Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
Make the window for a race in typefind and shutting down smaller until
we figure out the right locking here. Avoids #485753 usually.
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
Remove unneeded lock causing a race in typefind and shutting down.
Fixes#485753.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Also remove sinks when going to NULL because we might not complete the
state change to PAUSED, causing the PAUSED->READY state change not to
happen.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (type_found),
(gst_decode_bin_change_state):
Don't disconnect the have_type signal because we never reconnect it
later on. Instead keep a variable to see if we already detected a type.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(type_found):
Unlink the signal handler when we found the type, we're not going to do
anything sensible with more type_found signals anyway.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(sdp_check_header), (sdp_type_find), (plugin_init):
Add typefind function for application/sdp.
Remove some old dirac typefind code that was ifdeffed out.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_flush), (gst_queue_locked_enqueue),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_push_one), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
Also fix#476514 for queue2.
Original commit message from CVS:
2007-09-14 Julien MOUTTE <julien@moutte.net>
* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
typefind for QCP files (RFC #3625)
Original commit message from CVS:
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c:
Increase upper limit for audio queue a bit; fixes preroll problem
with playbin and decodebin2 when playing a quicktime trailer with
multichannel audio via http (#464666).
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
Original commit message from CVS:
2007-09-03 Johan Dahlin <jdahlin@async.com.br>
* gst/typefind/gsttypefindfunctions.c (plugin_init):
Add an audio/x-nsf typefind function for the nsfdec element.
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
The tcp and subparse plugins are under gst, but not totaly free of
dependencies. Handle selection inconfigure.ac, so that they show up
on the final list of what is build and what is not. Maybe they should
better be moved to ext.
Original commit message from CVS:
Patch by: Daniel Díaz <yosoy@danieldiaz.org>
* configure.ac:
* gst/Makefile.am:
Check if libxml provides HTML parser which subparse needs.
Fixes#451970.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
We need to set up delayed-linking whenever the caps are non-fixed,
not just when there are multiple types - use gst_pad_is_fixed()
to test.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
Original commit message from CVS:
Based on a patch by: Davyd <davyd at madeley dot id dot au>
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_int32),
(volume_process_int32_clamp), (volume_process_int24),
(volume_process_int24_clamp), (volume_process_int16),
(volume_process_int16_clamp), (volume_process_int8),
(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
* gst/volume/gstvolume.h:
Add support for int32, int24 and int8 to the volume element.
Fixes#445529.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_remove_probe), (queue_threshold_reached):
Patch by: Josep Torra Valles <josep@fluendo.com>
Fixes: #465015
Make sure we remove the check_queues buffer probe from the
correct queue to avoid racily going back to "buffering 99%" when
buffering is actually complete.
Also, fix the spelling of Josep's surname in the ChangeLog.
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (queue_threshold_reached),
(gen_source_element), (setup_substreams),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(gst_play_bin_handle_redirect_message):
Move connection-speed property from playbin to playbasebin so that we
can also configure it in source elements that have the connection-speed
property. Fixes#464028.
Add some debug info here and there.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes#464079.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes#460422.
Also set the default volume to the default value specified in the
GParamSpec.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes#463215.
Original commit message from CVS:
Patch by: Dan Williams <dcbw at redhat dot com>
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_get_streaminfo_value_array):
Don't return NULL when querying the stream info value array but instead
return an empty array. Fixes#459204.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
(gst_video_rate_query):
Use boilerplate.
Add latency query, might not be perfect yet but already works a lot
better. Fixes#442557.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
Fix the r_mask test for RGBA32 on little-endian.
Fix a stupid typo that would have obviously broken
compilation on big-endian, if anyone was testing.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
(paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add alpha to the color struct.
Use a default alpha value of 255 instead of 128.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
(setup_source):
Clear the dynamic pads counter when starting a new uri. This makes
reusing playbin work again.
Fixes#454264.
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:
* 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.
