Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Refine musepack typefinding a bit. Return MAXIMUM
probability when we detect stream version 7 to make
sure the mpeg audio typefinder doesn't trump us.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_init):
send events from src-pad to all sink-pads fixes#338657
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_remove_client_link):
* gst/tcp/gstmultifdsink.h:
Fix race condition in multifdsink that can lead to spurious
duplicate clients. this patch adds a new signal that is fired when
multifdsink has removed all references to the fd.
Fixes#339574.
Updated documentation.
API: client-fd-removed signal added
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
When asking g_value_array_new to prealloc elements, we may as well
ask for the right number of elements.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_swap_prev), (gst_video_rate_chain):
fix up docs
fix a leak when no caps negotiated
fix counting of input frames
* tests/check/elements/.cvsignore:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(GST_START_TEST), (videorate_suite):
add tests for these
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(resample_set_state_from_caps):
Add support for other formats audioresample can handle such as
32 bits in and float and 64 bits float. Fixes#301759
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
Fix wavpack typefinding to work in more cases (don't peek
for chunks of multiple hundred kBs at once, but process
things step-by-step in smaller units). Fixes#339786.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
* tests/check/Makefile.am:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(setup_videorate), (cleanup_videorate), (GST_START_TEST),
(videorate_suite), (main):
Fix an infinite loop if frames are passed in with wrongly ordered
timestamps. Fixes#339013.
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
fix typefinding on some ISO files. Fixes#339212.
Original commit message from CVS:
Patch by: Jan Schmidt
* gst/playback/gststreamselector.c:
(gst_stream_selector_bufferalloc):
Restore old StreamSelector behaviour.
Fixes#338419.
Original commit message from CVS:
* gst/audioresample/debug.h:
replace debug macros with variable number of parameters
by a simple alias to gstreamer standard debug macros
(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
supported by MSVC 6.0 and 7.1)
* gst/audioresample/resample.h:
define M_PI and rint for WIN32
* win32/common/libgstaudio.def:
* win32/common/libgstriff.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
add new exported functions
* win32/vs6:
update project files
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy):
Clean up our group elements properly in the case where it never
got committed - it still got added unconditionally to the bin.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
Don't remove our mute-probe if someone else already did so.
Don't set a 2nd one if there is already one pending on the pad.
* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
(do_playbin_seek):
When a seek fails, ensure that playbin is still set back to playing.
* gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
(mpeg_ts_type_find), (plugin_init):
Add a typefind function for mpeg-ts streams.
Original commit message from CVS:
2006-04-06 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_reset)
(gst_video_rate_init): Caps-related parameters should not be reset
by a flush -- move their inits to the instance init function.
(gst_video_rate_flush_prev): Don't complain if gst_pad_push
is not OK, just return the result.
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_class_init)
(gst_audio_test_src_get_times): Re-enable is-live=true, as was
broken by Stefan's commit on 24 March.
Original commit message from CVS:
2006-04-04 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
Whoops, fix bug introduced. Bad hacker!
Original commit message from CVS:
2006-04-04 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
Properly handle the case where you get EOS before any buffers are
received. Use gst_buffer_make_metadata_writable where appropriate.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element):
Error out gracefully when we can't create any of the usual
conversion elements for some reason. Also, don't try to
create an audioscale (sic) element that's not used anyway.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Don't post RESOURCE_NOT_FOUND error when we can't find a source
element for a particular protocol, that's confusing for users.
Instead, post a RESOURCE_FAILED error, so that our own error
message is actually shown in totem etc. (#336303).
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (convert_encoding),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
Text subtitle files may or may not be UTF-8. If it's not, we
don't really want to see '?' characters in place of non-ASCII
characters like accented characters. So let's assume the input
is UTF-8 until we come across text that is clearly not. If it's
not UTF-8, we don't really know what it is, so try the following:
(a) see whether the GST_SUBTITLE_ENCODING environment variable
is set; if not, check (b) if the current locale encoding is
non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
the current locale encoding is UTF-8 and the environment variable
was not set to any particular encoding. Not perfect, but better
than nothing (and better than before, I think) (fixes#172848).
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
* gst/adder/gstadder.c: (gst_adder_init):
use DEBUG_FUNCPTR for collectpads
Original commit message from CVS:
Patch by: Julien MOUTTE <julien at moutte dot net>
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
(gst_video_test_src_do_seek), (gst_video_test_src_create):
Partially handle 0 framerate, only EOS after the first frame
is missing.
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
* 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:
Patch for support of YVU9 AVI files (#334822)
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
(id3v1_type_find), (apetag_type_find), (plugin_init):
Can't do tag preferences via probability, as tags would then
lose against types that are recognised with MAXIMUM probability
(like .wav); so let all tag typefinders return MAXIMUM themselves
and order them via the rank. Split ID3v1 and ID3v2 typefinders so
that we can prefer APE to ID3v1 (fixes#335028).
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc):
Preserve the existing buggy streamselector behaviour by performing
a fallback buffer allocation when downstream isn't linked yet.
This should really be fixed in playbin by blocking pads until it's
linked them.
Also, use gst_pad_alloc_buffer instead of
gst_pad_alloc_buffer_and_set.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
Chain up to the parent finalize method.
Add 32-bit sample size to the template caps.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add the fourcc that the VMWare codec uses.
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc),
(gst_stream_selector_request_new_pad):
For the active pad, forward buffer-alloc requests, otherwise
return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
having to memcpy every frame when used by playbin.
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
Get negotiated caps from the sink pad, rather than the sink
pad's peer.
Original commit message from CVS:
Patch by: Sebastien Moutte <sebastien moutte net>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
(gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
(gst_ffmpeg_smpfmt_to_caps):
Replace __VA_ARGS__ caps creation macros with varargs functions.
Makes things compile on MSVC (#320765), looks nicer, and we can
tell the compiler to check for the NULL terminator.
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_init), (gst_video_scale_src_event):
Re-enable QoS after the release.
Rework videoscale to use the base class src_event handler.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
NULL-terminate array of mpeg4 video file extensions.
Fixes crash on PPC (#334226).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
Fix invalid memory access to region before peek'd data (#332964).
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Fix invalid memory access: make sure string passed to
regexec() is NUL-termianted.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mp3_type_find):
Refactor mpeg/audio typefinding to make it more maintainable
and easier to fine-tune. Make probing into middle of the file
work properly (fixes#333900, also see #152688).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(utf8_type_find_have_valid_utf8_at_offset):
Remove part from previous commit that was bogus:
g_utf8_validate() does in fact not accept embedded
zeroes, so we don't need to check for those (thanks
to Mike for the hint).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(utf8_type_find_count_embedded_zeroes),
(utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
Make plain/text typefinder more conservative: firstly, check
for embedded zeroes, which are perfectly valid UTF-8 characters,
but also a fairly good sign that something is not a plain text
file; secondly, probe into the middle of the file if possible.
If we can't probe into the middle, limit the probability value
to be returned to TYPE_FIND_POSSIBLE (see #333900).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Give id3 and ape tag typefinders a rank slightly higher
than PRIMARY to ensure they're always run before any of
the other typefinders (in particular wav and mp3) (#324186).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes#331903)
Handle EOS correctly.
Added docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* 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.signals:
Fix hierarchy, added some more elements to the docs.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_type):
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
Fix docs for ffmpegcolorspace.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (id3_type_find),
(apetag_type_find), (ape_type_find), (plugin_init):
Some typefinding fine-tuning:
- rank ID3/APE tags in order of preference via probabilities, so that
ID3v2 > APEv2 > APEv1 > ID3v1.
- three or four bytes don't really justify MAXIMUM probability,
change those to 'very likely' (musepack and monkeysaudio).
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
added defines of M_PI and M_PI_2
* gst/ffmpegcolorspace/avcodec.h:
removed #include "stdint.h" for win32 as _stdint.h is
autogenerated to win32/common
* win32/common/libgstaudio.def:
* win32/common/libgsttag.def:
added some exports
* win32/vs6:
some project files bugs corrected
* win32/vs7:
project files are reset to the default vs7 configuration
(they link to msvcr71.dll using default optimizations)
Original commit message from CVS:
* gst/videoscale/vs_scanline.c:
(vs_scanline_resample_nearest_RGBA):
Revert optimization in videoscale. It should go in liboil and have
an appropriate liboil function.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
Don't ignore return code from ffmpeg convert function.
* gst/ffmpegcolorspace/imgconvert.c: (img_convert):
Split out some long statements to ease debugging.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
(cmml_type_find), (plugin_init):
Fix CMML type find function to not require a specific minor version
of the CMML header.
Add an MPEG4 video elementary stream typefind function.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_caps_remove_format_info),
(gst_ffmpegcsp_get_unit_size):
The 'palette_data' field from incoming RGB caps shouldn't be
proxied on outgoing YUV caps; also, restrict unit size
adjustment in case of paletted data only to the unit that
actually has a palette. Fixes#330711.
Original commit message from CVS:
Reviewed by : Edward Hervey <edward@fluendo.com>
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(qt_type_find):
Better 3gp typefinding.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
Don't send EOS event here, the base class will send one for us.
* gst/playback/gstplaybasebin.c: (prepare_output):
Subpictures without video stream aren't allowed either.
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Fix debug statement copy'n'paste-o.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
(parse_subrip), (gst_sub_parse_format_autodetect):
Set right caps given that we send escaped text. Also,
honour <i></i>, <b></b> and <u></u> markers that can be found
in .srt files (fixes#310202).
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (get_our_ghost_pad),
(remove_element_chain), (cleanup_decodebin),
(gst_decode_bin_change_state): Make decodebin reusable by
fixing remove_element_chain first and then introduce a
cleaner in state change to ->NULL. (Closes#331678)
------------------------------------------------------
Original commit message from CVS:
2006-02-19 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
should be considered as row.
Original commit message from CVS:
2006-02-18 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_finalize), (gst_text_overlay_init),
(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
(gst_text_overlay_render_text),
(gst_text_overlay_text_pad_link),
(gst_text_overlay_text_pad_unlink),
(gst_text_overlay_text_event),
(gst_text_overlay_video_event), (gst_text_overlay_pop_text),
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
(gst_text_overlay_change_state): Refactoring of textoverlay
without collectpads. This now supports sparse subtitles coming
from a demuxer instead of a sub file. Seeking is still broken
though. Need to discuss with wtay some more on how to handle
seeking correctly.
* ext/pango/gsttextoverlay.h:
* gst/playback/gstplaybin.c: (setup_sinks): Support linking with
subtitles coming from the demuxer.
Original commit message from CVS:
Reviewed by Edward Hervey <edward@fluendo.com>
* gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
C-level optimization of the RGBA nearest neighbour function.
Eventually this might end up in liboil with vectorized versions.
Original commit message from CVS:
* gst/audioconvert/plugin.c: (plugin_init):
Register the GstAudioChannelPosition enum type with the type
system in the plugin_init function, so that it is known before
any element actually makes use of multi-channel stuff. This is
required for example if one wants to be able to deserialise/use
a caps string with channel positions before any pipeline has
been setup and started, like with gst-launch.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_vis_element):
Update vis bin docs.
Move queue after tee so we don't queue video buffers but
audio samples instead. Fixes problems where the video queue
is filled and the audio queue empty.
Original commit message from CVS:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
Revert Andy's newsegment change pending a more correct
fix.
Original commit message from CVS:
:
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(qt_type_find), (plugin_init):
detect more files as 3gp
group and reorder the iso file formats
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
In case we can't find the required number of consecutive
mpeg audio frames to positively identify an MPEG audio
stream, check if there's at least a valid mpeg audio
frame right at offset 0 and if so suggest mpeg/audio
caps with a very low probability (#153004).
Original commit message from CVS:
2006-02-07 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
a TIME segment if we get timestamped buffers. Requires recent
fixes in core to work properly.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (prepare_output):
Don't print the URI as part of the error message, it
makes error dialogs look rather ugly, especially if
the URI is very long or has characters in it that
need escaping.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (prepare_output):
Error out if we have only text or subtitles, but nothing
else. Also error out if we have subtitles but no video
stream.
Original commit message from CVS:
2006-02-06 Julien MOUTTE <julien@moutte.net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
Stick to seeking theory until i find the bug.
* gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Don't put essential function call into
g_return_*() macro, otherwise it'll all be
replaced by NOOPs when compiling with
G_DISABLE_CHECKS defined.
Original commit message from CVS:
* gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
Make test work again by connecting fakesinks to each decoded pad,
which makes the pipeline wait until each fakesink has a buffer
queued before going to PAUSED state. At that point we know the
decodebin pads are negotiated.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (au_type_find),
(paris_type_find), (ilbc_type_find), (plugin_init):
Fix typefinding for audio/x-au, audio/x-paris and
audio/iLBC-sh. We cannot use the START_WITH macros
here, because there can only be one typefind factory
with the same name (caps), so the second one would
replace the first one and the first one would never
be called when doing typefinding (see #161712).
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_create_sine_table), (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
initialize gst_controller before using
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Patch from Eric Jonas to support conversions to/from UYVY
(Fixes: #324626)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
Original commit message from CVS:
* gst/videoscale/vs_scanline.c: Oops, *that's* why I never
checked in this change -- it requires liboil features not
in 0.3.6. Revert parts.
Original commit message from CVS:
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_base_init), (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- a library should not call setlocale. see Libraries node in
gettext manual
- make sure all plugins that use translation do bindtextdomain
to point to the localedir
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
(setup_sinks), (plugin_init):
all this, and check for NULL when creating sinks
Original commit message from CVS:
2006-01-27 Julien MOUTTE <julien@moutte.net>
* gst/subparse/gstsubparse.c: (gst_subparse_type_find),
(plugin_init): Make typefinding of subtitles work again.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(wavpack_type_find), (m4a_type_find), (ircam_type_find),
(plugin_init):
Backport a bunch of typefinding fixes from the 0.8 branch.
Also, improve wavpack typefinding: if we can't peek the
entire wavpack block, try to parse the bits we can get and
see if we find what we're looking for in those.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (pad_probe):
Also consider the flush-start and tag events as unblockers
for the pad probes.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybin.c: (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
On the fly visualisation switch, works disabling, enabling as
well but it won't be able to enable vis in a playbin that was
created with no visualisation.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(free_pad_probes), (remove_fakesink), (pad_probe),
(close_pad_link), (gst_decode_bin_change_state):
Replace GstPadBlockCallback with pad probes that detect
first buffer AND eos before removing fakesink.
Fixes hang with demuxers doing EOS while pre-rolling.
Solves #328279
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property):
Comment out broken code that connects to the state-changed signal.
At this point, changing current stream selection is broken, but
stuff like gst-launch playbin current-audio=1 works and filters
to the chosen stream.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Fix playback for sources that emit raw audio or
raw video streams (e.g.: cd audio sources) (#325984).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(probe_triggered), (new_decoded_pad), (mute_group_type),
(set_active_source):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init),
(gst_stream_selector_set_property),
(gst_stream_selector_request_new_pad):
Reenable stream selection. These mechanisms need a complete overhaul
in the face of 0.8->0.10 changes though.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
(gst_audio_rate_change_state), (plugin_init):
Add debugging category.
Fix type issues.
Add case for incoming buffers without valid offset/offset_end.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Don't leak an autoaudiosink/alsasink when we generate
a new audio element. (old code, I guess)
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Support float audio in audiorate.
Use width rather than depth for selecting sample width.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.h:
Use GLib types here (that way we don't have to include the
generated _stdint.h header, which makes life easier for win32
folks that don't use autotools for the build) (#325990, patch
by: Sergey Scobich).
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
Original commit message from CVS:
* configure.ac:
* gst/volume/Makefile.am:
* gst/volume/demo.c:
move old example to tests/examples/volume/volune.c
* tests/examples/Makefile.am:
* tests/examples/seek/seek.c: (main):
change window-close event from "delete-event" to "destroy"
* tests/examples/volume/Makefile.am:
* tests/examples/volume/volume.c: (value_changed_callback),
(setup_gui), (message_received), (eos_message_received), (main):
fix event handling and bus usage
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad):
Fix non-C89 variable declaration not at the start of a block. Should
help some compilers.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_start):
Add start method to reset running time and number of frames sent
when starting up (fixes#324696; patch by: Michal Benes).
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
(gst_ogg_demux_activate_chain):
Extra debug output when activating/deactivating chains.
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(is_demuxer_element), (try_to_link_1), (remove_element_chain),
(unlinked):
Remove a queue from our list when it becomes unlinked.
Don't add queues to elements in class 'Demux' if they
can only produce one pad
Original commit message from CVS:
* ext/libvisual/visual.c: (make_valid_name):
change some char* into char[]
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
prepare to handle EOS and SEGMENT_DONE
Original commit message from CVS:
* gst/tcp/gsttcp.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
Add <string.h> includes for memset and FD_ZERO (fixes#323878;
patch by: Benjamin Pineau).
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
(gst_video_rate_chain):
Fix timestamping for videorate when the first buffer it sees has a
non-zero timestamp. Fix some misleading debug output.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c: (handoff):
Make sure the video frame buffer we return to apps via the
"frame" property always has caps set on it. Modify
_gst_gvalue_set_object() macro to handle NULL objects
gracefully too.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Adjust to some recent api changes and add wtays new cool seeking
capabillities
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_sub_parse_init),
(gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
(parser_state_init), (handle_buffer), (gst_sub_parse_chain),
(gst_sub_parse_sink_event), (gst_sub_parse_change_state):
Implement some sort of event handling that doesn't rely on
g_return_if_fail; make sure we always push the last chunk of an
.srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
state change function; remove some old cruft. Seeking is still
rather unlikely to work though.
* tools/.cvsignore:
Ignore more.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Work around refcount problem with g_value_set_object() that occur
if the core has been compiled against GLib-2.6 (g_value_set_object()
will only g_object_ref() the element, but the caller will
gst_object_unref() it and bad things will happen due to the way
GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
totem for people on FC4 using Thomas's 0.10 RPMs.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_stream_selector_chain):
3rd time's the charm. Correct ref-counting for discarded buffers.
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_dispose), (gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
* gst/playback/gststreamselector.h:
Add the active-pad property for playbin to use shortly. Ignore buffers
from any other pad, returning GST_FLOW_NOT_LINKED
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (queue_filled_cb):
Better use of the queues. Start with a small size queue and only increase
the size of the queues when the other queues are empty.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit), (probe_triggered):
* gst/playback/gstplaybasebin.h:
Prepare to handle errors betters.
* gst/playback/gstplaybin.c: (add_sink), (setup_sinks):
Set sinks to PAUSED first before adding and linking them so that
we don't interrupt dataflow.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (close_pad_link), (try_to_link_1):
Remove unused properties, and add queues between demuxers and decoders
so that a lot more files can preroll properly.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
remove silly include
* gst/tags/Makefile.am:
* gst/tags/gsttagediting.c:
* gst/tags/gsttageditingprivate.h:
* gst/tags/tagedit.vcproj:
remove directory, is as good as empty
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audiorate_chain):
Properly return GstFlowReturn from gst_pad_push in chain functions.
Original commit message from CVS:
2005-11-24 Andy Wingo <wingo@pobox.com>
* gst/tcp/gstmultifdsink.c (gst_multifdsink_handle_client_write):
Be threadsafe.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_chain):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
Use utility method for scaling clocktime for fractional framerates.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_caps_remove_format_info):
* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
Forward-port fixes from the 0.8 branch (patch by Luca Ognibene,
#318353); use gst_structure_has_name().
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(mpeg2_sys_type_find), (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find):
Terminate vararg functions with NULL instead of 0 to
make gcc4 happy.
Original commit message from CVS:
2005-11-21 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybin.c (gen_audio_element)
(gen_video_element): Use the new MISSING_PLUGIN core error
category.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
* gst/adder/gstadder.c: (gst_adder_init),
(gst_adder_request_new_pad), (gst_adder_collected),
(gst_adder_change_state):
Update for gst_collectpads_foo() to gst_collect_pads_foo()
API change.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
Remove obsolete vorbistag element and debug category.
* gst/playback/gstplaybasebin.c: (check_queue):
Don't divide by 0 when queue-threshold is 0.
* sys/ximage/ximagesink.c: (gst_ximagesink_set_property):
Don't modify an existing pixel-aspect-ratio if we fail to read
a new one.
Original commit message from CVS:
2005-11-18 Julien MOUTTE <julien@moutte.net>
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_fixate_caps): Introduce back caps fixate with
handling of PAR.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Unsetting IS_SINK flag from the fakesink, so decodebin
never behaves as a sink.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_client_queue_data),
(gst_multifdsink_render):
Don't leak GDP headers when using GDP mode (i.e. tcpserversink).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
Use autoaudiosink, it tends to be more widely available than
autoaudiiosink.
Original commit message from CVS:
2005-11-14 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybin.c (gen_audio_element): Use autoaudiosink
as well if it is available. Fixes#316442.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_init),
(gst_videotestsrc_src_fixate):
move fixation to a fixate function
remove negotiate function, basesrc's is good enough
fixes a bug for check when using the element alone
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_get_palette), (gst_ffmpeg_set_palette),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size), (gst_ffmpegcsp_transform):
Make palettes work again (see #132341). Use our own macros
for rounding up.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit), (new_decoded_pad),
(setup_substreams), (set_active_source):
Unlock GROUP_LOCK in failure cases, so that we don't deadlock when
trying to go to NULL if we failed to read a file.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_class_init), (gst_audiotestsrc_get_times),
(gst_audiotestsrc_create):
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_times), (gst_sinesrc_create):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_get_times),
(gst_videotestsrc_create):
The base class can now sync for us.
Original commit message from CVS:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_query_types), (gst_sinesrc_src_query),
(gst_sinesrc_newsegment):
Send newsegment event in TIME format, set duration if
num-buffers is set, fix duration querying.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/volume/gstvolume.c: (volume_set_caps):
Fix compilation on Solaris with Forte. (#320923)
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (remove_fakesink),
(pad_blocked), (close_pad_link), (new_pad), (no_more_pads):
Handle the case where a pad_block failed.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
(gst_adder_change_state):
Fix timestamps and fix deadlock when stopping the collectpads.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
When clearing an audioconvert context, set tmpbufsize to zero, so
we'll allocate it again later if required.
This fixes audioconvert re-negotiating formats, which previously
segfaulted with a NULL destination buffer.
Original commit message from CVS:
2005-10-24 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
And
here comes my change on caps for framerate and geometry range.
We are now accepting 1 to MAXINT for width and height, and from
0.0 to MAXDOUBLE for framerate. That allows duration less png
frames
to be blended correctly in videomixer.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (pad_unblocked),
(pad_blocked), (close_pad_link), (new_pad):
Don't try to remove elements twice.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types),
(gst_vorbisenc_src_query):
Implement position and duration queries.
* gst/playback/test3.c: (update_scale), (main):
Fix for async state changes and print nicer output.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
Don't use functions for position queries when handling
duration queries.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(gst_play_base_bin_change_state):
Fix leak.
Handle case where playbasebin is now ASYNC because
decodebin is.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(xml_check_first_element), (xml_type_find), (smil_type_find),
(plugin_init):
Add typefinding for SMIL and for generic XML. Based on patch by
Akos Maroy (#308663).
Original commit message from CVS:
2005-10-17 Andy Wingo <wingo@pobox.com>
* gst/tcp/gstmultifdsink.c: Convert to use the boilerplate macro.
* gst/tcp/gsttcp.c (gst_tcp_socket_read): Comment update.
Original commit message from CVS:
2005-10-17 Julien MOUTTE <julien@moutte.net>
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size): We are asked to compute a buffer
size
from caps, let's use the caps...
Original commit message from CVS:
2005-10-16 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin.c
(gst_element_set_state_like_a_crazy_man): New kraaaaaaazy
function!
(try_to_link_1): Increase kraziness level.
Original commit message from CVS:
- Don't use non-portable LL suffix on constants, since MSVC doesn't allow
them. These constants all fit into ints anyway.
- Continue to hate nano.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.
* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
(do_playbin_seek), (gst_play_bin_send_event):
Override send_event differently, so that we can takes bits of
functionality from GstPipeline (special handling for seeks,
including pausing/resuming, and resetting stream time) and
still get
the appropriate behaviour of only forwarding event to a single
sink,
rather than all of them.
Unfortunately requires a lot of code duplication, but the
alternatives are equally ugly in the end.
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
Alloc temp storage somewhere else where we can do it more
portable.
Original commit message from CVS:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_create),
(gst_tcpserversrc_start):
Don't block in accept while doing the state change, move
to poll and make cancellable.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
(plugin_init):
Add wavpack and spc typefind functions from 0.8 branch.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (tar_type_find),
(ar_type_find), (msdos_type_find), (plugin_init):
Add typefind functions for tar archives, ar archives,
RAR archives, and msdos-executables (dlls, exe, etc.).
Some of those would be wrongly identified as mpeg
streams of some sort before (#315550).
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),
(gst_adder_init), (gst_adder_request_new_pad),
(gst_adder_change_state):
Add query function to source pad, so adder reports the correct
time/sample position when queried (#315457); fix state change
function; use GST_DEBUG_FUNCPTR() for pad functions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
Fix leaks in typefind registration
Clean up the gratuitous commenting and whitespacing a little
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_finalize), (multifdsink_hash_remove),
(gst_multifdsink_stop):
Fix crasher when going to NULL multiple times.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (remove_groups), (setup_source):
* gst/playback/gstplaybin.c: (remove_sinks), (add_sink),
(setup_sinks), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Set state to NULL before removing from bin. Fix refcounting.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Correct refcounting in send_event() function. Previously was wrong
if the first sink was unable to handle the event.
Original commit message from CVS:
2005-10-03 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin.c (try_to_link_1)
(remove_element_chain): set element to NULL before removing it.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
and unlocking.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
Actually add the pad template.
(gst_tcpclientsink_get_type): We're a base sink. Woot, works.
* gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
I'm at it...
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsrc.c: Make interruptable -- code stolen
from fdsrc. Get caps in create() instead of start() so it can be
interrupted. Interruption somewhat untested.
* gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
Proper EOS handling.
Original commit message from CVS:
2005-09-27 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpserversrc.c:
* gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcp.c (gst_tcp_read_buffer): New function, factored
out of tcpclientsrc.c. Cancellable.
(gst_tcp_socket_read): Made private, cancellable, with better
diagnostics. Also the FIONREAD ioctl takes a int*, not a size_t*.
(gst_tcp_gdp_read_buffer): Made cancellable, actually returns the
whole buffer, and better diagnostics.
(gst_tcp_gdp_read_caps): Same.
* gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
Original commit message from CVS:
2005-09-26 Andy Wingo <wingo@pobox.com>
* gst/sine/gstsinesrc.h:
* gst/sine/gstsinesrc.c: Refactor, remove the table lookup code,
change the 'sync' property to 'is-live' and implement it halfway,
update for controller api change.
* gst/volume/gstvolume.c (volume_transform_ip): Update for
controller api change.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_send_event):
Only seek on one sink, the first one that succeeds.
Original commit message from CVS:
2005-09-21 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
thingies.
* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
can be called multiple times, dogs.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: free plugin list correctly
* gst/playback/gstplaybin.c: emit warning if autovideosink
and autoaudiosink can't be found (instead of segfaulting)
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc. Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
Cleanups, speedups, simplifications, added back support
for 24 bits.
Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
Original commit message from CVS:
* configure.ac:
In the output at the end, don't show the first plugin on the same
line as "Core plug-ins, always built:".
Indent the output as for other plugin categories
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
#define that can be used to not use peer buffer_alloc functions for
test purposes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_ximage_new),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
(gst_xvimage_buffer_get_type), (gst_xvimagesink_setcaps),
(gst_xvimagesink_show_frame):
Error case handling fixes. gst-launch fakesrc ! x[v]imagesink now
fails gracefully instead of XError aborting or deadlocking.
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
