Original commit message from CVS:
2005-08-29 Andy Wingo <wingo@pobox.com>
* check/elements/audioconvert.c: Convert from native endian, not
little endian.
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),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
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):
Use realloc else we lose our original data.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (ogg_find_peek):
Another from MikeS:
During typefinding, don't support negative offsets
(offsets from the end of the stream) in our typefind->peek() function
- nothing embedded in ogg ever needs them. However, we need to recognise
those requests and reject them, otherwise we return invalid pointers.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
(vorbisdec_finalize), (vorbis_handle_type_packet):
Big shout-out to MikeS for fixing this giant memory leak.
Huzzah!
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
Original commit message from CVS:
* check/Makefile.am:
* configure.ac:
add core's plugins to the mix so that playbin works
* check/generic/states.c: (GST_START_TEST):
set a 0 timeout on pipelines, so they don't force the next
state change
* gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
(gst_play_base_bin_change_state):
remove the crappy error handling and do GST error handling
Original commit message from CVS:
* check/Makefile.am:
* check/generic/states.c: (GST_START_TEST), (states_suite), (main):
add same test as to core, it bitches out on playbin atm.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
(gst_videoscale_prepare_size), (parse_caps),
(gst_videoscale_set_caps), (gst_videoscale_get_size),
(gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
(gst_videoscale_transform):
* gst/videoscale/gstvideoscale.h:
Refactor, make use of BaseTranform really well.
Original commit message from CVS:
* check/Makefile.am:
Add CHECK_CFLAGS and LDFLAGS
* gst/playback/gstplaybasebin.c: (fill_buffer):
GST_MESSAGE_SRC became a GObject
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
Original commit message from CVS:
2005-08-24 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lelement.c (gst_v4lelement_start)
(gst_v4lelement_stop): Call _start and _stop for xoverlay instead
of _open and _close.
* sys/v4l/gstv4lxoverlay.h:
* sys/v4l/gstv4lxoverlay.c (gst_v4l_xoverlay_set_xwindow_id): Open
an Xv connection here, instead of all the time. Make Xv only be
loaded if you axe for it. Kindof a workaround for buggy behaviour
of Xv when using remote xservers (XvQueryExtension would block).
(gst_v4l_xoverlay_stop, gst_v4l_xoverlay_start): New functions,
replace the _open and _close public API. Only start the xv
connection if necessary.
(gst_v4l_xoverlay_open, gst_v4l_xoverlay_close): Made static.
Original commit message from CVS:
* examples/seeking/seek.c: (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
Small seek updates.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* ext/alsa/gstalsasink.c (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
device-name property.
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.
* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.
* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
