Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Output segment with proper 'stop' value, makes flvdemux 100% compatible
with gnonlin.
Original commit message from CVS:
patch by: Alessandro Decina
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
pat-info is now a signal not a GObject property that
gets notified.
pat-info, pmt-info now instead of passing a GObject as
a parameter, pass a GstStructure.
New signals: nit-info, sdt-info, eit-info for DVB SI information
* sys/dvb/camconditionalaccess.c:
* sys/dvb/camconditionalaccess.h:
* sys/dvb/camdevice.c:
* sys/dvb/camdevice.h:
* sys/dvb/camswclient.c:
* sys/dvb/camswclient.h:
* sys/dvb/camutils.c:
* sys/dvb/camutils.h:
Cam code now uses the pmt GstStructure passed from mpegtsparse
signals rather than the GObject.
* sys/dvb/dvbbasebin.c:
Use new signals in mpegtsparse and use GstStructures as per
mpegtsparse's modified API.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
jitterbuffer can buffer an unlimited amount of time and thus has no
max_latency requirements.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_sink_event):
Don't try to flush the decoder on EOS when it was not initialized.
Fixes#498667
Original commit message from CVS:
2007-11-21 Julien Moutte <julien@fluendo.com>
* ext/sdl/sdlaudiosink.c: (gst_sdlaudio_sink_write): Fix build
on Mac OS X. (missing format parameter)
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/mpegtsparse/mpegtspacketizer.c: (foreach_stream_clear),
(remove_all), (mpegts_packetizer_clear):
Ensure that the plugin does not crash when the property pat-info is
queried before a PAT is available. It also ensures that the PAT info is
cleared when the changing from PLAYING to READY.
Fixes#487892.
Original commit message from CVS:
Patch by: Michael Kötter <m dot koetter at oraise dot de>
* ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_set_caps),
(alsaspdifsink_get_time), (alsaspdifsink_open),
(alsaspdifsink_set_params), (alsaspdifsink_delay), (plugin_init):
Fix sample rate and clocking.
Remove buffer_time and period_time as this seems to break on some
hardware. Fixes#485462.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
don't forget to handle the offset's
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
precalculate some many used values
Original commit message from CVS:
patch by: Armando Taffarel Neto <taffarel@solis.coop.br>
* gst/librfb/gstrfbsrc.c:
Set the timestamp for the output buffers
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/replaygain/rganalysis.c: (yule_filter):
Avoid slowdown from denormals when processing near-silence input data.
Spotted by Gabriel Bouvigne. Fixes#494499.
Original commit message from CVS:
* gst/flv/gstflvparse.c:
Add mapping for Nellymoser ASAO audio codec.
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we
actually have data to read at the end of the tag. This avoids trying
to allocate negative buffers.
Original commit message from CVS:
* gst/equalizer/demo.c: (draw_spectrum):
* gst/spectrum/demo-audiotest.c: (draw_spectrum):
* gst/spectrum/demo-osssrc.c: (draw_spectrum):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
Change the meaning of the magnitude values given in the
GstMessages by spectrum to decibel instead of
decibel+threshold.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
And continue to update docs. Also include some sample code
for the n-band equalizer in the docs.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Update docs and property ranges to the real values.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c:
Now do the scaling right for real. Also initialize a previously
uninitialized variable.
Original commit message from CVS:
* gst/equalizer/demo.c:
Make default volume a bit less. Improve layout by giving more space to
the slider with big-numbers and enable fill.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
Original commit message from CVS:
* ext/gio/gstgio.c: (plugin_init):
Remove nowadays unnecessary workaround for a crash.
* ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
(gst_gio_sink_start), (gst_gio_sink_stop),
(gst_gio_sink_unlock_stop):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_unlock_stop):
* ext/gio/gstgiosrc.h:
Make the finalize function safer, clean up everything that could stay
around.
Reset the cancellable instead of creating a new one after cancelling
some operation.
Don't store the GFile in the element, it's only necessary for creating
the streams.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (main):
Use autoaudiosink instead of alsasink and use a sine wave.
* gst/spectrum/gstspectrum.c:
Fix the magnitude calculation.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
(CDshowFakeSink.CDshowFakeSink):
* gst-libs/gst/dshow/gstdshowfakesink.h: (CDshowFakeSink.m_hres):
Fix crasher in constructor due to the base class's constructor
not necessarily being NULL-safe (depends on the SDK version used
apparently; #492406).
* sys/dshowsrcwrapper/gstdshowaudiosrc.c: (gst_dshowaudiosrc_prepare):
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (gst_dshowvideosrc_set_caps):
Fix a couple of MSVC compiler warnings (#492406).
Original commit message from CVS:
* gst/equalizer/demo.c: (main):
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_class_init), (setup_filter):
Allow setting 0 as bandwidth and handle this correctly.
Also handle a bandwidth of rate/2 properly.
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_class_init):
Make it possible to generate a N-band equalizer with 1 bands. The
previous limit of 2 was caused by a nowadays replaced calculation
doing a division by zero if number of bands was 1.
