Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize):
Unparent all bands from the equalizer when finalizing to stop
leaking them.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_add_internal), (gst_multiudpsink_remove):
* gst/udp/gstmultiudpsink.h:
Add property to automatically join a multicast group or not. This can be
useful when sharing a socket between multiple elements.
Fixes#509531.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/multipart/Makefile.am:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartdemux.h:
* gst/multipart/multipartmux.c:
* gst/multipart/multipartmux.h:
Re-add multipartdemux to the docs. Last round of section cleanup.
Original commit message from CVS:
As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use atoll to parse the rtptime with enough precision. Fixes#509329.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
Initialise variables to work around (false) 'foo might be used
uninitialized in this function' warnings by gcc-3.3.3 (#509298).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Generate the image-type values correctly. Leave them out of the caps
when outputting a "preview image" tag, since it only makes sense
to have one of those - the type is irrelevant.
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_open):
If we can, mark the mixer multiple open when we use it, in case
(for some reason) the process wants to open it again elsewhere.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
* gst/rtp/gstrtptheorapay.c:
Fix the clock rate to 90000 as required by the RFC.
Fixes#508644.
Original commit message from CVS:
Based on patch by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst/id3demux/id3v2frames.c: (parse_comment_frame):
Make sure the ISO 639-X language code in ID3v2 COMM frames
is actually valid UTF-8 (or rather: ASCII), so we don't end
up with non-UTF8 strings in tags if there's garbage in the
language field. Also make sure the language code is always
lower case. Fixes: #508291.
Original commit message from CVS:
reviewed by: Edward Hervey <edward.hervey@collabora.co.uk>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry), (_do_init),
(gst_videomixer_child_proxy_get_child_by_index),
(gst_videomixer_child_proxy_get_children_count),
(gst_videomixer_child_proxy_init), (gst_videomixer_reset),
(gst_videomixer_init), (gst_videomixer_request_new_pad),
(gst_videomixer_release_pad), (gst_videomixer_fill_queues):
Implement GstChildProxy interface.
Send newsegment at the right moment
Fixes#488879
Original commit message from CVS:
* gst/alpha/Makefile.am:
* gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
(gst_alpha_sink_event), (gst_alpha_chain),
(gst_alpha_change_state), (plugin_init):
Make the various properties of 'alpha' controllable. This allows doing
niceties like fade-in/fade-out.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (COMMON_VIDEO_CAPS_NO_FRAMERATE),
(videosink_templ):
Also fix up pad templates to indicate that image/jpeg doesn't
absolutely require the framerate property to be set (#504081).
Original commit message from CVS:
Based on patch by: Wouter Cloetens <wouter at mind be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
(gst_matroska_mux_finish), (gst_matroska_mux_collected):
* gst/matroska/matroska-mux.h:
Keep track of first and last timestamps for each incoming stream,
so we can calculate the total duration for live sources and other
input where we can't query the duration from the start or where
there's no constant framerate from which we can deduce the
duration; also use calculated/observed duration if it is bigger
than the previously queried duration. Furthermore, use
gst_pad_query_peer_duration() and take into account that it may
return TRUE but still a duration of CLOCK_TIME_NONE, which easily
screws up comparisons when using unsigned integers. Fixes#504081.
Original commit message from CVS:
* configure.ac:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_init), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_init),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_transform_ip):
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
(gst_audio_panorama_transform):
* gst/level/gstlevel.c: (gst_level_init):
Make elements GST_BUFFER_FLAG_GAP aware and call
gst_base_transform_set_gap_aware for this.
Bump core requirement to CVS.
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
Also sync GObject properties to the controller if operating
in passthrough mode.
Original commit message from CVS:
* gst/avi/gstavi.c:
increase rank because no known issues anymore ...
* gst/avi/gstavisubtitle.c:
send subtitle name to the srcpad
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Implement redirect for the DESCRIBE reply. Fixes#506025.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_loop):
* gst/wavparse/gstwavparse.c: (gst_wavparse_chain):
* sys/ximage/gstximagesrc.c: (composite_pixel):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x (it's
not really nice to abort in any case). Fixes#505745.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (IS_BOM_UTF8), (IS_BOM_UTF16_BE),
(IS_BOM_UTF16_LE), (IS_BOM_UTF32_BE), (IS_BOM_UTF32_LE),
(gst_avi_subtitle_extract_file), (gst_avi_subtitle_parse_gab2_chunk):
Detect other UTF byte order markers and convert to UTF-8 as
appropriate.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (src_template),
(gst_avi_subtitle_extract_utf8_file),
(gst_avi_subtitle_parse_gab2_chunk), (gst_avi_subtitle_chain),
(gst_avi_subtitle_base_init), (gst_avi_subtitle_class_init),
(gst_avi_subtitle_init), (gst_avi_subtitle_change_state):
* gst/avi/gstavisubtitle.h:
Refactor a bit; fix name extraction; don't assume all the data
in the chunk is actually subtitle data, there may be padding at
the end; fix GST_ELEMENT_ERROR usage; store extracted subtitle
file so it's there to send again after a seek (for future use).
Original commit message from CVS:
* gst/avi/Makefile.am:
* gst/avi/gstavi.c:
* gst/avi/gstavisubtitle.c:
* gst/avi/gstavisubtitle.h:
* tests/check/Makefile.am:
* tests/check/elements/avisubtitle.c:
* win32/common/config.h:
Add avi subtitle element for bug #442034. Need seeking support
and more support for character conversion.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
When subsequent files are read, if the file doesn't exist, send
an EOS instead of causing an error.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_url_link_frame):
Parse WOAF frames and put the result into GST_TAG_CONTACT,
which is where it would end up if the same information was
put in a vorbis comment (don't think it's worth adding a
new URI tag for this). Fixes#488112.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(next_start_code), (is_nal_equal), (gst_rtp_h264_pay_decode_nal),
(encode_base64), (gst_rtp_h264_pay_parse_sps_pps),
(gst_rtp_h264_pay_handle_buffer):
* gst/rtp/gstrtph264pay.h:
Use higher performance start-code searching.
Parse NALs and store SPS, PPS and profile in the caps so that they can
be used in the SDP. Fixes#502814.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
Copy timestamp from input to output. Not very perfect yet but better
than nothing. Fixes#503023.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
The transform_ip() methods should do nothing if in passthrough mode.
It might get non-writable buffers in that case but the buffer might
as well be writable.
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform):
The transform() methods won't be called in passthrough mode and
otherwise the buffer is always writable so don't check here.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event):
Fix seeking in .wav files again (#501775). Some people seem to think
they don't need to test their changes when they're just 'reflowing'
some code.
Original commit message from CVS:
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_init),
(gst_auto_video_sink_create_element_with_pretty_name),
(gst_auto_video_sink_find_best),
(gst_auto_video_sink_set_property),
(gst_auto_video_sink_get_property):
* gst/autodetect/gstautovideosink.h:
Fix docs.
Use same error reporting code as autoaudiosink.
Add property to filter sinks based on caps. Only select raw video sinks
by default for backwards compat.
API: GstAutoVideoSink::filter-caps
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_set_property),
(gst_auto_audio_sink_get_property):
* gst/autodetect/gstautoaudiosink.h:
Add property to filter sinks based on caps. Only select raw audio sinks
by default for backwards compat. Fixes#417420.
API: GstAutoAudioSink::filter-caps
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init),
(gst_rtp_h263_depay_process):
Code beautification.
Added debug statements.
Don't bit-shift everything, just do operations on last/first byte
instead.
Original commit message from CVS:
Patch by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process):
Fix wrong comparison in overrun check. Fixes#499239 some more.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_process):
* gst/rtp/gstrtph263depay.h:
Fix h263 depayloader so that ANY h263 decoder can handle the outgoing
stream.
Original commit message from CVS:
Based on Path by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
* gst/rtp/gstrtpmp4adepay.h:
Fix depayloading when multiple frames are inside one RTP packet.
Fixes#499239.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process):
Read the I flag for Mode A h263 rtp stream and set the
GST_BUFFER_FLAG_DELTA_UNIT accordingly.
Fixes#499383
