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* gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.Wim Taymans2007-09-262-8/+20
| | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: Set timestamps on RTP buffers in interleaved mode. Mark first buffers with a DISCONT. Remove flush hack now that sync for live sources has been figured out.
* gst/udp/gstudpsrc.c: Update documentation.Wim Taymans2007-09-261-3/+4
| | | | | | Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Update documentation.
* gst/qtdemux/gstrtpxqtdepay.*: Fail if we don't know the quicktime format.Wim Taymans2007-09-262-2/+13
| | | | | | | | Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_change_state): * gst/qtdemux/gstrtpxqtdepay.h: Fail if we don't know the quicktime format.
* Add support for the new GST_TAG_COMPOSER (#459809).Tim-Philipp Müller2007-09-251-0/+1
| | | | | | | | Original commit message from CVS: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * gst/apetag/gstapedemux.c: Add support for the new GST_TAG_COMPOSER (#459809).
* gst/law/: Compulsive clean-ups: use boilerplate macros, add debug ↵Tim-Philipp Müller2007-09-256-189/+157
| | | | | | | | | | | | | | | categories, fix up things to conform to symbol nome... Original commit message from CVS: * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/alaw.c: * gst/law/mulaw-conversion.h: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nomenklatura, etc.
* gst/law/: Use static tables for A-Law decoding and encoding; this makesLaurent Glayal2007-09-252-6/+246
| | | | | | | | | | | Original commit message from CVS: Based on patch by: Laurent Glayal <spglegle yahoo fr> * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: Use static tables for A-Law decoding and encoding; this makes A-Law decoding and encoding less CPU-intensive, but increases the binary size a bit. Leaving old code around for now, selectable by a define in the code. Fixes #435435.
* gst/qtdemux/qtdemux.c: Add fourccs for MPEG2 HDV streams. Fixes #479960.Sebastian Dröge2007-09-251-0/+7
| | | | | | | Original commit message from CVS: Patch by: <j at bootlab dot org> * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add fourccs for MPEG2 HDV streams. Fixes #479960.
* Massive leak fixing, plus code cleanups.Stefan Kost2007-09-242-4/+4
| | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/audioresample/gstaudioresample.c: * ext/x264/gstx264enc.c: * gst/dvdspu/gstdvdspu.c: * gst/dvdspu/gstdvdspu.h: * gst/festival/gstfestival.c: * gst/h264parse/gsth264parse.c: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtsparse.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/nuvdemux/gstnuvdemux.c: * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.c: * sys/vcd/vcdsrc.c: Massive leak fixing, plus code cleanups.
* gst/rtp/gstrtpamrdepay.c: Set outgoing packet duration because we can. Fixes ↵Wim Taymans2007-09-211-0/+3
| | | | | | | | #478244 some more. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process): Set outgoing packet duration because we can. Fixes #478244 some more.
* gst/rtp/gstrtpL16pay.c: Removed some unused code.Wim Taymans2007-09-197-8/+26
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer): Removed some unused code. * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer): * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer): * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet), (gst_rtp_theora_pay_flush_packet): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet): Try to preserve the incomming buffer duration on the outgoing packets. Fixes #478244.
* ChangeLog: Add missing newline.Stefan Kost2007-09-182-19/+1
| | | | | | | | | | | | | | Original commit message from CVS: * ChangeLog: Add missing newline. * gst/librfb/rfbdecoder.c: Fix the build (missing stdlib.h). * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: Use basetransform segment so that it is correctly managed on flushes and start/stop. Report message timestamp as stream time, which is what an application can understand. (Yes these are adapted from wim recent level element changes)
* gst/: Fix compiler warnings shown with Forte.Jan Schmidt2007-09-172-8/+17
| | | | | | | | | | Original commit message from CVS: * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message): Fix compiler warnings shown with Forte.
* gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to ↵Wim Taymans2007-09-171-4/+18
| | | | | | | | | | configure for some reason. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams), (gst_rtspsrc_dup_printf): Give meaningfull error when all streams failed to configure for some reason.
* gst/rtp/README: Update README with the design for synchronisation rules of ↵Wim Taymans2007-09-161-22/+144
| | | | | | | | | RTP on sender and receiver. Original commit message from CVS: * gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver.
* gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the ↵Sebastian Dröge2007-09-141-42/+32
| | | | | | | | | | | | element driving the pipeline is responsible f... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_loop), (gst_wavparse_chain): Don't push EOS from the chain function, the element driving the pipeline is responsible for this. The bug this was meant to fix seems to be queue not forwarding EOS in all cases (see #476514).
* gst/level/gstlevel.*: Use basetransform segment so that it is correctly ↵Wim Taymans2007-09-132-35/+1
| | | | | | | | | | | | | managed on flushes and start/stop. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start), (gst_level_transform_ip): * gst/level/gstlevel.h: Use basetransform segment so that it is correctly managed on flushes and start/stop. Report message timestamp as stream time, which is what an application can understand.
* gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes ↵Sebastian Dröge2007-09-131-32/+42
| | | | | | | | | #476514. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos), (gst_wavparse_loop), (gst_wavparse_chain): Add EOS logic for the push-based mode too. Fixes #476514.
* gst/law/: Fix law encoder timestamps.Wim Taymans2007-09-124-13/+29
| | | | | | | | | | Original commit message from CVS: * gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain): * gst/law/alaw-encode.h: * gst/law/mulaw-encode.c: (gst_mulawenc_init), (gst_mulawenc_chain): * gst/law/mulaw-encode.h: Fix law encoder timestamps.
* gst/: Printf format fixes (#476128).Peter Kjellerstedt2007-09-121-1/+1
| | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst-libs/gst/app/gstappsink.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvparse.c: * gst/interleave/deinterleave.c: * gst/switch/gstswitch.c: Printf format fixes (#476128).
* gst/udp/gstudpsrc.c: Make udpsrc timestamp outgoing buffers based on when ↵Wim Taymans2007-09-101-0/+2
| | | | | | | | | they were received. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create): Make udpsrc timestamp outgoing buffers based on when they were received. Also make it output a segment in time.
* gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups.Stefan Kost2007-09-101-5/+8
| | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups.
* gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add ↵Haakon Sporsheim2007-09-071-1/+1
| | | | | | | | | | | | | padding between the structure fields, si... Original commit message from CVS: Patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com> * gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, since that would lead to us sending RTP packets with broken headers (as is currently the case when compiling with MSVC). Also see similar fixes in libgstrtp in gst-plugins-base. (#474616; #471194)
* gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset ↵Sebastian Dröge2007-09-071-4/+4
| | | | | | | | | | calls. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_start), (gst_spectrum_transform_ip): Use the correct parameter order for the memset calls. Thanks to Christian Schaller for noticing.
* Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, ↵Sebastian Dröge2007-09-063-132/+365
| | | | | | | | | | | | | | | | | | | | | | | | | | | | float and double, use floats for the message... Original commit message from CVS: * configure.ac: * gst/spectrum/Makefile.am: * gst/spectrum/demo-audiotest.c: (draw_spectrum), (message_handler), (main): * gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler): * gst/spectrum/gstspectrum.c: (gst_spectrum_base_init), (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_property), (gst_spectrum_get_property), (gst_spectrum_start), (gst_spectrum_setup), (gst_spectrum_message_new), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message contents, average all FFTs done in one interval for better results, use a better windowing function, allow posting the phase in the message and actually do an FFT with the requested number of bands instead of interpolating. * tests/check/elements/spectrum.c: (GST_START_TEST), (spectrum_suite): Improve the units tests by checking for a 11025Hz sine wave and add unit tests for all 4 supported sample types.
* gst/qtdemux/: Don't assume tags are encoded as UTF-8 (#473670).Tim-Philipp Müller2007-09-052-6/+16
| | | | | | | Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: Don't assume tags are encoded as UTF-8 (#473670).
* gst/udp/gstmultiudpsink.c: Add property do configure destination ↵Wim Taymans2007-09-041-9/+99
| | | | | | | | | | | | | | | address/port pairs Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_set_clients_string), (gst_multiudpsink_get_clients_string), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal), (gst_multiudpsink_add), (gst_multiudpsink_clear_internal), (gst_multiudpsink_clear): Add property do configure destination address/port pairs API:GstMultiUDPSink::clients
* gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and ↵Stefan Kost2007-09-041-18/+33
| | | | | | | | improve debugs logs. Original commit message from CVS: * gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs.
* gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration ↵Stefan Kost2007-09-042-70/+151
| | | | | | | | | | | calculations. Appropriate use of uint64_scale_int... Original commit message from CVS: * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff out of loops.
* gst/avi/gstavidemux.c: Implement seek-query.Stefan Kost2007-09-031-0/+26
| | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: Implement seek-query.
* gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP ↵Wim Taymans2007-08-291-0/+2
| | | | | | | | | | packet not wait for preroll. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_dup_printf): Use new basesink async property to make sparse RTCP packet not wait for preroll.
* gst/audiofx/Makefile.am: Dist the right file.Jan Schmidt2007-08-271-1/+1
| | | | | | Original commit message from CVS: * gst/audiofx/Makefile.am: Dist the right file.
* gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values ↵Wim Taymans2007-08-231-5/+38
| | | | | | | | | | in the POSIX locale instead of the curre... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf), (gst_rtspsrc_get_float), (gst_rtspsrc_play): Make sure we generate and parse floating point values in the POSIX locale instead of the current locale.
* gst/rtsp/gstrtspsrc.*: Fix method detection again.Wim Taymans2007-08-222-22/+64
| | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: Fix method detection again. Keep track of when we must send a Range header. Use segment values for Range, Speed and Scale headers. Parse Speed and Scale headers to update the segment values.
* gst/rtp/: Added an H263 depayloader. Fixes #369392.Wim Taymans2007-08-206-25/+545
| | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init), (gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init), (gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps), (gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property), (gst_rtp_h263_depay_get_property), (gst_rtp_h263_depay_change_state), (gst_rtp_h263_depay_plugin_init): * gst/rtp/gstrtph263depay.h: Added an H263 depayloader. Fixes #369392. * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush): Make the H263+ pay/depayloader support H263-1998 and H263-2000 payloads. Also alow plain H263 on the h263p payloaders. Fixes #465040.
* gst/filter/: Add small comparision with the chebyshev filters in the docs.Sebastian Dröge2007-08-192-2/+10
| | | | | | | Original commit message from CVS: * gst/filter/gstbpwsinc.c: * gst/filter/gstlpwsinc.c: Add small comparision with the chebyshev filters in the docs.
* gst/audiofx/: Add small comparision with the windowed sinc filters in the docs.Sebastian Dröge2007-08-194-0/+20
| | | | | | | Original commit message from CVS: * gst/audiofx/audiochebyshevfreqband.c: * gst/audiofx/audiochebyshevfreqlimit.c: Add small comparision with the windowed sinc filters in the docs.
* gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit.Wim Taymans2007-08-181-84/+48
| | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop): Refactor the udp and interleaved loop function a bit.
* gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids ↵Wim Taymans2007-08-172-27/+72
| | | | | | | | | | | | | | | deadlocks when going to PAUSED. Fixes #455... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_connection_send), (gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain), (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455808.
* gst/debug/rndbuffersize.c: Fix debug statement.Wim Taymans2007-08-171-1/+1
| | | | | | Original commit message from CVS: * gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop): Fix debug statement.
* gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC.Wim Taymans2007-08-171-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos): Fix stray %u in debug line as spotted by Saur on IRC.
* Use generator macros for the process functions for the different sample ↵Sebastian Dröge2007-08-174-178/+110
| | | | | | | | | | | | | | | | | | | | | | | | types, add lower upper boundaries for the GOb... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: * tests/check/elements/lpwsinc.c: (GST_START_TEST): Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GObject properties so automatically generated UIs can use sliders and change frequency properties to floats to save a bit of memory, even ints would in theory be enough. Also rename frequency to cutoff for consistency reasons. * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-gstrtpmanager.xml: Regenerated for the above changes.
* gst/audiofx/: Use generator macros for the process functions for the ↵Sebastian Dröge2007-08-174-114/+120
| | | | | | | | | | | | | | | | | different sample types, add lower upper boundari... Original commit message from CVS: * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_class_init): * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_class_init): Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GObject properties so automatically generated UIs can use sliders and add a note about the number of poles as a too high number of poles combined with very low or very high frequencies will produce only noise. * docs/plugins/gst-plugins-good-plugins.args: Regenerated for the property changes.
* gst/rtsp/gstrtspsrc.*: Improve timeout handling.Wim Taymans2007-08-172-75/+189
| | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property), (gst_rtspsrc_flush), (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Improve timeout handling. Use the same socket for sending and receiving RTCP packets so that some servers can track clients better. Improve connection closed handling. Try to reconnect. Don't overwrite our content base with NULL. Improve debugging. Improve range parsing and handling. Remove flushing hack now that core does the right thing.
* gst/udp/gstmultiudpsink.*: Add support for getting and setting the socket to ↵Wim Taymans2007-08-173-11/+76
| | | | | | | | | | | | | | | use. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_init), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_close), (gst_multiudpsink_add): * gst/udp/gstmultiudpsink.h: Add support for getting and setting the socket to use. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_get_property): Add support for getting the currently used socket.
* gst/filter/gstbpwsinc.*: Implement latency query and only forward those ↵Sebastian Dröge2007-08-163-8/+300
| | | | | | | | | | | | | | | | | | | | samples downstream that actually contain the ... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (gst_bpwsinc_init), (process_32), (process_64), (bpwsinc_build_kernel), (bpwsinc_push_residue), (bpwsinc_transform), (bpwsinc_start), (bpwsinc_query), (bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property): * gst/filter/gstbpwsinc.h: Implement latency query and only forward those samples downstream that actually contain the data we want, i.e. drop kernel_length/2 in the beginning and append kernel_length/2 (created by convolving the filter kernel with zeroes) to the end. * tests/check/elements/bpwsinc.c: (GST_START_TEST): Adjust the unit test for this slightly changed behaviour. * gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel): Reset residue length only when actually creating a residue.
* gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.Sebastian Dröge2007-08-1610-4/+3793
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: reviewed by: Stefan Kost <ensonic@users.sf.net> * gst/audiofx/Makefile.am: * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_mode_get_type), (gst_audio_chebyshev_freq_band_base_init), (gst_audio_chebyshev_freq_band_dispose), (gst_audio_chebyshev_freq_band_class_init), (gst_audio_chebyshev_freq_band_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_band_set_property), (gst_audio_chebyshev_freq_band_get_property), (gst_audio_chebyshev_freq_band_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_band_transform_ip), (gst_audio_chebyshev_freq_band_start): * gst/audiofx/audiochebyshevfreqband.h: * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_mode_get_type), (gst_audio_chebyshev_freq_limit_base_init), (gst_audio_chebyshev_freq_limit_dispose), (gst_audio_chebyshev_freq_limit_class_init), (gst_audio_chebyshev_freq_limit_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_limit_set_property), (gst_audio_chebyshev_freq_limit_get_property), (gst_audio_chebyshev_freq_limit_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_limit_transform_ip), (gst_audio_chebyshev_freq_limit_start): * gst/audiofx/audiochebyshevfreqlimit.h: * gst/audiofx/audiofx.c: (plugin_init): Add Chebyshev lowpass/highpass and bandpass/bandreject elements. Fixes #464800. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/audiochebyshevfreqband.c: (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband), (GST_START_TEST), (audiochebyshevfreqband_suite), (main): * tests/check/elements/audiochebyshevfreqlimit.c: (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit), (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main): Add unit tests for the chebyshev filters. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-wavpack.xml: And add docs for the chebyshev filters. While doing that also run make update in docs/plugins.
* Make ro memory to share.Stefan Kost2007-08-161-1/+1
| | | | | | | Original commit message from CVS: * ext/annodex/gstcmmltag.c: * gst/rtp/gstrtpvorbispay.c: Make ro memory to share.
* gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we ↵Wim Taymans2007-08-161-2/+11
| | | | | | | | | have data available immediatly. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Improve UDP performance by avoiding a select() when we have data available immediatly.
* gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals.Wim Taymans2007-08-164-58/+273
| | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT), (gst_rtp_dec_class_init): * gst/rtsp/gstrtpdec.h: Add (dummy) SSRC management signals. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (find_stream), (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc), (on_timeout), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add connection-speed property. Add find_stream helper functions. Handle stream EOS based on BYE messages or SSRC timeout. Returns SUCCESS from the state change function as we hide our async elements from the parent.
* gst/filter/gstlpwsinc.*: Implement latency query and only forward those ↵Sebastian Dröge2007-08-162-6/+295
| | | | | | | | | | | | | | | | | | samples downstream that actually contain the ... Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_push_residue), (lpwsinc_transform), (lpwsinc_start), (lpwsinc_query), (lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property): * gst/filter/gstlpwsinc.h: Implement latency query and only forward those samples downstream that actually contain the data we want, i.e. drop kernel_length/2 in the beginning and append kernel_length/2 (created by convolving the filter kernel with zeroes) to the end. * tests/check/elements/lpwsinc.c: (GST_START_TEST): Adjust the unit test for this slightly changed behaviour.
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