Commit graph

4115 commits

Author SHA1 Message Date
Jan Schmidt
6391d87d09 configure.ac: Bump verion back to devel -> 0.10.20.1
Original commit message from CVS:
* configure.ac:
Bump verion back to devel -> 0.10.20.1
2008-06-20 08:52:21 +00:00
Sebastian Dröge
31f3f65d53 gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't increase the size of non-string image buffers by one as this
might in theory confuse decoders. Still increase it by one for string
image buffers to append '\0'.
2008-06-20 08:47:14 +00:00
Antoine Tremblay
1a71c15677 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
2008-06-20 08:45:13 +00:00
Jan Schmidt
01e689e359 Release 0.10.20
Original commit message from CVS:
Release 0.10.20
2008-06-18 14:36:28 +00:00
Jan Schmidt
5dd552cf45 configure.ac: 0.10.19.3 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.3 pre-release
2008-06-11 21:17:01 +00:00
David Schleef
526b2e63a2 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on win32.
Patch By: David Schleef <ds@schleef.org>
Fixes: #536874
2008-06-11 20:13:00 +00:00
Sebastian Dröge
bb595d8fd8 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
(gst_gio_base_src_create):
* ext/gio/gstgiobasesrc.h:
Try to read the requested number of bytes, even if the first
read returns less than requested, until nothing is read anymore
or we have the requested amount of bytes. This fixes playback of
files via Samba as Samba only allows to read 64k at once.
Implement a caching algorithm that makes sure that we read at
least 4k of data every time. Some elements will try to read a few
bytes, then seek, read again a few bytes and so on and this is
painfully slow as every operation has to go over DBus if GVfs is
used as backend.
Fixes bug #536849 and #536848.
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
(gst_gio_src_check_get_range):
Override check_get_range() to blacklist http/https URIs
and whitelist file URIs. More to be added on demand.
2008-06-11 09:35:51 +00:00
Jan Schmidt
71546edc4b configure.ac: 0.10.19.2 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.2 pre-release
2008-06-05 09:47:23 +00:00
Jan Schmidt
cb8b68c547 win32/common/: Add new API functions to the dll exports
Original commit message from CVS:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
Add new API functions to the dll exports
2008-06-04 21:48:27 +00:00
Michael Smith
2fdd607e95 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.
2008-06-04 17:42:38 +00:00
Tim-Philipp Müller
93db55c074 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
2008-06-04 17:12:40 +00:00
Peter Kjellerstedt
ec07ea9905 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
Original commit message from CVS:
* tests/check/Makefile.am:
Do not try to run the check tests for subparse unless it has been
built.
2008-06-04 16:06:49 +00:00
Peter Kjellerstedt
4d05d8ab6b tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
Do not try to run a test which requires vorbisenc unless we have
actually built it.
2008-06-04 16:00:26 +00:00
Peter Kjellerstedt
26cd5ea1c8 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.
2008-06-04 11:53:53 +00:00
Peter Kjellerstedt
c140528357 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.
2008-06-04 11:33:23 +00:00
Tim-Philipp Müller
44e087f8c9 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Document mixer track flags.
2008-06-04 10:18:42 +00:00
Antoine Tremblay
be2f6a8085 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
2008-06-04 05:58:38 +00:00
Sebastian Dröge
d57ab7cfdb gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
2008-06-04 05:44:06 +00:00
Sebastian Dröge
8b14d08115 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
2008-06-04 04:24:27 +00:00
John Millikin
f934d1c233 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
Original commit message from CVS:
Based on patch by: John Millikin <jmillikin gmail com>
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
(gst_vorbis_tag_add_coverart):
Retrieve COVERART tags from vorbis comments (#512333)
2008-06-03 20:01:58 +00:00
Tim-Philipp Müller
8b491df810 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Don't forget to add new enum value here too (should probably use
glib-mkenums here...).
2008-06-03 19:44:48 +00:00
Tim-Philipp Müller
cd9bb9a674 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
(gst_tag_image_data_to_image_buffer):
Add two utility functions to avoid code duplication (#512333):
API: add gst_tag_image_data_to_image_buffer()
API: add gst_tag_list_add_id3_image()
2008-06-03 19:29:06 +00:00
Sebastian Dröge
eb93e07392 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_check_channel_positions() to the exported symbols.
2008-06-03 08:54:29 +00:00
Sebastian Dröge
0de81029c8 API: Make gst_audio_check_channel_positions() public.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
2008-06-03 08:48:32 +00:00
Tim-Philipp Müller
d5b8246724 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
Original commit message from CVS:
* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
minrange and maxrange are scaled according to the frequency
multiplier.
2008-06-02 20:09:14 +00:00
Tim-Philipp Müller
d868077048 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
Original commit message from CVS:
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
Use gstvideo functions to calculate strides and plane offsets. Fixes
rendering issue ('ghost' images of the text on the chroma planes)
with widths or heights that are not multiples of 8 (#506659 and
probably also #485729).
* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
(main):
Test with odd height/width too.
2008-06-02 18:37:02 +00:00
Sebastian Dröge
1d37b272ce gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
2008-06-02 12:20:35 +00:00
Mark Nauwelaerts
9fa61c528d gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.
2008-05-31 19:57:57 +00:00
Mark Nauwelaerts
1985500efe ChangeLog surgery, mark API change
Original commit message from CVS:
ChangeLog surgery, mark API change
2008-05-31 19:50:59 +00:00
Mark Nauwelaerts
c660bbd6dd gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
2008-05-31 18:10:47 +00:00
Wim Taymans
11309247f3 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.
2008-05-30 15:29:20 +00:00
Sebastian Dröge
fdd708c418 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
2008-05-30 08:42:17 +00:00
Sebastian Dröge
79cf1cf8fd win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_clock_reset to the list of exported symbols.
2008-05-29 19:45:40 +00:00
Sebastian Dröge
ca7a0b8e9e tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).
2008-05-29 19:37:47 +00:00
Sebastian Dröge
b86a5d4303 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
2008-05-29 12:17:16 +00:00
Sebastian Dröge
45ef6b5e13 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously	conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 11:34:09 +00:00
Sebastian Dröge
31b677599a ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
Add sane defaults for the 7 and 8 channel layouts as those are
undefined in the Vorbis spec. Use NONE channel layouts when decoding
more than 8 channels instead of erroring out. Fixes bug #535356.
2008-05-29 07:02:50 +00:00
Wim Taymans
1a3053b241 Add theoraparse to the docs and fix some docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/theora/theoraparse.c:
Add theoraparse to the docs and fix some docs.
2008-05-28 16:10:20 +00:00
Wim Taymans
2855fb48ad gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.
2008-05-28 15:48:33 +00:00
Mark Nauwelaerts
17b17a566f gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.
2008-05-28 14:49:24 +00:00
Tim-Philipp Müller
b82c4cee0e tests/examples/seek/seek.c: Initialise error to NULL as we should.
Original commit message from CVS:
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
Initialise error to NULL as we should.
2008-05-28 11:31:44 +00:00
Sebastian Dröge
57c3aa9b66 gst/adder/gstadder.c: Implement latency query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.
2008-05-28 08:14:47 +00:00
Sebastian Dröge
4ccac97b40 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
2008-05-27 18:10:00 +00:00
Tim-Philipp Müller
5d121dd673 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
Original commit message from CVS:
* win32/vs6/libgstpbutils.dsp:
Add pbutils-enumtypes.c to sources (#518037).
2008-05-27 17:14:07 +00:00
Wim Taymans
35e4b75b86 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 16:20:17 +00:00
Tim-Philipp Müller
dc9eb0d6b8 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).
2008-05-27 16:11:32 +00:00
Tim-Philipp Müller
555feaa11b tests/check/pipelines/oggmux.c: Don't use deprecated function.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (test_pipeline):
Don't use deprecated function.
2008-05-27 10:57:56 +00:00
Wim Taymans
514b8fa456 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
2008-05-27 10:35:55 +00:00
Wim Taymans
13d7048f69 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.
2008-05-26 17:18:52 +00:00
Tim-Philipp Müller
fa38b99379 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
2008-05-26 10:29:20 +00:00
Tim-Philipp Müller
5ce4d71f82 tests/check/libs/video.c: More checks.
Original commit message from CVS:
* tests/check/libs/video.c:
More checks.
2008-05-26 10:26:00 +00:00
Tim-Philipp Müller
206f91995b Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
2008-05-25 20:51:35 +00:00
Wim Taymans
79a725148d gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
2008-05-23 14:14:28 +00:00
Tim-Philipp Müller
747d52adb3 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.
2008-05-22 22:35:40 +00:00
Jan Schmidt
d58def621b Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
2008-05-22 22:09:16 +00:00
Thijs Vermeir
88b1e8efcf gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.
2008-05-22 18:30:15 +00:00
Sjoerd Simons
1c424d9d93 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.
2008-05-22 11:59:33 +00:00
Felipe Contreras
75d05dc499 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
Original commit message from CVS:
* docs/Makefile.am:
Fix installing plugin documentation when gtk-doc is disabled.
2008-05-21 17:09:42 +00:00
Felipe Contreras
b5f896dad6 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h
2008-05-21 17:01:16 +00:00
Julien Moutte
0f80e462d9 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
2008-05-21 16:47:58 +00:00
Wim Taymans
2cdf18edff Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
2008-05-21 16:44:15 +00:00
Wim Taymans
c6b54c3d02 Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
Stefan Kost
eda6d89b8c gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.
2008-05-21 14:35:41 +00:00
Wim Taymans
3cd156cad5 docs/design/part-playbin2.txt: Add some leftover doc.
Original commit message from CVS:
* docs/design/part-playbin2.txt:
Add some leftover doc.
2008-05-21 11:52:30 +00:00
Sebastian Dröge
736b181916 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
2008-05-21 11:36:37 +00:00
Sebastian Dröge
7d605d4514 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
2008-05-21 11:30:58 +00:00
Henrik Eriksson
10ae17ced1 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.
2008-05-21 11:29:25 +00:00
Sebastian Dröge
74d46a9977 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.
2008-05-21 07:46:02 +00:00
Sebastian Dröge
d03bbd1e3e gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:39:56 +00:00
Sebastian Dröge
d47bd6d7bc gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
2008-05-21 07:28:04 +00:00
Antoine Tremblay
a8dda35c1b gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
2008-05-21 06:45:22 +00:00
Sebastian Dröge
3ee2676c2e gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.
2008-05-21 06:39:20 +00:00
Sebastian Dröge
e66b0a6642 gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
2008-05-21 05:48:05 +00:00
Wim Taymans
f36d9d6b08 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 16:26:53 +00:00
Tim-Philipp Müller
d0932b0aa1 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
Original commit message from CVS:
* configure.ac:
Require core CVS for GstBaseSrc buffer caps setting magic.
2008-05-20 14:35:42 +00:00
Sebastian Dröge
fcda3964dc gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
2008-05-20 12:26:32 +00:00
Sebastian Dröge
d76c4b4c65 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
2008-05-20 12:15:34 +00:00
Wim Taymans
d8dc371c0d ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
2008-05-20 11:13:27 +00:00
Wim Taymans
95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
Sebastian Dröge
b5a5d64713 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
2008-05-20 08:12:19 +00:00
Tim-Philipp Müller
28c01f5015 configure.ac: Error out if we don't have the required version of core.
Original commit message from CVS:
* configure.ac:
Error out if we don't have the required version of core.
2008-05-19 16:13:25 +00:00
Tim-Philipp Müller
7cb1276dac gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
2008-05-19 15:59:40 +00:00
Tim-Philipp Müller
cfc8f3c0d7 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
2008-05-19 14:09:08 +00:00
Sebastian Dröge
05cf63634e gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
2008-05-16 21:12:02 +00:00
Wim Taymans
86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
Tim-Philipp Müller
d92ff26d29 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:57:41 +00:00
Bernard B
d06df554a9 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 13:43:12 +00:00
Sebastian Dröge
6720c5beb8 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
2008-05-14 10:58:52 +00:00
Stefan Kost
5965f5e8a9 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Better debug logging in port value handling. Merging separate port
value loops into one.
2008-05-14 09:12:10 +00:00
Hannes Bistry
b9bc12afd8 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.
2008-05-13 16:02:19 +00:00
Sebastian Dröge
05349cc354 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
2008-05-13 13:04:24 +00:00
Sebastian Dröge
5800b1ac77 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstrtsp.def:
Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
symbols.
2008-05-13 11:37:15 +00:00
Sjoerd Simons
fd84ec0ca3 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
2008-05-13 10:59:49 +00:00
Sebastian Dröge
4d5870847f gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.
2008-05-13 09:14:44 +00:00
j^
1a154e1d3d ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone):
* ext/ogg/gstoggdemux.h:
Parse presentation time from skeleton streams and use it as offset
for the timestamps. Fixes bug #530068.
2008-05-13 07:28:21 +00:00
Wim Taymans
0c9b13988c gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-12 08:45:11 +00:00
Tim-Philipp Müller
1482332184 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
Fix docs: type and missing word.
2008-05-11 19:52:59 +00:00
Tim-Philipp Müller
fed34307db gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
2008-05-10 20:16:21 +00:00
Tim-Philipp Müller
104fed4d66 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
2008-05-10 18:19:17 +00:00
Jan Schmidt
f11cf32c3f Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.
2008-05-09 21:42:26 +00:00
Wim Taymans
fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 16:38:10 +00:00
Sebastian Dröge
531c6fb462 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
2008-05-09 08:34:52 +00:00
Edward Hervey
9fa3d7a294 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
2008-05-08 17:35:44 +00:00
Wouter Cloetens
a8a2b9c717 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
2008-05-08 14:46:27 +00:00
Sjoerd Simons
09163ca363 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Ole André Vadla Ravnås
7a22e13f03 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
Original commit message from CVS:
* win32/common/config.h.in:
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
use the real thing than having "???" unconditionally.
2008-05-07 19:50:27 +00:00
Wim Taymans
09f7dee84d gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 15:47:03 +00:00
Sebastian Dröge
b9a285021c gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
2008-05-06 12:35:09 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 12:12:16 +00:00
Wim Taymans
4a3db41f6d gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
2008-05-06 10:16:49 +00:00
Sebastian Dröge
9854bd07f6 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_base_audio_src_[sg]et_slave_method() to the exported
symbols.
2008-05-06 09:59:43 +00:00
Sebastian Dröge
9333eb4899 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 12:33:05 +00:00
Young-Ho Cha
76e3ffb61c gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
2008-05-05 11:14:48 +00:00
Sebastian Dröge
de277a5b2a gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
2008-05-05 10:03:51 +00:00
Edward Hervey
b98072f957 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
It's SorensOn and not SorensEn.
2008-05-05 07:41:03 +00:00
Tim-Philipp Müller
6451cbf5b7 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Fix description of video/x-flash-video.
2008-05-04 15:23:36 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Tim-Philipp Müller
1157de776a tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
2008-05-03 16:00:04 +00:00
Tim-Philipp Müller
005c1c8636 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
2008-05-03 15:45:23 +00:00
Tim-Philipp Müller
ee90cf1969 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
2008-05-03 15:39:04 +00:00
Tim-Philipp Müller
6de5983831 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
2008-05-03 12:09:16 +00:00
Wim Taymans
c6389eec57 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
2008-05-02 12:13:08 +00:00
Wim Taymans
f0f6476aff gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
2008-05-02 12:11:07 +00:00
Stefan Kost
2b843ca69f gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
2008-05-02 11:13:05 +00:00
Thijs Vermeir
32c304ad6f Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
2008-05-02 10:54:51 +00:00
Sebastian Dröge
abbce230e2 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
2008-05-02 10:02:05 +00:00
Tim-Philipp Müller
ea0d78e8e5 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
2008-05-01 19:11:42 +00:00
Tim-Philipp Müller
f8977b9e9e gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
2008-04-30 20:54:56 +00:00
Michael Smith
0947ecf74c ext/theora/theoradec.c: Cool kids don't divide by zero.
Original commit message from CVS:
* ext/theora/theoradec.c:
Cool kids don't divide by zero.
Treat PAR of x:0 as 1:1.
Fixes #530719.
2008-04-30 17:06:45 +00:00
Tim-Philipp Müller
5f6db60a4d gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
2008-04-30 14:37:52 +00:00
Michael Smith
802c45b10b gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
2008-04-28 22:18:49 +00:00
Wim Taymans
7916e386ca gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-28 08:51:38 +00:00
Wim Taymans
5b8afead80 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix the docs about the seqnum compare function, it returns a difference.
2008-04-25 07:37:09 +00:00
Edward Hervey
f75494578e ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_get_device_list): Don't return before freeing up
the allocated structures.
2008-04-24 09:27:35 +00:00
Stefan Kost
2b44c294ff gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Remove obsolete streaminfo code and fix a leak. Fixes #529546
2008-04-24 08:19:35 +00:00
Stefan Kost
e133a5f587 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Revert the event part, that should not go in.
2008-04-23 13:50:34 +00:00
Stefan Kost
6706362116 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c:
Don't leak GstPluginFeatures when filtering.
2008-04-23 13:45:29 +00:00
Stefan Kost
b773ac8cb2 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some logging for cases when grabbing the xv failed.
2008-04-23 08:58:42 +00:00
David Schleef
809f8542cf ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Update Ogg/Dirac muxing.  Removes the weird "KW-DIRAC" bos
packet.  Should conform to what we currently think is the
final Ogg/Dirac muxing spec.
2008-04-22 06:18:04 +00:00
David Schleef
ac9e02c1d9 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Fix typo that causes the overlay keying color to bright green
on a 16-bit display.  Dark grey good.  Bright green bad.
2008-04-22 06:13:43 +00:00
Stefan Kost
c9269a777f ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfsuri.c:
Add  FIXME comment about using uri-list for source and sink.
2008-04-21 13:47:06 +00:00
Sebastian Dröge
9587e5e1c0 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
vaargs functions to gint. Otherwise the fractions will get 0 set
instead of the correct value on big endian systems. Fixes bug #529018.
2008-04-20 11:42:37 +00:00
Sebastian Dröge
fb2dc81a97 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnome_vfs_sink_uri_get_protocols):
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnome_vfs_src_uri_get_protocols):
* ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
(gst_gnomevfs_get_supported_uris):
Get the list of supported URI schemes in a threadsafe way and use the
same list for the source and sink.
2008-04-20 10:17:23 +00:00
Sebastian Dröge
9f83e90909 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
Original commit message from CVS:
* ext/gio/gstgio.c: (_internal_get_supported_protocols),
(gst_gio_get_supported_protocols):
Don't generate a new supported protocols list on each call but cache
it. It's supposed to be static anyway, this way we only leak it once
per process.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_finalize),
(gst_gio_sink_set_property), (gst_gio_sink_get_property),
(gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_finalize),
(gst_gio_src_set_property), (gst_gio_src_get_property),
(gst_gio_src_start):
* ext/gio/gstgiosrc.h:
API: Add "file" properties where one can set a GFile as source/destination.
Add locking to the properties and use gst_element_class_set_details_simple()
instead of a static GstElementDetails struct.
2008-04-20 10:11:54 +00:00
Sebastian Dröge
0c73cdcbc8 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
2008-04-19 20:06:59 +00:00
Sebastian Dröge
b8d03daeab gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
(gst_rtp_payload_info_for_name):
Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
2008-04-19 16:33:24 +00:00
Tim-Philipp Müller
712c625b01 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
Original commit message from CVS:
* configure.ac:
Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
(NB: this only affects compilation of some of the examples).
Remove some configure.ac cruft that's not needed any longer.
2008-04-18 17:10:43 +00:00
Edward Hervey
9e630c23db gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Don't validate the payload if there isn't any.
Fixes #525915
2008-04-18 14:54:01 +00:00
Sebastian Dröge
66bbadadd0 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
Use g_atomic_int_set() instead of gst_atomic_int_set().
2008-04-17 07:33:46 +00:00
Sebastian Dröge
ee1cf43285 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Return NULL instead of a gchar * array with one NULL element if we
don't get any supported URI schemes from GIO.
2008-04-17 07:29:28 +00:00