmidt <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
Use new still-frame API from gst-plugins-base

2009-11-18 02:14:46 +0000 Jan Schmidt <[email protected]>

* configure.ac:
Bump gst-plugins-base requirement to 0.10.25.1

2009-11-17 17:59:13 -0800 Michael Smith <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: identify IMA adpcm in qt properly.

2009-11-18 01:27:37 +0000 Jan Schmidt <[email protected]>

* configure.ac:
* win32/common/config.h:
Back to development -> 0.10.17.1

2009-11-17 01:53:08 +0000 Jan Schmidt <[email protected]>

* gst-plugins-good.doap:
Add release 0.10.17 to the doap file