Commit graph

273 commits

Author SHA1 Message Date
Jan Schmidt
83e3fe189c configure.ac: Add dvdlpcmdec
Original commit message from CVS:

* configure.ac:
Add dvdlpcmdec

* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_reset),
(free_all_buffers), (gst_mpeg2dec_alloc_buffer):
Don't push buffers if the src pad isn't negotiated yet.

* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format),
(gst_audio_convert_buffer_from_default_format):
Add support for 24-bit width.

* gst/dvdlpcmdec/.cvsignore:
* gst/dvdlpcmdec/Makefile.am:
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_get_type),
(gst_dvdlpcmdec_base_init), (gst_dvdlpcmdec_class_init),
(gst_dvdlpcm_reset), (gst_dvdlpcmdec_init), (gst_dvdlpcmdec_link),
(gst_dvdlpcmdec_chain), (gst_dvdlpcmdec_change_state),
(plugin_init):
* gst/dvdlpcmdec/gstdvdlpcmdec.h:
New decoder for rearranging DVD LPCM into our audio/x-raw-int
format. Needs support for the channels maps if someone can find
a DVD LPCM track with > 2 channels.

* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_handle_dvd_event),
(gst_dvd_demux_send_discont), (gst_dvd_demux_handle_discont),
(gst_dvd_demux_get_audio_stream), (gst_dvd_demux_process_private):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_init_stream),
(gst_mpeg_demux_send_subbuffer), (gst_mpeg_demux_handle_src_query):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_reset),
(gst_mpeg_parse_parse_packhead), (gst_mpeg_parse_loop),
(gst_mpeg_parse_get_rate), (gst_mpeg_parse_convert_src),
(gst_mpeg_parse_handle_src_query),
(gst_mpeg_parse_handle_src_event):
Use audio/x-dvd-lpcm for LPCM output.
Add DTS output.
2005-02-08 11:08:15 +00:00
Andy Wingo
9c1c858d24 gst/audioconvert/bufferframesconvert.c
Original commit message from CVS:
2005-02-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate): New function, fixates to 256
frames per buffer by default. (Much better than 1.)
(buffer_frames_convert_init): Set the fixate function for both src
and sink pad.
(buffer_frames_convert_link): After success setting nonfixed caps,
get the negotiated caps so we can know how many buffer-frames it
will be. No idea how this worked at all before.
2005-02-04 15:40:38 +00:00
Ronald S. Bultje
97dd32f748 gst/audioconvert/gstaudioconvert.c: The return value of fixate_to does not imply that the requested value was set, so...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
The return value of fixate_to does not imply that the requested
value was set, so don't assume.
2005-01-07 18:17:52 +00:00
Thomas Vander Stichele
c1e1206414 fix for glib < 2.4
Original commit message from CVS:
fix for glib < 2.4
2004-12-24 15:12:56 +00:00
Benjamin Otte
05103c18ca gst/audioconvert/gstchannelmix.c: more overwriting protection due to modifying channels one by one instead of all at ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
more overwriting protection due to modifying channels one by one
instead of all at once
2004-11-28 20:41:23 +00:00
Ronald S. Bultje
836ea71ea1 gst/audioconvert/gstchannelmix.c: Normalize using absolute values.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_normalize):
Normalize using absolute values.
2004-11-28 19:22:07 +00:00
Benjamin Otte
110fd90222 gst/audioconvert/gstchannelmix.c: walk the samples backwards if out_channels > in_channels so we don't overwrite data
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
walk the samples backwards if out_channels > in_channels so we don't
overwrite data
2004-11-28 18:26:36 +00:00
Ronald S. Bultje
4fdf8d5b16 gst/audioconvert/: Implement a channel mixer.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_channels):
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_unset_matrix),
(gst_audio_convert_fill_identical),
(gst_audio_convert_fill_compatible),
(gst_audio_convert_detect_pos), (gst_audio_convert_fill_one_other),
(gst_audio_convert_fill_others),
(gst_audio_convert_fill_normalize),
(gst_audio_convert_fill_matrix), (gst_audio_convert_setup_matrix),
(gst_audio_convert_passthrough), (gst_audio_convert_mix):
* gst/audioconvert/gstchannelmix.h:
Implement a channel mixer.
2004-11-28 16:09:13 +00:00
Ronald S. Bultje
1de3e19fdb ext/a52dec/gsta52dec.c: Don't do sample adjusting anymore, we use float audio now.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_loop),
(gst_a52dec_change_state):
Don't do sample adjusting anymore, we use float audio now.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
Don't fixate to non-existing properties.
2004-11-27 20:22:42 +00:00
Christophe Fergeau
39f436c1a0 gst/audioconvert/gstaudioconvert.c: call parent dispose method
Original commit message from CVS:
2004-11-27  Christophe Fergeau  <teuf@gnome.org>

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
call parent dispose method
2004-11-27 14:41:51 +00:00
Ronald S. Bultje
3a0a2898af Surround sound support.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
2004-11-25 20:36:29 +00:00
Ronald S. Bultje
b463251864 gst/audioconvert/gstaudioconvert.c: Really don't touch read-only buffers (#156563).
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format):
Really don't touch read-only buffers (#156563).
2004-10-29 12:48:45 +00:00
Iain Holmes
02be6646cc Free the caps used for the try
Original commit message from CVS:
Free the caps used for the try
2004-09-16 11:34:50 +00:00
David Schleef
78deea7e4a configure.ac: remove NASM check, since we don't use it. Update dirac check to 0.4
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
2004-09-15 19:29:24 +00:00
Steve Lhomme
7b86915b8a more working plugins
Original commit message from CVS:
more working plugins
2004-07-27 21:41:30 +00:00
Steve Lhomme
a6c379fd7a rename GStreamer-0.8.lib to libgstreamer.lib
Original commit message from CVS:
rename GStreamer-0.8.lib to libgstreamer.lib
2004-07-27 09:57:33 +00:00
Steve Lhomme
6c8a44d0c9 avoid problems with math.h, fix release dependancy
Original commit message from CVS:
avoid problems with math.h, fix release dependancy
2004-07-27 09:48:51 +00:00
Steve Lhomme
17615126c2 more plugins supported under windows
Original commit message from CVS:
more plugins supported under windows
2004-07-26 13:20:10 +00:00
Benjamin Otte
d5f5085677 gst/audioconvert/gstaudioconvert.c: don't enfore negotiation from source side, it breaks sinesrc ! audioconvert ! oss...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
don't enfore negotiation from source side, it breaks
sinesrc ! audioconvert ! osssink
2004-07-23 17:40:16 +00:00
Andy Wingo
b028fc77a7 gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For float, "any" caps -> buffer_frames=[0,MAX].
Original commit message from CVS:
2004-07-11  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For
float, "any" caps -> buffer_frames=[0,MAX].

* gst/interleave/interleave.c (interleave_getcaps): Seems the core
doesn't intersect our caps with the template any more. Do it
ourselves.
(interleave_buffered_loop): Use g_newa instead of malloc/free.
2004-07-11 11:21:56 +00:00
Thomas Vander Stichele
c750d9bd49 don't assert in state change
Original commit message from CVS:
don't assert in state change
2004-07-09 10:56:51 +00:00
Benjamin Otte
b95a7dca6a gst/audioconvert/gstaudioconvert.c: fixate nicely even when the peer is not negotiating
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate nicely even when the peer is not negotiating
2004-05-26 14:47:23 +00:00
Benjamin Otte
4f845c50ce gst/audioconvert/gstaudioconvert.c: make sure we don't allow depth > width
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
make sure we don't allow depth > width
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate endianness to G_BYTE_ORDER as default
* gst/audioscale/gstaudioscale.c:
we don't handle another endianness as host-endianness
2004-05-25 20:14:10 +00:00
Benjamin Otte
2e050e0378 ext/vorbis/oggvorbisenc.c: properly fail when we can't setup the vorbis encoder due to unsupported settings
Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
(gst_oggvorbisenc_setup):
properly fail when we can't setup the vorbis encoder due to
unsupported settings
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
(gst_vorbisenc_setup):
same
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
fix case where warnings occured when one pad was unlinked while the
other's link function was called
2004-05-24 19:19:29 +00:00
Stéphane Loeuillet
1f1a7cbe84 first batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
Original commit message from CVS:

first batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
2004-05-21 22:39:30 +00:00
Thomas Vander Stichele
cba2022045 gst/audioconvert/gstaudioconvert.c: refactor/comment code
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
refactor/comment code
2004-05-03 13:25:22 +00:00
Benjamin Otte
1cd4212d70 gst/audioconvert/gstaudioconvert.c: fix memleak
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
fix memleak
2004-04-25 17:56:11 +00:00
Benjamin Otte
26cc5e8768 ext/mad/gstid3tag.c: remove leftover g_print
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init):
remove leftover g_print
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
don't try setting only a subset of the caps. We don't want to kill
autoplugging on purpose
2004-04-20 15:51:48 +00:00
Thomas Vander Stichele
d32724fb41 add debugging
Original commit message from CVS:
add debugging
2004-04-14 16:09:10 +00:00
Benjamin Otte
50d120f1ba ext/gnomevfs/gstgnomevfssink.c: fix erase signal - if any handler returns false the file will not be overwritten. If ...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(_gst_boolean_allow_overwrite_accumulator),
(gst_gnomevfssink_class_init):
fix erase signal - if any handler returns false the file will not be
overwritten. If no handler is connected, the file will not be
overwritten either.
renamed signal to "allow-overwrite"
* ext/mad/gstid3tag.c: (tag_list_to_id3_tag_foreach):
free string when adding it to ID3 failed
* ext/vorbis/vorbisdec.c: (vorbis_dec_event):
unref event when done
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
free caps
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_type_find):
fix invalid read
2004-04-09 18:55:10 +00:00
Andy Wingo
92fe387eea gst/audioconvert/bufferframesconvert.c: New element to convert buffer-frames for float streams. Not working nicely yet.
Original commit message from CVS:
2004-04-09  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c: New element to convert
buffer-frames for float streams. Not working nicely yet.
* gst/audioconvert/plugin.h:
* gst/audioconvert/plugin.c: New files.
* gst/audioconvert/Makefile.am: Build the new files.
* gst/audioconvert/gstaudioconvert.c: Initialize via plugin.[ch].
2004-04-09 12:39:30 +00:00
Benjamin Otte
0db7a00219 gst/audioconvert/gstaudioconvert.c: advertise buffer-frames correctly on sinkpads
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps):
advertise buffer-frames correctly on sinkpads
2004-04-05 13:18:56 +00:00
Benjamin Otte
87ffc58ab9 gst/audioconvert/gstaudioconvert.c: add a fixation function that pretty much does the right thing (fixes #137556)
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(_fixate_caps_to_int), (gst_audio_convert_fixate):
add a fixation function that pretty much does the right thing (fixes
#137556)
2004-03-21 02:54:37 +00:00
Thomas Vander Stichele
f83cb187de don't mix tabs and spaces
Original commit message from CVS:
don't mix tabs and spaces
2004-03-15 19:32:28 +00:00
Thomas Vander Stichele
4df3f18839 gst-indent
Original commit message from CVS:
gst-indent
2004-03-14 22:34:34 +00:00
Colin Walters
e93d93afdf gst/audioconvert/gstaudioconvert.c: Fix typo in width 8 conversion.
Original commit message from CVS:
2004-03-09  Colin Walters  <walters@verbum.org>

* gst/audioconvert/gstaudioconvert.c: Fix typo in width 8
conversion.
2004-03-10 04:01:50 +00:00
Benjamin Otte
c6b75500be add some 'what's this element and what is it not' doc
Original commit message from CVS:
add some 'what's this element and what is it not' doc
2004-03-06 15:31:25 +00:00
Benjamin Otte
33f79a881e gst/audioconvert/gstaudioconvert.c: do conversions from/to float correctly, fix some caps nego errors, export correct...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_buffer_from_default_format):
do conversions from/to float correctly, fix some caps nego errors,
export correct supported caps in template and getcaps, use correct
caps in try_set_caps functions
2004-03-06 13:26:12 +00:00
David Schleef
f0365ebe22 ext/aalib/gstaasink.c: Add fixate function. (bug #131128)
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_fixate), (gst_aasink_init):
Add fixate function. (bug #131128)
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_fixate):  Add fixate function.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Fix attempt to print a non-pointer using GST_PTR_FORMAT.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt):
Fix missing break that was causing ulaw to be interpreted as
raw int.
2004-03-06 04:51:15 +00:00
David Schleef
befdae8cda ext/faad/gstfaad.c: Fix negotiation.
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.
2004-03-06 00:42:20 +00:00
Benjamin Otte
043693d8d9 gst/audioconvert/gstaudioconvert.c: convert channels correctly. convert correctly to unsigned.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
convert channels correctly. convert correctly to unsigned.
2004-03-05 21:05:26 +00:00
Benjamin Otte
02c11b879e gst/audioconvert/gstaudioconvert.c: make float=>int conversion work correctly even in cornercases.
Original commit message from CVS:
2004-03-05  Benjamin Otte  <otte@gnome.org>

* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format):
make float=>int conversion work correctly even in cornercases.
2004-03-04 23:30:29 +00:00
Thomas Vander Stichele
8008cf22a4 PTR fix
Original commit message from CVS:
PTR fix
2004-02-25 18:02:19 +00:00
Thomas Vander Stichele
d31fb663bd assorted debug/warning fixes
Original commit message from CVS:
assorted debug/warning fixes
2004-02-25 17:45:54 +00:00
Andy Wingo
cd8976a9eb gst/interleave/interleave.c (interleave_buffered_loop): Always push only when channel->buffer is NULL. Prevents segfa...
Original commit message from CVS:
2004-02-25  Andy Wingo  <wingo@pobox.com>

* gst/interleave/interleave.c (interleave_buffered_loop): Always
push only when channel->buffer is NULL. Prevents segfaults doing
the state change after a nonlocal exit, like a scheme exception.

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
Handle the case where the intersected caps is empty.
2004-02-25 13:25:44 +00:00
Benjamin Otte
355318b59d gst/audioconvert/gstaudioconvert.c: set rank to PRIMARY
Original commit message from CVS:
2004-02-22  Benjamin Otte  <otte@gnome.org>

reported by: Stefan Kost <kost@imn.htwk-leipzig.de>

* gst/audioconvert/gstaudioconvert.c: (plugin_init):
set rank to PRIMARY
* gst/volume/gstvolume.c: (plugin_init):
set rank to NONE
fixes #134960

2004-02-22   Julio M. Merino Vidal <jmmv@menta.net>

reviewed by Benjamin Otte  <otte@gnome.org>

* ext/flac/gstflacenc.c: (gst_flacenc_chain):
escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
2004-02-22 15:31:30 +00:00
Andy Wingo
ce89f16818 gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and interleave respectively.
Original commit message from CVS:
2004-02-20  Andy Wingo  <wingo@pobox.com>

* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.

* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.

* configure.ac: Remove intfloat and oneton, add interleave.

* ext/sndfile/gstsf.c: Handle events better.

* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
2004-02-20 14:17:57 +00:00
Thomas Vander Stichele
0f003d87b3 throw error instead of assertion
Original commit message from CVS:
throw error instead of assertion
2004-02-16 11:45:32 +00:00
David Schleef
8fe7678826 gst/audioconvert/gstaudioconvert.c: Use gst_pad_try_set_caps_nonfixed().
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Use gst_pad_try_set_caps_nonfixed().
2004-01-27 09:05:22 +00:00
David Schleef
e4593b1536 gst/ac3parse/gstac3parse.c: update to checklist 5
Original commit message from CVS:
* gst/ac3parse/gstac3parse.c: update to checklist 5
* gst/adder/gstadder.c: rewrite negotiation.  update to checklist 5
* gst/audioconvert/gstaudioconvert.c: update to checklist 5
* gst/audioscale/gstaudioscale.c: same
* gst/auparse/gstauparse.c: same
* gst/avi/gstavidemux.c: same
2004-01-27 09:00:01 +00:00
Benjamin Otte
b552a2d9f0 gst-libs/gst/audio/audio.h: remove buffer-frames from audio caps
Original commit message from CVS:
2004-01-26  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
2004-01-26 03:54:21 +00:00
David Schleef
89303c580f ext/esd/esdsink.c: Remove property that handles osssink fallback.
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.
2004-01-15 21:05:17 +00:00
Thomas Vander Stichele
2a415f1abd fix up audioconvert caps nego, remove float stuff, remove rate stuff gst-launch-0.7 -v sinesrc ! audioconvert ! audi...
Original commit message from CVS:

fix up audioconvert caps nego, remove float stuff, remove rate stuff
gst-launch-0.7  -v sinesrc ! audioconvert ! audio/x-raw-int,rate=23000 ! wavenc ! filesink location=test.wav now writes a completely useless 23000 Hz wave file
2004-01-12 19:46:45 +00:00
David Schleef
61763a83f0 gst/audioconvert/gstaudioconvert.c: Test that pad is negotiated before getting its caps.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Test that pad is negotiated before getting its caps.
2004-01-12 18:59:57 +00:00
David Schleef
e095323df1 Negotiation fixes.
Original commit message from CVS:
Negotiation fixes.
2003-12-31 08:02:04 +00:00
David Schleef
3b60021408 Merge CAPS branch
Original commit message from CVS:
Merge CAPS branch
2003-12-22 01:47:09 +00:00
Andy Wingo
80fece4f4b remove copyright field from plugins
Original commit message from CVS:
remove copyright field from plugins
2003-12-04 10:37:39 +00:00
Leif Johnson
736153ab06 + checking in plugin category changes
Original commit message from CVS:
+ checking in plugin category changes
2003-11-16 22:02:21 +00:00
Iain Holmes
3d97918694 Audioconvert - Check!
Original commit message from CVS:
Audioconvert - Check!
Updated for new stuff
2003-11-01 11:41:42 +00:00
Andy Wingo
dc35dbb595 /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
Original commit message from CVS:
/GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
2003-10-08 16:08:22 +00:00
Thomas Vander Stichele
453e9b8871 reverting error patch before making a branch.
Original commit message from CVS:
reverting error patch before making a branch.
2003-09-16 10:00:02 +00:00
Benjamin Otte
d26698b5a1 converted gst_element_error to new format in gst/ - gettext pending
Original commit message from CVS:
converted gst_element_error to new format in gst/ - gettext pending
2003-09-15 00:34:44 +00:00
David Schleef
0228717f04 Remove redundant plugindir definition
Original commit message from CVS:
Remove redundant plugindir definition
2003-08-10 00:01:58 +00:00
Leif Johnson
2ad6bd23cf + some whitespace changes + adding dummy definitions to prepare for float caps
Original commit message from CVS:
+ some whitespace changes
+ adding dummy definitions to prepare for float caps
2003-07-19 22:58:41 +00:00
Ronald S. Bultje
b005531324 New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
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
2003-07-06 20:49:52 +00:00
Benjamin Otte
ffa5706370 compatibility fix for new GST_DEBUG stuff.
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.
2003-06-29 19:46:12 +00:00
David Schleef
b036fe4fda fix: width is in bytes, not bits. Remove incorrect assertion.
Original commit message from CVS:
fix: width is in bytes, not bits.  Remove incorrect assertion.
2003-06-10 06:32:59 +00:00
Benjamin Otte
b3c728ed0d *_is_writeable => *_is_writable (spelling)
Original commit message from CVS:
*_is_writeable => *_is_writable (spelling)
2003-04-16 18:36:29 +00:00
Benjamin Otte
333099fdeb change *_is_readonly to *_is_writeable. Hope the name stays now...
Original commit message from CVS:
change *_is_readonly to *_is_writeable. Hope the name stays now...
2003-04-15 21:35:08 +00:00
Benjamin Otte
f38ca0a7f4 - revert change to use a newly added gst_buffer_is_readonly (which wasn't newly added before)
Original commit message from CVS:
- revert change to use a newly added gst_buffer_is_readonly (which wasn't newly added before)
- walk buffer backwards when it might be possible that data is read out of overwritten parts when going forwards
2003-04-15 19:10:14 +00:00
Colin Walters
804522784e Change agressive -> aggressive.
Original commit message from CVS:
Change agressive -> aggressive.
2003-04-15 03:39:22 +00:00
Colin Walters
8aa9339f50 Initialize various variables so gcc won't complain.
Original commit message from CVS:
Initialize various variables so gcc won't complain.

Use GST_BUFFER_FLAG_IS_SET instead of unknown function gst_buffer_is_readonly.
2003-04-15 03:19:08 +00:00
Benjamin Otte
d41620bc75 Added initial version of audioconvert, a generic converter of integer audio/raw formats.
Original commit message from CVS:
Added initial version of audioconvert, a generic converter of integer audio/raw formats.
It currently supports conversion of
- channels (mono/stereo only, until someone tells me how to mix other channels)
- endianness (little/bi endian)
- signedness
- width (8, 1, 24 and 32 bits)
- depth (1 - width bits)
missing:
- enough testing (I intend to write a testsuite for this, but that's pending)
- samplerate conversion
- other goodies like format conversion etc
Expect bugs when using it.

problems this should solve:
- encoding wav files on big endian machines
- goom working with mono audio files in gst-player
- Iain's soundcard (that one is a problem in itself)
- complaints about missing conversion
- too many age old, nearly unmaintained plugins (stereo2mono etc.)
Have fun.
2003-04-14 01:20:30 +00:00