Age | Commit message (Collapse) | Author | Files | Lines |
|
pipelines of wavpackenc. As the wavpack stuff n...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c:
Add missing audioconverts in the example pipelines of wavpackenc. As
the wavpack stuff now needs input with 32 bit width (and random depth)
this is needed now. The example pipelines for the parser and decoder
are still fine.
|
|
gst_amrwbdec_base_init, gst_amrwbdec_class_in...
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c (gst_amrwbdec_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbdec_base_init, gst_amrwbdec_class_init):
* ext/amrwb/gstamrwbenc.c (gst_amrwbenc_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbenc_base_init, gst_amrwbenc_class_init):
* ext/amrwb/gstamrwbparse.c (gst_amrwbparse_debug, GST_CAT_DEFAULT,
_do_init, gst_amrwbparse_base_init, gst_amrwbparse_class_init):
First round of cleanups, that use GST_BOILERPLATE, GST_ELEMENT_DETAILS,
GST_DEBUG_FUNCPTR and add log-category.
|
|
Original commit message from CVS:
patch by: Stanislav Brabec <sbrabec@suse.cz>
* configure.ac:
* ext/amrwb/Makefile.am:
* ext/amrwb/amrwb-code/Makefile.am:
* ext/amrwb/amrwb-code/amrwb/Makefile.am:
* ext/amrwb/amrwb-code/amrwb/README:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbdec.h (__GST_AMRWBDEC_H__):
* ext/amrwb/gstamrwbenc.h (__GST_AMRWBENC_H__):
* ext/amrwb/gstamrwbparse.c:
* ext/amrwb/gstamrwbparse.h (__GST_AMRWBPARSE_H__):
* gst-libs/Makefile.am:
* gst-libs/ext/Makefile.am:
* gst-libs/ext/amrwb/Makefile.am:
* gst-libs/ext/amrwb/README:
Use external shared libamrwb. Fixes #423741 (with lots of cleanup).
|
|
Original commit message from CVS:
* ext/x264/gstx264enc.c (gst_x264_enc_init_encoder):
This needs a version check.
* gst/bayer/Makefile.am:
Fix the build.
|
|
#413818.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c:
Make redirection the default behavior. Fixes #413818.
|
|
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
Call bindtextdomain() to get localized strings.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
* ext/wavpack/gstwavpackparse.h:
Handle DISCONT buffers by correctly setting the DISCONT flag
on outgoing buffers when necessary.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
Send newsegment from the streaming thread.
|
|
seeking after the last sample. With the fix...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Remove old workaround that was needed when seeking after the last
sample. With the fixed error handling this works now as expected
without pushing the last sample although it wasn't requested.
|
|
handler, correctly work with stop position == -...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Handle segment seeks in the seek event handler, correctly work with
stop position == -1 and instead of stopping the task on seek just
pause it.
|
|
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_loop):
Add handling for segment seeks.
|
|
loop function. Before it was easy to get th...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer),
(gst_wavpack_parse_create_src_pad),
(gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop),
(gst_wavpack_parse_chain):
Correctly handle errors, especially in the loop function. Before it
was easy to get the task paused but no error being posted on the bus.
|
|
Original commit message from CVS:
* autogen.sh:
Require automake 1.7
* ext/alsaspdif/Makefile.am:
* ext/divx/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/neon/Makefile.am:
* ext/sdl/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/theora/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/xvid/Makefile.am:
* gst/modplug/Makefile.am:
Fix up Makefile.am accordingly.
|
|
data in quicktime because of sample rate mi...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_open_decoder):
FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in
quicktime because of sample rate mismatches.
Reenable overriding the implicit SBR behaviour (accidently changed?)
to allow playback of these files.
|
|
return codes when re- establishing links.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.
|
|
with width==32 and depth=[1,32] accept th...
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes #421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.
|
|
duration bug.
Original commit message from CVS:
2007-03-27 Julien MOUTTE <julien@moutte.net>
* ext/xvid/gstxviddec.c: (gst_xviddec_chain): Add some
debug log and fix a stupid output buffer duration bug.
|
|
'odd' widths and heights correctly yet.
Original commit message from CVS:
Patch by: Michal Benes <michal.benes at itonis tv>
Patch by: Josef Zlomek <josef.zlomek at itonis tv>
* configure.ac:
* ext/Makefile.am:
* ext/x264/Makefile.am:
* ext/x264/gstx264enc.c: (gst_x264_enc_me_get_type),
(gst_x264_enc_analyse_get_type),
(gst_x264_enc_timestamp_queue_init),
(gst_x264_enc_timestamp_queue_free),
(gst_x264_enc_timestamp_queue_put),
(gst_x264_enc_timestamp_queue_get), (gst_x264_enc_header_buf),
(gst_x264_enc_set_src_caps), (gst_x264_enc_sink_set_caps),
(gst_x264_enc_base_init), (gst_x264_enc_class_init),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_close_encoder), (gst_x264_enc_dispose),
(gst_x264_enc_sink_event), (gst_x264_enc_chain),
(gst_x264_enc_encode_frame), (gst_x264_enc_change_state),
(gst_x264_enc_set_property), (gst_x264_enc_get_property),
(plugin_init):
* ext/x264/gstx264enc.h:
Add libx264-based h264 encoder plugin (#421110). Probably doesn't
handle 'odd' widths and heights correctly yet.
|
|
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_init):
Alloc user agent string only once.
|
|
plugging loops with ranks is no clean solution...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Revert last commit, preventing infinite plugging loops with ranks
is no clean solution and in general there's no reason why one wants
to parse framed wavpack data again.
|
|
instead of bytes. This allows "wavpackenc ! wa...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Send the new segment event in time format instead of bytes. This
allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Accept framed and non-framed input, wavpackparse doesn't care. To
prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
..." pipelines.
|
|
can and should use it.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.
|
|
struct directly and not as a pointer to sav...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block):
* ext/wavpack/gstwavpackenc.h:
Put the write helpers into the GstWavpackEnc struct directly and not
as a pointer to save two small, but useless mallocs. This also makes
it possible to drop the finalize method.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
buffers the same way wavpackenc does it.
|
|
clip the buffer later and
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
BaseTransform-based elements will likely break because of wrong
unit-size. Also plug a possible memleak that happens when decoding
fails for some reason.
|
|
will not be used and could cause deadlocks.
Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.
|
|
Original commit message from CVS:
* ext/nas/nassink.c: (NAS_createFlow):
* ext/sndfile/gstsfsrc.c: (gst_sf_src_create):
Printf format string fixes.
|
|
which we will use in the future to run sele...
Original commit message from CVS:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.
|
|
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.c:
Use a general wavpack debug category for common code.
* ext/wavpack/gstwavpackstreamreader.c:
(gst_wavpack_stream_reader_set_pos_abs),
(gst_wavpack_stream_reader_set_pos_rel),
(gst_wavpack_stream_reader_write_bytes):
Use the general wavpack debug category here too and add debug
output to the functions that should not be called at all by
the wavpack library.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Change debugging category names to conform to the conventions.
|
|
Original commit message from CVS:
* ext/nas/nassink.c: (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_getcaps),
(gst_nas_sink_unprepare):
Some more cleanups/changes; use boilerplate macro.
|
|
and LIBS to Makefile.am; rename structure...
Original commit message from CVS:
* ext/nas/Makefile.am:
* ext/nas/README:
* ext/nas/nassink.c: (gst_nas_sink_get_type),
(gst_nas_sink_base_init), (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_finalize),
(gst_nas_sink_getcaps), (gst_nas_sink_prepare),
(gst_nas_sink_unprepare), (gst_nas_sink_delay),
(gst_nas_sink_reset), (gst_nas_sink_write),
(gst_nas_sink_set_property), (gst_nas_sink_get_property),
(gst_nas_sink_open), (gst_nas_sink_close), (NAS_flush),
(NAS_sendData), (NAS_EventHandler), (gst_nas_sink_sink_get_format),
(NAS_createFlow), (plugin_init):
* ext/nas/nassink.h:
Bunch of nassink clean-ups: make build by adding the right CFLAGS
and LIBS to Makefile.am; rename structure, macros and functions
according to canonical naming scheme; move some things around a bit;
use GST_CAT_DEFAULT instead of GST_CAT_* everywhere; remove README
file that didn't really contain any useful information anyway (the
useful bits have been moved into the 'host' property description).
|
|
Original commit message from CVS:
* ext/directfb/dfbvideosink.c: (gst_dfbvideosink_finalize):
Chain up in finalize.
|
|
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* ext/Makefile.am:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Move the app library to gst-libs/gst/app (duh!)
|
|
Original commit message from CVS:
Commit NAS Sink, closed bugzilla 345633
|
|
Original commit message from CVS:
* ext/dts/gstdtsdec.c: (gst_dtsdec_init), (gst_dtsdec_sink_event):
A few small clean-ups.
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
More debug output for failure cases.
|
|
downstream can do and let libdts do the dow...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* ext/dts/gstdtsdec.c: (gst_dtsdec_handle_frame),
(gst_dtsdec_change_state):
Don't do forced downmixing to stereo, but check what downstream
can do and let libdts do the downmixing based on that (#400555).
|
|
the unused ishttp member (#388050).
Original commit message from CVS:
Patch by: Lutz Mueller <lutz topfrose de>
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_class_init),
(gst_neonhttp_src_init), (gst_neonhttp_src_set_property),
(gst_neonhttp_src_set_uri), (gst_neonhttp_src_set_proxy),
(gst_neonhttp_src_send_request_and_redirect),
(gst_neonhttp_src_uri_set_uri):
* ext/neon/gstneonhttpsrc.h:
Simplify _set_uri() and _set_proxy() and remove the unused ishttp
member (#388050).
* tests/check/elements/neonhttpsrc.c: (GST_START_TEST):
Fix bogus URI to something that actually exists, otherwise we just
bypass the test (and also to something that doesn't redirect, since
neonhttpsrc doesn't seem to handle this very gracefully yet)
|
|
Original commit message from CVS:
Add patch from Bug 357055 from Chris Lord, adding support for Vorbis streams
|
|
Original commit message from CVS:
* configure.ac:
* ext/gsm/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/wavpack/Makefile.am:
* gst/equalizer/Makefile.am:
* gst/filter/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/replaygain/Makefile.am:
* gst/speed/Makefile.am:
Fix build with LDFLAGS='-Wl,-z,defs'.
|
|
Original commit message from CVS:
add missing \ in Makefile.am
|
|
timidity.cfg check.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs. Also fix typo in
timidity.cfg check.
* ext/timidity/gsttimidity.c: (plugin_init):
Also build if no config was detected at configure time.
|
|
Original commit message from CVS:
* configure.ac:
* ext/timidity/Makefile.am:
* ext/timidity/gsttimidity.c: (plugin_init):
* ext/timidity/gstwildmidi.c: (gst_wildmidi_base_init),
(gst_wildmidi_class_init), (gst_wildmidi_init),
(gst_wildmidi_src_convert), (gst_wildmidi_src_query),
(gst_wildmidi_get_upstream_size), (gst_wildmidi_get_segment),
(gst_wildmidi_get_new_segment_event), (gst_wildmidi_src_event),
(gst_wildmidi_activate), (gst_wildmidi_activatepull),
(gst_wildmidi_allocate_buffer), (gst_wildmidi_clip_buffer),
(gst_wildmidi_fill_buffer), (gst_wildmidi_get_buffer),
(gst_wildmidi_loop), (gst_wildmidi_change_state),
(gst_wildmidi_set_property), (gst_wildmidi_get_property),
(gst_wildmidi_typefind), (wildmidi_open_config), (plugin_init):
* ext/timidity/gstwildmidi.h:
Add second midi renderer. Fix some double frees and leaks. Clean up
logging.
|
|
Original commit message from CVS:
* ext/faad/gstfaad.c:
Also update the comment that describes the hack.
|
|
hacks needed.
Original commit message from CVS:
* configure.ac:
Tell the code which faad it is, so that we can adjust the hacks
needed.
* ext/faad/gstfaad.c:
Make our hacks dependent on the fadd lib in use.
|
|
Original commit message from CVS:
Patch by: Wouter Paesen <wouter@blue-gate.be>
* configure.ac:
* ext/Makefile.am:
* ext/timidity/Makefile.am:
* ext/timidity/gsttimidity.c: (gst_timidity_base_init),
(gst_timidity_class_init), (gst_timidity_init),
(gst_timidity_set_song_options), (gst_timidity_src_convert),
(gst_timidity_src_query), (gst_timidity_get_upstream_size),
(gst_timidity_get_segment), (gst_timidity_get_new_segment_event),
(gst_timidity_src_event), (gst_timidity_activate),
(gst_timidity_activatepull), (gst_timidity_allocate_buffer),
(gst_timidity_clip_buffer), (gst_timidity_fill_buffer),
(gst_timidity_get_buffer), (gst_timidity_loop),
(gst_timidity_change_state), (gst_timidity_typefind),
(plugin_init):
* ext/timidity/gsttimidity.h:
Add timitity midi render plugin (#403992)
|
|
incremented refcount.
Original commit message from CVS:
* ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_query):
GST_PAD_PARENT doesn't return a GstObject with an incremented refcount.
Switched to using gst_pad_get_parent().
|
|
Original commit message from CVS:
* ext/sndfile/gstsfsrc.c:
Fix build (installed setup).
|
|
Original commit message from CVS:
2007-02-05 Andy Wingo <wingo@pobox.com>
* ext/sndfile/Makefile.am:
* ext/sndfile/gstsfsrc.h:
* ext/sndfile/gstsfsrc.c: Port sfsrc to 0.10, pull or push, with
random access woo.
|
|
Original commit message from CVS:
2007-02-02 Andy Wingo <wingo@pobox.com>
* configure.ac:
* ext/Makefile.am
* ext/sndfile/Makefile.am:
* ext/sndfile/gstsf.c:
* ext/sndfile/gstsf.h:
* ext/sndfile/gstsfsink.c:
* ext/sndfile/gstsfsink.h: Port sfsink to 0.10. Works in pull or
push mode with interleaved float or int data.
|
|
autoplugged by autoaudiosink (which didn't hap...
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c: (plugin_init):
Set rank to NONE so that it doesn't get autoplugged by autoaudiosink
(which didn't happen previously because the klass string didn't
contain anything autoaudiosink was looking for).
|
|
reported as one sample less than it is
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_create_src_pad):
Fix a off by one that leads to the duration reported as one
sample less than it is
|
|
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c:
* gst/nsf/gstnsf.c:
Fix classification in GstElementDetails.
* ext/ladspa/gstladspa.c: (gst_ladspa_base_init),
(gst_ladspa_class_init):
Improve Klassification and reduce code slighly.
|
|
Original commit message from CVS:
* ext/ladspa/Makefile.am:
* ext/ladspa/gstladspa.c: (gst_ladspa_class_get_param_spec):
add GstController support to ladspa
|