ChangeLog 423 KB
Newer Older
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
	(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
	  Double-check that RGB input caps are really RGBA caps (apparently
	  the core doesn't always catch it if those caps aren't a subset of
	  our template caps, also see #421543). Fixes #429319 in a way.
	  Also, don't leak the pad template in the transform_caps function.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/alphacolor.c: (setup_alphacolor),
	(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
	(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
	(GST_START_TEST), (alphacolor_suite):
	  Add some basic unit tests for alphacolor.

18
19
20
21
22
23
24
25
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  If we get a fatal flow return in the loop function, first post the
	  error message and only then send the EOS event downstream, otherwise
	  applications might get an eos message before the error message and
	  think everything was ok (related to #429319).

26
27
28
29
30
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	Read the channel byte as an unsigned byte.

31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
	(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
	(gst_rtp_gsm_depay_setcaps):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
	(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
	(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
	(gst_ilbc_depay_get_property):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
	(gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
	(gst_rtp_pcmu_depay_setcaps):
	Make sure we configure the clock_rate in the baseclass in the setcaps
	function. Fixes #431282.

53
54
55
56
57
58
59
60
61
62
63
64
65
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_stream_free), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Parse server address from SDP.
	Hook up a udpsink to send RTCP back to the server.

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtsp/rtsptransport.h:
	Add some docs.

66
67
68
69
70
2007-04-25  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Make header field check conditional. Fixes #433135

71
72
73
74
75
76
77
78
79
80
81
82
2007-04-24  Tim-Philipp Müller  <tim at centricular dot 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-alphacolor.xml:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  Add minimal docs blurb to alphacolor; split out headers into
	  separate header file for gtk-doc.

83
84
85
86
87
88
2007-04-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c: (gst_progress_report_report):
	  Don't try to post NULL message (in case we can't query upstream
	  position or duration).

89
90
91
92
93
94
95
96
2007-04-18  Michael Smith  <msmith@fluendo.com>

	* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
	(gst_cutter_get_caps):
	* gst/cutter/gstcutter.h:
	  Fix some of the most obvious bugs in cutter. Now doesn't leak
	  everything if input is silent.

97
98
99
100
101
102
103
104
2007-04-18  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Wav apparently only supports width==GST_ROUND_UP(depth), everything
	else results in a invalid block align and invalid files.

105
106
107
108
109
110
111
2007-04-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Snaik <snaik32 gmail com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
	  Add missing break statement for BOX_HORIZONTAL case.

112
113
114
115
116
117
118
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	Use correct format strings for integer types.

119
120
121
122
123
124
125
126
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	(gst_wavparse_create_sourcepad):
	Use gst_riff_create_audio_template_caps () instead of the local caps.
	This makes updates of the local caps unecessary whenever libgstriff
	gets support for new formats.

127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
2007-04-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Brian Cameron  <brian.cameron at sun dot com>

	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/sunaudio/gstsunaudiosrc.h:
	  Fix and/or update copyright attributions (#430228).

143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
2007-04-13  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	Fix docs.

	* gst/rtsp/URLS:
	Add some more example urls.

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_chain_rtp):
	Better debugging.

	* gst/rtsp/gstrtspsrc.c: (request_pt_map),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_parse_rtpinfo):
	Remove unused code.

160
161
162
163
164
165
166
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Relax the audio/mpeg caps again and add FIXME: comment.

167
168
169
170
171
172
173
174
175
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	  More sanity check for the header fields. Fix type for 'rate' header
	  field.

176
177
178
179
180
181
182
183
184
185
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
	(gst_icydemux_unicodify):
	  If the metadata strings we get in the stream are not UTF-8, try to
	  interpret them according to the character encodings specified in the
	  GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
	  only fall back to locale/ISO-8859-1 if those aren't set or don't
	  work. Should fix #428901.

186
187
188
189
190
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c:
	Use the proper sync word for SPS and PPS.

191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
2007-04-12  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/rtp/Makefile.am:
	* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
	  fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
	* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
	  Add a simple hashing implementation that we can use to generate
	  a 24-bit ident value based on the codebooks for vorbis and theora.
	* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
	  gst_rtp_theora_pay_handle_buffer):
	* gst/rtp/gstrtpvorbisdepay.c
	  (gst_rtp_vorbis_depay_parse_configuration,
	  gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
	  gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
	  gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
	  Use the hashing function, ensuring that the same codebooks result
	  in the same ident and thus the same SDP description.
	  Various log fixes/changes.

211
212
213
214
215
216
217
218
219
2007-04-12  Wim Taymans  <wim@fluendo.com>

	Patch by: jerry tan <jerry dot tan at sun dot com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
	application's responsibility to make sure it open the device once.
	Remove a careless error if AUDIODEV is set. Fixes #392620.

220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
	* gst/rtsp/gstrtpdec.h:
	Make backward compat with rtpbin by adding the request-pt-map signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams):
	* gst/rtsp/gstrtspsrc.h:
	Implement request-pt-map signals instead of setting caps on the buffers
	for the session manager.

236
237
238
239
240
241
2007-04-11  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudp.c: (plugin_init):
	Register GstNetBuffer in plugin_init so that the type can be used from
	multiple threads without races.

242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
2007-04-10  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	Fix depayloader clock_rate and some cleanups.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	* gst/rtp/gstrtph264depay.h:
	Don't push codec_data in the adapter because it might get flushed when
	we get a discont.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Handle multiple AU per packet.

	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
	(gst_rtp_sv3v_depay_plugin_init):
	Disable rank, this one does not work.
	Remove timestamping, base class does that.

262
263
264
265
266
267
268
269
270
271
2007-04-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	  limit caps to the formats we announce in the template

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
	  fix some crashers/asserts when dealing with broken files

272
273
274
275
276
277
278
279
280
281
282
283
284
285
2007-04-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
	(gst_rtp_speex_depay_setcaps):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
	Fix some compiler warnings. Fixes #428182.

286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
2007-04-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
	(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_finalize),
	(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
	(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
	(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
	(create_rtcp), (gst_rtp_dec_request_new_pad),
	(gst_rtp_dec_release_pad):
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/gstrtsp.c: (plugin_init):
	Morph RTPDec into something compatible with RTPBin as a fallback.
	Various other style fixes.

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
	(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
	(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
	(new_session_pad), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Implement RTPBin session manager handling.
	Don't try to add empty properties to caps.
	Implement fallback session manager, handling.
	Don't combine errors from RTCP streams, just ignore them.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
	* gst/rtsp/rtsptransport.h:
	Implement fallback session manager.
	Make RTPBin the default one when available.

321
322
323
324
325
326
2007-04-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
	This element is ready to be autoplugged.

327
328
329
330
331
332
333
2007-04-05  Julien MOUTTE  <julien@moutte.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	Don't leave the offsets defined by upstream element on the
	compressed data buffer we are pushing downstream. Make them
	GST_BUFFER_OFFSET_NONE.

334
335
336
337
338
339
340
341
342
343
344
345
346
2007-04-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/README:
	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
	  Don't abort on out-of-memory. Use stream-nr as unsigned integer only.

Wim Taymans's avatar
Wim Taymans committed
347
348
349
350
351
2007-04-03  Wim Taymans  <wim@fluendo.com>

	* gst/smpte/barboxwipes.c:
	Fix error as spotted by Snaik <snaik32 at gmail dot com>

352
353
354
355
356
357
358
2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Support audio/x-raw-float in wav files. This only works with
	plugins-base CVS, using an older version doesn't have any
	disadvantages though.

359
360
361
362
363
364
365
366
367
2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Revert last change as we don't want plugins-good to depend on
	plugins-base CVS now.

368
369
370
371
372
373
374
375
376
377
378
379
380
381
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Require gst-plugins-base CVS for audioconvert with non-native
	float support and width/depth fix in libgstriff.

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Don't swap the floats ourself if they're not in native endianness.
	Instead let audioconvert handle this. Fixes #339838.

382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
	(gst_rtp_h263p_depay_change_state):
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	(gst_rtp_h264_depay_change_state):
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Flush adapter on disconts.

397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
	Use more efficient adapter and rtpbuffer methods when possible.

416
417
418
419
420
421
422
423
424
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps):
	Correctly handle width!=depth input.
	* gst/wavparse/gstwavparse.c:
	Already export in the caps that width==8 uses unsigned samples and
	everything else uses signed samples.

425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
2007-03-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

	* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
	(gst_dynudpsink_init), (gst_dynudpsink_set_property),
	(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
	(gst_dynudpsink_close):
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Rework the socket allocation a bit based on the sockfd argument so that
	it becomes usable.
	Add a closefd property to instruct the udp elements to close the custom
	file descriptors when going to READY. Fixes #423304.
	API:GstUDPSrc::closefd property
	API:GstDynUDPSink::closefd property

445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
2007-03-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
	(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
	(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
	(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
	(gst_rtp_h264_pay_plugin_init):
	* gst/rtp/gstrtph264pay.h:
	Added H264 payloader. Fixes #423782.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	Small fixes.

464
465
466
467
468
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Actually support depths from 1 to 32, not only 8 to 32.

469
470
471
472
473
474
2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Add support for wav files containing audio/x-raw-int with random
	depths between 1 and 32 bits.

475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
2007-03-28  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
	(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
	(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
	(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
	(gst_rtp_mp4a_depay_get_property),
	(gst_rtp_mp4a_depay_change_state),
	(gst_rtp_mp4a_depay_plugin_init):
	* gst/rtp/gstrtpmp4adepay.h:
	Added MP4A-LATM depayloader. Fixes #417792.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	Fixup depayloader, setting codec_data, using more efficient adaptor and
	rtpbuffer handling.

	* gst/rtsp/URLS:
	Add url to test above.

499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
2007-03-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
	(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Handle default clock-rates for static payload types, rearrange stuff so
	that the rtpmap field in the sdp can override the defaults.
	Parse RTP-Info field to get the seqnum and timebase fields that should
	go in the caps.
	Delay configuring caps after we got the RTP-Info from the PLAY reply from
	the server. 

516
517
518
519
520
521
522
523
2007-03-22  Wim Taymans  <wim@fluendo.com>

	Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Accept complex pipeline descriptions as an audio profile instead of just
	a single element. Fixes #420658.

524
525
526
527
528
529
2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
	  Rename registered type in preparation of GstTagDemux moving to
	  -base at some point in the future.

530
531
532
533
534
535
2007-03-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Streaming mode fixes: don't unref buffer we don't own any longer;
	  remove bogus adapter flush. Fixes #419338.

536
537
538
539
540
541
2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS: Change the format to key/value, add a bunch of
	  information, remove a bunch of requirements that are for
	  other GStreamer packages.

542
543
544
545
546
2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS: Fix a few things.  This file really needs a
	good once-over.

547
548
549
550
551
2007-03-15  Edward Hervey  <edward@fluendo.com>

	* sys/Makefile.am:
	Don't forget to distribute the sys/osxaudio/ directory.

552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
2007-03-15  Edward Hervey  <edward@fluendo.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudiosink.c:
	(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
	(gst_osx_audio_sink_getcaps),
	(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
	* sys/osxaudio/gstosxaudiosrc.c:
	(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
	(gst_osx_audio_src_create_ringbuffer):
	* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
	(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
	(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
	(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
	* sys/osxaudio/gstosxringbuffer.h:
	Activate osxaudio in gst-plugins-good with proper build setup.
	Add inlined documentation.
	Fix debug statements
	Fix ringbuffer when pausing.
	Fixes #323471

576
577
578
579
580
581
582
2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	Ported mulaw and alaw payloaders to use new base class

583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
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/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update translations.

599
600
601
602
603
2007-03-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix string replace error (AG_AG_GST_* => AG_GST_*).

604
605
606
607
608
609
2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking,
	  and SEEK_CUR+SEEK_END here as well.

610
611
612
613
614
615
2007-03-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking, 
	  and SEEK_CUR+SEEK_END.

616
617
618
619
620
621
622
623
624
2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
	  the image format a variable-length NUL-terminated string; in
	  versions before that the image format is a fixed-length string of
	  3 characters (see #348644 for a sample tag).
	  Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.

625
626
627
628
629
630
631
632
633
2007-03-10  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add new project files to MANIFEST.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.
	
634
635
636
637
638
639
640
641
642
2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
	(gst_avi_demux_parse_index):
	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Printf format fixes; also add some missing quotes in translated
	  strings. Fixes #416728 and #416727.

643
644
645
646
647
648
649
650
651
2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
	  Tim and I can't think of any reason the child audio sink needs to 
	  be set back to NULL after successfully determining that it can 
	  reach READY - it gets immediately set back to READY by the caller
	  anyway, causing an unnecessary close/open of any audio devices
	  involved.

652
653
654
655
656
657
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* po/LINGUAS:
	* po/ja.po:
	  Add ja.po file from #377306.

658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/sunaudio/gstsunaudio.c: (plugin_init):
	* sys/sunaudio/gstsunaudiomixertrack.c:
	(gst_sunaudiomixer_track_new):
	  Actually translate sunaudio mixer track labels instead of just
	  marking the strings as translatable (#377306); clean up weird
	  label string mapping code that serves no apparent purpose. Also
	  set the 'untranslated-label' property when creating mixer tracks
	  if the GstMixerTrack base class supports this.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/sunaudio.c: (GST_START_TEST),
	(sunaudio_suite):
	  Very minimalistic unit test for sunaudiomixer element (compiles, but not
	  actually tested on a system where sunaudiomixer is available).

676
677
678
679
680
2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Re-enable the states test and see if it works on the buildbots.

681
682
683
684
685
686
687
688
689
690
691
692
693
2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
	(gst_dvdec_src_negotiate), (gst_dvdec_chain),
	(gst_dvdec_change_state):
	* ext/dv/gstdvdec.h:
	Infer pixel-aspect-ratio from the video frame format if it isn't
	provided by the container, as happens when playing DV from AVI
	or Quicktime containers.

	Patch by: Wim Taymans <wim@fluendo.com>
	Fixes #380944

694
695
696
697
698
699
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	When activated, remove the udpsrc timeout, we have dataflow and timeouts
	will later be handled by the jitterbuffer.

700
701
702
703
704
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* ext/taglib/gstid3v2mux.cc:
	Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.
Jan Schmidt's avatar
Jan Schmidt committed
705
706
	
	Patch by: Alex Lancaster <alexl at users sourceforge net>
707

708
709
710
711
712
713
714
715
2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix stream position reporting after a seek. Fixes #416445.

716
717
718
719
720
721
722
723
724
725
2007-03-08  Wim Taymans  <wim@fluendo.com>

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_chain):
	Make avidemux accept optional header chunks in any order.
	Fixes #415446.

726
727
728
729
730
731
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable the states check until the remaining Valgrind errors
	are fixed or suppressed.

732
733
734
735
736
2007-03-08  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/.cvsignore:
	  Add audiodynamic check to .cvsignore

737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
2007-03-08  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_characteristics_get_type),
	(gst_audio_dynamic_mode_get_type),
	(gst_audio_dynamic_set_process_function),
	(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
	(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
	(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
	(gst_audio_dynamic_transform_hard_knee_compressor_int),
	(gst_audio_dynamic_transform_hard_knee_compressor_float),
	(gst_audio_dynamic_transform_soft_knee_compressor_int),
	(gst_audio_dynamic_transform_soft_knee_compressor_float),
	(gst_audio_dynamic_transform_hard_knee_expander_int),
	(gst_audio_dynamic_transform_hard_knee_expander_float),
	(gst_audio_dynamic_transform_soft_knee_expander_int),
	(gst_audio_dynamic_transform_soft_knee_expander_float),
	(gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new audiodynamic element which can act as a compressor or
	expander. Supported are hard-knee and soft-knee operation modes with
	user-specified ratio and threshold.
	Attack and release parameters are not yet implemented but will follow.
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Integrate audiodynamic into the docs.
	* tests/check/Makefile.am:
	* tests/check/elements/audiodynamic.c: (setup_dynamic),
	(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
	Add unit test for audiodynamic.

775
776
777
778
779
2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
	Free handles that we allocated when exiting via the error paths.

780
781
782
783
784
785
786
787
2007-03-07  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_class_init),
	(gst_level_set_caps), (gst_level_start), (gst_level_event),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	  Resolve message timestamps against the playback segment.

788
789
790
791
792
2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
	(gst_id3demux_sink_activate):
	  Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
793
794
795
	  caps passed to it (previously one code path assumed it took ownership
	  while another one assumed it didn't, while in fact it sometimes did and
	  sometimes didn't ...).
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812

	* configure.ac:
	* tests/files/Makefile.am:
	* tests/files/id3-407349-1.tag:
	* tests/files/id3-407349-2.tag:
	  Add directory where data for unit tests can be stored.

	* tests/Makefile.am:
	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
	(read_tags_from_file), (run_check_for_file),
	(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
	  Add unit test for id3demux, and in particular for bug #407349. Only
	  testing pull-mode for now; push mode doesn't work yet because the test
	  files are smaller than ID3_TYPE_FIND_MIN_SIZE.

813
814
815
816
817
2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Add missing backslash at end of line.

Jan Schmidt's avatar
Jan Schmidt committed
818
819
820
821
2007-03-06  Jan Schmidt  <thaytan@mad.scientist.com>

	Trigger rebuild.

822
823
824
825
826
827
828
829
830
831
832
833
2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	(parse_obsolete_tdat_frame):
	  Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
	  the four-digit number will be interpreted as a year, whereas it is
	  month and day in DDMM format. Instead, parse TDAT frames and fix up
	  the date in the GST_TAG_DATE tag later if we also extracted a year.
	  Fixes #407349.

834
835
836
837
838
839
840
841
2007-03-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid):
	Fix up the dispose logic so it doesn't leak, and fix setting of 
	the child state so that we don't set a child to our current state 
	just as we are changing it to something else.

842
843
844
845
846
847
848
849
2007-03-06  Wim Taymans  <wim@fluendo.com>

	* gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
	(gst_goom_chain):
	* gst/goom/gstgoom.h:
	Document, fix and improve goom adapter behaviour.
	Fixes #407006.

850
851
852
853
854
2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/esd/esdsink.c: (gst_esdsink_open):
	Unref static pad template after using it.

855
856
857
858
859
860
2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid):
	Fix up the reference counting of the child elements.

861
862
863
864
865
866
867
2007-03-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	Fix encoding-name case.

868
869
870
871
872
873
874
875
876
877
878
879
2007-03-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
	(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
	(gst_rtp_speex_depay_process):
	* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
	(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
	(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
	(gst_rtp_speex_pay_change_state):
	* gst/rtp/gstrtpspeexpay.h:
	Fix speex (de)payloader. Fixes #358040.

880
881
882
883
884
885
886
887
2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
	(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
	Install fakesink in NULL by fixing some broken logic. This obviates
	the need to manually set _IS_SINK.
	Add some comments and remove a little cruft while I'm at it.

888
889
890
891
892
2007-03-05  Wim Taymans  <wim@fluendo.com>

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
	Mark us as a sink when we have no fakesink in NULL. Fixes #414887.

Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
893
894
895
896
897
2007-03-04  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Update.

898
899
900
901
902
903
904
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Gah! Also disable gconfvideosink from the tests, otherwise
	it will instantiate autovideosink, and dfbvideosink and
	leak on the buildbots.

905
906
907
908
909
910
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
	(gst_cdio_cdda_src_finalize):
	Make sure we always destroy our libcdio handle.

911
912
913
914
915
916
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable autovideosink so the buildbots don't barf over memory
	leaked in the directfb sink.

917
918
919
920
921
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose):
	Chain up in dispose

922
923
924
925
926
927
928
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	(gst_multipart_find_pad_by_mime):
	Use gst_pad_new_from_static_template instead of
	static_pad_template_get+pad_new.

929
930
931
932
933
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_create):
	Catch the case where no clock has been set.

934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
	(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
	* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
	(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
	(gst_gconf_audio_src_finalize), (do_toggle_element):
	* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
	(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
	(do_toggle_element):
	* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
	(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
	(gst_gconf_video_src_finalize), (do_toggle_element):
	* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
	(gst_switch_sink_reset), (gst_switch_sink_set_child):
	* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
	* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
	* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	(gst_shout2send_init), (gst_shout2send_finalize):
	* gst/debug/testplugin.c: (gst_test_class_init),
	(gst_test_finalize):
	* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
	(gst_flxdec_dispose):
	* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize):
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
	* gst/rtsp/rtspextwms.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
	(gst_smpte_finalize):
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
	* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
	(gst_udpsink_finalize):
	* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
	(gst_wavparse_sink_activate):
	* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
	* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
	(gst_oss_src_finalize):
	* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_finalize):
	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):

	Fix a bunch of leaks shown by the newly-added states test.

982
983
984
985
986
987
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/dv/gstdvdec.c: (gst_dvdec_init):
	Use gst_pad_new_from_static_template instead of 
	static_pad_template_get+pad_new.

988
989
990
991
992
993
994
995
2007-03-03  Tim-Philipp Müller  <tim at centricular dot net>

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

	* ext/libcaca/Makefile.am:
	* gst/debug/Makefile.am:
	  Don't mix tabs and spaces (#414168).

996
997
998
999
1000
2007-03-02  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/generic/.cvsignore:
	  Ignore files to please buildbot.

1001
1002
1003
1004
1005
1006
1007
2007-03-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Unbreak my previous commit (swapped nominator & denominator). Tim,
	  thanks for spotting.

1008
1009
1010
1011
1012
1013
1014
1015
2007-03-02  Wim Taymans  <wim@fluendo.com>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
	(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
	(gst_cdio_cdda_src_finalize):
	Small code cleanups.
	Don't use pad_alloc as the base class cannot deal with the error codes.

Wim Taymans's avatar
Wim Taymans committed
1016
1017
1018
1019
1020
1021
2007-03-02  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create):
	Fix doc.

1022
1023
1024
1025
1026
1027
1028
1029
1030
2007-03-02  Stefan Kost  <ensonic@users.sf.net>

	Patch by: René Stadler <mail@renestadler.de>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Handle rounding better to not drop last sample frame. Fixes #356692

1031
1032
1033
1034
1035
1036
2007-03-02  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable cacasink from the states check too - it also calls exit(1)
	on us when it can't find a terminal to talk to.

1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
2007-03-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property):
	* gst/udp/gstudpsrc.h:
	Add support to strip proprietary headers. Fixes #350296.

1047
1048
1049
1050
1051
2007-03-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	Fix compilation.

1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
2007-03-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init),
	(gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process),
	(gst_rtp_mp2t_depay_set_property),
	(gst_rtp_mp2t_depay_get_property):
	* gst/rtp/gstrtpmp2tdepay.h:
	Add support to strip off proprietary headers. Fixes #350278.

Wim Taymans's avatar
Wim Taymans committed
1063
1064
1065
1066
1067
2007-03-02  Wim Taymans  <wim@fluendo.com>

	* ext/hal/hal.c:
	Fix compilation.

1068
1069
1070
1071
1072
1073
1074
1075
2007-03-02  Wim Taymans  <wim@fluendo.com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init),
	(gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property),
	(gst_sunaudiosrc_open):
	* sys/sunaudio/gstsunaudiosrc.h:
	Remove device-name from GstSunAudioSrc. Fixes #412597.

1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
2007-03-01  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/hal/gsthalaudiosink.c: (do_toggle_element):
	* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	Having NULL as UDI previously selected the default sink/src. Change
	this back but mention it in the debug output.
	* ext/hal/hal.c: (gst_hal_get_alsa_element),
	(gst_hal_get_oss_element), (gst_hal_get_string),
	(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
	(gst_hal_get_audio_src):
	* ext/hal/hal.h:
	Refactor a bit, check all error conditions, greatly improve debugging
	and fix some possible memory leaks. Also implement OSS support
	and allow specifying an UDI that points to a real device. For this the
	child device which supports ALSA (preferred) or OSS is used.
	As a side effect this makes it impossible now to get a alsasink in
	halaudiosrc and a alsasrc in halaudiosink.

1094
1095
1096
1097
1098
1099
1100
2007-03-01  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
	(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
	Errors from the udp sources are not fatal unless all of them are in
	error.

1101
1102
1103
1104
1105
1106
2007-03-01  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable aasink in the states test. I suspect this is the element that
	is calling exit(1) when it can't proceed.

1107
1108
1109
1110
1111
1112
2007-03-01  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Draw plugins in from the build tree sys/ dir, rather than picking
	up the already installed versions.

1113
1114
1115
1116
1117
2007-03-01  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
	Error out correctly when getting xcontext fails.

1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
2007-03-01  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
	Make state change to PAUSED NO_PREROLL because that's what it will be in
	the future and rtspsrc relies on it.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_change_state):
	Don't error out when we don't get an error from the state change
	function.

1129
1130
1131
1132
1133
1134
1135
1136
1137
1138
1139
2007-03-01  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/hal/gsthalaudiosink.c: (do_toggle_element):
	* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	  Check if the device UDI is set before trying to query HAL
	  about it and give a useful error message if it wasn't set.
	* ext/hal/hal.c: (gst_hal_get_string):
	  Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
	  gives an assertion failure in D-Bus when running with
	  DBUS_FATAL_WARNINGS=1.

1140
1141
1142
1143
1144
2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  Convert to new AG_GST style.

1145
1146
1147
1148
1149
1150
2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/Makefile.am:
	* tests/check/generic/states.c: (GST_START_TEST), (states_suite):
	  add test for states

1151
1152
1153
1154
1155
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/.cvsignore:
	Add new videofilter check to .cvsignore.

1156
1157
1158
1159
1160
1161
1162
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop), (gst_avi_demux_chain):
	Fix combined flow return. Fixes #412608.

1163
1164
1165
1166
1167
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* gst/videofilter/Makefile.am:
	Dist header..

1168
1169
1170
1171
1172
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* gst/videofilter/gstgamma.h:
	Add header too.

1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
1189
1190
2007-02-28  Wim Taymans  <wim@fluendo.com>

	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
	(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
	(gst_gamma_get_property), (gst_gamma_calculate_tables),
	(oil_tablelookup_u8), (gst_gamma_set_caps),
	(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
	Port gamma filter to 0.10. Fixes #412704.

	* tests/check/Makefile.am:
	* tests/check/elements/videofilter.c: (setup_filter),
	(cleanup_filter), (check_filter), (GST_START_TEST),
	(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
	Add unit tests for videofilters.

1191
1192
1193
1194
1195
1196
1197
1198
2007-02-28  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add another interesting test url.

	* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
	Don't allow getting header fields from data packets.

1199
1200
1201
1202
1203
1204
1205
1206
2007-02-28  Michael Smith  <msmith@fluendo.com>

	* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	(gst_shout2send_init), (gst_shout2send_start),
	(gst_shout2send_set_property), (gst_shout2send_get_property):
	* ext/shout2/gstshout2.h:
	  Add a property for username.

1207
1208
1209
1210
2007-02-27  Christian Schallerr <christian@fluendo.com>

	* sys/osxaudio: Add Pioneers of the inevitable to the copyright list

1211
1212
1213
1214
1215
2007-02-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/Makefile.am:
	Fix make check too.

1216
1217
1218
1219
1220
1221
2007-02-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/base64.c: (util_base64_encode):
	* gst/rtsp/base64.h:
	Commit missing files for base64 encoding.

1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235
1236
1237
1238
1239
2007-02-24  Tim-Philipp Müller  <tim at centricular dot net>

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

	* configure.ac:
	* ext/annodex/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/speex/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debug/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/level/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/videofilter/Makefile.am:
	  Fix build with LDFLAGS='-Wl,-z,defs' (#410997)

1240
1241
1242
1243
1244
1245
1246
1247
1248
2007-02-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/rtspconnection.c: (append_auth_header),
	(rtsp_connection_send), (rtsp_connection_set_auth):
	g_base64_encode is a GLib 2.12 function. Use an equivalent taken
	from icecast to replace it. Relicensed from GPL courtesy of Mike
	Smith.

1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
1262
1263
1264
1265
1266
1267
1268
1269
2007-02-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
	(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(append_auth_header), (rtsp_connection_send),
	(rtsp_connection_free), (rtsp_connection_set_auth):
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	* gst/rtsp/rtspurl.h:

	Implement simple Basic Authentication support so that urls like
	rtsp://user:pass@hostname/rtspstream work on hosts that require
	authentication.

1270
>>>>>>> 1.2755
1271
1272
1273
1274
1275
1276
1277
2007-02-22  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	Fix segfault when oppening a radio device.
	
1278
1279
1280
1281
1282
1283
1284
1285
2007-02-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_set_caps),
	(gst_level_transform_ip):
	* sys/v4l2/README:
	* tests/check/elements/level.c: (GST_START_TEST):
	  Fix level for multi-channel case.

1286
1287
1288
1289
1290
1291
1292
1293
2007-02-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	  Use function pointer for process function and add process functions
	  for float audio.

1294
1295
1296
1297
1298
1299
1300
1301
2007-02-19  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init):
	  Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
	  fixes #407369

1302
1303
1304
1305
1306
1307
1308
1309
1310
1311
1312
2007-02-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
	(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
	(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
	(gst_rtp_mp2t_pay_plugin_init):
	* gst/rtp/gstrtpmp2tpay.h:
	Added simple mpeg transport stream payloader.

1313
1314
1315
1316
1317
1318
1319
1320
1321
1322
2007-02-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add example H264 rtsp url.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	Don't convert values to lowercase or we might mess up base64 encoded
	properties.

1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
2007-02-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Fix case of string params.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	Fix depayloader, support more packet types.
	Add sync codes to make sure the packetizer can do its job.

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	Fix caps case again.

1338
1339
1340
1341
1342
2007-02-15  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	Set right caps on output buffers.

1343
1344
1345
1346
1347
1348
1349
2007-02-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/sdpmessage.c: (sdp_parse_line):
	As spotted by: Peter Kjellerstedt  <pkj at axis com>:
	Clear stack allocated SDPMedia struct before calling _init() on it.
	Clarify this in the docs as well.

1350
1351
1352
1353
1354
1355
1356
2007-02-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
	(do_change_child):
	Don't reset the profile when going switching states, as it makes
	the element non-reusable.

1357
1358
1359
1360
1361
1362
1363
1364
1365
1366
1367
1368
2007-02-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
	(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
	(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
	(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
	(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
	(sdp_parse_line):
	* gst/rtsp/sdpmessage.h:
	Based on patch by: jp.liu <jp_liu at astrocom dot cn>
	Fix memory management of SDP messages. Fixes #407793.

1369
1370
1371
1372
1373
1374
1375
1376
2007-02-14  Stefan Kost  <ensonic@users.sf.net>

	Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>

	* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
	Allow muxing video/x-h264 (was already in the caps). Fixes #407780.

2007-02-14  Wim Taymans  <wim@fluendo.com>
1377
1378
1379
1380
1381
1382

	Patch by: jp.liu <jp_liu at astrocom dot cn>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of password field in url. Fixes #407797.

1383
2007-02-14  Wim Taymans  <wim@fluendo.com>
1384
1385
1386
1387
1388
1389
1390
1391
1392
1393
1394
1395
1396
1397
1398
1399
1400
1401
1402
1403
1404
1405

	* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
	(gst_wavparse_reset), (gst_wavparse_init),
	(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
	(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
	(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	(gst_wavparse_loop), (gst_wavparse_chain),
	(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
	(plugin_init):
	* gst/wavparse/gstwavparse.h:
	Update docs.
	Use boilerplate.
	Various code cleanups.
	When the bitrate is not known (bps == 0 or compressed formats) let
	downstream element guestimate the duration and position and don't
	generate timestamps or durations. Fixes #405213.
	Fix EOS and ERROR conditions in chain mode, we just need to forward the
	error flowreturn upstream.

1406
1407
1408
1409
1410
1411
1412
1413
1414
1415
1416
1417
1418
1419
1420
1421
1422
1423
1424
1425
1426
1427
1428
1429
1430
1431
1432
1433
1434
1435
1436
1437
1438
2007-02-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/Makefile.am:
	* ext/gconf/gconf.c: (gst_gconf_get_string),
	(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
	(gst_gconf_render_bin_with_default):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
	(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
	(gst_gconf_audio_sink_dispose), (do_change_child),
	(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
	(cb_change_child), (gst_gconf_audio_sink_change_state):
	* ext/gconf/gstgconfaudiosink.h:
	* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
	(gst_switch_sink_class_init), (gst_switch_sink_reset),
	(gst_switch_sink_init), (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
	(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
	(gst_switch_sink_get_property), (gst_switch_sink_change_state):
	* ext/gconf/gstswitchsink.h:
	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
	(gst_auto_audio_sink_detect):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
	(gst_auto_video_sink_detect):
	Re-factor the gconfaudiosink into a "GstSwitchSink" base class
	and a child that implements the GConf key monitoring. The end goal of
	this is an audio sink that can be changed on the fly, but at the 
	moment it still only changes on the next READY transition.

1439
1440
1441
1442
1443
1444
1445
1446
1447
1448
1449
2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif

1450
1451
1452
1453
1454
1455
2007-02-13  Stefan Kost  <ensonic@users.sf.net>

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

1456
1457
1458
1459
1460
1461
1462
2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/monoscope/Makefile.am:
	* gst/monoscope/gstmonoscope.c:
	  Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
	  (but no LIBS, since we only use defines from the headers).

1463
1464
1465
1466
1467
1468
1469
1470
1471
1472
2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
	(gst_wavparse_stream_data):
	  Fix massive memory leak when operating in streaming mode due to
	  GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
	  Fixes #407057.

1473
1474
1475
1476
1477
1478
1479
1480
1481
1482
1483
1484
1485
1486
1487
1488
1489
2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	* gst/avi/gstavidemux.h:
	  Save some memory (8%) by repacking the index entry structure (more to
	  come). Add more FIXMEs to questionable parts.

1490
1491
1492
1493
1494
1495
1496
1497
1498
2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
	(gst_v4l2src_get_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init):
	  More FIXME comments and messaging changes.

1499
1500
1501
1502
1503
1504
1505
1506
1507
1508
1509
1510
1511
1512
2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	(gst_goom_change_state):
	* gst/goom/gstgoom.h:
	  Improved docs and use GST_DEBUG_FUNCPTR.

	* gst/level/gstlevel.c: (gst_level_class_init):
	  Use GST_DEBUG_FUNCPTR.

	* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
	(gst_monoscope_chain), (gst_monoscope_change_state):
	  Improved docs source cleanups.

1513
1514
1515
1516
1517
1518
1519
1520
1521
1522
1523
2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/Makefile.am:
	* gst/debug/gstdebug.c: (plugin_init):
	* gst/debug/gstpushfilesrc.c:
	* gst/debug/gstpushfilesrc.h:
	  Add code for a pushfilesrc element that implements a pushfile:// URI
	  handler, to make debugging push-mode operation of demuxer/decoders
	  that support both easier in connection with seek/playbin/etc.
	  The element isn't registered at the moment.

1524
1525
1526
1527
1528
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
1541
1542
1543
1544
1545
1546
1547
1548
1549
1550
1551
1552
1553
1554
1555
1556
1557
2007-02-11  Sébastien Moutte  <sebastien@moutte.net>

	* gst/avi/gstavimux.c:
	  Comment a #if 0 in caps template definition as VS6 seems to 
	do not support it.
	* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
	  Use gst_guint64_to_gdouble for conversion.
	* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
	  Move variables declaration before the first instruction.
	* gst/rtsp/rtspdefs.c:(rtsp_strresult):
	  Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
	  And don't include netdb.h for G_OS_WIN32
	* gst/rtsp/sdpmessage.c:(sdp_parse_line):
	  This initialization SDPMedia nmedia = {.media = NULL }; is not supported
	  by VS6 then use an other way to initialize SDPMedia structure.
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstdynudpnetutils.h:
	  Do not include <sys/time.h> for G_OS_WIN32
	* gst/udp/gstudpsrc.c:
	  Define socklen_t as int for G_OS_WIN32
	* win/common/config.h.in:
	  Undef HAVE_NETINET_IN_H
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstautogen.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstudp.dsp:
	  Add and update project files.
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	  Add a copy of udp enumtypes to win32/common as in core 
	  and base.
	
1558
1559
1560
1561
1562
1563
1564
1565
1566
1567
2007-02-11  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Activate monoscope when building with --enable-experimental. Fix
	  --enable-external configure switch description.

	* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
	  Help gst-indent.

1568
1569
1570
1571
1572
1573
2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  Explicitly cast result of pointer arithmetic to integer in order to
	  avoid compiler warnings on some 64-bit systems. Should fix #406018.

1574
1575
1576
1577
1578
2007-02-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c:
	  Some more docs.

1579
1580
1581
1582
1583
1584
1585
1586
2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/inspect/plugin-rtp.xml:
	  Update for new elements.

	* gst/debug/progressreport.h:
	  Commit newly-created header file as well.

1587
1588
1589
1590
1591
1592
1593
1594
1595
1596
1597
1598
2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* gst/debug/Makefile.am:
	* gst/debug/progressreport.c: (gst_progress_report_post_progress),
	(gst_progress_report_do_query), (gst_progress_report_report):
	  Make progressreport element post messages with the current progress
	  on the bus. Also add some basic docs for it.

1599
1600
1601
1602
1603
1604
1605
2007-01-30  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/hal/hal.c: (gst_hal_get_string):
	* ext/hal/hal.h:
	  Some small cleanups; deal with errors when parsing the HAL ALSA
	  capabilities a bit better.

1606
1607
1608
1609
1610
2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Let's try this again and use the right cast this time.

1611
1612
1613
1614
1615
1616
1617
2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Add cast to avoid compiler warnings with older GLib versions
	  where the nick/name members in GEnumValue are not declared as
	  constant strings.

1618
1619
1620
1621
1622
1623
1624
1625
1626
1627
1628
1629
1630
2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
	(gst_gconf_render_bin_from_key),
	(gst_gconf_get_default_audio_sink):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
	(do_toggle_element), (gst_gconf_audio_sink_set_property),
	(gst_gconf_audio_sink_get_property):
	  In gconfaudiosink, get the right key as the old key in do_toggle
	  (ie. one dependent on the profile selected). Log some more stuff so
	  we can see what's actually going on.

1631
1632
1633
1634
1635
1636
1637
1638
1639
1640
1641
1642
1643
1644
1645
2007-02-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
	(gst_audio_amplify_class_init), (gst_audio_amplify_init),
	(gst_audio_amplify_set_process_function),
	(gst_audio_amplify_setup):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_setup):
	* gst/audiofx/audioinvert.h:
	Some small cleanups and port both elements to the new GstAudioFilter
	base class to save a few lines of common code.
	* gst/audiofx/Makefile.am:
	Link against libgstaudio for the above changes

1646
1647
1648
1649
1650
2007-01-29  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/.cvsignore:
	Some more ignores.

1651
1652
1653
1654
1655
1656
1657
1658
1659
2007-01-26  Wim Taymans  <wim@fluendo.com>

	Patch by: charles <charlesg3 at gmail dot com>

	* ext/shout2/gstshout2.c: (gst_shout2send_init),
	(set_shout_metadata), (gst_shout2send_event):
	* ext/shout2/gstshout2.h:
	Properly handle tags in shout2send. Fixes #399825.

1660
1661
1662
1663
1664
1665
1666
2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_activate_streams):
	Convert SDP fields to upper/lowercase following the rules in the SDP to
	caps document. 

1667
1668
1669
1670
1671
1672
1673
1674
1675
1676
1677
1678
1679
1680
1681
1682
1683
1684
2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix case of encoding-name and key/value pairs to match the document.
	This is to make interoperation with SDP case-insensitive as required by
	the relevant RFCs.

1685
1686
1687
1688
1689
2007-01-25  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	Bump required -core/-base to CVS

1690
1691
1692
1693
1694
1695
1696
1697
2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
	(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
	* gst/rtp/gstrtpL16pay.h:
	Fill up to MTU using adapter.
	Timestamp rtp packets.

1698
1699
1700
1701
1702
1703
1704
2007-01-25  Edward Hervey  <edward@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
	Use G_GSIZE_FORMAT in print statements for portability.
	Fixes build on macosx.

1705
1706
1707
1708
1709
1710
1711
1712
1713
1714
1715
1716
1717
1718
1719
1720
1721
1722
1723
1724
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
	(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
	(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
	(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
	(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
	(gst_rtp_L16_depay_plugin_init):
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
	(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
	(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
	(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
	(gst_rtp_L16_pay_plugin_init):
	* gst/rtp/gstrtpL16pay.h:
	Port and enable raw audio payloader/depayloader. Needs a bit more work
	on the payloader side.

1725
1726
1727
1728
1729
1730
1731
1732
1733
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
	* gst/rtsp/gstrtspsrc.h:
	Only unblock the udp pads when we linked and activated them all.
	Fixes #395688.

1734
1735
1736
1737
1738
1739
1740
1741
1742
1743
1744
1745
1746
1747
1748
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
	(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
	(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
	(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
	(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
	* gst/rtp/gstrtpac3depay.h:
	Added simple AC3 depayloader (RFC 4184).

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	Fix a leak.

1749
1750
1751
1752
1753
1754
1755
1756
1757
1758
1759
1760
1761
1762
1763
1764
1765
1766
1767
1768
1769
1770
1771
1772
1773
1774
1775
1776
1777
1778
1779
1780
1781
1782
1783
1784
2007-01-24  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audioamplify.c:
	(gst_audio_amplify_clipping_method_get_type),
	(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
	(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
	(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
	(gst_audio_amplify_set_caps),
	(gst_audio_amplify_transform_int_clip),
	(gst_audio_amplify_transform_int_wrap_negative),
	(gst_audio_amplify_transform_int_wrap_positive),
	(gst_audio_amplify_transform_float_clip),
	(gst_audio_amplify_transform_float_wrap_negative),
	(gst_audio_amplify_transform_float_wrap_positive),
	(gst_audio_amplify_transform_ip):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new element "audioamplify". This allows scaling of raw audio
	samples, similar to the "volume" element, but provides different modes
	for clipping and allows unlimited amplification. It's mainly targeted
	for creative sound design and not as a replacement of the "volume"
	element. Fixes #397162
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for audioamplify and integrate them into the build system
	* tests/check/Makefile.am:
	* tests/check/elements/audioamplify.c: (setup_amplify),
	(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
	Add fairly extensive unit test suite for audioamplify

1785
1786
1787
1788
1789
1790
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
	Unblock pads after adding the pads to the element so that autopluggers
	get a change to link something. Possibly fixes #395688.

1791
1792
1793
1794
1795
1796
1797
1798
1799
1800
1801
1802
1803
1804
1805
1806
1807
1808
1809
1810
1811
1812
1813
1814
1815
1816
1817
1818
1819
1820
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
	(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
	(gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
	(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix caps with payload numbers.
	Add some fixed payload numbers to caps when possible.

1821
1822
1823
1824
1825
1826
1827
1828
1829
1830
1831
1832
1833
1834
1835
1836
1837
1838
1839
1840
1841
1842
1843
1844
1845
1846
1847
2007-01-23  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c: (plugin_init):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_init),
	(gst_audio_invert_set_property), (gst_audio_invert_get_property),
	(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
	(gst_audio_invert_transform_float),
	(gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Add new audiofx element "audioinvert". This element swaps the upper
	and lower half of samples and can be used for example for a
	wide-stereo effect. Fixes #396057
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for the audioinvert element and add them to the build system.
	* tests/check/Makefile.am:
	* tests/check/elements/audioinvert.c: (setup_invert),
	(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
	Add unit test suite for the audioinvert element.

1848
1849
1850
1851
1852
1853
1854
2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
	(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
	Parse config params as string and int.
	Parse and use AU header length

1855
1856
1857
1858
1859
1860
1861
1862
1863
1864
1865
1866
1867
2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
	(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
	* gst/smpte/gstmask.c: (_gst_mask_register):
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
	* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
	(gst_smpte_paint_triangle_clock):
	constify some static structs.
	Don't update the mask if nothing changed to the params.
	Make sure we never draw outside of the picture. Fixes #398325.

1868
1869
1870
1871
1872
1873
2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
	  Error out properly when pull_range fails while we're reading the
	  headers, instead of just pausing the task silently. Fixes #399338.

1874
1875
1876
1877
1878
1879
1880
1881
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	  Some more sanity checks to make sure the input formats match and the
	  input pads are actually negotiated, in case someone tries to feed
	  buffers from fakesrc or filesrc. Fixes #398299.
	  Also const-ify an array, just because we can.

1882
1883
1884
1885
1886
1887
1888
1889
1890
1891
2007-01-19  Edward Hervey  <edward@fluendo.com>

	* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
	Ignore previous commit, that was only valid for widths and heights
	that are multiples of 4.
	Copy over size/stride macros from jpegdec. This allows the element
	to work with any width,height...
	... but puts in evidence that the actual transformations only work
	with width/height that are multiples of 4.

1892
1893
1894
1895
1896
1897
1898
1899
1900
1901
2007-01-19  Edward Hervey  <edward@fluendo.com>

	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	Allocate buffers of the right size.
	The proper size of a I420 buffer in bytes is:
	
	    width * height * 3
	    ------------------
	            2

1902
1903
1904
1905
1906
1907
1908
2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_init):
	  Proxy getcaps on sink pads too, so that we either end up with the
	  same dimensions on all pads or error out if that's not possible
	  (seems to work even!). Fixes #398086, I think.

1909
1910
1911
1912
1913
1914
1915
1916
1917
1918
1919
1920
1921
1922
1923
1924
1925
1926
1927
1928
1929
1930
1931
1932
1933
1934
1935
1936
1937
1938
1939
1940
1941
1942
1943
1944
1945
1946
1947
1948
1949
1950
1951
1952
1953
1954
1955
1956
1957
1958
1959
1960
1961
1962
1963
1964
1965
1966
1967
1968
2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
	  fix integer properties with -1 as minimum value.

	* docs/plugins/inspect/plugin-1394.xml:
	* 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-dv.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-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-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.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-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 to CVS.

1969
1970
2007-01-18  Stefan Kost  <ensonic@users.sf.net>

1971
1972
	Patch by: Sebastian Dröge <slomo circular-chaos org>

1973
1974
1975
	* gst/audiofx/audiopanorama.c:
	  Fix doc section name (Fixes #397946)

1976
1977
1978
1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
1989
1990
1991
1992
2007-01-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2object.c:
	(gst_v4l2_object_install_properties_helper),
	(gst_v4l2_object_set_property_helper),
	(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_init), (gst_v4l2src_set_property),
	(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
	(gst_v4l2src_capture_deinit):
	  Fix EIO handing when capturing. Add new property to specify the number of
	  buffers to enque (and remove the borked num-buffers usage).

1993
1994
1995
1996
1997
1998
1999
2000
2001
2007-01-16  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
	(gst_audio_panorama_set_process_function):
	  Use a function array for process methods, add more docs and define the
	  startindex of enums.

2002
2003
2004
2005
2006
2007
2008
2009
2010
2011
2012
2013
2014
2015
2016
2017
2018
2019
2020
2021
2022
2007-01-14  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
	(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
	(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
	(gst_avi_mux_riff_get_avi_header),
	(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
	(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
	(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
	(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
	(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
	(gst_avi_mux_change_state):
	* gst/avi/gstavimux.h:
	* tests/check/elements/avimux.c: (teardown_src_pad):
	  Add support for more than one audio stream; write better AVIX
	  header; refactor code a bit; don't announce vorbis caps on our audio
	  sink pads since we don't support it anyway. Closes #379298.

2023
2024
2025
2026
2027
2028
2029
2030
2031
2032
2033
2034
2035
2036
2037
2038
2039
2040
2041
2042
2043
2044
2007-01-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c:
	(gst_audio_panorama_method_get_type),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_process_function),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_int_simple),
	(gst_audio_panorama_transform_s2s_int_simple),
	(gst_audio_panorama_transform_m2s_float_simple),
	(gst_audio_panorama_transform_s2s_float_simple):
	* gst/audiofx/audiopanorama.h:
	  Add 'method' property and provide a simple (non-psychoacustic)
	  processing method (#394859).

	* tests/check/elements/audiopanorama.c: (GST_START_TEST),
	(panorama_suite):
	  Tests for new method.

2045
2046
2047
2048
2049
2050
2051
2007-01-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
	  Set correct caps on outgoing pulled buffers, or things blow up
	  after recent core changes.

2052
2053
2054
2055
2056
2057
2058
2059
2060
2061
2062
2007-01-11  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_init),
	(gst_multipart_mux_request_new_pad),
	(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Return FLOW errors ASAP. Fixes #394977.
	Misc cleanups.

2063
2064
2065
2066
2067
2068
2069
2007-01-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	Check for stream pad before activating. 

2070
2071
2072
2073
2074
2075
2076
2077
2078
2079
2080
2081
2082
2083
2084
2085
2086
2087
2088
2089
2090
2091
2092
2093
2094
2095
2096
2097
2098
2099
2100
2101
2102
2103
2104
2105
2106
2107
2108
2109
2110
2111
2112
2113
2114
2115
2116
2117
2118
2119
2120
2121
2122
2123
2124
2125
2126
2127
2007-01-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/COPYING.MIT:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
	(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_open), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_send), (read_line),
	(parse_request_line), (parse_line), (rtsp_connection_read),
	(rtsp_connection_close):
	* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
	(rtsp_method_as_text), (rtsp_header_as_text),
	(rtsp_status_as_text), (rtsp_find_header_field),
	(rtsp_find_method):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
	(rtsp_ext_wms_configure_stream):
	* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
	(rtsp_message_new_request), (rtsp_message_init_request),
	(rtsp_message_new_response), (rtsp_message_init_response),
	(rtsp_message_init_data), (rtsp_message_unset),
	(rtsp_message_free), (rtsp_message_add_header),
	(rtsp_message_get_header), (rtsp_message_set_body),
	(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
	(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
	(sdp_message_dump):
	Allow url to be NULL to be able to use it for server connections.
	Can now send responses as well as requests.
	No longer hangs in an endless loop if EOF is received.
	Can now convert a status code to a text string.
	Return RTSP_HDR_INVALID for unknown headers.
	Return RTSP_INVALID for unknown methods.
	Copy CSeq and Session headers from the request.
	Only free memory corresponding to the currently set message type.
	Added const to function arguments as appropriate.
	Avoid a compiler warning when initializing nmedia.
	Use guint rather than gint to avoid compiler warnings.
	Fix crasher in wms extension.
	Factor out stream setup from open_connection.
	Delay activation of streams when actual data is received from the
	server, this prepares us to do proper protocol switching.
	Added new license.
	Fixes #380895.


2128
2129
2130
2131
2132
2133
2134
2135
2007-01-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo ubuntu com>

	* docs/plugins/Makefile.am:
	* gst/audiofx/audiopanorama.c:
	  Some small docs fixes (#394851).

Wim Taymans's avatar
Wim Taymans committed
2136
2137
2138
2139
2140
2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c:
	Fix docs.

2141
2142
2143
2144
2145
2146
2147
2148
2149
2150
2151
2152
2153
2154
2155
2156
2157
2158
2159
2160
2161
2162
2163
2164
2165
2166
2167
2168
2169
2170
2171
2172
2173
2174
2175
2176
2177
2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
	(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
	(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
	(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
	(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
	* gst/rtp/gstrtpmpvdepay.h:
	  Added RFC 2250 MPEG Video Depayloader.

	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	Fix Header file. Small cleanups.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
	(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
	(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
	(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
	(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
	(gst_rtp_mp4v_depay_change_state):
	Remove usused code. Remove Adapter from state Change. Added debug.

	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
	(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
	(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpadepay.h:
	Subclass base depayloader.
	Added debug.
	Support static payload type assignment as well.

	* gst/rtp/gstrtpmpapay.c:
	Fix caps.

2178
2179
2180
2181
2182
2183
2184
2185
2186
2187
2188
2189
2190
2191
2192
2193
2194
2007-01-08  Tim-Philipp Müller  <tim at centricular dot net>

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

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/smokecodec.c:
	  These libjpeg callbacks should return a 'boolean' (unsigned char
	  apparently) and not a 'gboolean' (which maps to gint). Fixes
	  warnings when compiling with MingW (#393427).

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  Use ioctlsocket on win32.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Some printf format fixes for win32.

2195
2196
2197
2198
2199
2200
2201
2202
2203
2204
2007-01-07  Sébastien Moutte  <sebastien@moutte.net>

	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	  Use gst_guint64_to_gdouble for conversion.
	* win32/vs6/libgstmatroska.dsp:
	  Add zlib to the link.
	* win32/vs6/libgstvideobox.dsp:
	  Update liboil library name (project is linked to 
	  liboil-0.3-0.lib now).
	  
2205
2206
2207
2208
2209
2210
2007-01-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/Makefile.am:
	  If zlib is available and used, we must link it explicitly for
	  things to work on MingW (fixes #392855).

2211
2212
2213
2214
2215
2216
2007-01-04  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/esdsink.c: (gst_esdsink_delay):
	  Don't return bogus values when esd_get_delay() fails for some
	  reason (#392189).

2217
2218
2219
2220
2221
2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/gstximagesrc.c: (composite_pixel):
	  Fix presumably copy'n'pasto for 16bpp depth.

2222
2223
2224
2225
2226
2227
2228
2229
2230
2231
2232
2233
2234
2235
2236
2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-mux.c:
	(gst_matroska_mux_audio_pad_setcaps):
	  The "signed" field in audio caps is of boolean type, trying to use
	  gst_structure_get_int() to extract it will fail. Fixing this makes
	  matroskamux accept raw audio input (#387121) (use at your own risk
	  though, due to the matroska spec being not entirely useful in this
	  respect).
	  Also fix up raw audio structures in template caps so that they
	  represent what our setcaps function will actually accept, so that
	  converters know what to convert to.
	  Finally, don't fail if there isn't an "endianness" field in 8-bit
	  PCM caps.

2237
2238
2239
2240
2241
2242
2243
2244
2245
2246
2247
2006-12-22  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  reapply consistent pad (de)activation

Jan Schmidt's avatar
Jan Schmidt committed
2248
2249
2250
2251
2252
2253
2254
2255
2006-12-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.5 doap entry

2256
2257
2258
2259
2260
2261
2262
=== release 0.10.5 ===

2006-12-21  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.5, "The Path of Thorns"

2263
2264
2265
2266
2267
2268
2269
2270
2271
2272
2273
2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  revert my freeze breakage

2274
2275
2276
2277
2278
2279
2280
2281
2282
2283
2284
2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  consistent pad (de)activation

2285
2286
2287
2288
2289
2290
2006-12-18  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* ext/Makefile.am:
	Disable LADPSA, as it has moved to the -bad module for the duration.

2291
2292
2293
2294
2295
2296
2297
2006-12-18  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	(gst_signal_processor_event):
	Reset flow_state back to _OK after a flush stop so that we exit our
	error state after the flush. Fixes #374213

2298
2299
2300
2301
2302
2303
2006-12-16  David Schleef  <ds@schleef.org>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Decent effort at porting to 0.10.  Needs cleanup on OS/X.

2304
2305
2306
2307
2308
2309
2310
2311
2312
2006-12-16  David Schleef  <ds@schleef.org>

	Patch by: Vijay Santhanam <vijay santhanam gmail com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Preliminary patch for porting osxvideosink

2313
2314
2315
2316
2317
2318
2319
2320
2321
2322
2323
2324
2325
2326
2006-12-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
	(gst_videomixer_set_master_geometry),
	(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
	(gst_videomixer_reset), (gst_videomixer_init),
	(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
	(gst_videomixer_release_pad), (gst_videomixer_collected),
	(gst_videomixer_change_state):
	Introduce some locking around the videomixer state so that it does not
	crash when adding/removing pads. Fixes #383043.

2327
2328
2329
2330
2331
2332
2333
2006-12-16  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Make sure libcaca can actually be used instead of just checking for
	  /usr/bin/caca-config, so we don't wrongly try to build cacasink when
	  cross-compiling (fixes #384587).

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
2334
2335
2336
2337
2338
2339
2340
2006-12-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	* gst-plugins-good.doap:
	* gst-plugins-good.spec.in:
	  adding doap file

2341
2342
2343
2344
2345
2346
2347
2006-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  libflac-1.1.3 changed API again, but we can't build against it yet,
	  so make sure our check doesn't use libflac-1.1.3 and add a comment
	  to this effect.

2348
2349
2350
2351
2352
2353
2006-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/effectv/gstquark.c: (gst_quarktv_transform),
	(gst_quarktv_planetable_clear):
	  Add some NULL pointer checks (possibly related to #385623).

2354
2355
2356
2357
2358
2359
2360
2361
2362
2363
2364
2006-12-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
	(gst_tag_demux_chain):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  In streaming mode, if the first buffer we get doesn't have an
	  offset, fix it up to be 0, otherwise trimming won't work later on
	  and we'll be typefinding application/x-id3, which may result in
	  decodebin plugging an endless number of id3demux elements as a
	  consequence. Fixes #385031.
	  
2365
2366
2367
2368
2369
2370
2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Ignore the buffer_time the sound device reports. Turns out it is 
	  sometimes completely bogus and we're better off without it.

2371
2372
2373
2374
2375
2376
2377
2378
2379
2380
2381
2382
2006-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	(gst_matroska_demux_video_caps):
	* gst/matroska/matroska-ids.c:
	(gst_matroska_track_init_video_context):
	* gst/matroska/matroska-ids.h:
	  Try harder to extract the framerate for video tracks correctly and
	  save it directly instead of converting it back and forth a few
	  times. Mostly makes a difference for very small framerates (<1).
	  Fixes #380199.

2383
2384
2385
2386
2387
2388
2389
2390
2391
2392
2393
2006-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init),
	(gst_gconf_audio_src_dispose), (do_toggle_element):
	* ext/gconf/gstgconfaudiosrc.h:
	  Remove gconf notify hook when the gconfaudiosrc element is
	  destroyed, otherwise the callback may be called on an
	  already-destroyed instance and bad things happen. Should fix
	  #378184.
	  Also ignore gconf key changes when the source is already running.

2394
2395
2396
2397
2398
2399
2400
2401
2402
2403
2404
2405
2006-12-09  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge  <mail at slomosnail de>

	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  We need to be able to read and parse any possible floating point string
	  format ("1,234" or "1.234") irrespective of the current locale. g_strod()
	  will parse the former only in certain locales though, so we really need
	  to canonicalise the separator to '.' and then use g_ascii_strtod() to
	  make sure we can parse either version at all times.
	  Fixes #382982 for real.

2406
2407
2408
2409
2410
2411
2412
2413
2414
2415
2416
2417
2418
2419
2420
2421
2422
2423
2424
2425
2426
2427
2428
2429
2430
2431
2432
2433
2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiosrc.c:

        Use the sunaudio debug category.

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
	(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
	(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
	(gst_sunaudiosink_open), (gst_sunaudiosink_close),
	(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
	(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
	(gst_sunaudiosink_reset):
	* sys/sunaudio/gstsunaudiosink.h:

	Uses the sunaudio debug category for all debug output
 	Implements the _delay() callback to synchronise video playback better
 	Change the segtotal and segsize values back to the parent class 
          defaults (taken from buffer_time and latency_times of 200ms and 10ms 
          respectively)
	Measure the samples written to the device vs. played.
	Keep track of segments in the device by writing empty eof frames, and
	sleep using a GCond when we get too far ahead and risk overrunning the
	sink's ringbuffer.

	Fixes: #360673

2434
2435
2436
2437
2438
2439
2440
2441
2442
2006-12-08  Wim Taymans  <wim@fluendo.com>

	Patch by: Sebastian Dröge  <mail at slomosnail de >

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
	(gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
	* gst/audiofx/audiopanorama.h:
	Fix audiopanorame with float samples. Fixes #383726.

2443
2444
2445
2446
2447
2448
2449
2450
2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_reset):
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open),
	(gst_sunaudiosrc_reset):

	Implement reset functions to unblock the src/sink more quickly on 
	state change requests.
2451
	Patch by: Brian Cameron <brian dot cameron at sun com>
2452

2453
2454
2455
2456
2457
2458
2459
2460
2461
2462
2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiomixer.c:
	(gst_sunaudiomixer_change_state):
	Construct the correct mixer device name when the AUDIODEV env var
	is set.

	Patch by: Jerry Tan <jerry.tan at sun dot com>
	Fixes: #383596

2463
2464
2465
2466
2467
2468
2469
2470
2471
2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	Apply patch to open the mixer control and set the MULTIPLE_OPEN
	ioctl. On solaris, the mixer device doesn't need opening non-blocking 
	- it can be opened by multiple processes by default, but needs the ioctl 	for multiple opens within 1 process.
	Patch by: Jerry Tan <jerry.tan at sun dot com>
	Fixes: #349015

2472
2473
2474
2475
2476
2477
2478
2479
2480
2481
2482
2483
2484
2485
2486
2487
2488
2489