Original commit message from CVS:
* tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
(msg_buffering), (connect_bus_signals), (main):
Add statusbar.
Add buffering support with feedback in the statusbar.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
(gst_decode_bin_set_property), (gst_decode_bin_get_property),
(analyze_new_pad), (connect_pad), (expose_pad),
(gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
(gst_decode_group_expose), (gst_decode_group_free),
(do_async_start), (do_async_done), (gst_decode_bin_change_state):
Remove fakesink hack, we can now implement this more elegantly.
Added property to bypass typefinding.
Removed underrun callback and demuxer pad probe, we now use the srcpad
probe to expose groups.
API::sink-caps property
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Guard against multiple emissions of the no_more_pads signal, which
happens when we are dealing with chained oggs.
* gst/playback/gsturidecodebin.c: (remove_decoders),
(make_decoder), (type_found), (setup_streaming), (source_new_pad),
(setup_source):
For streams, use our own typefind element and plug our queue after it.
We will need this to determine the type of buffering to use for the
queue soon.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet),
(theora_dec_chain):
Don't try to do anything fancy with the return code from pushing an
event, it does not have enough information to turn it into a
GST_FLOW_ERROR.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
(gst_ogg_demux_chain_elem_pad):
Add small debug line.
Pass return code from the internal decoder instead of the too generic
GST_FLOW_ERROR.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain):
Refix oggdemux, we only have a problem if we failed to find a chain and
we are not EOF.
Original commit message from CVS:
Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain):
When we fail to find a BOS page and we and up with no chain, error out
properly instead of segfaulting. Fixes#525665.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
(gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
The new-pad-group sequence is add-pads, no-more-pads, add-pads,
no-more-pads...
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_out_rates),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_set_property):
Update the estimated input data when we push out a buffer.
Add some debug info about the temp file.
Only forward src events when we are not using a temp file.
Don't block the duration query, we need to find something better.
Don't leak the temp filename.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
ms-gsm can have arbitrarty sample rates. See #481354.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_is_filled):
The queue is never filled when there are no buffers in the queue at all.
Fixes#523993.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (free_group), (gst_play_bin_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
(gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_encoding), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb), (perform_eos), (autoplug_select_cb),
(activate_group), (deactivate_group), (setup_next_source),
(save_current_group), (gst_play_bin_change_state):
Update some docs.
Add new locks and conds to protect pipeline creation and group
switching.
Implement the sub-uri property.
Keep track of pending uridecodebin creation and configure the output
pipeline after all streams are configured.
Propagate subtitle encoding to the uridecodebins.
Implement getting the video/audio/visualisation elements.
Use input-selector for stream switching.
If we are asked to do visualisation, prefer to autoplug raw sinks
instead of sinks that accept encoded data.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
(gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
(gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
(gst_play_sink_set_volume), (gst_play_sink_get_volume),
(gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
(gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add methods to get audio/video/vis elements.
Add methods to set the font description for the overlay.
Remove properties, we're using this element with its methods only.
Add support for subtitles.
Rearrange the locking a bit to not use the object lock for protecting
the pipeline construction.
Try to use the volume and mute property on the sink when its available.
Implement the mute option with volume when the sink does not have a mute
property.
Only add volume element when the sink has no volume property.
Only do visualisations with raw audio pads.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property), (gst_text_overlay_src_event),
(gst_text_overlay_text_event), (gst_text_overlay_video_event),
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
(gst_text_overlay_change_state):
* ext/pango/gsttextoverlay.h:
Add property to configure waiting for text on the textpad or not, with
the default behaviour being the old one (always wait for text before
rendering the video). This default behaviour is usually not the best one
because the text stream can very sparse and could require queueing a lot
of video.
Fix the flushing and EOS handing so that we don't mix up their meaning.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_factories),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
(gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (no_more_pads_full),
(new_decoded_pad_cb), (gen_source_element), (remove_decoders),
(proxy_autoplug_factories_signal), (make_decoder),
(source_new_pad), (setup_source):
Add a readonly source property and notify.
Add new lock for protecting the construction of the pipeline.
Keep track of the decodebins we plugged.
Correctly proxy the autoplug signal so that it actually continues.
Proxy subtitle-encoding to the decodebins.
Original commit message from CVS:
* tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (main):
Rearrange some buttons in playbin2 and make some other boxes insensitive
when needed.
Add language codes to subtitle selection boxes when we gind the right
tags for the streams.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding),
(gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
(deactivate_free_recursive):
Protect caps property with the object lock.
Protect encoding property with the object lock.
Keep list of elements we added that have the subtitle-encoding property.
Distribute the subtitle-encoding to all of the elements when it
changes.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Correctly set the supported URI schemes and don't leave
some schemes in the middle or at the start at NULL.
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Filter http and https protocols. GIO/GVfs handles them but it's
impossible to implement iradio/icecast with it. Better use
souphttpsrc or something else for this.
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
If getting the file informations by a query fails try it with the
seek-to-end trick too.
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_create_silence_buffer),
(gst_audio_convert_transform):
Make audioconvert GAP-aware by outputting silence buffers when the
input has the GAP flag set. This is up to 8x faster.
Based on a patch by Stefan Kost. Fixes bug #517813.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_double):
Use oil_scalarmultiply_f64_ns() for double processing when it's
available at compile time.
Original commit message from CVS:
* configure.ac:
Fix lrint/lrintf checks to actually work. These functions are
in libm on Linux at least so try to link to it.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
Use GST_STR_NULL when trying to print strings that could be NULL because
this might crash on some platforms. See #520808.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(read_line), (gst_rtsp_connection_read_internal):
Generic Windows fixes that makes libgstrtsp work on Windows when
coupled with the new GstPoll API. See #520808.
Original commit message from CVS:
Patch by: Milosz Derezynski <internalerror at gmail dot com>
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
If seeking to a new position succeeds don't simply return from
create() without creating a buffer. Do this only in the case
seeking to the new position fails. Fixes bug #523054.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
(gst_video_format_from_rgba32_masks):
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
(#522635).
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
Add unit test for the RGB caps parsing and creation, checking for
internal consistency of the new API and consistency of the API with
the old GST_VIDEO_CAPS_* defines.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Revert change that caused regression until a real fix is found.
Fixes#522203.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes#520300.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Add trivial function to compare GstNetAddress. See #520626.
API: GstNetBuffer::gst_netaddress_equal
