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
Original commit message from CVS:
* gst/Makefile.am:
GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
them twice
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Add new API to the defs
Original commit message from CVS:
Patch by: Mersad Jelacic <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
Original commit message from CVS:
* configure.ac:
Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
plug-ins are included/excluded. (#498222)
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for IMelody files, using audio/x-imelody.
See bug #519516.
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst/playback/gstplaybin2.c:
Make the function signature of the _get_*_tags() functions match
the signature of the vfuncs they implement, ie. return a
GstTagList rather than a GstStructure, which is more correct,
even if one is typedef'ed to the other (#518940).
Original commit message from CVS:
* tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
(fourcc_list_struct), (fourcc_list), (fourcc_get_size),
(paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
(paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
(paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
(paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
(gst_video_format_is_packed), (video_format_is_packed):
Add unit test that makes sure that the strides, offsets and
sizes returned for the various YUV formats by the new video API
match the old reference implementation in videotestsrc.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
YV12 is I420 with swapped components 1 and 2, so the offset of
component 1 for I420 should be the offset for component 2 for YV12
and vice versa.
Original commit message from CVS:
2008-02-29 Julien Moutte <julien@fluendo.com>
* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
(gst_mixer_message_parse_mute_toggled),
(gst_mixer_message_parse_record_toggled),
(gst_mixer_message_parse_volume_changed),
(gst_mixer_message_parse_option_changed):
De-cruft and fix message type assertions (NULL is not a really
valid mixer message type string).
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_vis_src_negotiate):
When negotiating, actually start from a format that we can support
instead of from the too generic template.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
Enable vis setting.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gen_vis_chain):
Implement vis switching while playing.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
(gst_rtsp_connection_flush):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Use GstPoll for the rtsp connection.
Original commit message from CVS:
* tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
(init_visualization_features), (vis_combo_cb), (shot_cb), (main):
Add combo box for visualisations, populate it with a factory list
of all visualisation plugins, configure vis plugin instance in
playbin2.
Original commit message from CVS:
* gst-libs/gst/cdda/sha1.c: (sha_transform):
Use memcpy() instead of upcasting a byte array to long *. This
fixes an unaligned memory access, resulting in SIGBUS on IA64.
This should be ported to GCheckSum once we can use GLib 2.16.
Partially fixes bug #500833.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
Push tag event after the newsegment event. Log the pointer of
the buffer we're actually going to push rather than the buffer
we're feeding to _make_metadata_writable().
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Comment smoke typefinder for now. The smokedec plugin needs one
frame per buffer but we have no parser yet, thus it simply crashes
in most situations.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for the smoke video codec. Copied from the jpeg plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mid_type_find),
(plugin_init):
Add midi typefinder, copied from the timidity plugin.
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
Original commit message from CVS:
* tests/check/gst-plugins-base.supp:
Add a suppression for a cached value in GIO that wasn't moved
while moving gio from -bad to -base.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* configure.ac:
Don't hardcode -Wall and -Werror for configure checks, this fails
with non-GCC compilers. Fixes bug #517991.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnome_vfs_sink_handle_event):
Return FALSE when seeking for a new segment fails instead
of silently ignoring the failure and appending every buffer
that comes for the new segment.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (find_property),
(gst_play_sink_find_property), (gen_video_chain),
(gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
Recursively search the sink element for a last-frame property so that we
can also find the property in autovideosink and friends that don't
always proxy the internal sink properties.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
Original commit message from CVS:
2008-02-19 Julien Moutte <julien@fluendo.com>
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
typefind lookup to fix typefinding on HD clips.
Original commit message from CVS:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
Fix up copyright (I rewrote the GStreamer-0.10 code for
this from scratch back in the days).
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
(create_element), (gst_play_frame_conv_convert):
* gst/playback/gstscreenshot.h:
Add screenshot conversion code from totem.
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
(gst_play_bin_class_init), (gst_play_bin_convert_frame),
(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
Implement frame property to get a color-unconverted snapshot.
Implement convert-frame action signal to get a converted snapshot image.
Configure connection speed in uridecodebin.
Document some more properties.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_get_last_frame):
* gst/playback/gstplaysink.h:
Use last-buffer property of the video sink to get a video snapshot.
* tests/examples/seek/seek.c: (shot_cb), (main):
Add snapshot button for playbin2 and use the frame property to save the
frame as a png in the current directory.
Original commit message from CVS:
* configure.ac:
Require CVS of core for new API in collectpads.
* gst/adder/gstadder.c:
Use new API to make adder sparse stream aware.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb):
Get the object data correct so that we can remove our channels
correctly.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Add option to disable async behaviour in the sinks when possible. This
makes it possible to avoid an audio queue when dealing with
visualisations.
Add option to add a queue for the audio path.
* tests/examples/seek/seek.c: (clear_streams), (update_streams),
(main):
Disable the vis checkbox to match the defaults of playbin2.
Only get the stream info when we need to.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
(gst_gio_base_src_set_stream):
* ext/gio/gstgiosink.c: (gst_gio_sink_start):
* ext/gio/gstgiosrc.c: (gst_gio_src_start):
Don't use async operations as they require a running main loop.
This makes us block again when closing streams and unable
to mount the enclosing volume of an URI if it isn't yet.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Move tee in front of the audio and vis pipelines.
Add queue for audio for now.
Add visualisation support.
* tests/examples/seek/seek.c: (main):
Visualisation is by default disabled.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (close_stream_cb):
* ext/gio/gstgiobasesrc.c: (close_stream_cb):
Improve debugging a bit.
* ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Try to mount the enclosing volume of a GFile if it isn't mounted
yet. This requires us to wait for an async operation to finish, done
with an nested GMainLoop. Authentication is not supported yet, will
come later.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
Add some more debug info.
Make sure we never return a negative delay. Fixes#516246.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
Revert patch that makes the sink hold the object lock when
calling snd_pcm_delay(), since it breaks playback for me.
Original commit message from CVS:
2008-02-12 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
some seek flags when changing rate.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
Original commit message from CVS:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
(gst_v4lsrc_buffer_finalize):
Correctly chain up the finalize method.
Original commit message from CVS:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
Add documentation and example code for giostreamsink/giostreamsrc.
* tests/check/pipelines/gio.c: (GST_START_TEST):
Ask the GMemoryOutputStream for the data instead of assuming that
the pointer to the data stayed the same. It could've been realloc'ed.
Original commit message from CVS:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
Make the documentation of giosink/giosrc complete, large parts
are based on the gnomevfssink/gnomevfssrc docs.
Original commit message from CVS:
* ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
* ext/alsa/gstalsasink.c: (set_swparams):
* ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
against libasound >= 1.0.16, since it's been deprecated in
0.10.16, and alignment is always 1 then, apparently. (#512899)
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
* gst/playback/gstplaysink.c: (gen_audio_chain):
Handle case where we can't create the volume element a bit
better (#514307).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
Add support for https protocol. Fixes#510229.
Original commit message from CVS:
2008-02-11 Julien Moutte <julien@fluendo.com>
Patch by: Alan Peevers <peeves@pacbell.net>
* ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
lock when calling alsa methods.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Bump rank of jpeg and png typefinders, which will return maximum
probability in the most common cases (thus short-circuiting more
expensive typefinders like the mp3 one for these two quite common
image types).
Original commit message from CVS:
Patch by: Branko Čibej <brane at xbc dot nu>
* sys/xvimage/xvimagesink.c:
Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
Fixes bug #515654.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Set is_dynamic as True if there are elements with both request
and sometimes src pad templates instead of breaking out when it
finds the first pad template that is a src.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb), (clear_streams),
(update_streams), (video_combo_cb), (audio_combo_cb),
(text_combo_cb), (volume_spinbutton_changed_cb), (main):
Add some stream switching and volume gui for playbin2.
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
Added marshal for streamselector Tags.
* gst/playback/gstplaybasebin.c: (set_active_source):
Streamselector now selects pads based on the pad object instead of its
name.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (get_group), (get_tags),
(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_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
Remove option to mute streams with the current-a/v/t property, we have
this functionality in the flags.
Add signals to notify when the number of A/V/T channels changed.
Add action signals to get tags for the A/V/T streams.
Implement setting the current A/V/T stream.
Rearrange some things to simplify stream selection.
Implement volume.
* gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
(gst_play_sink_get_volume), (gst_play_sink_set_property),
(gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
(activate_vis), (gst_play_sink_reconfigure):
* gst/playback/gstplaysink.h:
Add and implement volume setting methods.
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_finalize),
(gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad):
* gst/playback/gststreamselector.h:
Add pad properties for tags and status of pads.
Keep tags on pads.
Make active pad selection based on pad object instead of name.
Original commit message from CVS:
2008-02-08 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
(main): Make sure bus signals are reconnected when pressing STOP
and then PLAY again for a parse launch pipeline. Fix a ref leak
on the bus.
* win32/common/config.h: Updated.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Add the headers which need scanning for the GIO plugin. The rest of
the docs still need migrating.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Comment out a couple of other things which break the build when
GST_DISABLE_DEPRECATED isn't on but -Werror is.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Clear the addrinfo struct using memset. Fixes#514937.
Original commit message from CVS:
* gst/tcp/gstfdset.h:
Remove unused field to same some memory.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Mark action signals as such.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek),
(rate_spinbutton_changed_cb), (update_streams), (main):
Remove obsolete stream_time reset after flushing seek, core does that
automatically now.
Improve accuracy of speed spinbutton.
Only do playbin2 stuff when we actually use it.
Original commit message from CVS:
* tests/check/Makefile.am:
Revert previous change of the test environment's GST_PLUGIN_PATH.
The problem is not with the plugins, but with element factories
and only occurs if elements are split out from existing plugins
or if plugins change name (see #512740).
Original commit message from CVS:
* tests/check/Makefile.am:
Fix the tests environment's GST_PLUGIN_PATH: we want the directory
with the core's plugins first and our local build directories last,
since we might be building against an installed core, and that
core's plugin directory may contain older or other versions of
our own -base plugins, but we really do want to test our local
ones (if there are multiple plugins or element factories with the
same name, those inspected last will trump those read in earlier).
Fixes#512740 for the most part.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Use gmtime_r if available as gmtime is not MT-safe.
Fixes bug #511810.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(get_group), (get_n_pads), (gst_play_bin_get_property),
(pad_added_cb), (no_more_pads_cb), (perform_eos),
(autoplug_select_cb), (deactivate_group):
Remove stream-info, we going for something easier.
Refactor getting the current group.
Implement getting the number of audio/video/text streams.
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_get_property),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Add property for number of pads.
* tests/examples/seek/seek.c: (set_scale), (update_flag),
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (msg_async_done),
(msg_state_changed), (main):
Block slider callback when updating the slider position.
Add gui elements for controlling playbin2.
Add callback for async_done that updates position/duration.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes#511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Also remove the conditional registration of the signals
that disappeared with the ABI change in 0.10.14
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert patch to gstrtspconnection.c for brown paper bag
release of -base. Re-opens: #511825
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
Fixes: #513018
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraparse.h:
* ext/theora/theoradec.c:
* ext/theora/theoraparse.c:
Take a 2nd stab at handling libtheora granulepos changes in the decoder
and parser by inspecting the bitstream version of the incoming data.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Include Winsock2.h for VS6 and use a different way initialize
hints structure so it can build with VS6.
* win32/MANIFEST:
* win32/vs6/libgstsdp.dsp:
* win32/common/libgstsdp.def:
Add new files for libgstsdp.
* win32/vs6/grammar.dsp:
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstvolume.dsp:
Add new dependencies to the link list.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
Add test to see if hstrerror is available or if we need libresolv
(Solaris) for it, then use it in libgstrtsp.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_context_copy),
(gst_install_plugins_context_get_type):
* gst-libs/gst/pbutils/install-plugins.h:
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
for bindings.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_class_init),
(_theora_granule_frame), (_theora_granule_start_time),
(theora_dec_sink_convert), (theora_dec_decode_buffer):
Adapt for post-alpha meaning of granulepos, when we
have a newer version of libtheora.
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
(theora_enc_is_discontinuous), (theora_enc_chain):
Likewise.
* tests/check/Makefile.am:
Link libtheora into theoraenc test so we can check which version of
libtheora we're testing against.
* tests/check/pipelines/theoraenc.c: (check_libtheora),
(check_buffer_granulepos),
(check_buffer_granulepos_from_starttime), (GST_START_TEST),
(theoraenc_suite):
Adapt tests to check the values that are now defined for theora; make
the tests backwards-adapt the passed values if we're running against an
old libtheora.
Fixes#497964
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
Original commit message from CVS:
* autogen.sh:
Add -Wno-portability to the automake parameters to stop warnings
about GNU make extensions being used. We require GNU make in almost
every Makefile anyway.
* configure.ac:
Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
at the same time is required for per target flags.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
Post an error message if we can't pull as many bytes as we need
for the tag. This makes sure the user gets to see a proper error
message if a file with a partial ID3 tag is fed to decodebin, and
not a 'no ID3 tag demuxer' error, which would be confusing
(see #508138).
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Make sure we error out correctly if we can't activate one of
the elements we've added. Fixes#508138.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
(check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
Use snd_mixer_selem_set_{playback|capture}_volume_all() if
the volume is the same for all channels. This works around
some problem in alsa that leaves us with inconsistent state
for some reason (#486840).
Original commit message from CVS:
Patch by: Jerone Young <jerone at gmail com>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
If there's no mixer track by the name of 'Master' or 'Front',
check if there's one called 'PCM' before trying the generic
fallback logic (fixes#506928, where we pick 'Mic' as master
track for the AD1984 card in a Thinkpad T61/X61 laptop).
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type), (register_gst_play_flags),
(gst_play_flags_get_type):
* gst/playback/gstplay-enum.h:
Add enums for configuration flags.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (gst_play_bin_set_property),
(gst_play_bin_get_property), (no_more_pads_cb),
(autoplug_select_cb), (gst_play_bin_change_state):
Merge mode with flags.
Add more property getters/setters, defaults and docs.
Add properties to get number of audio/video/text streams.
Create sink object in _init so that we can always rely on it being
there.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gen_video_chain), (gen_audio_chain), (gen_vis_chain),
(activate_vis), (gst_play_sink_reconfigure),
(gst_play_sink_set_flags), (gst_play_sink_get_flags),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Use flags to configure the sink pipelines.
Add tee before audio pipeline so that we can use it for visualisations.
Start working on integrating visualisations.
Remove mode, we can do everything with the flags now.
Add method to configue the sink pipeline.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(check_buffer_timestamp), (check_buffer_duration):
Turn these functions into macros so we can see right away
where the failure occured.
Original commit message from CVS:
2008-01-05 Julien Moutte <julien@fluendo.com>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
debugging information to understand how X calculates the stride
for XvImages.
Original commit message from CVS:
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_choose_func),
(gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
(volume_setup):
* gst/volume/gstvolume.h:
Use GstAudioFilter as base class for the volume element instead of
plain GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/pbutils.h:
Use glib-enum generator to have a proper enum GType for
GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c:
Reenable theoraenc test, which fails on the buildbot but
not locally.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
Disable theoraenc test long enough to get the buildbot to
compile a recent -base.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Make sure we reset the slider value to 0.0 without racing against a
possible g_idle that sets it to something else.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add Location header so that we can start implementing redirects.
See #506025.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_chain):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x and don't
abort() in any case but properly report the error.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_primary_decoder):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
(cddabasesrc_suite):
Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
deprecated in the future (see #498924).
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-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
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:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add new GstVideFormat enum and write a bunch of helper functions
based around it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Yet another fix for broken software that produce files with an empty
blockalign field. Instead of completely failing, make a second attempt
at guessing the width/depth by looking at strf->size.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
for jpeg video streams.
Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
for the above modification.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
(gst_x_overlay_handle_events):
More guards (we don't want klass to end up being NULL).
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:
* tests/examples/seek/seek.c: (msg_segment_done), (main):
Don't go to READY on EOS as this avoids testing of seeking and
restarting after EOS, use the stop button when you want to READY.
Don't try to do a flushing seek in segment-done, it does not make
sense to use this for gapless playback and is not needed.
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:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_free):
Close control sockets. Fixes#503440.
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:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description for 'private' dts caps (who come up with that name?).
Original commit message from CVS:
* Makefile.am:
Add check-exports target and run it with 'make check'.
* configure.ac:
Be stricter about what we export in our libraries: change regexp so that
we only export _gst_foo(), but not __gst_foo().
* gst-libs/gst/cdda/base64.h: (rfc822_binary):
* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
Change internal functions to __gst_foo so they dont' get exported.
* win32/common/libgstaudio.def:
Add missing symbols.
Original commit message from CVS:
* ext/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
Use gst_tag_freeform_string_to_utf8() here, which also takes
into account any character sets specified by the user via
environment variables.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
No need for floating point operations here. avoids having to link
against the math library too.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats. Generate ADPCM description
dynamically depending on layout/format.
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-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain):
If we find a new serial number but it does not contain a BOS page, make
sure we initialize the chain to NULL because else we will try to scan it
and crash. Fixes#500763
Original commit message from CVS:
2007-11-24 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Increase the range of the
rate selector as I would like to test QOS behavior at higher
forward and reverse playback speed like say 64x.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes#498767.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
Original commit message from CVS:
Patch by: Joe Peterson <lavajoe at gentoo dot org>
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix compilation on FreeBSD (Gentoo). Fixes#498228.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): There's a nice macro to
check
GTK version, use it.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Try to support stable version
of GTK.
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:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix leaking headers. Fixes#496761.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
Don't leak the PAR on errors. Fixes#496731.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
(gst_tag_from_id3_user_tag):
Add mapping for audio cd discid tags, so we can extract
them from tags as well (see #347848). Also compare identifiers
in ID3v2 TXXX frames in a case-insensitive way to increase
compatibility when reading tags (discid vs. DiscID vs. DiscId).
Original commit message from CVS:
=== release 0.10.15 ===
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
releasing 0.10.15, "No need to argue"
Original commit message from CVS:
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstfft.dsp:
* win32/MANIFEST:
Add a project file for fft plugin and remove socket
based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Convert line endings back to DOS.
Fixes#496724
Original commit message from CVS:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.h:
Don't include malloc.h which doesn't exist on Mac OSX.
Instead, pull in glib.h and use g_malloc/g_free for
consistency. Fixes: #496548
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:
Patch by: Sebastien Moutte <sebastien moutte net>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
Fix some C99-isms and and a missing function that some versions of
MSVC don't like too much (#494346).
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Update vs6 projects files (#494346).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes#492098.
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:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
'Could not open resource for writing' is not an acceptable
error message when we can't open the audio device (see #492334),
even less so when we're trying to open it to record something.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* win32/common/libgstrtp.def:
Add some more missing symbols (#492813).
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
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-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
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:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
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-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes#489010.
Original commit message from CVS:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes#489010.
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
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:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
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-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
Make sure that before we clean up the X resources, we shutdown and join
the event thread.
Also make sure the event thread does not shut down immediatly after
startup because the running variable is not yet correctly set.
Fixes#378770.
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-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
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-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes#483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Replace g_return_if_val (as it could be disabled), with regular return
and warning.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
(gst_rtp_payload_info_for_name):
* gst-libs/gst/rtp/gstrtppayloads.h:
Added new file and header to deal with payload info.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Payload specific stuff is move to new headers.
Implement _default_clock rate using the new payload function.
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
(gst_sdp_parse_line):
* gst-libs/gst/sdp/gstsdpmessage.h:
Add some more comments.
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:
* win32/common/libgstaudio.def:
Add new exported functions.
* win32/vs6/grammar.dsp:
Add autogeneration and copy of some autegenerated files from win32/common
for rtsp library.
* win32/vs6/libgstaudioconvert.dsp:
Add gstaudioquantize.c to the build.
* win32/vs6/libgstinterfaces.dsp:
Add videoorientation.c to the build.
* win32/vs6/libgstriff.dsp:
Add libgsttag to the link libraries list.
* win32/vs6/libgstvolume.dsp:
Add liboil to the link.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstrtsp.dsp:
* win32/common/libgstrtsp.def:
Add files to build libgstrtsp library.
