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Original commit message from CVS:
Patch by: Cody Russell <bratsche at gnome org>
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
Avoid unnecessary class cast check in class_init
functions (#337747).
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::stop so that audioresample can clear it...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
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Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
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audioresample can handle such as 32 bits in and f...
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
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parameters by a simple alias to gstreamer sta...
Original commit message from CVS:
* gst/audioresample/debug.h:
replace debug macros with variable number of parameters
by a simple alias to gstreamer standard debug macros
(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
supported by MSVC 6.0 and 7.1)
* gst/audioresample/resample.h:
define M_PI and rint for WIN32
* win32/common/libgstaudio.def:
* win32/common/libgstriff.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
add new exported functions
* win32/vs6:
update project files
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Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add audioresample to docs.
* gst/audioconvert/gstaudioconvert.c:
Add revision date.
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_base_init), (gst_audioresample_class_init),
(gst_audioresample_init), (gst_audioresample_dispose),
(audioresample_get_unit_size), (audioresample_transform_caps),
(resample_set_state_from_caps), (audioresample_transform_size),
(audioresample_set_caps), (audioresample_event),
(audioresample_do_output), (audioresample_transform),
(audioresample_pushthrough), (gst_audioresample_set_property),
(gst_audioresample_get_property), (plugin_init):
* gst/audioresample/gstaudioresample.h:
Added docs.
Small code cleanups.
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Original commit message from CVS:
* gst/audioresample/resample.h:
As before, but for o_buf
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unsigned char *, not void *, since we do arithm...
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
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audioresample.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Don't leak all input buffers to audioresample.
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Original commit message from CVS:
expand tabs
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negotiation etc.. work fine.
Original commit message from CVS:
* gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
* gst/audioresample/buffer.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c: (resample_input_flush),
(resample_input_pushthrough), (resample_input_eos),
(resample_get_output_size_for_input),
(resample_get_input_size_for_output), (resample_get_output_size),
(resample_get_output_data):
* gst/audioresample/resample.h:
* gst/audioresample/resample_ref.c: (resample_scale_ref):
Fix audioresample, seek torture, new segments, reverse negotiation
etc.. work fine.
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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst/audioresample/gstaudioresample.c:
Segment update fix.
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Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
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Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
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Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
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Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
fix distcheck
* gst/audioresample/resample.c:
fix wrong docstring
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
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Original commit message from CVS:
use base class' newsegment to properly timestamp
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Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
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Original commit message from CVS:
add a check for audioresample
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Original commit message from CVS:
show some info on what's left in the queue
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transform size protect against possible mem corr...
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
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Original commit message from CVS:
fix broken header setup in Makefile.am
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Original commit message from CVS:
dist more
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Original commit message from CVS:
port audioresample to basetransform
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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
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