Commit graph

5951 commits

Author SHA1 Message Date
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
Wim Taymans
8c3b00be4d gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_create), (gst_app_src_set_caps),
(gst_app_src_get_caps), (gst_app_src_set_size),
(gst_app_src_get_size), (gst_app_src_set_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.
2008-05-07 10:38:23 +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
Wim Taymans
1275eda108 gst-libs/gst/app/gstappsink.*: Start some docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
2008-05-05 10:27:45 +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