Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
do not update controlled params, if buffer has no timestamp
Original commit message from CVS:
* configure.ac:
* gst/sine/Makefile.am:
* gst/volume/Makefile.am:
controllerized elements also need to link against controller-libs ;)
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstgconf.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_create):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
controllerized two audio plugins
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
Use realloc else we lose our original data.
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
Original commit message from CVS:
* check/Makefile.am:
* configure.ac:
add core's plugins to the mix so that playbin works
* check/generic/states.c: (GST_START_TEST):
set a 0 timeout on pipelines, so they don't force the next
state change
* gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
(gst_play_base_bin_change_state):
remove the crappy error handling and do GST error handling
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
(gst_videoscale_prepare_size), (parse_caps),
(gst_videoscale_set_caps), (gst_videoscale_get_size),
(gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
(gst_videoscale_transform):
* gst/videoscale/gstvideoscale.h:
Refactor, make use of BaseTranform really well.
Original commit message from CVS:
* check/Makefile.am:
Add CHECK_CFLAGS and LDFLAGS
* gst/playback/gstplaybasebin.c: (fill_buffer):
GST_MESSAGE_SRC became a GObject
Original commit message from CVS:
* gst/playback/gstplaybin.c: (remove_sinks):
Remove visualization from parent explicitely; works around some
apparent refcount issue that I haven't tracked down yet.
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
Original commit message from CVS:
2005-08-04 Andy Wingo <wingo@pobox.com>
* gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c
(gst_ffmpegcsp_get_size): Adapt to API changes.
* gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
Implement an in-place do-nothing transform.
Original commit message from CVS:
* configure.ac:
When testing for X libs, use the X CFlags
* gst/adder/gstadder.c: (gst_adder_change_state):
Stop the collectpads before calling parent state change function
on PAUSED->READY, otherwise we deadlock deactivating pads.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Switch to auto*sink elements as default sinks; add volume element
so that volume control in totem works.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
* gst/playback/gstplaybin.c: (setup_sinks),
(gst_play_bin_change_state):
Refcount fix and more comments.
Original commit message from CVS:
2005-07-20 Andy Wingo <wingo@pobox.com>
* gst/videoscale/vs_image.c (vs_image_scale_nearest_YUYV): Typo
fix (?), fixes a seggie mcfalterson (#310894).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer):
Use _new_custom() so we can set custom message types for buffering
messages.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_free):
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_init),
(gst_multifdsink_add), (gst_multifdsink_remove),
(gst_multifdsink_clear), (gst_multifdsink_get_stats),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_caps),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
(gst_multifdsink_stop):
* gst/tcp/gstmultifdsink.h:
0.8 backporting.
* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
Also draw image when not from a pool.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
Small debug additions.
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.
* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
Original commit message from CVS:
2005-07-05 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybasebin.c (fill_buffer):
message_new_application fixen.
x
Original commit message from CVS:
2005-07-04 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
Original commit message from CVS:
2005-07-01 Andy Wingo <wingo@pobox.com>
* ext/theora/theoradec.c (theora_dec_src_getcaps): Implement a
getcaps to do explicit caps. Needs to be done in all decoders,
possibly via a base class.
* configure.ac (GST_PLUGIN_LDFLAGS): Add videoscale.
* ext/ogg/gstoggdemux.c (gst_ogg_pad_typefind): No need to set
caps on the sink pad, just rely on the pad template. Also, setting
ANY caps on a pad is not valid because the caps are not fixed.
* sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc): Set the
caps on the buffer, and get the width from the desired_caps if
they're set.
(gst_ximagesink_renegotiate_size): Implement via setting the
desired_caps on the ximagesink.
(gst_ximagesink_setcaps): Only reset the width of the player if it
wasn't already set. Not sure if this is right.
(gst_ximagesink_show_frame): Memcpy only for normal buffers.
* sys/ximage/ximagesink.h (desired_caps): New field, is the caps
that the user wants. NULL unless the window has been resized.
* gst/volume/gstvolume.c (volume_transform): Adapt to
basetransform refcount changes.
Original commit message from CVS:
2005-07-01 Andy Wingo <wingo@pobox.com>
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h: Clean up, port to 0.9. Derives
from BaseTransform, implements a transform_caps. Removed dead code
including some PAR stuff that was never reached -- should probably
be added back somehow.
Original commit message from CVS:
2005-06-27 Andy Wingo <wingo@pobox.com>
* gst/videotestsrc/gstvideotestsrc.c
(gst_videotestsrc_activate_push): Activation API changes.
* gst/playback/gstdecodebin.c (gst_decode_bin_change_state)
(gst_decode_bin_dispose): Free dynamics in READY->NULL, because
they have refs on the decodebin.
* ext/ogg/gstoggdemux.c (gst_ogg_pad_class_init): Ref the right
parent class.
(gst_ogg_pad_typefind): Don't leak a pad ref.
(gst_ogg_chain_new_stream): gst_object_unref, not g_object_unref.
(gst_ogg_demux_sink_activate, gst_ogg_demux_sink_activate_push)
(gst_ogg_demux_sink_activate_pull): Changes for activation API.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
2nd argument of 'unknow-type' signal is a GstCaps and not a
GstMiniObject
Original commit message from CVS:
2005-06-25 Jan Schmidt <thaytan@mad.scientist.com>
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Set the worker thread's running flag to TRUE before starting the
thread.
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Catch a failure to add typefind to the bin.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
(gst_videorate_getcaps), (gst_videorate_setcaps),
(gst_videorate_event), (gst_videorate_chain):
Fixed videorate, fixating an already fixated caps is not
an error.
Original commit message from CVS:
2005-06-02 Andy Wingo <wingo@pobox.com>
* pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
* pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
list.
* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Don't
remove the typefind, the bin dispose will do it for us. When it's
removed and unreffed, the signal handler will be disconnected,
too.
(unlinked): It's too difficult to disconnect from unlinked
handlers, as they are on pads not elements. Just punt if the pads
aren't grandkids of the bin.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
(gst_ffmpegcsp_chain), (gst_ffmpegcsp_change_state):
No need to take the STREAM lock anymore.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (probe_triggered):
Fix missing unlock.
* gst/playback/gstplaybin.c: (add_sink):
First add, then link (otherwise pad link fails).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
Increase buffer for video, decrease buffer for other media types.
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Change names for debugging purposes.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
(gst_ffmpegcsp_chain):
Enable buffer alloc passthrough if the source and dest
formats are the same.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(fill_buffer), (check_queue), (queue_threshold_reached),
(queue_out_of_data):
* gst/playback/gstplaybasebin.h:
Post buffer-fullness on the bus.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
(gst_play_base_bin_class_init), (gst_play_base_bin_finalize),
(get_active_group), (get_building_group), (group_destroy),
(group_commit), (check_queue), (queue_overrun),
(queue_threshold_reached), (queue_out_of_data),
(gen_preroll_element), (remove_groups), (unknown_type),
(add_element_stream), (no_more_pads), (probe_triggered),
(preroll_unlinked), (new_decoded_pad), (setup_subtitle),
(setup_substreams), (setup_source), (finish_source),
(prepare_output), (muted_group_change_state),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_change_state):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_set_property),
(gen_video_element), (gen_text_element), (gen_audio_element),
(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
(gst_play_bin_change_state):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
(cb_probe), (gst_stream_info_new), (gst_stream_info_dispose),
(stream_info_change_state), (gst_stream_info_set_mute),
(gst_stream_info_get_property):
* gst/playback/gststreaminfo.h:
* gst/playback/gststreamselector.c: (gst_stream_selector_init),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
* gst/playback/gststreamselector.h:
Rough port of playbin. Needs some more work, but is mostly done,
and uses a few locks in important places, which should make stuff
like chain-switches clean. Still uses GST_STATE() in a few places,
which isn't all that good an idea, subtitles/elements disabled
because no elements to test with and thus probably broken, query
and event handling moved to GstBin, internal thread removed
alltogether because the pipeline does that for us now. Can play
Ogg/Vorbis files. Haven't tested anything else yet.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_handle_comment_packet): Post a
message to the bus with the tags. Still not sent downstream tho.
* gst/playback/gstdecodebin.c (remove_element_chain): Unref after
get_parent.
(remove_element_chain): Use OBJECT_PARENT instead of get_parent to
avoid refcounting hassles.
Original commit message from CVS:
2005-05-09 Andy Wingo <wingo@pobox.com>
* gst/volume/Makefile.am:
* gst/volume/demo.c
* gst/volume/gstvolume.h
* gst/volume/gstvolume.c: Port to 0.9 API, derive from
basetransform. Probably need an audio filter base class.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
Make caps writable before writing to it.
Fix negotiation in audioconvert some more.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_getcaps), (gst_videorate_setcaps),
(gst_videorate_blank_data), (gst_videorate_init),
(gst_videorate_event), (gst_videorate_chain),
(gst_videorate_change_state):
Port videorate, do a better job at negotiation while we're at
it.
Original commit message from CVS:
* configure.ac: Require liboil.
* gst/videotestsrc/gstvideotestsrc.c: Fix up liboil calls, add
a few more.
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
Well, unreffing a buffer right before pushing it is asking
for trouble..
Original commit message from CVS:
2005-05-05 Andy Wingo <wingo@pobox.com>
* gst/audiorate/gstaudiorate.c (gst_audiorate_class_init): Pacify
GObject.
* configure.ac: Return audiorate and subparse from the ghetto.
Re-enable -Wall -Werror.
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h: Port to 0.9. Can operate loop-based
or chain-based. Cleaned up a bit. Not tested.
Original commit message from CVS:
* gst/adder/Makefile.am:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init), (gst_adder_init),
(gst_adder_request_new_pad), (gst_adder_collected),
(gst_adder_change_state):
* gst/adder/gstadder.h:
Ported adder as an example of a mixer element using
collect pads. Needs more negotiation work.
Original commit message from CVS:
* examples/dynparams/Makefile.am: Move demo-dparams from gst/sine
to examples/dynparams. Examples do not belong interspersed with
source code.
* examples/dynparams/demo-dparams.c:
* gst/sine/Makefile.am:
* gst/sine/demo-dparams.c:
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter):
* gst/playback/gstdecodebin.c: (find_compatibles):
Work with staticpadtemplates in elementfactories.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
Declare variables at beginning of block and make gcc-2.95 happy
(fixes # 167482, patch by Gergely Nagy).
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpclientsrc.h:
Move some includes into the header, so that struct sockaddr_in is
defined when it should be defined on FreeBSD as well (fixes
#167483).
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
Don't pass uninitialised values to setsockopt() here either.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.h:
include missing header file
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_compatible):
use same sign for both channels when converting to/from compatible
channel. Previously used different signs made the signals cancel
each other out and appear like silence. (fixes#167269)
Original commit message from CVS:
Include "_stdint.h" instead of <stdint.h>. Fixes build on systems that do not have stdint.h, like Solaris 9 (fixes#166631).
Original commit message from CVS:
2005-02-04 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate): New function, fixates to 256
frames per buffer by default. (Much better than 1.)
(buffer_frames_convert_init): Set the fixate function for both src
and sink pad.
(buffer_frames_convert_link): After success setting nonfixed caps,
get the negotiated caps so we can know how many buffer-frames it
will be. No idea how this worked at all before.
Original commit message from CVS:
* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
(gst_tcpclientsink_finalize):
* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
(gst_tcpclientsrc_finalize):
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
(gst_tcpserversink_init), (gst_tcpserversink_finalize):
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
(gst_tcpserversrc_init), (gst_tcpserversrc_finalize):
Don't leak the hostname when shutting down.
In tcpserversrc, take a copy of the default hostname.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_chain):
D'oh, reference the palette data, not the palette structure.
Fixes color distortion in #132341.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
PAR can be non-fixed when not provided as argument (#162626).
