ChangeLog 514 KB
Newer Older
1 2 3 4 5 6
2007-10-19  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* tests/check/elements/avimux.c:
	  Add some debug and sync tests with the fix.	  

7 8 9 10 11 12 13 14 15
2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	When the socket is used by the app for other purposes, don't generate an
	error if there is activaty on the socket that is not data related.
	Fixes #487488.

16 17 18 19 20 21 22
2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	(gst_v4l2src_grab_frame):
	Add some more debug info. Generate an error when we run out of buffers
	for some reason. See #480557.

23 24 25 26 27 28 29
2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Anders Skargren <anders dot skargren at axis dot com>

	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Set marker bit correctly.

30 31 32 33 34
2007-10-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	Use allowed name for the GstStructure.

35 36 37 38 39 40
2007-10-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	* gst/autodetect/gstautoaudiosink.c:
	  Use new gst_bus_pop_filtered().

41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58
2007-10-13  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  When probing the formats and sizes a camera supports, make
	  sure the best ones (highest resolution, prefered format)
	  end up at the beginning of the probed caps and the less
	  desirable ones at the end.  This is important because the
	  order within the caps matters for things like fixation and
	  negotiation, ie. what format is chosen in the end.
	  With recent kernels, the current probing code will end up
	  querying the supported sizes from lowest resolution to
	  highest resolution, adding them to the probed caps in that
	  order, resulting to v4l2src fixating to the lowest possible
	  resolution if downstream does not express a size preference.
	  Also make up a somewhat random ranking of prefered output
	  formats for the same reason. Fixes #485828.
	
59 60 61 62 63 64 65 66 67 68 69 70 71
2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jason Kivlighn  <jkivlighn gmail com>

	* gst/id3demux/id3v2frames.c:
	  Extract license/copyright URIs from ID3v2 WCOP frames
	  (Fixes #447000).

	* tests/check/elements/id3demux.c:
	* tests/files/Makefile.am:
	* tests/files/id3-447000-wcop.tag:
	  Add simple unit test.

72 73 74 75 76 77
2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Add support for license/copyright URI tags (ID3v2 WCOP frame).
	  Prerequisite for #447000.

78 79 80 81 82 83
2007-10-08  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
	Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
	a GstClockTime.

84 85 86 87 88 89 90 91 92 93 94 95
2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	More seeking fixes, mostly passing around the new playback segment in
	order to configure it properly.
	Also reset base_time of udp sources when setting them back to PLAYING as
	a temporary hack until core supports seek in live sources properly.

96 97 98 99 100
2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpmp4adepay.c:
	Fix caps as to not confuse autopluggers.

101 102 103 104 105 106 107 108 109 110 111
2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/gstid3demux.h:
	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	  Port ID3 tag demuxer over to the new GstTagDemux in -base
	  (now would be a good time to test re-importing your music
	  collection).

112 113 114 115 116 117 118 119 120
2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/Makefile.am:
	* gst/apetag/gstapedemux.c:
	* gst/apetag/gstapedemux.h:
	* gst/apetag/gsttagdemux.c:
	* gst/apetag/gsttagdemux.h:
	  Port APE tag demuxer over to the new GstTagDemux in -base.

121 122 123 124 125 126 127 128 129 130 131 132 133
2007-10-05  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_internal_src_query),
	(gst_rtspsrc_handle_src_query), (new_session_pad),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_loop_send_cmd):
	Improve flushing behaviour.
	Set state of the udp sources to PAUSE/PLAYING correctly.
	Handle events and queries for UDP and TCP transport now.

Stefan Kost's avatar
Stefan Kost committed
134 135 136 137 138 139
2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	  Add log category.

140 141 142 143 144 145 146 147
2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Timo Hotti <Timo.Hotti@sysopendigia.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c:
	  Add unit tests for payloaders/depayloaders.

148 149 150 151 152 153
2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  Also save codec data for audio streams. Fixes #482495.

154 155 156 157 158 159
2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  Fix "Index entry has invalid stream nr 1".
	  Add support for muxing aac - work in progress (see #482495).

160 161 162 163 164 165 166 167 168
2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
	(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	* gst/rtsp/gstrtspsrc.h:
	Parse bandwidth modifiers, they are not yet configured in the session
	manager because we don't have an API for that yet.

169 170 171 172 173 174 175
2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	Use shiny new function in -base to get the default clock-rate.
	Update some docs.

176 177 178 179 180 181 182 183 184 185
2007-09-29  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add files to win32 manifest.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.

186 187 188 189 190 191 192 193 194
2007-09-28  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	In TCP mode, only timestamp the first buffer. TCP is not real time and
	it does not make sense to try to skew compensate, also some servers send
	the first batch of data in a burst.

195 196 197 198 199 200 201 202
2007-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c:
	  Fix setting the discont flag on the first buffer
	  pushed downstream for formats with private codec
	  data that needs to be deserialised into buffers
	  (such as vorbis and FLAC when in a matroska container).

203 204 205 206 207 208 209 210 211 212 213
2007-09-27  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Antoine Tremblay <hexa00 at gmail dot com>

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
	(gst_rtp_mp4v_pay_handle_buffer):
	* gst/rtp/gstrtpmp4vpay.h:
	Free the config string. Fixes #480707.
	Clean up the timestamp code a little.

214 215 216 217 218 219 220 221 222 223
2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	Set timestamps on RTP buffers in interleaved mode.
	Mark first buffers with a DISCONT.
	Remove flush hack now that sync for live sources has been figured out.

224 225 226 227 228
2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Update documentation.

229 230 231 232 233 234 235
2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
	(gst_rtp_xqt_depay_change_state):
	* gst/qtdemux/gstrtpxqtdepay.h:
	Fail if we don't know the quicktime format.

236 237 238 239 240 241 242 243
2007-09-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  Save the flow return from the last gst_pad_push() and
	  make sure we pass the right flow return value upstream
	  in the case of failure; minor clean-ups.

244 245 246 247 248 249 250
2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* gst/apetag/gstapedemux.c:
	  Add support for the new GST_TAG_COMPOSER (#459809).

251 252 253 254 255 256 257 258 259 260 261 262
2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/alaw.c:
	* gst/law/mulaw-conversion.h:
	  Compulsive clean-ups: use boilerplate macros, add debug
	  categories, fix up things to conform to symbol nomenklatura,
	  etc.

263 264 265 266 267 268 269 270 271 272 273
2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	  Use static tables for A-Law decoding and encoding; this makes
	  A-Law decoding and encoding less CPU-intensive, but increases
	  the binary size a bit. Leaving old code around for now,
	  selectable by a define in the code. Fixes #435435.

274 275 276 277 278 279 280
2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
	AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros
	in configure.ac.

281 282 283 284 285 286 287
2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: <j at bootlab dot org>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add fourccs for MPEG2 HDV streams. Fixes #479960.

288 289 290 291 292 293
2007-09-23  Stefan Kost  <ensonic@users.sf.net>

	* sys/oss/gstosshelper.c:
	  Use GST_WARNING instead of a g_critical. This situation is not caused
	  by the application.

294 295 296 297 298 299
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/LINGUAS:
	* po/nl.po:
	  Updated translations.

300 301 302 303 304 305 306
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mikel Olasagasti <hey_neken@mundurat.net>

	* po/eu.po:
	  Added Basque translation.

307 308 309 310 311 312 313 314
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Abel Cheung <abelcheung@gmail.com>

	* po/zh_HK.po:
	* po/zh_TW.po:
	  Added Chinese (traditional and Hong Kong) translation.

315 316 317 318 319 320 321
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jakub Bogusz <qboosh@pld-linux.org>

	* po/pl.po:
	  Added Polish translation.

322 323 324 325 326 327 328
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Ilkka Tuohela <hile@iki.fi>

	* po/fi.po:
	  Added Finnish translation.

329 330 331 332 333 334 335
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jorge González González <aloriel@gmail.com>

	* po/es.po:
	  Added Spanish translation.

336 337 338 339 340 341 342
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mogens Jaeger <mogens@jaeger.tf>

	* po/da.po:
	  Added Danish translation.

343 344 345 346 347 348 349
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Funda Wang <fundawang@linux.net.cn>

	* po/zh_CN.po:
	  Added Chinese (simplified) translation.

350 351 352 353 354 355 356
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Alexander Shopov <ash@contact.bg>

	* po/bg.po:
	  Added Bulgarian translation.

357 358 359 360 361
2007-09-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
	Set outgoing packet duration because we can. Fixes #478244 some more.

362 363 364 365 366 367 368 369 370
2007-09-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/cairo/gsttextoverlay.c:
	  Add info about static leak.
	
	* tests/check/Makefile.am:
	* tests/check/generic/states.c:
	  Improved state change unit test.

Stefan Kost's avatar
Stefan Kost committed
371 372 373 374 375 376
2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/.cvsignore:
	* tests/check/.cvsignore:
	  Ignore registries in any format.

377 378 379 380 381 382 383 384 385 386 387 388 389 390 391
2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
	Removed some unused code.

	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
	* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
	(gst_rtp_theora_pay_flush_packet):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
	Try to preserve the incomming buffer duration on the outgoing
	packets. Fixes #478244.

392 393 394 395 396 397 398
2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	  Work around compiler warnings with g++-4.2 when assigning a
	  string constant to a gchar * (partially fixes #478092).

399 400 401 402 403
2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We require core CVS now for gst_base_src_set_do_timestamp().

404 405 406 407 408 409 410 411 412
2007-09-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message):
	Fix compiler warnings shown with Forte.

413 414 415 416 417 418 419
2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_dup_printf):
	Give meaningfull error when all streams failed to configure for some
	reason.

420 421 422 423 424 425
2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/README:
	Update README with the design for synchronisation rules of RTP on
	sender and receiver.

426 427 428 429 430 431 432 433 434
2007-09-14  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
	(gst_wavparse_chain):
	Don't push EOS from the chain function, the element
	driving the pipeline is responsible for this. The bug
	this was meant to fix seems to be queue not forwarding
	EOS in all cases (see #476514).

435 436 437 438 439 440 441 442 443 444
2007-09-13  Wim Taymans  <wim.taymans@gmail.com>

	* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	Use basetransform segment so that it is correctly managed on flushes and
	start/stop.
	Report message timestamp as stream time, which is what an application
	can understand.

Sebastian Dröge's avatar
Sebastian Dröge committed
445 446 447 448 449 450 451 452
2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gsttaglibmux.c:
	* tests/check/elements/apev2mux.c:
	Update my mail address.

453 454 455 456 457 458
2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
	(gst_wavparse_loop), (gst_wavparse_chain):
	Add EOS logic for the push-based mode too. Fixes #476514.

459 460 461 462 463 464 465 466 467
2007-09-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
	* gst/law/alaw-encode.h:
	* gst/law/mulaw-encode.c: (gst_mulawenc_init),
	(gst_mulawenc_chain):
	* gst/law/mulaw-encode.h:
	Fix law encoder timestamps.

468 469 470 471 472 473 474 475
2007-09-12  Stefan Kost  <ensonic@users.sf.net>

	* ext/gconf/gstgconfaudiosink.c:
	  Fix warning when building without debug.

	* sys/oss/gstossmixertrack.c:
	  Use const like in alsamixertrack.c (fixes warnings).

476 477 478 479 480 481 482 483 484 485 486
2007-09-11  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Fix framerate detection code some more.
	Handle the case where there is a weird step in the stepwise framerates.
	Don't overwrite the min interval with the framerate, use a temp variable
	instead.
	Use max in the Continuous framerate intervals instead of step, which is
	1 according to the docs. Fixes #475424.

487 488 489 490 491 492
2007-09-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
	Make udpsrc timestamp outgoing buffers based on when they were received.
	Also make it output a segment in time.

493 494 495 496 497
2007-09-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Plug a little leak. Little code cleanups.

498 499 500 501 502 503
2007-09-09  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
	  flac versions, 's good for cross-compilation karma.

504 505 506 507 508 509 510 511 512 513 514
2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>

	* gst/rtp/gstrtph263pay.c:
	  Fix up header structure so that compilers don't add padding
	  between the structure fields, since that would lead to us
	  sending RTP packets with broken headers (as is currently the
	  case when compiling with MSVC). Also see similar fixes in
	  libgstrtp in gst-plugins-base. (#474616; #471194)

515 516 517 518 519 520 521
2007-09-07  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Don't overwrite our GValue with 0 but instead use the previously
	computed value. Fixes #471823 some more.

522 523 524 525 526 527
2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  No tabs in this file please, or gtk-doc will end up documenting
	  rather absurd class hierarchies.

528 529 530 531 532 533 534 535 536
2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	  If the new kid element fails to change state for some reason
	  (e.g. esdsink not being able to connect to the sound server),
	  forward the error message it posted on the bus instead of just
	  posting a generic 'Internal state change error: please file a
	  bug' error message. Fixes #471364.

537 538 539 540 541 542
2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  Don't assume tags are encoded as UTF-8 (#473670).

543 544 545 546 547 548 549 550
2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	  Implement LATENCY queries in the crudest way possible so I don't
	  have to use sync=false any longer when testing with videosinks.

Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
551 552 553 554 555
2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix build.

556 557 558 559 560 561 562
2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Add some more debugging in the framerate function.
	Iterate stepwise framerate up to and _including_ the max and if nothing
	was added to the list, add a dummy 0/1 to 100/1 framerate so that we
563
	don't end up with an empty list. Fixes #471823
564

565 566 567 568 569 570 571 572 573 574 575 576
2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_set_clients_string),
	(gst_multiudpsink_get_clients_string),
	(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
	(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
	(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
	(gst_multiudpsink_clear):
	Add property do configure destination address/port pairs
	API:GstMultiUDPSink::clients

577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592
2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/Makefile.am:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sdp:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H264-PCMA.sdp:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	Added some RTP example scripts for sending and receiving RTP streams.

593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614
2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
	(gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
	Restructure the setcaps function so that we can also compute the
	expected GStreamer output size of the video frames.
	Set frame_byte_size correctly so that read-based devices have a chance
	of working correctly.
	When grabbing a frame, discard frames that are not of the expected size.
	Some cameras don't output the right framesize for the first buffer.
	Try only a couple of times to get a valid frame, else error out.

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	(gst_v4l2_fill_lists), (gst_v4l2_get_input):
	Add some more debug info when scanning the device.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
	(gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
	(gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
	(gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
	Add some more debug info when dequeing a frame.

615 616 617 618 619
2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  More code cleanups. Add some more comment and improve debugs logs.

620 621 622 623 624 625 626 627
2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  Implement seek-query. Refactor duration calculations. Appropriate use
	  of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
	  out of loops.

628 629 630 631 632
2007-09-03  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Implement seek-query.

633 634 635 636 637 638 639
2007-08-29  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_dup_printf):
	Use new basesink async property to make sparse RTCP packet not wait for
	preroll.

640 641 642 643 644
2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audiofx/Makefile.am:
	Dist the right file.

645 646 647 648 649 650 651
2007-08-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
	(gst_rtspsrc_get_float), (gst_rtspsrc_play):
	Make sure we generate and parse floating point values in the POSIX
	locale instead of the current locale. 

652 653 654 655 656 657 658 659 660 661 662
2007-08-22  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Fix method detection again.
	Keep track of when we must send a Range header.
	Use segment values for Range, Speed and Scale headers.
	Parse Speed and Scale headers to update the segment values.

663 664 665 666 667 668 669
2007-08-22  Stefan Kost  <ensonic@users.sf.net>

	patch by: Mark Nauwelaerts <manauw@skynet.be>

	* sys/v4l2/v4l2src_calls.c:
	  Handle optional v4l2 ioctls gracefully.

670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691
2007-08-20  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
	(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
	(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
	(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
	(gst_rtp_h263_depay_get_property),
	(gst_rtp_h263_depay_change_state),
	(gst_rtp_h263_depay_plugin_init):
	* gst/rtp/gstrtph263depay.h:
	Added an H263 depayloader. Fixes #369392.

	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
	Make the H263+ pay/depayloader support H263-1998 and H263-2000
	payloads.
	Also alow plain H263 on the h263p payloaders. Fixes #465040.

692 693 694 695 696 697
2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	Add small comparision with the windowed sinc filters in the docs.

698 699 700 701 702 703 704 705
2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
	(audiochebyshevfreqband_suite):
	* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
	(audiochebyshevfreqlimit_suite):
	Also test 32 bit float mode and the type 2 variants of the filters.

706 707 708 709 710 711 712
2007-08-18  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop):
	Refactor the udp and interleaved loop function a bit.

713 714 715 716 717 718 719 720 721 722 723 724
2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
	(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect connection activity with a new lock, avoids deadlocks when going
	to PAUSED. Fixes #455808.

725 726 727 728 729
2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
	Fix debug statement.

730 731 732 733 734
2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
	Fix stray %u in debug line as spotted by Saur on IRC.

735 736 737 738 739 740 741 742 743 744 745 746 747 748
2007-08-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_class_init):
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_class_init):
	Use generator macros for the process functions for the different
	sample types, add lower upper boundaries for the GObject properties
	so automatically generated UIs can use sliders and add a note about
	the number of poles as a too high number of poles combined with
	very low or very high frequencies will produce only noise.
	* docs/plugins/gst-plugins-good-plugins.args:
	Regenerated for the property changes.

749 750 751 752 753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768 769
2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
	(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Improve timeout handling.
	Use the same socket for sending and receiving RTCP packets so that some
	servers can track clients better.
	Improve connection closed handling. Try to reconnect.
	Don't overwrite our content base with NULL.
	Improve debugging.
	Improve range parsing and handling.
	Remove flushing hack now that core does the right thing.

770 771 772 773 774 775 776 777 778 779 780 781 782
2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_init), (gst_multiudpsink_set_property),
	(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
	(gst_multiudpsink_close), (gst_multiudpsink_add):
	* gst/udp/gstmultiudpsink.h:
	Add support for getting and setting the socket to use.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_get_property):
	Add support for getting the currently used socket.

783 784 785 786 787 788 789 790 791 792 793 794 795 796 797 798 799 800 801 802 803 804 805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820 821 822 823 824 825 826 827 828 829 830 831 832 833 834 835 836 837 838 839 840 841 842 843 844 845
2007-08-16  Sebastian Dröge  <slomo@circular-chaos.org>

	reviewed by: Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_mode_get_type),
	(gst_audio_chebyshev_freq_band_base_init),
	(gst_audio_chebyshev_freq_band_dispose),
	(gst_audio_chebyshev_freq_band_class_init),
	(gst_audio_chebyshev_freq_band_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_band_set_property),
	(gst_audio_chebyshev_freq_band_get_property),
	(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
	(gst_audio_chebyshev_freq_band_start):
	* gst/audiofx/audiochebyshevfreqband.h:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_mode_get_type),
	(gst_audio_chebyshev_freq_limit_base_init),
	(gst_audio_chebyshev_freq_limit_dispose),
	(gst_audio_chebyshev_freq_limit_class_init),
	(gst_audio_chebyshev_freq_limit_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_limit_set_property),
	(gst_audio_chebyshev_freq_limit_get_property),
	(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
	(gst_audio_chebyshev_freq_limit_start):
	* gst/audiofx/audiochebyshevfreqlimit.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
	Fixes #464800.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/audiochebyshevfreqband.c:
	(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
	(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
	* tests/check/elements/audiochebyshevfreqlimit.c:
	(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
	(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
	Add unit tests for the chebyshev filters.

	* 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-1394.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	And add docs for the chebyshev filters. While doing
	that also run make update in docs/plugins.

Stefan Kost's avatar
Stefan Kost committed
846 847 848 849 850 851
2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/gstcmmltag.c:
	* gst/rtp/gstrtpvorbispay.c:
	  Make ro memory to share.

852 853 854 855 856 857
2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Improve UDP performance by avoiding a select() when we have data
	available immediatly.

858 859 860 861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876 877 878 879
2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
	(gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Add (dummy) SSRC management signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
	(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
	(on_timeout), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add connection-speed property.
	Add find_stream helper functions.
	Handle stream EOS based on BYE messages or SSRC timeout.
	Returns SUCCESS from the state change function as we hide our async
	elements from the parent.

880 881 882 883 884
2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/rndbuffersize.c:
	  Fix da leak.

885 886 887 888 889 890 891 892 893 894 895
2007-08-14  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/Makefile.am:
	* gst/debug/breakmydata.c:
	* gst/debug/gstdebug.c:
	* gst/debug/negotiation.c:
	* gst/debug/progressreport.c:
	* gst/debug/rndbuffersize.c:
	* gst/debug/testplugin.c:
	  Add new test element and clean-up the others a little.

896 897 898 899 900
2007-08-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of mp4a version 0 atoms. Fixes #465774.

901 902 903 904 905
2007-08-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpilbcdepay.c:
	  Include stdlib.

906 907 908 909 910 911
2007-08-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpmpvdepay.c:
	Set the mpegversion in the caps so that autoplugging does not get
	confused.

912 913 914 915 916 917 918
2007-08-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/uk.po:
	* po/vi.po:
	  Updated translations.

919 920 921 922 923
2007-08-08  Michael Smith <msmith@fluendo.com>

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Render right border in the correct location.

924 925 926 927 928 929 930 931
2007-08-08  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Olivier Crete <tester at tester dot ca>

	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Make mode property a string. Fixes #464475.

932 933 934 935 936
2007-08-05  Stefan Kost  <ensonic@users.sf.net>

	* ext/flac/gstflacenc.c:
	  Widen caps to match decoder a bit and add more FIXMEs.

937 938 939 940 941 942 943
2007-08-05  Stefan Kost  <ensonic@users.sf.net>

	patch by: Mark Nauwelaerts <manauw@skynet.be>

	* gst/avi/gstavimux.c:
	  Fix ODML index tag numbering. Fixes #463624.

944 945 946 947 948 949 950 951 952 953
2007-08-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_udp_sink):
	Fix default clock-rate for realmedia.
	Fix parsing of transport.
	Don't try to link NULL pads.

954 955 956 957 958 959 960 961
2007-07-30  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.skip:
	  Add POTFILES.skip with list of source files that aren't disted at the
	  moment but contain translatable strings. Should hopefully pacify
	  broken tools and make it clearer that these files are left out
	  intentionally (#461600).

962 963 964 965 966
2007-07-30  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
	If the buffer was entirely clipped ... don't try sending it :)

967 968 969 970 971 972 973 974 975 976 977 978
2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports):
	If we don't hav a session manager, set the caps on outgoing buffers
	ourselves.
	Force PAUSE/PLAY methods for now until the extensions can overwrite.
	Append final bit of the transport string even when it does not contain a
	placeholder.

979 980 981 982 983 984 985 986 987 988 989 990 991
2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
	(gst_rtsp_ext_list_connect):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
	Clean up the interface list.
	Allow connecting to interface signals for the extensions.
	Remove old extension code.
	Free list on cleanup.
	Allow extensions to send additional RTSP messages.

992 993 994 995 996
2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Handle a NULL gconf key gracefully by rendering the default element.

997 998 999 1000 1001
2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.h:
	Fix include path for extension interface.

1002 1003 1004 1005 1006
2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.h:
	Also remove a now unecessary variable here.

1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020
2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
	(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
	(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
	(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Don't save format information ourselves, this is already saved in
	GstAudioFilter.

1021 1022 1023 1024 1025 1026 1027 1028 1029
2007-07-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Use rank to filter out extensions.
	Add url to stream_select interface call.

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 1061 1062 1063 1064 1065 1066 1067 1068 1069 1070 1071 1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090
2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/base64.c:
	* gst/rtsp/base64.h:
	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
	(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
	(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
	(gst_rtsp_ext_list_setup_media),
	(gst_rtsp_ext_list_configure_stream),
	(gst_rtsp_ext_list_get_transports),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
	(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
	(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspconnection.c:
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c:
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtspmessage.c:
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsprange.c:
	* gst/rtsp/rtsprange.h:
	* gst/rtsp/rtsptransport.c:
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/rtspurl.c:
	* gst/rtsp/rtspurl.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.c:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/test.c:
	Use shiny new RTSP and SDP library.
	Implement RTSP extensions using the new interface.
	Remove a lot of old code.

1091 1092 1093 1094 1095
2007-07-24  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.

1096 1097 1098 1099 1100 1101
2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	Don't unref the outgoing buffer twice when dropping it because it's
	outside of the segment.

1102 1103 1104 1105 1106 1107 1108 1109 1110 1111
2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
	Use the new buffer clipping function from gstaudio here and
	require gst-plugins-base CVS.
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	For framed Wavpack buffers we require a valid timestamp.

1112 1113 1114 1115 1116 1117 1118 1119 1120 1121
2007-07-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
	(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
	Clip raw audio and video when we can, keep track of current output
	segment.
	Don't leak buffers and events when there is no output pad.
	Improve debugging here and there.

1122 1123 1124 1125 1126
2007-07-23  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Sync liboil check with plugins-base.

1127 1128 1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140
2007-07-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/Makefile.am:
	  Fix CFLAGS/LIBS.

	* ext/cdio/gstcdiocddasrc.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  Include stdlib

	* ext/cairo/Makefile.am:
	* gst/videofilter/Makefile.am:
	* tests/examples/level/Makefile.am:
	  Use $(LIBM) instead of -lm

1141 1142 1143 1144 1145
2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	  Add another example pipeline.

1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157 1158 1159 1160 1161
2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Alexander Eichner <alexeichi@yahoo.de>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Use define here.

	* sys/v4l2/gstv4l2tuner.c:
	(gst_v4l2_tuner_set_frequency_and_notify):
	  Don't touch the property - its still disabled.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
	(gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
	* sys/v4l2/v4l2src_calls.h:
	  Improve fallback format negotionation. Fixes #451388

1162 1163 1164 1165 1166
2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/videocrop.c: (GST_START_TEST):
	  Fix the test.

Stefan Kost's avatar
Stefan Kost committed
1167 1168 1169 1170 1171 1172 1173 1174 1175 1176 1177 1178
2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task),
	(gst_pngdec_sink_setcaps):
	  More docs. More logs in pngdec.

1179 1180 1181 1182 1183 1184 1185 1186 1187
2007-07-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Initialize num_buffers with minimum value.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame):
	  Handle frame-size query failure gracefully.

1188 1189 1190 1191 1192 1193 1194
2007-07-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of esds atoms inside mp4a atoms so that we can set correct
	codec_info for AAC audio. Fixes #457097 along with a whole other bunch
	of qt/aac files.

1195 1196 1197 1198 1199 1200
2007-07-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c:
	(gst_wavpack_dec_clip_outgoing_buffer):
	Fix buffer clipping to correctly clip to the segment stop.

1201 1202 1203 1204 1205 1206 1207 1208 1209
2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* tests/Makefile.am:
	Remove bogus check for libcheck, since we check for
	gstreamer-check and it pulls in the required info from there,
	and we weren't actually _using_ the information for libcheck
	ourselves anyway.

1210 1211 1212 1213 1214
2007-07-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Use pkg-config to locate check.

1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236
2007-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	* gst/debug/gstnavigationtest.c: (gst_navigationtest_transform):
	* gst/effectv/gstaging.c: (gst_agingtv_transform):
	* gst/effectv/gstdice.c: (gst_dicetv_transform):
	* gst/effectv/gstedge.c: (gst_edgetv_transform):
	* gst/effectv/gstquark.c: (gst_quarktv_transform):
	* gst/effectv/gstrev.c: (gst_revtv_transform):
	* gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform):
	* gst/effectv/gstvertigo.c: (gst_vertigotv_transform):
	* gst/effectv/gstwarp.c: (gst_warptv_transform):
	* gst/matroska/matroska-demux.c:
	(gst_matroska_demux_add_wvpk_header),
	(gst_matroska_demux_check_subtitle_buffer),
	(gst_matroska_decode_buffer):
	* gst/videofilter/gstvideoflip.c: (gst_video_flip_transform):
	  Fix build against core CVS.

1237 1238 1239 1240 1241 1242 1243
2007-07-10  Edward Hervey  <bilboed@gmail.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
	don't have enough granularity to convert that boolean into a
	GstFlowReturn.

1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254 1255
2007-07-06  Michael Smith <msmith@fluendo.com>

	* gst/law/alaw-decode.c: (alawdec_sink_setcaps),
	(gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain),
	(gst_alawdec_change_state):
	* gst/law/alaw-decode.h:
	* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	(gst_mulawdec_class_init), (gst_mulawdec_init),
	(gst_mulawdec_chain), (gst_mulawdec_change_state):
	* gst/law/mulaw-decode.h:
	  Fix capsnego bogosity in *law decoders. 

1256 1257 1258 1259 1260 1261 1262 1263 1264
2007-07-06  Michael Smith <msmith@fluendo.com>

	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
	(gst_smokeenc_setcaps), (gst_smokeenc_chain),
	(gst_smokeenc_change_state):
	* ext/jpeg/gstsmokeenc.h:
	  Remove stupidity in get/set caps functions.
	  Fix some refcounting problems.

1265 1266 1267 1268 1269 1270 1271
2007-07-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
	Remove endianness-flipping hack that seems to have been required
	only because of a bug in ffmpegcolorspace.
	Partially Fixes: #451908

1272 1273 1274 1275 1276
2007-07-05  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

1277 1278 1279 1280 1281 1282 1283 1284 1285
2007-07-03  Wim Taymans,,,  <set EMAIL_ADDRESS environment variable>

	Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>

	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Set the encoding-name in the rtp caps to all uppercase, as required by
	the caps spec.
	Some small cleanups in the error paths. Fixes #453037.

1286 1287 1288 1289 1290 1291 1292 1293 1294 1295
2007-06-28  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackparse.c:
	(gst_wavpack_parse_index_get_last_entry),
	(gst_wavpack_parse_index_get_entry_from_sample),
	(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	(gst_wavpack_parse_scan_to_find_sample):
	* ext/wavpack/gstwavpackparse.h:
	Use a GSList for the GArray that is used like a list anyway.

1296 1297 1298 1299 1300 1301 1302 1303 1304 1305 1306 1307 1308
2007-06-28  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
	(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
	  Add state change function where we set 0/1 as default framerate in
	  case our setcaps function isn't called, like it might not in a
	  filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
	  gdkpixbufdec trying to create caps with a 0/0 framerate.
	  Also post an error message on the bus if gst_pad_push() fails when
	  called from our sink event handler (+1 for flow returns for event
	  functions in 0.11) instead of failing silently.

1309 1310 1311 1312 1313
2007-06-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps):
	Cast stack args to the proper types. Fixes #451249.

1314 1315 1316 1317 1318 1319 1320 1321
2007-06-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (gst_rtspsrc_setup_streams):
	* gst/rtsp/gstrtspsrc.h:
	For container formats we only need to activate one of the streams so
	that we correctly signal no-more-pads. Fixes #451015.

1322 1323 1324 1325 1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338 1339 1340 1341 1342 1343 1344 1345 1346 1347 1348 1349 1350 1351 1352 1353 1354 1355 1356 1357 1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370 1371 1372 1373 1374 1375 1376
2007-06-25  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update docs with caps info.

1377 1378 1379 1380 1381
2007-06-25  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Add more files with translatable strings (#450878).

1382 1383 1384 1385 1386
2007-06-22  Jan Schmidt  <thaytan@noraisin.net>

	* MAINTAINERS:
	Updating all the maintainers files

Edward Hervey's avatar
Edward Hervey committed
1387 1388 1389 1390 1391 1392 1393 1394 1395 1396 1397 1398
2007-06-22  Edward Hervey  <edward@fluendo.com>

	* ext/flac/gstflactag.c: (gst_flac_tag_init):
	* gst/interleave/deinterleave.c: (deinterleave_init),
	(deinterleave_sink_link):
	* gst/interleave/interleave.c: (interleave_init):
	* gst/median/gstmedian.c: (gst_median_init):
	* gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init):
	Fix memory leaks.
	* tests/check/elements/id3demux.c: (pad_added_cb):
	Remove unused variable.

1399 1400 1401 1402 1403 1404 1405 1406
2007-06-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.h:
	Make the prototype of gst_gconf_get_key_for_sink_profile
	match the implementation.
	Patch by: Damien Carbery <damien dot carbery at sun dot com>
	Fixes: #449747

1407 1408 1409 1410 1411
2007-06-20  Michael Smith <msmith@fluendo.com>

	* gst/rtp/gstrtpdepay.c:
	  Fix description - rtpdepay is not a payloader.

1412 1413 1414 1415 1416 1417 1418
2007-06-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
	(qtdemux_video_caps):
	* gst/qtdemux/qtdemux_fourcc.h:
	  Add MJPG to the variants of motion jpeg.

1419 1420 1421 1422 1423 1424 1425 1426 1427 1428 1429 1430
2007-06-19  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
	* tests/check/elements/videocrop.c: (GST_START_TEST):
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	* tests/check/elements/wavpackparse.c: (GST_START_TEST):
	  Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
	  error flags are included and it errors out on compiler warnings
	  for CVS builds; remove unused variables in various unit tests.

1431 1432 1433 1434 1435 1436 1437 1438 1439
2007-06-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close), (rtsp_connection_free):
	Use threadsafe inet_ntop to convert an ip number to a string. 
	Fixes #447961.
	Don't leak fd (and ip) when freeing a connection without first closing
	it.

Jan Schmidt's avatar
Jan Schmidt committed
1440 1441 1442 1443 1444 1445 1446 1447
2007-06-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.6 to the doap file.

Jan Schmidt's avatar
Jan Schmidt committed
1448 1449 1450 1451 1452 1453 1454
=== release 0.10.6 ===

2007-06-18  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.6, "Wobble Board"

1455 1456 1457 1458 1459 1460
2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  Revert previous commit again, since we are frozen (sorry).

1461 1462 1463 1464 1465 1466 1467 1468 1469
2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  inet_ntoa() uses a static buffer internally, so we need to copy the
	  returned string if we want to store it for later (#447961).

1470 1471 1472 1473 1474 1475 1476 1477 1478 1479 1480 1481 1482 1483 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493 1494 1495 1496 1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514
2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* win32/vs6/autogen.dsp:
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstalaw.dsp:
	* win32/vs6/libgstalpha.dsp:
	* win32/vs6/libgstalphacolor.dsp:
	* win32/vs6/libgstapetag.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstauparse.dsp:
	* win32/vs6/libgstautodetect.dsp:
	* win32/vs6/libgstavi.dsp:
	* win32/vs6/libgstcutter.dsp:
	* win32/vs6/libgstdirectdraw.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgsteffectv.dsp:
	* win32/vs6/libgstflx.dsp:
	* win32/vs6/libgstgoom.dsp:
	* win32/vs6/libgsticydemux.dsp:
	* win32/vs6/libgstid3demux.dsp:
	* win32/vs6/libgstinterleave.dsp:
	* win32/vs6/libgstjpeg.dsp:
	* win32/vs6/libgstlevel.dsp:
	* win32/vs6/libgstmatroska.dsp:
	* win32/vs6/libgstmedian.dsp:
	* win32/vs6/libgstmonoscope.dsp:
	* win32/vs6/libgstmulaw.dsp:
	* win32/vs6/libgstmultipart.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsmpte.dsp:
	* win32/vs6/libgstspeex.dsp:
	* win32/vs6/libgstudp.dsp:
	* win32/vs6/libgstvideobalance.dsp:
	* win32/vs6/libgstvideobox.dsp:
	* win32/vs6/libgstvideocrop.dsp:
	* win32/vs6/libgstvideoflip.dsp:
	* win32/vs6/libgstvideomixer.dsp:
	* win32/vs6/libgstwaveform.dsp:
	* win32/vs6/libgstwavenc.dsp:
	* win32/vs6/libgstwavparse.dsp:
	Mark *.dsp & *.dsw as binary files and convert to DOS line
	endings, as they don't load into VS6 correctly otherwise.

1515 1516 1517 1518 1519 1520 1521 1522
2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect):
	Fix the MingW build. 
	Patch By: Vincent Torri <vtorri at univ-evry dot fr>
	Fixes: #446981

Jan Schmidt's avatar
Jan Schmidt committed
1523 1524 1525 1526 1527 1528
2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/.cvsignore:
	* tests/icles/.cvsignore:
	Hush the buildbots up

1529 1530 1531 1532 1533 1534 1535 1536 1537
2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/directdraw/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/waveform/Makefile.am:
	Make sure to dist everything needed for win32 builds.

1538 1539 1540 1541 1542 1543 1544
2007-06-14  Edward Hervey  <edward@fluendo.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	For AMR-NB streams, export the AMRSpecificBox as codec_data on the
	caps.
	Fixes #447458

1545 1546 1547 1548 1549 1550
2007-06-13  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	Make sure we allocate enough memory for the codec_data.
	Fixes #447210.

1551 1552 1553 1554 1555 1556 1557 1558 1559 1560 1561
2007-06-12  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add videocrop project file to the win32 manifest.
	* win32/vs6/gst_plugins_good.dsw:
	Add qtdemux,videocrop and waveform projects to the workspace.
	* win32/vs6/libgstqtdemux.dsp:
	Add zlib to the link list of qtdemux.
	* win32/vs6/libgstvideocrop.dsp:
	Add a project file for videocrop.

1562 1563 1564 1565 1566
2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* po/POTFILES.in:
	Add qtdemux for translation

1567 1568 1569 1570 1571 1572 1573 1574 1575 1576 1577 1578 1579 1580
2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* gst-plugins-good.spec.in:
	* sys/Makefile.am:
	* tests/check/Makefile.am:
	* tests/icles/Makefile.am:
	* tests/icles/videocrop-test.c:
	Move videocrop and osxvideo from -bad.

Jan Schmidt's avatar
Jan Schmidt committed
1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591 1592 1593 1594 1595
2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* 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-qtdemux.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* win32/MANIFEST:
	Move qtdemux from -bad.

	* gst-plugins-good.spec.in:
	Update spec file to reflect moving of qtdemux and wavpack

Jan Schmidt's avatar
Jan Schmidt committed
1596 1597
2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>
	
1598
	* win32/MANIFEST:
Jan Schmidt's avatar
Jan Schmidt committed
1599 1600 1601 1602 1603 1604 1605 1606 1607
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-directdraw.xml:
	* docs/plugins/inspect/plugin-directsound.xml:
	* docs/plugins/inspect/plugin-waveform.xml:
	Move the waveform plugin from -bad too. Update the inspect xml
	files to mention Plugins Good instead of Plugins Bad.

1608 1609 1610 1611 1612 1613 1614 1615 1616 1617 1618 1619 1620 1621 1622 1623 1624 1625 1626
2007-06-12  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
	(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
	finalization and resuscitation. No longer public.
	(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
	(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
	(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
	(gst_v4l2_buffer_pool_destroy): Make the pool follow common
	miniobject semantics, and be threadsafe.
	(gst_v4l2src_queue_frame): Remove this function, as we just call
	the ioctls directly in the two places where we queue buffers.
	(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
	directly.
	(gst_v4l2src_capture_init): Use the new buffer_pool_new function
	to allocate the pool, which also preallocates the GstBuffers.
	(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
	queueing the frames directly.
1627 1628
	(gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
	mmap buffers have been dequeued.
1629 1630 1631 1632 1633 1634 1635 1636 1637 1638 1639 1640 1641 1642 1643 1644

	* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
	real MiniObject instead of rolling our own refcounting and
	finalizing. Give it a lock.
	(struct _GstV4l2Buffer): Remove one intermediary object, having
	the buffers hold the struct v4l2_buffer directly.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
	capture_init so that it can set them on the buffers that it will
	create.
	(gst_v4l2src_get_read): For better or for worse, include the
	timestamping and offsetting code here; really we should be using
	bufferalloc though.
	(gst_v4l2src_get_mmap): Just make grab_frame return one of our
	preallocated, mmap'd buffers.

1645 1646 1647 1648 1649 1650 1651 1652 1653
2007-06-11  Wim Taymans  <wim@fluendo.com>

	Patch by: daniel fischer <dan at f3c dot com>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
	(gst_ximage_src_get_caps):
	Actually use the display_name property so that we can dump any
	available X display. Fixes #445905.

1654 1655 1656 1657 1658 1659 1660 1661
2007-06-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>

	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
	Add missing rate fields to caps. Fixes #441118.

1662 1663 1664 1665 1666 1667 1668
2007-06-10  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs8/gst-plugins-good.sln:
	Add DirectSound and DirectDraw sinks project files to
	workspace and solution files.

1669 1670 1671 1672 1673 1674 1675 1676 1677 1678 1679 1680 1681 1682 1683 1684 1685 1686 1687 1688 1689 1690
2007-06-10  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Josh Coalson <xflac at yahoo dot com>,
	updated by Alexis Ballier <aballier at gentoo dot org>:

	* configure.ac:
	* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
	(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
	(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
	(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
	(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
	(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
	(gst_flac_enc_chain), (gst_flac_enc_set_property),
	(gst_flac_enc_get_property), (gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
	
1691 1692 1693 1694 1695 1696
2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
	Remove workaround for bug #421543. This is fixed in core 0.10.13 and
	not necessary anymore as we need at least that core version. 

1697 1698 1699 1700 1701 1702 1703 1704 1705 1706 1707
2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain):
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	(gst_wavpack_parse_push_buffer):
	* ext/wavpack/gstwavpackparse.h:
	Improve discont handling by checking if the next Wavpack block has
	the expected, following block index.

1708 1709 1710 1711 1712
2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
	  Fix element description.

1713 1714 1715 1716 1717 1718 1719 1720 1721 1722 1723 1724 1725 1726 1727 1728 1729 1730
2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* 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/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* ext/Makefile.am:
	* tests/check/Makefile.am:
	  move wavpack plugin.  See #352605.

1731 1732 1733 1734 1735 1736 1737 1738 1739 1740 1741
2007-06-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* 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:
	* sys/Makefile.am:
	* win32/MANIFEST:
	Add DirectDraw & DirectSound plugins to the build and docs.

1742 1743 1744 1745 1746 1747
2007-06-08  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
	* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
	  When operating in pull mode, error out correct on not-linked.

1748 1749 1750 1751 1752 1753 1754
2007-06-06  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
	(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
	format and size if the ioctls are defined; should fix compilation
	on Linux < 2.16.19.

1755 1756 1757 1758 1759 1760 1761 1762
2007-06-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Printf fixes in debug statements; use LOG level for debug statements
	  that are printed for each and every frame; convert c++ comments to
	  C-style comments; not much point using g_try_malloc() if we then not
	  even check the return value.

1763 1764 1765 1766 1767 1768 1769 1770 1771
2007-06-05  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Bump requirements to released versions (core and base 0.10.13).

	* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  own implementation.

1772 1773 1774 1775 1776 1777 1778 1779 1780 1781 1782 1783 1784 1785 1786
2007-06-05  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
	some useless comments.

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
	frames before calling STREAMON, that might leave them in a state
	where they can't be dequeued if we go back to NULL without calling
	STREAMON, according to the docs.
	(gst_v4l2src_capture_start): Enqueue buffers here instead, right
	before we call STREAMON.
	(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
	failures. (For me this code hung.) The pool refcounting is still
	crack; added a note to that effect.

1787 1788 1789 1790 1791 1792 1793
2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
	(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
	Add support for mapping gst structure names to the MIME type equivalent.
	Implemented for audio/x-mulaw->audio/basic. Fixes #442874.

1794 1795 1796 1797 1798 1799 1800 1801 1802
2007-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
	(gst_wavenc_chain), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Properly write wav files with width!=depth by having the depth most
	significant bytes set and all others zero. Fixes #442535.

1803 1804 1805 1806 1807
2007-06-01  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c:
	Add include to make buildbot happy.

1808 1809 1810 1811 1812 1813 1814 1815 1816 1817 1818 1819 1820 1821 1822 1823 1824 1825 1826 1827 1828 1829 1830 1831 1832