Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
(volume_transform_ip):
Increase "volume" property to 10.0. Fixes#340369.
Set the process function to NULL when capsnego fails so that
we properly error out.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Refine musepack typefinding a bit. Return MAXIMUM
probability when we detect stream version 7 to make
sure the mpeg audio typefinder doesn't trump us.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
interpret the out[] buffer in the order the bytes are actually
put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
Other tests should use BYTE_ORDER since the array is filled in
with actual values
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
when a test fails, give an indication of which it is
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_init):
send events from src-pad to all sink-pads fixes#338657
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
(alsasink_parse_spec):
query witdh capabilities from alsa, fixes#338919
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_remove_client_link):
* gst/tcp/gstmultifdsink.h:
Fix race condition in multifdsink that can lead to spurious
duplicate clients. this patch adds a new signal that is fired when
multifdsink has removed all references to the fd.
Fixes#339574.
Updated documentation.
API: client-fd-removed signal added
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
When asking g_value_array_new to prealloc elements, we may as well
ask for the right number of elements.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
(gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_get_property),
(gst_gnome_vfs_src_send_additional_headers_callback),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
(gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Remove ICY handling (mostly) from gnomevfssrc, in favour of
proper shared support within icydemux.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_swap_prev), (gst_video_rate_chain):
fix up docs
fix a leak when no caps negotiated
fix counting of input frames
* tests/check/elements/.cvsignore:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(GST_START_TEST), (videorate_suite):
add tests for these
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (get_float_caps),
(GST_START_TEST), (audioconvert_suite):
Added check for correct clipping when doing float samples
in audioconvert.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(resample_set_state_from_caps):
Add support for other formats audioresample can handle such as
32 bits in and float and 64 bits float. Fixes#301759
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_render_text):
Don't strip newlines from the text. Also, center lines
within multi-line paragraphs (#339405).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
Fix wavpack typefinding to work in more cases (don't peek
for chunks of multiple hundred kBs at once, but process
things step-by-step in smaller units). Fixes#339786.
Original commit message from CVS:
2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
patch by: Wim Taymans
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek):
make sure correct newsegments are sent, so that the decoder
and the demuxer agree on timestamps. Fixes playback of a lot
of Ogg files that do not start from 0. Fixes#339833.
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
* tests/check/Makefile.am:
* tests/check/elements/videorate.c: (assert_videorate_stats),
(setup_videorate), (cleanup_videorate), (GST_START_TEST),
(videorate_suite), (main):
Fix an infinite loop if frames are passed in with wrongly ordered
timestamps. Fixes#339013.
Original commit message from CVS:
Patch by: Tim-Philipp Müller <tim at centricular dot net>
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
fix typefinding on some ISO files. Fixes#339212.
Original commit message from CVS:
Patch by: Jan Schmidt
* gst/playback/gststreamselector.c:
(gst_stream_selector_bufferalloc):
Restore old StreamSelector behaviour.
Fixes#338419.
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
Fix some memory leaks: on finalize, free buffers left in the queue
before destroying the queue; in _push(), unref rtp_buf even if
the process vfunc returned a NULL buffer as output buffer (#337548);
demote some recuring debug messages to LOG level.
