Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps):
* tests/check/elements/audioresample.c:
* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
* tests/check/elements/videorate.c:
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/elements/volume.c:
* tests/check/elements/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
Don't busy-wait in tests; this was causing test timeouts very
frequently when running under valgrind.
Original commit message from CVS:
* gst/tcp/README:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_client_queue_caps),
(gst_multi_fd_sink_client_queue_buffer),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_render):
* gst/tcp/gstmultifdsink.h:
make multifdsink properly deal with streamheader:
- streamheader is taken from caps
- buffers marked with IN_CAPS are not sent
- streamheaders are sent, on connection, from the caps of the
buffer where the client gets positioned to
- further streamheader changes are done every time the client
will receive a buffer with different caps
* tests/check/elements/multifdsink.c: (GST_START_TEST),
(gst_multifdsink_create_streamheader):
add tests for this
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Reinstate limit on channel count. Vorbis does not define the meaning
of > 6 channels, so they're just independent channels. Gstreamer
currently has no mechanism to represent N independent channels.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
Don't arbitrarily restrict channel counts and rate in vorbis.
In terms of effects likely on real-world files, this fixes 96kHz
playback of vorbis.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
value. Fixes g-critical on trying to play back ogg containing
unknown codec.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_create), (group_commit),
(setup_source):
* gst/playback/gstplaybasebin.h:
Make the subtitle detection work from any thread so we don't
deadlock. Fixes#343397.
Original commit message from CVS:
* gst/volume/Makefile.am:
Seriously, it's not *that* hard to get compilation right. Even
a drunk can do it ! Add LIBOIL CFLAGS and LIBS
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_class_init),
(gst_volume_init), (volume_process_float), (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps),
(volume_transform_ip), (plugin_init):
* gst/volume/gstvolume.h:
rewrite the passthrough check, split _int16 and _int16_clamp, fix
another property desc., remove unused param from process function
* tests/check/elements/volume.c: (volume_suite):
reactivate the passthrough test
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_reset),
(gst_visual_sink_setcaps), (gst_visual_sink_event),
(gst_visual_src_event), (get_buffer), (gst_visual_chain):
Handle DISCONT.
Use running time before doing QoS.
Handle mono too.
Original commit message from CVS:
* win32/common/libgstvideo.def:
export gst_video_calculate_display_ratio
* win32/vs6/libgstvideoscale.dsp:
add link to libgstvideo-0.10.lib
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_source_element):
Throw a more comprehensible error for rtsp:// URIs (rather
than erroring out with a negotiation error later on) until
we fix playbin to handle rtspsrc etc.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (raw_caps_factory),
(gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
(gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
(gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
* ext/vorbis/vorbisenc.h:
Multi-channel caps negotiation, so we can do proper multichannel
vorbis encoding, negotiated through audioconvert.
Original commit message from CVS:
* tests/check/elements/adder.c: (test_event_message_received),
(test_play_twice_message_received), (GST_START_TEST),
(adder_suite):
Added check to show that #339935 is fixed with ongoing
adder and collectpads fixes.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (gst_play_base_bin_dispose),
(set_encoding_element), (decodebin_element_added_cb),
(decodebin_element_removed_cb), (setup_subtitle), (setup_source),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
* gst/playback/gstplaybasebin.h:
Add 'subtitle-encoding' property to playbin, so applications can
force a subtitle encoding for non-UTF8 subtitles (#342268).
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
(gst_sub_parse_set_property):
Rename recently-added 'encoding' property to 'subtitle-encoding'
(so it can be proxied by playbin/decodebin in a generic way
with less danger of false positives).
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform can make better decisions on what caps to
negotiate.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_get_type):
Make it easier to copy&paste
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_set_mute),
(gst_volume_class_init), (volume_process_int16), (volume_set_caps),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume):
* gst/volume/gstvolume.h:
Add own debug category, move duplicate code to helper function, fix
property texts, add more comments and prepare ffor liboil-goodness
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
add test for mute and passtrough case, be a bit more verbose to track
failure
* tests/check/generic/states.c: (GST_START_TEST):
catch elements that fail to instantiate
Original commit message from CVS:
* tests/check/pipelines/simple-launch-lines.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisenc.c:
Comment out tests using parse_launch() if core was built without
parsing capabilities.
Original commit message from CVS:
* tests/check/Makefile.am:
Extra bonus points for whoever explains to ensonic that you are meant
to test unit tests thoroughly before commiting them, especially if
you know it's going to break.
De-activated element/adder tests.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
Marking caps conversion issues as GST_WARNING is way too verbose,
Moving them to GST_LOG.
Original commit message from CVS:
* README:
Replace current README (containing the release notes from
some 0.9.x version) with a proper README taken from the core.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
(gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
Improve the errors produced on bad output, including some human
readable description strings.
Handle the (theoretical for ximagesink) case where the XServer
has a different idea about the size required for a particular
frame and gives us too small a memory allocation.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
Improve the errors produced on bad output, including some human
readable description strings.
Handle RGB Xv formats properly by transforming them into our
big-endian caps description.
Use gst_caps_truncate to ensure that we never try and choose a
non-fixed caps in buffer_alloc.
Handle the case where the XServer has a different idea about the size
required for a particular frame and gives us too small a memory
allocation.
Use -1 to indicate 'no image format', because 0 is a valid XServer
image format number.
Put RGB Xv formats at the end of the caps, so that we always prefer
YUV format frames.
Iterate the available Xv Encodings to determine the maximum width and
height, and then return that in our caps.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
When there is only one unfinished pad and it receives an event that
doesn't match our requirements, we need to set alldone=FALSE so that
the fakesink is not removed yet.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
Use gst_type_find_helper_for_buffer() to find the type
of stream from the first packet.
* configure.ac:
Bump requirements to core CVS (needed for vorbis
typefinding to work).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
Else they play perfectly fine with qtdemux.
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* gst/audiorate/gstaudiorate.c:
make more debug catagories static
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (message_received),
(test_event_message_received), (GST_START_TEST),
(test_play_twice_message_received), (adder_suite):
added test case for using element twice, extra bonus points for anyone
who can make these test run reliably
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_chain):
Make work with time-stamped input buffers that do not
have a granulepos in BUFFER_OFFSET_END (like theora
buffers coming from matroskademux). Fixes#342448.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/tcp/Makefile.am:
fdstresstest doesn't need Gtk+, fix compilation if
gtk is not available (#342566).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
On second thought, just skip JUNK chunks automatically, so
the caller doesn't have to handle this. Fixes#342345.
Also, return GST_FLOW_UNEXPECTED if we get a short read,
not GST_FLOW_ERROR.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Don't bail out on JUNK chunks with a size of 0 (would try to
pull_range 0 bytes before, which sources don't like too much).
See #342345.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Use the gstutil scaling function to preserve 64 bits while calculating
output width and height from the display-aspect-ratio. (A continuation
of #341542)
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_buffer_alloc):
* sys/xvimage/xvimagesink.h:
When performing buffer allocations, remember the caps and image format
we return so that if the same caps are asked for next time we can
return them immediately without doing any caps intersections.
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/README:
Some new documentation
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs.
Not enabled in Makefile.am until approved.
Original commit message from CVS:
* tests/check/elements/alsa.c: (test_device_property_probe):
Fix test case: don't try to free NULL GValueArray when there
are no devices.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/alsa.c: (test_device_property_probe),
(alsa_suite), (main):
Add simple test that runs a device property probe on alsasrc,
alsasink and alsamixer. Disable valgrind check for now (too
many leaks in libasound, and valgrind ignored my suppressions
additions).
