| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
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
|
|
library now ... (#354451).
Original commit message from CVS:
Patch by: Rosfran Borges <rosfran dot borges at idnt org br>
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_start),
(gst_mythtv_src_next_program_chain):
Remove sleep calls, they've been moved into the library now ...
(#354451).
|
|
in pull mode, not the other way around. A...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query):
Fix the SEEKING query. We can seek if we are in pull mode, not the
other way around. Also set the correct format in the seeking query and
handle the case where the headers are not read yet and we can't say
anything about our seeking capabilities.
|
|
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
Fix spelling in 2 places: It's called Wavpack, not WavePack.
|
|
Original commit message from CVS:
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_set_property):
Use break here instead of goto.
|
|
deadlock in _set_property() in the error ...
Original commit message from CVS:
* ext/mythtv/gstmythtvsrc.c: (do_read_request_response),
(gst_mythtv_src_create), (gst_mythtv_src_get_position),
(gst_mythtv_src_do_seek), (gst_mythtv_src_start),
(gst_mythtv_src_next_program_chain), (gst_mythtv_src_get_size),
(gst_mythtv_src_handle_event), (gst_mythtv_src_handle_query),
(gst_mythtv_src_change_state), (gst_mythtv_src_set_property),
(gst_mythtv_src_uri_get_type):
Clean up a bit, mostly the debug statements; fix deadlock in
_set_property() in the error cases; fix up query function.
|
|
Original commit message from CVS:
2007-01-12 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_fixate)
(gst_signal_processor_ouija_caps, gst_signal_processor_prepare):
Remove fixate/ouija stuff, thankfully, due to the new
part-negotiation.txt pull-mode negotiation scheme.
(gst_signal_processor_setcaps_pull)
(gst_signal_processor_setcaps): Implement upstream set_caps pull
proxying for pull mode. Now this works: ladspa-sine-fcac !
audioconvert ! alsasink.
|
|
Original commit message from CVS:
MythTV client source plugin created.
|
|
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb),
(jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_acquire):
* ext/jack/gstjackaudiosink.h:
Improve docs.
|
|
Original commit message from CVS:
* ext/gsm/gstgsmenc.c: (gst_gsmenc_init):
Fix typo in docs.
|
|
hardcoded constants (#395536).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
(gst_wavpack_enc_correction_mode_get_type),
(gst_wavpack_enc_joint_stereo_mode_get_type):
Minor clean-up: use enum values instead of hardcoded constants (#395536).
|
|
Original commit message from CVS:
2007-01-06 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c
(gst_signal_processor_ouija_caps): Move around in the source
file...
(gst_signal_processor_prepare, gst_signal_processor_do_pulls):
Call ouija_caps in prepare() instead of do_pulls(), a bit earlier.
This allows us to have caps when we do the pad_alloc_buffer().
(gst_pad_alloc_buffer_and_set_caps): Use self->caps instead of the
pad caps, which might not be set yet.
|
|
Original commit message from CVS:
2007-01-06 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstsignalprocessor.c:
(gst_signal_processor_add_pad_from_template)
(gst_signal_processor_fixate): Add a fixate function, to assist in
pathological ladspa-sine-fcac ! fakesink can-activate-pull=true
cases.
(gst_signal_processor_prepare, gst_signal_processor_process): Add
nframes args so that getrange can tell ladspa how many frames to
process.
(gst_signal_processor_ouija_caps): setcaps needs to be called
before processing, which normally happens when chaining a buffer
to a pad. However in getrange mode with no sinks we need to check
explicitly for this condition, guess some caps to use, and use
those to setcaps(). Hence this mystical function.
(gst_signal_processor_do_pulls): Pull in bytes, not samples.
Divine the caps if necessary.
(gst_signal_processor_getrange): Interpret the length as bytes,
not samples.
(gst_signal_processor_chain): nframes=G_MAXUINT, will be limited
by incoming buffer sizes.
|
|
Original commit message from CVS:
move amrwb code
|
|
Original commit message from CVS:
move amrwb code
|
