Commit graph

169 commits

Author SHA1 Message Date
Sebastian Dröge
69e338d7dd alsa: Constify channel position table 2015-01-21 09:42:35 +01:00
Sebastian Dröge
8e6fb92886 alsa: Fix indention 2015-01-21 09:42:35 +01:00
Thomas Roos
485ad66a11 alsa: Allow to use 8 bit samples with ALSA
8 bit samples have no (0) as endianness, not the native endianness.

https://bugzilla.gnome.org/show_bug.cgi?id=739446
2015-01-21 09:42:35 +01:00
Takashi Iwai
76d807893c alsa: Add channel map API support
The initial support for the new ALSA chmap API.
Just translate the current chmap to GstAudioChannelPosition during the
setup.  No function to specify the channel map manually yet, so still
impossible to assign any non-standard positions or to configure in a
different order even if the hardware allows.

https://bugzilla.gnome.org/show_bug.cgi?id=709755
2013-10-09 19:05:53 +02:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-03 23:05:09 +00:00
Tim-Philipp Müller
1a69ec3fd3 alsa: if no formats in native endianness could be detected, try non-native endianness as well
This can happen, e.g. when using an USB sound card on
a big-endian device

https://bugzilla.gnome.org/show_bug.cgi?id=680904
2012-10-18 11:04:06 +01:00
Tim-Philipp Müller
1e329bb4f4 alsa: fix supported format detection
The format probing code was assuming there'd be one caps
structure for each separate width/depth combination like
we did in 0.10 all over the place: for one, we'd query
unsigned/signed formats together for the same width/height,
and we'd add the entire current structure to the probed
caps when we find a format is supported. Now that we have
all raw formats in a single structure, this is all not going
to work so well any more. We added the entire structure with
all possible formats to the caps if we support just one format.

Fix probing so that we only return the list of actually
supported raw audio formats (with native endianness) from
get_caps().
2012-10-18 11:03:07 +01:00
Tim-Philipp Müller
fc37cf5779 Silence some 'variable may be used uninitialized' compiler warnings
when compiling with -DG_DISABLE_ASSERT
2012-08-08 10:19:20 +01:00
Andoni Morales Alastruey
2434f2932b alsasink: check for spdif support only in the current device 2012-05-18 12:01:06 +02:00
Andoni Morales Alastruey
c6409806c1 alsasink: use the iec958 payloader to support non-payloaded input streams 2012-05-07 13:31:01 +02:00
Wim Taymans
6e054dfc3d alsa: fix small caps leak 2012-03-27 15:43:44 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
61a53092e4 alsa: merge instead of appending structures 2012-01-26 14:28:06 +01:00
Sebastian Dröge
2fc75efdce alsa: Port to the new multichannel caps 2012-01-05 10:34:20 +01:00
Wim Taymans
3254e79f04 alsa: fix negotiation
Don't assume the format is a string because now it is a list of string in the
template.
Chain up to the parent class implementation of get_caps.
2011-11-10 16:05:19 +01:00
Wim Taymans
06311362e9 fix compilation 2011-10-27 17:26:58 +02:00
Wim Taymans
8023f49d19 more audio caps porting 2011-08-19 17:41:22 +02:00
Wim Taymans
693919ff87 alsa: add method to retrieve the card name
Reuse an existing method to retrieve the card name.
2010-08-18 16:42:13 +02:00
Stefan Kost
3d0c70d3d8 alsa: release pcminfo after the strdup 2009-02-27 11:14:25 +02:00
Stefan Kost
c074e84360 alsa: cleanup name lookup.
We can break, once we have a name to make sure, we won't read it ever twice.
2009-02-26 18:01:05 +02:00
Julien Moutte
f0154849b0 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
Original commit message from CVS:
2008-02-29  Julien Moutte  <julien@fluendo.com>

* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
2008-02-29 18:44:36 +00:00
David Schleef
1beb98e6dc ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
Change alsa alloca's to malloc to fix warnings on gcc-4.2.
2007-09-16 01:56:21 +00:00
Tim-Philipp Müller
af6eee1084 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
Fix typo and compilation on big endian systems.
2007-08-29 14:22:04 +00:00
Jan Schmidt
fc50d2dc64 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
2007-08-24 15:28:33 +00:00
Tim-Philipp Müller
662c557cc2 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
Try to get devic-name from device string first, and from handle only
as fallback (seems to yield better results and is more robust
against buggy probing code on the application side).
2007-02-09 09:58:28 +00:00
Julien Puydt
880da4d8f1 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
Original commit message from CVS:
Based on patch by: Julien Puydt <julien.puydt at laposte net>
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
(gst_alsa_find_device_name):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
Improve device-name detection a bit, especially in the case where
the device is not actually open (#405020, #405024). Move common code
into gstalsa.c instead of duplicating it.
2007-02-08 15:43:26 +00:00
Wim Taymans
5fd36709af ext/alsa/gstalsa.c: Small code cleanup.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
Small code cleanup.
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
(gst_alsa_mixer_new):
Remove hack that always set the device to hw:0*.
Properly find the card name for whatever device was configured.
Do some better debugging.
Fixes #350784.
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_change_state):
Cleanups.
Handle setting of a NULL device name better.
2006-08-13 14:34:48 +00:00
Tim-Philipp Müller
ea41bfefd7 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
(gst_alsa_detect_channels):
* ext/alsa/gstalsasink.c:
Add support for cards that (only) do more than 8 channels,
like the Delta 44 (#345188).
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
unspecified channel position and cannot be combined with any
of the other audio channel positions; adjust position layout
checks accordingly (#345188).
2006-08-03 14:16:06 +00:00
Tim-Philipp Müller
e953cf18fd ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
Fix typo, so that alsasink also advertises 8 channels
if that's supported (tags: can, worms, open, alsa, ph34r).
2006-07-17 12:33:42 +00:00
Tim-Philipp Müller
9d3a69fc58 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_formats), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsa_detect_channels),
(gst_alsa_probe_supported_formats):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
Refactor and improve caps probing code: probe signedness
when we probe the supported formats/widths; set endianness
to the one we actually probed for (ie. cpu endianness).
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
(gst_alsasrc_close):
* ext/alsa/gstalsasrc.h:
Implement caps probing for alsasrc.
2006-05-16 15:52:17 +00:00
Andy Wingo
b05796c9d9 ext/alsa/: Port to 0.9.
Original commit message from CVS:
2005-08-19  Andy Wingo  <wingo@pobox.com>

* ext/alsa/gstalsamixertrack.h:
* ext/alsa/gstalsamixertrack.c:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixeroptions.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Port to 0.9.

* ext/alsa/Makefile.am: Build mixer, mixeroptions, mixertracks.
Remove gstalsa.c and alsaclock. No more cruft here.
2005-08-19 16:13:54 +00:00
Thomas Vander Stichele
e571f069d1 renamed to actual element names, so much nicer to look at
Original commit message from CVS:

* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
2005-08-05 18:51:29 +00:00
Wim Taymans
851547e321 ext/alsa/: Implement alsasink with simple open/write/close API.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_get_caps),
(gst_alsa_fixate_to_mimetype), (gst_alsa_fixate_field_nearest_int),
(gst_alsa_link), (gst_alsa_close_audio):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
(gst_alsasink_dispose), (gst_alsasink_base_init),
(gst_alsasink_class_init), (gst_alsasink_init),
(gst_alsasink_getcaps), (set_hwparams), (set_swparams),
(alsasink_parse_spec), (gst_alsasink_open), (gst_alsasink_close),
(xrun_recovery), (gst_alsasink_write), (gst_alsasink_delay),
(gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Implement alsasink with simple open/write/close API.
Make alsa dir build by disabling compilation of code.
2005-04-28 16:19:06 +00:00
Ronald S. Bultje
705aac4125 Add support for AMR-NB (mobile phone audio format; #155163, #163286).
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/amrnb/Makefile.am:
* ext/amrnb/amrnb.c: (plugin_init):
* ext/amrnb/amrnbdec.c: (gst_amrnbdec_get_type),
(gst_amrnbdec_base_init), (gst_amrnbdec_class_init),
(gst_amrnbdec_init), (gst_amrnbdec_link), (gst_amrnbdec_chain),
(gst_amrnbdec_state_change):
* ext/amrnb/amrnbdec.h:
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_get_type),
(gst_amrnbparse_base_init), (gst_amrnbparse_class_init),
(gst_amrnbparse_init), (gst_amrnbparse_formats),
(gst_amrnbparse_querytypes), (gst_amrnbparse_query),
(gst_amrnbparse_handle_event), (gst_amrnbparse_reserve),
(gst_amrnbparse_loop), (gst_amrnbparse_state_change):
* ext/amrnb/amrnbparse.h:
Add support for AMR-NB (mobile phone audio format; #155163, #163286).
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add AMR-NB/-WB raw formats.
* ext/alsa/gstalsa.c: (gst_alsa_link):
Keep valid time when changing format.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
Add some more format-specific options (#140141, #143555, #155163).
2005-01-28 10:36:12 +00:00
Ronald S. Bultje
af9b02817f ext/alsa/gstalsa.c: Fix for if items are already in list...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
Fix for if items are already in list...
2005-01-13 11:46:19 +00:00
Ronald S. Bultje
dc9ff9e5c3 ext/alsa/gstalsa.c: Fix dmix.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
Fix dmix.
2005-01-11 12:36:22 +00:00
Stéphane Loeuillet
dbebd65ced TODO: delete this file, it is by far outdated
Original commit message from CVS:
* TODO:
delete this file, it is by far outdated
* ext/alsa/gstalsa.1: remove
* ext/alsa/gstalsa.c: (add_rates), (add_channels), (gst_alsa_caps),
(gst_alsa_check_sample_rates), (gst_alsa_rates_probe),
(gst_alsa_get_caps):
Add HW probing for supported sample rates. Fixes #161704
2005-01-10 17:01:37 +00:00
Stéphane Loeuillet
6686d9e7ee ext/alsa/gstalsa.*: Add HW probing for period_count/size and buffer_size MIX/MAX
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_get_caps):
* ext/alsa/gstalsa.h:
Add HW probing for period_count/size and buffer_size MIX/MAX
Adjust default/user defined value if out of bounds
Should fix bug #162024
2005-01-10 04:09:43 +00:00
Ronald S. Bultje
71c41d27f8 ext/alsa/gstalsa.c: Reset variables on READY.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Reset variables on READY.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
(gst_matroska_mux_loop):
Require data before writing header.
2005-01-09 10:30:58 +00:00
Ronald S. Bultje
df80b706a7 configure.ac: Fix indentation, fix v4l2 plugin detection.
Original commit message from CVS:
* configure.ac:
Fix indentation, fix v4l2 plugin detection.
* ext/Makefile.am:
Fix libmms location (Maciej, use diff -u!).
* ext/alsa/gstalsa.c: (gst_alsa_init):
Initialize caps cache to NULL.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Only change state on audiosink if it exists.
2004-12-29 13:27:45 +00:00
Benjamin Otte
cffb318704 ext/alsa/gstalsa.*: refactor big chunks of the core caps negotiation code to make it a lot faster, because people cla...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
refactor big chunks of the core caps negotiation code to make it
a lot faster, because people claim it's really slow
(actually, just cache the getcaps when the device is opened)
2004-12-06 16:10:06 +00:00
Martin Soto
a931717c2e ext/alsa/: Make alsasink actually honor gst_element_set_clock and use that clock instead of ist internal one.
Original commit message from CVS:
2004-11-28  Martin Soto  <martinsoto@users.sourceforge.net>

* ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c (gst_alsa_set_clock):
Make alsasink actually honor gst_element_set_clock and use that
clock instead of ist internal one.
2004-11-28 13:35:44 +00:00
Ronald S. Bultje
e71527d6d9 ext/alsa/gstalsa.c: Don't omit the last (which incase of dmix is the only :) ) channel count. Don't set channels if <...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
Don't omit the last (which incase of dmix is the only :) )
channel count. Don't set channels if <= 2.
2004-11-26 11:47:24 +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
Benjamin Otte
1cb0235a08 ext/alsa/gstalsa.c: add debugging
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
add debugging
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
do a wait when we enter the loop func with no data available to
write instead of getting into an 100% CPU loop by just returning and
being called again by the scheduler
2004-11-13 01:08:31 +00:00
Ronald S. Bultje
05b7b2845b ext/alsa/gstalsa.c: Only set hardware parameters *after* negotiation. Before negotiation, it will set ANY and that se...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Only set hardware parameters *after* negotiation. Before
negotiation, it will set ANY and that seems to cause crashes
(see e.g. #151288, #153227).
2004-11-11 10:27:01 +00:00
Benjamin Otte
37af33bdda ext/alsa/gstalsa.c: buffer-frames property was missing
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
2004-11-09 06:08:22 +00:00
Jan Schmidt
4cf67a0834 Fixes a bunch of problems with finalize and dispose functions, either assumptions that dispose is only called once, o...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
(gst_alsa_finalize):
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
(gst_cdaudio_finalize):
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
(cdparanoia_finalize):
* ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
* ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_finalize):
* ext/flac/gstflacdec.c: (gst_flacdec_class_init),
(gst_flacdec_finalize):
* ext/flac/gstflacenc.c: (gst_flacenc_class_init),
(gst_flacenc_finalize):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
(gst_gnomevfssink_finalize):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
(gst_gnomevfssrc_finalize):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
(gst_fameenc_finalize):
* ext/nas/nassink.c: (gst_nassink_class_init),
(gst_nassink_finalize):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
(gst_sdlvideosink_class_init):
* ext/sndfile/gstsf.c: (gst_sf_dispose):
* gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
* gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
* gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_x_window_listener_dispose):
* gst/audioscale/gstaudioscale.c:
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
(play_on_demand_finalize):
* gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
(cdplayer_finalize):
* sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
(gst_glimagesink_class_init):
* sys/oss/gstosselement.c: (gst_osselement_class_init),
(gst_osselement_finalize):
* sys/oss/gstosssink.c: (gst_osssink_dispose):
* sys/oss/gstosssrc.c: (gst_osssrc_dispose):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
Fixes a bunch of problems with finalize and dispose functions,
either assumptions that dispose is only called once, or not calling
the parent class dispose/finalize function
2004-11-01 14:43:38 +00:00
Benjamin Otte
fac2d57868 fixes for G_DISABLE_ASSERT and friends
Original commit message from CVS:
* examples/dynparams/filter.c: (ui_control_create):
* examples/gstplay/player.c: (print_tag):
* ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
* ext/gdk_pixbuf/gstgdkanimation.c:
(gst_gdk_animation_iter_may_advance):
* ext/jack/gstjack.c: (gst_jack_request_new_pad):
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
(tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
* ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
* gst-libs/gst/media-info/media-info-test.c: (print_tag):
* gst/sine/demo-dparams.c: (main):
* gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
* testsuite/alsa/formats.c: (create_pipeline):
* testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
fixes for G_DISABLE_ASSERT and friends
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(plugin_init):
require mp3 typefinding to have at least MIN_HEADERS valid headers
add typefinding for AAC adts files
2004-08-03 14:28:12 +00:00
Benjamin Otte
f4bbdba991 ext/alsa/gstalsa.c: disable some of the debugging code for now. Writing debugging to a buffer is broken in current al...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_open_audio),
(gst_alsa_sw_params_dump), (gst_alsa_hw_params_dump),
(gst_alsa_close_audio):
disable some of the debugging code for now. Writing debugging to a
buffer is broken in current alsalib releases.
2004-07-15 20:32:41 +00:00