ChangeLog 318 KB
Newer Older
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
2006-11-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
        We depend on gsttag to generate the vorbis comments.

	* gst/rtp/gstrtpvorbisdepay.c:
	(gst_rtp_vorbis_depay_parse_configuration),
	(gst_rtp_vorbis_depay_setcaps),
	(gst_rtp_vorbis_depay_switch_codebook),
	(gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbisdepay.h:
	Parse configuration string in the depayloader.
	Implement selecting and switching to a new codebook.
	Receiving vorbis over RTP now works.

	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
	(gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_finish_headers),
	(gst_rtp_vorbis_pay_handle_buffer):
	* gst/rtp/gstrtpvorbispay.h:
	Set timestamps on outgoing buffers and RTP packets.
	Fix configuration string, prepend number of Packet headers.
	Fix encoding of ident string.
	Add delivery-method to caps.
	Streaming vorbis over RTP now works.

27
28
29
30
31
32
33
34
35
2006-11-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	(gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
	(gst_rtp_vorbis_pay_handle_buffer):
	* gst/rtp/gstrtpvorbispay.h:
        Generate a valid configuration string in the caps based on the
        vorbis headers.

36
37
38
39
40
41
42
43
2006-11-02  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cdio/gstcdio.c: (gst_cdio_get_cdtext):
	* ext/cdio/gstcdio.h:
	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
	  Move CD-TEXT utility function into common file so it can also be
	  used by a future cdioparanoiasrc.

44
45
46
47
48
49
50
51
52
53
54
2006-11-01  Edgard Lima <edgard.lima@indt.org.br>
	
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to
	xoverlay code that is still not implemented.

55
56
57
58
59
60
61
62
63
64
65
66
67
68
2006-11-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  We require a -base more recent than 0.10.9, so it's safe to use
	  GST_TYPE_TAG_IMAGE_TYPE unconditionally now.

	* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
	  Use _newsegment_full() now that we depend on a recent enough core.

	* gst/wavparse/gstwavparse.c:
	  Remove cruft that we don't need any longer now that we depend on
	  a recent enough -base.

69
70
71
72
73
74
75
76
2006-10-31  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
	(gst_rtpilbcpay_setcaps):
	Fix and activate ILBC pay and depayloaders. Fixes #368162.

77
78
79
80
81
82
2006-10-31  Wim Taymans  <wim@fluendo.com>

	* ext/speex/gstspeexdec.c: (speex_dec_convert),
	(speex_dec_sink_event), (speex_dec_chain_parse_header):
	Some small cleanups, use _scale.

83
84
85
86
87
2006-10-31  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	Use higher precision scale function.

88
89
90
91
92
93
94
95
96
97
98
2006-10-30  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Michal Benes  <michal dot benes at itonis tv>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_decode_buffer):
	  Fix several issues with encoded/compressed/encrypted/signed tracks;
	  also, remove superfluous newline characters from some debug
	  statements. (#366155)

99
100
101
102
103
104
105
106
107
108
109
110
111
112
2006-10-30  Wim Taymans  <wim@fluendo.com>

	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
	(gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
	(gst_smokedec_change_state):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
	(gst_smokeenc_init), (gst_smokeenc_finalize),
	(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
	(gst_smokeenc_resync), (gst_smokeenc_chain),
	(gst_smokeenc_set_property), (gst_smokeenc_get_property),
	(gst_smokeenc_change_state):
	Various cleanups, capsnego and leak fixes.

113
114
115
116
117
118
119
120
2006-10-30  Wim Taymans  <wim@fluendo.com>

	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
	Fix videomixer so that it can handle any combination of framerates.
	Fixes #367221.

121
122
123
124
125
126
127
128
129
130
131
2006-10-28  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_file_header),
	(gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix position query for audio. also fixes timestamps in streaming
	mode and bug #364958.
	Small cleanups.

132
133
134
135
136
137
138
139
2006-10-27  Wim Taymans  <wim@fluendo.com>

	* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain):
	* ext/libpng/gstpngenc.h:
	Fix strides. Fixes #364856.
	Cleanup capsnego.
	Set caps on outgoing buffers.

140
141
142
143
144
145
146
147
148
149
150
151
152
2006-10-18  Wim Taymans  <wim@fluendo.com>

	Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>

	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	(gst_rtp_pcma_pay_handle_buffer):
	* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
	Add static payload numbers in addition to the dynamic ones.
	Fixes #361639.

153
154
155
156
157
158
159
160
161
162
163
164
165
166
2006-10-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/rtspurl.h:
	Reuse already existing enum for lower transport.
	Add rtspt and rtspu protocols.
	Send redirect to rtspt when udp times out.

167
168
169
170
171
172
2006-10-18  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_data):
	Fix seeking some more, mostly for speed changes.

173
174
2006-10-18  Tim-Philipp Müller  <tim at centricular dot net>

175
	Patch by: Fredrik Persson  <frepe at bredband net>
176
177
178
179
180
181
182

	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	  Fix _set_channel(): remove useless g_object_notify() for "channel"
	  property that doesn't exist any longer and therefore now also
	  useless redirect (#338818).

183
184
185
186
187
188
189
2006-10-17  Wim Taymans  <wim@fluendo.com>

	* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
	Some drivers do not support unsetting the non-blocking flag once the
	device is opened. In those cases, close/open the device in
	non-blocking mode. Fixes #362673.

190
191
192
193
194
195
196
197
2006-10-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  dear stefan, framespersecond is not frameperiod, reverting but adding
	  comment

198
199
200
201
202
203
204
205
2006-10-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  Numerator is numerator and denominator is denominator. Say that aloud
	  5 times and retry after next beer.

206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
2006-10-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Josep Torra Valles  <josep at fluendo com>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/esd/esdsink.c: (gst_esdsink_write):
	* ext/flac/gstflacdec.c: (gst_flac_dec_length),
	(gst_flac_dec_read_seekable), (gst_flac_dec_chain),
	(gst_flac_dec_send_newsegment):
	* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
	(gst_flac_enc_tell_callback):
	* ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
	(smokecodec_parse_header), (smokecodec_decode):
	* gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
	* gst/debug/efence.c: (gst_fenced_buffer_alloc):
	* gst/goom/Makefile.am:
	* gst/goom/gstgoom.c:
	* gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
	* sys/sunaudio/gstsunaudiomixertrack.h:
	  Fix a bunch of problems discovered by the Forte compiler, mostly type
	  mixups and pointer arithmetics with void pointers. Fixes #362603.

233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeex.c: (plugin_init):
	* ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
	(gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
	(gst_speex_enc_class_init), (gst_speex_enc_finalize),
	(gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
	(gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
	(gst_speex_enc_src_query), (gst_speex_enc_sink_query),
	(gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
	(gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
	(gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
	(gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
	(gst_speex_enc_chain), (gst_speex_enc_get_property),
	(gst_speex_enc_set_property), (gst_speex_enc_change_state):
	* ext/speex/gstspeexenc.h:
	  Miscellaneous clean-ups, among other things: speexenc => enc to
	  enhance code readability; change speexenc => speex_enc; in chain
	  function unref input buffer in case of error; take reference in
	  event function; use boilerplate macro; use gst_pad_query_peer_*
	  convenience functions.

255
256
257
258
259
260
261
2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
	(gst_speexenc_set_last_msg), (gst_speexenc_setup),
	(gst_speexenc_set_header_on_caps):
	  Fix some mem leaks.

262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
2006-10-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some other URL.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
	(gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Work on fallback to TCP connection when the UDP socket times out.
	Handler server requests, just reply with OK for now.

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Added some more Real extension headers.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of urls with a ':' that is not part of the hostname:port
	part of the url.

283
284
285
286
287
288
289
290
291
292
2006-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
	* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Activate pad before adding it to the already-running element.

	* tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Activate newly-created pad too.

293
294
295
296
297
298
299
300
301
2006-10-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Sebastien Cote <sebas642 at yahoo dot ca>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
	(gst_udpsrc_start):
	Fix some leaks in caps and uris. Fixes #361252.

302
303
304
305
306
2006-10-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).

307
308
309
310
311
312
2006-10-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	Fix build as per the patch in #338818 comment 36.

313
314
315
316
317
2006-10-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
	  Activate pads before adding them to the source.

Wim Taymans's avatar
Wim Taymans committed
318
319
320
321
322
323
2006-10-06  Wim Taymans  <wim@fluendo.com>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
	* gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
	Activate pads before adding.

324
325
326
327
328
329
330
331
332
2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	(gst_multipart_find_pad_by_mime):
	Activate pads before adding.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	BOILERPLATE sets parent_class for us.

333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_alloc_udp_ports),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Rework how the transport string is constructed, try to share channels
	and udp ports.
	Make most of the stuff less dependant on RTP as we are also going to use
	it for RDT.
	Add support for transport specific session managers.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
	Implement _flush().

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Add generic error return code.

	* gst/rtsp/rtspext.h:
	Add support for pluggable tranport strings.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
	(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	Detect WMServer and activate the extension.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
	(rtsp_transport_get_manager), (rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Added methods to get mime/manager for certain transports.

Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
2006-10-05  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c:
	(gst_cairo_time_overlay_update_font_height):
	* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	* ext/libpng/gstpngdec.c: (user_endrow_callback):
	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_data):
	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	* gst/debug/efence.c: (gst_efence_buffer_alloc),
	(gst_fenced_buffer_copy):
	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_handle_message):
	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	  Printf format fixes.

397
398
399
400
401
2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	Dist new .h file too.

402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
	(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Factor out extension in separate module.
	Fix getcaps to filter against the padtemplate.
	Use Content-Base if the server gives one.
	Rework the transport parsing a bit for future extensions.
	Added some Real Header field definitions.

429
430
431
432
433
434
435
436
437
2006-10-04  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  added v4l2 stubs
	* gst-plugins-good.spec.in:
	  add v4l2

438
439
440
441
442
443
2006-10-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  Extract disc/album/medium number and count and try harder
	  to extract track number/count.

444
445
446
447
448
449
450
2006-10-03  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* sys/Makefile.am:
	  add build stuff for v4l2, needs --enable-experimental until
	  the last bits are resolved

451
452
453
454
455
456
457
458
459
2006-09-29  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Disable autodetect test temporarily, so that the build bots
	  update -bad and the ranks of unreliable video sinks in there.

	* tests/check/elements/autodetect.c: (GST_START_TEST):
	  Skip test if no usable videosink is found.

460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add some more URLs.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_finalize),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add timeout property to control UDP timeouts.
	Fix error messages.
	Also start a loop function when operating in UDP mode so that we can
	do some more stuff async.
	Handle element messages from udpsrc to detect timeouts. If a timeout
	happens we currently generate an error.
	API: rtspsrc::timeout property.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create):
	Really implement the timeout in microseconds and not milliseconds.

487
488
489
490
491
492
493
494
495
496
497
498
2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Added property to post a message on timeout.
	Updated docs.
	When restarting the select, initialize the fdsets again.
	Init control sockets so we don't accidentally close a random socket.
	API: GstUDPSrc::timeout property

499
500
501
502
503
504
505
506
2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
	Fix flag registration.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	Reading 0 also means 'no more commands'

507
508
509
510
511
512
513
514
2006-09-29  Wim Taymans  <wim@fluendo.com>

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

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Fix possible infinite loop when shutting down, a read can also return
	0 to indicate no more messages are available. Fixes #358156.

515
516
517
518
519
520
521
522
523
2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
	(gst_auto_audio_sink_find_best):
	* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
	Small cleanups.
	don't try to set "sync" property when it is not available.

524
525
526
527
528
529
530
531
532
533
534
535
2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/alpha/gstalpha.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	* gst/videomixer/videomixer.c:
	  Include stdlib.h in some more places, makes things compile
	  with uClibc and -Werror (#357592).

536
537
538
539
540
541
542
2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/jpeg/gstjpegdec.c:
	  Set minimum height to 8 (from 16), our code should handle
	  that fine. Some of the buttons on the apple trailer site
	  are apparently only 15 pixels high (see #357470).

543
544
545
546
547
548
549
550
551
552
553
2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_open):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Improve error reporting.

Wim Taymans's avatar
Wim Taymans committed
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
	(gst_rtp_mp2t_depay_plugin_init):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
	Fix klass typos.
	Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.

577
578
579
580
581
2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Need  -base CVS for gst_base_rtp_depayload_push_ts().

582
583
584
585
586
587
588
2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
	Don't check for a tag that is never there and check if we read the
	correct tag. Fixes seeking again.
	We must post an error when all pads are unlinked.

589
590
591
592
593
594
595
596
597
598
599
600
2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	(gst_rtp_vorbis_pay_reset_packet),
	(gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
	(gst_rtp_vorbis_pay_handle_buffer):
	More fixage, set endoder-params correctly in the payloader.

601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_base_init):
	  Make static pad templates static to appease valgrind's leak
	  detector.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/autodetect.c: (GST_START_TEST),
	(autodetect_suite):
	  Add simple test for the ghostpad lockup on shutdown fixed in core
	  CVS (audio bit disabled because it would need dozens of alsa
	  suppressions and I'm too lazy to add those now).

Wim Taymans's avatar
Wim Taymans committed
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
	Small cleanups.

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
	(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
	(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
	(gst_rtp_vorbis_depay_process),
	(gst_rtp_vorbis_depay_set_property),
	(gst_rtp_vorbis_depay_get_property),
	(gst_rtp_vorbis_depay_change_state),
	(gst_rtp_vorbis_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
	(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
	(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet),
	(gst_rtp_vorbis_pay_append_buffer),
	(gst_rtp_vorbis_pay_handle_buffer),
	(gst_rtp_vorbis_pay_plugin_init):
	* gst/rtp/gstrtpvorbispay.h:
	Add experimental vorbis pay and depayloaders.

645
646
647
648
649
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
	Fix profile-level-id parsing and setup.

650
651
652
653
654
655
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/README:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	Update README, simple cleanup.

656
657
658
659
660
661
662
663
664
665
666
667
668
2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with some examples.

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
	(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
	(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
	(gst_rtp_mp4g_pay_setcaps):
	* gst/rtp/gstrtpmp4gpay.h:
	Make optional RTP parameters of type STRING, as required by the
	application/x-rtp caps specification.

669
670
2006-09-20  Philippe Kalaf  <philippe.kalaf at collabora.co.uk>

671
	* gst/rtp/gstrtph263pdepay.c:
672
673
674
675
	* gst/rtp/gstrtph263ppay.c:
	Correctly calculate size of each H263+ RTP buffer taking into account MTU and
	RTP header.

676
677
678
679
680
2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	And makefile too.

681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
	(gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
	(decode_base64), (gst_rtp_asf_depay_setcaps),
	(gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
	(gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
	(gst_rtp_asf_depay_plugin_init):
	* gst/rtp/gstrtpasfdepay.h:
	Added preliminary ASF depayloader.

	* gst/rtp/gstrtph264depay.c: (decode_base64):
	Fix base64 decoding.

696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some test URLS.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_loop), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	When creating streams, give access to the complete SDP.
	Fix some leaks.
	Collect and merge global stream properties in stream caps.
	Preliminary support for WMServer.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Make connection interruptable.
	Refactor to make it reconnectable.
	Don't fail on short reads when reading data packets.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
	(rtsp_url_get_port):
	* gst/rtsp/rtspurl.h:
	Add methods for getting/setting the port.

	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_message_get_attribute_val), (sdp_media_get_attribute),
	(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
	(sdp_media_get_format), (sdp_parse_line),
	(sdp_message_parse_buffer):
	Fix headers. 
	Add methods for getting multiple attributes with the same name.
	Increase buffer size when parsing.
	Fix parsing of a=foo fields.

	* gst/rtsp/test.c: (main):
	Update to new connection API.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtsp.h:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Dual licensed under MIT and LGPL now.

752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Reorganize stream parsing and creation.
	Detect container formats in interleaved mode.
	Keep more state about the streams.
	Assume a server also supports PLAY if it does not say.
	Add unicast and interleaved properties to TCP transport requests to make
	some servers happy (WMServer).

	* gst/rtsp/sdpmessage.h:
	Add some defines for the standard Bandwidth types.

Wim Taymans's avatar
Wim Taymans committed
772
773
774
775
776
2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix build.

777
778
779
780
781
2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c:
	Add ms-gsm to the src template.

782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Small cleanups, added documentation.
	Try to clean up the requests and responses.
	Refactor parsing the supported methods.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
	(rtsp_connection_create), (rtsp_connection_send),
	(parse_response_status), (parse_request_line),
	(rtsp_connection_receive), (rtsp_connection_close),
	(rtsp_connection_free):
	* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
	(rtsp_transport_init), (rtsp_transport_parse),
	(rtsp_transport_free):
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
	(sdp_message_clean), (sdp_message_free), (sdp_media_new),
	(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
	Use g_return_val some more.

	* gst/rtsp/rtspdefs.h:
	Add more enum values to track initial states.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
	(rtsp_message_init_request), (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free),
	(rtsp_message_add_header), (rtsp_message_remove_header),
	(rtsp_message_get_header), (rtsp_message_set_body),
	(rtsp_message_take_body), (rtsp_message_get_body),
	(rtsp_message_steal_body), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Reorder arguments, object goes as the first one.
	Use g_return_val some more.

823
824
825
826
827
828
829
830
831
832
833
834
835
2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Export sometimes source pad with correct caps on the template, create
	the ghostpad from the template.
	Remove RTCP template as we never expose RTCP.
	Protect against invalid body size.
	Avoid memcpy when creating the output buffer.
	Properly post an error and send EOS when the loop function is shut down.

836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Make sure we can never set an invalid location.

	* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
	* gst/rtsp/rtspmessage.h:
	Added _steal_body method for future use.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
	Make freeing of NULL url return immediatly.

853
854
855
856
857
858
859
860
861
862
863
864
865
2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Use boilerplate.
	Make rtspsrc subclass GstBin to make state changes easier.
	Add Range header field on the PLAY request.

866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
	* gst/rtsp/rtspconnection.c: (inet_aton):
	Small cleanups.
	when multicast is selected as the transport, create UDP sources and
	connect to the multicast group.
	Move parsing and setting of caps to a common place.
	Fixes #349894.

Stefan Kost's avatar
Stefan Kost committed
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* ext/flac/gstflactag.c:
	* gst/alpha/gstalpha.c:
	* gst/debug/breakmydata.c:
	* gst/debug/negotiation.c:
	* gst/debug/testplugin.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videomixer/videomixer.c:
	* sys/sunaudio/gstsunaudiosrc.h:
903
	More G_OBJECT macro fixing.
Stefan Kost's avatar
Stefan Kost committed
904

905
906
907
908
909
910
911
912
913
2006-09-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Yves Lefebvre <ivanohe at abacom dot com>

	* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
	Correctly set the dwLength in strh.
	With this patch, the file duration is now displayed correctly in window
	media player and the AVI plays completely. Fixes #356147

914
915
916
917
918
919
920
921
922
923
2006-09-15  Wim Taymans  <wim@fluendo.com>

	Patch by: Darren Kenny <darren dot kenny at sun dot com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list):
	Set the output track as the MASTER so that the gnome-settings-daemon
	keybindings for changing the volume using the keyboard works.
	Fixes #356142.

924
925
926
927
928
929
930
931
2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	Fix documentation, it is not possible to control the framerate of jpegdec
	using filtered caps yet. Fixes #355210.
	Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
	stop when there is an error.

932
933
934
935
936
937
938
939
2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't interpret a first buffer with an offset of NONE as
	  'from the middle of the stream', but only a first buffer
	  that has a valid buffer offset that's non-zero (see #345449).

940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
	(gst_icydemux_typefind_or_forward):
	* gst/icydemux/gsticydemux.h:
	  When we merge/collect multiple incoming buffers for typefinding
	  purposes, keep an initial 0 offset on the first outgoing buffer
	  as well (otherwise id3demux won't work right). Fixes #345449.
	  Also Make buffer metadata writable before setting buffer caps.

	* tests/check/elements/icydemux.c: (typefind_succeed),
	(cleanup_icydemux), (push_data), (GST_START_TEST),
	(icydemux_suite):
	  Small test case for the above.

955
956
957
958
959
960
961
962
963
964
965
2006-09-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  More code reuse and better logging in _peek_chunk(). Reintroduce check
	  for chunk sizes before reading them (avoid oom). Better handling for 
	  invalid chunksizes when streaming.

966
967
968
969
970
971
972
2006-09-11  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_set_property):
	* gst/level/gstlevel.h:
          Fix type mixup in level->interval (gdouble<->guint64). Spotted by
          René Stadler

973
974
975
976
977
978
979
2006-09-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_data):
	  Revert one change to fix streaming avi (adapter size != data size).

980
981
982
983
984
985
986
987
988
989
990
991
992
993
2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Frédéric Riss  <frederic.riss at gmail dot com>

	* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
	(gst_matroska_demux_reset),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.h:
	  Add support for VOBSUB subtitle tracks and zlib-compressed
	  tracks. Make sure we start on a keyframe after a seek. (#343348)

994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
	(gst_matroska_demux_push_flac_codec_priv_data),
	(gst_matroska_demux_push_xiph_codec_priv_data),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	* gst/matroska/matroska-ids.h:
	  Add basic FLAC support (#311586), not perfect yet though, needs some
	  tweaking in flacdec; also, seeking could be better.
	  Do better bounds checking when deserialising vorbis stream headers
	  to make sure we don't read beyond the end of the buffer on bad input.

1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
2006-09-04  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
	Seeking back in a file containing a CMML stream errors out if the seek
	goes back up to the CMML headers. This is because after the seek the xml
	processing instruction <?xml ...?> is submitted to the xml parser again, 
	which results in an error. The attached patch fixes the problem. 
	Fixes #353908.

	* ext/annodex/gstcmmlenc.h:
	Fix authors name.


1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
2006-08-28  Andy Wingo  <wingo@pobox.com>

	* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
	New helper function to lessen the ifdefs.
	(GST_INFO_OBJECT): 
	(gst_dv1394src_iso_receive): Use it.
	(gst_dv1394src_create): Also use the control sockets in iec61883
	mode.
	(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
	handle for AVC operations; fixes #348233.

1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
2006-08-27  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audiofxgood.xml:
	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c:
	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c:
	* gst/audiofxgood/audiopanorama.c:
	* gst/audiofxgood/audiopanorama.h:
          Rename again (audiofxgood -> audiofx).

1048
1049
1050
1051
1052
1053
2006-08-27  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_stream_scan):
          Initialze variables.

1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
2006-08-25  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_init), (gst_avi_demux_finalize),
	(gst_avi_demux_reset), (gst_avi_demux_index_last),
	(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
	(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
	(gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
	(gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	More attempts to turn this into readable code.
	Don't leak adapters.
	Calculate duration according to index more efficiently.
	Don't try to act like we drive the pipeline in chain mode.

1075
1076
1077
1078
1079
2006-08-25  Wim Taymans  <wim@fluendo.com>

	* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt):
	Fix build.

1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
2006-08-25  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
	Do some extra sanity checks.
	Fixes #350340.

	* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
	(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
	(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
	Check if clip->start_time is valid before adding the clip to the
	track list.
	Reset enc->preamble going from PAUSED to READY.
	Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
	only used for EOS.
	Only post an error message if we were the one that created the fatal
	GstFlowReturn value.

	* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
	(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
	Parse the seconds field of the npt-sec time format using %llu rather than
	%d and check that the value scaled by GST_SECOND doesn't overflow.
	Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
	Lookup a clip's track with clip->track rather than clip->id which
	makes no sense.
	Identify a clip by its track and start time and not its xml id.
	do some more input checking and make sure we don't do undefined shifts.

	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
	(cmml_tag_message_pop), (check_headers), (push_clip_full),
	(push_clip), (push_empty_clip), (check_output_clip),
	(GST_START_TEST), (cmmldec_suite):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
	(check_headers), (push_clip), (check_clip_times), (check_clip),
	(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
	Added some more checks.

1120
1121
1122
1123
1124
1125
1126
1127
1128
1129
1130
1131
1132
1133
2006-08-24  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_transform_m2s_int),
	(gst_audio_panorama_transform_s2s_int),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float):
	* gst/audiofxgood/audiopanorama.h:
	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
          Make also the pan-property float (saves scaling and yields better
          resolution)

1134
1135
2006-08-24  Stefan Kost  <ensonic@users.sf.net>

1136
1137
1138
1139
1140
1141
1142
1143
1144
1145
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float):
          ChangeLog surgery to add cymax's real name


2006-08-24  Stefan Kost  <ensonic@users.sf.net>

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

1146
1147
1148
1149
1150
1151
1152
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_int),
	(gst_audio_panorama_transform_s2s_int),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float),
	(gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
1153
          Added float support
1154

1155
1156
1157
1158
1159
1160
2006-08-24  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofxgood/audiopanorama.c:
	(gst_audio_panorama_transform_m2s):
	  Fix docs & debug category. Add Fixme for volume pan levels.

1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
2006-08-24  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	  unbreak AVI index handling, some more debug, remove an obsolete
	  adapter_flush that caused streaming to wander off in the wild

1171
1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
2006-08-24  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex),
	(gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull):
	* gst/avi/gstavidemux.h:
	Some more cleanups. 
	Fix totalFrames parsing in ODML.
	Disable use of index for length calculation in case of ODML as this is
	broken now.

1186
1187
1188
1189
1190
2006-08-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata):
	  Use libgsttag helper function here too.

1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
1205
1206
1207
1208
1209
1210
1211
1212
1213
2006-08-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_init), (gst_avi_demux_dispose),
	(gst_avi_demux_reset), (gst_avi_demux_index_next),
	(gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	(gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk),
	(gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull),
	(gst_avi_demux_parse_subindex),
	(gst_avi_demux_read_subindexes_push),
	(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
	(sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_peek_tag),
	(gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain),
	(gst_avi_demux_sink_activate), (gst_avi_demux_activate_push),
	(gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	  Initial streaming support for avidemux (fixes #336465)

1214
1215
1216
1217
1218
1219
1220
1221
1222
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  There is no taglibmux element ...

	* gst/rtsp/gstrtspsrc.c:
	  Use '%' rather than '&perc;' in gtk-doc blurb, docs build
	  was complaining about unknown entity here.

1223
1224
1225
1226
1227
1228
1229
1230
1231
2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
	(gst_avi_demux_process_next_entry):
	* gst/avi/gstavidemux.h:
	Mark DISCONT.
	Remove old unused fields and reorder the struct a bit.

Wim Taymans's avatar
Wim Taymans committed
1232
1233
1234
1235
1236
1237
1238
1239
1240
1241
2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	* sys/oss/gstosssink.c: (gst_oss_sink_open),
	(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
	Small documentation updates.

1242
1243
1244
1245
1246
1247
1248
1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	(gst_avi_demux_next_data_buffer),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
	(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	Precalc most of the duration query for each stream.
	Make seeking more correct.
	Use GstSegment to track position and duration.
	Code cleanups and leak fixes.
	Calculate correct total duration based on index length.

1262
1263
1264
1265
1266
1267
1268
1269
2006-08-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
	(parse_insert_string_field):
	  If strings in text fields are marked ISO8859-1, but contain
	  valid UTF-8 already, then handle them as UTF-8 and ignore
	  the encoding. (#351794)

1270
1271
1272
1273
1274
1275
1276
1277
1278
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
	(gst_flac_dec_write), (gst_flac_dec_loop),
	(gst_flac_dec_sink_event), (gst_flac_dec_chain),
	(gst_flac_dec_src_query):
	* ext/flac/gstflacdec.h:
	  Make flac-in-ogg work (#352100).

1279
1280
1281
1282
1283
1284
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
	  Don't unref buffers of which we've already given away
	  ownership to the adapter.

1285
1286
1287
1288
1289
1290
1291
1292
1293
1294
1295
1296
1297
2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
	  Make metadata extraction actually work.

	* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
	(gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
	(gst_speexenc_chain):
	  Fix metadata writing: replace old code which wrote completely
	  broken tags with libgsttag-based code. Plus miscellaneous
	  code cleanups (use static pad templates etc.) and a bunch
	  of leak fixes.

1298
1299
1300
1301
1302
1303
1304
1305
1306
2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiopanorama/.cvsignore:
	* gst/audiopanorama/Makefile.am:
	* gst/audiopanorama/audiofx.c:
	* gst/audiopanorama/audiopanorama.c:
	* gst/audiopanorama/audiopanorama.h:
          die! die! die! you should never have been there

1307
1308
1309
1310
1311
2006-08-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
	Fix invalid memory access in audiopanorama test suite.

1312
1313
1314
1315
1316
2006-08-21  Edward Hervey  <edward@fluendo.com>

	* tests/check/elements/.cvsignore:
	ignore built file

1317
1318
1319
1320
1321
2006-08-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	Fix the build again.

1322
1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c: (plugin_init):
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_get_unit_size),
	(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s),
	(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
	  resubmit with the desired name *again*

Stefan Kost's avatar
Stefan Kost committed
1338
1339
1340
1341
1342
1343
2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
	* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
          use g_assert in _get_unit_size

1344
1345
1346
1347
1348
1349
1350
1351
1352
1353
1354
1355
1356
1357
1358
1359
2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-audiofxgood.xml:
          cleanup -unused.txt to make it useful, add previously missing docs

	* ext/Makefile.am:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/esd/gstesd.c: (plugin_init):
          reflow to get rid of two external symbols

	* gst/audiofxgood/audiofx.c: (plugin_init):
          re-add

Stefan Kost's avatar
Stefan Kost committed
1360
1361
1362
1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
1373
1374
1375
1376
1377
1378
1379
1380
2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_get_unit_size),
	(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s),
	(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
	* tests/check/Makefile.am:
	* tests/check/elements/audiopanorama.c: (setup_panorama_m),
	(setup_panorama_s), (cleanup_panorama), (GST_START_TEST),
	(panorama_suite), (main):
        Add audiofxgood plugin with audiopanorama element

1381
1382
1383
1384
1385
2006-08-18  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/Makefile.am:
	More Oss docs fixage. 

1386
1387
1388
1389
1390
1391
1392
1393
1394
1395
1396
1397
1398
1399
2006-08-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_base_init),
	(gst_rtp_sv3v_depay_class_init), (gst_rtp_sv3v_depay_init),
	(gst_rtp_sv3v_depay_finalize), (gst_rtp_sv3v_depay_setcaps),
	(gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_set_property),
	(gst_rtp_sv3v_depay_get_property),
	(gst_rtp_sv3v_depay_change_state),
	(gst_rtp_sv3v_depay_plugin_init):
	* gst/rtp/gstrtpsv3vdepay.h:
	Added experimental SVQ3 depayloader.

1400
1401
1402
1403
1404
1405
1406
1407
1408
2006-08-18  Edward Hervey  <edward@fluendo.com>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
	(gst_dvdemux_loop), (gst_dvdemux_change_state):
	* ext/dv/gstdvdemux.h:
	When handling seek requests, don't send the newsegment event from the
	calling thread. Instead save it so it can be sent from the streaming
	thread.

1409
1410
1411
1412
1413
1414
1415
1416
1417
1418
1419
2006-08-17  Wim Taymans  <wim@fluendo.com>

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

	* gst/multipart/multipartdemux.c: (multipart_parse_header):
	Accept leading whitespace before the boundary
	This patch makes the demuxer allow some whitespace before the actual
	boundary. This makes the demuxer work with the ``old'' gstreamer
	multipartmuxer again (which placed an extra \n before the start
	of the stream) Fixes #349068.

1420
1421
1422
1423
1424
2006-08-17  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	Error out on non-implemented stuff.

1425
1426
1427
1428
1429
1430
1431
1432
1433
1434
1435
1436
2006-08-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Andy Wingo <wingo at pobox dot com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setup),
	(gst_signal_processor_start), (gst_signal_processor_stop),
	(gst_signal_processor_cleanup), (gst_signal_processor_setcaps),
	(gst_signal_processor_pen_buffer), (gst_signal_processor_flush),
	(gst_signal_processor_do_pulls), (gst_signal_processor_do_pushes),
	(gst_signal_processor_change_state):
	Make ladspa elements reusable. Fixes #350006.

1437
1438
1439
1440
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
	Convert ' ' into '_'. Try to keep as many characters in the padtemplate
1441
	names as possible. Fixes #349901.
1442

1443
1444
1445
1446
1447
1448
1449
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
	(gst_signal_processor_do_pushes):
	A push() gives away our refcount so we should not use the buffer on the
	pen anymore.

1450
1451
1452
1453
1454
1455
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	(gst_oss_mixer_element_finalize):
	  Don't leak device string.

1456
1457
1458
1459
1460
1461
1462
1463
1464
1465
1466
1467
1468
1469
1470
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Require CVS of GStreamer core and -base (for
	  GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).

	* ext/taglib/gstid3v2mux.cc:
	  Write extended comment tags properly (#348762).

	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	(parse_comment_frame):
	  Extract COMM frames into extended comments, which makes it
	  easier to properly retain the description bit of the tag
	  and maintain this information when re-tagging (#348762).

1471
1472
1473
1474
1475
1476
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Don't try to run annodex unit tests if the annodex
	  plugin has not been built (Fixes #351116).

1477
1478
1479
1480
1481
1482
1483
1484
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_find_best):
	  When we can't find a usable audiosink, don't error out,
	  but use a fake sink instead and post a warning message
	  on the bus (#341278).

1485
1486
1487
1488
1489
1490
1491
1492
1493
1494
1495
1496
1497
1498
1499
1500
1501
1502
1503
1504
1505
1506
1507
1508
1509
1510
1511
1512
1513
1514
1515
1516
1517
1518
1519
1520
1521
1522
1523
1524
1525
1526
1527
1528
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
1541
1542
1543
1544
1545
1546
2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init):
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	  Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for
	  ossmixer's new device property.

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Add docs for OSS elements.

	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update to CVS version.
	  
1547
1548
1549
1550
1551
1552
1553
1554
1555
1556
1557
1558
1559
1560
1561
1562
1563
1564
1565
1566
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	Caps extra properties must be defined as strings for
	depayloaders because they are generated from an SDP.

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
	(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
	(gst_rtp_h264_depay_finalize), (decode_base64),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	(gst_rtp_h264_depay_set_property),
	(gst_rtp_h264_depay_get_property),
	(gst_rtp_h264_depay_change_state),
	(gst_rtp_h264_depay_plugin_init):
	* gst/rtp/gstrtph264depay.h:
	Added basic, not completely functional RFC 3984 H264 depayloader.

1567
1568
1569
1570
1571
1572
1573
1574
1575
1576
1577
1578
1579
1580
1581
1582
1583
1584
1585
1586
1587
1588
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
	Add pads after setting them up.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_finalize),
	(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Fix interleaved mode.
	 - Protect streaming with lock.
	 - Combine flows
	 - set caps on outgoing buffers.
	 - strip trailing \0 from data packets.
	 - Configure RTP/RTCP in stream.
	Use DEBUG_OBJECT more.

1589
1590
1591
1592
1593
2006-08-16  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	Turn a g_print into a DEBUG line.

1594
1595
1596
1597
1598
1599
1600
1601
1602
1603
1604
1605
1606
2006-08-13  Wim Taymans  <wim@fluendo.com>

	* sys/oss/gstossmixer.c: (gst_ossmixer_open), (gst_ossmixer_new):
	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	(gst_oss_mixer_element_init), (gst_oss_mixer_element_set_property),
	(gst_oss_mixer_element_get_property),
	(gst_oss_mixer_element_change_state):
	* sys/oss/gstossmixerelement.h:
	Small cleanups. Better error reporting.
	Add device property for the mixer instead of the hardcoded
	/dev/mixer. Fixes #350785.
	API: GstOssMixerElement::device property

1607
1608
1609
1610
1611
1612
1613
1614
1615
2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jens Granseuer <jensgr at gmx net>

	* gconf/Makefile.am:
	  Make --disable-schemas work right (they still need
	  to be copied to the installation directory, just not
	  applied). Fixes #351347 (also #344100).
	  
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1616
1617
1618
1619
2006-08-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac: back to HEAD

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1620
1621
1622
1623
1624
1625
1626
=== release 0.10.4 ===

2006-08-14  Thomas Vander Stichele <thomas at apestaart dot org>

	* configure.ac:
	  releasing 0.10.4, "Dear Leader"

1627
1628
1629
1630
1631
1632
1633
1634
1635
1636
2006-08-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	Patch by: Edward Hervey <edward@fluendo.com>

	* configure.ac:
	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_data):
	Send the newsegment event in the streaming thread.
	Fixes #347529

1637
1638
1639
1640
1641
1642
2006-08-08  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
	(gst_smokeenc_resync), (gst_smokeenc_chain):
	  Refuse sink caps in the encoder if width or height is not a
1643
1644
1645
	  multiple of 16, the encoder does not support that yet (#349939);
	  along the same lines, check the return value of the encoder
	  setup function; also remove some debug log clutter.
1646

1647
1648
1649
1650
1651
1652
1653
1654
1655
1656
1657
1658
1659
1660
1661
1662
1663
1664
1665
1666
1667
1668
2006-08-04  Andy Wingo  <wingo@pobox.com>

	* ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
	whether a processor can work in place or not, and for keeping
	track of its state. Change the FlowReturn instance variable from
	"state" to "flow_state", all callers changed.

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
	(gst_signal_processor_start, gst_signal_processor_stop)
	(gst_signal_processor_cleanup): New functions to manage the
	processor's state.
	(gst_signal_processor_setcaps): start() as well as setup() here.
	(gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
	(gst_signal_processor_change_state): Stop and cleanup the
	processor as we go to NULL.

	* ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
	INPLACE_BROKEN is not set.

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
	Do the alloc_buffer in bytes, not frames.
	
1669
1670
1671
1672
1673
2006-08-04  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	Fix rgb masks when recording in < 24bpp.

1674
1675
1676
1677
1678
1679
1680
1681
1682
1683
1684
1685
2006-08-04  Andy Wingo  <wingo@pobox.com>

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
	(gst_signal_processor_prepare)
	(gst_signal_processor_update_inputs)
	(gst_signal_processor_process, gst_signal_processor_pen_buffer)
	(gst_signal_processor_flush)
	(gst_signal_processor_sink_activate_push)
	(gst_signal_processor_src_activate_pull)
	(gst_signal_processor_change_state): Remove the last of the code
	that assumes that we process whole buffers at a time. Fix some
	debugging. Seems to work now in some cases.
Andy Wingo Wingo's avatar
BPB    
Andy Wingo Wingo committed
1686
	(gst_signal_processor_src_activate_pull): BPB
1687

1688
1689
2006-08-01  Andy Wingo  <wingo@pobox.com>

1690
1691
1692
1693
	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
	Fix nframes-choosing.
	(gst_signal_processor_init): Init pending_in and pending_out.

1694
1695
1696
1697
1698
1699
1700
1701
1702
1703
1704
1705
1706
1707
1708
1709
1710
1711
	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
	more default sample rate, although we never check that the sample
	rate actually gets set. Something for the future.
	(gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
	(gst_signal_processor_event): Refcount fixen.
	(gst_signal_processor_process): Pull the number of frames to
	process from the sizes of the buffers in the input pens.
	(gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
	(gst_signal_processor_do_pulls): Add an nframes argument, and use
	it instead of buffer_frames.
	(gst_signal_processor_getrange): Refcount fixen, pass nframes on
	to do_pulls.
	(gst_signal_processor_chain)
	(gst_signal_processor_sink_activate_push)
	(gst_signal_processor_src_activate_pull):  Refcount fixen.

	* ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.

1712
1713
1714
1715
2006-07-31  Stefan Kost  <ensonic@users.sf.net>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	(gst_signal_processor_process):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1716
1717
	  don't query buffer-frames from caps, add lots of debug-log,
	  try fix for assert (#349189)
1718

Wim Taymans's avatar
Wim Taymans committed
1719
1720
1721
1722
1723
2006-07-31  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c:
	Fix docs.

1724
1725
1726
1727
1728
1729
1730
1731
1732
1733
2006-07-29  Stefan Kost  <ensonic@users.sf.net>

	* ext/ladspa/gstsignalprocessor.c:
	(gst_signal_processor_add_pad_from_template),
	(gst_signal_processor_init), (gst_signal_processor_setcaps),
	(gst_signal_processor_process), (gst_signal_processor_pen_buffer),
	(gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
	(gst_signal_processor_sink_activate_push),
	(gst_signal_processor_src_activate_pull),
	(gst_signal_processor_change_state):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1734
1735
	 Add debugs logs here and there, add more error handling, add some
	 FIXME comments, filed #349189
1736

1737
1738
1739
1740
1741
1742
2006-07-29  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
	(gst_smokeenc_setcaps), (gst_smokeenc_chain):
	Set caps on buffer correctly.  Fixes bug #349155.

1743
1744
1745
1746
1747
1748
1749
1750
1751
1752
1753
1754
1755
1756
1757
1758
1759
1760
1761
1762
2006-07-28  Wim Taymans  <wim@fluendo.com>

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

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
	(gst_multipart_demux_class_init), (gst_multipart_demux_init),
	(gst_multipart_demux_finalize), (get_line_end),
	(multipart_parse_header), (multipart_find_boundary),
	(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
	(gst_multipart_set_property), (gst_multipart_get_property):
	Uses GstAdapter instead of own buffering.
	Actually parses the mime-type correctly (In tests the mime-type was
	always "" with the old version).
	Uses the Content-length header if available to speed up things.
	Reliably autoscans the boundary name by default.
	Fixes #349068.

	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	Don't start the stream with a \n.

1763
1764
1765
1766
1767
1768
1769
2006-07-28  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	  Open source with O_NONBLOCK (#349015).

1770
1771
1772
1773
1774
1775
1776
2006-07-28  Stefan Kost,,,  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
	(gst_avi_demux_massage_index):
	* gst/avi/gstavidemux.h:
	  Whitespace fixes and more debug

1777
1778
1779
1780
1781
1782
1783
1784
1785
1786
2006-07-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_create_element_with_pretty_name),
	(gst_auto_audio_sink_find_best),
	(gst_auto_audio_sink_change_state):
	  Get rid of old and unused magic sound-server properties stuff.
	  Add suffix to child sink's name that makes it easy to see from
	  the name alone which type it actually is (alsa, oss, esd, etc.).

1787
1788
1789
1790
1791
1792
1793
1794
1795
2006-07-27  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_set_property), (gst_udpsrc_get_property),
	(gst_udpsrc_start):
	* gst/udp/gstudpsrc.h:
	Rename "buffer" to "buffer-size" to make clear it is a size we set and
	not some sort of feature we enable.

1796
1797
1798
1799
1800
1801
2006-07-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Use CLOSE_SOCKET() here instead of close() to maintain
	  win32 workiness.

1802
1803
1804
1805
1806
1807
1808
1809
2006-07-27  Wim Taymans  <wim@fluendo.com>

	Patch by: Thijs Vermeir <thijs dot vermeir at barco 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_udpsrc_start):
	* gst/udp/gstudpsrc.h:
1810
	Added "buffer-size" property to control the kernel receive buffer size.
1811
1812
	Update documentation.
	Small cleanups. Fixes #348752.
1813
	API: buffer-size property
1814

1815
1816
1817
1818
1819
1820
1821
1822
1823
1824
1825
2006-07-26  Wim Taymans  <wim@fluendo.com>

	Patch by: Kai Vehmanen <kv2004 at eca dot cx>

	* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	(gst_rtp_pcma_pay_handle_buffer):
	* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush),
	(gst_rtp_pcmu_pay_handle_buffer):
	Fix timestamp calculation on outgoing RTP packets.
	Fixes #348675.

1826
1827
1828
1829
1830
1831
1832
1833
2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Fix writing of comment frames (should be COMM not TCOM),
	  is still sub-optimal though, since we don't retain or
	  extract the comment descriptions properly (#334375,
	  also see #334375).

1834
1835
1836
1837
1838
1839
1840
2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c:
	  #define 'fact' RIFF chunk if we are not compiling against
	  -base CVS (we don't want to depend on -base CVS for this
	  one define only, and also not for release order reasons).

1841
1842
1843
1844
1845
1846
1847
1848
2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Handle multiple tags of the same type properly. Re-inject
	  unparsed ID3v2 frames that we get as binary blobs from
	  id3demux into the tag again so we don't lose information
	  when retagging (#334375).

1849
1850
1851
1852
1853
1854
1855
1856
2006-07-25  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init):
	  Document newly-added properties properly, so that there is a
	  'Since: 0.10.4' in the plugin docs. Convert some property
	  names into canonical GObject style (GObject will do that
	  internally anyway).

1857
1858
1859
1860
1861
1862
1863
1864
2006-07-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist):
	  Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
	  well, and add the version to the blob's buffer caps, since that
	  information will be needed for deserialisation later on (#348644).

1865
1866
1867
1868
1869
1870
1871
2006-07-25  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
	(gst_avi_demux_parse_stream):
	 Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
	 indentation and spacing.

1872
1873
1874
1875
1876
1877
1878
1879
1880
1881
1882
1883
1884
1885
1886
1887
1888
1889
1890
1891
1892
1893
1894
1895
1896
1897
1898
1899
1900
1901
1902
1903
1904
1905
1906
1907
1908
1909
1910
1911
1912
1913
1914
1915
1916
1917
1918
1919
1920
1921
1922
1923
1924
1925
1926
1927
1928
1929
1930
1931
1932
1933
2006-07-24  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/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update files to CVS/Prerelease version, add esdsink docs.

	* ext/esd/esdsink.c:
	  Add gtk-doc blurb.

	* gst/rtp/gstrtpmp4vpay.c:
	  Fix typo in element description.

1934
1935
1936
1937
1938
1939
1940
1941
1942
1943
2006-07-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/esdsink.c: (gst_esdsink_open),
	(gst_esdsink_factory_init):
	  Prevent libesd from auto-spawning a sound daemon if it
	  is not already running. Now that we don't do evil stuff
	  like that any longer we can give esdsink a rank so that
	  autoaudiosink will try it as well if all other audio
	  sinks fail (#343051).

1944
1945
1946
1947
1948
2006-07-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/Makefile.am:
	  Oops, need to remove README from EXTRA_DIST as well.

1949
1950
1951
1952
1953
1954
1955
1956
1957
2006-07-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/README:
	  Remove, it contains nothing useful anyway.

	* ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
	(gst_esdsink_delay):
	  Some small clean-ups; use GST_BOILERPLATE etc.

1958
1959
1960
1961
1962
1963
1964
1965
2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst/law/alaw-decode.c: (alawdec_getcaps):
	* gst/law/alaw-encode.c: (alawenc_getcaps), (gst_alawenc_chain):
	* gst/law/mulaw-decode.c: (mulawdec_getcaps):
	* gst/law/mulaw-encode.c: (mulawenc_getcaps):
	Fix negotiation to deal with ANY/EMPTY caps instead of leaking.

1966
2006-07-24  Stefan Kost  <ensonic@users.sf.net>
1967
1968
1969
1970
1971
1972
1973
1974
1975
1976
1977
1978

	* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	(gst_wavparse_other), (gst_wavparse_perform_seek),
	(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	(gst_wavparse_pad_query):
	* gst/wavparse/gstwavparse.h:
	  Use information from 'fact' chunk for length calculation of compressed
	  samples. Calculate bps if bogus value is found in wav header (embeded
	  mp2/mp3).
	  

1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
1989
1990
1991
1992
1993
1994
1995
1996
1997
1998
1999
2000
2006-07-24  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Joni Valtanen  <joni dot valtanen at movial fi>

	* configure.ac:
	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.c: (gst_dynudpsink_init),
	(gst_dynudpsink_finalize), (gst_dynudpsink_close):
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init),
	(gst_multiudpsink_finalize), (gst_multiudpsink_close):
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudp.c: (plugin_init):
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
	(gst_udpsrc_start), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	* gst/udp/gstudpnetutils.c: (gst_udp_net_utils_win32_inet_aton),
	(gst_udp_net_utils_win32_wsa_startup):
	* gst/udp/gstudpnetutils.h:
	  Port udp plugin to win32 (#345288).

2001
2002
2003
2004
2005
2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_send):
	Remove unwanted DEBUG line.

2006
2007
2008
2009
2010
2011
2012
2013
2014
2006-07-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (plugin_init):
	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist):
	* gst/id3demux/id3tags.h:
	  On second thought, it might be wiser and more efficient
	  not to do tag registration from a streaming thread.

2015
2016
2017
2018
2019
2020
2021
2022
2023
2024
2006-07-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist),
	(id3demux_id3v2_frames_to_tag_list):
	  Put ID3v2 frames we can't parse as binary blobs into private
	  tags, so that they are not lost when retagging, at least once
	  id3v2mux has been taught to re-inject those frames again.
	  See bug #334375.

2025
2026
2027
2028
2029
2030
2031
2032
2033
2006-07-21  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_process_next_entry):
	Fix some leaks.

	* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	Don't use \n in debug lines.

2034
2035
2036
2037
2006-07-20  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
2038
	  Add annodex and icydemux, cleanup the sections a bit
2039

2040
2041
2042
2043
2044
2045
2046
2047
2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Alex Lancaster <alexl at users sourceforge net>

	* ext/taglib/gstid3v2mux.cc:
	  Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
	  ID3v2 TSSE frames (#347898).

2048
2049
2050
2006-07-18  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
2051
2052
	  Respect mpegversion for "video/mpeg" and give message in case of
	  unhandled versions.
2053

2054
2055
2056
2057
2058
2059
2060
2061
2062
2063
2064
2065
2006-07-17  Wim Taymans  <wim@fluendo.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip),
	(gst_pngdec_caps_create_and_set), (gst_pngdec_task),
	(gst_pngdec_chain), (gst_pngdec_sink_event),
	(gst_pngdec_libpng_init), (gst_pngdec_change_state),
	(gst_pngdec_sink_activate_push):
	* ext/libpng/gstpngdec.h:
	Use statically allocated segment instead of leaking.
	Various cleanups.
	Fix flush and seek handling.

2066
2067
2068
2069
2070
2071
2072
2073
2074
2075
2076
2077
2078
2079
2080
2081
2082
2083
2006-07-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init),
	(gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init),
	(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process),
	(gst_rtp_mp4g_depay_set_property),
	(gst_rtp_mp4g_depay_get_property),
	(gst_rtp_mp4g_depay_change_state),
	(gst_rtp_mp4g_depay_plugin_init):
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
	(gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps),
	(gst_rtp_mp4g_pay_flush):
	Added simple generic mpeg4 depayloader.
	Fix generic mpeg4 payloader.

2084
2085
2086
2087
2088
2006-07-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
	  Don't try doing state changes on a NULL pointer.

2089
2090
2091
2092
2093
2094
2095
2096
2097
2098
2099
2006-07-14  Wim Taymans  <wim@fluendo.com>

	Patch by: Sebastien Cote <sebas642 at yahoo dot ca>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_base_init),
	(gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_init),
	(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpamrdepay.h:
	rtpamrdec isn't a subclass of GstBaseRtpDepayload.
	Fixes #321191

2100
2101
2102
2103
2104
2105
2006-07-14  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_get_caps), (gst_ximage_src_class_init):
	Fix segfault when moving mouse pointer to the bottom right corner.

2106
2107
2108
2109
2110
2111
2112
2113
2114
2115
2116
2117
2118
2119
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init),
	(gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init),
	(gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process),
	(gst_rtp_mp2t_depay_set_property),
	(gst_rtp_mp2t_depay_get_property),
	(gst_rtp_mp2t_depay_change_state),
	(gst_rtp_mp2t_depay_plugin_init):
	* gst/rtp/gstrtpmp2tdepay.h:
	Added mpeg2 TS depayloader. Closing #347234.

2120
2121
2122
2006-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_close):
2123
2124
2125
	  Remove g_assert that shouldn't be there and was triggered
	  after trying to open a device that doesn't exist or can't
	  be opened for some other reason (#347972).
2126

2127
2128
2129
2130
2131
2132
2133
2134
2006-07-10  Edward Hervey  <edward@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_stream_header), (push_tag_lists):
	* gst/avi/gstavidemux.h:
	Don't push tag events found by gst_riff_parse_info() before outputting
	GST_EVENT_NEWSEGMENT.

2135
2136
2137
2138
2139
2140
2141
2142
2143
2006-07-10  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
	(rtsp_connection_close):
	* gst/rtsp/rtspdefs.h:
	replaced closesocket and close in code with one CLOSE_SOCKET. 
	Some more cleanups. Fixes #345301.

2144
2145
2146
2147
2148
2006-07-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	  Fix example pipeline in docs.

2149
2150
2151
2152
2153
2154
2155
2156
2157
2158
2006-07-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Rob Taylor <robtaylor at floopily dot org>

	* gst/udp/gstmultiudpsink.c: (join_multicast),
	(gst_multiudpsink_init_send), (gst_multiudpsink_add):
	If a destination is added before the stream is set to PAUSED, the
	multicast group is not joined as the socket is not created yet. 
	Also TTL and LOOP should also be set. Fixes #346921.

2159
2160
2161
2162
2163
2164
2165
2166
2167
2168
2169
2006-07-09  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_set_property), (gst_ximage_src_get_property),
	(gst_ximage_src_get_caps), (gst_ximage_src_class_init),
	(gst_ximage_src_init):
	* sys/ximage/gstximagesrc.h:
	Fix use-damage property to actually work :)
	Add startx, starty, endx, endy properties so screencasts other than full
	screen ones can work.

2170
2171
2172
2173
2174
2175
2176
2177
2178
2006-07-08  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_set_property), (gst_ximage_src_get_property),
	(gst_ximage_src_class_init), (gst_ximage_src_init):
	* sys/ximage/gstximagesrc.h:
	Add use_damage property to offer ability to choose whether to use
	XDamage or not.

2179
2180
2181
2182
2183
2184
2006-07-07  Wim Taymans  <wim@fluendo.com>

	* gst/goom/filters.c: (zoomFilterSetResolution):
	Avoid goom coredumping by clearing memory. 
	Fixes 345679.

2185
2186
2187
2188
2189
2006-07-05  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/libgstid3demux.dsp:
	Add a link to libgsttag-0.10.lib.

2190
2191
2192
2193
2194
2195
2196
2197
2198
2199
2200
2201
2006-07-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	(gst_tag_demux_read_range):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	(gst_id3demux_read_range):
	  Don't return FLOW_UNEXPECTED when a buffer is before
	  the start of the stream (which might happen with
	  large ID3v2 tags if the tag reading was done pullrange
	  based and we then switched to push mode later on).
	  Fixes regression introduced by commit from June 29th.

2202
2203
2204
2205
2206
2207
2208
2209
2210
2211
2212
2213
2214
2215
2006-07-05  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Make UTF-8 the default encoding when writing string
	  tags (before, our UTF-8 strings would automatically
	  be converted to ISO-8859-1 by taglib and written as
	  ISO-8859-1 fields if that was possible).

	* tests/check/elements/id3v2mux.c: (utf8_string_in_buf),
	(test_taglib_id3mux_check_tag_buffer), (identity_cb),
	(test_taglib_id3mux_with_tags):
	  Add test case that makes sure our UTF-8 strings have
	  actually been written into the tag as UTF-8.

2216
2217
2218
2219
2220
2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Let's try that again.

2221
2222
2223
2224
2225
2226
2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Disable monoscope plugin for now until it fulfills
	  all the requirements.

2227
2228
2229
2230
2231
2232
2233
2234
2235
2236
2237
2238
2239
2240
2006-07-03  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	* gst/monoscope/Makefile.am:
	* gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init),
	(gst_monoscope_class_init), (gst_monoscope_init),
	(gst_monoscope_finalize), (gst_monoscope_reset),
	(gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps),
	(gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain),
	(gst_monoscope_sink_event), (gst_monoscope_src_event),
	(gst_monoscope_change_state), (plugin_init):
	* gst/monoscope/gstmonoscope.h:
	  Port monoscope visualisation to 0.10.

2241
2242
2243
2244
2245
2246
2247
2248
2249
2006-07-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Return FLOW_UNEXPECTED when at the end of the file, not
	  FLOW_ERROR. Fixes 'internal stream error' errors that
	  would sometimes occur in totem when scrubbing to the
	  end of an ID3v1 tagged mp3 file.

2250
2251
2252
2253
2254
2255
2256
2257
2258
2006-07-03  Edward Hervey  <edward@fluendo.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback),
	(buffer_clip), (user_end_callback), (gst_pngdec_chain),
	(gst_pngdec_sink_event), (gst_pngdec_change_state):
	* ext/libpng/gstpngdec.h:
	Implement buffer clipping/dropping using GstSegment.
	This provides accurate seeking.

2259
2260
2261
2262
2263
2264
2265
2266
2267
2268
2269
2270
2271
2272
2273
2006-07-03  Edward Hervey  <edward@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
	(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	(gst_avi_demux_process_next_entry), (push_tag_lists),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	* gst/avi/gstavidemux.h:
	Proper aggregation of each stream's GstFlowReturn in order to figure out
	whether the task should stop or not.
	Don't send inline events before pushing out a NEW_SEGMENT, more
	specifically for GST_TAG_EVENT.
	Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
	sub-indexes.

2274
2275
2276
2277
2278
2279
2280
2281
2282
2283
2006-06-30  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list):
	  Move "Monitor" slider to input tab so it works more like
	  sdtaudiocontrol, which is what people on Solaris are used
	  to using for their mixer program (#346259).

2284
2285
2286
2287
2288
2006-06-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/elements/level.c: (GST_START_TEST):
	  fix a leak, clean up at the end

2289
2290
2291
2292
2293
2294
2295
2296
2006-06-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	(gst_matroska_demux_send_event),
	(gst_matroska_demux_loop_stream_parse_id):
	* gst/matroska/matroska-ids.h:
	  Send tag event after newsegment event.

2297
2298
2299
2300
2301
2302
2303
2304
2305
2306
2307
2308
2006-06-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	(gst_id3demux_read_range):
	  Make sure we don't return GST_FLOW_OK with a NULL buffer in
	  certain cases where a read beyond the end of the file is
	  requested. Fixes #345930.

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	(gst_tag_demux_read_range):
	  Fix same issue here as well.

2309
2310
2311
2312
2313
2314
2006-06-29  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
	
	Fix hypothetical crash.

2315
2316
2317
2318
2319
2320
2321
2322
2323
2324
2325
2006-06-28  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Do not modify the ports value. If the user has turned off the
	  built-in speakers, then we should not reset it in the prepare
	  function, since this causes the built-in speakers to turn
	  back on anytime the user changes a track in totem, rhythmbox,
	  etc. (#346066).

2326
2327
2328
2329
2330
2006-06-23  Wim Taymans  <wim@fluendo.com>

	* gst/goom/gstgoom.c: (gst_goom_src_negotiate):
	Fix double caps unref when negotiation fails.

2331
2332
2333
2334
2335
2336
2337
2338
2339
2340
2341
2342
2343
2344
2345
2346
2347
2348
2349
2350
2351
2352
2353
2354
2355
2356
2357
2358
2359
2360
2361
2362
2363
2364
2365
2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/annodex/gstcmmlparser.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/speex/gstspeexenc.c:
	* gst/alpha/gstalphacolor.c:
	* gst/cutter/gstcutter.c:
	* gst/debug/gstnavigationtest.c:
	* gst/icydemux/gsticydemux.c:
	* gst/level/gstlevel.c:
	* gst/multipart/multipart.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
	  plus two minor macro fixes.

2366
2367
2368
2369
2370
2371
2372
2373
2374
2375
2376
2377
2378
2379
2380
2381
2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c:
	(gst_matroska_demux_check_subtitle_buffer),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.c:
	(gst_matroska_track_init_subtitle_context):
	* gst/matroska/matroska-ids.h:
	  Try to fix up broken matroska files containing subtitle
	  streams with non-UTF8 character encodings (courtesy of
	  mkvmerge) using either the encoding specified in the
	  GST_SUBTITLE_ENCODING environment variable or the
	  current locale's character set if it is non-UTF8.
	  Fixes #337076.

2382
2383
2384
2385
2386
2387
2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Set image type from APIC frame as "image-type" field
	  of GST_TAG_IMAGE buffer caps (#344605).

2388
2389
2390
2391
2392
2393
2394
2395
2396
2397
2398
2399
2400
2401
2402
2403
2404
2405
2406
2407
2408
2409
2410
2411
2412
2413
2006-06-20  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/Makefile.am:
	* ext/flac/gstflacdec.c: (gst_flac_dec_init),
	(gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize),
	(gst_flac_dec_metadata_callback),
	(gst_flac_dec_metadata_callback_seekable),
	(gst_flac_dec_metadata_callback_stream),
	(gst_flac_dec_error_callback),
	(gst_flac_dec_error_callback_seekable),
	(gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable),
	(gst_flac_dec_read_stream), (gst_flac_dec_write),
	(gst_flac_dec_write_seekable), (gst_flac_dec_write_stream),
	(gst_flac_dec_loop), (gst_flac_dec_sink_event),
	(gst_flac_dec_chain), (gst_flac_dec_convert_sink),
	(gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query),
	(gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
	(gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate),
	(gst_flac_dec_sink_activate_push),
	(gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
	* ext/flac/gstflacdec.h:
	  Support chain-based operation, should make flac-over-DAAP
	  work (#340492).

2414
2415
2416
2417
2418
2006-06-20  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	Doc updates, merge some unused symbols.

2419
2420
2421
2422
2423
2424
2425
2426
2427
2428
2006-06-20  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	Added documentation for the rtsp plugin. Fixes #345393.

2429
2430
2431
2432
2433
2434
2006-06-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
	(rtsp_connection_close), (rtsp_connection_free):
	Use better G_OS_* macros. Fixes #345301 some more.

2435
2436
2437
2438
2439
2440
2441
2442
2443
2444
2445
2446
2447
2448
2449
2450
2451
2452
2453
2454
2455
2456
2457
2458
2459
2460
2461
2462
2463
2464
2465
2466
2467
2006-06-20  Wim Taymans  <wim@fluendo.com>

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

	* sys/sunaudio/Makefile.am:
	* sys/sunaudio/gstsunaudio.c: (plugin_init):
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
	(gst_sunaudiomixer_ctrl_list_tracks),
	(gst_sunaudiomixer_ctrl_get_volume),
	(gst_sunaudiomixer_ctrl_set_volume),
	(gst_sunaudiomixer_ctrl_set_mute),
	(gst_sunaudiomixer_ctrl_set_record):
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	(gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
	(gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
	(gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
	(gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
	(gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
	(gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
	(gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
	(gst_sunaudiosrc_reset):
	* sys/sunaudio/gstsunaudiosrc.h:
	Add a SunAudio source plugin.
	Support stereo and right/left channel gain in the mixer plugin.
	Support the RECORD flag so that you can switch between line-input and
	microphone in gnome-volume-control.
	Code cleanups like using an enumerator for track number instead of an 
	integer. Fixes #344923.