Original commit message from CVS:
* REQUIREMENTS: Change the format to key/value, add a bunch of
information, remove a bunch of requirements that are for
other GStreamer packages.
Original commit message from CVS:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END here as well.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
the image format a variable-length NUL-terminated string; in
versions before that the image format is a fixed-length string of
3 characters (see #348644 for a sample tag).
Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
Printf format fixes; also add some missing quotes in translated
strings. Fixes#416728 and #416727.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
Tim and I can't think of any reason the child audio sink needs to
be set back to NULL after successfully determining that it can
reach READY - it gets immediately set back to READY by the caller
anyway, causing an unnecessary close/open of any audio devices
involved.
Original commit message from CVS:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_new):
Actually translate sunaudio mixer track labels instead of just
marking the strings as translatable (#377306); clean up weird
label string mapping code that serves no apparent purpose. Also
set the 'untranslated-label' property when creating mixer tracks
if the GstMixerTrack base class supports this.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/sunaudio.c: (GST_START_TEST),
(sunaudio_suite):
Very minimalistic unit test for sunaudiomixer element (compiles, but not
actually tested on a system where sunaudiomixer is available).
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
(gst_dvdec_src_negotiate), (gst_dvdec_chain),
(gst_dvdec_change_state):
* ext/dv/gstdvdec.h:
Infer pixel-aspect-ratio from the video frame format if it isn't
provided by the container, as happens when playing DV from AVI
or Quicktime containers.
Patch by: Wim Taymans <wim@fluendo.com>
Fixes#380944
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
When activated, remove the udpsrc timeout, we have dataflow and timeouts
will later be handled by the jitterbuffer.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
Fix stream position reporting after a seek. Fixes#416445.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_chain):
Make avidemux accept optional header chunks in any order.
Fixes#415446.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
Free handles that we allocated when exiting via the error paths.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_sink_activate):
Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
caps passed to it (previouslly one code path assumes it takes ownership
while another one assumes it doesn't).
* configure.ac:
* tests/files/Makefile.am:
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
Add directory where data for unit tests can be stored.
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
(read_tags_from_file), (run_check_for_file),
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
Add unit test for id3demux, and in particular for bug #407349. Only
testing pull-mode for now; push mode doesn't work yet because the test
files are smaller than ID3_TYPE_FIND_MIN_SIZE.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_obsolete_tdat_frame):
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
the four-digit number will be interpreted as a year, whereas it is
month and day in DDMM format. Instead, parse TDAT frames and fix up
the date in the GST_TAG_DATE tag later if we also extracted a year.
Fixes#407349.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the dispose logic so it doesn't leak, and fix setting of
the child state so that we don't set a child to our current state
just as we are changing it to something else.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the reference counting of the child elements.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
Install fakesink in NULL by fixing some broken logic. This obviates
the need to manually set _IS_SINK.
Add some comments and remove a little cruft while I'm at it.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
Mark us as a sink when we have no fakesink in NULL. Fixes#414887.
Original commit message from CVS:
* tests/check/Makefile.am:
Gah! Also disable gconfvideosink from the tests, otherwise
it will instantiate autovideosink, and dfbvideosink and
leak on the buildbots.
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Make sure we always destroy our libcdio handle.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable autovideosink so the buildbots don't barf over memory
leaked in the directfb sink.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
(gst_multipart_find_pad_by_mime):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* ext/libcaca/Makefile.am:
* gst/debug/Makefile.am:
Don't mix tabs and spaces (#414168).
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Small code cleanups.
Don't use pad_alloc as the base class cannot deal with the error codes.
Original commit message from CVS:
Patch by: René Stadler <mail@renestadler.de>
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Handle rounding better to not drop last sample frame. Fixes#356692
Original commit message from CVS:
* tests/check/Makefile.am:
Disable cacasink from the states check too - it also calls exit(1)
on us when it can't find a terminal to talk to.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Having NULL as UDI previously selected the default sink/src. Change
this back but mention it in the debug output.
* ext/hal/hal.c: (gst_hal_get_alsa_element),
(gst_hal_get_oss_element), (gst_hal_get_string),
(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
(gst_hal_get_audio_src):
* ext/hal/hal.h:
Refactor a bit, check all error conditions, greatly improve debugging
and fix some possible memory leaks. Also implement OSS support
and allow specifying an UDI that points to a real device. For this the
child device which supports ALSA (preferred) or OSS is used.
As a side effect this makes it impossible now to get a alsasink in
halaudiosrc and a alsasrc in halaudiosink.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
Errors from the udp sources are not fatal unless all of them are in
error.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable aasink in the states test. I suspect this is the element that
is calling exit(1) when it can't proceed.
Original commit message from CVS:
* tests/check/Makefile.am:
Draw plugins in from the build tree sys/ dir, rather than picking
up the already installed versions.
Original commit message from CVS:
2007-03-01 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
Error out correctly when getting xcontext fails.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Check if the device UDI is set before trying to query HAL
about it and give a useful error message if it wasn't set.
* ext/hal/hal.c: (gst_hal_get_string):
Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
gives an assertion failure in D-Bus when running with
DBUS_FATAL_WARNINGS=1.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another interesting test url.
* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
Don't allow getting header fields from data packets.
Original commit message from CVS:
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_start),
(gst_shout2send_set_property), (gst_shout2send_get_property):
* ext/shout2/gstshout2.h:
Add a property for username.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (append_auth_header),
(rtsp_connection_send), (rtsp_connection_set_auth):
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
from icecast to replace it. Relicensed from GPL courtesy of Mike
Smith.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init):
Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
fixes#407369
Original commit message from CVS:
* gst/rtsp/URLS:
Add example H264 rtsp url.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
Don't convert values to lowercase or we might mess up base64 encoded
properties.
Original commit message from CVS:
* gst/rtp/README:
Fix case of string params.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Fix depayloader, support more packet types.
Add sync codes to make sure the packetizer can do its job.
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
Fix caps case again.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
As spotted by: Peter Kjellerstedt <pkj at axis com>:
Clear stack allocated SDPMedia struct before calling _init() on it.
Clarify this in the docs as well.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
(do_change_child):
Don't reset the profile when going switching states, as it makes
the element non-reusable.
Original commit message from CVS:
Patch by: jp.liu <jp_liu at astrocom dot cn>
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix parsing of password field in url. Fixes#407797.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes#405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
