Commit graph

1282 commits

Author SHA1 Message Date
Wim Taymans
6f3511bfb6 fix some typos
Fix some typos in the doc string of the new
gst_rtsp_options_as_string() method.
2009-01-29 14:00:30 +01:00
Wim Taymans
484a025f6d Add new RTSP message method to set header
Add gst_rtsp_message_take_header() that takes ownership of the passed header
value. This allows us to avoid an allocations and memory copy in some
situations.
API: GstRTSPMessage::gst_rtsp_message_take_header()
2009-01-29 11:55:10 +01:00
Wim Taymans
e8bd8cab41 Add method to serialize RTSP options
Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
string.
API: GstRTSP::gst_rtsp_options_as_text()
2009-01-28 11:48:01 +01:00
Jan Schmidt
63c9ede3d0 Extend and clean up git ignores 2009-01-23 23:16:11 +00:00
Wim Taymans
a7f2540f77 Add more codec ids for RIFF formats
Handle codec ID for various other AAC formats.
Sync the list of possible codec ids with that of ffmpeg.
Fixes #567255
2009-01-23 11:33:29 +01:00
Wim Taymans
26256b95c8 Reset queued_bytes counter when flushing
Set the amount of queued bytes in the internal queue back to 0 when we clear the
queue.
Fixes #567982
2009-01-23 11:11:31 +01:00
Wim Taymans
509f561ef3 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base 2009-01-22 13:12:02 +01:00
Sebastian Dröge
2e8f9921c9 Reduce the number of allocations for creating FFT contexts
Reduce the number of allocations from 2 to 1 for every FFT
context by allocating enough memory for the FFT context
and passing parts of it to the kissfft allocation functions.
2009-01-22 12:54:35 +01:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00
Sebastian Dröge
4d3ff205be gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
Original commit message from CVS:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
Use correct struct alignment everywhere to prevent unaligned
memory accesses, resulting in SIGBUS on sparc and probably others.
Fixes bug #500833.
2009-01-16 11:44:04 +00:00
Sebastian Dröge
98ea758763 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Forward unknown events upstream to allow latency configuration.
Fixes bug #567960.
2009-01-16 11:40:02 +00:00
Jan Schmidt
80ac3b565e gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
Store the returned signal id in the right slot when
registering the pull-buffer signal.
Fixes #567168
Spotted by: Thomas Vander Stichele  <thomas at apestaart dot org>
2009-01-09 23:13:17 +00:00
Tim-Philipp Müller
d629c9fc17 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c:
Small docs addition to clarify that one really mustn't free
the constant GList returned (#566812).
2009-01-09 17:17:50 +00:00
Wim Taymans
1f6297f051 Add GType for GstRTSPUrl and expose a copy function because we can.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
(gst_rtsp_url_get_type), (gst_rtsp_url_copy):
* gst-libs/gst/rtsp/gstrtspurl.h:
* win32/common/libgstrtsp.def:
Add GType for GstRTSPUrl and expose a copy function because we can.
API: gst_rtsp_url_copy()
Fixes #567027.
2009-01-08 17:18:24 +00:00
Sebastian Dröge
ba03cb6080 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Make the GType of GstCDDABaseSrcMode public for bindings.
Fixes bug #566837.
2009-01-07 10:50:15 +00:00
José Alburquerque
7431789249 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes #566723.
2009-01-06 17:30:31 +00:00
Tim-Philipp Müller
ada70bb159 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Make debug categories static. Use _element_class_set_details_simple().
2009-01-06 11:10:29 +00:00
Tim-Philipp Müller
d2b82026c8 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(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_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
(gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
(gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
(gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer)::
* gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
* gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_finalize), (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_is_seekable), (gst_app_src_check_get_range),
(gst_app_src_query), (gst_app_src_do_seek), (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_stream_type), (gst_app_src_get_stream_type),
(gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full),
(gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
* gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
Move private data into a private instance struct. Add padding to
instance and class structures exposed in public headers. Add
Since markers to the gtk-doc blurbs (#566750).
2009-01-06 10:56:45 +00:00
Jan Schmidt
1b2dc5f3a8 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Fix up build flags and include statement for the new generated
enumtypes files, to fix dist.
2009-01-06 10:16:16 +00:00
Jan Schmidt
08393941a8 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* 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/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-app.xml:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* tests/examples/Makefile.am:
* tests/examples/app/Makefile.am:
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
2009-01-05 23:04:57 +00:00
Wim Taymans
0a4c1bc64c gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
2009-01-05 17:13:13 +00:00
Edward Hervey
70a35897fb gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
* gst-libs/gst/tag/gsttagdemux.h:
Add GType for GstTagDemuxResult enum.
2008-12-31 13:31:55 +00:00
Edward Hervey
98ad43fcdd gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
This will help bindings to use it.
2008-12-31 13:01:30 +00:00
Edward Hervey
e2fcc71650 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c:
* win32/MANIFEST:
* win32/common/audio-enumtypes.c:
(gst_audio_channel_position_get_type),
(gst_ring_buffer_state_get_type),
(gst_ring_buffer_seg_state_get_type),
(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
* win32/common/audio-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
* win32/common/multichannel-enumtypes.h:
* win32/vs6/grammar.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs7/libgstaudio.vcproj:
* win32/vs8/libgstaudio.vcproj:
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
audio- in order to wrap all enums declarations of that library.
This modification should not matter since that header file is not a
public header (it will be included by public headers).
Modify win32 crap^Wfiles accordingly.
2008-12-31 11:20:26 +00:00
Edward Hervey
20adaa1328 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Complete Sebastien's commit from the 13th by exporting the
_slave_method_get_type() methods.
2008-12-30 17:55:07 +00:00
Wim Taymans
0ab6c0fbc0 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
Original commit message from CVS:
* 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_query),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
* gst-libs/gst/app/gstappsrc.h:
Add properties and methods to configure and retrieve the min and max
latencies.
2008-12-29 16:45:20 +00:00
Wim Taymans
a579eba73d gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2.  Fixes #564929.
2008-12-20 12:45:03 +00:00
Sebastian Dröge
4ed1f5d6fd gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
Make GstAudioSrcSlaveMethod get_type() function non-static
as it's public now.
* win32/common/libgstaudio.def:
* win32/common/libgstnetbuffer.def:
Add some missing functions to the list of exported symbols.
2008-12-19 13:03:00 +00:00
Andrew Feren
a628077e96 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
Original commit message from CVS:
Patch by: Andrew Feren <acferen at yahoo dot com>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
(gst_netaddress_get_address_bytes),
(gst_netaddress_set_address_bytes):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Make gst_netaddress_get_ip4_address fail for v6 addresses.
Make gst_netaddress_get_ip6_address either fail or return the v4
address as a transitional v6 address.
Add two convenience functions:
API: gst_netaddress_get_address_bytes()
API: gst_netaddress_set_address_bytes()
Fixes #564896.
2008-12-18 12:37:33 +00:00
Wim Taymans
8567ee2149 Add appsrc and appsink documentation.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
Add appsrc and appsink documentation.
2008-12-17 13:51:46 +00:00
Wim Taymans
24685b5df0 examples/app/: Fix example to unref after emiting the push-buffer action.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes #564482.
2008-12-15 12:02:26 +00:00
Sebastian Dröge
04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-13 06:57:09 +00:00
Edward Hervey
c5ae184910 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
2008-12-12 19:41:28 +00:00
Edward Hervey
c4295a07b9 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mapping for VP6 in avi/riff.
2008-12-12 07:15:22 +00:00
Luis Menina
a4493595a6 gst/: Include glib.h instead of a specific GLib header. Including single
Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
2008-12-10 08:19:13 +00:00
Julien Moutte
b7f763b23f gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
Original commit message from CVS:
2008-12-09  Julien Moutte  <julien@fluendo.com>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
2008-12-09 18:30:10 +00:00
Stefan Kost
b3cc87185a gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Fix handling of odd chunks in riff metadata.
2008-12-09 17:21:37 +00:00
Olivier Crete
3c9df39c15 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-08 12:08:32 +00:00
이문형
933186aaa1 gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
2008-12-01 19:36:35 +00:00
Wim Taymans
af354dbef3 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
2008-11-27 16:47:41 +00:00
이문형
d80a5c9dbc gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
A successful gst_poll_wait() doesn't always mean successful connect() on
Windows.  We should check errors by calling gst_poll_fd_has_error().
See #561924.
2008-11-27 11:16:44 +00:00
Wim Taymans
b2004e3d05 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
Fix typo in the docs.
2008-11-25 15:33:30 +00:00
Wim Taymans
6983c1c85b gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-25 10:32:49 +00:00
Stefan Kost
a8264f66c7 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
2008-11-24 20:11:52 +00:00
Stefan Kost
7f937c99d4 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
2008-11-24 12:56:54 +00:00
Michael Smith
77c3f8bb7f gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Fix win32 build. Oops.
2008-11-20 22:06:05 +00:00
Michael Smith
4f04294e45 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Use WSAGetLastError() rather than errno/h_errno on win32.
2008-11-20 21:40:49 +00:00
Michael Smith
eabff64640 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Support WMA Lossless properly.
2008-11-20 21:20:27 +00:00
Jan Schmidt
66ba67723e gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Fix random type causing a docs warning.
2008-11-14 21:39:09 +00:00
Wim Taymans
9c32e1f152 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
(gst_rtp_buffer_get_payload_type),
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
(gst_rtp_buffer_set_timestamp),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
Avoid expensive type checks we already did as part of the
_validate() function that should be called first.
2008-11-13 15:37:40 +00:00