ChangeLog 460 KB
Newer Older
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
2007-06-12  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
	(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
	finalization and resuscitation. No longer public.
	(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
	(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
	(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
	(gst_v4l2_buffer_pool_destroy): Make the pool follow common
	miniobject semantics, and be threadsafe.
	(gst_v4l2src_queue_frame): Remove this function, as we just call
	the ioctls directly in the two places where we queue buffers.
	(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
	directly.
	(gst_v4l2src_capture_init): Use the new buffer_pool_new function
	to allocate the pool, which also preallocates the GstBuffers.
	(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
	queueing the frames directly.
20
21
	(gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
	mmap buffers have been dequeued.
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37

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

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

38
39
40
41
42
43
44
45
46
2007-06-11  Wim Taymans  <wim@fluendo.com>

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

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

47
48
49
50
51
52
53
54
2007-06-11  Wim Taymans  <wim@fluendo.com>

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

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

55
56
57
58
59
60
61
2007-06-10  Sebastien Moutte  <sebastien@moutte.net>

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

62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
2007-06-10  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* configure.ac:
	* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
	(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
	(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
	(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
	(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
	(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
	(gst_flac_enc_chain), (gst_flac_enc_set_property),
	(gst_flac_enc_get_property), (gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
	
84
85
86
87
88
89
2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

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

90
91
92
93
94
95
96
97
98
99
100
2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

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

101
102
103
104
105
2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

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

106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

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

124
125
126
127
128
129
130
131
132
133
134
2007-06-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* sys/Makefile.am:
	* win32/MANIFEST:
	Add DirectDraw & DirectSound plugins to the build and docs.

135
136
137
138
139
140
2007-06-08  Tim-Philipp Müller  <tim at centricular dot net>

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

141
142
143
144
145
146
147
2007-06-06  Andy Wingo  <wingo@pobox.com>

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

148
149
150
151
152
153
154
155
2007-06-06  Tim-Philipp Müller  <tim at centricular dot net>

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

156
157
158
159
160
161
162
163
164
2007-06-05  Tim-Philipp Müller  <tim at centricular dot net>

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

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

165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
2007-06-05  Andy Wingo  <wingo@pobox.com>

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

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

180
181
182
183
184
185
186
2007-06-05  Wim Taymans  <wim@fluendo.com>

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

187
188
189
190
191
192
193
194
195
2007-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

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

196
197
198
199
200
2007-06-01  Wim Taymans  <wim@fluendo.com>

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

201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
2007-06-01  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (add_date_header),
	(rtsp_connection_send), (parse_response_status),
	(parse_request_line), (parse_line), (rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspmessage.c: (key_value_foreach),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_remove_header), (rtsp_message_append_headers),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Improves version checking, allowing an RTSP server to reply with "505
	RTSP Version not supported.
	Adds a Date header to all messages.
	Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
	want to be able to send a response even if something in the request was
	invalid. EINVAL is only used when passing wrong arguments to functions.
	Do not handle an invalid method in parse_request_line(). Defer this to
	the caller so it can respond with "405 Method Not Allowed".
	Improves parsing of the timeout parameter to the Session header,
	allowing whitespace after the semicolon. 
	Avoids a compiler warning due to variables shadowing a function argument.

228
229
230
231
232
233
234
235
236
237
238
239
240
241
2007-06-01  Wim Taymans  <wim@fluendo.com>

	Based on Patch by: Daniel Charles <dcharles at ti dot com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
	(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
	(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpamrpay.h:
	Add support for AMR-WB.
	Small cleanups such as using BOILERPLATE.

242
243
244
245
246
2007-05-31  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
	Fix compile warning when debug is disabled as spotted bu Saur on IRC.

247
248
2007-05-30  Andy Wingo  <wingo@pobox.com>

249
250
251
252
	* sys/v4l2/gstv4l2object.h: 
	* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
	unintended changes.

253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
	* sys/v4l2/v4l2src_calls.h: 
	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
	the format list in the order that the driver gives it to us.
	(gst_v4l2src_probe_caps_for_format_and_size)
	(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
	based on the capabilities of the device.
	(gst_v4l2src_grab_frame): Update for object variable renaming.
	(gst_v4l2src_set_capture): Update to be strict in its parameters,
	as in the set_caps below.
	(gst_v4l2src_capture_init): Update for object variable renaming,
	and reflow.
	(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
	(gst_v4l2src_capture_deinit): Update for object variable renaming.
	(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
	(gst_v4l2src_get_fps): Remove; these functions don't have much
	meaning outside of an atomic set_caps method.
	(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
	known.

	* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
	call to update_fps; not sure about this change.
	(gst_v4l2_tuner_set_norm): Work around the fact that for the
	moment we don't have an update_fps_func.

	* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
	structures in the object, just store what we need. Do store the
	probed caps of the device. Don't store the current frame rate.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
	update_fps_function, for now. Update for new object variable
	naming.
	(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
	new object variable naming.
	(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
	(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
	(gst_v4l2src_get_caps): Rework to probe the device for supported
	frame sizes and frame rates.
	(gst_v4l2src_set_caps): Rework to be strict in the given
	parameters: if someone asks us to have a certain size and rate,
	that is what we configure.
	(gst_v4l2src_get_read): Update for object variable naming. Don't
	leak buffers on short reads.
	(gst_v4l2src_get_mmap): Update for object variable naming, and add
	comments.
	(gst_v4l2src_create): Update for object variable naming.

299
300
301
302
303
304
305
306
2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
	(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
	* gst/avi/gstavidemux.h:
	  Parse subtitle text streams instead of erroring out (#442034). Still
	  needs a parser for the subtitles to actually show up.

307
308
309
310
311
312
313
314
315
2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
	(gst_avi_demux_loop):
	  Make _push_event() return TRUE if the event could be pushed on at
	  least one pad and not only if it could be pushed on all pads,
	  otherwise we'll end up posting an error message on EOS if one or
	  more source pads are not connected.

316
317
318
319
320
2007-05-28  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Use renamed RTP bin.

321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
2007-05-28  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>

	* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	(gst_video_box_set_property), (gst_video_box_transform_caps),
	(video_box_recalc_transform), (gst_video_box_set_caps),
	(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
	(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
	(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
	(gst_video_box_i420_i420), (gst_video_box_transform),
	(plugin_init):
	Add AYUV->AYUV and AYUV->I420 formats. 
	Fix negotiation and I420->AYUV conversion.
	Fixes #429329.

337
338
339
340
341
342
343
344
2007-05-26  Wim Taymans  <wim@fluendo.com>

	* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	Use different variables for nested for loops so that the outer loop
	functions properly and speex files with multiple frames per buffer work
	properly.
	Fixes #441408.

345
346
347
348
349
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
	  Don't leak newsegment events.

350
351
352
353
354
355
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
	  drags it in.

356
357
358
359
360
361
362
363
364
365
366
367
368
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(notgst_value_array_append_buffer),
	(gst_flac_enc_process_stream_headers),
	(gst_flac_enc_write_callback), (gst_flac_enc_chain),
	(gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	  Collect headers, add "streamheader" field to output caps and set
	  BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
	  produces output according to the official FLAC-to-Ogg mapping
	  instead of completely broken files. Fixes #426044.

369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
	(gst_id3demux_send_new_segment), (gst_id3demux_chain),
	(gst_id3demux_sink_event):
	* gst/id3demux/gstid3demux.h:
	* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
	(gst_tag_demux_chain), (gst_tag_demux_sink_event),
	(gst_tag_demux_send_new_segment):
	Handle and adjust new-segment events so that downstream really
	sees a stream with the tag pieces stripped off the front and back.
	Fixes strangeness in seeking when mp3 decoders use the new-segment
	byte position to estimate their current playback position timestamp
	and then the arriving buffers don't match up.

384
385
386
387
388
389
2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
	  Don't unnecessarily perform a READY->NULL->READY transition on the
	  detected audio sink when starting up. Fixes: #440127

390
391
392
393
394
395
2007-05-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
	(gst_flac_enc_chain):
	  Don't crash in chain function if setcaps hasn't been called.

396
397
398
399
400
2007-05-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
	Init value to avoid infinte loops.

401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
2007-05-24  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	(rtsp_connection_send), (rtsp_connection_receive):
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
	Fix for new API.

	* gst/rtsp/rtspconnection.c: (add_auth_header),
	Only add authorisation and session headers when sending messages.

	* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_unset), (rtsp_message_add_header),
	(rtsp_message_remove_header), (rtsp_message_get_header),
	(rtsp_message_append_headers), (dump_key_value),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Add support for multiple headers of the same type by storing the parsed
	headers in a GArray instaed of a hashtable.

426
427
428
429
430
431
432
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
	Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
	safer shutdown.

433
434
435
436
437
438
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Added signal for backwards compat.

439
440
441
442
443
444
445
446
447
448
449
450
2007-05-21  Sebastian Dröge  <slomo@circular-chaos.org>
	
	Patch by: René Stadler <mail at renestadler dot de>

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Use audioconvert for converting from non-native endianness floats
	in auparse instead of doing it ourself. Fixes #424527.
	This needs the audioconvert from plugins-base CVS.
	
451
452
453
454
455
456
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_flush):
	Fix enum registration.

457
458
459
460
461
462
463
464
465
466
467
2007-05-21  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
	(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
	(gst_rtp_h263p_pay_flush):
	* gst/rtp/gstrtph263ppay.h:
	Add new fragmentation mode base on GOB headers. Fixes #438940.

468
469
470
471
472
2007-05-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
	  Printf format fix.

473
474
475
476
477
478
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Don't crash when an unsupported transport error was returned by the
	server, just try to configure the next stream. Fixes #439255.

479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Add TCP timeout property and use it for all TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_write), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	Make connect and writes cancelable and make them use the timeout.

494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Refactor timeout handling.
	Also send keep-alive when dealing with TCP transport.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_free), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	* gst/rtsp/rtspconnection.h:
	Use a timer to handle the session timeouts, add some methods to deal
	with timeouts.

510
511
512
513
514
515
516
517
518
519
520
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Ignore streams that fail the setup command, we will retry with a
	different transport later on.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_configure_stream):
	Fix encoding name case.

521
522
523
524
525
2007-05-16  Edward Hervey  <edward@fluendo.com>

	* ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
	Fix build on macosx.

526
527
528
529
530
531
2007-05-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
	Replace direct comparison of a string with the string literal "" with
	a comparison of the first character with '\0'. Fixes #438926.

532
533
534
535
536
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c (gst_break_my_data_init):
	  One more try. This should be the proper fix now.

537
538
539
540
541
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c:
	  Ooops, no // comments please.

542
543
544
545
546
547
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c: (gst_break_my_data_class_init),
	(gst_break_my_data_init):
	  Fix gst_buffer_is_writable() assertion.

548
549
550
551
552
2007-05-14  David Schleef  <ds@schleef.org>

	* sys/v4l2/gstv4l2src.c: Add support for Bayer images as
	  video/x-raw-bayer.  Fixes #314160.

553
554
555
556
557
558
559
560
561
562
563
564
565
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtptheoradepay.c: (decode_base64),
	(gst_rtp_theora_depay_parse_configuration):
	* gst/rtp/gstrtptheorapay.c: (encode_base64),
	(gst_rtp_theora_pay_finish_headers),
	(gst_rtp_theora_pay_handle_buffer):
	Update theora pay/depayloader in a similar to vorbis.

	* gst/rtp/gstrtpvorbisdepay.c:
	(gst_rtp_vorbis_depay_parse_configuration):
	Update docs.

566
567
568
569
570
571
572
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
	When we try to execute a method that is not supported by the server,
	don't error out but remove the method from the accepted methods so that
	we never try to perform this method again.

573
574
575
576
577
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	Remove annoying _dump_mem.

578
579
580
581
582
583
584
585
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
	Parse range correctly.

	* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	The baseurl now always has a '/' at the start.

586
587
588
589
590
591
592
593
594
595
596
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	Factor out caps configuration and configure more stuff such as the time
	ranges and speed/scale values.

	* gst/rtsp/rtsptransport.c:
	Add Copyright after non-trival fixes.

597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
	(rtsp_message_get_header):
	* gst/rtsp/rtspmessage.h:
	Make channel guint8 where possible.
	Make rtsp_message_init_data() take the channel as a guint8.

	* gst/rtsp/rtspdefs.c:
	Fixed a typo: Timout -> Timeout

	* gst/rtsp/rtspdefs.h:
	Make RTSP_CHECK() behave as a statement.

	* gst/rtsp/sdpmessage.c:
	Avoid a compiler warning in INIT_ARRAY().
	Fixes #437692.

619
620
621
622
623
624
625
626
627
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
	(rtsp_url_get_request_uri):
	* gst/rtsp/rtspurl.h:
	Add support for query parameters to RTSP URLs.

628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
	(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	(rtsp_transport_parse), (rtsp_transport_as_text):
	* gst/rtsp/rtsptransport.h:
	Add validation to rtsp_transport_parse().
	Add rtsp_transport_as_text() to generate an RTSP header from an
	RTSPTransport.
	Change ssrc to guint (was a string) since that is what it is, even
	though it is sent as a hex string.
	Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
	incorrect, which can be seen when looking at the examples in the RFC).
	Fixes #437670.

646
647
648
649
650
651
652
653
654
655
2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	Patch by: Eric Anholt

	* sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
	  gst_ximage_src_ximage_get):
	Use union of all damage between frames to make it faster.
	Fixes bug #342463.
	Also fix crasher when cursor is at bottom right of window.

656
657
658
659
660
661
662
2007-05-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
	  streaming mode regression for file from #343837 with 'bext' chunk
	  before the 'fmt' chunk.

663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.h:
	Preliminary seek support.
	Activate internal pads so that we can receive events on them.
	Don't try to parse a range string when it's NULL.

678
679
680
681
682
683
684
685
686
687
688
689
690
691
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with new RTP variables that will be used for
	synchronisation.

	* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
	(gst_rtp_vorbis_depay_parse_configuration),
	(gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (encode_base64),
	(gst_rtp_vorbis_pay_finish_headers),
	(gst_rtp_vorbis_pay_handle_buffer):
	Update vorbis pay and depayloader to draft-04.

692
693
694
695
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	UDP MCAST is actually the default for RTP/AVP.
696
697
698
699
700
2007-05-13  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 can build 
	in_data += (filter->width / 8).
701

702
703
704
705
706
707
708
709
710
711
2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
	  gst_ximage_src_ximage_get):
	* sys/ximage/gstximagesrc.h (last_ximage):
	When using Damage actually keep the last frame, and not assume
	that the buffer we get already has the last frame on it.
	Copy the cursor over if we specify a non-zero start x and
	start y.

712
713
714
715
716
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Make UDP the default transport when not specified.

717
718
719
720
721
2007-05-09  David Schleef  <ds@schleef.org>

	* gst/level/gstlevel.c:
	  Revert last change.

722
723
724
725
726
727
728
729
730
731
732
733
734
2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
	(gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 know the size of data
	pointed when moving the pointer.
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Move instructions after variables declaration.
	* win32/vs6/autogen.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update vs6 project files.

735
736
737
738
739
740
741
742
743
744
745
746
2007-05-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
	* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
	(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
	(rtsp_range_free):
	* gst/rtsp/rtsprange.h:
	Add code to parse time ranges.
	Report DURATION on the stream when possible.

747
748
749
750
751
752
753
2007-05-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
	(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
	(gst_videomixer_collected):
	  Fix strides calculation for AYUV (it's just width*4) (#436910).

754
755
756
757
758
759
760
761
2007-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	Sync the GObject properties before each processing step to properly
	work with the controller.

762
763
764
765
766
767
768
769
770
771
772
773
774
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state):
	Let more error state trickle down so that we can catch more error
	cases.
	Handle keep-alive a little smarter by selecting a method the server
	actually supports.
	Fix a race in UDP streaming shutdown.

775
776
777
778
779
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
	Ignore errors when trying to use the keep-alive messages.

780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport):
	Send RTCP messages back to the server over the TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Factor out and expose lowlevel _write and _read methods.
	Implement sending data messages to the server.

799
800
801
802
803
804
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
	(gst_multipart_mux_collected):
	Fix timestamps on outgoing buffers.

805
806
807
808
809
810
811
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c:
	(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Emit NEWSEGMENT events before pushing the first buffer.

812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	Refactor transport configuration code.
	Create internal pads for TCP transport so that we can implement events
	and queries.
	Handle events and queries.
	Parse range from the SDP.
	Fix race in pause handler where the connection could still be flushing.

833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Fix race when multiple udp sources post timeouts, just act on the first
	received timeout.
	Protect stream list with a recursive lock to fix some races.
	Flush connection when we need to do a reconnect or stop.
	Make state lock recursive.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close):
	Some small cleanups.

852
853
854
855
856
857
858
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	Only set DISCONT when there actually is a discont or when we just
	started.

859
860
861
862
863
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflac.c: (plugin_init):
	Call bindtextdomain() to get localized strings.

864
865
866
867
868
869
870
871
872
873
874
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Be a bit more clever when dealing with VBR files with FACT tags, we
	don't want to timestamp buffers in that case but the estimated BPS can
	be used for seeking.
	Only send close segment in the streaming thread.

875
876
877
878
879
880
881
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	Correctly post an error on the bus if something went wrong in the loop
	function. This fixes a few cases where the task was paused and nothing
	happened anymore.

882
883
884
885
886
887
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix compilation of deprecated test just because I'm too lazy to delete
	it.

888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	Fix sending RTCP to the right place.
	Fix bug in reffing the wrong UDP element.
	Use new pad names for the session manager.
	Implement handling server requests in interleaved and UDP modes.
	Handle session keep-alive in UDP modes.
	Remove GCond for handling UDP timeouts.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive), (rtsp_connection_close):
	* gst/rtsp/rtspconnection.h:
	Store connection IP address for later.
	Add timeout args to all operations that might block forever.
	Parse session timeout.
	Only close sockets when not already closed.

	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Add timeout return value and error string.

	* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
	Add small comment.

920
921
922
923
924
925
926
927
928
2007-05-01  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
	* gst/rtp/gstrtpmp4vpay.h:
	Handle NEWSEGMENT and FLUSH events. Fixes #434824.

929
930
931
932
933
934
935
936
937
2007-04-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Remove v4l2src from docs, since it breaks the docs build, and the
	  plugin is only built if --enable-experimental is used anyway.

	* docs/plugins/Makefile.am:
	  Spaces => tab.

938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (leave_multicast),
	(gst_multiudpsink_add), (gst_multiudpsink_remove):
	Add code to drop membership of a multicast group.

	* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
	(gst_udpsink_set_uri):
	Implement URI handler.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	Use URI handler to make udpsink instace.
	Improve code to configure port and destination.

953
954
955
956
957
958
959
960
961
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	Fix multicast detection.
	Don't try to join a multicast group if the address is not multicast.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
	Small debug improvement.

962
963
964
965
966
967
968
969
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message):
	Ignore ASYNC state messages from the udpsink, it's irrelevant for the
	parent.

970
971
972
973
974
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpilbcdepay.h:
	Fix mode property when specified as an arg.

975
976
977
978
979
980
981
982
2007-04-26  Edward Hervey  <edward@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-osxaudio.xml:
	Add documentation for osxaudio plugin.

983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect state changes with a lock.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(parse_line):
	* gst/rtsp/rtspconnection.h:
	Remove some unused stuff.

998
999
1000
1001
1002
1003
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Handle the case where there are exactly 0 bytes to read and the ioctl
	did not report an error. Fixes #433530.

1004
1005
1006
1007
1008
1009
1010
1011
1012
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Apply DISCONT to buffers.
	Only apply timestamp to the first sample after a DISCONT, too many VBR
	files cause random jitter in the timestamps. Fixes #433119.

1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property):
	* gst/rtsp/gstrtpdec.h:
	Add dummy latency property to be backwards compat with rtpbin.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Add latency property and configure in the session manager.
	Don't set invalid clock-base and seqnum-base on caps, some servers
	sometimes don't send them.

1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
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.

1047
1048
1049
1050
1051
1052
1053
1054
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).

1055
1056
1057
1058
1059
2007-04-25  Wim Taymans  <wim@fluendo.com>

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

1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
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.

1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
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.

1095
1096
1097
1098
1099
2007-04-25  Stefan Kost  <ensonic@users.sf.net>

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

1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
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.

1112
1113
1114
1115
1116
1117
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).

1118
1119
1120
1121
1122
1123
1124
1125
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.

1126
1127
1128
1129
1130
1131
1132
1133
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.

1134
1135
1136
1137
1138
1139
1140
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.

1141
1142
1143
1144
1145
1146
1147
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.

1148
1149
1150
1151
1152
1153
1154
1155
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.

1156
1157
1158
1159
1160
1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
1171
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).

1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
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.

1189
1190
1191
1192
1193
1194
1195
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.

1196
1197
1198
1199
1200
1201
1202
1203
1204
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.

1205
1206
1207
1208
1209
1210
1211
1212
1213
1214
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.

1215
1216
1217
1218
1219
2007-04-12  Wim Taymans  <wim@fluendo.com>

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

1220
1221
1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235
1236
1237
1238
1239
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.

1240
1241
1242
1243
1244
1245
1246
1247
1248
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.

1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
1262
1263
1264
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.

1265
1266
1267
1268
1269
1270
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.

1271
1272
1273
1274
1275
1276
1277
1278
1279
1280
1281
1282
1283
1284
1285
1286
1287
1288
1289
1290
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.

1291
1292
1293
1294
1295
1296
1297
1298
1299
1300
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

1301
1302
1303
1304
1305
1306
1307
1308
1309
1310
1311
1312
1313
1314
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.

1315
1316
1317
1318
1319
1320
1321
1322
1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
1338
1339
1340
1341
1342
1343
1344
1345
1346
1347
1348
1349
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.

1350
1351
1352
1353
1354
1355
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.

1356
1357
1358
1359
1360
1361
1362
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.

1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
1373
1374
1375
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
1376
1377
1378
1379
1380
2007-04-03  Wim Taymans  <wim@fluendo.com>

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

1381
1382
1383
1384
1385
1386
1387
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.

1388
1389
1390
1391
1392
1393
1394
1395
1396
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.

1397
1398
1399
1400
1401
1402
1403
1404
1405
1406
1407
1408
1409
1410
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.

1411
1412
1413
1414
1415
1416
1417
1418
1419
1420
1421
1422
1423
1424
1425
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.

1426
1427
1428
1429
1430
1431
1432
1433
1434
1435
1436
1437
1438
1439
1440
1441
1442
1443
1444
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.

1445
1446
1447
1448
1449
1450
1451
1452
1453
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.

1454
1455
1456
1457
1458
1459
1460
1461
1462
1463
1464
1465
1466
1467
1468
1469
1470
1471
1472
1473
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

1474
1475
1476
1477
1478
1479
1480
1481
1482
1483
1484
1485
1486
1487
1488
1489
1490
1491
1492
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.

1493
1494
1495
1496
1497
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.

1498
1499
1500
1501
1502
1503
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.

1504
1505
1506
1507
1508
1509
1510
1511
1512
1513
1514
1515
1516
1517
1518
1519
1520
1521
1522
1523
1524
1525
1526
1527
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.

1528
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
1541
1542
1543
1544
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. 

1545
1546
1547
1548
1549
1550
1551
1552
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.

1553
1554
1555
1556
1557
1558
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.

1559
1560
1561
1562
1563
1564
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.

1565
1566
1567
1568
1569
1570
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.

1571
1572
1573
1574
1575
2007-03-17  David Schleef  <ds@schleef.org>

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

1576
1577
1578
1579
1580
2007-03-15  Edward Hervey  <edward@fluendo.com>

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

1581
1582
1583
1584
1585
1586
1587
1588
1589
1590
1591
1592
1593
1594
1595
1596
1597
1598
1599
1600
1601
1602
1603
1604
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

1605
1606
1607
1608
1609
1610
1611
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

1612
1613
1614
1615
1616
1617
1618
1619
1620
1621
1622
1623
1624
1625
1626
1627
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.

1628
1629
1630
1631
1632
2007-03-14  Tim-Philipp Müller  <tim at centricular dot net>

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

1633
1634
1635
1636
1637
1638
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.

1639
1640
1641
1642
1643
1644
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.

1645
1646
1647
1648
1649
1650
1651
1652
1653
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'.

1654
1655
1656
1657
1658
1659
1660
1661
1662
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.
	
1663
1664
1665
1666
1667
1668
1669
1670
1671
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.

1672
1673
1674
1675
1676
1677
1678
1679
1680
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.

1681
1682
1683
1684
1685
1686
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

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

1687
1688
1689
1690
1691
1692
1693
1694
1695
1696
1697
1698
1699
1700
1701
1702
1703
1704
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).

1705
1706
1707
1708
1709
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.

1710
1711
1712
1713
1714
1715
1716
1717
1718
1719
1720
1721
1722
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

1723
1724
1725
1726
1727
1728
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.

1729
1730
1731
1732
1733
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
1734
1735
	
	Patch by: Alex Lancaster <alexl at users sourceforge net>
1736

1737
1738
1739
1740
1741
1742
1743
1744
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.

1745
1746
1747
1748
1749
1750
1751
1752
1753
1754
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.

1755
1756
1757
1758
1759
1760
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.

1761
1762
1763
1764
1765
2007-03-08  Sebastian Dröge  <slomo@circular-chaos.org>

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

1766
1767
1768
1769
1770
1771
1772
1773
1774
1775
1776
1777
1778
1779
1780
1781
1782
1783
1784
1785
1786
1787
1788
1789
1790
1791
1792
1793
1794
1795
1796
1797
1798
1799
1800
1801
1802
1803
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.

1804
1805
1806
1807
1808
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.

1809
1810
1811
1812
1813
1814
1815
1816
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.

1817
1818
1819
1820
1821
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
1822
1823
1824
	  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 ...).
1825
1826
1827
1828
1829
1830
1831
1832
1833
1834
1835
1836
1837
1838
1839
1840
1841

	* 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.

1842
1843
1844
1845
1846
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
1847
1848
1849
1850
2007-03-06  Jan Schmidt  <thaytan@mad.scientist.com>

	Trigger rebuild.

1851
1852
1853
1854
1855
1856
1857
1858
1859
1860
1861
1862
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.

1863
1864
1865
1866
1867
1868
1869
1870
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.

1871
1872
1873
1874
1875
1876
1877
1878
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.

1879
1880
1881
1882
1883
2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

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

1884
1885
1886
1887
1888
1889
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.

1890
1891
1892
1893
1894
1895
1896
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.

1897
1898
1899
1900
1901
1902
1903
1904
1905
1906
1907
1908
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.

1909
1910
1911
1912
1913
1914
1915
1916
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.

1917
1918
1919
1920
1921
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
1922
1923
1924
1925
1926
2007-03-04  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Update.

1927
1928
1929
1930
1931
1932
1933
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.

1934
1935
1936
1937
1938
1939
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.

1940
1941
1942
1943
1944
1945
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.

1946
1947
1948
1949
1950
2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

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

1951
1952
1953
1954
1955
1956
1957
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.

1958
1959
1960
1961
1962
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.

1963
1964
1965
1966
1967
1968
1969
1970
1971
1972
1973
1974
1975
1976
1977
1978
1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
1989
1990
1991
1992
1993
1994
1995
1996
1997
1998
1999
2000
2001
2002
2003
2004
2005
2006
2007
2008
2009
2010
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.

2011
2012
2013
2014
2015
2016
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.

2017
2018
2019
2020
2021
2022
2023
2024
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).

2025
2026
2027
2028
2029
2007-03-02  Stefan Kost  <ensonic@users.sf.net>

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

2030
2031
2032
2033
2034
2035
2036
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.

2037
2038
2039
2040
2041
2042
2043
2044
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
2045
2046
2047
2048
2049
2050
2007-03-02  Wim Taymans  <wim@fluendo.com>

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

2051
2052
2053
2054
2055
2056
2057
2058
2059
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

2060
2061
2062
2063
2064
2065
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.

2066
2067
2068
2069
2070
2071
2072
2073
2074
2075
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.

2076
2077
2078
2079
2080
2007-03-02  Wim Taymans  <wim@fluendo.com>

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

2081
2082
2083
2084
2085
2086
2087
2088
2089
2090
2091
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
2092
2093
2094
2095
2096
2007-03-02  Wim Taymans  <wim@fluendo.com>

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

2097
2098
2099
2100
2101
2102
2103
2104
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.

2105
2106
2107
2108
2109
2110
2111
2112
2113
2114
2115
2116
2117
2118
2119
2120
2121
2122
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.

2123
2124
2125
2126
2127
2128
2129
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.

2130
2131
2132
2133
2134
2135
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.

2136
2137
2138
2139
2140
2141
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.

2142
2143
2144
2145
2146
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.

2147
2148
2149
2150
2151
2152
2153
2154
2155
2156
2157
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.

2158
2159
2160
2161
2162
2163
2164
2165
2166
2167
2168
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.

2169
2170
2171
2172
2173
2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  Convert to new AG_GST style.

2174
2175
2176
2177
2178
2179
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

2180
2181
2182
2183
2184
2007-02-28  Wim Taymans  <wim@fluendo.com>

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

2185
2186
2187
2188
2189
2190
2191
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.

2192
2193
2194
2195
2196
2007-02-28  Wim Taymans  <wim@fluendo.com>

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

2197
2198
2199
2200
2201
2007-02-28  Wim Taymans  <wim@fluendo.com>

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

2202
2203
2204
2205
2206
2207
2208
2209
2210
2211
2212
2213
2214
2215
2216
2217
2218
2219
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.

2220
2221
2222
2223
2224
2225
2226
2227
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.

2228
2229
2230
2231
2232
2233
2234
2235
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.

2236
2237
2238
2239
2007-02-27  Christian Schallerr <christian@fluendo.com>

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

2240
2241
2242
2243
2244
2007-02-26  Jan Schmidt  <thaytan@mad.scientist.com>

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

2245
2246
2247
2248
2249
2250
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.

2251
2252
2253
2254
2255
2256
2257
2258
2259
2260
2261
2262
2263
2264
2265
2266
2267
2268
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)

2269
2270
2271
2272
2273
2274
2275
2276
2277
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.

2278
2279
2280
2281
2282
2283
2284
2285
2286
2287
2288
2289
2290
2291
2292
2293
2294
2295
2296
2297
2298
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.

2299
>>>>>>> 1.2755
2300
2301
2302
2303
2304
2305
2306
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.
	
2307
2308
2309
2310
2311
2312
2313
2314
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.

2315
2316
2317
2318
2319
2320
2321
2322
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.

2323
2324
2325
2326
2327
2328
2329
2330
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

2331
2332
2333
2334
2335
2336
2337
2338
2339
2340
2341
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.

2342
2343
2344
2345
2346
2347
2348
2349
2350
2351
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.

2352
2353
2354
2355
2356
2357
2358
2359
2360
2361
2362
2363
2364
2365
2366
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.

2367
2368
2369
2370
2371
2007-02-15  Wim Taymans  <wim@fluendo.com>

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

2372
2373
2374
2375
2376
2377
2378
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.

2379
2380
2381
2382
2383
2384
2385
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.

2386
2387
2388
2389
2390
2391
2392
2393
2394
2395
2396
2397
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.

2398
2399
2400
2401
2402
2403
2404
2405
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>
2406
2407
2408
2409
2410
2411

	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.

2412
2007-02-14  Wim Taymans  <wim@fluendo.com>
2413
2414
2415
2416
2417
2418
2419
2420
2421
2422
2423
2424