Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Small docs fixes/updates.
* gst-libs/gst/video/gstvideosink.h:
Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
removed from the base sink API between 0.9.6 and 0.9.7).
API: add GST_VIDEO_SINK_CAST and use it for the height/width
accessor macros, so we don't do a runtime GObject type check every
time we use them.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Declare variables at the beginning of a block. Fixes#383195.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Fix GstBaseRTPAudioPayload structure so the whole GObject
inheritance business actually works (parent class instance structure
must always come first in the derived class instance structure).
Original commit message from CVS:
* configure.ac:
Bump liboil requirement to 0.3.8.
* gst-libs/gst/riff/riff-media.c:
Add Dirac fourcc.
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.h:
Use liboil's stdint.h.
* gst/videotestsrc/videotestsrc.c:
Remove liboil related ifdef's, since they aren't needed now, and
won't work with future versions.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
add h263/h264 variants to the caps, Fixes#363118
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Use g_strerror instead of strerror so we get UTF-8.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_init):
Fix and activate base audio payloader.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
(gst_riff_parse_info):
If strings in INFO chunk are not UTF-8, do something similar to
what we do for ID3v1 tags: check a number of environment variables
(GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
character sets to try, otherwise try the current locale and/or fall
back on ISO-8859-1. Fixes#360552.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
Original commit message from CVS:
* gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
(gst_tuner_set_channel), (gst_tuner_get_channel),
(gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
(gst_tuner_set_frequency), (gst_tuner_get_frequency),
(gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
(gst_tuner_find_channel_by_name):
Fix some function guards, add some more function guards.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Don't crash when ringbuffer is not yet created.
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
Fixes#361634.
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Activate pads befre adding them to running elements.
Original commit message from CVS:
Patch by: Sebastien Cote <sebas642 at yahoo.ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_finalize):
Fix two small memory leaks (#361456).
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-libs/gst/interfaces/xoverlay.c:
(gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
Some more guards against invalid input.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_change_state):
Also call parent state change function to activate pads.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg1_parse_header), (mpeg1_sys_type_find):
Add some more debug info in mpeg typefinding.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp):
the source pad always uses fixed caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse dates that are followed by a time as well (#357532).
* tests/check/libs/tag.c: (test_vorbis_tags):
Add unit test for this.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_base_init):
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
(gst_tag_register_musicbrainz_tags):
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
depend on libgsttag. This is required so we can extract/read tags like
DISCID without depending on libgstcddabasesrc (which used to register
them).
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
tags (also see #347848).
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
Log vorbis comments we are actually writing. Const-ify array.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Added MPEG-4 AAC and id and caps. Fixes#357289
Added WMA9 Lossless id.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Small cleanups.
Fix some leaks.
Refactored the process method and added methods to push from the process
vmethod.
Use _scale functions.
API: gst_base_rtp_depayload_push_ts
API: gst_base_rtp_depayload_push
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
timestamps are uint.
Original commit message from CVS:
* gst-libs/gst/interfaces/videoorientation.c:
(gst_video_orientation_iface_init),
(gst_video_orientation_get_hflip),
(gst_video_orientation_get_vflip),
(gst_video_orientation_get_hcenter),
(gst_video_orientation_get_vcenter),
(gst_video_orientation_set_hflip),
(gst_video_orientation_set_vflip),
(gst_video_orientation_set_hcenter),
(gst_video_orientation_set_vcenter):
Add since tags to new API docs, ChangeLog surgery (forgot API keyword
in ChangeLog)
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Early morning compilation fix.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Reorder the audio formats a bit for clarity.
Detect and create caps for MSGSM and MSN (WAV49).
Fixes#356596.
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
Small cleanups, move error handling out of normal flow for clarity.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Do the delay calculation in the source/sink base classes as this is
specific for the capture/playback mode.
Try to fixate a bit better, like round depth up to a multiple of 8
bigger than width.
Handle underruns correctly by marking DISCONT on buffers and adjusting
timestamps to handle the gap.
Set offset/offset_end correctly on buffers.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Remove resync and underrun recovery from the ringbuffer.
Fix ringbuffer read code on under/overrun.
Original commit message from CVS:
* configure.ac:
We require 0.10.10.1 now because of _wait_preroll().
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Use gst_base_sink_wait_preroll().
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk), (gst_riff_parse_file_header),
(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
(gst_riff_parse_info):
Protect public functions against bad input.
Do some cleanups.
Fix documentation.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add voxware audio IDs (even if we can't play it) (#351795).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
Const-ify some arrays and use G_N_ELEMENTS instead
of wasting oodles of RAM on terminator bits.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* tests/check/libs/tag.c: (GST_START_TEST):
And the same for _to_vorbiscomment_buffer(): allow
id_data_len == 0 for speex.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer):
Allow id_data_len == 0 (needed for vorbis comments in Speex files).
Also add some checks to make sure we don't memcmp() beyond the end of
vorbiscomment buffer if the ID to check for is larger than the buffer.
* tests/check/libs/tag.c: (GST_START_TEST):
Some more tests for gst_tag_list_from_vorbiscomment_buffer().
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Don't try to GObject scan the netbuffer as it's not a GObject.
Fixes#351308.
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Document GstNetBuffer.
