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2007-09-19gst/real/gstrealvideodec.*: Don't generate an error for occasional decoding ↵Wim Taymans2-11/+67
errors. Original commit message from CVS: * gst/real/gstrealvideodec.c: (gst_real_video_dec_chain), (open_library), (gst_real_video_dec_init), (gst_real_video_dec_set_property), (gst_real_video_dec_get_property), (gst_real_video_dec_class_init): * gst/real/gstrealvideodec.h: Don't generate an error for occasional decoding errors. Add max-errors property. Error out when we receive max-errors in a row. Fixes #478159.
2007-09-19gst/librfb/gstrfbsrc.c: Add password property (write only)Thijs Vermeir3-11/+55
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: Add password property (write only) * gst/librfb/rfbdecoder.c: Read the reason on failure Use the password property for authentication * gst/librfb/rfbdecoder.h: Add defines for version checking
2007-09-19gst/librfb/: VNC Authentication should be working now temperaly with fake ↵Thijs Vermeir6-5/+776
password 'testtest' Original commit message from CVS: * gst/librfb/Makefile.am: * gst/librfb/d3des.c: * gst/librfb/d3des.h: * gst/librfb/rfbdecoder.c: * gst/librfb/vncauth.c: * gst/librfb/vncauth.h: VNC Authentication should be working now temperaly with fake password 'testtest'
2007-09-18gst/librfb/rfbdecoder.*: Added some documentation about security handling ↵Thijs Vermeir2-8/+48
start implementing security handling for rf... Original commit message from CVS: * gst/librfb/rfbdecoder.c: * gst/librfb/rfbdecoder.h: Added some documentation about security handling start implementing security handling for rfb 3.3
2007-09-18gst/spectrum/: Handling window resize.Stefan Kost2-29/+64
Original commit message from CVS: * gst/spectrum/demo-audiotest.c: * gst/spectrum/demo-osssrc.c: Handling window resize.
2007-09-18ChangeLog: Add missing newline.Stefan Kost3-19/+2
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)
2007-09-17Added a new property for the rfb versionThijs Vermeir4-6/+112
Original commit message from CVS: Added a new property for the rfb version
2007-09-17gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one ↵Wim Taymans6-24/+56
was created first in the ssrc demuxer. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Link to the right pads regardless of which one was created first in the ssrc demuxer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsource.c: (calculate_jitter): Improve debugging. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links): * gst/rtpmanager/gstrtpssrcdemux.h: Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2007-09-16gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.Wim Taymans6-139/+341
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): Use lock to protect variable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop): Reconstruct GST timestamp from RTP timestamps based on measured clock skew and sync offset. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_set_tail_changed), (rtp_jitter_buffer_set_clock_rate), (rtp_jitter_buffer_get_clock_rate), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek): * gst/rtpmanager/rtpjitterbuffer.h: Measure clock skew. Add callback to be notfied when a new packet was inserted at the tail. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Remove clock skew detection, it's move to the jitterbuffer now.
2007-09-15gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.Wim Taymans3-6/+44
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session): Also set NTP base time on new sessions. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Use the right lock to protect our variables. Fix some comment. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp), (gst_rtp_session_chain_send_rtp), (create_send_rtp_sink): Implement getcaps on the sender sinkpad so that payloaders can negotiate the right SSRC.
2007-09-12gst/rtpmanager/: Various leak fixes.Wim Taymans7-12/+108
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (free_session), (get_client), (free_client), (gst_rtp_bin_associate), (free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_finalize): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize): * gst/rtpmanager/rtpsession.h: Various leak fixes.
2007-09-12gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so ↵Wim Taymans8-39/+306
that we can generate better Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base), (gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp): Calculate and configure the NTP base time so that we can generate better NTP times in SR packets. Set caps on new ghostpad. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Clean debug statement. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: Add ntp-ns-base property to convert running_time to NTP time. Handle NEWSEGMENT events on send and recv RTP pads so that we can calculate the running time and thus NTP time of the packets. Simplify getting the current NTP time using the pipeline clock. Implement internal links functions. Use the buffer timestamp to calculate the NTP time instead of the clock. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links), (gst_rtp_ssrc_demux_src_query): * gst/rtpmanager/gstrtpssrcdemux.h: Implement internal links function. Calculate the diff between different streams, this might be used later to get the inter stream latency. * gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp): Simple cleanup. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr): Make the clock skew window a little bigger. Apply the clock skew to all buffers, not just one with a new timestamp. Calculate and debug sender clock drift. Use extended last timestamp to interpollate for SR reports.
2007-09-12gst/: Printf format fixes (#476128).Peter Kjellerstedt4-8/+11
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).
2007-09-07gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset ↵Sebastian Dröge1-4/+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.
2007-09-06gst/mpegtsparse/mpegtsparse.c: Fix the build (missing stdlib.h).Stefan Kost1-0/+3
Original commit message from CVS: * gst/mpegtsparse/mpegtsparse.c: Fix the build (missing stdlib.h).
2007-09-06gst/spectrum/fix_fft.c: Remove fixed point FFT as it's not used anymore.Sebastian Dröge1-453/+0
Original commit message from CVS: * gst/spectrum/fix_fft.c: Remove fixed point FFT as it's not used anymore.
2007-09-06Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, ↵Sebastian Dröge5-143/+378
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.
2007-09-05gst/real/gstrealvideodec.c: Add some more debugging.Wim Taymans1-15/+33
Original commit message from CVS: * gst/real/gstrealvideodec.c: (gst_real_video_dec_chain), (gst_real_video_dec_setcaps): Add some more debugging. Don't set LONG for width/height in caps. Set correct output buffer size when caps changed. The custom message sent to the decoder should not include the format and subformat. Fixes #471554.
2007-09-04gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with ↵Tim-Philipp Müller1-2/+3
-Wall -Werror (#473562). Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
2007-09-04Nosefart -> NES Sound FormatJohan Dahlin1-1/+1
Original commit message from CVS: Nosefart -> NES Sound Format
2007-09-04gst/nsf/gstnsf.*: Add support for (very) basic tagging.Johan Dahlin2-0/+21
Original commit message from CVS: 2007-09-03 Johan Dahlin <johan@gnome.org> * gst/nsf/gstnsf.c: (gst_nsfdec_finalize), (start_play_tune): * gst/nsf/gstnsf.h: Add support for (very) basic tagging.
2007-09-03gst/rtpmanager/: Updated example pipelines in docs.Wim Taymans12-435/+1306
Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: Updated example pipelines in docs. Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync. Set the default latency correctly. Add some more points where we can get caps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Add ts-offset property to control timestamping. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Various cleanups. Feed rtpsession manager with NTP time based on pipeline clock when handling RTP packets and RTCP timeouts. Perform all RTCP with the system clock. Set caps on RTCP outgoing buffers. * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc), (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain): * gst/rtpmanager/gstrtpssrcdemux.h: Also demux RTCP messages. * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_rb), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_send_bye), (session_start_rtcp), (session_report_blocks), (session_cleanup), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Remove the get_time callback, the GStreamer part will feed us with enough timing information. Split sync timing and RTCP timing information. Factor out common RB handling for SR and RR. Send out SR RTCP packets for lip-sync. Move SR and RR packet info generation to the source. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_new_sr), (rtp_source_get_new_rb), (rtp_source_get_last_sr): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Use caps on incomming buffers to get timing information when they are there. Calculate clock scew of the receiver compared to the sender and adjust the rtp timestamps. Calculate the round trip in sources. Do SR and RR calculations in the source.
2007-08-31gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release ↵Wim Taymans1-13/+23
buffers from the jitterbuffer so that we can h... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop): Use extended timestamp to release buffers from the jitterbuffer so that we can handle the rtp wraparound correctly.
2007-08-29gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.Wim Taymans3-10/+69
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Improve Comments. * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_parse_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink), (create_send_rtp_sink): Also parse the sink caps for clock-rate instead of only relying on the result of the signal. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Make sure we fetch the clock rate for payloads we are sending out so that we can use it for SR reports.
2007-08-29gst/switch/gstswitch.c (gst_switch_chain, gst_switch_set_property): If all ↵Zaheer Abbas Merali1-11/+114
information is known at time of setting st... Original commit message from CVS: * gst/switch/gstswitch.c (gst_switch_chain, gst_switch_set_property): If all information is known at time of setting start-time property, send new segments then.
2007-08-29gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the ↵Wim Taymans6-8/+145
session manager so that it can generate ... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_event_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: Distribute synchronisation parameters to the session manager so that it can generate correct SR packets for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time), (rtp_session_set_timestamp_sync), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Add methods for setting sync parameters. Set correct RTP time in SR packets using the sync params. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Record last RTP <-> GST timestamp so that we can use them to convert NTP to RTP timestamps in SR packets.
2007-08-28gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.Wim Taymans4-7/+68
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map): Add some more advanced example pipelines. * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_send_rtcp): Add some debug and FIXME. Release LOCK when performing session cleanup. * gst/rtpmanager/rtpsession.c: (session_report_blocks): Add some debug. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (rtp_source_send_rtp): Make sure we always send RTP packets with the session SSRC.
2007-08-28gst/dvdspu/gstdvdspu.c: Don't need this include (fixes compilation in ↵Tim-Philipp Müller1-1/+0
uninstalled setup). Original commit message from CVS: * gst/dvdspu/gstdvdspu.c: Don't need this include (fixes compilation in uninstalled setup).
2007-08-27gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer ↵Wim Taymans1-1/+14
latency into account. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): When synchronizing buffers, take peer latency into account. Don't try to add our latency to invalid peer max latency values.
2007-08-27gst/flv/gstflvdemux.c: Make sure we initialize the seek result.Julien Moutte1-0/+1
Original commit message from CVS: 2007-08-27 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull): Make sure we initialize the seek result.
2007-08-27gst/dvdspu/Makefile.am: Commit the makefile too.Jan Schmidt1-0/+12
Original commit message from CVS: * gst/dvdspu/Makefile.am: Commit the makefile too.
2007-08-27dvdspu element donated by Fluendo. It implements a DVD Sub-PictureJan Schmidt9-1323/+2123
Original commit message from CVS: * configure.ac: * gst/dvdspu/.cvsignore: * gst/dvdspu/Notes.txt: * gst/dvdspu/gstdvdspu-render.c: (dvdspu_recalc_palette), (dvdspu_update_palettes), (dvdspu_clear_comp_buffers), (dvdspu_get_nibble), (dvdspu_get_rle_code), (dvdspu_draw_rle_run), (rle_end_x), (dvdspu_render_line), (dvdspu_update_chgcol), (dvdspu_render_line_with_chgcol), (dvdspu_blend_comp_buffers), (gstdvdspu_render_spu): * gst/dvdspu/gstdvdspu.c: (dvdspu_base_init), (dvdspu_class_init), (dvdspu_init), (dvdspu_clear), (dvdspu_dispose), (dvdspu_finalize), (dvdspu_flush_spu_info), (dvdspu_buffer_alloc), (dvdspu_src_event), (dvdspu_video_set_caps), (dvdspu_video_proxy_getcaps), (dvdspu_video_event), (dvdspu_video_chain), (dvspu_handle_vid_buffer), (dvdspu_redraw_still), (gstdvdspu_parse_chg_colcon), (dvdspu_exec_cmd_blk), (dvdspu_finish_spu_buf), (dvdspu_setup_cmd_blk), (dvdspu_handle_new_spu_buf), (dvdspu_handle_dvd_event), (dvdspu_dump_dcsq), (dvdspu_advance_spu), (dvdspu_check_still_updates), (dvdspu_subpic_chain), (dvdspu_subpic_event), (dvdspu_change_state), (gstdvdspu_plugin_init): * gst/dvdspu/gstdvdspu.h: dvdspu element donated by Fluendo. It implements a DVD Sub-Picture Unit, decoding and overlaying DVD subtitles and menu graphics. * gst/mpeg2sub/.cvsignore: * gst/mpeg2sub/Makefile.am: * gst/mpeg2sub/Notes.txt: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpeg2sub/gstmpeg2subt.h: * gst/mpeg2sub/mpeg2subt.vcproj: Delete old and broken mpeg2subt element that was never ported from 0.8
2007-08-24gst/flv/gstflvdemux.c: Remove some useless ifdef.Julien Moutte1-2/+0
Original commit message from CVS: 2007-08-24 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvdemux.c: (gst_flv_demux_flush), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event), (gst_flv_demux_src_event): Remove some useless ifdef.
2007-08-24Make switch more reliable and also not lock up when sink pad caps change.Zaheer Abbas Merali1-11/+26
Original commit message from CVS: * examples/switch/switcher.c (main): * gst/switch/gstswitch.c (gst_switch_chain): Make switch more reliable and also not lock up when sink pad caps change.
2007-08-24gst/flv/gstflvdemux.c: Implement seeking in push mode.Julien Moutte2-34/+165
Original commit message from CVS: 2007-08-24 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvdemux.c: (gst_flv_demux_flush), (gst_flv_demux_cleanup), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event), (gst_flv_demux_src_event): Implement seeking in push mode. * gst/flv/gstflvdemux.h:
2007-08-24gst/mpegtsparse/: Update licences to reflect LGPL-ness of these files also.Zaheer Abbas Merali4-0/+43
Original commit message from CVS: * gst/mpegtsparse/flutspatinfo.c: * gst/mpegtsparse/flutspatinfo.h: * gst/mpegtsparse/flutspmtinfo.c: * gst/mpegtsparse/flutspmtinfo.h: Update licences to reflect LGPL-ness of these files also.
2007-08-23Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers ↵Tim-Philipp Müller12-327/+327
a GType that's different than the GstRTPF... Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.signals: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPFoo types that farsight registers (luckily GType names are case sensitive). Should finally fix #430664.
2007-08-23Port festival plugin to GStreamer-0.10 (#461377).Jonathan Matthew2-61/+113
Original commit message from CVS: Patch by: Jonathan Matthew <jonathan at kaolin wh9 net> * configure.ac: * gst/festival/Makefile.am: * gst/festival/gstfestival.c: Port festival plugin to GStreamer-0.10 (#461377).
2007-08-22gst/flv/: Handle pixel aspect ratio through metadata tags like ASF does. ↵Julien Moutte3-1/+28
Fluendo muxer supports this and Original commit message from CVS: 2007-08-22 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_pull_tag): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Handle pixel aspect ratio through metadata tags like ASF does. Fluendo muxer supports this and Flash players can support it as well this way.
2007-08-22gst/flv/: Make sure we don't try filling up the index if no times object was ↵Julien Moutte2-23/+40
parsed. Fix the way we decide to push ta... Original commit message from CVS: 2007-08-22 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag): * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we don't try filling up the index if no times object was parsed. Fix the way we decide to push tags and emit no-more-pads. Fix some printf typing in debugging.
2007-08-21gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have ↵Wim Taymans1-4/+6
no latency configured, just push the buf... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_set_property): When drop-on-latency is set but we have no latency configured, just push the buffer as fast as possible. Fix typo in comment.
2007-08-21gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling.Wim Taymans2-5/+5
Original commit message from CVS: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_get_ts_diff): * gst/rtpmanager/rtpjitterbuffer.h: Fix undefined overflow prone ts_diff handling.
2007-08-21gst/modplug/gstmodplug.cc: Fix compiler warning.Wim Taymans1-1/+1
Original commit message from CVS: * gst/modplug/gstmodplug.cc: Fix compiler warning.
2007-08-21gst/modplug/gstmodplug.cc: Don't use NULL caps for srcpad is not yet connected.Stefan Kost1-24/+29
Original commit message from CVS: * gst/modplug/gstmodplug.cc: Don't use NULL caps for srcpad is not yet connected.
2007-08-20Add mpeg transport stream parser written by:Zaheer Abbas Merali12-0/+2468
Original commit message from CVS: * configure.ac: * gst/mpegtsparse/Makefile.am: * gst/mpegtsparse/flutspatinfo.c: * gst/mpegtsparse/flutspatinfo.h: * gst/mpegtsparse/flutspmtinfo.c: * gst/mpegtsparse/flutspmtinfo.h: * gst/mpegtsparse/flutspmtstreaminfo.c: * gst/mpegtsparse/flutspmtstreaminfo.h: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtspacketizer.h: * gst/mpegtsparse/mpegtsparse.c: * gst/mpegtsparse/mpegtsparse.h: * gst/mpegtsparse/mpegtsparsemarshal.list: Add mpeg transport stream parser written by: Alessandro Decina. Includes a couple of files from the Fluendo transport stream demuxer that Fluendo have kindly allowed to be licenced under LGPL also.
2007-08-19gst/filter/: Add small comparision with the chebyshev filters in the docs.Sebastian Dröge2-2/+10
Original commit message from CVS: * gst/filter/gstbpwsinc.c: * gst/filter/gstlpwsinc.c: Add small comparision with the chebyshev filters in the docs.
2007-08-17Use generator macros for the process functions for the different sample ↵Sebastian Dröge4-178/+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.
2007-08-16gst/filter/gstbpwsinc.*: Implement latency query and only forward those ↵Sebastian Dröge3-8/+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.
2007-08-16gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.Wim Taymans3-14/+49
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Fix EOS handling. Convert some DEBUG into WARNINGs. Pause task when flushing. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink): Use system clock for RTCP session management timeouts. * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout): Release the session lock when emiting signals.
2007-08-16gst/filter/gstlpwsinc.*: Implement latency query and only forward those ↵Sebastian Dröge2-6/+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.