Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_class_init), (arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Better algorith for the center frequencies. Subtract band filters from
input for negative gains. Rework the gain mapping.
Original commit message from CVS:
2007-07-19 Julien MOUTTE <julien@moutte.net>
* configure.ac:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
(gst_flv_demux_cleanup), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
(gst_flv_demux_seek_to_prev_keyframe), (gst_flv_demux_loop),
(gst_flv_demux_sink_activate),
(gst_flv_demux_sink_activate_push),
(gst_flv_demux_sink_activate_pull), (gst_flv_demux_sink_event),
(gst_flv_demux_change_state), (gst_flv_demux_dispose),
(gst_flv_demux_base_init), (gst_flv_demux_class_init),
(gst_flv_demux_init), (plugin_init):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (FLV_GET_BEUI24), (FLV_GET_STRING),
(gst_flv_demux_query_types), (gst_flv_demux_query),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_type), (gst_flv_parse_header):
* gst/flv/gstflvparse.h: Adds a first draft of an FLV demuxer.
It does not do seeking yet, it supports pull and push mode so
YES
you can use it to play youtube videos directly from an HTTP uri.
Not so much testing done yet but it parses metadata, reply to
duration queries, etc...
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
Add example to the docs. Fix buffer-offset-end and add some debug.
Original commit message from CVS:
Patch by: Hans de Goede <j.w.r.degoede at hhs dot nl>
* gst/modplug/gstmodplug.cc:
add several missing supported mime-types to the modplug plugin.
Fixes#456901.
Original commit message from CVS:
* gst/multifile/Makefile.am:
* gst/multifile/gstmultifile.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesink.h:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
Add .h files to be able to add it to the docs.
Original commit message from CVS:
* gst/videosignal/gstvideodetect.c: (gst_video_detect_420),
(gst_video_detect_set_property), (gst_video_detect_get_property):
* gst/videosignal/gstvideodetect.h:
Add property to adjust the center, sensitivity is now the distance from
this center.
Original commit message from CVS:
* gst/videosignal/gstvideodetect.c: (gst_video_detect_420),
(gst_video_detect_set_property), (gst_video_detect_get_property),
(gst_video_detect_class_init):
* gst/videosignal/gstvideodetect.h:
* gst/videosignal/gstvideomark.c: (gst_video_mark_draw_box),
(gst_video_mark_420), (gst_video_mark_set_property),
(gst_video_mark_get_property), (gst_video_mark_class_init):
* gst/videosignal/gstvideomark.h:
Add left and bottom offset properties to control the position of the
pattern.
Original commit message from CVS:
* examples/switch/switcher.c (my_bus_callback, switch_timer,
last_message_received, main):
* gst/switch/gstswitch.c (gst_switch_release_pad,
gst_switch_request_new_pad, gst_switch_chain, gst_switch_event,
gst_switch_set_property, gst_switch_get_property,
gst_switch_get_linked_pad, gst_switch_getcaps,
gst_switch_bufferalloc, gst_switch_dispose, gst_switch_init):
* gst/switch/gstswitch.h (switch_mutex, GST_SWITCH_LOCK,
GST_SWITCH_UNLOCK):
Add an extra lock to protect against certain variables instead of
using the object lock. Fix case where caps are different in the
sink pads causes deadlock. Update example to use different caps
on each sink pad.
Original commit message from CVS:
* ext/timidity/gsttimidity.c: (gst_timidity_loop):
* ext/timidity/gstwildmidi.c: (gst_wildmidi_loop):
* gst/tta/gstttaparse.c: (gst_tta_parse_loop):
When driving the pipeline, also post an error when we get a
not-linked flow return from downstream.
Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c:
(mpeg_util_parse_extension_packet), (mpeg_util_parse_sequence_hdr),
(mpeg_util_parse_picture_hdr):
* gst/mpegvideoparse/mpegvideoparse.c:
(mpegvideoparse_handle_sequence), (mpegvideoparse_drain_avail):
Fix some silly bugs with calculating the guard sizes.
Properly compare the old sequence header structure with the new one.
Don't error out on an invalid sequence - just ignore it.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_decode):
Printf fix in debug statement; also print the right number there.
Original commit message from CVS:
* ext/libmms/gstmms.h:
No reason to use gpointers instead of typed pointes here as far as I
can see.
* ext/mythtv/gstmythtvsrc.c:
* ext/neon/gstneonhttpsrc.c:
* gst/switch/gstswitch.c:
Don't use gtk-doc magic markers for things that aren't meant to be
parsed by gtk-doc. Makes gtk-doc complain a bit less.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use lock to protect from concurrent access.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain),
(gst_real_audio_dec_setcaps):
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain):
Don't crash when we get a buffer and our input caps haven't been set
yet; also, don't leak all the input buffers (realaudiodec only).
Original commit message from CVS:
* ext/x264/gstx264enc.c (gst_x264_enc_init_encoder):
This needs a version check.
* gst/bayer/Makefile.am:
Fix the build.
Original commit message from CVS:
* configure.ac:
* gst/bayer/Makefile.am:
* gst/bayer/gstbayer.c:
* gst/bayer/gstbayer2rgb.c:
Add a Bayer-to-RGB converter. You know you want one, uh-huh.
Partial fix for #314160.
Original commit message from CVS:
* gst/switch/gstswitch.c (ARG_ACTIVE_SOURCE, ARG_STOP_VALUE,
ARG_LAST_TS, parent_class, gst_switch_release_pad,
gst_switch_request_new_pad, gst_switch_chain, gst_switch_event,
gst_switch_set_property, gst_switch_get_property, gst_switch_getcaps,
gst_switch_dispose, gst_switch_init, gst_switch_class_init):
* gst/switch/gstswitch.h (previous_sinkpad, nb_sinkpads, stop_value,
current_start, last_ts):
Allow application to provide a stop timestamp, so a new segment
update can be sent before switching.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps),
(gst_real_audio_dec_finalize):
* gst/real/gstrealaudiodec.h:
* gst/real/gstrealvideodec.c: (open_library), (close_library):
* gst/real/gstrealvideodec.h:
Use GModule instead of using dlsym() directly. Fixes#430598.
Original commit message from CVS:
* gst/speed/gstspeed.c: (speed_src_event), (speed_sink_event),
(speed_chain), (speed_change_state):
Fix event handling a bit by replacing completely dubious code
written by someone else with completely dubious code written
by me. Should at least fix#412077 though.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Add some debug info.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_send_rtp):
Store real user name in the session.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* 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_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.
Original commit message from CVS:
* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
(gst_y4m_encode_setcaps):
* tests/check/elements/y4menc.c: (GST_START_TEST):
Plug some leaks; try to make build bot happy again.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
Original commit message from CVS:
* gst/nsf/types.h:
Rename #ifndef header guard symbol to something less generic, so
types.h doesn't get skipped over when compiling on MingW. Include
GLib headers and use those to set the endianness and the basic
types so that this isn't entirely broken for non-x86 architectures.
Original commit message from CVS:
* gst/mve/gstmvedemux.c: (gst_mve_audio_init):
Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on
MingW (no idea though why we add a BYTE_ORDER endianness field if
the audio is compressed).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
Some more custom marshallers.
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
(pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
* gst/rtpmanager/gstrtpbin.h:
Prepare for caching pt maps.
Connect to signals to collect pt maps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add request_clock_rate signal.
Use scale insteat of scale_int because the later does not deal with
negative numbers.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_chain):
* gst/rtpmanager/gstrtpptdemux.h:
Implement request-pt-map signal.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
* gst/rtpmanager/gstrtpbin.h:
Provide a clock.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Add some debug and comments.
Fix double unref() in error cases.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Try to recover from packet loss a little better.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (qtdemux_parse_samples):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Process 'ctts' atoms, which are present in AVC ISO files (.mov files
with h264 video).
Use the offset present in 'ctts' to calculate the PTS for each packet
and set the PTS on outgoing buffers.
Fixes#423283
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
Remove 'channel-positions' field when munging input caps into
1-channel output caps (I guess technically we should set the
position for each channel on the output caps if it's non-NONE,
but I'll save that as a task for another day).
Original commit message from CVS:
* gst/vmnc/vmncdec.c: (gst_vmnc_dec_class_init),
(gst_vmnc_dec_init), (vmnc_dec_finalize), (gst_vmnc_dec_reset),
(vmnc_handle_wmvi_rectangle), (render_colour_cursor),
(render_cursor), (vmnc_make_buffer), (vmnc_handle_wmvd_rectangle),
(vmnc_handle_wmve_rectangle), (vmnc_handle_wmvf_rectangle),
(vmnc_handle_wmvg_rectangle), (vmnc_handle_wmvh_rectangle),
(vmnc_handle_wmvj_rectangle), (render_raw_tile), (render_subrect),
(vmnc_handle_raw_rectangle), (vmnc_handle_copy_rectangle),
(vmnc_handle_hextile_rectangle), (vmnc_handle_packet),
(vmnc_dec_setcaps), (vmnc_dec_chain_frame), (vmnc_dec_chain),
(vmnc_dec_set_property), (vmnc_dec_get_property):
Redesign to include a parser for raw files (no timestamps in that
mode yet, though).
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_remove_pads), (gst_deinterleave_process),
(gst_deinterleave_chain):
Don't leak input buffer in chain function; maintain our own list of
source pads - there are no guarantees about the order of the list
in the GstElement struct, and we want a very specific order; lastly,
some more debugging.
Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_util_find_start_code),
(collect_packets), (set_par_from_dar), (set_fps_from_code),
(mpeg_util_parse_extension_packet), (mpeg_util_parse_sequence_hdr),
(mpeg_util_parse_picture_hdr):
* gst/mpegvideoparse/mpegpacketiser.h:
* gst/mpegvideoparse/mpegvideoparse.c:
(mpegvideoparse_handle_sequence), (mpegvideoparse_handle_picture),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (plugin_init):
* gst/mpegvideoparse/mpegvideoparse.h:
Move the MPEG specific byte parsing into the mpegpacketiser code.
Add parsing of picture types, that just feeds into a debug message
for now.
Fix some 64-bit format strings.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_init):
A 10 band EQ should be initialized to 1 bands and not to 3.
Original commit message from CVS:
* configure.ac:
* gst/mpeg1videoparse/Makefile.am:
* gst/mpeg1videoparse/gstmp1videoparse.c:
* gst/mpeg1videoparse/gstmp1videoparse.h:
* gst/mpeg1videoparse/mp1videoparse.vcproj:
* gst/mpegvideoparse/Makefile.am:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_packetiser_init),
(mpeg_packetiser_free), (mpeg_packetiser_add_buf),
(mpeg_packetiser_flush), (mpeg_find_start_code),
(get_next_free_block), (complete_current_block),
(append_to_current_block), (start_new_block), (handle_packet),
(collect_packets), (mpeg_packetiser_handle_eos),
(mpeg_packetiser_get_block), (mpeg_packetiser_next_block):
* gst/mpegvideoparse/mpegpacketiser.h:
* gst/mpegvideoparse/mpegvideoparse.c: (mpegvideoparse_get_type),
(gst_mpegvideoparse_base_init), (gst_mpegvideoparse_class_init),
(mpv_parse_reset), (gst_mpegvideoparse_init),
(gst_mpegvideoparse_dispose), (set_par_from_dar),
(set_fps_from_code), (mpegvideoparse_parse_seq),
(gst_mpegvideoparse_time_code), (gst_mpegvideoparse_flush),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (gst_mpegvideoparse_change_state),
(plugin_init):
* gst/mpegvideoparse/mpegvideoparse.h:
* gst/mpegvideoparse/mpegvideoparse.vcproj:
Port mpeg1videoparse to 0.10 and give it rank SECONDARY-1, so
that it's below existing decoders.
Rename it to mpegvideoparse to reflect that it handles MPEG-1 and
MPEG-2 now.
Re-write the parsing code so that it collects packets differently
and timestamps Picture packets correctly.
Add a list of FIXME's at the top.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.
Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init),
(gst_iir_equalizer_band_get_type),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_finalize), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_setup), (plugin_init):
* gst/equalizer/gstiirequalizer.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_class_init),
(gst_iir_equalizer_nbands_init),
(gst_iir_equalizer_nbands_set_property),
(gst_iir_equalizer_nbands_get_property):
* gst/equalizer/gstiirequalizernbands.h:
Refactor plugin into a base class and a first subclass (nband eq). The
nband eq uses GstChildProxy and is controlable. More subclasses will
follow.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
Share qtdemux debug category across all files, otherwise all debugging
in files other than qtdemux.c would end up in the default category.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_event), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
One FIXME less, by resolving message timestamps against the playback
segment.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_transform_ip):
Fix and cleanup default property values.
Add FIXMEs for stuff that looks rather wrong.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (message_handler):
* gst/spectrum/demo-osssrc.c: (message_handler):
Remove two obsolete and confusing comments.
Original commit message from CVS:
* ext/dts/gstdtsdec.c: (gst_dtsdec_init), (gst_dtsdec_sink_event):
A few small clean-ups.
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
More debug output for failure cases.
Original commit message from CVS:
* configure.ac:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Add a new plugin/library to make it easy for apps to shove
data into a pipeline.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_init):
* gst/real/gstrealvideodec.c: (gst_real_video_dec_init):
Use gst_pad_use_fixed_caps() on source pads, to avoid negotiation
errors in certain situations (e.g. dec ! cs ! ximagesink and the
imagesink window is resized); also, some minor clean-ups.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace):
Rename "values" property to "band-values" and change type into a
GValueArray, so it's more easily bindable and the range of the
values passed in is defined and checked etc.; also do some
locking.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_transform_packed_complex):
Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
* tests/icles/videocrop-test.c: (check_bus_for_errors),
(test_with_caps), (main):
Block streaming thread before changing filter caps while the
pipeline is running so that we don't get random not-negotiated
errors just because GStreamer can't handle that yet.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
(gst_multi_file_sink_class_init):
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init):
* gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer),
(gst_mve_video_palette), (gst_mve_video_code_map),
(gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create),
(gst_mve_demux_chain):
* gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk):
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/mve/mvevideodec16.c: (ipvideo_copy_block):
* gst/mve/mvevideodec8.c: (ipvideo_copy_block):
* gst/mve/mvevideoenc16.c: (mve_encode_frame16):
* gst/mve/mvevideoenc8.c: (mve_encode_frame8):
Use proper print statements.
Fixes build on mac os x.
<wingo> oo look at me my name is edward i'm hacking on macos wooo
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads):
Use fixed caps on src pads.
(gst_deinterleave_remove_pads): Remove src pads, not sink pads. I
seem to have reverse midas disease!
(gst_deinterleave_process): Proxy timestamps, offsets, durations,
and set caps on outgoing buffers. Fixes#395597, I think.
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/interleave.c (gst_interleave_init): Init the
activation mode properly.
(gst_interleave_src_setcaps, gst_interleave_src_getcaps)
(gst_interleave_init): Set a setcaps and getcaps function on the
src pad, so that we can implement pull-mode negotiation.
(gst_interleave_sink_setcaps): Renamed from
gst_interleave_setcaps, as it only does the sink logic now.
Implement both for pull-mode and push-mode.
(gst_interleave_process): Set caps on our outgoing buffer.
(gst_interleave_src_activate_pull): Fix some more bogus casts.
What is up with this.
Original commit message from CVS:
* gst/mve/gstmvedemux.c: (gst_mve_demux_get_src_query_types),
(gst_mve_demux_handle_src_query), (gst_mve_demux_handle_src_event),
(gst_mve_add_stream):
Support SEEKING query (bad news now delivered properly!); add event
function to source pads to make sure seeks aren't propagated
upstream, even if they aren't handled.
Original commit message from CVS:
2007-01-07 Andy Wingo <wingo@pobox.com>
* configure.ac:
* gst/interleave/Makefile.am:
* gst/interleave/plugin.h:
* gst/interleave/plugin.c:
* gst/interleave/interleave.c:
* gst/interleave/deinterleave.c: New elements interleave and
deinterleave, implement channel interleaving and deinterleaving.
The interleaver can operate in pull or push mode but the
deinterleaver is more like a demuxer and can only operate in push
mode.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_finalize):
Don't call the RAFreeDecoder since it randomly causes segfaults.
* gst/real/gstrealaudiodec.h:
indent properly.
Original commit message from CVS:
Patch by: Lutz Mueller <lutz@topfrose.de>
* gst/real/Makefile.am:
* gst/real/gstreal.c: (plugin_init):
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain),
(gst_real_audio_dec_setcaps), (gst_real_audio_dec_init),
(gst_real_audio_dec_base_init), (gst_real_audio_dec_change_state),
(gst_real_audio_dec_finalize), (gst_real_audio_dec_set_property),
(gst_real_audio_dec_get_property), (gst_real_audio_dec_class_init):
* gst/real/gstrealaudiodec.h:
Added RealAudio wrapper elementfactory.
Modified structures so it can also work on x86_64 using the
adequate .so .
Original commit message from CVS:
* configure.ac:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_moov):
Check for zlib and if available pass it explicitly to the linker
when linking qtdemux. If not available (or --disable-external has
been specified!), disable the bits in qtdemux that use it. Fixes
build on MingW (#392856).
Original commit message from CVS:
* configure.ac:
Real video .so are now also available for x86_64, so we can build the
Real plugin on i386 AND x86_64.
* gst/real/Makefile.am:
* gst/real/gstreal.c: (plugin_init):
New plugin file for real .so wrapper plugins.
* gst/real/gstrealvideodec.c: (gst_real_video_dec_alloc_buffer),
(gst_real_video_dec_decode), (gst_real_video_dec_chain),
(gst_real_video_dec_activate_push), (gst_real_video_dec_setcaps),
(open_library), (close_library), (gst_real_video_dec_init),
(gst_real_video_dec_base_init), (gst_real_video_dec_finalize),
(gst_real_video_dec_set_property),
(gst_real_video_dec_get_property), (gst_real_video_dec_class_init):
* gst/real/gstrealvideodec.h:
Moved RealVideo element to separate file
Cleaned up code some more.
Make it work on x86_64.
Try several possible locations for .so
Separate opening/closing libraries in separate functions.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_post_progress),
(gst_qtdemux_chain):
Don't post BUFFERING messages in streaming mode if the stream
headers are behind the movie data; instead, post "progress" element
messages as a temporary solution. Apps might get confused and do
silly things to the pipeline state if they see buffering messages
from different sources and don't realize they come from different
sources (#387160).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_chain),
(gst_qtdemux_add_stream):
Don't output g_warning for an unsupported format, just send a
GST_ELEMENT_WARNING and don't add the pad.
Fix the case where it doesn't check for a NULL pad in streaming mode.
Fixes#387137
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix crash dereferencing NULL pointer if there's no stco atom.
Fixes#387122.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event):
We don't support seeking in streaming mode, so don't even try.
Implement seeking query so apps can query seekability properly
(see #365414). Fix duration query.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add AMR-WB to the list of supported formats.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
Fix non-working redirects from inetfilm.com (handle 'alis' reference
data type as well). Fixes#378613.
