Original commit message from CVS:
patch by: Wim Taymans <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Revert patch which sends timestamps only on keyframes, as it
breaks playback with current gst-ffmpeg.
Fixes: #515562
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
* tests/check/elements/multifile.c: (GST_START_TEST):
Close some memory leaks spotted by the unit test. Fixes bug #515697.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* gst/spectrum/gstspectrum.c:
* tests/examples/spectrum/.cvsignore:
* tests/examples/spectrum/Makefile.am:
* tests/examples/spectrum/spectrum-example.c:
Add a simple example application for the spectrum element, include it
in the docs, and fix some documentation ambiguities.
Fixes: #348085
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/spectrum/Makefile.am:
Fix includes order
* tests/check/Makefile.am:
Exclude v4l2src from the states test - it takes too long to start.
* tests/check/elements/spectrum.c:
Make the test run properly with CK_FORK=no
Original commit message from CVS:
2008-02-08 Julien Moutte <julien@fluendo.com>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock): Flag
keyframe and delta units correctly when dealign with a
BlockGroup.
Fixes: #514397
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
Need to use gsize here for the size, fixes compiler warning.
* tests/examples/equalizer/.cvsignore:
* tests/examples/equalizer/Makefile.am:
* tests/examples/spectrum/.cvsignore:
* tests/examples/spectrum/Makefile.am:
Add missing files to fix the build.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
Use g_file_[sg]et_contents() instead of using stdio functions.
Should be less error prone.
* tests/check/elements/multifile.c:
Create a temporary directory using standard functions instead of
creating a directory in the current dir.
Original commit message from CVS:
* 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:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
* gst/audiofx/audiowsinclimit.h:
* tests/check/Makefile.am:
* tests/check/elements/audiowsincband.c:
* tests/check/elements/audiowsinclimit.c:
Move the lpwsinc and bpwsinc elements from gst-plugins-bad into
the audiofx plugin, and rename to audiowsinclimit and audiowsincband
respectively.
Fixes: #467666
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_chain):
* tests/check/elements/icydemux.c:
Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without
caps, and add a somewhat useful debug message. Plus test.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c:
Include unistd.h only if HAVE_UNISTD_H is defined
* win32/common/config.h.in:
* win32/common/config.h:
Define socklen_t as it seems it's not defined in default
Visual Studio headers.
* win32/vs6/libgstalpha.dsp:
* win32/vs6/libgstapetag.dsp:
* win32/vs6/libgstavi.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstvideomixer.dsp:
Update project file dependencies and add new source files
Original commit message from CVS:
Patch by: Bjarne Rosengren <bjarne at axis dot com>
* gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
Don't leak buffers when we don't push them downstream.
Fixes bug #514965.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
Add a fixme comment.
* gst/selector/gstoutputselector.c:
Fix same leak as in input-selector.
* tests/icles/output-selector-test.c:
Improve the test.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex):
If there's no entries in the subindex, don't try to do anything stupid,
just return.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/avi/gstavisubtitle.c:
Add documentation for avisubtitle and change class to
Codec/Parser/Subtitle
Original commit message from CVS:
* gst/alpha/Makefile.am:
* gst/alpha/gstalpha.c:
Re-write the 'alpha' plugin to be BaseTransform based, simplifying
some stuff, and making buffer-alloc and resizing work automatically.
No longer crashes on odd frame widths and heights, although there
seems to be a disagreement with ffmpegcolorspace about what size
an AYUV frame with odd height should be.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data):
GStreamer timestamps are PTS values while AVI only knows about DTS
timestamps. Make sure we only copy the DTS as the buffer timestamp when
we are dealing with a key frame.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use g_ascii_strtoll() instead of atoll, which is only
available in C99.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Don't implement get_unit_size() ourselves, the GstAudioFilter base
class already does this for us.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Add symbols from -unused.txt to the right place.
* gst/dvdspu/gstdvdspu.c:
* gst/dvdspu/gstdvdspu.h:
Coherent namespace usage.
* gst/spectrum/gstspectrum.c:
Fix broken XML fragment in doc snippet even more.
Original commit message from CVS:
Based on a patch by:
Victor STINNER <victor dot stinner at haypocalc dot com>
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Set variable to NULL after freeing it to prevent double frees
or make failures by another use of it afterwards more obvious
and fix use of it after the freeing.
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
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
(gst_tag_lib_mux_adjust_event_offsets):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
* sys/osxaudio/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Jordi Jaen Pallares <jordijp at gmail dot com>
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init),
(gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize),
(gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer):
* gst/rtp/gstrtpmp2tpay.h:
Fill the MTU with as many packets as possible. Fixes#491323.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Fix some more leaks. Fixes#497007.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_tcp):
Fix 3 pad leaks. Fixes#496983.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Fix small leak. Fixes#497017.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension),
(qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add suppport for theora in quicktime according to XiphQT.
Original commit message from CVS:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
We don't want the same string multiple times in a tag list for the
same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
this doesn't happen and remove special-case code for GST_TAG_GENRE.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_parse_range):
Don't leak event, don't leak range (fixes#496752).
Original commit message from CVS:
Patch by: Arek Korbik <arkadini@gmail.com>
* gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
Detect RGBA/BGRA correctly on little endian systems.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw skynet be>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_dvd_clut_change_event),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.h:
Extract palette data for dvd subpicture streams and send it
downstream as custom gstreamer dvd event (fixes#453417).
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/wavparse/gstwavparse.c:
Return the result in _activate_pull(). Don't ref element there.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event):
Ref the element when we should, but not when we its not needed. Reflow
the event_handling to not leak the event.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/replaygain/rganalysis.c: (yule_filter):
Avoid slowdown from denormals when processing near-silence input data.
Spotted by Gabriel Bouvigne. Fixes#494499.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(qtdemux_parse_samples):
Properly free QTDemuxSamples array.
Protect table write with a sensible check, some files apparently DO contain
stts values starting with 0 :(
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/qtdemux/qtdemux.c:
Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that
previous commit messed up.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/qtdemux/qtdemux.c:
Sync _handle_src_event() with oggdemux. In avidemux also ref the
element when we should, but not when we its not needed.
Original commit message from CVS:
* gst/equalizer/demo.c: (draw_spectrum):
* gst/spectrum/demo-audiotest.c: (draw_spectrum):
* gst/spectrum/demo-osssrc.c: (draw_spectrum):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
Change the meaning of the magnitude values given in the
GstMessages by spectrum to decibel instead of
decibel+threshold.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
And continue to update docs. Also include some sample code
for the n-band equalizer in the docs.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Update docs and property ranges to the real values.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c:
Now do the scaling right for real. Also initialize a previously
uninitialized variable.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Return FALSE if we can't handle a query instead of changing the
format. Ignore fact when dealing with mpeg audio.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (main):
Use autoaudiosink instead of alsasink and use a sine wave.
* gst/spectrum/gstspectrum.c:
Fix the magnitude calculation.
Original commit message from CVS:
* gst/equalizer/demo.c: (main):
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_class_init), (setup_filter):
Allow setting 0 as bandwidth and handle this correctly.
Also handle a bandwidth of rate/2 properly.
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_class_init):
Make it possible to generate a N-band equalizer with 1 bands. The
previous limit of 2 was caused by a nowadays replaced calculation
doing a division by zero if number of bands was 1.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* configure.ac:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstdynudpsink.h:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstmultiudpsink.h:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsink.h:
Fix includes for MSVC and GLib-2.14.0 (#492388).
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
No more pipe define since GLib-2.14.0, need to use _pipe() directly.
Original commit message from CVS:
* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
(gst_mulawdec_chain):
* gst/law/mulaw-decode.h:
Calculate outgoing buffer duration if incoming buffer didn't have a
valid duration.
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/demo.c: (on_window_destroy), (on_configure_event),
(on_gain_changed), (on_bandwidth_changed), (on_freq_changed),
(draw_spectrum), (message_handler), (main):
Add small demo application based on the spectrum demo applications
that gets white noise as input, pushes it through an equalizer and
paints the spectrum. For every equalizer band it's possible to set
gain, bandwidth and frequency.
* gst/equalizer/gstiirequalizer.c: (setup_filter):
Add some guarding against too large or too small frequencies and
bandwidths. Also improve debugging a bit.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (arg_to_scale),
(setup_filter), (gst_iir_equalizer_compute_frequencies):
Replace filters with a bit better filters for which we can actually
find documentation, which don't change anything on zero gain, etc.
Make the frequency property of the bands writable, rename the
band-width property to bandwidth and change the meaning to the
frequency difference between bandedges, change the meaning of the
gain property to dB instead of a weird scale between -1 and 1 that
has no real meaning.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie):
Smarter combine_flow code that also deals with downstream elements
returning UNEXPECTED when they receive data out of the segment
boundaries. Fixes#491305.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_request_new_pad):
Let's not call every request pad we create "sink%d", that'll
create problems if there's to be more than one pad. Fixes#490682.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/interleave.c:
Add unit test for the above.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved):
Fix race when pausing a RTSP stream in interleaved.
Fixes#475784.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize):
Use correct unref function for buffers. #488844.
Original commit message from CVS:
Based on patch by: Laurent Glayal <spglegle yahoo fr>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
When the socket is used by the app for other purposes, don't generate an
error if there is activaty on the socket that is not data related.
Fixes#487488.
Original commit message from CVS:
Patch by: Anders Skargren <anders dot skargren at axis dot com>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
Set marker bit correctly.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init),
(gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init),
(setup_filter), (gst_iir_equalizer_setup):
* gst/equalizer/gstiirequalizer.h:
Move bandwidth property to the separate bands and add float64 support.
Original commit message from CVS:
Based on patch by: Jason Kivlighn <jkivlighn gmail com>
* gst/id3demux/id3v2frames.c:
Extract license/copyright URIs from ID3v2 WCOP frames
(Fixes#447000).
* tests/check/elements/id3demux.c:
* tests/files/Makefile.am:
* tests/files/id3-447000-wcop.tag:
Add simple unit test.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
a GstClockTime.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
More seeking fixes, mostly passing around the new playback segment in
order to configure it properly.
Also reset base_time of udp sources when setting them back to PLAYING as
a temporary hack until core supports seek in live sources properly.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c:
* gst/id3demux/gstid3demux.h:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c:
Port ID3 tag demuxer over to the new GstTagDemux in -base
(now would be a good time to test re-importing your music
collection).
Original commit message from CVS:
* gst/apetag/Makefile.am:
* gst/apetag/gstapedemux.c:
* gst/apetag/gstapedemux.h:
* gst/apetag/gsttagdemux.c:
* gst/apetag/gsttagdemux.h:
Port APE tag demuxer over to the new GstTagDemux in -base.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_internal_src_query),
(gst_rtspsrc_handle_src_query), (new_session_pad),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_loop_send_cmd):
Improve flushing behaviour.
Set state of the udp sources to PAUSE/PLAYING correctly.
Handle events and queries for UDP and TCP transport now.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Fix "Index entry has invalid stream nr 1".
Add support for muxing aac - work in progress (see #482495).
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
* gst/rtsp/gstrtspsrc.h:
Parse bandwidth modifiers, they are not yet configured in the session
manager because we don't have an API for that yet.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
Use shiny new function in -base to get the default clock-rate.
Update some docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
Fix setting the discont flag on the first buffer
pushed downstream for formats with private codec
data that needs to be deserialised into buffers
(such as vorbis and FLAC when in a matroska container).
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_handle_buffer):
* gst/rtp/gstrtpmp4vpay.h:
Free the config string. Fixes#480707.
Clean up the timestamp code a little.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
Set timestamps on RTP buffers in interleaved mode.
Mark first buffers with a DISCONT.
Remove flush hack now that sync for live sources has been figured out.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Fail if we don't know the quicktime format.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
* gst/apetag/gstapedemux.c:
Add support for the new GST_TAG_COMPOSER (#459809).
Original commit message from CVS:
* gst/law/alaw-decode.c:
* gst/law/alaw-decode.h:
* gst/law/alaw-encode.c:
* gst/law/alaw-encode.h:
* gst/law/alaw.c:
* gst/law/mulaw-conversion.h:
Compulsive clean-ups: use boilerplate macros, add debug
categories, fix up things to conform to symbol nomenklatura,
etc.
Original commit message from CVS:
Based on patch by: Laurent Glayal <spglegle yahoo fr>
* gst/law/alaw-decode.c:
* gst/law/alaw-encode.c:
Use static tables for A-Law decoding and encoding; this makes
A-Law decoding and encoding less CPU-intensive, but increases
the binary size a bit. Leaving old code around for now,
selectable by a define in the code. Fixes#435435.
