Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* gst/playback/gstplaybasebin.c: (set_subtitles_visible),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(setup_sinks), (playbin_set_subtitles_visible):
Make switching off of subtitles work. To avoid all kind of
problems with unlinking of the subtitle input, we just keep
the subtitle inputs linked as they are and tell textoverlay
not to render them. Fixes#373011.
Other subtitle switching issues (esp. when there are both
external and in-stream subtitles) remain. They'll be solved
in playbin2.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
Add a "blink" pattern. Turn on the pain. Apologies. It's useful
for testing vertical refresh synchronization.
Original commit message from CVS:
* configure.ac:
* gst/volume/gstvolume.c: (gst_volume_init):
Use new gst_base_transform_set_gap_aware() function as volume
correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
for this.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
(reset_rate_timer), (update_in_rates), (update_out_rates),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_chain), (gst_queue_loop):
Use separate timers for input and output rates.
Pause measuring the output rate when we block for more data.
See #503262.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_chain):
Pause the timer to measure the input rate when we block because the
queue is filled. See #503262.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
Expose the right pad in the right place with the right element.
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes#502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
Original commit message from CVS:
* gst/playback/Makefile.am:
Group decodebin2 and uridecodebin into the same plugin so that they
can share the GEnumType.
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
Add signal to sort factories instead of the more awkward autoplug-select
signal.
Modify autoplug_select so that we can try, skip or expose the
autopluggin of an element on a pad.
* gst/playback/gstfactorylists.c: (compare_ranks),
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
(element_filter), (gst_factory_list_get_elements),
(gst_factory_list_debug), (gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Simplify the API, allow getting elements based on mask.
* gst/playback/gstplay-marshal.list:
Add some more marshallers.
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
(autoplug_select_cb), (activate_group):
Add support for managing non-raw sinks by providing a custom element and
sink list to decodebin2.
Try to plug non-raw sinks when decodebin2 using autoplug-select of
decodebin2.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
* gst/playback/gstplaysink.h:
Add support for raw and non-raw sinks.
Add support to force sinks selected by playbin2.
Don't plug raw converters for non-raw sinks.
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
(plugin_init):
Use right accumulators.
Proxy new signal.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
(gst_selector_pad_class_init), (gst_selector_pad_init),
(gst_selector_pad_finalize), (gst_selector_pad_reset),
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_stream_selector_get_type),
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_set_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_is_active_sinkpad),
(gst_stream_selector_activate_sinkpad),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Improve streamselector, make it select and unselect the current pad more
intelligently.
Subclass GstPad for the sinkpads of the selector.
Handle segments more correctly.
Fix caps negotiation.
Implement release_pad.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_group_check_if_drained), (source_pad_event_probe),
(remove_fakesink):
Add drained signal fired when decodebin finishes decoding the data.
Remove deprecated STATE_DIRTY message.
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
(analyse_source), (proxy_drained_signal), (make_decoder),
(source_new_pad), (value_list_append_structure_list),
(handle_redirect_message), (handle_message):
Proxy the new drained signal.
Handle pad removed from decodebin.
Handle redirect messages by sorting multiple redirections based on the
connection speed.
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes#491722).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value. Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes#430677.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Post nice/more useful error message if we don't have a decoder for
the primary type.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
Be a bit more useful, unblock the pads after we fired the no-more-pads
signal so that we can use the signal to inspect and connect all pads
without having to keep extra state outside of decodebin.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_continue),
(gst_uri_decode_bin_class_init), (no_more_pads_full):
Implement default signal handler so that we return TRUE when nothing is
connected.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (analyze_new_pad):
Move subtitle encoding property to decodebin2 so that it can set the
property value on all elements that it autoplugs and that require it.
Make caps refcounting more consistent in get/set.
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
(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), (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(make_decoder):
Proxy properties and relevant signals from the internal decodebin.
Make properties MT safe.
Original commit message from CVS:
Inspired by patch of: René Stadler <mail at renestadler dot de>
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
(find_compatibles):
* gst/playback/gstplay-marshal.list:
Remove the autoplug-sort signal and replace it with a binding friendly
autoplug-select signal.
Add an autoplug-factories signal that can be used to generate a list of
factories to try to autoplug.
Add the GstPad to the autoplugging signal args as it might be needed to
make a good factory selection.
Fix up the marshallers for this. Fixes#407282.
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.
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:
* 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:
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:
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.
