Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily. Looks like a problem with the 'jess'
visualisation.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
(read_body), (gst_rtsp_connection_receive):
Make sure we can not cancel in the middle of receiving a message.
Fixes#475731.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c:
Increase upper limit for audio queue a bit; fixes preroll problem
with playbin and decodebin2 when playing a quicktime trailer with
multichannel audio via http (#464666).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_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:
Allow othe clocks than the internal clock to be used for the pipeline.
Add property to disable clock provide.
API: GstBaseAudioSrc::provide-clock
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_class_init):
Correctly chain up finalize with the parent class to prevent
memory leaks. Fixes#474880.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(is_multicast_address), (gst_sdp_message_as_text),
(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
(gst_sdp_media_init), (gst_sdp_media_uninit),
(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Separate INIT_ARRAY() and related macros into two versions, one for
structures and one for pointers (e.g., INIT_ARRAY() and
INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
lists of emails and phone numbers.
Add missing const as appropriate.
Change all gint to guint since they all actually represent unsigned
values.
Do not use time as a variable name as it shadows the global time().
Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
Actually implement gst_sdp_message_add_time().
Make gst_sdp_message_add_time() take repeat times as an argument.
Store repeat times in GstSDPTime as a GArray rather than as gchar**.
Corrected the definition of gst_sdp_media_get_bandwidth() (was
misspelled as badwidth).
gst-indented and a little clean up. Fixes#471067.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
Original commit message from CVS:
2007-09-03 Johan Dahlin <jdahlin@async.com.br>
* gst/typefind/gsttypefindfunctions.c (plugin_init):
Add an audio/x-nsf typefind function for the nsfdec element.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_set_gst_timestamp):
Add some more docs for the queue-delay property and fix a typo in a
comment.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Fix typo.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Be less silly with the segment start, just apply the clock-base to the
timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Deprecate the queue handling thread thing and remove the code.
Use new method to calculate the extended timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_copy_entry):
Use g_strndup which does exactly what we want.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add helper function to compare seqnums.
Add helper function to calculate extended timestamps.
API: gst_rtp_buffer_compare_seqnum()
API: gst_rtp_buffer_ext_timestamp()
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_get_entry),
(gst_rtcp_packet_sdes_copy_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix and document SDES item data function.
Add new function that makes a proper copy of SDES item data.
API: gst_rtcp_packet_sdes_copy_entry()
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
The tcp and subparse plugins are under gst, but not totaly free of
dependencies. Handle selection inconfigure.ac, so that they show up
on the final list of what is build and what is not. Maybe they should
better be moved to ext.
Original commit message from CVS:
Patch by: Daniel Díaz <yosoy@danieldiaz.org>
* configure.ac:
* gst/Makefile.am:
Check if libxml provides HTML parser which subparse needs.
Fixes#451970.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that. Fixes#470456.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
We need to set up delayed-linking whenever the caps are non-fixed,
not just when there are multiple types - use gst_pad_is_fixed()
to test.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail):
Add missing separator in PID fallback case.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
Original commit message from CVS:
Based on a patch by: Davyd <davyd at madeley dot id dot au>
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_int32),
(volume_process_int32_clamp), (volume_process_int24),
(volume_process_int24_clamp), (volume_process_int16),
(volume_process_int16_clamp), (volume_process_int8),
(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
* gst/volume/gstvolume.h:
Add support for int32, int24 and int8 to the volume element.
Fixes#445529.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
When calculating the first timestamp of the buffers, don't go below 0
and clip the samples because the offset was on the eos page.
Fixes#466717.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
(gst_ogg_demux_collect_chain_info):
Also submit the eos page when trying to find the first timestamp.
See #466717.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes#467667.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Small cleanups.
On shutdown, don't read the control socket yet.
Set timeout value correctly in all cases.
Add function to check if the server accepts reads or writes.
API: gst_rtsp_connection_poll()
* gst-libs/gst/rtsp/gstrtspdefs.h:
Fix compilation with -pedantic.
Add enum for _poll.
