Age | Commit message (Collapse) | Author | Files | Lines |
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Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_change_state)
(gst_stream_selector_block, gst_stream_selector_switch): Use the
cond mechanism instead of blocked pads. Patch 8/12.
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(gst_stream_selector_wait) (gst_selector_p...
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector):
* gst/switch/gstswitch.c (gst_stream_selector_wait)
(gst_selector_pad_chain, gst_stream_selector_init)
(gst_stream_selector_dispose): Add infrastructure for new blocking
mechanism that does not use gst_pad_set_blocked, which does not
work on sink pads. Patch 7/12.
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Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_class_init)
(gst_stream_selector_get_property): Add last-stop-time readable
property. Patch 6/12.
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Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector): Add some
state variables.
* gst/switch/gstswitch.c (gst_stream_selector_push_pending_stop)
(gst_selector_pad_chain): Push any pending stop event.
(gst_stream_selector_set_active_pad)
(gst_stream_selector_set_property): Factor out setting the active
pad to a function. Close the segment of the previous active pad if
told to do so via a stop_time != GST_CLOCK_TIME_NONE.
(gst_stream_selector_switch): Implement switch vmethod. Patch 5/12.
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signal. This implementation will be replace...
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_block): Implement
the block() signal. This implementation will be replaced in future
patches, however. Patch 4/12.
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Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass): Fix
vmethod prototypes.
* gst/switch/gstswitch.c (gst_stream_selector_class_init)
(gst_stream_selector_block, gst_stream_selector_switch): Add empty
vmethod definitions. Patch 3/12.
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Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (gst_stream_selector_class_init): Add
`block' and `switch' signals.
* gst/switch/Makefile.am:
* gst/switch/gstswitch-marshal.list: Add foo to generate a
marshaller for the `switch' signal. Patch 2/12.
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registered as the "switch" plugin, with "GstS...
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h:
* gst/switch/gstswitch.c: Replace with files from
gststreamselector.[ch], registered as the "switch" plugin, with
"GstSwitch" types. Patch 1/12.
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reset on newsegment events.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_dispose),
(gst_video_parse_sink_event):
Free the adapter on dispose and correctly reset on newsegment events.
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upstream and if it can't handle the query try ...
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event),
(gst_video_parse_src_event), (gst_video_parse_src_query):
Improve duration query by first asking upstream and if it can't handle
the query try to get the duration in bytes from upstream and convert.
For seeks, try if upstream handles this already first and do our
conversion to byte format only if it doesn't and if we get a
newsegment event in time format keep it and only do our conversions
if the event has another format.
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video/x-raw-gray. Also send downstream elements d...
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c:
(gst_video_parse_format_get_type),
(gst_video_parse_endianness_get_type),
(gst_video_parse_class_init), (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property),
(gst_video_parse_format_to_fourcc),
(gst_video_parse_update_block_size), (gst_video_parse_chain),
(gst_video_parse_sink_event):
Add support for video/x-raw-rgb and video/x-raw-gray. Also send
downstream elements downstream, not upstream.
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list inside each stream with a list of su...
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
Hash streams by pid again. Add a linked list inside each
stream with a list of sub_tables. Fix multiple sections
as it was borked with my last commit.
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Original commit message from CVS:
update spec file and add two missing files for disting
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pad, implement seeking and use ANY caps f...
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_src_event), (gst_video_parse_src_query_type):
Implement a query type function for the src pad, implement seeking
and use ANY caps for the sink pad as the element doesn't care what
caps the input has and everything is handled via properties.
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Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_convert),
(gst_video_parse_sink_event):
Handle -1 values for the CONVERT query too.
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supported too and allow
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event):
Add YV12 to the pad templates as it is supported too and allow
-1 as stop position for NEWSEGMENT events.
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the build.
Original commit message from CVS:
* gst/videoparse/Makefile.am:
Add $(GST_PLUGINS_BASE_CFLAGS) to CFLAGS to fix the build.
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property):
Use g_value_[sg]et_enum() for enum properties, g_value_[sg]et_int()
gives a g_critical().
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handle queries and conversion. Works much lik...
Original commit message from CVS:
* gst/videoparse/README:
* gst/videoparse/gstvideoparse.c:
Add a bunch of features: handle format specification, handle
queries and conversion. Works much like a normal parser now.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
(gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
(gst_rtp_ssrc_demux_change_state):
Clean up the dynamic pads when going to READY.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize),
(gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize),
(rtp_session_send_bye):
* gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
Fix some leaks.
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passing them to gst_tag_list_add().
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Don't strdup (and thus leak) codec name strings when passing
them to gst_tag_list_add().
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(on_ssrc_sdes):
Post a message when the SDES infor changes for a source.
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
Update some comments.
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element event handler does something differen...
Original commit message from CVS:
Based on patch by: <mutex at runbox dot com>
* gst/videoparse/gstvideoparse.c: (gst_video_parse_src_query):
Forward the query upstream, the default element event handler does
something different. Fixes #502879.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_sdes), (rtp_session_process_sdes):
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.c:
* gst/rtpmanager/rtpstats.h:
Add signal to notify of an SDES change.
Fix object type in the signal callbacks.
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the session managers with them.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name),
(gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose SDES items as properties and configure the session managers with
them.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_set_property):
Fix SSRC property.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
* gst/rtpmanager/rtpjitterbuffer.c:
Update comment.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_set_property), (gst_rtp_session_get_property):
Define some GObject properties to set SDES and other configuration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_ssrc_sdes), (rtp_session_set_bandwidth),
(rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
(rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
(rtp_session_get_sdes_string), (obtain_source),
(rtp_session_get_internal_source), (rtp_session_process_sdes),
(rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
(is_rtcp_time):
* gst/rtpmanager/rtpsession.h:
Add signal when new SDES infor has been found for a source.
Create properties for SDES and other info.
Simplify the SDES API.
Add method for getting the internal source object of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_finalize), (rtp_source_set_property),
(rtp_source_get_property), (rtp_source_set_callbacks),
(rtp_source_get_ssrc), (rtp_source_set_as_csrc),
(rtp_source_is_as_csrc), (rtp_source_is_active),
(rtp_source_is_validated), (rtp_source_is_sender),
(rtp_source_received_bye), (rtp_source_get_bye_reason),
(rtp_source_set_sdes), (rtp_source_set_sdes_string),
(rtp_source_get_sdes), (rtp_source_get_sdes_string),
(rtp_source_get_new_sr), (rtp_source_get_new_rb):
* gst/rtpmanager/rtpsource.h:
Add GObject properties for various things.
Don't leak the bye reason.
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Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Fix list of supported and known codecs.
Emit tag with the codec name so it gets properly reported in totem and
other applications.
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Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_transform_ip):
Fix compilation.
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Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_transform_ip):
Don't process buffers in passthrough mode.
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there's no need for checking if the elemen...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
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passthrough mode to get consistent property values.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
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Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c: (crc_tab):
Mark crc values table as constant.
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sub_table_identifier, not by pid. So hash with t...
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
A sub table is identified by the pair table_id and
sub_table_identifier, not by pid. So hash with that.
* sys/dvb/dvbbasebin.c:
Make sure initial pids are added properly to filter,
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app thread when unsetting `queue-buffers', ...
Original commit message from CVS:
2007-12-05 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_switch_set_property): Don't push
buffers from app thread when unsetting `queue-buffers', it's
dangerous and the chain function will do it for us anyway.
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bus messages.
Original commit message from CVS:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Remove signals for pat, pmt, nit, eit, sdt. Replace with bus
messages.
* sys/dvb/dvbbasebin.c:
Instead of attaching to signals, use the bus messages.
Also fix up so the dvbsrc starts only outputting the info tables
like PAT, CAT, NIT, SDT, EIT instead of the whole ts.
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Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
Should fix the 64-bit build
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Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Add CoRRE encoding
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relicenced from MPL.
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
CRC check the psi pids. CRC checking code relicenced from MPL.
Thanks Fluendo.
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Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
Use glib macro for swapping
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Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Disable CopyRect encoding by default
Add RRE encoding
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Original commit message from CVS:
2007-11-26 Julien Moutte <julien@fluendo.com>
* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
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multiple times.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c:
Use dispose and finalize. Dispose can be called multiple times.
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only the changes noted in README again.
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.
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flvdemux 100% compatible with gnonlin.
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.
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notified.
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.
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have a resampler state already.
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.
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Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
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latency changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
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the latency of the resampler. These will g...
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.
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amount of time and thus has no max_latency ...
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.
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