Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
Original commit message from CVS:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
(img_get_alpha_info):
Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
of the existing BGRA32 and RGBA32 formats with the alpha at the other
end of the word. Partially fixes#451908
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
(gst_adder_request_new_pad):
Make getcaps more robust by not using the proxycaps function. This makes
sure that we don't end up recursively calling getcaps upstream.
See #316248.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
format, as produced by some dc1394 cameras like the iSight.
See http://www.fourcc.org/yuv.php#IYU1
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes#360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
Use other metrics as well when estimating the buffer level.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
When creating the groups, allow for a 5 second, unlimited buffers
preroll phase after which we expose the group.
When the group is exposed, use a small number of buffers up to a 2
second limit. Also disconnect the overrun signal from multiqueue when we
exposed the group because it is not needed anymore.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Don't error out on 0-sized packets, just emit a warning because this is
not a fatal error. Fixes#451145.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
The chain should be freed if we error out here, else it will leak.
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
(cleanup_decodebin):
Don't forget to *properly* remove the signals, else it will leak.
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
(main):
Destroy and recreate parse-launch based pipeline after stop to be able
to play again. Reorder some code and add more comments.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
When handling a delayed-caps notification case, mark
the group as dynamic so that the nbdynamic count is
incremented and decremented correctly. Fixes: #449156
Patch by: Wim Taymans <wim@fluendo.com>
Original commit message from CVS:
2007-06-19 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
Update tmpbuf for all neccesary rows, not just one, as is required
when downscaling.
Fixes#402076.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
(eos_buffer_probe):
Add a test that ensures we set DELTA_UNIT on all non-header,
non-video buffers, if we have a video stream.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_process_best_pad):
Move setting delta_pad to earlier, where we inspect all pads, so
that leading audio pages don't get DELTA_UNIT unset if they come
before the first DELTA_UNIT from video pages. Fixes the newly-added
test. Fixes#385527.
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
fails on the p5-ppc64 build bot and the failure looks like it is due
to the same issue as #348114, ie. a compiler bug.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
Fix compilation on mingw. Fixes#446972.
Original commit message from CVS:
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_enqueue):
Fix a division by zero when the max percent is <= 0. Fixes#446572.
also update the buffering status when receiving events. Fixes#446551.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
(gst_queue_handle_src_query):
Wait for preroll before attempting to forward a duration query upstream.
Fixes#445505.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
The BOS page of the first Dirac video stream needs to come before
the BOS page of any Vorbis streams or other audio streams, just like
it is with Theora.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_init),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
(gst_queue_src_activate_pull):
Add pull based scheduling and fix some deadlocks. Fixes#444523.
Does not yet completely work because duration queries upstream won't
block yet.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes#394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
Include stdio to define fseeko.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
(gst_v4lsrc_query):
Make v4lsrc output segments in GST_FORMAT_TIME. Fixes#442553.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (no_more_pads_full),
(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
(gst_uri_decode_bin_change_state):
Make sure we name srcpads uniquely even when using different internal
decodebins.
Signal no-more-pads when no more dynamic elements exist.
Remove pads on cleanup.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
(caps_notify_group_cb), (gst_decode_group_new),
(gst_decode_group_free):
Add support for delayed caps fixation when autoplugging.
Optimize cases where a multiqueue is not needed/wanted, like right after
anything that is not a demuxer.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
consideratly speedup ogg chain detection by not trying to find a base
timestamp for skeleton streams.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove_flush),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Add support for remuve_flush.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
* ext/theora/theoraenc.c: (theora_enc_sink_event),
(theora_enc_chain):
Add draft design for forcing keyframes in encoders and implement in
theoraenc.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun),
(no_more_pads_full):
Stop buffering when the group is commited because the queues filled up.
Fixes#442024.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes#417992.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Handle unknown or invalid pads without crashing, as might occur if
a media file like an mp3 is specified as a subtitle file.
Fixes: #410039
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
(setup_sinks):
Block the subtitle bin output queue before ghosting it and linking,
then unblock after. This avoids spurious not-linked errors caused
by the queue starting up (because it gets linked when it is ghosted).
Fixes: #350299
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
file. Avoids flukes where the input gets typefound to some valid but
useless type.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
Add unit test for gnomevfssink seeking and position reporting for
file:// URIs.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
* ext/gnomevfs/gstgnomevfssink.h:
Fix position reporting, especially after a seek (from upstream),
see #412648.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Fix a locking-order bug I introduced with my changes the other day.
Patch by Mike Smith.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes#415001.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_show_frame):
When we create our own window, indicate that we handle the
WM_DELETE client message from the window manager, so that it won't
kill our window (and our app) along with it. Handle ClientMessage,
post an error on the bus, and close the window. Further buffers
arriving will result in a FlowError because the window has been
destroyed.
Fixes: #393975
Clean up the X event handling loop and make them the same for
both xvimagesink and ximagesink while I'm at it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
Original commit message from CVS:
* configure.ac:
Bump libtheora requirement to 1.0alpha5 for the pixformat check
(also has a .pc file, so we don't need the fallback check any
longer). Fixes#438840.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
Original commit message from CVS:
* docs/Makefile.am: Install docs even when --disable-gtk-doc
is disabled. This matches the behavior of gtk+. Fixes#349099.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Some more chained streaming ogg timestamp fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page):
Add some FIXMEs.
Fix chain start/stop segment handling based on patch by
<ahalda at cs dot mcgill dot ca> see #320984.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_collect_chain_info):
If there is a stream in a chain without any data packets, ignore the
stream in the total length calculations. Might be related to #436820.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Consolidate and re-work our mpeg system stream detection to probe
more packets and produce a higher confidence result. Fixes a
regression caused by lowering the typefind probability last year
- related to bug #397810. Remove the redundant MPEG-1 specific
typefind function, as the new one detects both MPEG-1 & MPEG-2
happily.
Also cleanup the MPEG elementary and MPEG-TS detection functions a
little.
Tested against my media test directory, with some improvements and
no regressions.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (stop_cb), (main):
* tests/examples/seek/seek.c: (do_seek):
Some small cosmetic changes.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_type_packet), (theora_dec_change_state):
Don't push events (newsegment, tags) before initialising the
decoder.
This is neccesary for seeking to work correctly in gnonlin.