ChangeLog 1.15 MB
Newer Older
1 2 3 4 5 6 7 8
2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c: (cb_probe):
	  Ignore video-codec tag for audio streams and ignore audio-codec tags
	  for video streams. Should make codec name collection a bit more
	  robust against sloppy demuxers that send tag events containing both
	  tags down each pad.

9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (update_rates):
	Tweak the buffering thresholds a little.
	Update the buffer size with the previously calculate rate instead of
	only when we calculate a new rate so that we get smoother buffering
	updates.

	* gst/playback/Makefile.am:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
	(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
	(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
	(gst_uri_decode_bin_get_property), (unknown_type),
	(add_element_stream), (no_more_pads_full), (no_more_pads),
	(source_no_more_pads), (new_decoded_pad), (array_has_value),
	(gen_source_element), (has_all_raw_caps), (analyse_source),
	(remove_decoders), (make_decoder), (remove_source),
	(source_new_pad), (setup_source), (decoder_query_init),
	(decoder_query_duration_fold), (decoder_query_duration_done),
	(decoder_query_position_fold), (decoder_query_position_done),
	(decoder_query_latency_fold), (decoder_query_latency_done),
	(decoder_query_seeking_fold), (decoder_query_seeking_done),
	(decoder_query_generic_fold), (gst_uri_decode_bin_query),
	(gst_uri_decode_bin_change_state), (plugin_init):
	New element that intergrates a source, optional buffering element and
	decodebin.

36 37 38 39 40 41 42
2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Bump libtheora requirement to 1.0alpha5 for the pixformat check
	  (also has a .pc file, so we don't need the fallback check any
	  longer). Fixes #438840.

43 44 45 46 47 48 49 50 51 52 53 54
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
	(apply_segment), (apply_buffer), (update_buffering),
	(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_filled),
	(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
	(plugin_init):
	fix build.

55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/Makefile.am:
	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
	(gst_queue_getcaps), (gst_queue_bufferalloc),
	(gst_queue_acceptcaps), (update_time_level), (apply_segment),
	(apply_buffer), (update_buffering), (reset_rate_timer),
	(update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_empty),
	(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
	(gst_queue_loop), (gst_queue_handle_src_event),
	(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
	(gst_queue_src_activate_push), (gst_queue_change_state),
	(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
	On our way to playbin2 this is the new network queue that does buffering
	all by itself using high and low watermarks. It can also measure up and
	downstream bandwidth to optimally size the queue.

75 76 77 78 79 80 81
2007-05-17  Michael Smith <msmith@fluendo.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
	  Use the segment->last_stop value to calculate the next timestamp to
	  generate after a seek; not the segment->start value.

82 83 84 85 86
2007-05-15  David Schleef  <ds@schleef.org>

	* docs/Makefile.am: Install docs even when --disable-gtk-doc
	  is disabled.  This matches the behavior of gtk+.  Fixes #349099.

87 88 89 90 91 92
2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
	Some more chained streaming ogg timestamp fixes.

93 94 95 96 97 98 99 100 101
2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_handle_page):
	Add some FIXMEs.
	Fix chain start/stop segment handling based on patch by
	<ahalda at cs dot mcgill dot ca> see #320984.

102 103 104 105 106
2007-05-15  Michael Smith <msmith@fluendo.com>

	* configure.ac:
	  We don't require a C++ compiler. So don't require one.

107 108 109 110 111 112 113 114 115 116
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track):
	  Apply some of the cleanup Tim suggested in #152864 afterwards.

117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
	  _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
	  gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
	  gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
	  gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
	  gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
	* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
	* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
	  gst_alsa_mixer_element_interface_supported,
	  gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
	  gst_alsa_mixer_element_set_property,
	  gst_alsa_mixer_element_get_property,
	  gst_alsa_mixer_element_change_state):
	* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
	* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
	  gst_mixer_option_changed):
	* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
	  volume_changed, option_changed, _gst_reserved):
	  Implement notification for alsamixer. Fixes #152864

147 148 149 150 151 152
2007-05-14  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add support for video/x-raw-bayer.

153 154 155 156 157 158
2007-05-12  David Schleef  <ds@schleef.org>

	* sys/xvimage/xvimagesink.c:
	  Add some sanity checking for the XVImage size returned by X.
	  Related to #377400.

159 160 161 162 163 164 165 166
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Parse and use additional caps fields as described in updated
	application/x-rtp caps spec.

167 168 169 170 171 172 173
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_collect_chain_info):
	If there is a stream in a chain without any data packets, ignore the
	stream in the total length calculations. Might be related to #436820.

174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194
2007-05-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
	(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
	(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
	(mpeg_video_type_find), (mpeg_video_stream_type_find),
	(plugin_init):

	Consolidate and re-work our mpeg system stream detection to probe
	more packets and produce a higher confidence result. Fixes a
	regression caused by lowering the typefind probability last year
	- related to bug #397810. Remove the redundant MPEG-1 specific 
	typefind function, as the new one detects both MPEG-1 & MPEG-2
	happily.

	Also cleanup the MPEG elementary and MPEG-TS detection functions a
	little. 

	Tested against my media test directory, with some improvements and
	no regressions.

195 196 197 198 199 200 201
2007-05-10  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
	(queue_out_of_data):
	Connect to the new queue "pushing" signal instead of the broken
	"running" one.

202 203 204 205 206 207 208 209 210 211
2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	Move variable declaration before the first instruction.
	* gst/videotestsrc/videotestsrc.c:
	Define M_PI if it's not defined yet.
	* win32/common/libgstrtp.def:
	Add new exported functions.

212 213 214 215 216
2007-05-09  Michael Smith <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  gst_pad_push_event() does not return a GstFlowReturn!

217 218 219 220 221 222
2007-05-09  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/scrubby.c: (stop_cb), (main):
	* tests/examples/seek/seek.c: (do_seek):
	Some small cosmetic changes.

223 224 225 226 227 228 229 230
2007-05-08  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
	  gst_adder_change_state):
	* gst/adder/gstadder.h (bps, offset, collect_event, segment,
	  segment_pending, segment_position, segment_rate):
	  Handle playback-rate on adder.

231 232 233 234 235 236 237 238 239 240
2007-05-07  Michael Smith <msmith@fluendo.com>

	* ext/theora/gsttheoradec.h:
	* ext/theora/theoradec.c: (gst_theora_dec_reset),
	(theora_dec_sink_event), (theora_handle_comment_packet),
	(theora_handle_type_packet), (theora_dec_change_state):
	  Don't push events (newsegment, tags) before initialising the
	  decoder.
	  This is neccesary for seeking to work correctly in gnonlin.

241 242 243 244 245 246 247 248 249 250 251 252 253 254
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst/adder/gstadder.c:
	* gst/audiotestsrc/gstaudiotestsrc.c
	  (gst_audio_test_src_create_white_noise):
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
	  VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
	  volume_sink_template, volume_src_template, gst_volume_init,
	  volume_process_double, volume_process_int16,
	  volume_process_int16_clamp):
	  Doc fixes and formatting.

255 256 257 258 259 260 261
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  Minimal check for volume's GstController usability; also another
	  test for #422295.

262 263 264 265 266 267 268
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix it so that it (a) makes sense and (b) doesn't break
	  everything cdda-related including the unit test.

269 270 271 272 273 274
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix build when disabling asserts.

275 276 277 278 279 280 281 282 283 284 285 286 287
2007-05-03  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
	  When XShm is not available, we might get row strides that are not
	  rounded up to multiples of four; this is bad, because virtually
	  every RGB-processing element in GStreamer assumes rowstrides are
	  rounded up to multiples of four, so let's allocate at least enough
	  memory to avoid crashes in this case. The image will still be
	  displayed distorted though if this happens, so that still needs
	  fixing (maybe by allocating a bigger image with an 'even' width
	  and then clipping it appropriately when rendering - something for
	  Xlib aficionados in any case).

288 289 290 291 292 293
2007-05-03  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If a buffer doesn't have a timestamp, assume it's contiguous with
	  the previous buffer, and synthesise timestamps appropriately.

294 295 296 297 298 299
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* tests/check/elements/videorate.c: (GST_START_TEST):
	Set buffer timestamp to a valid value in order to test the buffer
	really does stay in videorate.

300 301 302 303 304 305
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	There is no sensible way to handle incoming buffers which don't have a
	valid timestamp. We therefore discard them and wait for the next one.

306 307 308 309 310 311
2007-05-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
	* gst/playback/gstdecodebin2.c: (plugin_init):
	  Better error message for text files.

312 313 314 315 316
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
	Fix offset bug in generation RR packets.

317 318 319 320 321 322 323 324 325
2007-04-27  Julien MOUTTE  <julien@moutte.net>

	* ext/theora/theoradec.c: (_theora_granule_time),
	(theora_dec_push_forward), (theora_handle_data_packet),
	(theora_dec_decode_buffer): Calculate buffer duration correctly
	to generate a perfect stream (#433888).
	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont): Glib provides ABS.

326 327 328 329 330 331 332 333 334 335 336
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix RB block parsing and writing.
	Add support for constructing BYE packets.

337 338 339 340 341 342 343 344 345
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
	(gst_base_audio_src_create):
	* po/POTFILES.in:
	  When posting a warning message because samples were dropped, post
	  something more intelligible than he default error message for clock
	  errors which is just confusing in this context (#432984).

346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
	(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
	(read_packet_header), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
	(gst_rtcp_packet_sdes_get_item_count),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_entry),
	(gst_rtcp_packet_sdes_next_entry),
	(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Implement code to write SR, RR and SDES packets.

363 364 365 366 367 368 369
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>

	* sys/ximage/ximagesink.c:
	  Fix build if XShm is not available (#432362).

370 371 372 373 374 375 376
2007-04-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
	Initalize the AudioConvertCtx with zeroes, otherwise it will contain
	pointers to random memory which are passed to g_free() when
	audio_convert_prepare_context() is called the first time.

377 378 379 380 381 382 383 384 385 386 387 388
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams <dcbw redhat com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  Don't leak incoming buffer if gst_pad_push() returns a
	  non-OK flow. Fixes #432755.
	 
	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	  Unit test for the above by Yours Truly.

389 390 391 392 393 394
2007-04-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
	(gst_adder_sink_event), (gst_adder_collected):
	  Fix non-flushing segmented seeks, Fixes #340060 for me

395 396 397 398 399 400 401 402 403 404 405
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Olivier Crete  <tester at tester ca>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init),
	(gst_base_rtp_audio_payload_init),
	(gst_base_rtp_audio_payload_dispose):
	  Chain up to parent class in dispose function; get rid of
	  unnecessary 'diposed' flag in private structure (#415001).

406 407 408 409 410 411 412 413 414
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init):
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	  Some minor docs fixes and additions; also add missing 'Since' bits.

415 416 417 418 419 420 421 422 423 424 425 426
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Zeeshan Ali  <zeenix gmail com>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
	(gst_base_rtp_audio_payload_push):
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  The recently-added gst_base_rtp_audio_payload_push() should take an
	  object of type GstBaseRTPAudioPayload as first argument (#431672).

427 428 429 430 431
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioresample/gstaudioresample.c:
	  Make more functions static, just because we can.

432 433 434 435 436
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/audioresample.c:
	  Add unit test for audioresample shutdown crasher (#420106).

437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452
2007-04-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/subparse/gstsubparse.c:
	* gst/subparse/samiparse.c:
	  Use GST_DISABLE_XML here

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_navigation_send_event):
	* sys/xvimage/xvimagesink.h:
	  Include stdlib.h when using atoi.
	  
	* tests/check/elements/playbin.c: (playbin_suite):
	  Use GST_DISABLE_REGISTRY here

453 454 455 456 457 458 459 460
2007-04-19  Michael Smith  <msmith@fluendo.com>

	* ext/theora/gsttheoraenc.h:
	* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
	(theora_enc_sink_event), (theora_enc_change_state):
	  Track initialisation state; don't try to use encoder state if we're
	  not initialised (it'll segfault).

461 462 463 464 465
2007-04-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/.cvsignore:
	Fix build.

466 467 468 469 470
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Allow random depths between 1 and 32 instead of only multiplies of 8.

471 472 473 474 475 476
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Set the maximum number of channels for PCM and float in the correct
	place to have it also used when creating the template caps.

477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Correctly support 4, 6 and 8 channels with normal PCM and float
	wav files.

	Fix the depth and signedness calculation in extensible wav files and
	also handle 1, 2, 4, 6, 8 channels here when a file without channel
	mask is found.

	Add support for float, alaw and mulaw in extensible wav files.

	This allows correct playback of all but 5 files from
	http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
	
	(gst_riff_create_audio_template_caps):
	Add voxware and float formats to the template caps.	

495 496 497 498 499 500 501 502 503 504 505
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Vincent Torri <vtorri at univ-evry dot fr>

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
	Fix unused variable warning if HAVE_LOCALTIME_R is undefinied

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
	Use the correct format strings for integer formats.

506 507 508 509 510 511 512
2007-04-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
	  Don't use pad_alloc_buffer_and_set_caps to create a small header
	  packet, or, worse, to create a big temporary video buffer using the
	  src pad.

513 514 515 516 517 518 519
2007-04-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, buffer_probe_cb, GST_START_TEST):
	  Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.

520 521 522 523 524 525 526
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
	  streamheader_suite):
	  Add another test set up for failure

527 528 529 530 531 532 533
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/Makefile.am:
	* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
	  GST_START_TEST, streamheader_suite, main):
	  Add a test for the streamheader bug Wim fixed.

534 535 536 537 538
2007-04-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/theora/theoradec.c: (theora_dec_sink_event):
	Fix misleading comment.

539 540 541 542 543
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  More sanity checks for the header fields.

544 545 546 547 548 549
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Try encodings from all environment variables, not just those in the
	  first environment variable that is set.

550 551 552 553 554 555 556 557 558 559
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_chain):
	Add some debug.

	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	Added check for videorate changing caps handling. Closes #421834.

560 561 562 563 564 565
2007-04-12  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  Use scale functions to avoid overflow when calculating duration of 
	  vorbis buffers.

566 567 568 569 570 571 572 573 574 575
2007-04-12  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_freeform_string_to_utf8):
	  API: add gst_tag_freeform_string_to_utf8() (#405072).

	* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
	  Use gst_tag_freeform_string_to_utf8() here.

576 577 578 579 580 581 582 583 584
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
	(gst_gdp_pay_sink_event):
	Make sure we set the IN_CAPS flag correctly.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Get the IN_CAPS flag before we call functions that mess with the flags.

585 586 587 588 589 590 591 592 593
2007-04-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
	  gst_gdp_pay_chain, gst_gdp_pay_sink_event):
	  Only stamp buffers with offset/offset_end right before they get
	  pushed.  This ensures offset continuity, which was not the case
	  before as shown by
	  gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE

594 595 596 597 598 599
2007-04-06  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybin.c: (add_sink),
	(gst_play_bin_change_state):
	Activate sync in playbin, we are ready to handle it for live streams.

600 601 602 603 604 605
2007-04-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/playbin.c:
	(test_sink_usage_video_only_stream), (playbin_suite):
	  Add small test for stream-info-value-array code paths.

606 607 608 609 610 611 612
2007-04-05  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving):
	Don't try to create invalid calibration parameters by making the
	internal time go backwards, instead make external time go forward.

613 614 615 616 617 618 619 620
2007-04-05  Wim Taymans  <wim@fluendo.com>

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

	* gst/playback/gstplaybasebin.c: (add_stream):
	Fix leak in add_stream(), when g_value_set_object() increases the
	refcount of streaminfo object. Fixes #426250.

621 622 623 624 625 626 627 628 629 630 631 632
2007-04-03  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add a test pattern called "circular", which has concentric
	  rings with varying radial frequency.  The main purpose of this
	  pattern is to test fidelity loss in a filter or scaler element.
	  Notably, this pattern is scale invariant, and is optimally viewed
	  with a width (and height) of 400.

633 634 635 636 637 638 639 640 641 642 643 644
2007-04-03  Wim Taymans  <wim@fluendo.com>

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

	* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
	(deactivate_free_recursive):
	Decodebin2 doesn't unref pads it obtains in some occasions:
	- multiqueue src pads, when either connecting further or exposing
	- sink pads of new autoplugged elements
	- peer pads when recursively freeing elements
	Fixes #425455.

645 646 647 648 649 650
2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Add audio/x-raw-float support, now that audioconvert support
	non-native endianness floats.

651 652 653 654 655 656
2007-03-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	  gstreamer-plugins-base.pc doesn't exist, it's
	  gstreamer-plugins-base-0.10.pc.

René Stadler's avatar
René Stadler committed
657 658 659 660 661 662 663 664 665 666 667 668 669 670 671 672 673 674
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: René Stadler <mail at renestadler dot de>
	with some minor changes

	* gst-libs/gst/floatcast/floatcast.h:
	Use more efficient float endianness conversion functions that don't
	involve 2 function calls per value.
	* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	(check_default), (audio_convert_prepare_context):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_parse_caps), (make_lossless_changes):
	Support non-native endianness floats as input and output.
	Fixes #339838.
	* tests/check/elements/audioconvert.c: (verify_convert),
	(GST_START_TEST):
	Add unit tests for the non-native endianness float conversions.

675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691 692
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_base_init),
	(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Add Private structure.
	Bring element code to 2007.
	Parse clock-base caps param and use it when generating the
	newsegment.
	Reset variables before going to PAUSED.
	Fix some docs.

Wim Taymans's avatar
Wim Taymans committed
693 694 695 696 697 698 699 700 701 702 703 704 705 706 707 708 709 710 711 712 713 714 715 716 717 718 719 720 721 722 723 724 725 726 727 728
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_get_adapter):
	Add RTCP docs.
	Fix some more docs.

	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
	(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
	(gst_rtcp_buffer_get_packet_count), (read_packet_header),
	(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
	(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
	(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
	(gst_rtcp_packet_sr_get_sender_info),
	(gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
	(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
	(gst_rtcp_packet_sdes_get_chunk_count),
	(gst_rtcp_packet_sdes_first_chunk),
	(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
	(gst_rtcp_packet_bye_get_ssrc_count),
	(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_get_reason_len),
	(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Add new helper object for parsing and creating RTCP messages.

729 730 731 732 733 734
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	PCM samples with width=8 must be always unsigned, no matter what
	depth they have.

735 736 737 738 739 740 741 742
2007-03-29  Andy Wingo  <wingo@pobox.com>

	* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
	perfect offsets also, not just timestamps.

	* tests/check/elements/videorate.c (test_more): Test that given
	any incoming offsets, that videorate produces perfect offsets.

743 744 745 746 747
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/riff/riff-ids.h:
	Add some more RIFF formats.

748 749 750 751 752 753 754 755 756
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	(gst_rtp_buffer_default_clock_rate):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Fix fixed payload names and docs.
	Added method to get the default clock rates of fixed payload types.
	API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()

757 758 759 760 761
2007-03-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* tests/check/pipelines/.cvsignore:
	Add new vorbisdec test to cvsignore.

762 763 764 765 766 767 768 769 770 771 772 773 774 775 776 777 778 779 780
2007-03-28  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
	(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
	(gst_base_audio_sink_set_property),
	(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
	(clock_convert_external), (gst_base_audio_sink_resample_slaving),
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_async_play):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	Store private stuff in GstBaseAudioSinkPrivate.
	Add configurable clock slaving modes property.
	API:: GstBaseAudioSink::slave-method property
	Some more latency reporting tweaks.
	Added skew based clock slaving correction and make it the default until
	the resampling method is more robust.

781 782 783 784 785 786 787 788 789 790 791 792 793 794 795
2007-03-27  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/audioconvert.c:
	Add docs to the integer pack functions and implement proper
	rounding. Before we had rounding towards negative infinity, i.e.
	always the smaller number was taken. Now we use natural rounding,
	i.e. rounding to the nearest integer and to the one with the largest
	absolute value for X.5. The old rounding introduced some minor
	distortions. Fixes #420079
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix one unit test that assumed the old rounding and added unit tests
	for checking signed/unsigned int16 <-> signed/unsigned int16 with
	depth 8, one for signed int16 <-> unsigned int16 and one for the new
	rounding from signed int32 to signed/unsigned int16.

796 797 798 799 800 801
2007-03-27  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
	(gst_audio_convert_transform_caps):
	  Fix typo in debug line introduced recently, as pointed out on irc.

802 803 804 805 806 807 808 809
2007-03-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	* tests/check/libs/tag.c: (GST_START_TEST):
	  Make sure we parse floating-point numbers in vorbis comments
	  correctly with either '.' or ',' as separator, no matter what
	  the current locale is. Add unit test for this too.

810 811 812 813 814 815 816 817 818 819 820 821
2007-03-26  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler  <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
	  When writing out floating-point numbers to vorbis comment tags, always
	  use the same character as separator no matter what the current locale is
	  (fixes #423051).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit tests for replaygain tags in vorbis comments (closes #423055).

822 823 824 825 826 827 828 829 830 831 832
2007-03-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
	  vorbis_handle_data_packet):
	  Correctly set DURATION to generate a timestamp-continuous stream.
	  One bug left at the end; see
	  ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
	* tests/check/Makefile.am:
	* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
	  Add a test to check this.  Without the above patch this test fails.

833 834 835 836 837
2007-03-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/rtp/Makefile.am:
	The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.

838 839 840 841 842 843 844 845
2007-03-23  Michael Smith  <msmith@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_reset), (gst_video_rate_chain):
	  If videorate changes caps, we can no longer use the old buffer
	  (which may have a different size, incompatible with our caps).
	  So don't do that; just duplicate the new frame more times.

846 847 848 849 850 851
2007-03-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	Remove playbin's override of the set_clock vmethod. It's irrelevant
	after Wim's commit on the 19th.

852 853 854 855 856 857 858
2007-03-22  Wim Taymans  <wim@fluendo.com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
	(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
	* ext/gnomevfs/gstgnomevfssrc.h:
	Don't cache file sizes. Fixes #341078.

859 860 861 862 863
2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (add_sink):
	  Use GST_PTR_FORMAT to log caps. 

864 865 866 867 868 869 870 871
2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/subparse/samiparse.c: (handle_start_font):
	  Special-case some more colour names that pango doesn't handle by
	  default. Fixes #420578.

872 873 874 875 876 877 878 879
2007-03-20  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  If we get a zero-sized input buffer, don't pass it to libvorbis, as
	  that marks EOS internally. After that, libvorbis will buffer all
	  input data, and encode none of it, eventually leading to memory
	  exhaustion.

880 881 882 883 884 885 886 887 888 889 890 891
2007-03-19  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (remove_fakesink):
	Don't post STATE_DIRTY anymore.

	* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
	(gst_play_bin_change_state):
	Remove stream_time reset in seek handling, core does that now.
	Disable clocking for live pipelines by forcing a NULL clock to the
	complete pipeline, core is too smart now for our previous hack.
	We can always autoplug in PAUSED now.

892 893 894 895 896
2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS:  Update this file, change the formatting to make
	it more consistent, plus more machine readable.

897 898 899 900 901 902 903
2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(strip_width_64), (append_with_other_format):
	  Previous fix was too simplistic, and broke the tests. Use a better
	  approach; only strip 64 from widths for integer audio.

904 905 906 907 908 909 910 911
2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(gst_audio_convert_transform_caps):
	  We don't support 64 bit integer audio, so don't try to claim we can.
	  Stops us producing caps don't match our template caps.
	  Update comments.

912 913 914 915 916 917 918 919
2007-03-15  Michael Smith  <msmith@fluendo.com>

	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont), (audioresample_transform):
	  Don't trigger discontinuities for very small imperfections; a filter
	  flush will sound bad, and many plugins have rounding errors leading
	  to these.

920 921
2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 

922 923
	Patch by Olivier Crete <olivier.crete@collabora.co.uk>

924 925
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
926 927 928
	API: add "min-ptime" property to RTP base audio payloader.
	API: add gst_base_rtp_audio_payload_push().
	API: add gst_base_rtp_audio_payload_get_adapter().
929 930 931
	Fixes #415001
	Indentation/whitespace/documentation fixes.

932 933 934 935 936 937 938 939 940 941 942 943
2007-03-14  Julien MOUTTE  <julien@moutte.net>

	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	(audioresample_transform_size), (audioresample_do_output),
	(audioresample_transform), (audioresample_pushthrough): Handle
	discontinuous streams.
	* gst/audioresample/gstaudioresample.h:
	* tests/check/elements/audioresample.c:
	(test_discont_stream_instance), (GST_START_TEST),
	(audioresample_suite): Add a test for discontinuous streams.
	* win32/common/config.h: Updated.

944 945 946 947 948 949 950 951 952 953 954 955 956 957 958 959 960
2007-03-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update translations from translation project.

961 962 963 964 965 966 967 968
2007-03-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioresample/debug.h:
	* gst/audioresample/resample.c: (resample_init):
	  Since I really am not interested in a debug line for each sample
	  being processed, move the library's debugging to its own category,
	  libaudioresample

969 970 971 972 973 974 975 976
2007-03-13  Michael Smith  <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  Since the plugin doesn't support anything other than 4:2:0 right
	  now, post an error and fail if we get something else. Won't matter
	  until libtheora supports the other pixel formats, but hopefully
	  that'll be soon...

977 978 979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994
2007-03-10  Sebastien Moutte  <sebastien@moutte.net>

	* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
	Use gst_guint64_to_gdouble for conversion.
	* win32/MANIFEST:
	Add new files to the win32 MANIFEST.
	* win32/common/libgstaudio.def:
	* win32/common/libgstpbutils.def:
	Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstplaybin.dsp:
	Change the link to libgstpbutils.lib.
	* win32/vs6/libgstdecodebin2.dsp:
	Add a new project for decodebin2.
	* win32/vs6/libgstpbutils.dsp:
	Add a new project for pbutils.

995 996 997 998 999 1000 1001 1002 1003
2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Also accept partial dates with only year and month,
	  like 1999-12-00 (fixes #410396 even more).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit test for the above.

1004 1005 1006 1007 1008 1009
2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	  Add unit test for MPL2 subtitle format (#413799).

1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020 1021 1022 1023
2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Kamil Pawlowski  <kamilpe gmail com>

	* gst/subparse/Makefile.am:
	* gst/subparse/gstsubparse.c:
	(gst_sub_parse_data_format_autodetect),
	(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
	(gst_subparse_type_find):
	* gst/subparse/gstsubparse.h:
	* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
	* gst/subparse/mpl2parse.h:
	  Add support for MPL2 subtitle format (#413799).

1024 1025 1026 1027 1028
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We require core CVS for the new buffer metadata copy functions.

1029 1030 1031 1032 1033 1034
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/tag/gstid3tag.c:
	Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.

Alex Lancaster's avatar
Alex Lancaster committed
1035 1036
	Patch by: Alex Lancaster <alexl at users sourceforge net>

1037 1038 1039 1040 1041 1042
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
	(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
	Improve adapter usage and comments.

Wim Taymans's avatar
Wim Taymans committed
1043 1044 1045 1046 1047 1048 1049 1050 1051 1052 1053 1054
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* ext/pango/gsttextrender.c: (gst_text_render_chain):
	* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
	* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
	Use new metadata copy function.

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_transform):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
	Basetransform copied the metadata for us.

1055 1056 1057 1058 1059 1060 1061 1062 1063 1064 1065 1066 1067 1068 1069 1070
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
	(gst_text_overlay_video_event):
	  Some more logging. Only accept newsegment events in TIME format and
	  send a WARNING message if they are not in TIME format.

	* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
	(gst_sub_parse_chain), (gst_sub_parse_sink_event):
	* gst/subparse/gstsubparse.h:
	  No need to allocate GstSegment structure dynamically, just put it
	  into the instance structure; ignore newsegment events in BYTE
	  format and in particular don't let it overwrite our saved TIME
	  segment from the last seek.

1071 1072 1073 1074 1075 1076
2007-03-09  Michael Smith  <msmith@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
	  Replace AC3 typefinder with one that isn't terrible, and actually
	  works usefully.

1077 1078 1079 1080 1081 1082 1083
2007-03-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_transform):
	  fix error category and translatable string
	  

1084 1085 1086 1087 1088 1089
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	  Fix up utils => pbutils here too.

1090 1091 1092 1093 1094 1095
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstsubparse.c: (handle_buffer):
	  Break out of loop in chain function as soon as possible if we get
	  a non-OK flow return.

1096 1097 1098 1099 1100 1101
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Unref the mixer if the state change fails too (if the
	alsa devices are inaccessible, for example)

1102 1103 1104 1105 1106 1107 1108 1109 1110 1111 1112 1113
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Don't test libvisual elements in the states check, because libvisual
	seems to leak internally.

	Re-enable the alsa and states tests now that there's new suppressions
	in gst.supp.

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Don't leak the alsamixer we instantiated.

1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126 1127 1128
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state), (gst_ximagesink_reset),
	(gst_ximagesink_finalize):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
	(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
	Move some cleanup stuff from the state change handler into a _reset()
	function that can be called from _finalize(). This ensures that things
	get freed even if (for some reason) the NULL->READY state transition
	fails in the parent class.
	Even if a parent state change fails, process our downward state change
	logic instead of bailing out early.
	Free the correct xcontext pointer in ximagesink's xcontext_clear.

1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	Extra log line.

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
	* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
	Use pango_font_description_set_family_static instead of 
	pango_font_description_set_family to save a string copy (it was
	leaking due to the strdup anyway)

	* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
	* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
	* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
	Chain up in finalize.

1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157 1158 1159 1160
2007-03-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/interfaces/mixertrack.c:
	(gst_mixer_track_class_init), (gst_mixer_track_get_property),
	(gst_mixer_track_set_property):
	  API: add "untranslated-label" property which should be set by
	  implementations at construct time (#414645).

	* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Set "untranslated-label" when constructing mixer track objects.

	* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
	  Unit test to check the above.

1161 1162 1163 1164 1165
2007-03-07  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	Fix confusing debug message.

1166 1167 1168 1169 1170
2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-plugins-base.doap:
	update doap file with new version

Jan Schmidt's avatar
Jan Schmidt committed
1171 1172 1173 1174 1175
2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

Jan Schmidt's avatar
Jan Schmidt committed
1176 1177 1178 1179 1180 1181 1182
=== release 0.10.12 ===

2007-03-07  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.12, "Zombie Horde"

1183 1184 1185 1186 1187
2007-03-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump version to 0.10.11.4 pre-release

1188 1189 1190 1191 1192 1193 1194
2007-03-06  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	Fix regression that made GStreamer skip the first samples of audio.
	Fixes #414684.

1195 1196 1197 1198 1199
2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump version to 0.10.11.3 pre-release

1200 1201 1202 1203 1204
2007-03-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* po/POTFILES.in:
	  Update paths for the rename from utils to pbutils to fix the build.

1205 1206 1207 1208 1209 1210
2007-03-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/Makefile.am:
	  Change directory to install headers in from gst/utils to gst/pbutils
	  as well.

1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254 1255 1256 1257 1258 1259 1260 1261
2007-03-05  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/descriptions.c:
	(gst_pb_utils_get_source_description),
	(gst_pb_utils_get_sink_description),
	(gst_pb_utils_get_decoder_description),
	(gst_pb_utils_get_encoder_description),
	(gst_pb_utils_get_element_description),
	(gst_pb_utils_add_codec_description_to_tag_list),
	(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
	* gst-libs/gst/pbutils/descriptions.h:
	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/pbutils/missing-plugins.c:
	(gst_missing_uri_source_message_new),
	(gst_missing_uri_sink_message_new),
	(gst_missing_element_message_new),
	(gst_missing_decoder_message_new),
	(gst_missing_encoder_message_new),
	(gst_missing_plugin_message_get_description):
	* gst-libs/gst/pbutils/missing-plugins.h:
	* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
	* gst-libs/gst/pbutils/pbutils.h:
	* gst-libs/gst/utils/Makefile.am:
	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/base-utils.h:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/descriptions.h:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/install-plugins.h:
	* gst-libs/gst/utils/missing-plugins.c:
	* gst-libs/gst/utils/missing-plugins.h:
	* gst-plugins-base.spec.in:
	* gst/playback/Makefile.am:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybasebin.c: (setup_subtitle),
	(gen_source_element):
	* gst/playback/gstplaybin.c: (plugin_init):
	* tests/check/Makefile.am:
	* tests/check/libs/pbutils.c: (GST_START_TEST),
	(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
	* tests/check/libs/utils.c:
	  rename utils to pbutils

1262 1263 1264 1265 1266 1267 1268 1269 1270 1271
2007-03-02  Jan Schmidt  <thaytan@mad.scientist.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
	Add documentation for decodebin2 that indicates that the API
	is still unstable.

1272 1273 1274 1275 1276
2007-03-01  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Update to 0.10.11.2 (0.10.12 pre-release)

1277 1278 1279 1280 1281 1282
2007-03-01  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	base time is irrelevant here.

1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298
2007-03-01  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	Improve debugging.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_query), (gst_base_audio_sink_event),
	(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
	Improve latency and clock slaving calculations.
	Improve slave clock calibration.

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_commit_full):
	When we are asked to render N sample to 0 bytes, return N.

1299 1300 1301 1302 1303 1304 1305 1306 1307 1308 1309 1310 1311 1312 1313 1314 1315
2007-03-01  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	(gst_alsasink_write), (gst_alsasink_reset):
	* ext/alsa/gstalsasink.h:
	Remove unused dispose function.
	Rename lock to not interfere with alsasrc lock.

	* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
	(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
	(gst_alsasrc_read), (gst_alsasrc_reset):
	* ext/alsa/gstalsasrc.h:
	Implement finalize function.
	Use lock to protect alsa access.
	Implement _reset.
	Fine tune sw params.

1316 1317 1318 1319 1320
2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  Convert to new AG_GST style.

1321 1322 1323 1324 1325 1326 1327 1328 1329
2007-02-28  Wim Taymans  <wim@fluendo.com>

	Patch by: Ed Catmur <ed at catmur dot co dot uk>

	* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
	(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
	Fix race condition when rapidly switching visualisations in playbin.
	Fixes #401029.

1330 1331 1332 1333 1334 1335
2007-02-28  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Include local stuff before system installed things in LDFLAGS and
	CFLAGS.

1336 1337 1338 1339 1340
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
	Improve debugging.

1341 1342 1343 1344 1345 1346 1347 1348
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
	(gst_v4lsrc_fixate), (gst_v4lsrc_query):
	* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
	Fix duration and timestamping, taking latency into account.
	Implement latency query.

1349 1350 1351 1352 1353 1354 1355 1356 1357 1358 1359 1360 1361 1362
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	(gst_audio_clock_new):
	Fix clock name.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_init), (gst_base_audio_sink_query):
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
	(gst_base_audio_src_create):
	Improve latency query code.
	Use proper clock names.

1363 1364 1365 1366 1367 1368 1369
2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/generic/states.c: (GST_START_TEST):
	  Copy the states.c test from core again
	* tests/check/Makefile.am:
	  ignore cdio and cdparanoiasrc

1370 1371 1372 1373 1374 1375 1376
2007-02-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index), (check_default),
	(audio_convert_prepare_context), (audio_convert_convert):
	  Also make valgrind happy and avoid copying data in some cases.

1377 1378 1379 1380 1381 1382 1383 1384 1385 1386 1387 1388 1389
2007-02-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
	(gst_audio_convert_transform_caps):
	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Don't run inplace if that overwrites source data as we go. Add more
	  tests. Fixes #339837 even more.

1390 1391 1392 1393 1394 1395 1396
2007-02-27  Julien MOUTTE  <julien@moutte.net>

	* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
	(msg_segment_done): Fix various seeking bugs (Slider was not
	updating when doing a non flushing seek, Reverse playback 
	on segment seek was wrong).

1397 1398 1399 1400 1401 1402
2007-02-26  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/seek.c: (stop_seek):
	When we stop scrubbing, don't leave the pipeline PLAYING when we
	requested a PAUSED state.

1403 1404 1405 1406 1407 1408 1409 1410 1411 1412 1413
2007-02-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse date strings in vorbis comments that have an invalid (zero)
	  month or day (#410396).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Test case for the above.

1414 1415 1416 1417 1418 1419 1420 1421 1422
2007-02-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Loïc Minier <lool+gnome at via ecp fr>

	* configure.ac:
	* ext/alsa/Makefile.am:
	* gst/audiotestsrc/Makefile.am:
	  Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).

1423 1424 1425 1426 1427 1428
2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: point out that the application needs to assist playbin
	  with buffering.

1429 1430 1431 1432 1433 1434 1435 1436 1437 1438 1439
2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	* tests/check/libs/utils.c: (missing_msg_check_getters):
	  Change GStreamer marker prefix in detail string from 'gstreamer.net'
	  to just 'gstreamer'. Document the caps string component of the
	  decoder/encoder detail a bit better, since not everyone will be
	  familiar with the GStreamer media type/caps system (but they better
	  enjoy nested itemized lists).

1440 1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452 1453
2007-02-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
	  Fix copying of GstNetBuffer (would crash before, or at least lead to
	  invalid memory access, #410772), for now by copying the GstBuffer copy
	  code from the core over here so we can copy the GstBuffer fields on a
	  provided buffer instance (of type GstNetBuffer in this case). Would be
	  better to fix this with some support by the core though (and in the long
	  run change the broken GstBuffer/GstMiniObject copy semantics, #393099).

	* tests/check/Makefile.am:
	  Enable unit test for GstNetBuffer.

1454 1455 1456 1457 1458 1459
2007-02-22  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_init): Disable pull-mode activation until we
	figure out how to make audio sinks go to PLAYING.

1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470 1471 1472
2007-02-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
	(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
	* gst/audioconvert/gstchannelmix.h:
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Add float as an intermediate format, as well as float mixing. Enable
	  test that was failing before. Fixes #339837

1473 1474 1475 1476 1477 1478
2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/examples/seek/seek.c: (do_seek):
	Undo the previous commit: -1 as a stop time implies that the stop
	time is the end of file, clearing any previously configured segment.

1479 1480 1481 1482 1483
2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/examples/seek/seek.c: (do_seek):
	Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.

1484 1485 1486 1487 1488 1489
2007-02-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c: (volume_process_int16),
	(volume_process_int16_clamp), (volume_set_caps):
	  Unbreak volume, value remains gint.

1490 1491 1492 1493 1494 1495 1496 1497 1498 1499
2007-02-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_set_volume),
	(gst_volume_init), (volume_process_double), (volume_process_float),
	(volume_process_int16), (volume_process_int16_clamp),
	(volume_set_caps), (volume_transform_ip), (volume_update_volume):
	* gst/volume/gstvolume.h:
	  Extend float audio support (double) and some int->uint cleanups.

1500 1501 1502 1503 1504 1505 1506 1507 1508
2007-02-20  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
	(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
	(sort_end_pads), (gst_decode_group_expose),
	(gst_decode_group_hide):
	Don't free groups from the streaming threads. Just put them aside and
	free them in dispose.

1509 1510 1511 1512 1513 1514 1515 1516 1517 1518 1519
2007-02-20  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (connect_element),
	(pad_added_group_cb), (gst_decode_group_check_if_blocked),
	(sort_end_pads), (gst_decode_group_expose):
	Handle dynamic pads within groups.
	Sort pads before exposing them in order to make playbin happy.
	There still is a race with the multiqueue filling up. This should be
	solved separately.
	Fixes #398721

1520 1521 1522 1523 1524 1525 1526 1527
2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	  Some more docs (and descriptions for two subtitle formats).

1528 1529 1530 1531 1532
2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/audio.c:
	  Fix documentation.

1533 1534 1535 1536 1537 1538 1539
2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Yves Lefebvre  <ivanohe abacom com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
	  Don't leak caps. Fixes #408278.

1540 1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552
2007-02-15  Stefan Kost  <ensonic@users.sf.net>

	* ext/cdparanoia/gstcdparanoiasrc.h:
	* ext/ogg/gstoggdemux.h:
	* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
	(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
	(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/interfaces/videoorientation.h:
	* gst/adder/gstadder.h:
	  More docs coverage and some ChangeLog surgery (add missing names)

Wim Taymans's avatar
Wim Taymans committed
1553 1554 1555 1556 1557 1558 1559 1560
2007-02-15  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c:
	(gst_ximagesink_calculate_pixel_aspect_ratio):
	* sys/xvimage/xvimagesink.c:
	(gst_xvimagesink_calculate_pixel_aspect_ratio):
	Small constifications.

1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575 1576 1577 1578
2007-02-15  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
	(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
	(gst_base_audio_sink_async_play),
	(gst_base_audio_sink_change_state):
	Answer latency query.
	Use configured latency when syncing.
	Fix clock slaving.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
	(gst_base_audio_src_query), (gst_base_audio_src_change_state):
	Fix possible memleak.
	Implement latency query.
	Small cleanups.

1579 1580 1581 1582 1583 1584 1585
2007-02-15  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
	Ignore errors in reset, these are not fatal. They also grab the element
	lock which is already taking when this function is called. Fixes
	#405451.

1586 1587 1588 1589 1590
2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Remove 'tests/examples/xerror/Makefile' from output files again.

1591 1592 1593 1594 1595 1596
2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Also crossref against gst-plugins-base-libs.

1597 1598 1599 1600 1601 1602 1603 1604 1605 1606 1607 1608 1609
2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.

	* gst-libs/gst/audio/audio.h:
	  Source formatting.

	* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
	  Add own debug category.

1610 1611 1612 1613 1614 1615 1616 1617
2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
	  (#403597).

1618 1619 1620 1621 1622 1623 1624 1625 1626 1627 1628
2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (setup_source):
	  When we have external subtitles and wait for the subtitle decodebin
	  to get up and running, we set up a (sync) bus handler for the
	  subtitle decodebin, so we can stop waiting when it posts an error
	  message. However, we should do that before we set the subtitle
	  decodebin's state to playing, otherwise things are racy and we might
	  miss error messages posted before we had a chance to set up the bus.
	  This should finally fix totem hanging on .txt pseudo-subtitle files.
	  
1629 1630 1631 1632 1633 1634 1635 1636 1637 1638 1639 1640 1641 1642 1643 1644 1645 1646 1647 1648 1649 1650 1651
2007-02-10  Sébastien Moutte  <sebastien at moutte dot net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	  Use gst_gdouble_to_guint64 for conversions.
	* win32/common/config.h.in:
	  Add a define for GST_INSTALL_PLUGINS_HELPER
	* win32/common/libgstaudio.def:
	* win32/common/libgstcdda.def:
	* win32/common/libgstnetbuffer.def:
	* win32/common/libgstrtp.def:
	* win32/common/libgutils.def:
	  Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstnetbuffer.dsp:
	* win32/vs6/libgstplaybin.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstvorbis.dsp:
	* win32/vs6/libgstcdda.dsp:
	* win32/vs6/libgstgdp.dsp:
	* win32/vs6/libgstutils.dsp:
	  Update and add new project files.

1652 1653 1654 1655 1656 1657 1658 1659 1660 1661
2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
	(subrip_remove_unhandled_tags), (parse_subrip):
	  For SubRip (.srt) subtitles, ignore all markup tags we don't
	  handle (like font tags, for example).

	* tests/check/elements/subparse.c:
	  Add test for this.

1662 1663 1664 1665 1666 1667
2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (add_fakesink),
	(gst_decode_bin_change_state):
	* gst/playback/gstdecodebin2.c: (add_fakesink),
	(gst_decode_bin_change_state):
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
1668
	  Don't error out if there is no fakesink in the NULL to READY state
1669 1670 1671 1672 1673 1674 1675 1676 1677
	  change, since when decodebin is re-used, we're only adding the
	  fakesink element in READY to PAUSED.

	* tests/check/elements/decodebin.c:
	(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
	(decodebin_suite):
	  Minimal unit test to make sure we can use the same decodebin
	  instance twice (at least with audiotestsrc input).

1678 1679 1680 1681 1682 1683 1684
2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
	  Try to get devic-name from device string first, and from handle only
	  as fallback (seems to yield better results and is more robust
	  against buggy probing code on the application side).

1685 1686 1687 1688 1689 1690 1691 1692 1693 1694 1695 1696 1697
2007-02-08  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Julien Puydt <julien.puydt at laposte net>

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
	(gst_alsa_find_device_name):
	* ext/alsa/gstalsa.h:
	* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
	  Improve device-name detection a bit, especially in the case where
	  the device is not actually open (#405020, #405024). Move common code
	  into gstalsa.c instead of duplicating it.

1698 1699 1700 1701 1702
2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c:
	  Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.

1703 1704 1705 1706 1707 1708 1709 1710 1711 1712 1713 1714 1715 1716 1717 1718 1719
2007-02-06  Julien MOUTTE  <julien@moutte.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_clear),
	(gst_xvimagesink_interface_supported),
	(gst_xvimagesink_probe_get_properties),
	(gst_xvimagesink_probe_probe_property),
	(gst_xvimagesink_probe_needs_probe),
	(gst_xvimagesink_probe_get_values),
	(gst_xvimagesink_property_probe_interface_init),
	(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init),
	(gst_xvimagesink_get_type):
	* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
	for XVAdaptors so that one can choose the adaptor to use with 
	gstreamer-properties.

1720 1721 1722 1723 1724
2007-02-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	  Also mention that a conversion from double to float is suboptimal still.

1725 1726 1727 1728 1729 1730 1731
2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstaudiofilter.c:
	(gst_audio_filter_class_init), (gst_audio_filter_change_state):
	  Clear our formats structure and free the caps contained in it when
	  shutting down.

1732 1733 1734 1735 1736 1737 1738
2007-02-05  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_callback): Update basesink->offset so that we
	pull monotonically increasing offsets instead of, um, seeking back
	to 0 each time. Fixes alsasrc ! alsasink!

1739 1740 1741 1742 1743 1744
2007-02-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videoscale/gstvideoscale.c:
	  A width and height of 1 makes us crash, so increase minimum size to
	  2x2 pixels until someone feels like fixing this (#404512).

1745 1746 1747 1748 1749 1750
2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
	  Add small test to make sure request pads are cleaned up properly
	  even if oggmux never changes state out of NULL.

1751 1752 1753 1754 1755 1756
2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/utils.c: (GST_START_TEST):
	  Fix unit test. Turns out things work much better when you
	  NULL-terminate string arrays. Should make p5 build bot happy again.

1757 1758 1759 1760 1761 1762 1763 1764 1765 1766 1767 1768 1769 1770
2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	(gst_audio_filter_template_base_init),
	(gst_audio_filter_template_class_init),
	(gst_audio_filter_template_init),
	(gst_audio_filter_template_set_property),
	(gst_audio_filter_template_get_property),
	(gst_audio_filter_template_setup),
	(gst_audio_filter_template_filter),
	(gst_audio_filter_template_filter_inplace), (plugin_init):
	  Oops, forgot to commit fixed-up example.

1771 1772 1773 1774 1775 1776 1777 1778 1779 1780 1781 1782 1783 1784 1785
2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
	(gst_audio_filter_class_init), (gst_audio_filter_init),
	(gst_audio_filter_set_caps),
	(gst_audio_filter_class_add_pad_templates):
	* gst-libs/gst/audio/gstaudiofilter.h:
	  Port GstAudioFilter to 0.10. This change technically breaks
	  API and ABI (and thus also every library developer's heart),
	  but seems justifiable on the grounds that the base class was
	  completely unusable before (ie. would crash immediately when
	  actually used). Fixes #403963 (and eventually also #403572).
	  Also document all of this a bit.

1786 1787 1788 1789 1790 1791 1792 1793 1794
2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/install-plugins.c:
	(gst_install_plugins_spawn_child):
	* tests/check/libs/utils.c:
	(test_base_utils_install_plugins_do_callout):
	  Lowering log level to see why things fail on the p5 build bot;
	  fix some typos in unit test messages.

1795 1796 1797 1798 1799 1800 1801
2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/utils.c:
	(test_base_utils_install_plugins_do_callout):
	  Don't hard-code temp directory for test helper; use GLib functions
	  to write out file and do error checking etc.

1802 1803 1804 1805 1806 1807 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 1833 1834 1835 1836 1837
2007-02-02  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/Makefile.am:
	* gst-libs/gst/utils/base-utils.h:
	* gst-libs/gst/utils/install-plugins.c:
	(gst_install_plugins_context_set_xid),
	(gst_install_plugins_context_new),
	(gst_install_plugins_context_free),
	(gst_install_plugins_get_helper),
	(gst_install_plugins_spawn_child),
	(gst_install_plugins_return_from_status),
	(gst_install_plugins_installer_exited),
	(gst_install_plugins_async), (gst_install_plugins_sync),
	(gst_install_plugins_return_get_name),
	(gst_install_plugins_installation_in_progress):
	* gst-libs/gst/utils/install-plugins.h:
	  API: add API for applications to initiate installation of missing
	  plugins, ie. gst_install_plugins_async() primarily.
	  Based on libgimme-codec by Ryan Lortie.

	* configure.ac:
	  Add --with-install-plugins-helper configure option so distros can specify
	  the path of the helper script or program to call when plugin installation
	  is requested (distros: please do any argument munging in this helper
	  script instead of patching GStreamer to pass arguments differently
	  to another program directly).

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Build and document new API.

	* tests/check/libs/utils.c: (result_cb),
	(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
	(libgstbaseutils_suite):
	  Some simple checks for the new API.