Original commit message from CVS:
* sys/glsink/glimagesink.c: (gst_glimage_sink_init_display):
Update the cached caps after opening the display so that we report
only the supported caps formats, not just the template caps.
Fixes: #439405
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Remove the event-loop-in-separate-thread modifications, because MacOSX
is $#@(*%$# ! For those wondering, the event handling needs to be done
in the main thread after all..
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
Use a separate thread/task for the cocoa event_loop, else it wouldn't
stop.
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:
* sys/directdraw/Makefile.am:
* sys/directsound/Makefile.am:
Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save
and restore the various flags in the directdraw/directsound
detection section. Apparently improves cross-compiling for win32
with mingw32 under some circumstances (#437539).
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:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
Add docs for Windows sinks.
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:
* ext/wavpack/gstwavpack.c: (plugin_init):
Call bindtextdomain() to get localized strings.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
* ext/wavpack/gstwavpackparse.h:
Handle DISCONT buffers by correctly setting the DISCONT flag
on outgoing buffers when necessary.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
Send newsegment from the streaming thread.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Remove old workaround that was needed when seeking after the last
sample. With the fixed error handling this works now as expected
without pushing the last sample although it wasn't requested.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_handle_seek_event):
Handle segment seeks in the seek event handler, correctly work with
stop position == -1 and instead of stopping the task on seek just
pause it.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer),
(gst_wavpack_parse_create_src_pad),
(gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop),
(gst_wavpack_parse_chain):
Correctly handle errors, especially in the loop function. Before it
was easy to get the task paused but no error being posted on the bus.
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:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
Commit result of running scanobj-update
Original commit message from CVS:
* configure.ac:
Don't build equalizer unless we have core from CVS (it won't
work with earlier versions due to GstChildProxy brokeness).
Also up requirements to last released core/base.
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:
* ext/faad/gstfaad.c: (gst_faad_open_decoder):
FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in
quicktime because of sample rate mismatches.
Reenable overriding the implicit SBR behaviour (accidently changed?)
to allow playback of these files.
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:
* 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:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes#421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.
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:
2007-03-27 Julien MOUTTE <julien@moutte.net>
* ext/xvid/gstxviddec.c: (gst_xviddec_chain): Add some
debug log and fix a stupid output buffer duration bug.
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:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Revert last commit, preventing infinite plugging loops with ranks
is no clean solution and in general there's no reason why one wants
to parse framed wavpack data again.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Send the new segment event in time format instead of bytes. This
allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Accept framed and non-framed input, wavpackparse doesn't care. To
prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
..." pipelines.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block):
* ext/wavpack/gstwavpackenc.h:
Put the write helpers into the GstWavpackEnc struct directly and not
as a pointer to save two small, but useless mallocs. This also makes
it possible to drop the finalize method.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
buffers the same way wavpackenc does it.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
BaseTransform-based elements will likely break because of wrong
unit-size. Also plug a possible memleak that happens when decoding
fails for some reason.
Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.
Original commit message from CVS:
* sys/osxvideo/osxvideosink.m:
Emit 'have-ns-view' message when working in embedded mode. The message
will contain a pointer to the newly created NSView.
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:
* tests/icles/equalizer-test.c: (equalizer_set_band_value),
(equalizer_set_all_band_values),
(equalizer_set_band_value_and_wait),
(equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
(main):
Port the example to new equalizer api.
Original commit message from CVS:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix leaks when running a NSApp.
Accept any kind of resolutions.
Works in fullscreen. Can maximize.
Only thing left before being able to move this to -good is documentation
and embedded window support.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Handle display mode changes during playback.
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:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.c:
Use a general wavpack debug category for common code.
* ext/wavpack/gstwavpackstreamreader.c:
(gst_wavpack_stream_reader_set_pos_abs),
(gst_wavpack_stream_reader_set_pos_rel),
(gst_wavpack_stream_reader_write_bytes):
Use the general wavpack debug category here too and add debug
output to the functions that should not be called at all by
the wavpack library.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Change debugging category names to conform to the conventions.
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/nas/nassink.c: (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_getcaps),
(gst_nas_sink_unprepare):
Some more cleanups/changes; use boilerplate macro.
Original commit message from CVS:
* ext/nas/Makefile.am:
* ext/nas/README:
* ext/nas/nassink.c: (gst_nas_sink_get_type),
(gst_nas_sink_base_init), (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_finalize),
(gst_nas_sink_getcaps), (gst_nas_sink_prepare),
(gst_nas_sink_unprepare), (gst_nas_sink_delay),
(gst_nas_sink_reset), (gst_nas_sink_write),
(gst_nas_sink_set_property), (gst_nas_sink_get_property),
(gst_nas_sink_open), (gst_nas_sink_close), (NAS_flush),
(NAS_sendData), (NAS_EventHandler), (gst_nas_sink_sink_get_format),
(NAS_createFlow), (plugin_init):
* ext/nas/nassink.h:
Bunch of nassink clean-ups: make build by adding the right CFLAGS
and LIBS to Makefile.am; rename structure, macros and functions
according to canonical naming scheme; move some things around a bit;
use GST_CAT_DEFAULT instead of GST_CAT_* everywhere; remove README
file that didn't really contain any useful information anyway (the
useful bits have been moved into the 'host' property description).
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
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:
Patch by: Young-Ho Cha <ganadist at chollian net>
* ext/dts/gstdtsdec.c: (gst_dtsdec_handle_frame),
(gst_dtsdec_change_state):
Don't do forced downmixing to stereo, but check what downstream
can do and let libdts do the downmixing based on that (#400555).
Original commit message from CVS:
Patch by: Lutz Mueller <lutz topfrose de>
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_class_init),
(gst_neonhttp_src_init), (gst_neonhttp_src_set_property),
(gst_neonhttp_src_set_uri), (gst_neonhttp_src_set_proxy),
(gst_neonhttp_src_send_request_and_redirect),
(gst_neonhttp_src_uri_set_uri):
* ext/neon/gstneonhttpsrc.h:
Simplify _set_uri() and _set_proxy() and remove the unused ishttp
member (#388050).
* tests/check/elements/neonhttpsrc.c: (GST_START_TEST):
Fix bogus URI to something that actually exists, otherwise we just
bypass the test (and also to something that doesn't redirect, since
neonhttpsrc doesn't seem to handle this very gracefully yet)
Original commit message from CVS:
* tests/check/Makefile.am:
Draw plugins in from the build tree sys/ dir, rather than
picking up the already installed versions.
Original commit message from CVS:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Disable the cocoa event loop since it's a huge memory leak. Should only
matter if the sink isn't used within an NSApp (which has already got
a coca event loop).
Remove all unused code.
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:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove include of unused headers.
* sys/waveform/gstwaveformplugin.c:
* sys/waveform/gstwaveformsink.c:
* sys/waveform/gstwaveformsink.h:
* win32/vs6/libgstwaveform.dsp:
Add a new waveform plugin which includes an audio sink
element using the WaveForm win32 API.
* win32/MANIFEST:
Add the new project file form waveform plugin.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawplugin.c:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Prepare the plugin to move to good:
Remove unused/untested code (rendering to an extern surface,
yuv format rendering).Use GST_(DEBUG/*)_OBJECT macros
Rename all functions from gst_directdrawsink to gst_directdraw_sink.
Add gtk doc section
Fix a bug in gst_directdraw_sink_show_frame, memcpy line by line
respecting destination surface stride.
* sys/directsound/gstdirectsoundplugin.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Prepare the plugin to move to good:
Rename all functions from gst_directsoundsink to gst_directsound_sink.
Add gtk doc section
* win32/common/config.h.in:
* win32/MANIFEST:
Add config.h.in
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs. Also fix typo in
timidity.cfg check.
* ext/timidity/gsttimidity.c: (plugin_init):
Also build if no config was detected at configure time.
Original commit message from CVS:
* Makefile.am:
Add win32 MANIFEST
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Clear unused code and add comments.
Remove yuv from template caps, it only supports RGB
actually.
Implement XOverlay interface and remove window and fullscreen
properties.
Add debug logs.
Test for blit capabilities to return only the current colorspace if
the hardware can't blit for one colorspace to another.
* sys/directsound/gstdirectsoundsink.c:
Add some debugs.
* win32/MANIFEST:
Add VS7 project files and solution.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectdraw.dsp:
* win32/vs6/libgstdirectsound.dsp:
* win32/vs6/libgstqtdemux.dsp:
Update project files.
Original commit message from CVS:
* configure.ac:
Tell the code which faad it is, so that we can adjust the hacks
needed.
* ext/faad/gstfaad.c:
Make our hacks dependent on the fadd lib in use.
Original commit message from CVS:
* ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_query):
GST_PAD_PARENT doesn't return a GstObject with an incremented refcount.
Switched to using gst_pad_get_parent().
Original commit message from CVS:
* configure.ac:
Increase required libsndfile version to a version that's known to
have the function sf_write_sync() to make the build bots happy.
