Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Streaming mode fixes: don't unref buffer we don't own any longer;
remove bogus adapter flush. Fixes#419338.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_init):
A 10 band EQ should be initialized to 1 bands and not to 3.
Original commit message from CVS:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END here as well.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
the image format a variable-length NUL-terminated string; in
versions before that the image format is a fixed-length string of
3 characters (see #348644 for a sample tag).
Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
Printf format fixes; also add some missing quotes in translated
strings. Fixes#416728 and #416727.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
Tim and I can't think of any reason the child audio sink needs to
be set back to NULL after successfully determining that it can
reach READY - it gets immediately set back to READY by the caller
anyway, causing an unnecessary close/open of any audio devices
involved.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
When activated, remove the udpsrc timeout, we have dataflow and timeouts
will later be handled by the jitterbuffer.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
Fix stream position reporting after a seek. Fixes#416445.
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init),
(gst_iir_equalizer_band_get_type),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_finalize), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_setup), (plugin_init):
* gst/equalizer/gstiirequalizer.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_class_init),
(gst_iir_equalizer_nbands_init),
(gst_iir_equalizer_nbands_set_property),
(gst_iir_equalizer_nbands_get_property):
* gst/equalizer/gstiirequalizernbands.h:
Refactor plugin into a base class and a first subclass (nband eq). The
nband eq uses GstChildProxy and is controlable. More subclasses will
follow.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_chain):
Make avidemux accept optional header chunks in any order.
Fixes#415446.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* 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:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
Share qtdemux debug category across all files, otherwise all debugging
in files other than qtdemux.c would end up in the default category.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_event), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
One FIXME less, by resolving message timestamps against the playback
segment.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_sink_activate):
Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
caps passed to it (previouslly one code path assumes it takes ownership
while another one assumes it doesn't).
* configure.ac:
* tests/files/Makefile.am:
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
Add directory where data for unit tests can be stored.
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
(read_tags_from_file), (run_check_for_file),
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
Add unit test for id3demux, and in particular for bug #407349. Only
testing pull-mode for now; push mode doesn't work yet because the test
files are smaller than ID3_TYPE_FIND_MIN_SIZE.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_obsolete_tdat_frame):
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
the four-digit number will be interpreted as a year, whereas it is
month and day in DDMM format. Instead, parse TDAT frames and fix up
the date in the GST_TAG_DATE tag later if we also extracted a year.
Fixes#407349.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_transform_ip):
Fix and cleanup default property values.
Add FIXMEs for stuff that looks rather wrong.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
(gst_multipart_find_pad_by_mime):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* ext/libcaca/Makefile.am:
* gst/debug/Makefile.am:
Don't mix tabs and spaces (#414168).
Original commit message from CVS:
Patch by: René Stadler <mail@renestadler.de>
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Handle rounding better to not drop last sample frame. Fixes#356692
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
Errors from the udp sources are not fatal unless all of them are in
error.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another interesting test url.
* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
Don't allow getting header fields from data packets.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (append_auth_header),
(rtsp_connection_send), (rtsp_connection_set_auth):
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
from icecast to replace it. Relicensed from GPL courtesy of Mike
Smith.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
Original commit message from CVS:
* gst/rtsp/URLS:
Add example H264 rtsp url.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
Don't convert values to lowercase or we might mess up base64 encoded
properties.
Original commit message from CVS:
* gst/rtp/README:
Fix case of string params.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Fix depayloader, support more packet types.
Add sync codes to make sure the packetizer can do its job.
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
Fix caps case again.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
As spotted by: Peter Kjellerstedt <pkj at axis com>:
Clear stack allocated SDPMedia struct before calling _init() on it.
Clarify this in the docs as well.
Original commit message from CVS:
Patch by: jp.liu <jp_liu at astrocom dot cn>
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix parsing of password field in url. Fixes#407797.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes#405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
Original commit message from CVS:
* gst/monoscope/Makefile.am:
* gst/monoscope/gstmonoscope.c:
Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
(but no LIBS, since we only use defines from the headers).
Original commit message from CVS:
Based on patch by: Jonathan Matthew <jonathan at kaolin wh9 net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
(gst_wavparse_stream_data):
Fix massive memory leak when operating in streaming mode due to
GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
Fixes#407057.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Save some memory (8%) by repacking the index entry structure (more to
come). Add more FIXMEs to questionable parts.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/gstdebug.c: (plugin_init):
* gst/debug/gstpushfilesrc.c:
* gst/debug/gstpushfilesrc.h:
Add code for a pushfilesrc element that implements a pushfile:// URI
handler, to make debugging push-mode operation of demuxer/decoders
that support both easier in connection with seek/playbin/etc.
The element isn't registered at the moment.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Comment a #if 0 in caps template definition as VS6 seems to
do not support it.
* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
Use gst_guint64_to_gdouble for conversion.
* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
Move variables declaration before the first instruction.
* gst/rtsp/rtspdefs.c:(rtsp_strresult):
Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
And don't include netdb.h for G_OS_WIN32
* gst/rtsp/sdpmessage.c:(sdp_parse_line):
This initialization SDPMedia nmedia = {.media = NULL }; is not supported
by VS6 then use an other way to initialize SDPMedia structure.
* gst/udp/gstdynudpsink.h:
* gst/udp/gstdynudpnetutils.h:
Do not include <sys/time.h> for G_OS_WIN32
* gst/udp/gstudpsrc.c:
Define socklen_t as int for G_OS_WIN32
* win/common/config.h.in:
Undef HAVE_NETINET_IN_H
* win32/vs6/gst_plugins_good.dsw:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstautogen.dsp:
* win32/vs6/libgstaudiofx.dsp:
* win32/vs6/libgstudp.dsp:
Add and update project files.
* win32/common/gstudp-enumtypes.c:
* win32/common/gstudp-enumtypes.h:
Add a copy of udp enumtypes to win32/common as in core
and base.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
Explicitly cast result of pointer arithmetic to integer in order to
avoid compiler warnings on some 64-bit systems. Should fix#406018.
Original commit message from CVS:
* docs/plugins/inspect/plugin-rtp.xml:
Update for new elements.
* gst/debug/progressreport.h:
Commit newly-created header file as well.
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.hierarchy:
* gst/debug/Makefile.am:
* gst/debug/progressreport.c: (gst_progress_report_post_progress),
(gst_progress_report_do_query), (gst_progress_report_report):
Make progressreport element post messages with the current progress
on the bus. Also add some basic docs for it.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
Add cast to avoid compiler warnings with older GLib versions
where the nick/name members in GEnumValue are not declared as
constant strings.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
(gst_audio_amplify_class_init), (gst_audio_amplify_init),
(gst_audio_amplify_set_process_function),
(gst_audio_amplify_setup):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_setup):
* gst/audiofx/audioinvert.h:
Some small cleanups and port both elements to the new GstAudioFilter
base class to save a few lines of common code.
* gst/audiofx/Makefile.am:
Link against libgstaudio for the above changes
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace):
Rename "values" property to "band-values" and change type into a
GValueArray, so it's more easily bindable and the range of the
values passed in is defined and checked etc.; also do some
locking.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_transform_packed_complex):
Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
* tests/icles/videocrop-test.c: (check_bus_for_errors),
(test_with_caps), (main):
Block streaming thread before changing filter caps while the
pipeline is running so that we don't get random not-negotiated
errors just because GStreamer can't handle that yet.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_activate_streams):
Convert SDP fields to upper/lowercase following the rules in the SDP to
caps document.
Original commit message from CVS:
* gst/rtp/README:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix case of encoding-name and key/value pairs to match the document.
This is to make interoperation with SDP case-insensitive as required by
the relevant RFCs.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
(gst_multi_file_sink_class_init):
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init):
* gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer),
(gst_mve_video_palette), (gst_mve_video_code_map),
(gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create),
(gst_mve_demux_chain):
* gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk):
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/mve/mvevideodec16.c: (ipvideo_copy_block):
* gst/mve/mvevideodec8.c: (ipvideo_copy_block):
* gst/mve/mvevideoenc16.c: (mve_encode_frame16):
* gst/mve/mvevideoenc8.c: (mve_encode_frame8):
Use proper print statements.
Fixes build on mac os x.
<wingo> oo look at me my name is edward i'm hacking on macos wooo
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
* gst/rtp/gstrtpL16pay.h:
Fill up to MTU using adapter.
Timestamp rtp packets.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
Use G_GSIZE_FORMAT in print statements for portability.
Fixes build on macosx.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
(gst_rtp_L16_depay_plugin_init):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
(gst_rtp_L16_pay_plugin_init):
* gst/rtp/gstrtpL16pay.h:
Port and enable raw audio payloader/depayloader. Needs a bit more work
on the payloader side.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
* gst/rtsp/gstrtspsrc.h:
Only unblock the udp pads when we linked and activated them all.
Fixes#395688.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes#397162
* 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:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
Unblock pads after adding the pads to the element so that autopluggers
get a change to link something. Possibly fixes#395688.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes#396057
* 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:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
Parse config params as string and int.
Parse and use AU header length
Original commit message from CVS:
* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
* gst/smpte/gstmask.c: (_gst_mask_register):
* gst/smpte/gstmask.h:
* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
(gst_smpte_paint_triangle_clock):
constify some static structs.
Don't update the mask if nothing changed to the params.
Make sure we never draw outside of the picture. Fixes#398325.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
Error out properly when pull_range fails while we're reading the
headers, instead of just pausing the task silently. Fixes#399338.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Some more sanity checks to make sure the input formats match and the
input pads are actually negotiated, in case someone tries to feed
buffers from fakesrc or filesrc. Fixes#398299.
Also const-ify an array, just because we can.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
Ignore previous commit, that was only valid for widths and heights
that are multiples of 4.
Copy over size/stride macros from jpegdec. This allows the element
to work with any width,height...
... but puts in evidence that the actual transformations only work
with width/height that are multiples of 4.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Allocate buffers of the right size.
The proper size of a I420 buffer in bytes is:
width * height * 3
------------------
2
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_init):
Proxy getcaps on sink pads too, so that we either end up with the
same dimensions on all pads or error out if that's not possible
(seems to work even!). Fixes#398086, I think.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo circular-chaos org>
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
(gst_audio_panorama_set_process_function):
Use a function array for process methods, add more docs and define the
startindex of enums.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
* tests/check/elements/avimux.c: (teardown_src_pad):
Add support for more than one audio stream; write better AVIX
header; refactor code a bit; don't announce vorbis caps on our audio
sink pads since we don't support it anyway. Closes#379298.
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads):
Use fixed caps on src pads.
(gst_deinterleave_remove_pads): Remove src pads, not sink pads. I
seem to have reverse midas disease!
(gst_deinterleave_process): Proxy timestamps, offsets, durations,
and set caps on outgoing buffers. Fixes#395597, I think.
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/interleave.c (gst_interleave_init): Init the
activation mode properly.
(gst_interleave_src_setcaps, gst_interleave_src_getcaps)
(gst_interleave_init): Set a setcaps and getcaps function on the
src pad, so that we can implement pull-mode negotiation.
(gst_interleave_sink_setcaps): Renamed from
gst_interleave_setcaps, as it only does the sink logic now.
Implement both for pull-mode and push-mode.
(gst_interleave_process): Set caps on our outgoing buffer.
(gst_interleave_src_activate_pull): Fix some more bogus casts.
What is up with this.
