Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Don't crash by dividing by zero (see sample movie in #126922).
Original commit message from CVS:
* ext/dvdnav/README:
Update the README to use dvddemux
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
Ensure getcaps returns a subset of the template caps
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_base_init),
(gst_mpeg2subt_init):
Ensure getcaps returns a subset of the template caps
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_video_stream),
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_set_cur_subpicture):
* gst/mpegstream/gstdvddemux.h:
Set the explicit caps on the current_video pad before pushing
anything
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream):
Free caps used to gst_pad_set_explicit_caps, which takes a const
GstCaps *
Original commit message from CVS:
reviewed by Benjamin Otte <otte@gnome.org>
* gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init):
create a NULL-initialized array of pads, so we don't think they
exist already. (fixes#143130)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
(gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
* gst/qtdemux/qtdemux.h:
Bitch. Also known as seeking, querying & co.
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
(gst_osssink_change_state):
* sys/oss/gstosssink.h:
Resyncing is for weenies, this hack is no longer needed and was
broken anyway (since it - unintendedly - always leaves resync to
TRUE).
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxaparse.h:
some renaming
add some checks/sanity
prepare for seek addition
* sys/sunaudio/gstsunaudio.c:
remove exported dupe init function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header): Patch from dcm@acm.org (David Moore)
to allow qtdemux to use non-seekable streams. (bug #142272)
Original commit message from CVS:
* configure.ac: Add sunaudio
* examples/Makefile.am: make gstplay depend on gconf
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette),
(convert_table_lookup), (img_convert): remove c99-isms
* gst/ffmpegcolorspace/imgconvert_template.h: make a constant
unsigned, to fix a warning on Solaris
* gst/mpeg1sys/systems.c: bcopy->memcpy
* gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy
* sys/Makefile.am: Add sunaudio
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain): Fix crash when ESD
is killed while we're playing.
* gst/qtdemux/qtdemux.c: (qtdemux_parse): call
gst_element_no_more_pads().
Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)
* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)
* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
Original commit message from CVS:
* ext/audiofile/gstafsrc.c: (gst_afsrc_get):
Remove old debug output
* ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
(gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
(gst_dvdec_set_property), (gst_dvdec_get_property):
Change the quality setting to an enum, so it works from gst-launch
Don't renegotiate a non-linked pad. Allows audio only decoding.
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
(gst_deinterlace_link), (gst_deinterlace_init):
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link):
Some caps negotiation fixes
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c :
Add mpegversion to CAPS to make it link
Rank is as GST_RANK_SECONDARY instead of NONE
Original commit message from CVS:
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_get_type):
whoever that was: DO NOT IMPORT PRIVATE SYMBOLS THAT ARE NOT IN
HEADERS. Had to be said.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_type_get),
(qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_dump_co64),
(qtdemux_dump_unknown), (qtdemux_parse_tree), (qtdemux_parse_udta),
(qtdemux_tag_add), (get_size), (gst_qtdemux_handle_esds): More qtdemux
hackage -- parse a lot more atoms, extract a few tags. One might even
mistake this for tag support. Maybe it is.
* gst/qtdemux/qtdemux.h:
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
(qtdemux_parse_trak), (get_size), (gst_qtdemux_handle_esds): Hacked
up qtdemux to make it spit out codec_data. Do _not_ look at this
code; you will no longer respect me.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpegdec_get_type):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
(gst_jpegenc_getcaps):
move format setting to inner loop
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcolorspace_getcaps):
use GST_PAD_CAPS if available so that we use already negotiated
caps
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_moov), (qtdemux_parse):
extra debugging
* sys/qcam/qcam-Linux.c: (qc_lock_wait), (qc_unlock):
* sys/qcam/qcam-os.c: (qc_lock_wait), (qc_unlock):
move hardcoded path to DEFINE
Original commit message from CVS:
* ext/divx/gstdivxdec.c: (plugin_init):
Remove comment that makes no sense.
* ext/mad/gstid3tag.c: (gst_id3_tag_set_property):
Fix for obvious typo that resulted in warnings during gst-register.
* ext/xvid/gstxviddec.c: (gst_xviddec_src_link),
(gst_xviddec_sink_link):
Fix caps negotiation a bit better.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
We call this 'codec_data', not 'esds'.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(qtdemux_parse), (qtdemux_type_get), (qtdemux_dump_mvhd),
(qtdemux_dump_tkhd), (qtdemux_dump_stsd), (qtdemux_dump_unknown),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
A number of new features and hacks to extract the esds atom and
put it into the caps. (bug #137724)
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_caps), (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Add MS RLE support. I added some functions to read out strf chunks
into strf chunks and the data behind it. This is usually color
palettes (as in RLE, but also in 8-bit RGB). Also use those during
caps creation. Lastly, add ADPCM (similar to wavparse - which
should eventually be rifflib based).
* gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
(gst_matroska_demux_init), (gst_matroska_demux_reset):
* gst/matroska/matroska-demux.h:
Remove placeholders for some prehistoric tagging system. Didn't add
support for any tag system really anyway.
* gst/qtdemux/qtdemux.c:
Add support for audio/x-m4a (MPEG-4) through spider.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
(gst_wavparse_loop):
ADPCM support (#135862). Increase max. buffer size because we
cannot split buffers for ADPCM (screws references) and I've seen
files with 2048 byte chunks. 4096 seems safe for now.
Original commit message from CVS:
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(gst_mp1videoparse_real_chain), (gst_mp1videoparse_change_state):
* gst/mpeg1videoparse/gstmp1videoparse.h:
Fix for some slight mis-cuts in buffer parsing, and for some
potential overflows or faults-causers. Adds disconts. Also fixes
#139105 while we're at it.
Original commit message from CVS:
a52dec: Use a debug category, Output timestamps correctly
Emit tag info, Handle events, tell liba52dec about cpu
capabilities so it can use MMX etc.
dvdec: Fix a crasher accessing invalid memory
dvdnavsrc:Some support for byte-format seeking.
Small fixes for still frames and menu button overlays
mpeg2dec: Use a debug category. Adjust the report level of several items to
LOG. Call mpeg2_custom_fbuf to mark our buffers as 'custom buffers'
so it doesn't lose the GstBuffer pointer
navseek: Add the navseek debug element for seeking back and forth in a
video stream using arrow keys.
mpeg2subt:Pretty much a complete rewrite. Now a loopbased element. May still
require work to properly synchronise subtitle buffers.
mpegdemux:
dvddemux: Don't attempt to create subbuffers of size 0
Reduce a couple of error outputs to warnings.
y4mencode:Output the y4m frame header correctly
Original commit message from CVS:
* ext/hermes/gsthermescolorspace.c: (plugin_init): decrease rank
by 2 to not interfere with other colorspaces.
* ext/pango/gsttextoverlay.c: (plugin_init): change rank to NONE
* gst/colorspace/gstcolorspace.c: (plugin_init): decrease rank by
one to not interfere with ffmpeg_colorspace.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
aren't in the caps.
* gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
* gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
Original commit message from CVS:
* gst/modplug/gstmodplug.cc:
handle events - don't do crap when a discont arrives that's not
necessary
This allows correct loading and playback of mods in Rhythmbox
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/gconf/Makefile.am:
* pkgconfig/Makefile.am:
move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
they get rebuilt properly
* configure.ac:
when checking for vorbis, try pkgconfig first.
* gst/modplug/gstmodplug.cc:
add fixate function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix for obvious mistake, where we first shift the offset and then
read a samplesize element assuming the old offset. Note that this
part still has something weird, i.e. my movies containing those
don't actually play well, but at least there's something that looks
like sound now.
Original commit message from CVS:
* configure.ac: the Hermes library controls hermescolorspace, not
colorspace.
* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_base_init),
(gst_mpeg2dec_init): minor pet peeve: disable code with #ifdef,
not /* */
* ext/sdl/sdlvideosink.c: Change XID to unsigned long.
* ext/sdl/sdlvideosink.h: ditto.
* gst/colorspace/gstcolorspace.c: Fix old comments about Hermes
Original commit message from CVS:
* ext/mikmod/gstmikmod.c: (gst_mikmod_init), (gst_mikmod_loop),
(gst_mikmod_change_state):
* ext/mikmod/gstmikmod.h:
make mikmod's loop function not loop infinitely and call
gst_element_yield anymore
* gst/modplug/gstmodplug.cc:
fix pad negotiation
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Fix crash (j might be greater than n_samples, in which case we're
writing outside the allocated space for the array) and memleak.
Original commit message from CVS:
2004-03-06 Christophe Fergeau <teuf@gnome.org>
* ext/faac/gstfaac.c: (gst_faac_chain):
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
* ext/libpng/gstpngenc.c: (user_write_data):
* ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
* gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
Fix several misuse of gst_buffer_merge (it doesn't take ownership
of any buffer), should fix some leaks. I hope I didn't unref buffers
that shouldn't be...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c: Add glib header
* pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
Original commit message from CVS:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init),
(gst_videodrop_chain), (gst_videodrop_change_state):
* gst/videodrop/gstvideodrop.h:
Work based on timestamp of input data, not based on the expected
framerate from the input. The consequence is that this element now
not only scales framerates, but also functions as a framerate
corrector or framerate stabilizer/constantizer.
Original commit message from CVS:
2004-02-27 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
Original commit message from CVS:
2004-02-20 Benjamin Otte <otte@gnome.org>
* ext/xine/Makefile.am:
* ext/xine/gstxine.h:
* ext/xine/xine.c:
* ext/xine/xineaudiodec.c:
* ext/xine/xinecaps.c:
add first version of xine plugin wrapper. Currently only wraps the
QDM2 win32 DLL, and even that only in proof-of-concept quality.
* configure.ac:
* ext/Makefile.am:
add xine plugin wrapper, disabled by default. Use --enable-xine to
build. Note that it'll segfault on gst-register if you don't remove
the goom and tvtime post plugins from xine.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(qtdemux_parse), (qtdemux_parse_trak), (qtdemux_audio_caps):
add extradata parsing for QDM2.
change around debugging prints.
Original commit message from CVS:
2004-02-15 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_loop): More fixes for
correct data refcounting.
Original commit message from CVS:
2004-02-15 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_change_state),
(gst_switch_class_init): Cleaning the sinkpads correctly on state
change, mostly the EOS flag.
Original commit message from CVS:
2004-02-14 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_connect_visualization): Disable
visualization until i find a way to fix switch correctly.
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head): Fix a bug when
EOS arrives.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_dispose), (gst_switch_class_init):
Reworked switch to get a more correct behaviour with events and refing
of data stored in sinkpads.
* gst/switch/gstswitch.h: Adding an eos flag for every sinkpad so that
we don't pull from a pad in EOS.
Original commit message from CVS:
2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
set explicit caps before adding the element, so the autopluggers can
plug correctly.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(mpeg2_sys_type_find), (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find):
fix memleaks in typefind functions. gst_type_find_suggest takes a const
argument.
Original commit message from CVS:
2004-01-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/mpeg1videoparse/gstmp1videoparse.c:
(gst_mp1videoparse_real_chain):
Committed wrong version last week... Grr... Didn't notice until now.
Original commit message from CVS:
2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
(gst_ffmpeg_caps_to_extradata), (gst_ffmpeg_caps_to_pixfmt):
* gst/qtdemux/qtdemux.c: (plugin_init), (qtdemux_parse_trak),
(qtdemux_video_caps):
* gst/qtdemux/qtdemux.h:
Add SVQ3 specific flags to qtdemux and ffmpeg
Original commit message from CVS:
2004-01-25 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_identity_handoff), (gst_play_set_location),
(gst_play_set_visualization), (gst_play_connect_visualization): Another
try in visualization implementation. Still have an issue with switch
blocking when pulling from video_queue and only audio comes out of
spider.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
release method. And check if the pad is usable before pulling.
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Additional pad usability check.
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
(gst_mp1videoparse_real_chain):
Fix MPEG video stream parsing. The original plugin had several
issues, including not timestamping streams where the source was
not timestamped (this happens with PTS values in mpeg system
streams, but MPEG video is also a valid stream on its own so
that needs timestamps too). We use the display time code for that
for now. Also, if one incoming buffer contains multiple valid
frames, we push them all on correctly now, including proper EOS
handling. Lastly, several potential segfaults were fixed, and we
properly sync on new sequence/gop headers to include them in next,
not previous frames (since they're header for the next frame, not
the previous). Also see #119206.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
(bpf_from_header):
Move caps setting so we only do it after finding several valid
MPEG-1 fraes sequentially, not right after the first one (which
might be coincidental).
* gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Add unsynced MPEG video stream typefinding, and change some
probability values so we detect streams rightly. The idea is as
follows: I can have an unsynced system stream which contains
video. In the current code, I would randomly get a type for either
system or video stream type found, because the probabilities are
being calculated rather randomly. I now use fixed values, so we
always prefer system stream if that was found (and that is how it
should be). If no system stream was found, we can still identity
the stream as video-only.
Original commit message from CVS:
2004-01-20 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_request_new_pad),
(gst_switch_init): Fixed switch element : proxying link and setting
caps from src to sink on request.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
from the template.
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofilter.h:
Add bytes_per_sample and size and n_samples calculation.
* gst-libs/gst/audio/gstaudiofilterexample.c:
Remove, now autogenerated.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
Moved from gstaudiofilterexample, object name changed, code added
so that it actually works.
* gst-libs/gst/audio/make_filter:
Script to build an audiofilter subclass from the template.
* gst/colorspace/Makefile.am:
* gst/colorspace/yuv2yuv.c:
Remove file, since it's GPL, and we don't use it.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_class_init): Remove property
that handles osssink fallback.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_getcaps):
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Add audio/x-qdm2 for QDM2 audio.
* gst/sine/gstsinesrc.c: (gst_sinesrc_get):
* gst/sine/gstsinesrc.h: Add example of how to implement tags.
* gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
Decrease minimum size to 16x16.
* gst/wavparse/gstwavparse.c:
Convert disabled pad template caps to new caps.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_chain): Throw element error when display cannot
be opened. Increase minimum framerate to 1.0. Check the data
free function on a buffer to make sure it is the type we expect
before manipulating it.
Original commit message from CVS:
2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/yuv2yuv.c: (gst_colorspace_yuy2_to_i420),
(gst_colorspace_i420_to_yv12):
Fix compiling... Didn't test if it actually works.
Original commit message from CVS:
* configure.ac:
* gst/colorspace/Makefile.am:
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/gstcolorspace.h:
* gst/colorspace/yuv2rgb.c:
* gst/colorspace/yuv2rgb.h:
Duplicate the ext/hermes colorspace plugin, and remove Hermes
code and GPL code. Fix for new caps negotiation. Rewrite
much of the format handling code, and some of the conversion
code. Basically, rewrote almost everything. This element
handles I420, YV12 to RGB conversions.
* ext/hermes/Makefile.am:
* ext/hermes/gsthermescolorspace.c:
Rename colorspace to hermescolorspace. Fix negotiation issues.
Remove non-Hermes related code. This element handles lots of
RGB to RGB conversions, but no YUV.
* ext/hermes/gstcolorspace.c:
* ext/hermes/gstcolorspace.h:
* ext/hermes/rgb2yuv.c:
* ext/hermes/yuv2rgb.c:
* ext/hermes/yuv2rgb.h:
* ext/hermes/yuv2rgb_mmx16.s:
* ext/hermes/yuv2yuv.c:
* ext/hermes/yuv2yuv.h:
Remove old code.
Original commit message from CVS:
2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (plugin_init):
qtdemux requires bytestream
Original commit message from CVS:
2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Improve mpeg2enc detection. This is for distributions that do
ship mjpegtools, but without mpeg2enc. Also does object check
for might there ever be ABI incompatibility.
* ext/mpeg2enc/gstmpeg2enc.cc:
Add Andrew as second maintainer (he's helping me), and also add
an error if no caps was set. This happens if I pull before capsnego
and that's something I should solve sometime else.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix time parsing.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
(gst_matroska_mux_track_header):
Add caps to templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
Add mpegversion=1 to prevent confusion with MPEG/AAC.
* gst/mpegstream/gstmpegdemux.c:
Remove layer since it causes warnings about unfixed caps.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Fix obvious typo (we error out if caps were set, we should of
course error out if *no* caps were set).
* sys/oss/gstosselement.c: (gst_osselement_convert):
Fix format conversion, we confused bits/bytes.
* sys/oss/gstosselement.h:
Improve documentation for 'bps'.
* sys/v4l/TODO:
Remove stuff about plugins that need removing - this was done
ages ago.
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
(gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
(gst_v4lsrc_src_query):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
(gst_v4l2src_src_convert), (gst_v4l2src_src_query):
Add get_query_types(), get_formats() and query() functions.
Original commit message from CVS:
Sorry Dave... Add mpegversion=1 to mp3 caps everywhere so that the autoplugger uses mad and not faad for mp3 decoding. This should fix mp3 playback.
Original commit message from CVS:
Adding a new plugin: switch.
It takes N input and only has 1 output. You can "switch" the forwarded input through properties ("nb_sources", "active_source") and i will probably add tuner interface support soon.
It should be able to handle any kind of data passing through it.
It is still a work in progress don't consider it usable for production yet.
Original commit message from CVS:
tagging stuff and build fixes. In detail:
- make gdk-pixbuf loader work when distchecking
- fix invalid syntax in ffmpeg Makefile. wildcards for EXTRA_DIST are not allowed. This broke builds where distdir != srcdir
- fix ffmpeg cvs grabbing when srcdir != distdir
- new id3tag plugin for id3 tag reading/writing (uses mad's libid3tag)
- mad and libid3tag require mad/libid3tag v0.15. Fixed configure to require that
- added ogg demuxer in ext/ogg. The demuxer does not handle events yet. Especially getting seeking right will require some effort or code copying from libvorbis.
- added raw vorbis detection to typefinding. oggdemux requires a typefind function to detect its contents.
- tags plugin in gst/tags. Provides API in <gst/tags/gsttagediting.h>. API includes tag matching GStreamer <=> ID3 and GStreamer <=> vorbis and writing/reading vorbiscomments or ID3v1 tags. Also included is a simple vorbiscomment reader/writer. Writing will not really work though until someone writes oggmux.
- various build fixes. Mostly missing (DIST)CLEANFILES.
- vorbisenc handles tag writing.
Now it's YOUR turn to fix and write more plugins that handle writing/reading of tags. :)
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
Original commit message from CVS:
Woah, I'm f***ing annoyed that someonme never tests his changes and figures out that every freakin' format is identified as text/plain
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
Original commit message from CVS:
Add new element: frame dropper. This element inserts/drops frames to go from a certain input framerate to a certain output framerate. It's extremely simple and that's why it's so cute.
Original commit message from CVS:
Adding needed license information. The patch was reviewed and approved by
Christian Shaller. Ronald Bultje and Benjamin also responded with
comments.
Original commit message from CVS:
Fixes to make it compile without GNOME, and with a modern (>= 0.3)
version of GStreamer. Now that I got it compiled, I want to delete
it.
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h
I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
Original commit message from CVS:
This adds width/height properties to qtdemux, so that it connects to ffdec_*... still doesn't work, because the buffer allocation in ffdec_ allocs too small buffers (edge emulation failure or so?), which causes a segfault. I'm working on that
Original commit message from CVS:
+ simplifying the filter's structure, but it's still not working perfectly
+ starting to wonder if/how midi integration is possible ...
Original commit message from CVS:
Handle compressed headers. Fix inappropriate use of bytestream_flush().
Code cleanup. Added getcaps and _link functions for src pads. Extract
and set the size,rate,channels correctly. Fix some of the caps to
agree with avidemux and/or ffmpeg.
Original commit message from CVS:
fix Makefiles for C++ libraries. They should now work with Forte. This needs a new libtool, update autogen.sh to reflect this.
Original commit message from CVS:
Made the SWAP and PREPARE_3D_LINE macros work with gints rather than
using typeof(), since typeof() is a gcc-extension and does not work
with other compilers. This is okay since every place these macros
are used, gints are passed in. I renamed SWAP to SWAP_INT to reflect
that it is not so generic.
Original commit message from CVS:
Corrected the configure.ac so it actually works. Updated some c files
so that they build on Solaris. This mostly involved supporting ISO
style variable-argument macros.
Original commit message from CVS:
Updated autogen.sh/configure.ac and various Makefiles to make the
configure script set up all gcc specific compiler arguments, rather
than hardcoding them in the Makefile.am files
Original commit message from CVS:
fixes to mp3 typefinding:
- removed workaround that detected files with valid ID3v2 tag as mp3 (not needed anymore)
Invalid files didn't occur because of broken length in the tag but because of padding
in the beginning of the audio data most of the time.
- fixed various assorted stuff in the old typefind function (like not adjusting buffer
size after skipping)
- added 2nd typefind function to detect mp3 streams (fixes#94113)
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
Original commit message from CVS:
Use bytestream to get buffer from sinkpad ( gst-player should play mod file now ;)
Move module types build from mikmod to modplug
Original commit message from CVS:
Added measures and beats to the playondemand filter so it can act like an audio
sequencer. Currently defines three extra globally visible functions, might
eventually want to put them in an interface instead ?
Original commit message from CVS:
- reimplemented using organic masks, rendered with gouraud shaded triangles
- implemented more masks
- implemented adjustable border
Original commit message from CVS:
- Ripped some ID3 tag parsing from libid3 for typefind.
- Added ID3V1 to fypefind.
- Don't check for a valid mp3 header after finding the ID3 tag as some ID3
tags seem broken.
Original commit message from CVS:
qtdemux.c:315: warning: implicit declaration of function `free'
qtdemux.c:331: warning: implicit declaration of function `malloc'
Original commit message from CVS:
new filter subdir for standard audio filters
first filter uses code from vorbis to implement an iir filter
not optimized yet, iir code uses doubles and plugin uses float
Original commit message from CVS:
add ranks only for plugins who participate in autoplugging. If you have a file that used to autoplug but doesn't anymore, then let me know or add a rank to the missing element.
Original commit message from CVS:
adding new quicktime parser:
- openquicktime free (hense gst/qtdemux)
- no more seeks for parsing -> better for network streams
- uses GstByteStream
- less memcpy's
- long ChangeLog record in pompous style
Original commit message from CVS:
* a hack to work around intltool's brokenness
* a current check for mpeg2dec
* details->klass reorganizations
* an element browser that uses details->klass
* separated cdxa parse out from the avi directory
Original commit message from CVS:
fixed rest of warning for gcc 3 in /gst.
fixed some Makefiles: s/-m486/-mcpu=i486/
disabled mpegaudioparse plugin. What good is this rotten code for anyway?
Original commit message from CVS:
* removal of //-style comments
* don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.
Original commit message from CVS:
s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
@-substitued variables variables are defined as make variables automagically,
and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag
Original commit message from CVS:
* s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way
* added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register,
and attempt to run though
* imposed some restrictions on the naming of request pads to better allow for reverse parsing
* added '%s' to reverse parsing
* added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out
* fixen on launch-gui
* added pkg-config stuff for the editor's libs
Original commit message from CVS:
Uhm, if I'll be fixing errors like this all over just because I'm enabling
plugin debug output for the first time, I'm in for a world of hurt over
the next few hours...
Original commit message from CVS:
* changes to spider:
* add padtemplates so the connect functions can connect two request-pad
elements
* add a hack in gstelement.c. please look at this, Company, and see how
we can get around this
* add backwards caps-propagation support in identity, int2float, float2int,
adder, speed, volume
Original commit message from CVS:
made changes everywhere to accomodate for the headers being in
<gst/(lib)/...>
we'll need to conclude this fast because we will also need to change stuff in core real soon for the libs in order to fix everything
and I can't do it right now because I disabled all of the plugins here ;)
