Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes#430677.
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:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes#463215.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes#360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes#339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes#420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index), (check_default),
(audio_convert_prepare_context), (audio_convert_convert):
Also make valgrind happy and avoid copying data in some cases.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes#339837 even more.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes#339837
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS:
2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
Patch by: Josep Torre Valles <josep@fluendo.com>
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
* gst/videotestsrc/videotestsrc.h:
A few array const-ifications.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(gst_audio_convert_transform_caps):
Get structure-name just once.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform can make better decisions on what caps to
negotiate.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix#341696: crash when mixing L+R+C to mono or stereo.
* tests/check/Makefile.am:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
(audioconvert_suite):
Add test for the above, including some generic framework bits for
testing multichannel things.
Original commit message from CVS:
* gst/audioconvert/plugin.c: (plugin_init):
Register the GstAudioChannelPosition enum type with the type
system in the plugin_init function, so that it is known before
any element actually makes use of multi-channel stuff. This is
required for example if one wants to be able to deserialise/use
a caps string with channel positions before any pipeline has
been setup and started, like with gst-launch.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
When clearing an audioconvert context, set tmpbufsize to zero, so
we'll allocate it again later if required.
This fixes audioconvert re-negotiating formats, which previously
segfaulted with a NULL destination buffer.
Original commit message from CVS:
- Don't use non-portable LL suffix on constants, since MSVC doesn't allow
them. These constants all fit into ints anyway.
- Continue to hate nano.
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
Alloc temp storage somewhere else where we can do it more
portable.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
Cleanups, speedups, simplifications, added back support
for 24 bits.
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.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
Use realloc else we lose our original data.
Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.
* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
Original commit message from CVS:
2005-07-04 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
Make caps writable before writing to it.
Fix negotiation in audioconvert some more.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.h:
include missing header file
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_compatible):
use same sign for both channels when converting to/from compatible
channel. Previously used different signs made the signals cancel
each other out and appear like silence. (fixes#167269)
Original commit message from CVS:
2005-02-04 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate): New function, fixates to 256
frames per buffer by default. (Much better than 1.)
(buffer_frames_convert_init): Set the fixate function for both src
and sink pad.
(buffer_frames_convert_link): After success setting nonfixed caps,
get the negotiated caps so we can know how many buffer-frames it
will be. No idea how this worked at all before.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
The return value of fixate_to does not imply that the requested
value was set, so don't assume.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
more overwriting protection due to modifying channels one by one
instead of all at once
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
walk the samples backwards if out_channels > in_channels so we don't
overwrite data
