ChangeLog 1.02 MB
Newer Older
1 2 3 4 5 6 7 8 9 10
2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	Set caps on outgoing buffers.

	* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
	(gst_video_rate_event), (gst_video_rate_chain):
	* gst/videorate/gstvideorate.h:
	Fix videorate some more. Fixes #357977

11 12 13 14 15 16 17
2006-09-28  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/adder.c: (adder_suite):
	  Don't set timeout to 6 seconds when we're running
	  in valgrind ... (and how is 6 seconds longer than
	  the default anyway?)

18 19 20 21 22 23 24 25 26 27 28
2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	(gst_audio_rate_sink_event), (gst_audio_rate_convert),
	(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
	Keep sink and src segment to keep track of time and support more
	input formats.
	Fix bogus next_offset and run_time calculation, don't understand how
	this could have worked before. Fixes #357976.
	Remove some unneeded vars.

29 30 31 32 33 34
2006-09-28  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (remove_sinks):
	  Only remove visualisation from visbin if there is a visbin (or:
	  don't throw warnings when closing totem without playing a file).

35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51
2006-09-27  Wim Taymans  <wim@fluendo.com>

	* 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.

52 53 54 55 56
2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Another attempt to make the gen64 buildbot happy.

57 58 59 60 61 62 63 64 65
2006-09-27  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>

	* ext/libvisual/visual.c: (gst_visual_clear_actors),
	(gst_visual_chain), (gst_visual_change_state):
	  Libvisual plugin was not passing audio data to libvisual 0.4.0 
	  correctly. Fixes #357800

66 67 68 69 70 71
2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
	  Add timeout to _get_state() so we see which pipeline it is
	  that causes trouble on the gen64 build bot.

72 73 74 75 76 77 78 79
2006-09-27  Wim Taymans  <wim@fluendo.com>

	* 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.

Wim Taymans's avatar
Wim Taymans committed
80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96
2006-09-27  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudioclock.c:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	Added docs for the audio libs.

97 98 99 100 101
2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Temporarily disable test that fails on the bots for unknown reasons.

102 103 104 105 106 107 108
2006-09-26 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 

	* 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

109 110 111 112 113 114 115 116 117
2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
	(add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
	(is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
	(new_pad):
	Cleanups and small leak fixes.
	Added Depayloaders to valid list of autopluggable elements.

118 119 120 121 122 123 124 125 126 127 128 129 130 131 132
2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_vis_blocked), (gst_play_bin_set_property),
	(gen_video_element), (gen_text_element), (gen_audio_element),
	(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
	(gst_play_bin_set_clock_func), (gst_play_bin_change_state):
	Detect NO_PREROLL state change returns and disable clock distribution to
	the sinks so that sync is disabled.
	Avoid some type checking and do simple casts instead.
	Small cleanups, fix some FIXMEs.
	Be more robust when linking user specified elements, catch an report
	errors. Fixes #357404.
	Fix some leaks in the error paths.

133 134 135 136 137
2006-09-25  Stefan Kost  <ensonic@users.sf.net>

	* ChangeLog:
	  ChangeLog surgery for missing bug-number

138 139 140 141 142 143 144
2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/playback/test.c:
	  Fix compilation with uClibc and -Werror (#357591).

145 146 147 148 149 150 151 152
2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

153 154 155 156 157 158 159 160
2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

161 162 163 164 165 166 167 168
2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  See if this makes the build bots happy.

	* tests/check/libs/cddabasesrc.c:
	  UTF8-ise my name.

169 170 171 172 173 174 175 176 177
2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha <ganadist at chollian dot net>

	* gst/subparse/samiparse.c: (handle_start_font),
	(fix_invalid_entities):
	  More case-insensitivity for certain tags; recognise entities with
	  decimal codes as special entities as well (#357330).

178 179 180 181 182
2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/Makefile.am:
	  Need to build tag directory before cdda.

183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204
2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

205 206 207 208 209 210
2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (gen_preroll_element):
	Improve buffering a bit by avoiding a deadlock because we cannot assume
	the underrun is always called.

211 212 213 214 215 216 217 218 219 220
2006-09-23  Wim Taymans  <wim@fluendo.com>

	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.

221 222 223 224 225 226 227 228
2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c:
	  Fix misleading docs addition.

	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Get rid of compiler warning the right way.

229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250
2006-09-22  Wim Taymans  <wim@fluendo.com>

	* 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.

251 252 253 254 255
2006-09-22  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/xoverlay.c:
	  Remove unused statement from doc example.

256 257 258 259 260 261 262 263 264 265 266 267 268 269 270
2006-09-21  Stefan Kost  <ensonic@users.sf.net>

	* 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)

271 272 273 274 275 276 277 278 279 280 281 282
2006-09-21  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
	(create_rgb_conversions), (rgb_conversion_free),
	(right_shift_colour), (fix_expected_colour), (check_rgb_buf),
	(got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
	  Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
	  but disable for now since it doesn't pass (something wrong with
	  RGBA somewhere).

283 284 285 286 287 288 289 290 291 292 293 294
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (group_commit),
	(queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
	(queue_out_of_data), (gen_preroll_element),
	(preroll_remove_overrun), (probe_triggered):
	Refactor handling of overrun detection.
	Separate handling of group completion and deadlock detection when doing
	network buffering. This should fix some deadlocks that were not detected
	because the group was completed.
	Add more comments, improve debugging.

295 296 297 298 299 300
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/gdpdepay.c: (GST_START_TEST):
	* tests/check/libs/audio.c:
	Some more compilation fixes.

301 302 303 304 305 306 307
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	Early morning compilation fix.

Wim Taymans's avatar
Wim Taymans committed
308 309 310 311 312 313 314 315 316
2006-09-20  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/gdpdepay.c: (GST_START_TEST):
	* tests/check/elements/multifdsink.c: (GST_START_TEST):
	* tests/check/elements/videorate.c: (GST_START_TEST):
	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (eos_buffer_probe):
	Fix some warnings.

317 318 319 320 321 322 323 324
2006-09-20  Stefan Kost  <ensonic@users.sf.net>

	* 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):
	  Handcrafted merge to help CVS understanding what I changed and what
	  not.

325 326 327 328 329 330
2006-09-20  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_get_times):
	  change colorkey behaviour back according to #354773 comment 6/7

331 332 333 334 335 336 337 338 339 340 341 342 343
2006-09-19  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
	(gst_multi_fd_sink_recover_client),
	(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
	(gst_multi_fd_sink_get_property):
	* gst/tcp/gstmultifdsink.h:
	  Implement stubbed out properties unit-type, units-soft-max,
	  units-max, to allow specifying maximum sizes in units other than
	  buffers.
	  Fixes #355935

344 345 346 347 348 349 350 351 352 353 354 355 356
2006-09-19  Wim Taymans  <wim@fluendo.com>

	* 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.

357
2006-09-18  Stefan Kost  <ensonic@users.sf.net>
358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/videoorientation.c:
	(gst_video_orientation_get_type),
	(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):
	* gst-libs/gst/interfaces/videoorientation.h:
374
	  API: Add new interface to control video orientation (fixes #354908)
375

376 377 378 379 380 381 382 383 384
2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* gst/videotestsrc/gstvideotestsrc.c:
	  Use G_UNLIKELY in _create and log one more detail.
	  
	(gst_video_test_src_get_times), (gst_video_test_src_create):
	* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
	  Use gst_util_uint64_scale_int in _get_times().

385 386 387 388 389
2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
	  Give better warning message (add object and detail).

390 391 392 393 394 395 396
2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_get_times):
	  xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
	  #354773), use gst_util_uint64_scale_int in _get_times()

397 398 399 400 401 402
2006-09-18  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
	  Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
	  always true, leading to dropping all timestamps.

403 404 405 406
2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* ext/libvisual/visual.c: (gst_vis_src_negotiate),
	(gst_visual_chain), (gst_visual_change_state):
407
	  update to work also with libvisual 0.4 API, fix double unref (#355914)
408 409 410 411 412 413 414 415 416
	  
	* tools/gst-launch-ext.1.in:
	* tools/gst-visualise.1.in:
	  remove references to old man-pages

	* tests/examples/seek/seek.c: (main):
	  add real meadi-buttons, add tool-tips for the seek-options, arrange
	  seek options in a table

417 418 419 420 421 422 423 424
2006-09-18  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
	(gst_ogg_mux_push_buffer):
	  Don't generate out-of-order timestamps from oggmux, instead clamp
	  output timestamps to be >= the previously output ts.
	  Fixes #355595

425 426 427 428 429 430 431
2006-09-18  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_multi_fd_sink_class_init):
	  Updates, fixes, and typo corrections for multifdsink. No functional
	  changes.

432 433 434 435 436 437
2006-09-17  Michael Smith  <msmith@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
	  Don't crash on truncated files - check that we got an 8 byte buffer
	  before trying to memcmp it.

438 439 440 441 442 443 444 445 446
2006-09-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (get_active_source):
	  Make stream-switching appear instant to the application
	  (ie. make sure that a g_object_get on 'current-foo' returns
	  the stream previously set with g_object_set(). Totem needs
	  this to update stream-related meta-info (like audio-codec)
	  correctly when switching streams.

447 448 449 450 451 452 453 454
2006-09-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
	(gst_alsa_mixer_ensure_track_list):
	  Try harder to guess which mixer track is the master mixer
	  track (instead of just taking the first one that has a pvolume).
	  Fixes #342228.

455 456 457 458 459 460
2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	(gst_audio_convert_transform_caps):
          Get structure-name just once.

461 462 463 464 465 466 467 468 469 470 471 472
2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audioresample.c: (GST_START_TEST):
	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	* tests/check/elements/volume.c: (GST_START_TEST):
	* tests/check/elements/vorbisdec.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
	(test_pipeline), (GST_START_TEST):
	* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
          Fix big batch of compiler warnings.

473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491
2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* 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

492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513
2006-09-15  Wim Taymans  <wim@fluendo.com>

	* 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.

514 515 516 517 518 519 520 521 522 523 524 525 526 527 528
2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (fill_buffer), (check_queue),
	(queue_threshold_reached), (gst_play_base_bin_set_property),
	(gst_play_base_bin_get_property):
	* gst/playback/gstplaybasebin.h:
	Don't use a 0 low watermark when buffering, it is catching starvation
	way too late. Instead, use a 3 second queue with 30 and 95
	percent low/high watermarks. 
	Added queue-min-threshold property to configure low watermark.
	Use new _buffering message API.
	Make queue_threshold variable big enough to store a uint64 time value.
	API: playbin::queue-min-threshold property.

529 530 531 532 533 534 535 536 537
2006-09-15  Wim Taymans  <wim@fluendo.com>

	* 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().

538 539 540 541 542 543
2006-09-15  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
	* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
	Use DEBUG_OBJECT more.

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
544 545
=== release 0.10.10 ===

546 547 548 549 550 551 552 553 554 555
2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Michael Smith <msmith at fluendo dot com>

	* gst/tcp/gstmultifdsink.c: (is_sync_frame),
	(gst_multi_fd_sink_client_queue_buffer),
	(gst_multi_fd_sink_new_client):
	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(multifdsink_suite):
	  Fix implementation of sync-method 'next-keyframe'
556
	  Closes #354594
557

558 559 560 561 562 563 564 565
2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Wim Taymans <wim at fluendo dot com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	This patch removes the RANDOM flag that was incorrectly introduced with
	revision 1.91.  Fixes #354590

566 567 568 569 570 571
2006-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Random variation in Makefile line to see if it makes the
	  gen64-base-full bot any happier.

572 573 574 575 576
2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c: (oggmux_suite):
	  Disable test that fails at the moment (killed after timeout).

577 578 579 580 581 582 583 584 585 586 587 588 589 590
2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: James Livingston  <doclivingston at gmail.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/.cvsignore:
	* tests/check/pipelines/oggmux.c: (get_page_codec),
	(check_chain_final_state), (fail_if_audio), (validate_ogg_page),
	(eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
	(test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
	(test_theora_vorbis), (oggmux_suite):
	  Add simple unit test for oggmux from #337026 with checking for the
	  EOS flags disabled for the time being.

591 592 593 594 595 596 597
2006-09-04  Wim Taymans  <wim@fluendo.com>

	patch by: Alessandro Dessina <alessandro nnva org>

	* ext/ogg/gstoggmux.c:
	Add cmml caps to oggmux. Fixes #353912

598 599 600 601 602 603 604
2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Returning a return value often helps. In this case, we
	  don't need the return value anyway, so just get rid of it.
	  Should make build bots much happier.

605 606 607 608 609 610 611 612 613 614 615 616 617 618
2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
	(paint_get_structure), (gst_video_test_src_get_size),
	(gst_video_test_src_smpte), (gst_video_test_src_snow),
	(gst_video_test_src_unicolor), (paint_setup_AYUV),
	(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
	(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
	* gst/videotestsrc/videotestsrc.h:
	  Add support for AYUV and the various RGBA formats. Initialise
	  fields of paintinfo structs allocated on the stack.

	* tests/check/elements/videotestsrc.c: (right_shift_colour),
	(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
619
	(check_rgb_buf), (videotestsrc_suite):
620 621
	  Add unit tests for videotestsrc's RGB output.

622 623 624 625 626 627 628 629 630 631 632 633 634
2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_pattern_get_type),
	(gst_video_test_src_set_pattern):
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
	(gst_video_test_src_black), (gst_video_test_src_white),
	(gst_video_test_src_red), (gst_video_test_src_green),
	(gst_video_test_src_blue):
	* gst/videotestsrc/videotestsrc.h:
	  Add more uni-colour patterns ("white", "red", "green", and "blue").

635 636 637 638 639
2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
	  Fix stride for YVYU, should be word-aligned (#353658).

640 641 642 643 644
2006-08-31  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/adder/gstadder.c: (gst_adder_src_event):
	  Fix build.

645 646 647 648 649 650 651 652 653 654
2006-08-31  Edward Hervey  <edward@fluendo.com>

	* gst/adder/gstadder.c: (forward_event_func),
	(gst_adder_src_event), (gst_adder_collected),
	(gst_adder_change_state):
	* gst/adder/gstadder.h:
	Remember the start position asked in the incoming seeks, so we can
	output GST_EVENT_NEW_SEGMENT with a correct position value (instead
	of assuming it will always be 0).

655 656 657 658 659 660 661
2006-08-31  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
	(gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_loop):
	Send the GST_EVENT_NEW_SEGMENT from the streaming thread.

662 663 664 665 666 667 668 669
2006-08-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	  Return FALSE instead of returning a random false unit
	  size when the format isn't known/supported (even if
	  this shouldn't happen under normal circumstances).

670 671 672 673 674 675 676 677 678 679 680
2006-08-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
	(gst_gnome_vfs_src_start):
	Try harder to get the size from a uri by using _info_uri() when
	_info_from_handle() does not give us enough info. 
	Also follow symlinks when getting the size.
	Partially Fixes #332864.

681 682 683 684 685 686 687 688 689 690 691 692 693 694
2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Viktor Peters  <viktor dot peters at gmail dot com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
	(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
	(gst_alsa_mixer_set_record):
	* ext/alsa/gstalsamixertrack.c:
	(gst_alsa_mixer_track_update_alsa_capabilities),
	(alsa_track_has_cap), (gst_alsa_mixer_track_new),
	(gst_alsa_mixer_track_update):
	* ext/alsa/gstalsamixertrack.h:
	  Improve and fix mixer track handling, in particular better handling
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
695 696 697
	  of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create
	  separate track objects for tracks that have both capture and playback
	  volume (and label them differently as well so they're not mistakenly
698 699 700 701 702 703 704 705 706
	  assumed to be duplicates); classify mixer tracks that only affect
	  the audible volume of something (rather than the capture volume)
	  as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
	  for capture tracks to correspond to alsa-pswitch alsa-cswitch
	  (following the meaning documented in the mixer interface header
	  file); add support for alsa's exclusive cswitch groups; update/sync
	  state/flags better if mixer settings are changed by another
	  application. Fixes #336075.

707 708 709 710 711
2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: add section about BUFFERING messages sent by playbin.

712 713 714 715 716 717 718 719 720
2006-08-29  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain):
	  Ignore explicit DISCONT marked on buffers (which is often spurious,
	  particularly when using multiple segments), in favour of solely
	  using the timestamps/durations.

721 722 723 724 725 726 727 728 729 730
2006-08-29  Edward Hervey  <edward@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	Don't rely on incoming buffers offset anymore, since it is completely
	broken when using multiple segments.
	Instead convert the incoming buffers timestamp to running time, and
	then convert that value to the offsets.
	Also inform GstSegment of the last outputted stop position, which is
	needed if we received several segments with an unknown stop value.

731 732 733 734 735
2006-08-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  fix buffer unreffing on a header push failure

736 737 738 739 740 741 742
2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
	(gst_audio_rate_chain):
	Make the metadata of the buffer writable before changing its
	flags.

743 744 745 746 747 748 749 750 751 752 753
2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	(gst_audio_rate_setcaps), (gst_audio_rate_init),
	(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
	(gst_audio_rate_chain), (gst_audio_rate_change_state):
	Fix audiorate some more.
	Reset and resync counters on flush and READY.
	Handle the DISCONT flag correctly.
	Use GstSegment to track position.
	Fail when not negotiated.
754
	Fixes #353234.
755

756 757 758 759 760 761
2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Fix spelling.
	  Remove accidently included debug line.

762 763 764 765 766 767 768 769
2006-08-25  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Small cleanups.
	If a buffer is received with no caps, make the buffer metadata
	writable and set the caps, making sure that we don't screw up the
	refcounts.

770 771 772 773 774 775 776 777 778 779 780 781 782
2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
	(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
	  Fix memory leaks and misleading debug messages, add a couple of
	  comments.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
	(gst_multi_fd_sink_render):
	  Do not use gst_buffer_make_writable() in a basesink render method,
	  as it may incorrectly unref the buffer. Instead, use convoluted
	  dance to avoid copying the buffer except when we need to.

783 784 785 786 787 788 789 790 791 792
2006-08-25  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c:
	(gst_vorbis_enc_buffer_check_discontinuous):
	  Allow very small discontinuities in the timestamps. These we can't
	  do anything useful with anyway (because vorbis's timestamps have
	  only sample granularity), and are commonly produced by elements with
	  minor bugs. Allow up to 1/2 a sample out.
	  Fixes #351742.

793 794 795 796 797 798 799
2006-08-24  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
	(play_scrub_toggle_cb), (main):
	Add a checkbox to enable play scrubbing. Makes it possible to disable
	normal scrubbing.

800 801 802 803 804
2006-08-23  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/.cvsignore:
	  make buildbot happy

805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820 821 822
2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
	(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
	(gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
	(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
	(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	(gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
	(gst_ogm_text_parse_strip_trailing_zeroes),
	(gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
	(gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
	  Refactor ogm parse, do better input checking, misc. clean-ups.
	  Cache incoming events and push them once the source pad has
	  been created. Don't pass unterminated strings to sscanf().
	  Strip trailing zeroes from subtitle text output, since they
	  are not valid UTF-8. Don't push vorbiscomment packets on
	  the subtitle text pad. Output perfect streams if possible.

823 824 825 826 827 828
2006-08-23  Wim Taymans  <wim@fluendo.com>

	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	Waits for tasks to settle down so that we clean up correctly for 
	valgrind.

829 830 831 832 833 834
2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
	  Unit test fixes: \377 is more likely to fit into 8 bits than \777;
	  actually return return value in taglists_are_equal.

835 836 837 838 839 840 841
2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Fix crash due to broken bitstream parsing on x86-64: can't make
	  any assumptions about sizeof(struct) due to alignment/packing
	  differences on different architectures. Fixes #351790.

842 843 844 845 846 847 848 849 850 851 852
2006-08-22  Wim Taymans  <wim@fluendo.com>

	* 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.

853 854 855 856 857 858
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

859 860 861 862 863 864 865 866 867
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

868 869 870 871 872 873 874 875
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

876 877 878 879 880 881 882 883 884 885 886
2006-08-21  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gdp.xml:
	* gst/gdp/Makefile.am:
	* tests/check/Makefile.am:
	  Move GDP plugin to -base from -bad.  Closes #347783.

887 888 889 890 891 892
2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* 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
893
	  vorbiscomment buffer if the ID to check for is larger than the buffer.
894 895 896 897

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Some more tests for gst_tag_list_from_vorbiscomment_buffer().

898 899 900 901 902 903 904
2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
	(gst_vorbis_enc_set_metadata):
	  Use vorbis comment utility functions from libgsttag
	  instead of re-inventing the wheel (partially fixes #347091).

905 906 907 908 909 910
2006-08-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix leaks. Wait for state transitions that might happen ASYNC, as well
	as some that won't.

911 912 913 914 915 916 917 918 919 920 921 922
2006-08-21  Wim Taymans  <wim@fluendo.com>

	* 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.

923 924 925 926 927 928
2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Add testcase for caps-size-explosion

929 930 931 932
2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_get_unit_size), (set_structure_widths):
933
	  Lower debug, use g_assert in _get_unit_size
934 935 936 937 938 939 940 941

	* 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

Wim Taymans's avatar
Wim Taymans committed
942 943 944 945 946 947 948 949 950 951 952
2006-08-18  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
	(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
	(gst_rtp_buffer_get_payload_buffer):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Document GstRTPBuffer.
	Added function to efficiently strip payload headers.
	API: gst_rtp_buffer_get_payload_subbuffer()

953 954 955 956 957
2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
	(gst_tag_to_vorbis_comments):
	  Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
958
	  tags and deserialise them properly as well (#347091).
959 960 961 962 963 964
	  Add some more gtk-doc blurbs and also some g_return_if_fail().

	* tests/check/libs/tag.c: (GST_START_TEST),
	(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
	  More tests.

965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980 981 982 983
2006-08-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogg.c: (plugin_init):
	* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
	(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
	(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
	(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
	(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
	(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
	Added ogg-in-avi parser element. Fixes #140139.

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
	Fixed a bug in oggdemux debug code.

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Recognise Ogg in the AVI extensible wave format.

984 985 986 987 988 989 990 991 992 993 994 995 996 997 998
2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  Make buffer durations add up (duration should be next_ts-ts for
	  perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
	  from CVS.

	* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
	(test_buffer_timestamps), (cddabasesrc_suite):
	  Add unit test for the above.

	* tests/check/Makefile.am:
	  Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
	  to see what happens.

999 1000 1001 1002 1003 1004 1005 1006 1007
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
	(gst_alsasink_open):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
	(gst_alsasrc_open):
	Avoid setting and using a NULL device name.
	Print more info when we fail to open a device.

1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
	  API: add gst_tag_parse_extended_comment() (#351426).

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
	  Add unit test for gst_tag_parse_extended_comment().

1020 1021 1022 1023 1024 1025
2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
	  Fix leak (#351502).

Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057 1058 1059 1060
2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* gst/playback/gstplaybin.c:
	  Document playbin.
	  
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	  Update to CVS version.

1061 1062 1063 1064 1065 1066 1067
2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_set_property), (gst_play_bin_get_property),
	(value_list_append_structure_list),
	(gst_play_bin_handle_redirect_message),
	(gst_play_bin_handle_message):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1068
	  API: GstPlayBin::connection-speed
1069 1070 1071 1072 1073
	  Add "connection-speed" property; re-order redirect messages with
	  multiple redirect locations depending on the minimum bitrate if
	  that information is available and a connection speed is set
	  (#350399).

1074 1075 1076 1077 1078
2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Update max volume to the same value that the volume element uses.

1079 1080 1081 1082 1083
2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
	Less uglyness..

1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095
2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	(gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
	Add some more debug info.
	Don't crash when a seek failed.
	Actually return the result of the seek instead of TRUE.
	Ignore multiple BOS pages with the same serial so that we don't create
	the same stream multiple times.
	Post an error when we fail to do the initial seek.

1096 1097 1098 1099 1100 1101 1102 1103 1104 1105 1106 1107 1108 1109 1110 1111 1112 1113 1114
2006-08-13  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
	(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
	Small code cleanup.

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
	(gst_alsa_mixer_new):
	Remove hack that always set the device to hw:0*.
	Properly find the card name for whatever device was configured.
	Do some better debugging.
	Fixes #350784.

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_set_property),
	(gst_alsa_mixer_element_change_state):
	Cleanups.
	Handle setting of a NULL device name better.

1115 1116 1117 1118 1119
2006-08-11  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c:
	Don't clip float values. Fixes #350900.

1120 1121
2006-08-11  Andy Wingo  <wingo@pobox.com>

1122 1123
	* gst/tcp/gsttcp.c: Really fix the build?

1124 1125 1126
	* gst/tcp/gsttcp.h: For now, always disable deprecation here --
	fixes the build.

1127 1128 1129 1130 1131
2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
	  Float caps shouldn't have a "signed" field.

1132 1133 1134 1135 1136 1137
2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
	  Implement SEEKING query in its most basic form, so that we can
	  at least check if we're seekable or not (#350655).

1138 1139 1140 1141 1142 1143 1144 1145 1146
2006-08-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  The checks here are not even close to anything that would
	  justify MAXIMUM probability, lowering to POSSIBLE until someone
	  fixes the checks (case at hand: quicktime redirection files
	  might start with 00 00 01 XX and pass the checks here just
	  fine, see #350399).

1147 1148 1149 1150 1151 1152 1153 1154 1155
2006-08-08  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
	  Better detection for multipart/x-mixed-replace: accept leading
	  whitespaces before the boundary marker as well (as our very own
	  multipartmux used to produce) (#349068).

1156 1157 1158 1159 1160 1161 1162 1163 1164
2006-08-07  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha  <ganadist at chollian 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):
	  Detect DTS audio streams (#350157).

1165 1166 1167 1168 1169 1170
2006-08-05  Andy Wingo  <wingo@pobox.com>

	* ext/theora/gsttheoraparse.h:
	* ext/theora/theoraparse.c (gst_theora_parse_class_init)
	(theora_parse_dispose, theora_parse_set_property)
	(theora_parse_get_property, theora_parse_munge_granulepos)
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1171 1172 1173
	(theora_parse_push_buffer, theora_parse_change_state):
	API: GstTheoraParse::synchronization-points
	Add a property 'synchronization-points' to fix badly synchronized oggs.
1174

1175 1176 1177 1178 1179 1180 1181 1182 1183
2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/audio.c: (structure_contains_channel_positions),
	(fixed_caps_have_channel_positions), (GST_START_TEST),
	(audio_suite), (main):
	  Add a few tests for the channel position stuff in libgstaudio.

1184 1185 1186 1187 1188 1189 1190 1191 1192 1193 1194 1195 1196 1197 1198 1199
2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
	(gst_alsa_detect_channels):
	* ext/alsa/gstalsasink.c:
	  Add support for cards that (only) do more than 8 channels,
	  like the Delta 44 (#345188).

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions):
	* gst-libs/gst/audio/multichannel.h:
	  API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
	  unspecified channel position and cannot be combined with any
	  of the other audio channel positions; adjust position layout
	  checks accordingly (#345188).

1200 1201 1202 1203 1204
2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Recognise ancient RealAudio files (see #349779).

1205 1206 1207 1208 1209 1210 1211
2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jens Granseuer  <jensgr at gmx net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Add typefinder for Interplay's MVE format (#348973).

1212 1213 1214 1215 1216 1217 1218 1219 1220 1221
2006-08-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Marcel Moreaux <marcelm at luon dot net>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_add_to_queue):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Handle RTP sequence number rollover.
	Disable jitterbuffer by default.

1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238
2006-07-28  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audioresample/gstaudioresample.c: (audioresample_stop),
	(audioresample_set_caps):
	Don't leak references to the incoming caps. Clean them up when
	stopping.

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	(gst_video_scale_finalize):
	Don't leak our temporary pixel buffer.

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
	(GST_START_TEST), (simple_launch_lines_suite):

	Fix leaks and re-enable the test for valgrind checking.

1239 1240 1241 1242 1243 1244 1245 1246
2006-07-28  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
	(plugin_init):
	  Add typefind function for multipart/x-mixed-replace (#348916).

1247 1248 1249 1250 1251 1252 1253
2006-07-28  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c: (gst_adder_setcaps),
	(gst_adder_query_duration):
	Fix leak in duration query.
	Reflow some docs and notes.

1254 1255 1256 1257 1258 1259 1260
2006-07-28  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
	(vorbisenc_suite):
	  Enable Andy's extra vorbisenc test, now that it passes. Also fix one
	  aspect of it.

1261 1262 1263 1264 1265 1266 1267 1268 1269 1270 1271 1272
2006-07-28  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
	(gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
	(gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	* ext/vorbis/vorbisenc.h:
	  Handle discontinuities in the input vorbis stream correctly,
	  so that the output is properly timestamped (and has good granulepos
	  values). Needs some oggmux fixes too.

1273 1274 1275 1276 1277 1278 1279 1280 1281 1282 1283
2006-07-27  Wim Taymans  <wim@fluendo.com>

	patch by: Kai Vehmanen <kv2004 eca cx>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_change_state):
	Don't send multiple newsegments with different formats.
	Fixes #348677.

1284 1285 1286 1287 1288 1289 1290
2006-07-26  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
	Make seeking in ogg more accurate again by doing the more correct
	granuletime to stream time conversion.

1291 1292 1293 1294 1295 1296 1297 1298
2006-07-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_new_client):
	  debug a little more understandably
	  do not use goto as a substitute for break, especially if
	  break is also being used

1299 1300 1301 1302 1303 1304
2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
	* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Remove GLib-2.6 compatibility cruft.

1305 1306 1307 1308 1309 1310
2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Don't try to align a sample to an unknown value.

1311 1312 1313 1314 1315 1316 1317
2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
	When the audio clock is slaved to another clock, never try to align
	samples but trust the rate interpolation algorithm.

1318 1319 1320 1321 1322 1323 1324 1325 1326 1327 1328 1329 1330 1331
2006-07-24  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	Don't try to calculate silence samples, base class does this much
	better now.

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_acquire):
	Calculate silence samples correctly.

	* gst-libs/gst/audio/gstringbuffer.h:
	Add _CAST macro.

1332 1333 1334 1335 1336 1337 1338
2006-07-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
	  Limit search for the first markup tag to the first few kB of
	  the file. If we don't find one there, it's highly unlikely that
	  this is an XML(-ish) file.

1339 1340
2006-07-21  Andy Wingo  <wingo@pobox.com>

1341 1342 1343
	* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
	test to the one in vorbisenc. Also commented out.

1344
	* tests/check/pipelines/vorbisenc.c: 
1345 1346 1347
	(test_discontinuity): New test, commented out until Mike lands
	some elite vorbisenc patches.

1348 1349 1350 1351 1352 1353 1354
	* tests/check/pipelines/theoraenc.c: Port to bufferstraw.
	Bufferstraw was actually factored out of these tests. Now we share
	code yay.

	* configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
	for bufferstraw addition to gstcheck.

1355 1356 1357 1358 1359
2006-07-21  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (clip_buffer):
	Better clipping.

1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370
2006-07-21  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
	(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
	(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
	Fix leak.
	Avoid type casting when we can.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
	Fix mem leak.

1371 1372 1373 1374 1375 1376 1377
2006-07-20  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_change_state):
	  Make state change fail if the specified device can't be opened
	  for some reason.

1378 1379 1380 1381 1382 1383
2006-07-20  Wim Taymans  <wim@fluendo.com>

	* gst/playback/test.c: (gen_video_element), (gen_audio_element),
	(cb_newpad), (main):
	Example of a small audio/video player using decodebin.

1384 1385 1386
2006-07-20  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1387
	  Add 'fact' chunk id
1388

1389 1390 1391 1392 1393 1394 1395 1396 1397 1398 1399 1400 1401 1402
2006-07-19  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_change_state):
	Don't assert when not negotiated but post a meaningfull 
	error message. Fixes #347918.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	Add comment about better default MTU size.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	Small cleanups, start docs.

1403 1404 1405 1406 1407 1408 1409 1410 1411
2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Martin Szulecki

	* sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
	  If "device-name" is requested and the device is not
	  open, try to temporarily open it to obtain this
	  information (#342494).

1412 1413 1414 1415 1416 1417 1418 1419 1420
2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstid3tag.c:
	  Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).

	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Some more random const-ifications.

1421 1422 1423 1424 1425
2006-07-18  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c:
	(gst_riff_create_video_template_caps):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1426 1427 1428
	  Add more FOURCCs (sort list to make stuff easier to find),
	  add comment what those 16 bytes in struct _gst_riff_strh according to
	  one avi-dumper are
1429

1430 1431 1432 1433 1434 1435 1436
2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions),
	(gst_audio_fixate_channel_positions):
	  Const-ify two arrays.

1437 1438 1439 1440 1441 1442
2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
	  Fix typo, so that alsasink also advertises 8 channels
	  if that's supported (tags: can, worms, open, alsa, ph34r).

1443 1444 1445 1446 1447 1448 1449 1450
2006-07-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
	*sigh*, when is the compiler going to warn when the comments
	are out-of-sync with the code.. Refix case of busted theora
	headers with 0 granule pos.

1451 1452 1453 1454 1455 1456 1457 1458 1459 1460
2006-07-14  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_wait),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	Fix 99% cpu load by waiting for absolute times on the
	clock. Fixes #347300.

1461 1462
2006-07-14  Andy Wingo  <wingo@pobox.com>

1463 1464 1465 1466 1467 1468 1469
	* ext/theora/gsttheoraparse.h: 
	* ext/theora/theoraparse.c (theora_parse_drain_event_queue)
	(theora_parse_push_headers, theora_parse_clear_queue)
	(theora_parse_drain_queue_prematurely, )
	(theora_parse_sink_event, theora_parse_change_state): Queue events
	until we initialized our state, like in vorbisparse.

1470 1471 1472 1473 1474 1475 1476 1477 1478 1479
	* ext/vorbis/vorbisparse.h: 
	* ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
	(vorbis_parse_push_headers, vorbis_parse_clear_queue)
	(vorbis_parse_drain_queue_prematurely, )
	(vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
	until we have initialized our state. Fixes seeking after an
	initial pad block.

2006-07-14  Andy Wingo  <wingo@pobox.com>

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1480
	Patch by: Iain Holmes <iaingnome@gmail.com>
1481 1482 1483
	
	* ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.

1484 1485 1486 1487 1488
2006-07-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump nano back to CVS

1489 1490 1491 1492 1493 1494 1495
=== release 0.10.9 ===

2006-07-13  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.9, "I walk the line"

1496 1497 1498 1499 1500 1501 1502
2006-07-14  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (stop_pipeline):
	  Move a g_cond_signal to earlier to avoid sometimes deadlocking
	  (commonly happens when running this test under valgrind) when trying
	  to remove the buffer probe.

1503 1504 1505 1506 1507
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
	Fix missing g_unlock from the previous commit

1508 1509 1510 1511 1512 1513 1514 1515 1516
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_change_state):
	Implement a locking order to ensure we always take the object lock
	before the x_lock and never vice-versa.

1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527 1528 1529 1530 1531 1532
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (find_compatibles):
	Fix a caps leak when linking (#347304)

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
	(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
	Don't leak shared memory resources. Use the object lock to protect
	against the xcontext disappearing while returning a buffer from the
	pipeline. (#347304)

1533 1534 1535 1536 1537 1538 1539 1540
2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
	(vorbis_handle_comment_packet):
	gst_tag_list_merge() returns a new object. Take that into account when
	using it. This avoids memleak.
	Revert previous commit which is not needed.

1541 1542 1543 1544 1545
2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
	Reset the decoder in finalize so that all fields get cleared.

1546 1547 1548 1549 1550 1551 1552 1553 1554
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_set_clock),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
	Don't try to post an error message when setting the clock fails
	as this can happen when adding an element to a bin which will then
	deadlock. Fixes #347296.

1555 1556 1557 1558 1559 1560 1561 1562
2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet):
	Post tag messages on the bus even if we're not initialized.
	If we're not initialized, we still postpone the event pushing of tags.

1563 1564 1565 1566 1567 1568 1569
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Revert last two changes that broke the freeze.

1570 1571 1572 1573 1574
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	basesink calculates silence sample correctly for us.

1575 1576 1577 1578 1579 1580 1581
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Calculate correct silence samples so we don't fill our ringbuffer
	with noise.

1582 1583 1584 1585 1586 1587 1588 1589 1590 1591
2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(gst_vorbis_dec_reset), (vorbis_dec_sink_event),
	(vorbis_handle_comment_packet), (vorbis_handle_type_packet):
	* ext/vorbis/vorbisdec.h:
	Delay sending events (newsegment, tags) until the decoder is properly
	initialized.
	Fixes #347295

1592 1593 1594 1595 1596 1597 1598
2006-07-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audioconvert.c: (get_float_mc_caps),
	(get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
	  Patch from #347221 adding a test for audioconvert
	  channel remappings.

1599 1600 1601 1602 1603 1604 1605 1606 1607 1608
2006-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
	(gst_ssa_parse_parse_line):
	  Don't include the terminating NUL in the buffer size,
	  it's only there for extra paranoia (would add random
	  '*' characters at the end of each subtitle since the
	  terminator itself is not valid UTF-8 technically).
	  Also fix indenting after boilerplate macro.

1609 1610 1611 1612 1613 1614 1615 1616 1617
2006-07-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (close_pad_link):
	  Also emit 'unknown-type' signal (which should really be
	  called unhandled-type) if we found potential decoders/demuxers
	  in the registry but none of them worked in the end (as in the
	  case where the plugins don't exist any longer but are still
	  listed in the registry). Fixes #329798.

1618 1619 1620 1621 1622 1623 1624
2006-07-08  Andy Wingo  <wingo@pobox.com>

	* theoraparse.c (theora_parse_push_buffer)
	(theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
	Add some more debugging. Fix granulepos reconstruction in the face
	of discontinuities.

1625 1626 1627 1628 1629 1630 1631 1632 1633 1634 1635 1636 1637 1638 1639 1640 1641 1642 1643 1644 1645
2006-07-06  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init),
	(gst_base_audio_sink_provide_clock):
	Use gobject_class instead of G_OBJECT_CLASS (klass)

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_init),
	(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
	(gst_base_audio_src_get_time),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
	(gst_base_audio_src_create_ringbuffer):
	Fix latency and buffer-time constants and properties ala basesink.
	Implement pull based scheduling. Fixes #346527.
	Set default blocksize in GstBaseSrc to 0, we default to pushing out
	one segment.
	Refuse slaving to another clock instead of silently not working.
	Only provide a clock when we are actually able to do so.
	Various small cleanups and compiler hints.

1646 1647 1648 1649 1650 1651 1652 1653
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Lutz Mueller <lutz at topfrose de>

	* gst/typefind/gsttypefindfunctions.c: (html_type_find),
	(plugin_init):
	  Add typefinding for text/html (#346581).

1654 1655 1656 1657 1658 1659 1660
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	(xml_check_first_element), (xml_type_find), (smil_type_find):
	  Fix SMIL typefinding, make xml_check_first_element() more
	  useful.

1661 1662 1663 1664 1665 1666 1667 1668 1669 1670 1671 1672
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	(gst_play_base_bin_finalize), (decodebin_element_added_cb),
	(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
	* gst/playback/gstplaybasebin.h:
	  Protect list of elements with a subtitle-encoding property and
	  the subtitle encoding member itself with a lock of their own
	  instead of using the object lock. This prevents a dead-lock in
	  the element-remove callback in some circumstances when shutting
	  down playbin.

1673 1674 1675 1676 1677 1678 1679
2006-07-05  Sebastien Moutte  <sebastien@moutte.net>

	* win32/common/libgsttag.def:
	Export some new functions.
	* win32/vs6/libgstogg.dsp:
	Add a link to libgsttag-0.10.lib.

1680 1681 1682 1683 1684
2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Some const-ification.

1685 1686 1687 1688 1689 1690
2006-07-04  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
	Improve checking if we are dealing with a stream. Added some
	more uris that need buffering.

1691 1692 1693 1694 1695
2006-07-03  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_do_clip):
	Remove unused variable.

1696 1697 1698 1699 1700 1701 1702
2006-07-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	  include lcov.mak
	* configure.ac:
	  add GCOV_LIBS to GST_LIBS

1703 1704 1705 1706 1707 1708 1709 1710
2006-07-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Michael Sheldon  <webmaster at mikeasoft com>

	* ext/alsa/gstalsasrc.c:
	  Add 32 bps to template caps and increase channels range
	  from [1,2] to [1,MAX]. See #346326.

1711 1712 1713 1714 1715
2006-06-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Recognise 'WMVA' video codec fourcc (#345879).
	  
1716 1717 1718 1719 1720
2006-06-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	 
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c: 	 
	  Fixed nasty memory leak

1721 1722 1723 1724 1725 1726
2006-06-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
	(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
	  fix logging

1727 1728 1729 1730 1731 1732 1733 1734 1735 1736 1737
2006-06-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
	(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
	Protect remove_fakesink using a mutex, so that we don't try and
	remove the fakesink simultaneously from multiple threads.

	When going from READY to PAUSED, restore the fakesink, so that
	it is there when decodebin gets reused.

1738 1739