Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
Push tag event after the newsegment event. Log the pointer of
the buffer we're actually going to push rather than the buffer
we're feeding to _make_metadata_writable().
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Comment out a couple of other things which break the build when
GST_DISABLE_DEPRECATED isn't on but -Werror is.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Clear the addrinfo struct using memset. Fixes#514937.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Use gmtime_r if available as gmtime is not MT-safe.
Fixes bug #511810.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes#511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Also remove the conditional registration of the signals
that disappeared with the ABI change in 0.10.14
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert patch to gstrtspconnection.c for brown paper bag
release of -base. Re-opens: #511825
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Include Winsock2.h for VS6 and use a different way initialize
hints structure so it can build with VS6.
* win32/MANIFEST:
* win32/vs6/libgstsdp.dsp:
* win32/common/libgstsdp.def:
Add new files for libgstsdp.
* win32/vs6/grammar.dsp:
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstvolume.dsp:
Add new dependencies to the link list.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
Add test to see if hstrerror is available or if we need libresolv
(Solaris) for it, then use it in libgstrtsp.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_context_copy),
(gst_install_plugins_context_get_type):
* gst-libs/gst/pbutils/install-plugins.h:
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
for bindings.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
Post an error message if we can't pull as many bytes as we need
for the tag. This makes sure the user gets to see a proper error
message if a file with a partial ID3 tag is fed to decodebin, and
not a 'no ID3 tag demuxer' error, which would be confusing
(see #508138).
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/pbutils.h:
Use glib-enum generator to have a proper enum GType for
GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add Location header so that we can start implementing redirects.
See #506025.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add new GstVideFormat enum and write a bunch of helper functions
based around it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Yet another fix for broken software that produce files with an empty
blockalign field. Instead of completely failing, make a second attempt
at guessing the width/depth by looking at strf->size.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
for jpeg video streams.
Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
for the above modification.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
(gst_x_overlay_handle_events):
More guards (we don't want klass to end up being NULL).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_free):
Close control sockets. Fixes#503440.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description for 'private' dts caps (who come up with that name?).
Original commit message from CVS:
* Makefile.am:
Add check-exports target and run it with 'make check'.
* configure.ac:
Be stricter about what we export in our libraries: change regexp so that
we only export _gst_foo(), but not __gst_foo().
* gst-libs/gst/cdda/base64.h: (rfc822_binary):
* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
Change internal functions to __gst_foo so they dont' get exported.
* win32/common/libgstaudio.def:
Add missing symbols.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
No need for floating point operations here. avoids having to link
against the math library too.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats. Generate ADPCM description
dynamically depending on layout/format.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes#498767.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
