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197 commits

Author SHA1 Message Date
Sebastian Dröge
e0f9b4fffc audioconvert: Port to the new multichannel caps 
audioconvert still needs support for mixing all the new
channel positions, see:
https://bugzilla.gnome.org/show_bug.cgi?id=666506
 2012-01-05 10:34:19 +01:00
René Stadler
5f3c8eb680 audioconvert, videoconvert: fix caps leak in transform_caps 2011-11-12 01:38:37 +01:00
Vincent Penquerc'h
7ca4b51b01 audioconvert: truncate caps in _fixate 
Otherwise the resulting caps may not be fixed.
 2011-11-10 14:38:09 +00:00
Edward Hervey
1c10fbcd33 audioconvert: We can handle channels conversion 2011-10-17 12:28:58 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Josep Torra
5629ed74b3 Fix debug statements 
Fixes build on MacOSX

Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
 2011-08-10 11:15:41 +02:00
Sebastian Dröge
4fcd621101 audioconvert: Use new gst_caps_is_subset_structure() API 
This prevents one copy of every structure and creating a new caps
instance.
 2011-05-27 14:10:50 +02:00
Sebastian Dröge
d590bce5f7 audioconvert: Optimize transform_caps() 
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.

This makes gst_pad_get_caps() on an audiotestsrc ! audioconvert !
audioconvert ! audioconvert ! fakesink pipeline about 1.7 times faster.
 2011-05-27 13:13:42 +02:00
Sebastian Dröge
d8e0af1fc1 gst: Update for the GstBaseTransform::transform_caps() changes 2011-05-27 12:13:14 +02:00
Sebastian Dröge
a9b134d1a9 Merge branch 'master' into 0.11 
Conflicts:
	docs/plugins/gst-plugins-base-plugins.hierarchy
	docs/plugins/gst-plugins-base-plugins.interfaces
	docs/plugins/gst-plugins-base-plugins.prerequisites
 2011-05-20 12:26:57 +02:00
Stefan Kost
f514be993c audioconvert: cleanup helper code 
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
 2011-05-19 23:41:08 +03:00
Sebastian Dröge
c020add91e audioconvert: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps 
Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
 2011-05-09 13:05:12 +02:00
Sebastian Dröge
f10a8f0986 gst: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 11:35:53 +02:00
Wim Taymans
3b03e23559 plugins: port some plugins to the new memory API 2011-03-27 16:35:28 +02:00
Benjamin Otte
5e21fa5e0e gst_element_class_set_details => gst_element_class_set_details_simple 
Also change my email from the old university one to the current one.
 2010-03-16 17:41:50 +01:00
Stefan Kost
bbb531619c audioconvert: remove unused array 2009-11-16 22:51:17 +02:00
Stefan Kost
319baefeba audioconvert: track active conversion in perf log 2009-10-12 21:43:42 +03:00
Edward Hervey
8cd1b5209b gst: Remove dead assignments and resulting unused variables 2009-08-08 15:54:02 +02:00
Stefan Kost
8b24a3a057 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Remove now obsolete note in the docs.
 2008-07-11 18:06:33 +00:00
Stefan Kost
2b33c755b6 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... 
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* 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.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
 2008-07-10 21:06:06 +00:00
Sebastian Dröge
fdd708c418 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
 2008-05-30 08:42:17 +00:00
Sebastian Dröge
fcda3964dc gst/audioconvert/gstaudioconvert.c: Fix logic in last commit. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
 2008-05-20 12:26:32 +00:00
Sebastian Dröge
d76c4b4c65 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
 2008-05-20 12:15:34 +00:00
Sebastian Dröge
b5a5d64713 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
 2008-05-20 08:12:19 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel... 
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
 2008-05-06 12:12:16 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... 
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
 2008-03-22 15:00:53 +00:00
Sebastian Dröge
88136fc11a gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_create_silence_buffer),
(gst_audio_convert_transform):
Make audioconvert GAP-aware by outputting silence buffers when the
input has the GAP flag set. This is up to 8x faster.
Based on a patch by Stefan Kost. Fixes bug #517813.
 2008-03-21 15:58:44 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... 
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
 2008-03-03 06:04:31 +00:00
Tim-Philipp Müller
5861f366a0 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes #430677.
 2007-10-31 17:54:48 +00:00
Sebastian Dröge
dbb857b93b gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
 2007-06-28 20:37:58 +00:00
Sebastian Dröge
84c824b952 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
 2007-04-24 18:58:25 +00:00
René Stadler
6ac8ff9ec3 with some minor changes 
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
 2007-03-29 18:42:34 +00:00
Michael Smith
e1544977a6 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
 2007-03-27 11:31:17 +00:00
Michael Smith
3bc107dd77 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
 2007-03-16 17:29:09 +00:00
Michael Smith
5759241eb4 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
 2007-03-16 16:42:23 +00:00
Thomas Vander Stichele
e81b9ec719 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
fix error category and translatable string
 2007-03-09 12:22:53 +00:00
Stefan Kost
2c67c89457 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes #339837 even more.
 2007-02-28 11:47:45 +00:00
Tim-Philipp Müller
b827e4d539 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
 2007-02-07 13:05:01 +00:00
Stefan Kost
66727fe484 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Also mention that a conversion from double to float is suboptimal still.
 2007-02-06 14:00:31 +00:00
Tim-Philipp Müller
b7cf10eb6d gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
 2007-02-02 11:21:48 +00:00
Stefan Kost
36180a5f3d gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837) 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
 2007-02-02 09:48:53 +00:00
Wim Taymans
75d5fcb62e gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Don't fail on 0 sized buffers. Fixes #396835.
 2007-01-30 11:29:17 +00:00
Tim-Philipp Müller
f306655319 gst/: A few array const-ifications. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
* gst/videotestsrc/videotestsrc.h:
A few array const-ifications.
 2006-09-23 15:24:55 +00:00
Stefan Kost
f2fbfdc124 gst/audioconvert/gstaudioconvert.c: Get structure-name just once. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(gst_audio_convert_transform_caps):
Get structure-name just once.
 2006-09-17 11:24:21 +00:00
Stefan Kost
c2d7af84c1 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
 2006-08-20 13:05:44 +00:00
Tim-Philipp Müller
8a223b920d gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
Float caps shouldn't have a "signed" field.
 2006-08-10 13:01:31 +00:00
Michael Smith
8e09be1bd9 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform  can make better decisions on what caps to
negotiate.
 2006-05-29 11:04:48 +00:00
Stefan Kost
e972defd3e make GstElementDetails const 
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
 2006-04-28 19:46:37 +00:00
j^
08047f5cfe better/unified long descriptions 
Original commit message from CVS:
Patch by: j^ <j at bootlab dot org>
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
better/unified long descriptions
Fixes #336477
 2006-03-29 14:00:08 +00:00
Stefan Kost
2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
 2006-03-24 10:42:11 +00:00
Wim Taymans
af09257fd0 docs/plugins/: Add audioresample to docs. 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add audioresample to docs.
* gst/audioconvert/gstaudioconvert.c:
Add revision date.
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_base_init), (gst_audioresample_class_init),
(gst_audioresample_init), (gst_audioresample_dispose),
(audioresample_get_unit_size), (audioresample_transform_caps),
(resample_set_state_from_caps), (audioresample_transform_size),
(audioresample_set_caps), (audioresample_event),
(audioresample_do_output), (audioresample_transform),
(audioresample_pushthrough), (gst_audioresample_set_property),
(gst_audioresample_get_property), (plugin_init):
* gst/audioresample/gstaudioresample.h:
Added docs.
Small code cleanups.
 2006-03-02 18:23:55 +00:00
Thomas Vander Stichele
5f83aa7dfa expand tabs 
Original commit message from CVS:
expand tabs
 2005-12-06 19:42:02 +00:00
Wim Taymans
201fd910de gst/audioconvert/gstaudioconvert.c: Post errors. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
Post errors.
 2005-12-02 10:17:35 +00:00
Thomas Vander Stichele
aa454b5383 remove sinesrc some more 
Original commit message from CVS:
remove sinesrc some more
 2005-11-23 15:36:58 +00:00
Jan Schmidt
1cc82e9138 Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) 
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_fixate):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_caps):
* gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_src_fixate):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
* gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
 2005-11-21 14:29:53 +00:00
Tim-Philipp Müller
9136b3d6aa gst/audioconvert/gstaudioconvert.c: Fix typo in docs. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix typo in docs.
 2005-11-08 12:18:14 +00:00
Wim Taymans
fc8ce00673 Bye bye buffer-frames. 
Original commit message from CVS:
* check/elements/audioconvert.c:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_identification_packet), (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (raw_caps_factory):
* gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
* gst/volume/gstvolume.c:
Bye bye buffer-frames.
 2005-10-19 17:02:46 +00:00
Thomas Vander Stichele
d9d1b4a934 some documentation for audioconvert 
Original commit message from CVS:
some documentation for audioconvert
 2005-09-23 14:41:31 +00:00
Wim Taymans
b6bc76642a gst/audioconvert/gstaudioconvert.c: And enable 24 bits mode as well.. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
And enable 24 bits mode as well..
 2005-09-15 13:52:27 +00:00
Jan Schmidt
0f4fa24d8e check/: Add extra tests for basetransform based components. 
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.
 2005-09-09 17:53:47 +00:00
Thomas Vander Stichele
a23795d4d3 disable 24 bit until it gets fixed 
Original commit message from CVS:
disable 24 bit until it gets fixed
 2005-09-02 23:16:15 +00:00
Andy Wingo
c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL. 
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
 2005-08-28 17:52:45 +00:00
Wim Taymans
98fbd82d1c gst/audioconvert/: Oops, allocate enough space to perform the channel mix. 
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):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
Oops, allocate enough space to perform the channel mix.
 2005-08-26 17:30:41 +00:00
Wim Taymans
ceb84de916 gst/audioconvert/: Cleanups, librarify a bit, optimize, better negotiation and more. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* 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):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
(gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
(gst_audio_convert_transform_ip), (gst_audio_convert_transform):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_fill_identical),
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
(gst_channel_mix_mix):
* gst/audioconvert/gstchannelmix.h:
Cleanups, librarify a bit, optimize, better negotiation and more.
 2005-08-26 15:43:56 +00:00
Thomas Vander Stichele
43332aed85 plug some leaks 
Original commit message from CVS:
plug some leaks
 2005-08-25 17:32:34 +00:00
Thomas Vander Stichele
41a43b86a8 port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes 
Original commit message from CVS:
port audioconvert to basetransform
fix ffmpegcsp and videoscale for basetransform changes
 2005-08-24 13:32:52 +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
a6d89f51fa gst/audioconvert/gstaudioconvert.c: Convert me to BaseTransform!! help.. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
Convert me to BaseTransform!! help..
 2005-07-29 17:07:39 +00:00
Wim Taymans
567802ca2c gst/audioconvert/gstaudioconvert.c: Timestamp buffers correctly. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
Timestamp buffers correctly.

* gst/playback/gstplaybin.c: (gen_video_element):
Make internal fakesink silent.
 2005-07-16 13:58:21 +00:00
Wim Taymans
a46a991d26 ext/theora/theoradec.c: Prepare for better timestamp fix later. 
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.

* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first

* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.

* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
 2005-07-06 15:14:38 +00:00
Andy Wingo
3a7f5a2e06 gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate): No refcount leakage. 
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
 2005-07-04 10:40:17 +00:00
Andy Wingo
1f40231de5 configure.ac: Enable -Werror. 
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* configure.ac: Enable -Werror.

* ext/theora/theoradec.c (theora_dec_src_getcaps):
* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate):
* gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int)
(gst_audio_convert_fixate):
* gst/sine/gstsinesrc.c (gst_sinesrc_src_fixate)
(gst_sinesrc_create): Fixate func changes.

* sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
(gst_ximagesink_buffer_alloc): Unused var.
 2005-07-04 10:37:25 +00:00
Andy Wingo
55d437af1c gst/: Ghost pad API fixes. 
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/gconf/gconf.c:
* gst/playback/test.c:
* gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
fixes.

* gst/audioconvert/gstaudioconvert.c: RPAD fixes.

* ext/theora/theoraenc.c (theora_enc_chain):
* ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.

* ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
RealPad.
 2005-06-08 22:18:05 +00:00
Wim Taymans
5474600d4f gst-libs/gst/audio/: Various small cleanups. 
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.
 2005-05-25 19:52:14 +00:00
Ronald S. Bultje
78016b40cf gst/audioconvert/gstaudioconvert.c: Implement instant setup switching. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_chain), (gst_audio_convert_link_src),
(gst_audio_convert_setcaps):
Implement instant setup switching.
 2005-05-23 17:28:02 +00:00
Wim Taymans
04fa67937d Leak fixes in oggdemux. 
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
(gst_ogg_demux_chain_unlocked):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_change_state):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_init),
(gst_ffmpegcsp_bufferalloc), (gst_ffmpegcsp_chain),
(gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
(gst_ffmpegcsp_get_property):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_imagepool_clear),
(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_free),
(gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id):
Leak fixes in oggdemux.
Some cleanups in audioconvert.
Make passthrough work along with buffer_alloc etc.
Make buffer_alloc and buffer recycling actually work in
xvimagesink.
 2005-05-17 17:41:32 +00:00
David Schleef
d90ee5bfa3 Port from GstData to GstMiniObject. 
Original commit message from CVS:
Port from GstData to GstMiniObject.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
(gst_ogg_mux_collected):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_comment_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_set_header_on_caps), (theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_comment_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/volume/gstvolume.c: (volume_transform):
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
(gst_ximagesink_buffer_alloc):
* sys/ximage/ximagesink.h:
 2005-05-16 15:35:52 +00:00
Wim Taymans
8aeaf8ed14 Make caps writable before writing to it. 
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
Make caps writable before writing to it.
Fix negotiation in audioconvert some more.
 2005-05-09 17:07:27 +00:00
Wim Taymans
3ec8704c9e Fixed negotiation wrt _peer_get_caps() 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
* gst/sine/Makefile.am:
* gst/sine/gstsinesrc.c: (gst_sinesrc_get_type),
(gst_sinesrc_class_init), (gst_sinesrc_init),
(gst_sinesrc_src_fixate), (gst_sinesrc_setcaps),
(gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start),
(gst_sinesrc_update_freq):
* gst/sine/gstsinesrc.h:
* gst/tcp/gstmultifdsink.c:
* sys/xvimage/xvimagesink.c:
Fixed negotiation wrt _peer_get_caps()
Some cleanups.
 2005-05-06 17:13:49 +00:00
Wim Taymans
3e5febcdd1 gst/: Don't ignore _push() return values. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_change_state), (gst_audio_convert_channels):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
(gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
(gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
(gst_videotestsrc_init), (gst_videotestsrc_loop):
Don't ignore _push() return values.
Make sure no processing is done when shutting down.
Videotestsrc pad activation fix.
 2005-05-05 09:49:08 +00:00
Wim Taymans
468d6d4326 gst/audioconvert/: Link against audio libs. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_channels):
Link against audio libs.
Fix audio convert plugin.
 2005-04-20 09:45:33 +00:00
Wim Taymans
1dae961cbf Plugin port to 0.9, ogg/theora playback should work in the seek example now. 
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
 2005-03-31 09:43:49 +00:00
Benjamin Otte
c196377ff6 gst/audioconvert/gstaudioconvert.c: create channel conversion matrix when linking 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(gst_audio_convert_channels):
create channel conversion matrix when linking
* gst/audioconvert/.cvsignore:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/channelmixtest.c: (main):
add (ugly) test that ensures stereo <=> mono conversion works
correctly
 2005-02-13 17:39:22 +00:00
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
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
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
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