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).
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.
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.
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.
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.
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.
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.
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.
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.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_set_index),
(gst_flv_demux_get_index):
Fix locking and refcounting on the index.
Original commit message from CVS:
2007-08-14 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_adapter_flush), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_do_seek),
(gst_flv_demux_handle_seek), (gst_flv_demux_sink_event),
(gst_flv_demux_src_event), (gst_flv_demux_query),
(gst_flv_demux_change_state), (gst_flv_demux_set_index),
(gst_flv_demux_get_index), (gst_flv_demux_dispose),
(gst_flv_demux_class_init): First method for seeking in pull
mode using the index built step by step or coming from metadata.
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Parse
more metadata types and keyframes index.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
Fix a segfault with more than one channel and don't rebuild
the kernel & residue with every buffer.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
(gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Add support for a bandreject mode and allow specifying the window
function that should be used.
* gst/filter/gstlpwsinc.c:
And another small formatting fix.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
(bpwsinc_transform), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Apply the same changes to the bandpass filter:
- Support double input
- Fix processing for input with >1 channels
- Specify frequency in Hz
- Specify actual filter kernel length
- Use transform instead of transform_ip as we're working
out of place anyway
- Factor out filter kernel generation and update the filter
kernel when the properties are set
Fix bandpass filter kernel generation to actually generate
a bandpass filter by creating a highpass instead of a second
lowpass.
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Small formatting fix.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Specify the actual filter length instead of a weird
2N+1. Setting the property will round to the next odd number.
Also remove now obsolete FIXMEs.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type),
(gst_lpwsinc_class_init), (gst_lpwsinc_init),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Allow choosing between hamming and blackman window. The blackman
window provides a better stopband attenuation but a bit slower
rolloff.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (process_32), (process_64),
(lpwsinc_build_kernel):
Fix processing if the input has more than one channel.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip),
(bpwsinc_set_property), (bpwsinc_get_property):
"this" is a C++ keyword, use "self" instead.
Add TODOs and FIXMEs and remove two wrong FIXMEs.
* gst/filter/gstlpwsinc.c:
Add FIXMEs and a new TODO.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32),
(process_64), (lpwsinc_build_kernel), (lpwsinc_setup),
(lpwsinc_get_unit_size), (lpwsinc_transform),
(lpwsinc_set_property), (lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Add double support, replace "this" with "self" as the former
is a C++ keyword.
Implement the frequency property in Hz instead of fraction
of sampling frequency.
Remove some unecessary FIXMEs and add some TODOs, add some
required locking and refactor the kernel generation into a
separate function that is also called when the properties
change now.
And use BaseTransform::transform instead of transform_ip
as the convolution is done out of place anyway. Should
be done in place later.
Original commit message from CVS:
* configure.ac:
* gst/stereo/Makefile.am:
* gst/stereo/gststereo.c: (gst_stereo_base_init),
(gst_stereo_class_init), (gst_stereo_init),
(gst_stereo_transform_ip), (gst_stereo_set_property),
(gst_stereo_get_property):
* gst/stereo/gststereo.h:
Port the stereo element to GStreamer 0.10.
Original commit message from CVS:
* gst/filter/Makefile.am:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_setup):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_setup):
* gst/filter/gstlpwsinc.h:
Use GstAudioFilter as base class and don't leak the memory
of the filter kernel and residue.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_activate_push), (open_library),
(gst_real_video_dec_init), (gst_real_video_dec_finalize):
* gst/real/gstrealvideodec.h:
Remove some old unused vars.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
Small cleanups.
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(open_library):
Remove fragment and timestamp correction code from the decoder to make
the caps and buffer contents compatible with matroska/ffdec_rvx0/...
Original commit message from CVS:
Patch by: Ian Munro <imunro at netspace net au>
* gst/bayer/gstbayer2rgb.c:
Include our own "_stdint.h" instead of <stdint.h> (which may not
be available).
* gst/speed/gstspeed.h:
Native HP-UX compiler dosn't seem to like enum typedefs before the
actual enum was defined.
* gst/vmnc/vmncdec.c:
Fix wrong usage of GST_ELEMENT_ERROR macro (#461373).
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
Use the proper context variable when setting the password !
LOG => WARNING for errors.
Give proper path when opening the codec (needs a '/' at the end).
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.
Original commit message from CVS:
Patch by: Jonathan Matthew <jonathan at kaolin wh9 net>).
* gst/modplug/gstmodplug.cc:
Fix modplug duration query. Fixes#384294.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Fix caps for 24 bit raw PCM audio (2).
Fixes#383471.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_video_caps):
Handle more H263 variants.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_event):
Call the base class handler. Fixes#380610.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Remove some asserts and replace them with a proper error
message. Fixes#379261.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Don't parse extra sample params for raw pcm. Fixes#374914.
Original commit message from CVS:
* configure.ac:
* gst/videoparse/Makefile.am:
* gst/videoparse/gstvideoparse.c:
A little pluggy to make sense out of the random chunks we get
from multifilesrc.
Original commit message from CVS:
* gst/multifilesink/Makefile.am:
* gst/multifilesink/gstmultifilesink.c:
* gst/multifilesink/gstmultifilesink.h:
* gst/multifilesink/multifilesink.vcproj:
Remove the old one.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_tree),
(qtdemux_parse_trak):
Handle unbounded length streams a bit better. Fixes#367696.
Original commit message from CVS:
* configure.ac:
* gst/multifilesink/Makefile.am:
* gst/multifilesink/gstmultifilesink.c:
* gst/multifilesink/gstmultifilesink.h:
I copied over filesink a while ago and modified it to work
as multifilesink. Might as well check it in. This could
use some work before being declared useful.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_create_src_pad):
* gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_create_pads):
* tests/check/elements/wavpackparse.c: (wavpackparse_found_pad):
Activate pads before adding them to running element.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo com>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(next_entry_size), (qtdemux_inflate), (qtdemux_parse_moov),
(qtdemux_parse_tree), (qtdemux_parse_trak), (qtdemux_tag_add_str),
(qtdemux_tag_add_num), (qtdemux_tag_add_date),
(qtdemux_tag_add_gnre):
Make compile with Forte compiler, mostly don't do pointer arithmetic
with void pointers (#362626).
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo com>
* gst/nsf/fds_snd.c:
* gst/nsf/mmc5_snd.c:
* gst/nsf/nsf.c:
* gst/nsf/vrc7_snd.c:
* gst/nsf/vrcvisnd.c:
Fix some things the Forte compiler warns about (#362626).
Original commit message from CVS:
* configure.ac:
* gst/deinterlace/Makefile.am:
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_base_init),
(gst_deinterlace_class_init), (gst_deinterlace_init),
(gst_deinterlace_stop), (gst_deinterlace_transform_caps),
(gst_deinterlace_set_caps), (gst_deinterlace_transform_ip),
(gst_deinterlace_set_property), (gst_deinterlace_get_property):
* gst/deinterlace/gstdeinterlace.h:
Port simple deinterlacer from 0.8. Use at your own risk, don't blame
me for anything it does or does not do to your precious pictures.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc),
(gst_qtdemux_loop_state_header):
Printf format fixes.
* sys/dvb/gstdvbsrc.c:
Use "_stdint.h".
Original commit message from CVS:
2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
Patch by: Josep Torre Valles <josep@fluendo.com>
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain),
(gst_faad_close_decoder):
Some cleanups.
Added some more debugging.
Don't ever ignore unlinked, we're not a demuxer.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
Activate pad before adding it to the element.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_transform_ip):
Removed cruft code that was just commented out. Removed some obsolete
debug logs statements.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_event):
Implements stop() to clear the adapter and event() to clear the
adapter on FLUSH_STOP and EOS.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_set_property):
* gst/spectrum/gstspectrum.h:
Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by
René Stadler
Original commit message from CVS:
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (main):
Use more defines
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_caps),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Apply some of the spectrum cleanup changes suggested in #348085.
Original commit message from CVS:
* configure.ac:
Bump requirements of -base (videocrop test case needs this).
* gst/videocrop/gstvideocrop.c:
Document sloppy handling of subsampled chroma planes if
left/top cropping is an odd number.
* tests/check/elements/videocrop.c: (handoff_cb),
(videocrop_test_cropping_init_context),
(videocrop_test_cropping_deinit_context),
(videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST),
(videocrop_suite), (main):
Add another unit test that crops the input to 1x1 (and checks
that that pixel has the expected values in a number of formats).
Original commit message from CVS:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init),
(gst_video_crop_transform_packed),
(gst_video_crop_transform_planar):
Some quick tests indicate that it doesn't make a great deal
of sense to use liboil here, at least not for the memcpy()s
we do, so remove liboil usage until there is clear evidence
it actually makes a positive difference somewhere.
Original commit message from CVS:
* configure.ac:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_base_init),
(gst_video_crop_class_init), (gst_video_crop_init),
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_get_unit_size), (gst_video_crop_transform_packed),
(gst_video_crop_transform_planar), (gst_video_crop_transform),
(gst_video_crop_transform_dimension),
(gst_video_crop_transform_dimension_value),
(gst_video_crop_transform_caps), (gst_video_crop_set_caps),
(gst_video_crop_set_property), (gst_video_crop_get_property),
(plugin_init):
Port/rewrite videocrop from scratch for GStreamer-0.10, and make
it support all formats videoscale supports (#345653).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
(gst_qtdemux_do_seek):
Reset each streams last_flow to GST_FLOW_OK.
(gst_qtdemux_activate_segment):
Removing mystic modifications for good.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(qtdemux_parse_tree):
put back 'segment start<=stop' change that was mystically reverted by
the last commit
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_add_stream), (qtdemux_parse_trak),
(qtdemux_video_caps):
Make sure segment start<=stop in weird quicktime files.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
(qtdemux_node_dump_foreach), (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Some more constification.
Fix some paletted data formats again.
Fix ulaw/alaw in qt.
Set correct caps for raw RGB.
Add support for yuv2, which is like Yuv2.
Add support for raw audio with the NONE fourcc, which is like raw.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_redirects_sort_func),
(qtdemux_process_redirects), (qtdemux_parse_tree):
Extract all references/redirections if there is more
than one and sort them; also extract minimum required
bitrate information if available. (#350399)
Original commit message from CVS:
2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Fix event parsing by gdpdepay. Fixes#349916.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
(gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
Consume all events except EOS because we generate events from
the gdp payload instead. Fixes#349204
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (audioresample_stop),
(audioresample_set_caps):
Don't leak references to the incoming caps. Clean them up when
stopping.
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_finalize):
Don't leak our temporary pixel buffer.
* tests/check/Makefile.am:
* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
(GST_START_TEST), (simple_launch_lines_suite):
Fix leaks and re-enable the test for valgrind checking.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
proxying get/set caps is the wrong thing to do, since we really
do change caps quite fundamentally
* tests/check/elements/gdpdepay.c:
* tests/check/elements/gdppay.c:
remove declaration of buffers, it's already done in gstcheck.h
Original commit message from CVS:
* gst/nsf/nsf.c: (nsf_load):
Really fix compilation. Apparently it's not enough to
just check the return value for errors, but we need to
check for short reads as well (now if only we handled
them too ...). Fixes#347935.
Original commit message from CVS:
2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
remove parent_class setting, BOILERPLATE does this
(gst_gdp_pay_reset_streamheader):
fix typo in comment
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie):
Store duration in uint64 too instead of clipping.
When we do a keyframe seek and the requested time is at the
keyframe, don't seek back to the beginning of the keyframe.
Fixes#347439.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
add more plugins and elements to docs
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
fix segfaults due to wrong g_free
add example
* gst/gdp/gstgdppay.c:
add example
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (on_window_destroy),
(draw_spectrum), (message_handler), (main):
* gst/spectrum/demo-osssrc.c: (on_window_destroy), (draw_spectrum),
(message_handler), (main):
port to use message to get results, cleanly exit when closing the window
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_dispose),
(gst_spectrum_set_property), (gst_spectrum_get_property),
(gst_spectrum_set_caps), (gst_spectrum_start),
(gst_spectrum_message_new), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
port to derive from basetransform and send results via messages
(like level element)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_parse_trak):
Combine return values from src pad pushes.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
(gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample),
(gst_qtdemux_add_stream):
Don't crash on files with 0 samples, EOS immediatly instead.
Fixes#344944.
Original commit message from CVS:
* configure.ac:
Check for X before using X_CFLAGS in the check for opengl (#343866).
* ext/musepack/Makefile.am:
* ext/wavpack/Makefile.am:
* gst/speed/Makefile.am:
Add missing GST_LIBS, fixes build on cygwin (#343866).
Original commit message from CVS:
* configure.ac:
enable building of GDP elements
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
(gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
(gst_gdp_pay_change_state):
* gst/gdp/gstgdppay.h:
add version 1.0
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
(gst_gdp_pay_get_property):
add crc-header and crc-payload properties
don't error out on some things that are recoverable
* tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
add test for crc
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment):
Clip the outputed NEWSEGMENT stop time to the configured segment stop
time.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (plugin_init):
po/POTFILES.in:
Throw an error when the file is encrypted. Move plugin_init stuff
to the end of the file, add stuff for i18n, make debug category
static.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_sink_caps),
(gst_spectrum_get_sink_caps), (gst_spectrum_chain):
Use boilerplate macro, fix strings to match plugin-moval-requirements
Original commit message from CVS:
* gst/spectrum/Makefile.am:
Link to base libraries
* gst/spectrum/demo-osssrc.c: (main):
use new threshhold property
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_dispose),
(gst_spectrum_set_property), (gst_spectrum_set_sink_caps),
(gst_spectrum_get_sink_caps), (gst_spectrum_chain),
(gst_spectrum_change_state):
* gst/spectrum/gstspectrum.h:
Use gst_adapter, support multiple-channels, add threshold property for
result, add docs, fix resulting spectrum range (was including mirrored
results)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds):
Figure out the real audio type in mp4a boxes by parsing the
optional descriptors in the optional esds box. Promote the
default AAC to mp3 when indicated. Fixes#330632.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_dump_unknown),
(qtdemux_parse_trak), (gst_qtdemux_handle_esds):
Parse version 2 sample descriptions.
Don't #define gst_util_dump_mem(), use something more
specific instead to avoid confusion.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
(qtdemux_dump_mvhd):
Don't cause side effects in a debugging function.
Also report duration in push mode since we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(resample_set_state_from_caps):
Add support for other formats audioresample can handle such as
32 bits in and float and 64 bits float. Fixes#301759
Original commit message from CVS:
Patch by: j^ <j at bootlab dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps):
Never treat video streams as an audio stream.
Add qtdrw mime type.
Fixes#339041
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
For VBR audio, don't try to calculate the samples_per_frame.
Fixes#338935.
Original commit message from CVS:
* gst/audioresample/debug.h:
replace debug macros with variable number of parameters
by a simple alias to gstreamer standard debug macros
(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
supported by MSVC 6.0 and 7.1)
* gst/audioresample/resample.h:
define M_PI and rint for WIN32
* win32/common/libgstaudio.def:
* win32/common/libgstriff.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
add new exported functions
* win32/vs6:
update project files
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (gst_qtdemux_add_stream), (qtdemux_dump_stsz),
(qtdemux_dump_stco), (qtdemux_parse_trak):
Don't make rounding errors in timestamp/duration calculations.
Fix timestamps for AMR and IMA4. Fixes (#337436).
Create a dummy segment even when there is no edit list.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_do_seek), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop):
Use duration as segment stop position if none is
explicitly configured.
Also perform EOS when we run past the segment stop.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_go_back),
(gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
(gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
(gst_qtdemux_chain), (qtdemux_parse_tree), (qtdemux_parse_trak):
More cleanups, added comments.
Mark discontinuities on outgoing buffers.
Post better errors when something goes wrong.
Handle EOS and segment end properly.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_push_event), (gst_qtdemux_go_back),
(gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
(gst_qtdemux_handle_src_event), (plugin_init),
(gst_qtdemux_change_state), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop), (gst_qtdemux_chain),
(qtdemux_sink_activate_pull), (gst_qtdemux_add_stream),
(qtdemux_parse), (qtdemux_parse_tree), (qtdemux_parse_trak),
(qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds):
* gst/qtdemux/qtdemux.h:
Handle stss boxes so we can mark and find keyframes.
Implement correct accurate and keyframe seeking.
Use _DEBUG_OBJECT when possible.
Original commit message from CVS:
* gst/modplug/libmodplug/Makefile.am:
* gst/modplug/libmodplug/load_it.cpp:
Try that again (not only should it be MODPLUG_ instead
of MODFILE, also that define is already set in stdafx.h;
what we really need is some more #ifndefs).
Original commit message from CVS:
* gst/modplug/libmodplug/Makefile.am:
More gcc-4.1 fixes (we don't need file saving, so just
define MODPLUG_NO_FILESAVE. That way, the compiler won't
complain about modplug ignoring the return value of fwrite
any longer and we might even save a few bytes as well).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(gst_qtdemux_init), (gst_qtdemux_dispose),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
Series of memleak fixes:
- Unref the GstAdapter in finalize.
- Use gst_pad_new_from_static_template(), shorter and safer.
- Free unused QtDemuxStream when not used.
Original commit message from CVS:
2006-03-11 Christophe Fergeau <teuf@gnome.org>
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* configure.ac:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h: added new element to add Xing headers
to MP3 files (this allows decoder to figure out the length of VBR
files)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Extract disc number and count from files that use
'disk' instead of 'disc' as node identifier for that
(fixes#332066).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak):
Use GST_WARNING instead of GST_ERROR for all the too short/long atoms
when parsing.
Also let's be a bit less vulgar in our warning messages :)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Can't divide through zero (suppress warning in case of
stream with one single still picture) (see #327083)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak), (qtdemux_video_caps):
Add support for palettised Apple SMC videos (#327075, based on
patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>).
Original commit message from CVS:
Reviewed by : Edward Hervey <edward@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video.
Add image/png for fourcc 'png '
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
Don't GST_LOG timestamps from nonexistent index
entries (#331582).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header):
Check that the size of the returned buffer is of the correct size
because the parser assumes that.
Fixes#331543.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event),
(gst_qtdemux_loop), (qtdemux_sink_activate_pull):
Don't stop the task if the pad isn't linked.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_post_buffering),
(gst_qtdemux_chain):
When buffering MDAT data, show the user something is
happening by posting 'buffering' messages on the bus.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_src_query), (gst_qtdemux_change_state),
(next_entry_size), (gst_qtdemux_chain):
* gst/qtdemux/qtdemux.h:
Make push-based work if mdat atom is before moov atom.
Don't answer duration query. This should be transformed into replying
FALSE to seek events.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (next_entry_size), (gst_qtdemux_chain):
Handle the case where data atoms are before moov atoms in push-based mode.
Errors out gracefully.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
(extract_initial_length_and_fourcc),
(gst_qtdemux_loop_state_header), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop_header), (next_entry_size), (gst_qtdemux_chain),
(qtdemux_sink_activate), (qtdemux_sink_activate_pull),
(qtdemux_sink_activate_push), (qtdemux_parse_trak):
* gst/qtdemux/qtdemux.h:
QtDemux can now work push-based.
It still needs some love for seeking.
Original commit message from CVS:
* configure.ac:
* gst/cdxaparse/Makefile.am:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxaparse.h:
Port cdxaparse, makes VCD playback work.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
use the correct variable to check if we can calculate
the last chunk. Looks like an obvious bug, and makes
the dump of offsets comparable to other tools
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
clean up some debugging, using _OBJECT, moving recurring
messages to LOG level
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
(gst_qtdemux_handle_src_event), (gst_qtdemux_loop_header),
(qtdemux_inflate), (qtdemux_parse), (qtdemux_parse_trak),
(qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds),
(qtdemux_video_caps), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
Some QT demux loving.
Handle seeking in a less broken way.
Fix AMR caps to match the AMR decoder.
Set first timestamp on AMR samples to 0 for now.
Remove some \n in DEBUG strings.
Use _scale_int for maximum precision.
Original commit message from CVS:
* gst/apetag/Makefile.am:
* gst/apetag/apedemux.c:
* gst/apetag/apedemux.h:
* gst/apetag/apetag.c:
Remove old files, apetag is in gst-plugins-good now.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
More coherent framerate setting on caps.
If sample_size is available, use that for the samples' duration in
the index. This enables single frame streams to work (and I imagine
fixes some other cases).
Tested on testsuite, no regression.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_audio_caps):
'twos' and 'sowt' fourcc can be 16bit or 8bit audio.
Fix 8bit case (#327133, based on patch by: Fabrizio
Gennari <fabrizio dot ge at tiscali dot it>).
Also, "G_LITTLE_ENDIAN" and "G_BIG_ENDIAN" are not
valid literals for endianness in caps strings,
only "LITTLE_ENDIAN" and "BIG_ENDIAN" are valid.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_send_event), (gst_qtdemux_handle_src_event),
(gst_qtdemux_change_state), (gst_qtdemux_loop_header):
* gst/qtdemux/qtdemux.h:
Fix seeking for quicktime files. Could still use some more
love and sophistication.
Original commit message from CVS:
reviewed by: Edward Hervey <edward@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add support for Indeo3 video in Quicktime files.
Closes#326524
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
Original commit message from CVS:
* ext/swfdec/gstswfdec.c: (gst_swfdec_class_init),
(gst_swfdec_chain), (gst_swfdec_render):
Add debugging category and return GstFlowReturn in the right places
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link):
Get something from the peer pad once we've checked if there is a peer pad.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(qtdemux_tree_get_child_by_type), (qtdemux_parse_trak),
(qtdemux_video_caps):
Couple of fixes
Original commit message from CVS:
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio):
* gst/festival/gstfestival.c: (socket_receive_file_to_buff):
* gst/vbidec/vbidata.c:
* gst/vbidec/vbidata.h:
* gst/vbidec/vbiscreen.c:
* sys/dxr3/ac3_padder.c:
don't use doc comments for non-docs
change some char* into char[]
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst),
(gst_faad_update_caps):
Assume that an unknown channel mapping with 2 channels
is stereo and play it that way instead of erroring.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
Handle e.g. jpeg streams with 0 duration frames as having 0 framerate.
Debug fixes. Some 64 bit variable fixes
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream):
Memleak fixes.
Send out EOS for valid reasons (couldn't pull_range() from upstream
for example).
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
Handle gracefully the consequence of "Maximum number of scalefactor
bands exceeded", which results in 0 channels with samplerates of 0.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state):
Do upward transitions, then call parent state_change, then do
downward transitions.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
Convert to fractional framerates
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Add support for custom genre tags.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(qtdemux_parse_tree):
Remove 'got-redirect' signal and post element message
on the bus instead.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_parse_header):
newsegment API update.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use
the second arg for the class, because G_OBJECT_GET_CLASS (self)
returns the wrong thing.
(gst_signal_processor_add_pad_from_template): Make pads of the
right type.
* ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make
writable param specs G_PARAM_CONSTRUCT so default values work.
(gst_ladspa_init): Use the second arg for the class.
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):
Add debug category, remove Close() call that made it crash
whenever reusing, renegotiating or anything; Close() actually
free()s the handle and should only be called on READY->NULL.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Actually set caps on buffer (in addition to pad), also.
Original commit message from CVS:
Add query function to GstSpeed, so that the stream length and current position get adjusted when queried (note that current position queries may still be wrong if the audio sink returns values based on buffer timestamps instead of passing on the query
Original commit message from CVS:
* gst/librfb/Makefile.am: Testing stuff before committing is
for wimps... and people with fast machines. Fix stupid
mistake.
Original commit message from CVS:
* configure.ac: Pull in librfb from my CVS tree, because it is
too small and annoying to be separate. Move rfbsrc plugin
to gst/.
* ext/Makefile.am:
* ext/librfb/Makefile.am:
* ext/librfb/gstrfbsrc.c:
* gst/librfb/Makefile.am:
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfb.c:
* gst/librfb/rfb.h:
* gst/librfb/rfbbuffer.c:
* gst/librfb/rfbbuffer.h:
* gst/librfb/rfbbytestream.c:
* gst/librfb/rfbbytestream.h:
* gst/librfb/rfbcontext.h:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
* gst/librfb/rfbutil.h:
Original commit message from CVS:
* ext/mpeg2dec/gstmpeg2dec.c:
Don't send things to NULL PAD_PEERs
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_chain):
Copy-on-write the incoming buffer.
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegclock.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
(normal_seek), (gst_mpeg_demux_handle_src_event):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegpacketize.h:
* gst/mpegstream/gstmpegparse.c:
(gst_mpeg_parse_update_streaminfo), (gst_mpeg_parse_reset),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead),
(gst_mpeg_parse_loop), (gst_mpeg_parse_get_rate),
(gst_mpeg_parse_convert_src), (gst_mpeg_parse_handle_src_query),
(gst_mpeg_parse_handle_src_event), (gst_mpeg_parse_change_state):
* gst/mpegstream/gstmpegparse.h:
* gst/mpegstream/gstrfc2250enc.h:
Various changes to the way time is computed that make seeking and
total time estimation much better here.
Use G_BEGIN/END_DECLS instead of __cplusplus
* gst/videocrop/gstvideocrop.c: (gst_video_crop_chain):
Use gst_buffer_stamp instead of only copying the TIMESTAMP
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header):
Re-apply patch from #142272 that allows non-seekable sources,
re-proposed by Daniel Drake <dsd@gentoo.org>.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_event), (gst_a52dec_chain):
Add some debug output. Check that a discont has a valid
time associated.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop):
Ignore TAG events. A little extra debug for broken timestamps.
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
(dvdnavsrc_change_state):
Ensure we send a discont to engage the link before we send any
other events.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
(dvdreadsrc_finalize), (_close), (_open), (_seek_title),
(_seek_chapter), (seek_sector), (dvdreadsrc_get),
(dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
dvd://title works in totem because typefinding sends a seek that ends
up going back to chapter 1 regardless.
* ext/mpeg2dec/gstmpeg2dec.c:
* ext/mpeg2dec/gstmpeg2dec.h:
Output correct timestamps and handle disconts.
* ext/ogg/gstoggdemux.c: (get_relative):
Small guard against a null dereference.
* ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
(gst_textoverlay_set_property):
Free memory when done. Don't call gst_event_filler_get_duration on
EOS events. Use GST_LOG and GST_WARNING instead of g_message and
g_warning.
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
(draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain):
Draw solid lines, prettier colours.
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
Add a default palette that'll work for some movies.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
(gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
* gst/mpegstream/gstmpegparse.h:
Use PTM/NAV events when for timestamp adjustment when connected to
dvdnavsrc. Don't use many discont events where one suffices.
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (gst_play_base_bin_add_element):
* gst/playback/gstplaybasebin.h:
Make sure we remove subtitles from the same bin we put them in.
* gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
(gst_subparse_buffer_format_autodetect),
(gst_subparse_change_state):
Fix some memleaks and invalid accesses.
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
(oggskel_type_find), (cmml_type_find), (plugin_init):
Some typefind functions for Annodex v3.0 files
* gst/wavparse/gstwavparse.h:
GstRiffReadClass is the correct parent class.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (gst_qtdemux_handle_esds):
More memory leak fixes (#149162).
Original commit message from CVS:
* ext/dv/gstdvdec.c:
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
I'm a bad boy. using /1001. to force C to do float division
and not integer division (as it did in my last commit)
Thanks to David I. Lehn for pointing this mistake.
Original commit message from CVS:
* ext/dv/gstdvdec.c:
* ext/libfame/gstlibfame.c:
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
replace framerate aproximations by their real value
(24000/1001, 30000/1001, 60000/1001)
Finish fixing bug #164049
Original commit message from CVS:
* ext/musepack/gstmusepackreader.cpp:
* gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
Some work on tags - still doesn't work in playbin...
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Handle events...
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/tta/gstttaparse.c: (gst_tta_src_event):
Fix gcc-2.95 compile (#163485).
Original commit message from CVS:
* gst/games/gstpuzzle.c: (nav_event_handler):
- handle nav events differently: forward every event no matter if it
was handled or not.
- translate events
You can now cheat by using navigationtest ! puzzle and moving the
mouse close to the edge of a tile. ;)
Original commit message from CVS:
* gst/games/gstpuzzle.c: (gst_puzzle_get_type),
(gst_puzzle_class_init), (gst_puzzle_finalize):
no memleaks, please
(gst_puzzle_create), (gst_puzzle_init),
(gst_puzzle_set_property), (gst_puzzle_setup):
change initialization code around so we don't reshuffle on resize
(draw_puzzle):
fix another stupid typo
Original commit message from CVS:
* gst/games/gstpuzzle.c: (gst_puzzle_base_init):
make RGB endianness work correctly
(gst_puzzle_show), (gst_puzzle_swap), (gst_puzzle_move):
refactor and fix race with initial shuffling
(nav_event_handler):
allow using the mouse to puzzle
(draw_puzzle):
insist on tiles having width and height as multiples of 4 to get
clean YUV image handling
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_handle_xevents), (gst_xvimagesink_buffer_alloc):
s/DEBUG/LOG/ for common messages
(gst_xvimagesink_navigation_send_event):
fix mouse event translation to not include screen PAR
* sys/ximage/ximagesink.c: (gst_ximagesink_navigation_send_event):
fix mouse event translation to actually work
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
Extract TrackNumber metadata + clean up code
* gst/games/gstvideoimage.c: (gst_video_image_draw_rectangle):
Hope this is the good fix (var used unitialised)
Original commit message from CVS:
* configure.ac:
* gst/games/Makefile.am:
* gst/games/gstpuzzle.c:
add a puzzle game with...
* gst/games/gstvideoimage.c:
* gst/games/gstvideoimage.h:
... full colorspace support (that includes YUV9 and RGB16)) stolen
from videotestsrc and made into something that would be a nice
library for a lot of other plugins.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_type_get), (qtdemux_audio_caps):
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(plugin_init):
Add 3GP (variables name Q3GP because they can't start with a
number). Add samr audio fourcc (used in .3gp files), decoder
is work in progress. Also do a GST_WARNING instead of ERROR
in case of unknown nodes, to decrease output.
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
(qtdemux_parse_udta), (qtdemux_tag_add), (gst_qtdemux_handle_esds):
Change all g_print()s to debugging. Add a bunch of consistency
checks.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
Original commit message from CVS:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-osssrc.c: (spectrum_chain), (main),
(idle_func):
Fix demo and reenable it. Yes, I'm currently playing with audio
analysis tools
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse), (gst_qtdemux_handle_esds):
An esds box is not a container.
Fix parsing of mp4v boxes.
Do not try to renegotiate fps for each frame. Need to
find a better method. This should fix mp4 playback.
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_loop):
Actually _do_ negotiation. Pass gdouble as arg instead
of guint64 for the framerate.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Don't crash by dividing by zero (see sample movie in #126922).
Original commit message from CVS:
* ext/dvdnav/README:
Update the README to use dvddemux
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
Ensure getcaps returns a subset of the template caps
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_base_init),
(gst_mpeg2subt_init):
Ensure getcaps returns a subset of the template caps
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_video_stream),
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_set_cur_subpicture):
* gst/mpegstream/gstdvddemux.h:
Set the explicit caps on the current_video pad before pushing
anything
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream):
Free caps used to gst_pad_set_explicit_caps, which takes a const
GstCaps *
Original commit message from CVS:
reviewed by Benjamin Otte <otte@gnome.org>
* gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init):
create a NULL-initialized array of pads, so we don't think they
exist already. (fixes#143130)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
(gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
* gst/qtdemux/qtdemux.h:
Bitch. Also known as seeking, querying & co.
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
(gst_osssink_change_state):
* sys/oss/gstosssink.h:
Resyncing is for weenies, this hack is no longer needed and was
broken anyway (since it - unintendedly - always leaves resync to
TRUE).
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxaparse.h:
some renaming
add some checks/sanity
prepare for seek addition
* sys/sunaudio/gstsunaudio.c:
remove exported dupe init function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header): Patch from dcm@acm.org (David Moore)
to allow qtdemux to use non-seekable streams. (bug #142272)
Original commit message from CVS:
* configure.ac: Add sunaudio
* examples/Makefile.am: make gstplay depend on gconf
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette),
(convert_table_lookup), (img_convert): remove c99-isms
* gst/ffmpegcolorspace/imgconvert_template.h: make a constant
unsigned, to fix a warning on Solaris
* gst/mpeg1sys/systems.c: bcopy->memcpy
* gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy
* sys/Makefile.am: Add sunaudio
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain): Fix crash when ESD
is killed while we're playing.
* gst/qtdemux/qtdemux.c: (qtdemux_parse): call
gst_element_no_more_pads().
Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)
* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)
* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
Original commit message from CVS:
* ext/audiofile/gstafsrc.c: (gst_afsrc_get):
Remove old debug output
* ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
(gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
(gst_dvdec_set_property), (gst_dvdec_get_property):
Change the quality setting to an enum, so it works from gst-launch
Don't renegotiate a non-linked pad. Allows audio only decoding.
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
(gst_deinterlace_link), (gst_deinterlace_init):
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link):
Some caps negotiation fixes
Original commit message from CVS:
* gst/cdxaparse/gstcdxaparse.c :
Add mpegversion to CAPS to make it link
Rank is as GST_RANK_SECONDARY instead of NONE
